update for 1.1.7

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
avcodec/parser: reset indexes on realloc failure
2025-12-06 06:49:59 +01:00 · 2013-10-11 15:39:23 +02:00 · 2013-10-11 15:36:12 +02:00 · 2013-10-11 15:36:12 +02:00 · 2013-10-11 15:28:11 +02:00 · 2013-10-11 14:49:50 +02:00
2696 changed files with 96625 additions and 215567 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -6,8 +6,6 @@
 *.dylib
 *.exe
 *.exp
-*.gcda
-*.gcno
 *.h.c
 *.ilk
 *.lib
@@ -26,13 +24,11 @@
 /ffprobe
 /ffserver
 /config.*
-/coverage.info
 /version.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
 /doc/*.pod
-/doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/examples/decoding_encoding
@@ -41,13 +37,11 @@
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
-/doc/examples/pc-uninstalled
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
 /doc/fate.txt
 /doc/doxy/html/
 /doc/print_options
-/lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
@@ -57,23 +51,18 @@
 /tests/data/
 /tests/rotozoom
 /tests/tiny_psnr
-/tests/tiny_ssim
 /tests/videogen
 /tests/vsynth1/
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
-/tools/crypto_bench
 /tools/cws2fws
 /tools/fourcc2pixfmt
 /tools/ffescape
 /tools/ffeval
-/tools/ffhash
 /tools/graph2dot
 /tools/ismindex
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
 /tools/trasher
-/tools/seek_print
-/tools/zmqsend
--- a/59
+++ b/59
@@ -1,6 +1,55 @@
-See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
-get the names of people who have contributed to FFmpeg.
+This file contains the names of some of the people who have contributed to
+FFmpeg. The names are sorted alphabetically by last name.  As this file is
+currently quite outdated and git serves as a much better tool for determining
+authorship, it remains here for historical reasons only.

-To check the log, you can type the command "git log" in the FFmpeg
-source directory, or browse the online repository at
-http://source.ffmpeg.org.
+Dénes Balatoni
+Michel Bardiaux
+Fabrice Bellard
+Patrice Bensoussan
+Alex Beregszaszi
+BERO
+Thilo Borgmann
+Mario Brito
+Ronald Bultje
+Alex Converse
+Maarten Daniels
+Reimar Doeffinger
+Tim Ferguson
+Brian Foley
+Arpad Gereoffy
+Philip Gladstone
+Vladimir Gneushev
+Roine Gustafsson
+David Hammerton
+Wolfgang Hesseler
+Marc Hoffman
+Falk Hueffner
+Aurélien Jacobs
+Steven Johnson
+Zdenek Kabelac
+Robin Kay
+Todd Kirby
+Nick Kurshev
+Benjamin Larsson
+Loïc Le Loarer
+Daniel Maas
+Mike Melanson
+Loren Merritt
+Jeff Muizelaar
+Michael Niedermayer
+François Revol
+Peter Ross
+Måns Rullgård
+Stefano Sabatini
+Roman Shaposhnik
+Oded Shimon
+Dieter Shirley
+Konstantin Shishkov
+Juan J. Sierralta
+Ewald Snel
+Sascha Sommer
+Leon van Stuivenberg
+Roberto Togni
+Lionel Ulmer
+Reynaldo Verdejo
--- a/1198
+++ b/1198
--- a/30
+++ b/30
@@ -33,28 +33,16 @@ Specifically, the GPL parts of FFmpeg are
    - vf_geq.c
    - vf_histeq.c
    - vf_hqdn3d.c
+    - vf_hue.c
    - vf_kerndeint.c
-    - vf_mcdeint.c
    - vf_mp.c
-    - vf_owdenoise.c
-    - vf_perspective.c
-    - vf_phase.c
    - vf_pp.c
-    - vf_pullup.c
-    - vf_sab.c
    - vf_smartblur.c
-    - vf_spp.c
-    - vf_stereo3d.c
    - vf_super2xsai.c
    - vf_tinterlace.c
    - vf_yadif.c
    - vsrc_mptestsrc.c

-Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
-the configure parameter --enable-version3 will activate this licensing option
-for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
-COPYING.GPLv3 to learn the exact legal terms that apply in this case.
-
 There are a handful of files under other licensing terms, namely:

 * The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
@@ -64,6 +52,11 @@ There are a handful of files under other licensing terms, namely:
  You must also indicate any changes including additions and deletions to
  those three files in the documentation.

+Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
+the configure parameter --enable-version3 will activate this licensing option
+for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
+COPYING.GPLv3 to learn the exact legal terms that apply in this case.
+

 external libraries
 ==================
@@ -74,15 +67,8 @@ affect the licensing of binaries resulting from the combination.
 compatible libraries
 --------------------

-The following libraries are under GPL:
-    - frei0r
-    - libcdio
-    - libutvideo
-    - libvidstab
-    - libx264
-    - libxavs
-    - libxvid
-When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
+The libcdio, libx264, libxavs and libxvid libraries are under GPL. When
+combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
 passing --enable-gpl to configure.

 The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
--- a/116
+++ b/116
@@ -7,8 +7,8 @@ FFmpeg code.
 Please try to keep entries where you are the maintainer up to date!

 Names in () mean that the maintainer currently has no time to maintain the code.
-A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
-patches and related discussions.
+A CC after the name means that the maintainer prefers to be CC-ed on patches
+and related discussions.


 Project Leader
@@ -43,24 +43,16 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================

-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
+documentation                           Mike Melanson
+website                                 Robert Swain, Lou Logan
 build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
+mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer


-Communication
-=============
-
-website                                 Robert Swain, Lou Logan
-mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
-Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Lou Logan
-Launchpad                               Timothy Gu
-
-
 libavutil
 =========

@@ -70,23 +62,11 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer

 Other:
-  bprint                                Nicolas George
-  bswap.h
-  des                                   Reimar Doeffinger
-  eval.c, eval.h                        Michael Niedermayer
-  float_dsp                             Loren Merritt
-  hash                                  Reimar Doeffinger
  intfloat*                             Michael Niedermayer
-  integer.c, integer.h                  Michael Niedermayer
-  lzo                                   Reimar Doeffinger
-  mathematics.c, mathematics.h          Michael Niedermayer
-  mem.c, mem.h                          Michael Niedermayer
-  opencl.c, opencl.h                    Wei Gao
-  opt.c, opt.h                          Michael Niedermayer
  rational.c, rational.h                Michael Niedermayer
-  rc4                                   Reimar Doeffinger
-  ripemd.c, ripemd.h                    James Almer
-  timecode                              Clément Bœsch
+  mathematics.c, mathematics.h          Michael Niedermayer
+  integer.c, integer.h                  Michael Niedermayer
+  bswap.h


 libavcodec
@@ -97,6 +77,10 @@ Generic Parts:
    avcodec.h                           Michael Niedermayer
  utility code:
    utils.c                             Michael Niedermayer
+    mem.c                               Michael Niedermayer
+    opt.c, opt.h                        Michael Niedermayer
+  arithmetic expression evaluator:
+    eval.c                              Michael Niedermayer
  audio and video frame extraction:
    parser.c                            Michael Niedermayer
  bitstream reading:
@@ -127,8 +111,6 @@ Generic Parts:
    libpostproc/*                       Michael Niedermayer
  table generation:
    tableprint.c, tableprint.h          Reimar Doeffinger
-  fixed point FFT:
-    fft*                                Zeljko Lukac

 Codecs:
  4xm.c                                 Michael Niedermayer
@@ -147,8 +129,8 @@ Codecs:
  binkaudio.c                           Peter Ross
  bmp.c                                 Mans Rullgard, Kostya Shishkov
  cavs*                                 Stefan Gehrer
-  cdxl.c                                Paul B Mahol
  celp_filters.*                        Vitor Sessak
+  cdxl.c                                Paul B Mahol
  cinepak.c                             Roberto Togni
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
@@ -159,10 +141,9 @@ Codecs:
  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
-  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
+  dv.c                                  Roman Shaposhnik
  eacmv*, eaidct*, eat*                 Peter Ross
-  exif.c, exif.h                        Thilo Borgmann
  ffv1.c                                Michael Niedermayer
  ffwavesynth.c                         Nicolas George
  flac*                                 Justin Ruggles
@@ -171,9 +152,9 @@ Codecs:
  g722.c                                Martin Storsjo
  g726.c                                Roman Shaposhnik
  gifdec.c                              Baptiste Coudurier
+  h264*                                 Loren Merritt, Michael Niedermayer
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
-  h264*                                 Loren Merritt, Michael Niedermayer
  huffyuv.c                             Michael Niedermayer
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
@@ -182,14 +163,13 @@ Codecs:
  interplayvideo.c                      Mike Melanson
  ivi*                                  Kostya Shishkov
  jacosub*                              Clément Bœsch
-  jpeg2000*                             Nicolas Bertrand
  jpeg_ls.c                             Kostya Shishkov
  jvdec.c                               Peter Ross
  kmvc.c                                Kostya Shishkov
  lcl*.c                                Roberto Togni, Reimar Doeffinger
  libcelt_dec.c                         Nicolas George
-  libdirac*                             David Conrad
  libgsm.c                              Michel Bardiaux
+  libdirac*                             David Conrad
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
@@ -197,8 +177,8 @@ Codecs:
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
-  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
  libxavs.c                             Stefan Gehrer
+  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
@@ -209,7 +189,6 @@ Codecs:
  mpc*                                  Kostya Shishkov
  mpeg12.c, mpeg12data.h                Michael Niedermayer
  mpegvideo.c, mpegvideo.h              Michael Niedermayer
-  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
  msvideo1.c                            Mike Melanson
@@ -235,7 +214,6 @@ Codecs:
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
-  smvjpegdec.c                          Ash Hughes
  snow.c                                Michael Niedermayer, Loren Merritt
  sonic.c                               Alex Beregszaszi
  srt*                                  Aurelien Jacobs
@@ -249,7 +227,6 @@ Codecs:
  truespeech.c                          Kostya Shishkov
  tscc.c                                Kostya Shishkov
  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
-  ttaenc.c                              Paul B Mahol
  txd.c                                 Ivo van Poorten
  ulti*                                 Kostya Shishkov
  v410*.c                               Derek Buitenhuis
@@ -260,8 +237,8 @@ Codecs:
  vda_h264_dec.c                        Xidorn Quan
  vima.c                                Paul B Mahol
  vmnc.c                                Kostya Shishkov
-  vorbis_dec.c                          Denes Balatoni, David Conrad
  vorbis_enc.c                          Oded Shimon
+  vorbis_dec.c                          Denes Balatoni, David Conrad
  vp3*                                  Mike Melanson
  vp5                                   Aurelien Jacobs
  vp6                                   Aurelien Jacobs
@@ -274,7 +251,6 @@ Codecs:
  wnv1.c                                Kostya Shishkov
  xan.c                                 Mike Melanson
  xbm*                                  Paul B Mahol
-  xface                                 Stefano Sabatini
  xl.c                                  Kostya Shishkov
  xvmc.c                                Ivan Kalvachev
  xwd*                                  Paul B Mahol
@@ -296,15 +272,11 @@ libavdevice
    libavdevice/avdevice.h


-  dshow.c                               Roger Pack
-  fbdev_enc.c                           Lukasz Marek
  iec61883.c                            Georg Lippitsch
-  lavfi                                 Stefano Sabatini
  libdc1394.c                           Roman Shaposhnik
-  pulse_audio_enc.c                     Lukasz Marek
-  sdl                                   Stefano Sabatini
  v4l2.c                                Luca Abeni
  vfwcap.c                              Ramiro Polla
+  dshow.c                               Roger Pack

 libavfilter
 ===========
@@ -313,33 +285,10 @@ Generic parts:
  graphdump.c                           Nicolas George

 Filters:
-  af_adelay.c                           Paul B Mahol
-  af_aecho.c                            Paul B Mahol
-  af_afade.c                            Paul B Mahol
  af_amerge.c                           Nicolas George
-  af_aphaser.c                          Paul B Mahol
-  af_aresample.c                        Michael Niedermayer
-  af_astats.c                           Paul B Mahol
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
-  af_biquads.c                          Paul B Mahol
-  af_compand.c                          Paul B Mahol
-  af_ladspa.c                           Paul B Mahol
  af_pan.c                              Nicolas George
-  avf_avectorscope.c                    Paul B Mahol
-  vf_blend.c                            Paul B Mahol
-  vf_colorbalance.c                     Paul B Mahol
-  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
-  vf_drawbox.c/drawgrid                 Andrey Utkin
-  vf_extractplanes.c                    Paul B Mahol
-  vf_histogram.c                        Paul B Mahol
-  vf_il.c                               Paul B Mahol
-  vf_mergeplanes.c                      Paul B Mahol
-  vf_psnr.c                             Paul B Mahol
-  vf_scale.c                            Michael Niedermayer
-  vf_separatefields.c                   Paul B Mahol
-  vf_stereo3d.c                         Paul B Mahol
-  vf_telecine.c                         Paul B Mahol
  vf_yadif.c                            Michael Niedermayer

 Sources:
@@ -359,14 +308,12 @@ Muxers/Demuxers:
  4xm.c                                 Mike Melanson
  adtsenc.c                             Robert Swain
  afc.c                                 Paul B Mahol
-  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
-  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
+  aiff.c                                Baptiste Coudurier
  ape.c                                 Kostya Shishkov
  ass*                                  Aurelien Jacobs
  astdec.c                              Paul B Mahol
  astenc.c                              James Almer
  avi*                                  Michael Niedermayer
-  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
  avr.c                                 Paul B Mahol
  bink.c                                Peter Ross
  brstm.c                               Paul B Mahol
@@ -388,8 +335,8 @@ Muxers/Demuxers:
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
-  img2*.c                               Michael Niedermayer
  ipmovie.c                             Mike Melanson
+  img2*.c                               Michael Niedermayer
  ircam*                                Paul B Mahol
  iss.c                                 Stefan Gehrer
  jacosub*                              Clément Bœsch
@@ -403,11 +350,11 @@ Muxers/Demuxers:
  matroskadec.c                         Aurelien Jacobs
  matroskaenc.c                         David Conrad
  metadata*                             Aurelien Jacobs
-  mgsts.c                               Paul B Mahol
  microdvd*                             Aurelien Jacobs
+  mgsts.c                               Paul B Mahol
  mm.c                                  Peter Ross
  mov.c                                 Michael Niedermayer, Baptiste Coudurier
-  movenc.c                              Baptiste Coudurier, Matthieu Bouron
+  movenc.c                              Michael Niedermayer, Baptiste Coudurier
  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
  mpegenc.c                             Michael Niedermayer
@@ -452,7 +399,6 @@ Muxers/Demuxers:
  voc.c                                 Aurelien Jacobs
  wav.c                                 Michael Niedermayer
  wc3movie.c                            Mike Melanson
-  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
  wv.c                                  Kostya Shishkov
@@ -460,9 +406,7 @@ Muxers/Demuxers:

 Protocols:
  bluray.c                              Petri Hintukainen
-  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
-  libssh.c                              Lukasz Marek
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni

@@ -487,7 +431,7 @@ Operating systems / CPU architectures
 Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
-MIPS                                    Mans Rullgard, Nedeljko Babic
+MIPS                                    Mans Rullgard
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
@@ -501,8 +445,9 @@ x86                                     Michael Niedermayer
 Releases
 ========

-2.1                                     Michael Niedermayer
-2.0                                     Michael Niedermayer
+1.2                                     Michael Niedermayer
+1.1                                     Michael Niedermayer
+1.0                                     Michael Niedermayer

 If you want to maintain an older release, please contact us

@@ -510,10 +455,8 @@ If you want to maintain an older release, please contact us
 GnuPG Fingerprints of maintainers and contributors
 ==================================================

-Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
-Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
 Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
@@ -521,10 +464,8 @@ Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
 Bœsch Clément                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
-FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
-Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Justin Ruggles                3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
@@ -541,4 +482,3 @@ Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
-Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
--- a/18
+++ b/18
@@ -20,7 +20,7 @@ INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))

 OBJS        = cmdutils.o $(EXEOBJS)
 OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
@@ -28,6 +28,7 @@ TOOLS-$(CONFIG_ZLIB) += cws2fws
 BASENAMES   = ffmpeg ffplay ffprobe ffserver
 ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+ALLMANPAGES = $(BASENAMES:%=%.1)

 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
@@ -43,7 +44,7 @@ FFLIBS := avutil
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
 EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README

-SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
+SKIPHEADERS = cmdutils_common_opts.h

 include $(SRC_PATH)/common.mak

@@ -153,16 +154,25 @@ clean::
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
+	$(RM) coverage.info
 	$(RM) -r coverage-html
-	$(RM) -rf coverage.info lcov

 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h
+	$(RM) config.* .version version.h libavutil/avconfig.h libavcodec/codec_names.h

 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)

+# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
+coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
+	$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
+
+coverage-html: coverage.info
+	$(Q)mkdir -p $@
+	$(Q)genhtml -o $@ $<
+	$(Q)touch $@
+
 check: all alltools examples testprogs fate

 include $(SRC_PATH)/doc/Makefile
--- a/2
+++ b/2
@@ -1 +1 @@
-2.1.1
+1.1.7
--- a/2
+++ b/2
@@ -1 +1 @@
-2.1.1
+1.1.7
--- a/cmdutils.c
+++ b/cmdutils.c
@@ -36,7 +36,9 @@
 #include "libavresample/avresample.h"
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
+#if CONFIG_POSTPROC
 #include "libpostproc/postprocess.h"
+#endif
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
@@ -47,7 +49,6 @@
 #include "libavutil/eval.h"
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
-#include "libavutil/cpu.h"
 #include "cmdutils.h"
 #include "version.h"
 #if CONFIG_NETWORK
@@ -57,16 +58,12 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
-#if CONFIG_OPENCL
-#include "libavutil/opencl.h"
-#endif
-

 static int init_report(const char *env);

 struct SwsContext *sws_opts;
-AVDictionary *swr_opts;
-AVDictionary *format_opts, *codec_opts, *resample_opts;
+SwrContext *swr_opts;
+AVDictionary *format_opts, *codec_opts;

 const int this_year = 2013;

@@ -78,6 +75,9 @@ void init_opts(void)
    if(CONFIG_SWSCALE)
        sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
                              NULL, NULL, NULL);
+
+    if(CONFIG_SWRESAMPLE)
+        swr_opts = swr_alloc();
 }

 void uninit_opts(void)
@@ -87,10 +87,11 @@ void uninit_opts(void)
    sws_opts = NULL;
 #endif

-    av_dict_free(&swr_opts);
+    if(CONFIG_SWRESAMPLE)
+        swr_free(&swr_opts);
+
    av_dict_free(&format_opts);
    av_dict_free(&codec_opts);
-    av_dict_free(&resample_opts);
 }

 void log_callback_help(void *ptr, int level, const char *fmt, va_list vl)
@@ -112,21 +113,6 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
    fflush(report_file);
 }

-static void (*program_exit)(int ret);
-
-void register_exit(void (*cb)(int ret))
-{
-    program_exit = cb;
-}
-
-void exit_program(int ret)
-{
-    if (program_exit)
-        program_exit(ret);
-
-    exit(ret);
-}
-
 double parse_number_or_die(const char *context, const char *numstr, int type,
                           double min, double max)
 {
@@ -144,7 +130,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
    else
        return d;
    av_log(NULL, AV_LOG_FATAL, error, context, numstr, min, max);
-    exit_program(1);
+    exit(1);
    return 0;
 }

@@ -155,7 +141,7 @@ int64_t parse_time_or_die(const char *context, const char *timestr,
    if (av_parse_time(&us, timestr, is_duration) < 0) {
        av_log(NULL, AV_LOG_FATAL, "Invalid %s specification for %s: %s\n",
               is_duration ? "duration" : "date", context, timestr);
-        exit_program(1);
+        exit(1);
    }
    return us;
 }
@@ -214,10 +200,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
    return po;
 }

-/* _WIN32 means using the windows libc - cygwin doesn't define that
- * by default. HAVE_COMMANDLINETOARGVW is true on cygwin, while
- * it doesn't provide the actual command line via GetCommandLineW(). */
-#if HAVE_COMMANDLINETOARGVW && defined(_WIN32)
+#if HAVE_COMMANDLINETOARGVW
 #include <windows.h>
 #include <shellapi.h>
 /* Will be leaked on exit */
@@ -301,7 +284,7 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
    if (po->flags & OPT_STRING) {
        char *str;
        str = av_strdup(arg);
-        av_freep(dst);
+//         av_freep(dst);
        *(char **)dst = str;
    } else if (po->flags & OPT_BOOL || po->flags & OPT_INT) {
        *(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
@@ -317,13 +300,12 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
        int ret = po->u.func_arg(optctx, opt, arg);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR,
-                   "Failed to set value '%s' for option '%s': %s\n",
-                   arg, opt, av_err2str(ret));
+                   "Failed to set value '%s' for option '%s'\n", arg, opt);
            return ret;
        }
    }
    if (po->flags & OPT_EXIT)
-        exit_program(0);
+        exit(0);

    return 0;
 }
@@ -383,7 +365,7 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
            opt++;

            if ((ret = parse_option(optctx, opt, argv[optindex], options)) < 0)
-                exit_program(1);
+                exit(1);
            optindex += ret;
        } else {
            if (parse_arg_function)
@@ -402,16 +384,6 @@ int parse_optgroup(void *optctx, OptionGroup *g)
    for (i = 0; i < g->nb_opts; i++) {
        Option *o = &g->opts[i];

-        if (g->group_def->flags &&
-            !(g->group_def->flags & o->opt->flags)) {
-            av_log(NULL, AV_LOG_ERROR, "Option %s (%s) cannot be applied to "
-                   "%s %s -- you are trying to apply an input option to an "
-                   "output file or vice versa. Move this option before the "
-                   "file it belongs to.\n", o->key, o->opt->help,
-                   g->group_def->name, g->arg);
-            return AVERROR(EINVAL);
-        }
-
        av_log(NULL, AV_LOG_DEBUG, "Applying option %s (%s) with argument %s.\n",
               o->key, o->opt->help, o->val);

@@ -498,15 +470,6 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
    }
 }

-static const AVOption *opt_find(void *obj, const char *name, const char *unit,
-                            int opt_flags, int search_flags)
-{
-    const AVOption *o = av_opt_find(obj, name, unit, opt_flags, search_flags);
-    if(o && !o->flags)
-        return NULL;
-    return o;
-}
-
 #define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
 int opt_default(void *optctx, const char *opt, const char *arg)
 {
@@ -515,9 +478,6 @@ int opt_default(void *optctx, const char *opt, const char *arg)
    char opt_stripped[128];
    const char *p;
    const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class();
-#if CONFIG_AVRESAMPLE
-    const AVClass *rc = avresample_get_class();
-#endif
    const AVClass *sc, *swr_class;

    if (!strcmp(opt, "debug") || !strcmp(opt, "fdebug"))
@@ -527,23 +487,23 @@ int opt_default(void *optctx, const char *opt, const char *arg)
        p = opt + strlen(opt);
    av_strlcpy(opt_stripped, opt, FFMIN(sizeof(opt_stripped), p - opt + 1));

-    if ((o = opt_find(&cc, opt_stripped, NULL, 0,
+    if ((o = av_opt_find(&cc, opt_stripped, NULL, 0,
                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) ||
        ((opt[0] == 'v' || opt[0] == 'a' || opt[0] == 's') &&
-         (o = opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
+         (o = av_opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
        av_dict_set(&codec_opts, opt, arg, FLAGS);
        consumed = 1;
    }
-    if ((o = opt_find(&fc, opt, NULL, 0,
-                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+    if ((o = av_opt_find(&fc, opt, NULL, 0,
+                              AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        av_dict_set(&format_opts, opt, arg, FLAGS);
-        if (consumed)
-            av_log(NULL, AV_LOG_VERBOSE, "Routing option %s to both codec and muxer layer\n", opt);
+        if(consumed)
+            av_log(NULL, AV_LOG_VERBOSE, "Routing %s to codec and muxer layer\n", opt);
        consumed = 1;
    }
 #if CONFIG_SWSCALE
    sc = sws_get_class();
-    if (!consumed && opt_find(&sc, opt, NULL, 0,
+    if (!consumed && av_opt_find(&sc, opt, NULL, 0,
                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
        // XXX we only support sws_flags, not arbitrary sws options
        int ret = av_opt_set(sws_opts, opt, arg, 0);
@@ -556,23 +516,13 @@ int opt_default(void *optctx, const char *opt, const char *arg)
 #endif
 #if CONFIG_SWRESAMPLE
    swr_class = swr_get_class();
-    if (!consumed && (o=opt_find(&swr_class, opt, NULL, 0,
-                                    AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
-        struct SwrContext *swr = swr_alloc();
-        int ret = av_opt_set(swr, opt, arg, 0);
-        swr_free(&swr);
+    if (!consumed && av_opt_find(&swr_class, opt, NULL, 0,
+                               AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
+        int ret = av_opt_set(swr_opts, opt, arg, 0);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
            return ret;
        }
-        av_dict_set(&swr_opts, opt, arg, FLAGS);
-        consumed = 1;
-    }
-#endif
-#if CONFIG_AVRESAMPLE
-    if ((o=opt_find(&rc, opt, NULL, 0,
-                       AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
-        av_dict_set(&resample_opts, opt, arg, FLAGS);
        consumed = 1;
    }
 #endif
@@ -625,11 +575,9 @@ static void finish_group(OptionParseContext *octx, int group_idx,
    g->swr_opts    = swr_opts;
    g->codec_opts  = codec_opts;
    g->format_opts = format_opts;
-    g->resample_opts = resample_opts;

    codec_opts  = NULL;
    format_opts = NULL;
-    resample_opts = NULL;
 #if CONFIG_SWSCALE
    sws_opts    = NULL;
 #endif
@@ -665,7 +613,7 @@ static void init_parse_context(OptionParseContext *octx,
    octx->nb_groups = nb_groups;
    octx->groups    = av_mallocz(sizeof(*octx->groups) * octx->nb_groups);
    if (!octx->groups)
-        exit_program(1);
+        exit(1);

    for (i = 0; i < octx->nb_groups; i++)
        octx->groups[i].group_def = &groups[i];
@@ -687,11 +635,11 @@ void uninit_parse_context(OptionParseContext *octx)
            av_freep(&l->groups[j].opts);
            av_dict_free(&l->groups[j].codec_opts);
            av_dict_free(&l->groups[j].format_opts);
-            av_dict_free(&l->groups[j].resample_opts);
 #if CONFIG_SWSCALE
            sws_freeContext(l->groups[j].sws_opts);
 #endif
-            av_dict_free(&l->groups[j].swr_opts);
+            if(CONFIG_SWRESAMPLE)
+                swr_free(&l->groups[j].swr_opts);
        }
        av_freep(&l->groups);
    }
@@ -708,7 +656,6 @@ int split_commandline(OptionParseContext *octx, int argc, char *argv[],
                      const OptionGroupDef *groups, int nb_groups)
 {
    int optindex = 1;
-    int dashdash = -2;

    /* perform system-dependent conversions for arguments list */
    prepare_app_arguments(&argc, &argv);
@@ -723,12 +670,8 @@ int split_commandline(OptionParseContext *octx, int argc, char *argv[],

        av_log(NULL, AV_LOG_DEBUG, "Reading option '%s' ...", opt);

-        if (opt[0] == '-' && opt[1] == '-' && !opt[2]) {
-            dashdash = optindex;
-            continue;
-        }
        /* unnamed group separators, e.g. output filename */
-        if (opt[0] != '-' || !opt[1] || dashdash+1 == optindex) {
+        if (opt[0] != '-' || !opt[1]) {
            finish_group(octx, 0, opt);
            av_log(NULL, AV_LOG_DEBUG, " matched as %s.\n", groups[0].name);
            continue;
@@ -800,7 +743,7 @@ do {                                                                           \
        return AVERROR_OPTION_NOT_FOUND;
    }

-    if (octx->cur_group.nb_opts || codec_opts || format_opts || resample_opts)
+    if (octx->cur_group.nb_opts || codec_opts || format_opts)
        av_log(NULL, AV_LOG_WARNING, "Trailing options were found on the "
               "commandline.\n");

@@ -809,18 +752,6 @@ do {                                                                           \
    return 0;
 }

-int opt_cpuflags(void *optctx, const char *opt, const char *arg)
-{
-    int ret;
-    unsigned flags = av_get_cpu_flags();
-
-    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
-        return ret;
-
-    av_force_cpu_flags(flags);
-    return 0;
-}
-
 int opt_loglevel(void *optctx, const char *opt, const char *arg)
 {
    const struct { const char *name; int level; } log_levels[] = {
@@ -837,13 +768,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
    int level;
    int i;

-    tail = strstr(arg, "repeat");
-    av_log_set_flags(tail ? 0 : AV_LOG_SKIP_REPEATED);
-    if (tail == arg)
-        arg += 6 + (arg[6]=='+');
-    if(tail && !*arg)
-        return 0;
-
    for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) {
        if (!strcmp(log_levels[i].name, arg)) {
            av_log_set_level(log_levels[i].level);
@@ -857,7 +781,7 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
               "Possible levels are numbers or:\n", arg);
        for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++)
            av_log(NULL, AV_LOG_FATAL, "\"%s\"\n", log_levels[i].name);
-        exit_program(1);
+        exit(1);
    }
    av_log_set_level(level);
    return 0;
@@ -950,6 +874,7 @@ static int init_report(const char *env)
           tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday,
           tm->tm_hour, tm->tm_min, tm->tm_sec,
           filename.str);
+    av_log_set_level(FFMAX(av_log_get_level(), AV_LOG_VERBOSE));
    av_bprint_finalize(&filename, NULL);
    return 0;
 }
@@ -967,12 +892,24 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg)
    max = strtol(arg, &tail, 10);
    if (*tail) {
        av_log(NULL, AV_LOG_FATAL, "Invalid max_alloc \"%s\".\n", arg);
-        exit_program(1);
+        exit(1);
    }
    av_max_alloc(max);
    return 0;
 }

+int opt_cpuflags(void *optctx, const char *opt, const char *arg)
+{
+    int ret;
+    unsigned flags = av_get_cpu_flags();
+
+    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
+        return ret;
+
+    av_force_cpu_flags(flags);
+    return 0;
+}
+
 int opt_timelimit(void *optctx, const char *opt, const char *arg)
 {
 #if HAVE_SETRLIMIT
@@ -986,26 +923,6 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg)
    return 0;
 }

-#if CONFIG_OPENCL
-int opt_opencl(void *optctx, const char *opt, const char *arg)
-{
-    char *key, *value;
-    const char *opts = arg;
-    int ret = 0;
-    while (*opts) {
-        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
-        if (ret < 0)
-            return ret;
-        ret = av_opencl_set_option(key, value);
-        if (ret < 0)
-            return ret;
-        if (*opts)
-            opts++;
-    }
-    return ret;
-}
-#endif
-
 void print_error(const char *filename, int err)
 {
    char errbuf[128];
@@ -1058,10 +975,12 @@ static void print_all_libs_info(int flags, int level)
    PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
    PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
    PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
-    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
+//    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
    PRINT_LIB_INFO(swscale,  SWSCALE,  flags, level);
    PRINT_LIB_INFO(swresample,SWRESAMPLE,  flags, level);
+#if CONFIG_POSTPROC
    PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
+#endif
 }

 static void print_program_info(int flags, int level)
@@ -1241,8 +1160,7 @@ static void print_codec(const AVCodec *c)
    printf("%s %s [%s]:\n", encoder ? "Encoder" : "Decoder", c->name,
           c->long_name ? c->long_name : "");

-    if (c->type == AVMEDIA_TYPE_VIDEO ||
-        c->type == AVMEDIA_TYPE_AUDIO) {
+    if (c->type == AVMEDIA_TYPE_VIDEO) {
        printf("    Threading capabilities: ");
        switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
                                   CODEC_CAP_SLICE_THREADS)) {
@@ -1323,7 +1241,7 @@ static unsigned get_codecs_sorted(const AVCodecDescriptor ***rcodecs)
        nb_codecs++;
    if (!(codecs = av_calloc(nb_codecs, sizeof(*codecs)))) {
        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
-        exit_program(1);
+        exit(1);
    }
    desc = NULL;
    while ((desc = avcodec_descriptor_next(desc)))
@@ -1476,60 +1394,36 @@ int show_protocols(void *optctx, const char *opt, const char *arg)

 int show_filters(void *optctx, const char *opt, const char *arg)
 {
-    const AVFilter av_unused(*filter) = NULL;
+    AVFilter av_unused(**filter) = NULL;
    char descr[64], *descr_cur;
    int i, j;
    const AVFilterPad *pad;

-    printf("Filters:\n"
-           "  T.. = Timeline support\n"
-           "  .S. = Slice threading\n"
-           "  ..C = Commmand support\n"
-           "  A = Audio input/output\n"
-           "  V = Video input/output\n"
-           "  N = Dynamic number and/or type of input/output\n"
-           "  | = Source or sink filter\n");
+    printf("Filters:\n");
 #if CONFIG_AVFILTER
-    while ((filter = avfilter_next(filter))) {
+    while ((filter = av_filter_next(filter)) && *filter) {
        descr_cur = descr;
        for (i = 0; i < 2; i++) {
            if (i) {
                *(descr_cur++) = '-';
                *(descr_cur++) = '>';
            }
-            pad = i ? filter->outputs : filter->inputs;
+            pad = i ? (*filter)->outputs : (*filter)->inputs;
            for (j = 0; pad && pad[j].name; j++) {
                if (descr_cur >= descr + sizeof(descr) - 4)
                    break;
                *(descr_cur++) = get_media_type_char(pad[j].type);
            }
            if (!j)
-                *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
-                                  ( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|';
+                *(descr_cur++) = '|';
        }
        *descr_cur = 0;
-        printf(" %c%c%c %-16s %-10s %s\n",
-               filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.',
-               filter->flags & AVFILTER_FLAG_SLICE_THREADS    ? 'S' : '.',
-               filter->process_command                        ? 'C' : '.',
-               filter->name, descr, filter->description);
+        printf("%-16s %-10s %s\n", (*filter)->name, descr, (*filter)->description);
    }
 #endif
    return 0;
 }

-void show_colors(void *optctx, const char *opt, const char *arg)
-{
-    const char *name;
-    const uint8_t *rgb;
-    int i;
-
-    printf("%-32s #RRGGBB\n", "name");
-
-    for (i = 0; name = av_get_known_color_name(i, &rgb); i++)
-        printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]);
-}
-
 int show_pix_fmts(void *optctx, const char *opt, const char *arg)
 {
    const AVPixFmtDescriptor *pix_desc = NULL;
@@ -1551,11 +1445,11 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg)
    while ((pix_desc = av_pix_fmt_desc_next(pix_desc))) {
        enum AVPixelFormat pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
        printf("%c%c%c%c%c %-16s       %d            %2d\n",
-               sws_isSupportedInput (pix_fmt)              ? 'I' : '.',
-               sws_isSupportedOutput(pix_fmt)              ? 'O' : '.',
-               pix_desc->flags & AV_PIX_FMT_FLAG_HWACCEL   ? 'H' : '.',
-               pix_desc->flags & AV_PIX_FMT_FLAG_PAL       ? 'P' : '.',
-               pix_desc->flags & AV_PIX_FMT_FLAG_BITSTREAM ? 'B' : '.',
+               sws_isSupportedInput (pix_fmt)      ? 'I' : '.',
+               sws_isSupportedOutput(pix_fmt)      ? 'O' : '.',
+               pix_desc->flags & PIX_FMT_HWACCEL   ? 'H' : '.',
+               pix_desc->flags & PIX_FMT_PAL       ? 'P' : '.',
+               pix_desc->flags & PIX_FMT_BITSTREAM ? 'B' : '.',
               pix_desc->name,
               pix_desc->nb_components,
               av_get_bits_per_pixel(pix_desc));
@@ -1687,62 +1581,6 @@ static void show_help_muxer(const char *name)
        show_help_children(fmt->priv_class, AV_OPT_FLAG_ENCODING_PARAM);
 }

-#if CONFIG_AVFILTER
-static void show_help_filter(const char *name)
-{
-#if CONFIG_AVFILTER
-    const AVFilter *f = avfilter_get_by_name(name);
-    int i, count;
-
-    if (!name) {
-        av_log(NULL, AV_LOG_ERROR, "No filter name specified.\n");
-        return;
-    } else if (!f) {
-        av_log(NULL, AV_LOG_ERROR, "Unknown filter '%s'.\n", name);
-        return;
-    }
-
-    printf("Filter %s\n", f->name);
-    if (f->description)
-        printf("  %s\n", f->description);
-
-    if (f->flags & AVFILTER_FLAG_SLICE_THREADS)
-        printf("    slice threading supported\n");
-
-    printf("    Inputs:\n");
-    count = avfilter_pad_count(f->inputs);
-    for (i = 0; i < count; i++) {
-        printf("       #%d: %s (%s)\n", i, avfilter_pad_get_name(f->inputs, i),
-               media_type_string(avfilter_pad_get_type(f->inputs, i)));
-    }
-    if (f->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)
-        printf("        dynamic (depending on the options)\n");
-    else if (!count)
-        printf("        none (source filter)\n");
-
-    printf("    Outputs:\n");
-    count = avfilter_pad_count(f->outputs);
-    for (i = 0; i < count; i++) {
-        printf("       #%d: %s (%s)\n", i, avfilter_pad_get_name(f->outputs, i),
-               media_type_string(avfilter_pad_get_type(f->outputs, i)));
-    }
-    if (f->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS)
-        printf("        dynamic (depending on the options)\n");
-    else if (!count)
-        printf("        none (sink filter)\n");
-
-    if (f->priv_class)
-        show_help_children(f->priv_class, AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_FILTERING_PARAM |
-                                          AV_OPT_FLAG_AUDIO_PARAM);
-    if (f->flags & AVFILTER_FLAG_SUPPORT_TIMELINE)
-        printf("This filter has support for timeline through the 'enable' option.\n");
-#else
-    av_log(NULL, AV_LOG_ERROR, "Build without libavfilter; "
-           "can not to satisfy request\n");
-#endif
-}
-#endif
-
 int show_help(void *optctx, const char *opt, const char *arg)
 {
    char *topic, *par;
@@ -1763,10 +1601,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
        show_help_demuxer(par);
    } else if (!strcmp(topic, "muxer")) {
        show_help_muxer(par);
-#if CONFIG_AVFILTER
-    } else if (!strcmp(topic, "filter")) {
-        show_help_filter(par);
-#endif
    } else {
        show_help_default(topic, par);
    }
@@ -1778,7 +1612,7 @@ int show_help(void *optctx, const char *opt, const char *arg)
 int read_yesno(void)
 {
    int c = getchar();
-    int yesno = (av_toupper(c) == 'Y');
+    int yesno = (toupper(c) == 'Y');

    while (c != '\n' && c != EOF)
        c = getchar();
@@ -1899,8 +1733,10 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
    if (!codec)
        codec            = s->oformat ? avcodec_find_encoder(codec_id)
                                      : avcodec_find_decoder(codec_id);
+    if (!codec)
+        return NULL;

-    switch (st->codec->codec_type) {
+    switch (codec->type) {
    case AVMEDIA_TYPE_VIDEO:
        prefix  = 'v';
        flags  |= AV_OPT_FLAG_VIDEO_PARAM;
@@ -1966,13 +1802,13 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
 {
    if (new_size >= INT_MAX / elem_size) {
        av_log(NULL, AV_LOG_ERROR, "Array too big.\n");
-        exit_program(1);
+        exit(1);
    }
    if (*size < new_size) {
        uint8_t *tmp = av_realloc(array, new_size*elem_size);
        if (!tmp) {
            av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
-            exit_program(1);
+            exit(1);
        }
        memset(tmp + *size*elem_size, 0, (new_size-*size) * elem_size);
        *size = new_size;
@@ -1980,3 +1816,150 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
    }
    return array;
 }
+
+static int alloc_buffer(FrameBuffer **pool, AVCodecContext *s, FrameBuffer **pbuf)
+{
+    const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(s->pix_fmt);
+    FrameBuffer *buf;
+    int i, ret;
+    int pixel_size;
+    int h_chroma_shift, v_chroma_shift;
+    int edge = 32; // XXX should be avcodec_get_edge_width(), but that fails on svq1
+    int w = s->width, h = s->height;
+
+    if (!desc)
+        return AVERROR(EINVAL);
+    pixel_size = desc->comp[0].step_minus1 + 1;
+
+    buf = av_mallocz(sizeof(*buf));
+    if (!buf)
+        return AVERROR(ENOMEM);
+
+    avcodec_align_dimensions(s, &w, &h);
+
+    if (!(s->flags & CODEC_FLAG_EMU_EDGE)) {
+        w += 2*edge;
+        h += 2*edge;
+    }
+
+    if ((ret = av_image_alloc(buf->base, buf->linesize, w, h,
+                              s->pix_fmt, 32)) < 0) {
+        av_freep(&buf);
+        av_log(s, AV_LOG_ERROR, "alloc_buffer: av_image_alloc() failed\n");
+        return ret;
+    }
+    /* XXX this shouldn't be needed, but some tests break without this line
+     * those decoders are buggy and need to be fixed.
+     * the following tests fail:
+     * cdgraphics, ansi
+     */
+    memset(buf->base[0], 128, ret);
+
+    avcodec_get_chroma_sub_sample(s->pix_fmt, &h_chroma_shift, &v_chroma_shift);
+    for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
+        const int h_shift = i==0 ? 0 : h_chroma_shift;
+        const int v_shift = i==0 ? 0 : v_chroma_shift;
+        if ((s->flags & CODEC_FLAG_EMU_EDGE) || !buf->linesize[i] || !buf->base[i])
+            buf->data[i] = buf->base[i];
+        else
+            buf->data[i] = buf->base[i] +
+                           FFALIGN((buf->linesize[i]*edge >> v_shift) +
+                                   (pixel_size*edge >> h_shift), 32);
+    }
+    buf->w       = s->width;
+    buf->h       = s->height;
+    buf->pix_fmt = s->pix_fmt;
+    buf->pool    = pool;
+
+    *pbuf = buf;
+    return 0;
+}
+
+int codec_get_buffer(AVCodecContext *s, AVFrame *frame)
+{
+    FrameBuffer **pool = s->opaque;
+    FrameBuffer *buf;
+    int ret, i;
+
+    if(av_image_check_size(s->width, s->height, 0, s) || s->pix_fmt<0) {
+        av_log(s, AV_LOG_ERROR, "codec_get_buffer: image parameters invalid\n");
+        return -1;
+    }
+
+    if (!*pool && (ret = alloc_buffer(pool, s, pool)) < 0)
+        return ret;
+
+    buf              = *pool;
+    *pool            = buf->next;
+    buf->next        = NULL;
+    if (buf->w != s->width || buf->h != s->height || buf->pix_fmt != s->pix_fmt) {
+        av_freep(&buf->base[0]);
+        av_free(buf);
+        if ((ret = alloc_buffer(pool, s, &buf)) < 0)
+            return ret;
+    }
+    av_assert0(!buf->refcount);
+    buf->refcount++;
+
+    frame->opaque        = buf;
+    frame->type          = FF_BUFFER_TYPE_USER;
+    frame->extended_data = frame->data;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
+        frame->base[i]     = buf->base[i];  // XXX h264.c uses base though it shouldn't
+        frame->data[i]     = buf->data[i];
+        frame->linesize[i] = buf->linesize[i];
+    }
+
+    return 0;
+}
+
+static void unref_buffer(FrameBuffer *buf)
+{
+    FrameBuffer **pool = buf->pool;
+
+    av_assert0(buf->refcount > 0);
+    buf->refcount--;
+    if (!buf->refcount) {
+        FrameBuffer *tmp;
+        for(tmp= *pool; tmp; tmp= tmp->next)
+            av_assert1(tmp != buf);
+
+        buf->next = *pool;
+        *pool = buf;
+    }
+}
+
+void codec_release_buffer(AVCodecContext *s, AVFrame *frame)
+{
+    FrameBuffer *buf = frame->opaque;
+    int i;
+
+    if(frame->type!=FF_BUFFER_TYPE_USER) {
+        avcodec_default_release_buffer(s, frame);
+        return;
+    }
+
+    for (i = 0; i < FF_ARRAY_ELEMS(frame->data); i++)
+        frame->data[i] = NULL;
+
+    unref_buffer(buf);
+}
+
+void filter_release_buffer(AVFilterBuffer *fb)
+{
+    FrameBuffer *buf = fb->priv;
+    av_free(fb);
+    unref_buffer(buf);
+}
+
+void free_buffer_pool(FrameBuffer **pool)
+{
+    FrameBuffer *buf = *pool;
+    while (buf) {
+        *pool = buf->next;
+        av_freep(&buf->base[0]);
+        av_free(buf);
+        buf = *pool;
+    }
+}
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -51,18 +51,8 @@ extern const int this_year;
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
-extern AVDictionary *swr_opts;
-extern AVDictionary *format_opts, *codec_opts, *resample_opts;
-
-/**
- * Register a program-specific cleanup routine.
- */
-void register_exit(void (*cb)(int ret));
-
-/**
- * Wraps exit with a program-specific cleanup routine.
- */
-void exit_program(int ret);
+extern struct SwrContext *swr_opts;
+extern AVDictionary *format_opts, *codec_opts;

 /**
 * Initialize the cmdutils option system, in particular
@@ -81,11 +71,6 @@ void uninit_opts(void);
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);

-/**
- * Override the cpuflags.
- */
-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
-
 /**
 * Fallback for options that are not explicitly handled, these will be
 * parsed through AVOptions.
@@ -101,9 +86,9 @@ int opt_report(const char *opt);

 int opt_max_alloc(void *optctx, const char *opt, const char *arg);

-int opt_codec_debug(void *optctx, const char *opt, const char *arg);
+int opt_cpuflags(void *optctx, const char *opt, const char *arg);

-int opt_opencl(void *optctx, const char *opt, const char *arg);
+int opt_codec_debug(void *optctx, const char *opt, const char *arg);

 /**
 * Limit the execution time.
@@ -138,7 +123,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
 * not zero timestr is interpreted as a duration, otherwise as a
 * date
 *
- * @see av_parse_time()
+ * @see parse_date()
 */
 int64_t parse_time_or_die(const char *context, const char *timestr,
                          int is_duration);
@@ -177,8 +162,6 @@ typedef struct OptionDef {
                                   an int containing element count in the array. */
 #define OPT_TIME  0x10000
 #define OPT_DOUBLE 0x20000
-#define OPT_INPUT  0x40000
-#define OPT_OUTPUT 0x80000
     union {
        void *dst_ptr;
        int (*func_arg)(void *, const char *, const char *);
@@ -259,11 +242,6 @@ typedef struct OptionGroupDef {
     * are terminated by a non-option argument (e.g. ffmpeg output files)
     */
    const char *sep;
-    /**
-     * Option flags that must be set on each option that is
-     * applied to this group
-     */
-    int flags;
 } OptionGroupDef;

 typedef struct OptionGroup {
@@ -275,9 +253,8 @@ typedef struct OptionGroup {

    AVDictionary *codec_opts;
    AVDictionary *format_opts;
-    AVDictionary *resample_opts;
    struct SwsContext *sws_opts;
-    AVDictionary *swr_opts;
+    struct SwrContext *swr_opts;
 } OptionGroup;

 /**
@@ -488,12 +465,6 @@ int show_layouts(void *optctx, const char *opt, const char *arg);
 */
 int show_sample_fmts(void *optctx, const char *opt, const char *arg);

-/**
- * Print a listing containing all the color names and values recognized
- * by the program.
- */
-void show_colors(void *optctx, const char *opt, const char *arg);
-
 /**
 * Return a positive value if a line read from standard input
 * starts with [yY], otherwise return 0.
@@ -507,7 +478,7 @@ int read_yesno(void);
 * @param filename file to read from
 * @param bufptr location where pointer to buffer is returned
 * @param size   location where size of buffer is returned
- * @return >= 0 in case of success, a negative value corresponding to an
+ * @return 0 in case of success, a negative value corresponding to an
 * AVERROR error code in case of failure.
 */
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
@@ -545,11 +516,52 @@ FILE *get_preset_file(char *filename, size_t filename_size,
 */
 void *grow_array(void *array, int elem_size, int *size, int new_size);

-#define media_type_string av_get_media_type_string
-
 #define GROW_ARRAY(array, nb_elems)\
    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)

+typedef struct FrameBuffer {
+    uint8_t *base[4];
+    uint8_t *data[4];
+    int  linesize[4];
+
+    int h, w;
+    enum AVPixelFormat pix_fmt;
+
+    int refcount;
+    struct FrameBuffer **pool;  ///< head of the buffer pool
+    struct FrameBuffer *next;
+} FrameBuffer;
+
+/**
+ * Get a frame from the pool. This is intended to be used as a callback for
+ * AVCodecContext.get_buffer.
+ *
+ * @param s codec context. s->opaque must be a pointer to the head of the
+ *          buffer pool.
+ * @param frame frame->opaque will be set to point to the FrameBuffer
+ *              containing the frame data.
+ */
+int codec_get_buffer(AVCodecContext *s, AVFrame *frame);
+
+/**
+ * A callback to be used for AVCodecContext.release_buffer along with
+ * codec_get_buffer().
+ */
+void codec_release_buffer(AVCodecContext *s, AVFrame *frame);
+
+/**
+ * A callback to be used for AVFilterBuffer.free.
+ * @param fb buffer to free. fb->priv must be a pointer to the FrameBuffer
+ *           containing the buffer data.
+ */
+void filter_release_buffer(AVFilterBuffer *fb);
+
+/**
+ * Free all the buffers in the pool. This must be called after all the
+ * buffers have been released.
+ */
+void free_buffer_pool(FrameBuffer **pool);
+
 #define GET_PIX_FMT_NAME(pix_fmt)\
    const char *name = av_get_pix_fmt_name(pix_fmt);

--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -14,12 +14,8 @@
    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
    { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
-    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
-    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
-    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
+    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
+    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
    { "report"     , 0,        {(void*)opt_report}, "generate a report" },
    { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
-#if CONFIG_OPENCL
-    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
-#endif
+    { "cpuflags"   , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },
--- a/common.mak
+++ b/common.mak
@@ -32,7 +32,7 @@ ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
 YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm

-HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
+HOSTCCFLAGS = $(IFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)

 define COMPILE
@@ -99,10 +99,8 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)

-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))

-SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
@@ -119,12 +117,12 @@ $(HOSTPROGS): %$(HOSTEXESUF): %.o
 	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)

 $(OBJS):     | $(sort $(dir $(OBJS)))
-$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
+$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(TOOLOBJS): | tools

-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HOBJS))

 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -1,31 +0,0 @@
-/*
- * Work around the class() function in AIX math.h clashing with
- * identifiers named "class".
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_COMPAT_AIX_MATH_H
-#define FFMPEG_COMPAT_AIX_MATH_H
-
-#define class class_in_math_h_causes_problems
-
-#include_next <math.h>
-
-#undef class
-
-#endif /* FFMPEG_COMPAT_AIX_MATH_H */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -1,879 +0,0 @@
-// Avisynth C Interface Version 0.20
-// Copyright 2003 Kevin Atkinson
-
-// This program is free software; you can redistribute it and/or modify
-// it under the terms of the GNU General Public License as published by
-// the Free Software Foundation; either version 2 of the License, or
-// (at your option) any later version.
-//
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License for more details.
-//
-// You should have received a copy of the GNU General Public License
-// along with this program; if not, write to the Free Software
-// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-// NOTE: this is a partial update of the Avisynth C interface to recognize
-// new color spaces added in Avisynth 2.60. By no means is this document
-// completely Avisynth 2.60 compliant.
-
-#ifndef __AVISYNTH_C__
-#define __AVISYNTH_C__
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-typedef unsigned char BYTE;
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVISYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 4 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_A=1<<4,
-      AVS_PLANAR_R=1<<5,
-      AVS_PLANAR_G=1<<6,
-      AVS_PLANAR_B=1<<7,
-      AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31,
-
-      AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
-      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
-
-      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
-      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
-      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
-      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
-
-      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
-      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
-
-      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
-      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
-      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
-      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
-
-      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
-      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
-
-      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
-      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  //  AVS_CS_YV12  = 1<<3  Reserved
-  //  AVS_CS_I420  = 1<<4  Reserved
-  AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
-
-  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
-  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
-  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:0 planar
-  AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
-  AVS_CS_IYUV  = AVS_CS_I420,
-  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
-  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
-  AVS_CS_Y8    = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-  };
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  BYTE * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  volatile long sequence_number;
-
-  volatile long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  volatile long refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-  int row_sizeUV, heightUV;
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_sizeUV;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_sizeUV;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->heightUV;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on an AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                             INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                   int cachehints, int frame_range);
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                       AVS_FilterInfo * * fi,
-                                       AVS_Value child, int store_child);
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-  AVS_CPUF_SSE3       = 0x100,   //  PIV+, K8 Venice
-  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
-  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
-  AVS_CPUF_SSE4_1     = 0x400,
-  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
-};
-
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                const char * name, const char * params,
-                                AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-
-#ifdef AVSC_NO_DECLSPEC
-// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
-/*
-  The following functions needs to have been declared, probably from windows.h
-
-  void* malloc(size_t)
-  void free(void*);
-
-  HMODULE LoadLibrary(const char*);
-  void* GetProcAddress(HMODULE, const char*);
-  FreeLibrary(HMODULE);
-*/
-
-
-typedef struct AVS_Library AVS_Library;
-
-#define AVSC_DECLARE_FUNC(name) name##_func name
-
-struct AVS_Library {
-  HMODULE handle;
-
-  AVSC_DECLARE_FUNC(avs_add_function);
-  AVSC_DECLARE_FUNC(avs_at_exit);
-  AVSC_DECLARE_FUNC(avs_bit_blt);
-  AVSC_DECLARE_FUNC(avs_check_version);
-  AVSC_DECLARE_FUNC(avs_clip_get_error);
-  AVSC_DECLARE_FUNC(avs_copy_clip);
-  AVSC_DECLARE_FUNC(avs_copy_value);
-  AVSC_DECLARE_FUNC(avs_copy_video_frame);
-  AVSC_DECLARE_FUNC(avs_create_script_environment);
-  AVSC_DECLARE_FUNC(avs_delete_script_environment);
-  AVSC_DECLARE_FUNC(avs_function_exists);
-  AVSC_DECLARE_FUNC(avs_get_audio);
-  AVSC_DECLARE_FUNC(avs_get_cpu_flags);
-  AVSC_DECLARE_FUNC(avs_get_error);
-  AVSC_DECLARE_FUNC(avs_get_frame);
-  AVSC_DECLARE_FUNC(avs_get_parity);
-  AVSC_DECLARE_FUNC(avs_get_var);
-  AVSC_DECLARE_FUNC(avs_get_version);
-  AVSC_DECLARE_FUNC(avs_get_video_info);
-  AVSC_DECLARE_FUNC(avs_invoke);
-  AVSC_DECLARE_FUNC(avs_make_writable);
-  AVSC_DECLARE_FUNC(avs_new_c_filter);
-  AVSC_DECLARE_FUNC(avs_new_video_frame_a);
-  AVSC_DECLARE_FUNC(avs_release_clip);
-  AVSC_DECLARE_FUNC(avs_release_value);
-  AVSC_DECLARE_FUNC(avs_release_video_frame);
-  AVSC_DECLARE_FUNC(avs_save_string);
-  AVSC_DECLARE_FUNC(avs_set_cache_hints);
-  AVSC_DECLARE_FUNC(avs_set_global_var);
-  AVSC_DECLARE_FUNC(avs_set_memory_max);
-  AVSC_DECLARE_FUNC(avs_set_to_clip);
-  AVSC_DECLARE_FUNC(avs_set_var);
-  AVSC_DECLARE_FUNC(avs_set_working_dir);
-  AVSC_DECLARE_FUNC(avs_sprintf);
-  AVSC_DECLARE_FUNC(avs_subframe);
-  AVSC_DECLARE_FUNC(avs_subframe_planar);
-  AVSC_DECLARE_FUNC(avs_take_clip);
-  AVSC_DECLARE_FUNC(avs_vsprintf);
-};
-
-#undef AVSC_DECLARE_FUNC
-
-
-AVSC_INLINE AVS_Library * avs_load_library() {
-  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (library == NULL)
-    return NULL;
-  library->handle = LoadLibrary("avisynth");
-  if (library->handle == NULL)
-    goto fail;
-
-#define __AVSC_STRINGIFY(x) #x
-#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
-#define AVSC_LOAD_FUNC(name) {\
-  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
-  if (library->name == NULL)\
-    goto fail;\
-}
-
-  AVSC_LOAD_FUNC(avs_add_function);
-  AVSC_LOAD_FUNC(avs_at_exit);
-  AVSC_LOAD_FUNC(avs_bit_blt);
-  AVSC_LOAD_FUNC(avs_check_version);
-  AVSC_LOAD_FUNC(avs_clip_get_error);
-  AVSC_LOAD_FUNC(avs_copy_clip);
-  AVSC_LOAD_FUNC(avs_copy_value);
-  AVSC_LOAD_FUNC(avs_copy_video_frame);
-  AVSC_LOAD_FUNC(avs_create_script_environment);
-  AVSC_LOAD_FUNC(avs_delete_script_environment);
-  AVSC_LOAD_FUNC(avs_function_exists);
-  AVSC_LOAD_FUNC(avs_get_audio);
-  AVSC_LOAD_FUNC(avs_get_cpu_flags);
-  AVSC_LOAD_FUNC(avs_get_error);
-  AVSC_LOAD_FUNC(avs_get_frame);
-  AVSC_LOAD_FUNC(avs_get_parity);
-  AVSC_LOAD_FUNC(avs_get_var);
-  AVSC_LOAD_FUNC(avs_get_version);
-  AVSC_LOAD_FUNC(avs_get_video_info);
-  AVSC_LOAD_FUNC(avs_invoke);
-  AVSC_LOAD_FUNC(avs_make_writable);
-  AVSC_LOAD_FUNC(avs_new_c_filter);
-  AVSC_LOAD_FUNC(avs_new_video_frame_a);
-  AVSC_LOAD_FUNC(avs_release_clip);
-  AVSC_LOAD_FUNC(avs_release_value);
-  AVSC_LOAD_FUNC(avs_release_video_frame);
-  AVSC_LOAD_FUNC(avs_save_string);
-  AVSC_LOAD_FUNC(avs_set_cache_hints);
-  AVSC_LOAD_FUNC(avs_set_global_var);
-  AVSC_LOAD_FUNC(avs_set_memory_max);
-  AVSC_LOAD_FUNC(avs_set_to_clip);
-  AVSC_LOAD_FUNC(avs_set_var);
-  AVSC_LOAD_FUNC(avs_set_working_dir);
-  AVSC_LOAD_FUNC(avs_sprintf);
-  AVSC_LOAD_FUNC(avs_subframe);
-  AVSC_LOAD_FUNC(avs_subframe_planar);
-  AVSC_LOAD_FUNC(avs_take_clip);
-  AVSC_LOAD_FUNC(avs_vsprintf);
-
-#undef __AVSC_STRINGIFY
-#undef AVSC_STRINGIFY
-#undef AVSC_LOAD_FUNC
-
-  return library;
-
-fail:
-  free(library);
-  return NULL;
-}
-
-AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (library == NULL)
-    return;
-  FreeLibrary(library->handle);
-  free(library);
-}
-#endif
-
-#endif
--- a/compat/avisynth/avisynth_c_25.h
+++ b/compat/avisynth/avisynth_c_25.h
@@ -1,68 +0,0 @@
-//  Copyright (c) 2011 FFmpegSource Project
-//
-//  Permission is hereby granted, free of charge, to any person obtaining a copy
-//  of this software and associated documentation files (the "Software"), to deal
-//  in the Software without restriction, including without limitation the rights
-//  to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
-//  copies of the Software, and to permit persons to whom the Software is
-//  furnished to do so, subject to the following conditions:
-//
-//  The above copyright notice and this permission notice shall be included in
-//  all copies or substantial portions of the Software.
-//
-//  THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-//  IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-//  FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-//  AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
-//  LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
-//  OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
-//  THE SOFTWARE.
-
-/* these are defines/functions that are used and were changed in the switch to 2.6
- * and are needed to maintain full compatility with 2.5 */
-
-enum {
-  AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-};
-
-AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
-    switch (plane)
-    {
-        case AVS_PLANAR_U: case AVS_PLANAR_V:
-            if (p->pitchUV)
-                return p->height>>1;
-            return 0;
-    }
-    return p->height;}
-
-AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
-    int r;
-    switch (plane)
-    {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-        if (p->pitchUV)
-            return p->row_size>>1;
-        else
-            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-        if (p->pitchUV)
-        {
-            r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-            if (r < p->pitchUV)
-                return r;
-            return p->row_size>>1;
-        }
-        else
-            return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-        r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-        if (r <= p->pitch)
-            return r;
-        return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
-    { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@@ -1,727 +0,0 @@
-// Avisynth C Interface Version 0.20
-// Copyright 2003 Kevin Atkinson
-
-// This program is free software; you can redistribute it and/or modify
-// it under the terms of the GNU General Public License as published by
-// the Free Software Foundation; either version 2 of the License, or
-// (at your option) any later version.
-//
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License for more details.
-//
-// You should have received a copy of the GNU General Public License
-// along with this program; if not, write to the Free Software
-// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
-// http://www.gnu.org/copyleft/gpl.html .
-//
-// As a special exception, I give you permission to link to the
-// Avisynth C interface with independent modules that communicate with
-// the Avisynth C interface solely through the interfaces defined in
-// avisynth_c.h, regardless of the license terms of these independent
-// modules, and to copy and distribute the resulting combined work
-// under terms of your choice, provided that every copy of the
-// combined work is accompanied by a complete copy of the source code
-// of the Avisynth C interface and Avisynth itself (with the version
-// used to produce the combined work), being distributed under the
-// terms of the GNU General Public License plus this exception.  An
-// independent module is a module which is not derived from or based
-// on Avisynth C Interface, such as 3rd-party filters, import and
-// export plugins, or graphical user interfaces.
-
-#ifndef __AVXSYNTH_C__
-#define __AVXSYNTH_C__
-
-#include "windowsPorts/windows2linux.h"
-#include <stdarg.h>
-
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// Constants
-//
-
-#ifndef __AVXSYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 3 };
-#endif
-
-enum {AVS_SAMPLE_INT8  = 1<<0,
-      AVS_SAMPLE_INT16 = 1<<1,
-      AVS_SAMPLE_INT24 = 1<<2,
-      AVS_SAMPLE_INT32 = 1<<3,
-      AVS_SAMPLE_FLOAT = 1<<4};
-
-enum {AVS_PLANAR_Y=1<<0,
-      AVS_PLANAR_U=1<<1,
-      AVS_PLANAR_V=1<<2,
-      AVS_PLANAR_ALIGNED=1<<3,
-      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
-      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
-
-  // Colorspace properties.
-enum {AVS_CS_BGR = 1<<28,
-      AVS_CS_YUV = 1<<29,
-      AVS_CS_INTERLEAVED = 1<<30,
-      AVS_CS_PLANAR = 1<<31};
-
-  // Specific colorformats
-enum {
-  AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
-  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
-};
-
-enum {
-  AVS_IT_BFF = 1<<0,
-  AVS_IT_TFF = 1<<1,
-  AVS_IT_FIELDBASED = 1<<2};
-
-enum {
-  AVS_FILTER_TYPE=1,
-  AVS_FILTER_INPUT_COLORSPACE=2,
-  AVS_FILTER_OUTPUT_TYPE=9,
-  AVS_FILTER_NAME=4,
-  AVS_FILTER_AUTHOR=5,
-  AVS_FILTER_VERSION=6,
-  AVS_FILTER_ARGS=7,
-  AVS_FILTER_ARGS_INFO=8,
-  AVS_FILTER_ARGS_DESCRIPTION=10,
-  AVS_FILTER_DESCRIPTION=11};
-
-enum {  //SUBTYPES
-  AVS_FILTER_TYPE_AUDIO=1,
-  AVS_FILTER_TYPE_VIDEO=2,
-  AVS_FILTER_OUTPUT_TYPE_SAME=3,
-  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
-
-enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
-};
-
-#define AVS_FRAME_ALIGN 16
-
-typedef struct AVS_Clip AVS_Clip;
-typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoInfo
-//
-
-// AVS_VideoInfo is layed out identicly to VideoInfo
-typedef struct AVS_VideoInfo {
-  int width, height;    // width=0 means no video
-  unsigned fps_numerator, fps_denominator;
-  int num_frames;
-
-  int pixel_type;
-
-  int audio_samples_per_second;   // 0 means no audio
-  int sample_type;
-  INT64 num_audio_samples;
-  int nchannels;
-
-  // Imagetype properties
-
-  int image_type;
-} AVS_VideoInfo;
-
-// useful functions of the above
-AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
-        { return (p->width!=0); }
-
-AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
-        { return (p->audio_samples_per_second!=0); }
-
-AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_BGR); }
-
-AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
-
-AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
-
-AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type&AVS_CS_YUV ); }
-
-AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
-
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
-
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return ((p->pixel_type & c_space) == c_space); }
-
-AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
-
-AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
-        { return !!(p->pixel_type & AVS_CS_PLANAR); }
-
-AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_FIELDBASED); }
-
-AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
-        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
-
-AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_BFF); }
-
-AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
-        { return !!(p->image_type & AVS_IT_TFF); }
-
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
-
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
-
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
-
-AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
-        { return p->audio_samples_per_second; }
-
-
-AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
-{
-    switch (p->sample_type) {
-      case AVS_SAMPLE_INT8:  return sizeof(signed char);
-      case AVS_SAMPLE_INT16: return sizeof(signed short);
-      case AVS_SAMPLE_INT24: return 3;
-      case AVS_SAMPLE_INT32: return sizeof(signed int);
-      case AVS_SAMPLE_FLOAT: return sizeof(float);
-      default: return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
-        { return p->nchannels*avs_bytes_per_channel_sample(p);}
-
-AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
-        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
-
-AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
-
-AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
-        { return bytes / avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
-        { return samples * avs_bytes_per_audio_sample(p); }
-
-AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
-        { return p->nchannels; }
-
-AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
-        { return p->sample_type;}
-
-// useful mutator
-AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
-        { p->image_type|=property; }
-
-AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
-        { p->image_type&=~property; }
-
-AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
-        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
-
-AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
-{
-    unsigned x=numerator, y=denominator;
-    while (y) {   // find gcd
-      unsigned t = x%y; x = y; y = t;
-    }
-    p->fps_numerator = numerator/x;
-    p->fps_denominator = denominator/x;
-}
-
-AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
-{
-        return (x->pixel_type == y->pixel_type)
-                || (avs_is_yv12(x) && avs_is_yv12(y));
-}
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_VideoFrame
-//
-
-// VideoFrameBuffer holds information about a memory block which is used
-// for video data.  For efficiency, instances of this class are not deleted
-// when the refcount reaches zero; instead they're stored in a linked list
-// to be reused.  The instances are deleted when the corresponding AVS
-// file is closed.
-
-// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrameBuffer {
-  unsigned char * data;
-  int data_size;
-  // sequence_number is incremented every time the buffer is changed, so
-  // that stale views can tell they're no longer valid.
-  long sequence_number;
-
-  long refcount;
-} AVS_VideoFrameBuffer;
-
-// VideoFrame holds a "window" into a VideoFrameBuffer.
-
-// AVS_VideoFrame is layed out identicly to IVideoFrame
-// DO NOT USE THIS STRUCTURE DIRECTLY
-typedef struct AVS_VideoFrame {
-  int refcount;
-  AVS_VideoFrameBuffer * vfb;
-  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
-} AVS_VideoFrame;
-
-// Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
-
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
-
-AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
-        return p->row_size; }
-
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_size>>1;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_size>>1;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
-        return p->height;}
-
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->height>>1;
-                        return 0;
-        }
-        return p->height;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
-
-AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
-
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
-// makes a shallow copy of a video frame
-AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
-  {avs_release_video_frame(f);}
-AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
-  {return avs_copy_video_frame(f);}
-#endif
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Value
-//
-
-// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
-// and avs_release_value appropiaty as you would if AVS_Value was
-// a pointer.
-
-// To maintain source code compatibility with future versions of the
-// avisynth_c API don't use the AVS_Value directly.  Use the helper
-// functions below.
-
-// AVS_Value is layed out identicly to AVSValue
-typedef struct AVS_Value AVS_Value;
-struct AVS_Value {
-  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
-               // for some function e'rror
-  short array_size;
-  union {
-    void * clip; // do not use directly, use avs_take_clip
-    char boolean;
-    int integer;
-    INT64 integer64; // match addition of __int64 to avxplugin.h
-    float floating_pt;
-    const char * string;
-    const AVS_Value * array;
-  } d;
-};
-
-// AVS_Value should be initilized with avs_void.
-// Should also set to avs_void after the value is released
-// with avs_copy_value.  Consider it the equalvent of setting
-// a pointer to NULL
-static const AVS_Value avs_void = {'v'};
-
-AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
-AVSC_API(void, avs_release_value)(AVS_Value);
-
-AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
-AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
-AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
-AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
-AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
-AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
-AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
-AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
-AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-AVSC_INLINE int avs_as_bool(AVS_Value v)
-        { return v.d.boolean; }
-AVSC_INLINE int avs_as_int(AVS_Value v)
-        { return v.d.integer; }
-AVSC_INLINE const char * avs_as_string(AVS_Value v)
-        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
-AVSC_INLINE double avs_as_float(AVS_Value v)
-        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
-AVSC_INLINE const char * avs_as_error(AVS_Value v)
-        { return avs_is_error(v) ? v.d.string : 0; }
-AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
-        { return v.d.array; }
-AVSC_INLINE int avs_array_size(AVS_Value v)
-        { return avs_is_array(v) ? v.array_size : 1; }
-AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
-        { return avs_is_array(v) ? v.d.array[index] : v; }
-
-// only use these functions on am AVS_Value that does not already have
-// an active value.  Remember, treat AVS_Value as a fat pointer.
-AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
-AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
-AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
-AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
-#endif
-AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_Clip
-//
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_release_clip)(AVS_Clip *);
-AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
-
-AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
-
-AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
-
-AVSC_API(int, avs_get_version)(AVS_Clip *);
-
-AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
-// The returned video frame must be released with avs_release_video_frame
-
-AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
-// return field parity if field_based, else parity of first field in frame
-
-AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
-                                  INT64 start, INT64 count);
-// start and count are in samples
-
-AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
-                                        int cachehints, size_t frame_range);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// This is the callback type used by avs_add_function
-typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
-                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
-
-typedef struct AVS_FilterInfo AVS_FilterInfo;
-struct AVS_FilterInfo
-{
-  // these members should not be modified outside of the AVS_ApplyFunc callback
-  AVS_Clip * child;
-  AVS_VideoInfo vi;
-  AVS_ScriptEnvironment * env;
-  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
-  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
-                                  INT64 start, INT64 count);
-  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
-                                        int frame_range);
-  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
-
-  // Should be set when ever there is an error to report.
-  // It is cleared before any of the above methods are called
-  const char * error;
-  // this is to store whatever and may be modified at will
-  void * user_data;
-};
-
-// Create a new filter
-// fi is set to point to the AVS_FilterInfo so that you can
-//   modify it once it is initilized.
-// store_child should generally be set to true.  If it is not
-//    set than ALL methods (the function pointers) must be defined
-// If it is set than you do not need to worry about freeing the child
-//    clip.
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
-                                      AVS_FilterInfo * * fi,
-                                      AVS_Value child, int store_child);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-
-/////////////////////////////////////////////////////////////////////
-//
-// AVS_ScriptEnvironment
-//
-
-// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
-enum {
-                                /* slowest CPU to support extension */
-  AVS_CPU_FORCE        = 0x01,   // N/A
-  AVS_CPU_FPU          = 0x02,   // 386/486DX
-  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
-  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
-  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
-  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
-  AVS_CPU_3DNOW        = 0x40,   // K6-2
-  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
-  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
-                                 // which only Hammer will have anyway)
-};
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
-
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
-AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
-
-AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
-AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
-
-AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
- // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
-
-AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
-                                     const char * name, const char * params,
-                                     AVS_ApplyFunc apply, void * user_data);
-
-AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
-
-AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
-                               AVS_Value args, const char** arg_names);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
-// The returned value must be be released with avs_release_value
-
-AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
-
-AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
-
-//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
-//void avs_pop_context(AVS_ScriptEnvironment *);
-
-AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
-                                          const AVS_VideoInfo * vi, int align);
-// align should be at least 16
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#ifndef AVSC_NO_DECLSPEC
-AVSC_INLINE
-AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
-                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-
-AVSC_INLINE
-AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
-                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
-#endif
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
-
-AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
-
-typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
-AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
-
-AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
-// The returned video frame must be be released
-
-AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
-
-AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
-
-// avisynth.dll exports this; it's a way to use it as a library, without
-// writing an AVS script or without going through AVIFile.
-AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-// this symbol is the entry point for the plugin and must
-// be defined
-AVSC_EXPORT
-const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
-
-
-#if defined __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
-
-
-AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
-// The returned video frame must be be released
-#if defined __cplusplus
-}
-#endif // __cplusplus
-
-#endif //__AVXSYNTH_C__
--- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h
+++ b/compat/avisynth/windowsPorts/basicDataTypeConversions.h
@@ -1,85 +0,0 @@
-#ifndef __DATA_TYPE_CONVERSIONS_H__
-#define __DATA_TYPE_CONVERSIONS_H__
-
-#include <stdint.h>
-#include <wchar.h>
-
-#ifdef __cplusplus
-namespace avxsynth {
-#endif // __cplusplus
-
-typedef int64_t __int64;
-typedef int32_t __int32;
-#ifdef __cplusplus
-typedef bool BOOL;
-#else
-typedef uint32_t BOOL;
-#endif // __cplusplus
-typedef void* HMODULE;
-typedef void* LPVOID;
-typedef void* PVOID;
-typedef PVOID HANDLE;
-typedef HANDLE HWND;
-typedef HANDLE HINSTANCE;
-typedef void* HDC;
-typedef void* HBITMAP;
-typedef void* HICON;
-typedef void* HFONT;
-typedef void* HGDIOBJ;
-typedef void* HBRUSH;
-typedef void* HMMIO;
-typedef void* HACMSTREAM;
-typedef void* HACMDRIVER;
-typedef void* HIC;
-typedef void* HACMOBJ;
-typedef HACMSTREAM* LPHACMSTREAM;
-typedef void* HACMDRIVERID;
-typedef void* LPHACMDRIVER;
-typedef unsigned char BYTE;
-typedef BYTE* LPBYTE;
-typedef char TCHAR;
-typedef TCHAR* LPTSTR;
-typedef const TCHAR* LPCTSTR;
-typedef char* LPSTR;
-typedef LPSTR LPOLESTR;
-typedef const char* LPCSTR;
-typedef LPCSTR LPCOLESTR;
-typedef wchar_t WCHAR;
-typedef unsigned short WORD;
-typedef unsigned int UINT;
-typedef UINT MMRESULT;
-typedef uint32_t DWORD;
-typedef DWORD COLORREF;
-typedef DWORD FOURCC;
-typedef DWORD HRESULT;
-typedef DWORD* LPDWORD;
-typedef DWORD* DWORD_PTR;
-typedef int32_t LONG;
-typedef int32_t* LONG_PTR;
-typedef LONG_PTR LRESULT;
-typedef uint32_t ULONG;
-typedef uint32_t* ULONG_PTR;
-//typedef __int64_t intptr_t;
-typedef uint64_t _fsize_t;
-
-
-//
-// Structures
-//
-
-typedef struct _GUID {
-  DWORD Data1;
-  WORD  Data2;
-  WORD  Data3;
-  BYTE  Data4[8];
-} GUID;
-
-typedef GUID REFIID;
-typedef GUID CLSID;
-typedef CLSID* LPCLSID;
-typedef GUID IID;
-
-#ifdef __cplusplus
-}; // namespace avxsynth
-#endif // __cplusplus
-#endif //  __DATA_TYPE_CONVERSIONS_H__
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -1,77 +0,0 @@
-#ifndef __WINDOWS2LINUX_H__
-#define __WINDOWS2LINUX_H__
-
-/*
- * LINUX SPECIFIC DEFINITIONS
-*/
-//
-// Data types conversions
-//
-#include <stdlib.h>
-#include <string.h>
-#include "basicDataTypeConversions.h"
-
-#ifdef __cplusplus
-namespace avxsynth {
-#endif // __cplusplus
-//
-// purposefully define the following MSFT definitions
-// to mean nothing (as they do not mean anything on Linux)
-//
-#define __stdcall
-#define __cdecl
-#define noreturn
-#define __declspec(x)
-#define STDAPI       extern "C" HRESULT
-#define STDMETHODIMP HRESULT __stdcall
-#define STDMETHODIMP_(x) x __stdcall
-
-#define STDMETHOD(x)    virtual HRESULT x
-#define STDMETHOD_(a, x) virtual a x
-
-#ifndef TRUE
-#define TRUE  true
-#endif
-
-#ifndef FALSE
-#define FALSE false
-#endif
-
-#define S_OK                (0x00000000)
-#define S_FALSE             (0x00000001)
-#define E_NOINTERFACE       (0X80004002)
-#define E_POINTER           (0x80004003)
-#define E_FAIL              (0x80004005)
-#define E_OUTOFMEMORY       (0x8007000E)
-
-#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
-#define FAILED(hr)              ((hr) & 0x80000000)
-#define SUCCEEDED(hr)           (!FAILED(hr))
-
-
-//
-// Functions
-//
-#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
-#define MAKEWORD(a,b) ((a << 8) | (b))
-
-#define lstrlen                             strlen
-#define lstrcpy                             strcpy
-#define lstrcmpi                            strcasecmp
-#define _stricmp                            strcasecmp
-#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
-#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
-// Windows uses (new, old) ordering but GCC has (old, new)
-#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
-
-#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
-#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
-#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
-
-#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
-
-#ifdef __cplusplus
-}; // namespace avxsynth
-#endif // __cplusplus
-
-#endif //  __WINDOWS2LINUX_H__
--- a/compat/strtod.c
+++ b/compat/strtod.c
@@ -19,6 +19,7 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

+#include <ctype.h>
 #include <limits.h>
 #include <stdlib.h>

@@ -48,7 +49,7 @@ double avpriv_strtod(const char *nptr, char **endptr)
    double res;

    /* Skip leading spaces */
-    while (av_isspace(*nptr))
+    while (isspace(*nptr))
        nptr++;

    if (!av_strncasecmp(nptr, "infinity", 8)) {
--- a/compat/tms470/math.h
+++ b/compat/tms470/math.h
@@ -1,24 +1,3 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_COMPAT_TMS470_MATH_H
-#define FFMPEG_COMPAT_TMS470_MATH_H
-
 #include_next <math.h>

 #undef INFINITY
@@ -26,5 +5,3 @@

 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
-
-#endif /* FFMPEG_COMPAT_TMS470_MATH_H */
--- a/1223
+++ b/1223
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -2,10 +2,10 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.

 The last version increases were:
-libavcodec:    2013-03-xx
-libavdevice:   2013-03-xx
+libavcodec:    2012-01-27
+libavdevice:   2011-04-18
 libavfilter:   2012-06-22
-libavformat:   2013-03-xx
+libavformat:   2012-01-27
 libavresample: 2012-10-05
 libpostproc:   2011-04-18
 libswresample: 2011-09-19
@@ -15,167 +15,11 @@ libavutil:     2012-10-22

 API changes, most recent first:

-2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
-  Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
-
-2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
-  Add av_get_known_color_name().
-
-2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
-  Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
-  av_opt_get_channel_layout() and av_opt_set_channel_layout().
-
-2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
-  Change default scaler to bicubic
-
-2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
-  Add av_codec_get_max_lowres()
-
-2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
-  Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
-  decoders
-
-2013-08-xx - xxxxxxx - lavfi 3.11.0 - avfilter.h
-  Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
-  implementations.
-
-2013-09-21 - xxxxxxx - lavu 52.16.0 - pixfmt.h
-  Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
-  AV_PIX_FMT_NV20.
-
-2013-09-16 - c74c3fb / 3feb3d6 - lavu 52.44.100 / 52.15.0 - mem.h
-  Add av_reallocp.
-
-2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
-  avcodec_close() argument can be NULL.
-
-2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
-  avformat_close_input() argument can be NULL and point on NULL.
-
-2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
-  Add av_format_get_probe_score().
-
-2013-08-15 - 1e0e193 - lsws 2.5.100 -
-  Add a sws_dither AVOption, allowing to set the dither algorithm used
-
-2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
-  Add a render2 alternative to the render callback function.
-
-2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
-  Add allocation function for AVVDPAUContext, allowing
-  to extend it in the future without breaking ABI/API.
-
-2013-08-10 - 67a580f / 5a9a9d4 - lavc 55.25.100 / 55.16.0 - avcodec.h
-  Extend AVPacket API with av_packet_unref, av_packet_ref,
-  av_packet_move_ref, av_packet_copy_props, av_packet_free_side_data.
-
-2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
-  Deprecate the bitstream-related members from struct AVVDPAUContext.
-  The bistream buffers no longer need to be explicitly freed.
-
-2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
-  Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
-  and select the AV_PIX_FMT_VDPAU format with get_format() instead.
-
-2013-08-05 - 4ee0984 / a0ad5d0 - lavu 52.41.100 / 52.14.0 - pixfmt.h
-  Deprecate AV_PIX_FMT_VDPAU_*. Use AV_PIX_FMT_VDPAU instead.
-
-2013-08-02 - 82fdfe8 / a8b1927 - lavc 55.20.100 / 55.11.0 - avcodec.h
-  Add output_picture_number to AVCodecParserContext.
-
-2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h
-  Add avcodec_chroma_pos_to_enum()
-  Add avcodec_enum_to_chroma_pos()
-
-2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
-  Deprecate avfilter_graph_parse() in favor of the equivalent
-  avfilter_graph_parse_ptr().
-
-2013-06-24 - af5f9c0 / 95d5246 - lavc 55.17.100 / 55.10.0 - avcodec.h
-  Add MPEG-2 AAC profiles
-
-2013-06-25 - af5f9c0 / 95d5246 - lavf 55.10.100 - avformat.h
-  Add AV_DISPOSITION_* flags to indicate text track kind.
-
-2013-06-15 - 99b8cd0 - lavu 52.36.100
-  Add AVRIPEMD:
-   av_ripemd_alloc()
-   av_ripemd_init()
-   av_ripemd_update()
-   av_ripemd_final()
-
-2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
-  Add av_realloc_array and av_reallocp_array
-
-2013-05-30 - 682b227 - lavu 52.35.100
-  Add AVSHA512:
-   av_sha512_alloc()
-   av_sha512_init()
-   av_sha512_update()
-   av_sha512_final()
-
-2013-05-24 - 8d4e969 / 129bb23 - lavfi 3.10.0 / 3.70.100 - avfilter.h
-  Add support for slice multithreading to lavfi. Filters supporting threading
-  are marked with AVFILTER_FLAG_SLICE_THREADS.
-  New fields AVFilterContext.thread_type, AVFilterGraph.thread_type and
-  AVFilterGraph.nb_threads (accessible directly or through AVOptions) may be
-  used to configure multithreading.
-
-2013-05-24 - fe40a9f / 2a6eaea - lavu 52.12.0 / 52.34.100 - cpu.h
-  Add av_cpu_count() function for getting the number of logical CPUs.
-
-2013-05-24 - 0c25c39 / b493847 - lavc 55.7.0 / 55.12.100 - avcodec.h
-  Add picture_structure to AVCodecParserContext.
-
-2013-05-17 - 3a751ea - lavu 52.33.100 - opt.h
-  Add AV_OPT_TYPE_COLOR value to AVOptionType enum.
-
-2013-05-13 - e398416 - lavu 52.31.100 - mem.h
-  Add av_dynarray2_add().
-
-2013-05-12 - 1776177 - lavfi 3.65.100
-  Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags.
-
-2013-04-19 - 380cfce - lavc 55.4.100
-  Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec.
-
-2013-04-18 - 7c1a002 - lavf 55.3.100
-  The matroska demuxer can now output proper verbatim ASS packets. It will
-  become the default starting lavf 56.0.100.
-
-2013-04-10 - af0d270 - lavu 25.26.100 - avutil.h,opt.h
-  Add av_int_list_length()
-  and av_opt_set_int_list().
-
-2013-03-30 - 5c73645 - lavu 52.24.100 - samplefmt.h
-  Add av_samples_alloc_array_and_samples().
-
-2013-03-29 - ef7b6b4 - lavf 55.1.100 - avformat.h
-  Add av_guess_frame_rate()
-
-2013-03-20 - 8d928a9 - lavu 52.22.100 - opt.h
-  Add AV_OPT_TYPE_DURATION value to AVOptionType enum.
-
-2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
-  Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
-
-2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
-  Add av_escape() and av_bprint_escape() API.
-
-2013-02-24 - b59cd08 - lavfi 3.41.100 - buffersink.h
-  Add sample_rates field to AVABufferSinkParams.
-
-2013-01-17 - a1a707f - lavf 54.61.100
-  Add av_codec_get_tag2().
-
-2013-01-01 - 2eb2e17 - lavfi 3.34.100
-  Add avfilter_get_audio_buffer_ref_from_arrays_channels.
-
-2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
+2012-12-20 - xxxxxxx - lavfi 3.28.100 - avfilter.h
  Add AVFilterLink.channels, avfilter_link_get_channels()
  and avfilter_ref_get_channels().

-2012-12-15 - 96d815fc - lavc 54.80.100 - avcodec.h
+2012-12-15 - 2ada584d - lavc 54.80.100 - avcodec.h
  Add pkt_size field to AVFrame.

 2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h
@@ -288,102 +132,6 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
  Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.

-2013-05-15 - ff46809 / e6c4ac7 - lavu 52.32.100 / 52.11.0 - pixdesc.h
-  Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*.
-
-2013-04-03 - 6fc58a8 / 507b1e4 - lavc 55.7.100 / 55.4.0 - avcodec.h
-  Add field_order to AVCodecParserContext.
-
-2013-04-19 - f4b05cd / 5e83d9a - lavc 55.5.100 / 55.2.0 - avcodec.h
-  Add CODEC_FLAG_UNALIGNED to allow decoders to produce unaligned output.
-
-2013-04-11 - lavfi 3.53.100 / 3.8.0
-  231fd44 / 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate
-            avfilterhraph.h, user applications should include just avfilter.h
-  86070b8 / bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and
-            avfilter_graph_add_filter().
-  4fde705 / 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the
-            filter.
-  710b0aa / 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter().
-  46de9ba / 1ba95a9 - Add avfilter_init_dict().
-  16fc24b / 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags.
-  f4db6bf / 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs.
-  835cc0f / fa2a34c - Add avfilter_next(), deprecate av_filter_next().
-            Deprecate avfilter_uninit().
-
-2013-04-09 - lavfi 3.51.100 / 3.7.0 - avfilter.h
-  0594ef0 / b439c99 - Add AVFilter.priv_class for exporting filter options through the
-            AVOptions API in the similar way private options work in lavc and lavf.
-  44d4488 / 8114c10 - Add avfilter_get_class().
-  Switch all filters to use AVOptions.
-
-2013-03-19 - 17ebef2 / 2c328a9 - lavu 52.20.100 / 52.9.0 - pixdesc.h
-  Add av_pix_fmt_count_planes() function for counting planes in a pixel format.
-
-2013-03-16 - ecade98 / 42c7c61 - lavfi 3.47.100 / 3.6.0
-  Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count.
-
-2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.100 / 55.0.0, lavf 55.0.100 / 55.0.0,
-lavd 54.4.100 / 54.0.0, lavfi 3.5.0
-  36099df / 8e401db, 532f31a / 1cec062 - add a new API for reference counted buffers and buffer
-                     pools (new header libavutil/buffer.h).
-  2653e12 / 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data.
-            Add av_packet_from_data() function for constructing packets from
-            av_malloc()ed data.
-  c4e8821 / 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add
-            AVFrame.buf/extended_buf to allow reference counting for the AVFrame
-            data. Add new API for working with reference-counted AVFrames.
-  80e9e63 / 759001c - add the refcounted_frames field to AVCodecContext to make audio and
-            video decoders return reference-counted frames. Add get_buffer2()
-            callback to AVCodecContext which allocates reference-counted frames.
-            Add avcodec_default_get_buffer2() as the default get_buffer2()
-            implementation.
-            Deprecate AVCodecContext.get_buffer() / release_buffer() /
-            reget_buffer(), avcodec_default_get_buffer(),
-            avcodec_default_reget_buffer(), avcodec_default_release_buffer().
-            Remove avcodec_default_free_buffers(), which should not have ever
-            been called from outside of lavc.
-            Deprecate the following AVFrame fields:
-                * base -- is now stored in AVBufferRef
-                * reference, type, buffer_hints -- are unnecessary in the new API
-                * hwaccel_picture_private, owner, thread_opaque -- should not
-                  have been acessed from outside of lavc
-                * qscale_table, qstride, qscale_type, mbskip_table, motion_val,
-                  mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
-                  which are not exported anymore.
-  a05a44e / 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add
-            av_buffersrc_add_frame(), deprecate av_buffersrc_buffer().
-            Add av_buffersink_get_frame() and av_buffersink_get_samples(),
-            deprecate av_buffersink_read() and av_buffersink_read_samples().
-            Deprecate AVFilterBufferRef and all functions for working with it.
-
-2013-03-17 - 6c17ff8 / 12c5c1d - lavu 52.19.100 / 52.8.0 - avstring.h
-  Add av_isdigit, av_isgraph, av_isspace, av_isxdigit.
-
-2013-02-23 - 71cf094 / 9f12235 - lavfi 3.40.100 / 3.4.0 - avfiltergraph.h
-  Add resample_lavr_opts to AVFilterGraph for setting libavresample options
-  for auto-inserted resample filters.
-
-2013-01-25 - e7e14bc / 38c1466 - lavu 52.17.100 / 52.7.0 - dict.h
-  Add av_dict_parse_string() to set multiple key/value pairs at once from a
-  string.
-
-2013-01-25 - 25be630 / b85a5e8 - lavu 52.16.100 / 52.6.0 - avstring.h
-  Add av_strnstr()
-
-2013-01-15 - e7e0186 / 8ee288d - lavu 52.15.100 / 52.5.0 - hmac.h
-  Add AVHMAC.
-
-2013-01-13 - 8ee7b38 / 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h
-  Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding.
-
-2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h
-  Add AV_PIX_FMT_VDPAU flag.
-
-2013-01-07 - 249fca3 / 074a00d - lavr 1.1.0
-  Add avresample_set_channel_mapping() for input channel reordering,
-  duplication, and silencing.
-
 2012-12-29 - 2ce43b3 / d8fd06c - lavu 52.13.100 / 52.3.0 - avstring.h
  Add av_basename() and av_dirname().

@@ -462,7 +210,7 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h
  Add AVFMT_FLAG_NOBUFFER for low latency use cases.

-2012-07-10 - fbe0245 / f3e5e6f - lavu 51.65.100 / 51.37.0
+2012-07-10 - 5fade8a - lavu 51.37.0
  Add av_malloc_array() and av_mallocz_array()

 2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -31,9 +31,9 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         = 2.1.1
+PROJECT_NUMBER         = 1.1.7

-# With the PROJECT_LOGO tag one can specify a logo or icon that is included
+# With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
 # pixels and the maximum width should not exceed 200 pixels. Doxygen will
 # copy the logo to the output directory.
@@ -277,7 +277,7 @@ SUBGROUPING            = YES
 # be useful for C code in case the coding convention dictates that all compound
 # types are typedef'ed and only the typedef is referenced, never the tag name.

-TYPEDEF_HIDES_STRUCT   = YES
+TYPEDEF_HIDES_STRUCT   = NO

 # The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
 # determine which symbols to keep in memory and which to flush to disk.
@@ -409,7 +409,7 @@ INLINE_INFO            = YES
 # alphabetically by member name. If set to NO the members will appear in
 # declaration order.

-SORT_MEMBER_DOCS       = NO
+SORT_MEMBER_DOCS       = YES

 # If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the
 # brief documentation of file, namespace and class members alphabetically
@@ -709,7 +709,7 @@ INLINE_SOURCES         = NO
 # doxygen to hide any special comment blocks from generated source code
 # fragments. Normal C and C++ comments will always remain visible.

-STRIP_CODE_COMMENTS    = NO
+STRIP_CODE_COMMENTS    = YES

 # If the REFERENCED_BY_RELATION tag is set to YES
 # then for each documented function all documented
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -6,6 +6,7 @@ LIBRARIES-$(CONFIG_AVFORMAT)   += libavformat
 LIBRARIES-$(CONFIG_AVDEVICE)   += libavdevice
 LIBRARIES-$(CONFIG_AVFILTER)   += libavfilter

+COMPONENTS-yes = $(PROGS-yes)
 COMPONENTS-$(CONFIG_AVUTIL)     += ffmpeg-utils
 COMPONENTS-$(CONFIG_SWSCALE)    += ffmpeg-scaler
 COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
@@ -14,11 +15,9 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters

-MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
-MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
-MANPAGES    = $(MANPAGES1) $(MANPAGES3)
-PODPAGES    = $(PROGS-yes:%=doc/%.pod)  $(PROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
+MANPAGES    = $(COMPONENTS-yes:%=doc/%.1)    $(LIBRARIES-yes:%=doc/%.3)
+PODPAGES    = $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html)     \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
@@ -60,22 +59,12 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 doc/%.html: TAG = HTML
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
-
-doc/%-all.html: TAG = HTML
-doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<

 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
+doc/%.pod: doc/%.texi $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
-
-doc/%-all.pod: TAG = POD
-doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@
+	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@

 doc/%.1 doc/%.3: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
@@ -88,46 +77,23 @@ $(DOCS) doc/doxy/html: | doc/
 doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
 	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^

-install-doc: install-html install-man
-
-install-html:
-
 install-man:

-ifdef CONFIG_HTMLPAGES
-install-progs-$(CONFIG_DOC): install-html
-
-install-html: $(HTMLPAGES)
-	$(Q)mkdir -p "$(DOCDIR)"
-	$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
-endif
-
 ifdef CONFIG_MANPAGES
 install-progs-$(CONFIG_DOC): install-man

 install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
-	$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
-	$(Q)mkdir -p "$(MANDIR)/man3"
-	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
+	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
 endif

-uninstall: uninstall-doc
-
-uninstall-doc: uninstall-html uninstall-man
-
-uninstall-html:
-	$(RM) -r "$(DOCDIR)"
+uninstall: uninstall-man

 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
-	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))

 clean:: docclean

-distclean:: docclean
-	$(RM) doc/config.texi
-
 docclean:
 	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html
--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -1,7 +1,7 @@
 Release Notes
 =============

-* 2.1 "Fourier"  October, 2013
+* 1.1 "Fire Flower"  January, 2013


 General notes
@@ -14,3 +14,12 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
+
+Of big interest to our Windows users, FFmpeg now supports building with the MSVC
+compiler. Since MSVC does not support C99 features used extensively by FFmpeg,
+this has been accomplished using a converter that turns C99 code to C89. See the
+platform-specific documentation for more detailed documentation on building
+FFmpeg with MSVC.
+
+The used output sample format for several audio decoders has changed, make
+sure you always check/use AVCodecContext.sample_fmt or AVFrame.format.
--- a/doc/avtools-common-opts.texi
+++ b/doc/avtools-common-opts.texi
@@ -1,33 +1,32 @@
-All the numerical options, if not specified otherwise, accept a string
-representing a number as input, which may be followed by one of the SI
-unit prefixes, for example: 'K', 'M', or 'G'.
-
-If 'i' is appended to the SI unit prefix, the complete prefix will be
-interpreted as a unit prefix for binary multiplies, which are based on
-powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
-prefix multiplies the value by 8. This allows using, for example:
-'KB', 'MiB', 'G' and 'B' as number suffixes.
+All the numerical options, if not specified otherwise, accept in input
+a string representing a number, which may contain one of the
+SI unit prefixes, for example 'K', 'M', 'G'.
+If 'i' is appended after the prefix, binary prefixes are used,
+which are based on powers of 1024 instead of powers of 1000.
+The 'B' postfix multiplies the value by 8, and can be
+appended after a unit prefix or used alone. This allows using for
+example 'KB', 'MiB', 'G' and 'B' as number postfix.

 Options which do not take arguments are boolean options, and set the
 corresponding value to true. They can be set to false by prefixing
-the option name with "no". For example using "-nofoo"
-will set the boolean option with name "foo" to false.
+with "no" the option name, for example using "-nofoo" in the
+command line will set to false the boolean option with name "foo".

@anchor{Stream specifiers}
@section Stream specifiers
 Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
-are used to precisely specify which stream(s) a given option belongs to.
+are used to precisely specify which stream(s) does a given option belong to.

 A stream specifier is a string generally appended to the option name and
-separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
-@code{a:1} stream specifier, which matches the second audio stream. Therefore, it
+separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
+@code{a:1} stream specifier, which matches the second audio stream. Therefore it
 would select the ac3 codec for the second audio stream.

-A stream specifier can match several streams, so that the option is applied to all
+A stream specifier can match several streams, the option is then applied to all
 of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
 streams.

-An empty stream specifier matches all streams. For example, @code{-codec copy}
+An empty stream specifier matches all streams, for example @code{-codec copy}
 or @code{-codec: copy} would copy all the streams without reencoding.

 Possible forms of stream specifiers are:
@@ -36,21 +35,20 @@ Possible forms of stream specifiers are:
 Matches the stream with this index. E.g. @code{-threads:1 4} would set the
 thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
-'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
-stream number @var{stream_index} of this type. Otherwise, it matches all
+@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
+'d' for data and 't' for attachments. If @var{stream_index} is given, then
+matches stream number @var{stream_index} of this type. Otherwise matches all
 streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
-in the program with the id @var{program_id}. Otherwise, it matches all streams in the
-program.
+If @var{stream_index} is given, then matches stream number @var{stream_index} in
+program with id @var{program_id}. Otherwise matches all streams in this program.
@item #@var{stream_id}
-Matches the stream by a format-specific ID.
+Matches the stream by format-specific ID.
@end table

@section Generic options

-These options are shared amongst the ff* tools.
+These options are shared amongst the av* tools.

@table @option

@@ -59,18 +57,10 @@ Show license.

@item -h, -?, -help, --help [@var{arg}]
 Show help. An optional parameter may be specified to print help about a specific
-item. If no argument is specified, only basic (non advanced) tool
-options are shown.
+item.

 Possible values of @var{arg} are:
@table @option
-@item long
-Print advanced tool options in addition to the basic tool options.
-
-@item full
-Print complete list of options, including shared and private options
-for encoders, decoders, demuxers, muxers, filters, etc.
-
@item decoder=@var{decoder_name}
 Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@@ -87,9 +77,6 @@ Print detailed information about the demuxer named @var{demuxer_name}. Use the
 Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.

-@item filter=@var{filter_name}
-Print detailed information about the filter name @var{filter_name}. Use the
-@option{-filters} option to get a list of all filters.
@end table

@item -version
@@ -98,6 +85,14 @@ Show version.
@item -formats
 Show available formats.

+The fields preceding the format names have the following meanings:
+@table @samp
+@item D
+Decoding available
+@item E
+Encoding available
+@end table
+
@item -codecs
 Show all codecs known to libavcodec.

@@ -128,39 +123,18 @@ Show available sample formats.
@item -layouts
 Show channel names and standard channel layouts.

-@item -colors
-Show recognized color names.
-
-@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
+@item -loglevel @var{loglevel} | -v @var{loglevel}
 Set the logging level used by the library.
-Adding "repeat+" indicates that repeated log output should not be compressed
-to the first line and the "Last message repeated n times" line will be
-omitted. "repeat" can also be used alone.
-If "repeat" is used alone, and with no prior loglevel set, the default
-loglevel will be used. If multiple loglevel parameters are given, using
-'repeat' will not change the loglevel.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
-Show nothing at all; be silent.
@item panic
-Only show fatal errors which could lead the process to crash, such as
-and assert failure. This is not currently used for anything.
@item fatal
-Only show fatal errors. These are errors after which the process absolutely
-cannot continue after.
@item error
-Show all errors, including ones which can be recovered from.
@item warning
-Show all warnings and errors. Any message related to possibly
-incorrect or unexpected events will be shown.
@item info
-Show informative messages during processing. This is in addition to
-warnings and errors. This is the default value.
@item verbose
-Same as @code{info}, except more verbose.
@item debug
-Show everything, including debugging information.
@end table

 By default the program logs to stderr, if coloring is supported by the
@@ -202,61 +176,7 @@ ffmpeg -cpuflags -sse+mmx ...
 ffmpeg -cpuflags mmx ...
 ffmpeg -cpuflags 0 ...
@end example
-Possible flags for this option are:
-@table @samp
-@item x86
-@table @samp
-@item mmx
-@item mmxext
-@item sse
-@item sse2
-@item sse2slow
-@item sse3
-@item sse3slow
-@item ssse3
-@item atom
-@item sse4.1
-@item sse4.2
-@item avx
-@item xop
-@item fma4
-@item 3dnow
-@item 3dnowext
-@item cmov
-@end table
-@item ARM
-@table @samp
-@item armv5te
-@item armv6
-@item armv6t2
-@item vfp
-@item vfpv3
-@item neon
-@end table
-@item PowerPC
-@table @samp
-@item altivec
-@end table
-@item Specific Processors
-@table @samp
-@item pentium2
-@item pentium3
-@item pentium4
-@item k6
-@item k62
-@item athlon
-@item athlonxp
-@item k8
-@end table
-@end table

-@item -opencl_options options (@emph{global})
-Set OpenCL environment options. This option is only available when
-FFmpeg has been compiled with @code{--enable-opencl}.
-
-@var{options} must be a list of @var{key}=@var{value} option pairs
-separated by ':'. See the ``OpenCL Options'' section in the
-ffmpeg-utils manual for the list of supported options.
@end table

@section AVOptions
@@ -281,12 +201,11 @@ muxer:
 ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example

-All codec AVOptions are per-stream, and thus a stream specifier
-should be attached to them.
+All codec AVOptions are obviously per-stream, so the chapter on stream
+specifiers applies to them

-Note: the @option{-nooption} syntax cannot be used for boolean
-AVOptions, use @option{-option 0}/@option{-option 1}.
+Note @option{-nooption} syntax cannot be used for boolean AVOptions,
+use @option{-option 0}/@option{-option 1}.

-Note: the old undocumented way of specifying per-stream AVOptions by
-prepending v/a/s to the options name is now obsolete and will be
-removed soon.
+Note2 old undocumented way of specifying per-stream AVOptions by prepending
+v/a/s to the options name is now obsolete and will be removed soon.
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -17,46 +17,9 @@ Below is a description of the currently available bitstream filters.

@section aac_adtstoasc

-Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
-bitstream filter.
-
-This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
-ADTS header and removes the ADTS header.
-
-This is required for example when copying an AAC stream from a raw
-ADTS AAC container to a FLV or a MOV/MP4 file.
-
@section chomp

-Remove zero padding at the end of a packet.
-
-@section dump_extra
-
-Add extradata to the beginning of the filtered packets.
-
-The additional argument specifies which packets should be filtered.
-It accepts the values:
-@table @samp
-@item a
-add extradata to all key packets, but only if @var{local_header} is
-set in the @option{flags2} codec context field
-
-@item k
-add extradata to all key packets
-
-@item e
-add extradata to all packets
-@end table
-
-If not specified it is assumed @samp{k}.
-
-For example the following @command{ffmpeg} command forces a global
-header (thus disabling individual packet headers) in the H.264 packets
-generated by the @code{libx264} encoder, but corrects them by adding
-the header stored in extradata to the key packets:
-@example
-ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
-@end example
+@section dump_extradata

@section h264_mp4toannexb

@@ -123,6 +86,6 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

@section noise

-@section remove_extra
+@section remove_extradata

@c man end BITSTREAM FILTERS
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -62,7 +62,7 @@ documented.

@section libcelt

-libcelt decoder wrapper.
+libcelt decoder wrapper

 libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
 Requires the presence of the libcelt headers and library during configuration.
@@ -70,7 +70,7 @@ You need to explicitly configure the build with @code{--enable-libcelt}.

@section libgsm

-libgsm decoder wrapper.
+libgsm decoder wrapper

 libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
 the presence of the libgsm headers and library during configuration. You need
@@ -80,7 +80,7 @@ This decoder supports both the ordinary GSM and the Microsoft variant.

@section libilbc

-libilbc decoder wrapper.
+libilbc decoder wrapper

 libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
 audio codec. Requires the presence of the libilbc headers and library during
@@ -101,7 +101,7 @@ value is 0 (disabled).

@section libopencore-amrnb

-libopencore-amrnb decoder wrapper.
+libopencore-amrnb decoder wrapper

 libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
 Narrowband audio codec. Using it requires the presence of the
@@ -158,45 +158,4 @@ ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@end table

-@section libzvbi-teletext
-
-Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
-subtitles. Requires the presence of the libzvbi headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libzvbi}.
-
-@subsection Options
-
-@table @option
-@item txt_page
-List of teletext page numbers to decode. You may use the special * string to
-match all pages. Pages that do not match the specified list are dropped.
-Default value is *.
-@item txt_chop_top
-Discards the top teletext line. Default value is 1.
-@item txt_format
-Specifies the format of the decoded subtitles. The teletext decoder is capable
-of decoding the teletext pages to bitmaps or to simple text, you should use
-"bitmap" for teletext pages, because certain graphics and colors cannot be
-expressed in simple text. You might use "text" for teletext based subtitles if
-your application can handle simple text based subtitles. Default value is
-bitmap.
-@item txt_left
-X offset of generated bitmaps, default is 0.
-@item txt_top
-Y offset of generated bitmaps, default is 0.
-@item txt_chop_spaces
-Chops leading and trailing spaces and removes empty lines from the generated
-text. This option is useful for teletext based subtitles where empty spaces may
-be present at the start or at the end of the lines or empty lines may be
-present between the subtitle lines because of double-sized teletext charactes.
-Default value is 1.
-@item txt_duration
-Sets the display duration of the decoded teletext pages or subtitles in
-miliseconds. Default value is 30000 which is 30 seconds.
-@item txt_transparent
-Force transparent background of the generated teletext bitmaps. Default value
-is 0 which means an opaque (black) background.
-@end table
-
@c man end SUBTILES DECODERS
--- a/doc/default.css
+++ b/doc/default.css
@@ -1,7 +1,3 @@
-a.summary-letter {
-    text-decoration: none;
-}
-
 a {
    color: #2D6198;
 }
@@ -49,16 +45,11 @@ body {
    text-align: center;
 }

-h1 a, h2 a, h3 a, h4 a {
-    text-decoration: inherit;
-    color: inherit;
-}
-
-h1, h2, h3, h4 {
+h1, h2, h3 {
    padding-left: 0.4em;
    border-radius: 4px;
-    padding-bottom: 0.25em;
-    padding-top: 0.25em;
+    padding-bottom: 0.2em;
+    padding-top: 0.2em;
    border: 1px solid #6A996A;
 }

@@ -72,22 +63,15 @@ h1 {

 h2 {
    color: #313131;
-    font-size: 1.0em;
+    font-size: 0.9em;
    background-color: #ABE3AB;
 }

 h3 {
-    color: #313131;
-    font-size: 0.9em;
-    margin-bottom: -6px;
-    background-color: #BBF3BB;
-}
-
-h4 {
    color: #313131;
    font-size: 0.8em;
    margin-bottom: -8px;
-    background-color: #D1FDD1;
+    background-color: #BBF3BB;
 }

 img {
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -1,155 +1,23 @@
@chapter Demuxers
@c man begin DEMUXERS

-Demuxers are configured elements in FFmpeg that can read the
+Demuxers are configured elements in FFmpeg which allow to read the
 multimedia streams from a particular type of file.

 When you configure your FFmpeg build, all the supported demuxers
 are enabled by default. You can list all available ones using the
-configure option @code{--list-demuxers}.
+configure option "--list-demuxers".

 You can disable all the demuxers using the configure option
-@code{--disable-demuxers}, and selectively enable a single demuxer with
-the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
-with the option @code{--disable-demuxer=@var{DEMUXER}}.
+"--disable-demuxers", and selectively enable a single demuxer with
+the option "--enable-demuxer=@var{DEMUXER}", or disable it
+with the option "--disable-demuxer=@var{DEMUXER}".

-The option @code{-formats} of the ff* tools will display the list of
+The option "-formats" of the ff* tools will display the list of
 enabled demuxers.

 The description of some of the currently available demuxers follows.

-@section applehttp
-
-Apple HTTP Live Streaming demuxer.
-
-This demuxer presents all AVStreams from all variant streams.
-The id field is set to the bitrate variant index number. By setting
-the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
-the caller can decide which variant streams to actually receive.
-The total bitrate of the variant that the stream belongs to is
-available in a metadata key named "variant_bitrate".
-
-@section asf
-
-Advanced Systems Format demuxer.
-
-This demuxer is used to demux ASF files and MMS network streams.
-
-@table @option
-@item -no_resync_search @var{bool}
-Do not try to resynchronize by looking for a certain optional start code.
-@end table
-
-@anchor{concat}
-@section concat
-
-Virtual concatenation script demuxer.
-
-This demuxer reads a list of files and other directives from a text file and
-demuxes them one after the other, as if all their packet had been muxed
-together.
-
-The timestamps in the files are adjusted so that the first file starts at 0
-and each next file starts where the previous one finishes. Note that it is
-done globally and may cause gaps if all streams do not have exactly the same
-length.
-
-All files must have the same streams (same codecs, same time base, etc.).
-
-The duration of each file is used to adjust the timestamps of the next file:
-if the duration is incorrect (because it was computed using the bit-rate or
-because the file is truncated, for example), it can cause artifacts. The
-@code{duration} directive can be used to override the duration stored in
-each file.
-
-@subsection Syntax
-
-The script is a text file in extended-ASCII, with one directive per line.
-Empty lines, leading spaces and lines starting with '#' are ignored. The
-following directive is recognized:
-
-@table @option
-
-@item @code{file @var{path}}
-Path to a file to read; special characters and spaces must be escaped with
-backslash or single quotes.
-
-All subsequent directives apply to that file.
-
-@item @code{ffconcat version 1.0}
-Identify the script type and version. It also sets the @option{safe} option
-to 1 if it was to its default -1.
-
-To make FFmpeg recognize the format automatically, this directive must
-appears exactly as is (no extra space or byte-order-mark) on the very first
-line of the script.
-
-@item @code{duration @var{dur}}
-Duration of the file. This information can be specified from the file;
-specifying it here may be more efficient or help if the information from the
-file is not available or accurate.
-
-If the duration is set for all files, then it is possible to seek in the
-whole concatenated video.
-
-@end table
-
-@subsection Options
-
-This demuxer accepts the following option:
-
-@table @option
-
-@item safe
-If set to 1, reject unsafe file paths. A file path is considered safe if it
-does not contain a protocol specification and is relative and all components
-only contain characters from the portable character set (letters, digits,
-period, underscore and hyphen) and have no period at the beginning of a
-component.
-
-If set to 0, any file name is accepted.
-
-The default is -1, it is equivalent to 1 if the format was automatically
-probed and 0 otherwise.
-
-@end table
-
-@section flv
-
-Adobe Flash Video Format demuxer.
-
-This demuxer is used to demux FLV files and RTMP network streams.
-
-@table @option
-@item -flv_metadata @var{bool}
-Allocate the streams according to the onMetaData array content.
-@end table
-
-@section libgme
-
-The Game Music Emu library is a collection of video game music file emulators.
-
-See @url{http://code.google.com/p/game-music-emu/} for more information.
-
-Some files have multiple tracks. The demuxer will pick the first track by
-default. The @option{track_index} option can be used to select a different
-track. Track indexes start at 0. The demuxer exports the number of tracks as
-@var{tracks} meta data entry.
-
-For very large files, the @option{max_size} option may have to be adjusted.
-
-@section libquvi
-
-Play media from Internet services using the quvi project.
-
-The demuxer accepts a @option{format} option to request a specific quality. It
-is by default set to @var{best}.
-
-See @url{http://quvi.sourceforge.net/} for more information.
-
-FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
-enabled.
-
@section image2

 Image file demuxer.
@@ -167,7 +35,7 @@ same for all the files in the sequence.
 This demuxer accepts the following options:
@table @option
@item framerate
-Set the frame rate for the video stream. It defaults to 25.
+Set the framerate for the video stream. It defaults to 25.
@item loop
 If set to 1, loop over the input. Default value is 0.
@item pattern_type
@@ -245,10 +113,6 @@ to read from. Default value is 0.
 Set the index interval range to check when looking for the first image
 file in the sequence, starting from @var{start_number}. Default value
 is 5.
-@item ts_from_file
-If set to 1, will set frame timestamp to modification time of image file. Note
-that monotonity of timestamps is not provided: images go in the same order as
-without this option. Default value is 0.
@item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
@@ -262,64 +126,33 @@ Use @command{ffmpeg} for creating a video from the images in the file
 sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
 input frame rate of 10 frames per second:
@example
-ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
@end example

@item
 As above, but start by reading from a file with index 100 in the sequence:
@example
-ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
@end example

@item
 Read images matching the "*.png" glob pattern , that is all the files
 terminating with the ".png" suffix:
@example
-ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
+ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
@end example
@end itemize

-@section mpegts
+@section applehttp

-MPEG-2 transport stream demuxer.
+Apple HTTP Live Streaming demuxer.

-@table @option
-
-@item fix_teletext_pts
-Overrides teletext packet PTS and DTS values with the timestamps calculated
-from the PCR of the first program which the teletext stream is part of and is
-not discarded. Default value is 1, set this option to 0 if you want your
-teletext packet PTS and DTS values untouched.
-@end table
-
-@section rawvideo
-
-Raw video demuxer.
-
-This demuxer allows to read raw video data. Since there is no header
-specifying the assumed video parameters, the user must specify them
-in order to be able to decode the data correctly.
-
-This demuxer accepts the following options:
-@table @option
-
-@item framerate
-Set input video frame rate. Default value is 25.
-
-@item pixel_format
-Set the input video pixel format. Default value is @code{yuv420p}.
-
-@item video_size
-Set the input video size. This value must be specified explicitly.
-@end table
-
-For example to read a rawvideo file @file{input.raw} with
-@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
-size of @code{320x240}, and a frame rate of 10 images per second, use
-the command:
-@example
-ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
-@end example
+This demuxer presents all AVStreams from all variant streams.
+The id field is set to the bitrate variant index number. By setting
+the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
+the caller can decide which variant streams to actually receive.
+The total bitrate of the variant that the stream belongs to is
+available in a metadata key named "variant_bitrate".

@section sbg

@@ -351,6 +184,37 @@ the script is directly played, the actual times will match the absolute
 timestamps up to the sound controller's clock accuracy, but if the user
 somehow pauses the playback or seeks, all times will be shifted accordingly.

+@section concat
+
+Virtual concatenation script demuxer.
+
+This demuxer reads a list of files and other directives from a text file and
+demuxes them one after the other, as if all their packet had been muxed
+together.
+
+The timestamps in the files are adjusted so that the first file starts at 0
+and each next file starts where the previous one finishes. Note that it is
+done globally and may cause gaps if all streams do not have exactly the same
+length.
+
+All files must have the same streams (same codecs, same time base, etc.).
+
+This script format can currently not be probed, it must be specified explicitly.
+
+@subsection Syntax
+
+The script is a text file in extended-ASCII, with one directive per line.
+Empty lines, leading spaces and lines starting with '#' are ignored. The
+following directive is recognized:
+
+@table @option
+
+@item @code{file @var{path}}
+Path to a file to read; special characters and spaces must be escaped with
+backslash or single quotes.
+
+@end table
+
@section tedcaptions

 JSON captions used for @url{http://www.ted.com/, TED Talks}.
@@ -372,4 +236,4 @@ Example: convert the captions to a format most players understand:
 ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example

-@c man end DEMUXERS
+@c man end INPUT DEVICES
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -11,23 +11,29 @@

@chapter Developers Guide

-@section Notes for external developers
+@section API
+@itemize @bullet
+@item libavcodec is the library containing the codecs (both encoding and
+decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
+it.

-This document is mostly useful for internal FFmpeg developers.
-External developers who need to use the API in their application should
-refer to the API doxygen documentation in the public headers, and
-check the examples in @file{doc/examples} and in the source code to
-see how the public API is employed.
+@item libavformat is the library containing the file format handling (mux and
+demux code for several formats). Look at @file{ffplay.c} to use it in a
+player. See @file{doc/examples/muxing.c} to use it to generate audio or video
+streams.

-You can use the FFmpeg libraries in your commercial program, but you
-are encouraged to @emph{publish any patch you make}. In this case the
-best way to proceed is to send your patches to the ffmpeg-devel
-mailing list following the guidelines illustrated in the remainder of
-this document.
+@end itemize

-For more detailed legal information about the use of FFmpeg in
-external programs read the @file{LICENSE} file in the source tree and
-consult @url{http://ffmpeg.org/legal.html}.
+@section Integrating libavcodec or libavformat in your program
+
+You can integrate all the source code of the libraries to link them
+statically to avoid any version problem. All you need is to provide a
+'config.mak' and a 'config.h' in the parent directory. See the defines
+generated by ./configure to understand what is needed.
+
+You can use libavcodec or libavformat in your commercial program, but
+@emph{any patch you make must be published}. The best way to proceed is
+to send your patches to the FFmpeg mailing list.

@section Contributing

@@ -51,16 +57,13 @@ and should try to fix issues their commit causes.
@subsection Code formatting conventions

 There are the following guidelines regarding the indentation in files:
-
@itemize @bullet
@item
 Indent size is 4.
-
@item
 The TAB character is forbidden outside of Makefiles as is any
 form of trailing whitespace. Commits containing either will be
 rejected by the git repository.
-
@item
 You should try to limit your code lines to 80 characters; however, do so if
 and only if this improves readability.
@@ -114,17 +117,13 @@ int myfunc(int my_parameter)

 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
-
@itemize @bullet
@item
 the @samp{inline} keyword;
-
@item
@samp{//} comments;
-
@item
 designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
-
@item
 compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
@@ -136,72 +135,46 @@ clarity and performance.
 All code must compile with recent versions of GCC and a number of other
 currently supported compilers. To ensure compatibility, please do not use
 additional C99 features or GCC extensions. Especially watch out for:
-
@itemize @bullet
@item
 mixing statements and declarations;
-
@item
@samp{long long} (use @samp{int64_t} instead);
-
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
-
@item
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize

@subsection Naming conventions
-All names should be composed with underscores (_), not CamelCase. For example,
-@samp{avfilter_get_video_buffer} is an acceptable function name and
-@samp{AVFilterGetVideo} is not. The exception from this are type names, like
+All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
+a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
 for example structs and enums; they should always be in the CamelCase

-There are the following conventions for naming variables and functions:

+There are following conventions for naming variables and functions:
@itemize @bullet
@item
 For local variables no prefix is required.
-
@item
-For file-scope variables and functions declared as @code{static}, no prefix
-is required.
-
+For variables and functions declared as @code{static} no prefixes are required.
@item
-For variables and functions visible outside of file scope, but only used
-internally by a library, an @code{ff_} prefix should be used,
-e.g. @samp{ff_w64_demuxer}.
-
+For variables and functions used internally by the library, @code{ff_} prefix
+should be used.
+For example, @samp{ff_w64_demuxer}.
@item
-For variables and functions visible outside of file scope, used internally
-across multiple libraries, use @code{avpriv_} as prefix, for example,
-@samp{avpriv_aac_parse_header}.
-
+For variables and functions used internally across multiple libraries, use
+@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
@item
-Each library has its own prefix for public symbols, in addition to the
-commonly used @code{av_} (@code{avformat_} for libavformat,
-@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
-Check the existing code and choose names accordingly.
-Note that some symbols without these prefixes are also exported for
-retro-compatibility reasons. These exceptions are declared in the
-@code{lib<name>/lib<name>.v} files.
+For exported names, each library has its own prefixes. Just check the existing
+code and name accordingly.
@end itemize

-Furthermore, name space reserved for the system should not be invaded.
-Identifiers ending in @code{_t} are reserved by
-@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
-Also avoid names starting with @code{__} or @code{_} followed by an uppercase
-letter as they are reserved by the C standard. Names starting with @code{_}
-are reserved at the file level and may not be used for externally visible
-symbols. If in doubt, just avoid names starting with @code{_} altogether.
-
@subsection Miscellaneous conventions
-
@itemize @bullet
@item
 fprintf and printf are forbidden in libavformat and libavcodec,
 please use av_log() instead.
-
@item
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
@@ -244,158 +217,125 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:

@enumerate
@item
-Contributions should be licensed under the
-@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
-including an "or any later version" clause, or, if you prefer
-a gift-style license, the
-@uref{http://www.isc.org/software/license/, ISC} or
-@uref{http://mit-license.org/, MIT} license.
-@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
-an "or any later version" clause is also acceptable, but LGPL is
-preferred.
-If you add a new file, give it a proper license header. Do not copy and
-paste it from a random place, use an existing file as template.
-
+   Contributions should be licensed under the LGPL 2.1, including an
+   "or any later version" clause, or the MIT license.  GPL 2 including
+   an "or any later version" clause is also acceptable, but LGPL is
+   preferred.
@item
-You must not commit code which breaks FFmpeg! (Meaning unfinished but
-enabled code which breaks compilation or compiles but does not work or
-breaks the regression tests)
-You can commit unfinished stuff (for testing etc), but it must be disabled
-(#ifdef etc) by default so it does not interfere with other developers'
-work.
-
+   You must not commit code which breaks FFmpeg! (Meaning unfinished but
+   enabled code which breaks compilation or compiles but does not work or
+   breaks the regression tests)
+   You can commit unfinished stuff (for testing etc), but it must be disabled
+   (#ifdef etc) by default so it does not interfere with other developers'
+   work.
@item
-The commit message should have a short first line in the form of
-a @samp{topic: short description} as a header, separated by a newline
-from the body consisting of an explanation of why the change is necessary.
-If the commit fixes a known bug on the bug tracker, the commit message
-should include its bug ID. Referring to the issue on the bug tracker does
-not exempt you from writing an excerpt of the bug in the commit message.
-
+   You do not have to over-test things. If it works for you, and you think it
+   should work for others, then commit. If your code has problems
+   (portability, triggers compiler bugs, unusual environment etc) they will be
+   reported and eventually fixed.
@item
-You do not have to over-test things. If it works for you, and you think it
-should work for others, then commit. If your code has problems
-(portability, triggers compiler bugs, unusual environment etc) they will be
-reported and eventually fixed.
-
+   Do not commit unrelated changes together, split them into self-contained
+   pieces. Also do not forget that if part B depends on part A, but A does not
+   depend on B, then A can and should be committed first and separate from B.
+   Keeping changes well split into self-contained parts makes reviewing and
+   understanding them on the commit log mailing list easier. This also helps
+   in case of debugging later on.
+   Also if you have doubts about splitting or not splitting, do not hesitate to
+   ask/discuss it on the developer mailing list.
@item
-Do not commit unrelated changes together, split them into self-contained
-pieces. Also do not forget that if part B depends on part A, but A does not
-depend on B, then A can and should be committed first and separate from B.
-Keeping changes well split into self-contained parts makes reviewing and
-understanding them on the commit log mailing list easier. This also helps
-in case of debugging later on.
-Also if you have doubts about splitting or not splitting, do not hesitate to
-ask/discuss it on the developer mailing list.
+   Do not change behavior of the programs (renaming options etc) or public
+   API or ABI without first discussing it on the ffmpeg-devel mailing list.
+   Do not remove functionality from the code. Just improve!

+   Note: Redundant code can be removed.
@item
-Do not change behavior of the programs (renaming options etc) or public
-API or ABI without first discussing it on the ffmpeg-devel mailing list.
-Do not remove functionality from the code. Just improve!
-
-Note: Redundant code can be removed.
-
+   Do not commit changes to the build system (Makefiles, configure script)
+   which change behavior, defaults etc, without asking first. The same
+   applies to compiler warning fixes, trivial looking fixes and to code
+   maintained by other developers. We usually have a reason for doing things
+   the way we do. Send your changes as patches to the ffmpeg-devel mailing
+   list, and if the code maintainers say OK, you may commit. This does not
+   apply to files you wrote and/or maintain.
@item
-Do not commit changes to the build system (Makefiles, configure script)
-which change behavior, defaults etc, without asking first. The same
-applies to compiler warning fixes, trivial looking fixes and to code
-maintained by other developers. We usually have a reason for doing things
-the way we do. Send your changes as patches to the ffmpeg-devel mailing
-list, and if the code maintainers say OK, you may commit. This does not
-apply to files you wrote and/or maintain.
+   We refuse source indentation and other cosmetic changes if they are mixed
+   with functional changes, such commits will be rejected and removed. Every
+   developer has his own indentation style, you should not change it. Of course
+   if you (re)write something, you can use your own style, even though we would
+   prefer if the indentation throughout FFmpeg was consistent (Many projects
+   force a given indentation style - we do not.). If you really need to make
+   indentation changes (try to avoid this), separate them strictly from real
+   changes.

+   NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
+   then either do NOT change the indentation of the inner part within (do not
+   move it to the right)! or do so in a separate commit
@item
-We refuse source indentation and other cosmetic changes if they are mixed
-with functional changes, such commits will be rejected and removed. Every
-developer has his own indentation style, you should not change it. Of course
-if you (re)write something, you can use your own style, even though we would
-prefer if the indentation throughout FFmpeg was consistent (Many projects
-force a given indentation style - we do not.). If you really need to make
-indentation changes (try to avoid this), separate them strictly from real
-changes.
-
-NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
-then either do NOT change the indentation of the inner part within (do not
-move it to the right)! or do so in a separate commit
+   Always fill out the commit log message. Describe in a few lines what you
+   changed and why. You can refer to mailing list postings if you fix a
+   particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
+   Recommended format:
+   area changed: Short 1 line description

+   details describing what and why and giving references.
@item
-Always fill out the commit log message. Describe in a few lines what you
-changed and why. You can refer to mailing list postings if you fix a
-particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
-Recommended format:
-area changed: Short 1 line description
-
-details describing what and why and giving references.
-
+   Make sure the author of the commit is set correctly. (see git commit --author)
+   If you apply a patch, send an
+   answer to ffmpeg-devel (or wherever you got the patch from) saying that
+   you applied the patch.
@item
-Make sure the author of the commit is set correctly. (see git commit --author)
-If you apply a patch, send an
-answer to ffmpeg-devel (or wherever you got the patch from) saying that
-you applied the patch.
-
+   When applying patches that have been discussed (at length) on the mailing
+   list, reference the thread in the log message.
@item
-When applying patches that have been discussed (at length) on the mailing
-list, reference the thread in the log message.
-
+    Do NOT commit to code actively maintained by others without permission.
+    Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+    timeframe (12h for build failures and security fixes, 3 days small changes,
+    1 week for big patches) then commit your patch if you think it is OK.
+    Also note, the maintainer can simply ask for more time to review!
@item
-Do NOT commit to code actively maintained by others without permission.
-Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
-timeframe (12h for build failures and security fixes, 3 days small changes,
-1 week for big patches) then commit your patch if you think it is OK.
-Also note, the maintainer can simply ask for more time to review!
-
+    Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+    are sent there and reviewed by all the other developers. Bugs and possible
+    improvements or general questions regarding commits are discussed there. We
+    expect you to react if problems with your code are uncovered.
@item
-Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
-are sent there and reviewed by all the other developers. Bugs and possible
-improvements or general questions regarding commits are discussed there. We
-expect you to react if problems with your code are uncovered.
-
+    Update the documentation if you change behavior or add features. If you are
+    unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+    maintainer(s) will review and commit your stuff.
@item
-Update the documentation if you change behavior or add features. If you are
-unsure how best to do this, send a patch to ffmpeg-devel, the documentation
-maintainer(s) will review and commit your stuff.
-
+    Try to keep important discussions and requests (also) on the public
+    developer mailing list, so that all developers can benefit from them.
@item
-Try to keep important discussions and requests (also) on the public
-developer mailing list, so that all developers can benefit from them.
-
+    Never write to unallocated memory, never write over the end of arrays,
+    always check values read from some untrusted source before using them
+    as array index or other risky things.
@item
-Never write to unallocated memory, never write over the end of arrays,
-always check values read from some untrusted source before using them
-as array index or other risky things.
-
+    Remember to check if you need to bump versions for the specific libav*
+    parts (libavutil, libavcodec, libavformat) you are changing. You need
+    to change the version integer.
+    Incrementing the first component means no backward compatibility to
+    previous versions (e.g. removal of a function from the public API).
+    Incrementing the second component means backward compatible change
+    (e.g. addition of a function to the public API or extension of an
+    existing data structure).
+    Incrementing the third component means a noteworthy binary compatible
+    change (e.g. encoder bug fix that matters for the decoder). The third
+    component always starts at 100 to distinguish FFmpeg from Libav.
@item
-Remember to check if you need to bump versions for the specific libav*
-parts (libavutil, libavcodec, libavformat) you are changing. You need
-to change the version integer.
-Incrementing the first component means no backward compatibility to
-previous versions (e.g. removal of a function from the public API).
-Incrementing the second component means backward compatible change
-(e.g. addition of a function to the public API or extension of an
-existing data structure).
-Incrementing the third component means a noteworthy binary compatible
-change (e.g. encoder bug fix that matters for the decoder). The third
-component always starts at 100 to distinguish FFmpeg from Libav.
-
+    Compiler warnings indicate potential bugs or code with bad style. If a type of
+    warning always points to correct and clean code, that warning should
+    be disabled, not the code changed.
+    Thus the remaining warnings can either be bugs or correct code.
+    If it is a bug, the bug has to be fixed. If it is not, the code should
+    be changed to not generate a warning unless that causes a slowdown
+    or obfuscates the code.
@item
-Compiler warnings indicate potential bugs or code with bad style. If a type of
-warning always points to correct and clean code, that warning should
-be disabled, not the code changed.
-Thus the remaining warnings can either be bugs or correct code.
-If it is a bug, the bug has to be fixed. If it is not, the code should
-be changed to not generate a warning unless that causes a slowdown
-or obfuscates the code.
-
-@item
-Make sure that no parts of the codebase that you maintain are missing from the
-@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
-your name after it.
-If at some point you no longer want to maintain some code, then please help
-finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
+    If you add a new file, give it a proper license header. Do not copy and
+    paste it from a random place, use an existing file as template.
@end enumerate

 We think our rules are not too hard. If you have comments, contact us.

+Note, these rules are mostly borrowed from the MPlayer project.
+
@anchor{Submitting patches}
@section Submitting patches

@@ -418,6 +358,11 @@ The tool is located in the tools directory.
 Run the @ref{Regression tests} before submitting a patch in order to verify
 it does not cause unexpected problems.

+Patches should be posted as base64 encoded attachments (or any other
+encoding which ensures that the patch will not be trashed during
+transmission) to the ffmpeg-devel mailing list, see
+@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
+
 It also helps quite a bit if you tell us what the patch does (for example
 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
 and has no lrint()')
@@ -425,13 +370,6 @@ and has no lrint()')
 Also please if you send several patches, send each patch as a separate mail,
 do not attach several unrelated patches to the same mail.

-Patches should be posted to the
-@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
-mailing list. Use @code{git send-email} when possible since it will properly
-send patches without requiring extra care. If you cannot, then send patches
-as base64-encoded attachments, so your patch is not trashed during
-transmission.
-
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -446,51 +384,40 @@ send a reminder by email. Your patch should eventually be dealt with.

@enumerate
@item
-Did you use av_cold for codec initialization and close functions?
-
+    Did you use av_cold for codec initialization and close functions?
@item
-Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
-AVInputFormat/AVOutputFormat struct?
-
+    Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+    AVInputFormat/AVOutputFormat struct?
@item
-Did you bump the minor version number (and reset the micro version
-number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
-
+    Did you bump the minor version number (and reset the micro version
+    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
-Did you register it in @file{allcodecs.c} or @file{allformats.c}?
-
+    Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
-Did you add the AVCodecID to @file{avcodec.h}?
-When adding new codec IDs, also add an entry to the codec descriptor
-list in @file{libavcodec/codec_desc.c}.
-
+    Did you add the AVCodecID to @file{avcodec.h}?
+    When adding new codec IDs, also add an entry to the codec descriptor
+    list in @file{libavcodec/codec_desc.c}.
@item
-If it has a FourCC, did you add it to @file{libavformat/riff.c},
-even if it is only a decoder?
-
+    If it has a fourCC, did you add it to @file{libavformat/riff.c},
+    even if it is only a decoder?
@item
-Did you add a rule to compile the appropriate files in the Makefile?
-Remember to do this even if you're just adding a format to a file that is
-already being compiled by some other rule, like a raw demuxer.
-
+    Did you add a rule to compile the appropriate files in the Makefile?
+    Remember to do this even if you're just adding a format to a file that is
+    already being compiled by some other rule, like a raw demuxer.
@item
-Did you add an entry to the table of supported formats or codecs in
-@file{doc/general.texi}?
-
+    Did you add an entry to the table of supported formats or codecs in
+    @file{doc/general.texi}?
@item
-Did you add an entry in the Changelog?
-
+    Did you add an entry in the Changelog?
@item
-If it depends on a parser or a library, did you add that dependency in
-configure?
-
+    If it depends on a parser or a library, did you add that dependency in
+    configure?
@item
-Did you @code{git add} the appropriate files before committing?
-
+    Did you @code{git add} the appropriate files before committing?
@item
-Did you make sure it compiles standalone, i.e. with
-@code{configure --disable-everything --enable-decoder=foo}
-(or @code{--enable-demuxer} or whatever your component is)?
+    Did you make sure it compiles standalone, i.e. with
+    @code{configure --disable-everything --enable-decoder=foo}
+    (or @code{--enable-demuxer} or whatever your component is)?
@end enumerate


@@ -498,109 +425,77 @@ Did you make sure it compiles standalone, i.e. with

@enumerate
@item
-Does @code{make fate} pass with the patch applied?
-
+    Does @code{make fate} pass with the patch applied?
@item
-Was the patch generated with git format-patch or send-email?
-
+    Was the patch generated with git format-patch or send-email?
@item
-Did you sign off your patch? (git commit -s)
-See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
-of sign off.
-
+    Did you sign off your patch? (git commit -s)
+    See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+    of sign off.
@item
-Did you provide a clear git commit log message?
-
+    Did you provide a clear git commit log message?
@item
-Is the patch against latest FFmpeg git master branch?
-
+    Is the patch against latest FFmpeg git master branch?
@item
-Are you subscribed to ffmpeg-devel?
-(the list is subscribers only due to spam)
-
+    Are you subscribed to ffmpeg-devel?
+    (the list is subscribers only due to spam)
@item
-Have you checked that the changes are minimal, so that the same cannot be
-achieved with a smaller patch and/or simpler final code?
-
+    Have you checked that the changes are minimal, so that the same cannot be
+    achieved with a smaller patch and/or simpler final code?
@item
-If the change is to speed critical code, did you benchmark it?
-
+    If the change is to speed critical code, did you benchmark it?
@item
-If you did any benchmarks, did you provide them in the mail?
-
+    If you did any benchmarks, did you provide them in the mail?
@item
-Have you checked that the patch does not introduce buffer overflows or
-other security issues?
-
+    Have you checked that the patch does not introduce buffer overflows or
+    other security issues?
@item
-Did you test your decoder or demuxer against damaged data? If no, see
-tools/trasher, the noise bitstream filter, and
-@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
-should not crash, end in a (near) infinite loop, or allocate ridiculous
-amounts of memory when fed damaged data.
-
+    Did you test your decoder or demuxer against damaged data? If no, see
+    tools/trasher and the noise bitstream filter. Your decoder or demuxer
+    should not crash or end in a (near) infinite loop when fed damaged data.
@item
-Does the patch not mix functional and cosmetic changes?
-
+    Does the patch not mix functional and cosmetic changes?
@item
-Did you add tabs or trailing whitespace to the code? Both are forbidden.
-
+    Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
-Is the patch attached to the email you send?
-
+    Is the patch attached to the email you send?
@item
-Is the mime type of the patch correct? It should be text/x-diff or
-text/x-patch or at least text/plain and not application/octet-stream.
-
+    Is the mime type of the patch correct? It should be text/x-diff or
+    text/x-patch or at least text/plain and not application/octet-stream.
@item
-If the patch fixes a bug, did you provide a verbose analysis of the bug?
-
+    If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
-If the patch fixes a bug, did you provide enough information, including
-a sample, so the bug can be reproduced and the fix can be verified?
-Note please do not attach samples >100k to mails but rather provide a
-URL, you can upload to ftp://upload.ffmpeg.org
-
+    If the patch fixes a bug, did you provide enough information, including
+    a sample, so the bug can be reproduced and the fix can be verified?
+    Note please do not attach samples >100k to mails but rather provide a
+    URL, you can upload to ftp://upload.ffmpeg.org
@item
-Did you provide a verbose summary about what the patch does change?
-
+    Did you provide a verbose summary about what the patch does change?
@item
-Did you provide a verbose explanation why it changes things like it does?
-
+    Did you provide a verbose explanation why it changes things like it does?
@item
-Did you provide a verbose summary of the user visible advantages and
-disadvantages if the patch is applied?
-
+    Did you provide a verbose summary of the user visible advantages and
+    disadvantages if the patch is applied?
@item
-Did you provide an example so we can verify the new feature added by the
-patch easily?
-
+    Did you provide an example so we can verify the new feature added by the
+    patch easily?
@item
-If you added a new file, did you insert a license header? It should be
-taken from FFmpeg, not randomly copied and pasted from somewhere else.
-
+    If you added a new file, did you insert a license header? It should be
+    taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
-You should maintain alphabetical order in alphabetically ordered lists as
-long as doing so does not break API/ABI compatibility.
-
+    You should maintain alphabetical order in alphabetically ordered lists as
+    long as doing so does not break API/ABI compatibility.
@item
-Lines with similar content should be aligned vertically when doing so
-improves readability.
-
+    Lines with similar content should be aligned vertically when doing so
+    improves readability.
@item
-Consider to add a regression test for your code.
-
+    Consider to add a regression test for your code.
@item
-If you added YASM code please check that things still work with --disable-yasm
-
+    If you added YASM code please check that things still work with --disable-yasm
@item
-Make sure you check the return values of function and return appropriate
-error codes. Especially memory allocation functions like @code{av_malloc()}
-are notoriously left unchecked, which is a serious problem.
-
-@item
-Test your code with valgrind and or Address Sanitizer to ensure it's free
-of leaks, out of array accesses, etc.
+    Make sure you check the return values of function and return appropriate
+    error codes. Especially memory allocation functions like @code{av_malloc()}
+    are notoriously left unchecked, which is a serious problem.
@end enumerate

@section Patch review process
@@ -653,145 +548,4 @@ message or introductionary message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.


-@subsection Visualizing Test Coverage
-
-The FFmpeg build system allows visualizing the test coverage in an easy
-manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
-the following steps:
-
-@enumerate
-@item
-    Configure to compile with instrumentation enabled:
-    @code{configure --toolchain=gcov}.
-
-@item
-    Run your test case, either manually or via FATE. This can be either
-    the full FATE regression suite, or any arbitrary invocation of any
-    front-end tool provided by FFmpeg, in any combination.
-
-@item
-    Run @code{make lcov} to generate coverage data in HTML format.
-
-@item
-    View @code{lcov/index.html} in your preferred HTML viewer.
-@end enumerate
-
-You can use the command @code{make lcov-reset} to reset the coverage
-measurements. You will need to rerun @code{make lcov} after running a
-new test.
-
-@subsection Using Valgrind
-
-The configure script provides a shortcut for using valgrind to spot bugs
-related to memory handling. Just add the option
-@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
-to your configure line, and reasonable defaults will be set for running
-FATE under the supervision of either the @strong{memcheck} or the
-@strong{massif} tool of the valgrind suite.
-
-In case you need finer control over how valgrind is invoked, use the
-@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
-your configure line instead.
-
-@anchor{Release process}
-@section Release process
-
-FFmpeg maintains a set of @strong{release branches}, which are the
-recommended deliverable for system integrators and distributors (such as
-Linux distributions, etc.). At regular times, a @strong{release
-manager} prepares, tests and publishes tarballs on the
-@url{http://ffmpeg.org} website.
-
-There are two kinds of releases:
-
-@enumerate
-@item
-@strong{Major releases} always include the latest and greatest
-features and functionality.
-
-@item
-@strong{Point releases} are cut from @strong{release} branches,
-which are named @code{release/X}, with @code{X} being the release
-version number.
-@end enumerate
-
-Note that we promise to our users that shared libraries from any FFmpeg
-release never break programs that have been @strong{compiled} against
-previous versions of @strong{the same release series} in any case!
-
-However, from time to time, we do make API changes that require adaptations
-in applications. Such changes are only allowed in (new) major releases and
-require further steps such as bumping library version numbers and/or
-adjustments to the symbol versioning file. Please discuss such changes
-on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
-
-@anchor{Criteria for Point Releases}
-@subsection Criteria for Point Releases
-
-Changes that match the following criteria are valid candidates for
-inclusion into a point release:
-
-@enumerate
-@item
-Fixes a security issue, preferably identified by a @strong{CVE
-number} issued by @url{http://cve.mitre.org/}.
-
-@item
-Fixes a documented bug in @url{https://trac.ffmpeg.org}.
-
-@item
-Improves the included documentation.
-
-@item
-Retains both source code and binary compatibility with previous
-point releases of the same release branch.
-@end enumerate
-
-The order for checking the rules is (1 OR 2 OR 3) AND 4.
-
-
-@subsection Release Checklist
-
-The release process involves the following steps:
-
-@enumerate
-@item
-Ensure that the @file{RELEASE} file contains the version number for
-the upcoming release.
-
-@item
-Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
-
-@item
-Announce the intent to do a release to the mailing list.
-
-@item
-Make sure all relevant security fixes have been backported. See
-@url{https://ffmpeg.org/security.html}.
-
-@item
-Ensure that the FATE regression suite still passes in the release
-branch on at least @strong{i386} and @strong{amd64}
-(cf. @ref{Regression tests}).
-
-@item
-Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
-supplementing files that contain @code{gpg} signatures
-
-@item
-Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
-push an annotated tag in the form @code{nX}, with @code{X}
-containing the version number.
-
-@item
-Propose and send a patch to the @strong{ffmpeg-devel} mailing list
-with a news entry for the website.
-
-@item
-Publish the news entry.
-
-@item
-Send announcement to the mailing list.
-@end enumerate
-
@bye
--- a/doc/devices.texi
+++ b/doc/devices.texi
@@ -1,21 +0,0 @@
-@chapter Device Options
-@c man begin DEVICE OPTIONS
-
-The libavdevice library provides the same interface as
-libavformat. Namely, an input device is considered like a demuxer, and
-an output device like a muxer, and the interface and generic device
-options are the same provided by libavformat (see the ffmpeg-formats
-manual).
-
-In addition each input or output device may support so-called private
-options, which are specific for that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the device
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-@c man end DEVICE OPTIONS
-
-@include indevs.texi
-@include outdevs.texi
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -71,7 +71,7 @@ Force middle/side encoding.
 Set AAC encoder coding method. Possible values:

@table @samp
-@item faac
+@item 0
 FAAC-inspired method.

 This method is a simplified reimplementation of the method used in FAAC, which
@@ -82,13 +82,13 @@ distortion below threshold band by band.
 The quality of this method is comparable to the two loop searching method
 descibed below, but somewhat a little better and slower.

-@item anmr
+@item 1
 Average noise to mask ratio (ANMR) trellis-based solution.

 This has a theoretic best quality out of all the coding methods, but at the
 cost of the slowest speed.

-@item twoloop
+@item 2
 Two loop searching (TLS) method.

 This method first sets quantizers depending on band thresholds and then tries
@@ -97,7 +97,7 @@ all quantizers and adjusting some individual quantizer a little.

 This method produces similar quality with the FAAC method and is the default.

-@item fast
+@item 3
 Constant quantizer method.

 This method sets a constant quantizer for all bands. This is the fastest of all
@@ -107,6 +107,13 @@ the methods, yet produces the worst quality.

@end table

+@subsection Tips and Tricks
+
+According to some reports
+(e.g. @url{http://d.hatena.ne.jp/kamedo2/20120729/1343545890}), setting the
+@option{cutoff} option to 15000 Hz greatly improves the quality of the output
+quality. As a result, we encourage you to do the same.
+
@section ac3 and ac3_fixed

 AC-3 audio encoders.
@@ -494,18 +501,14 @@ Selected by Encoder (default)

@end table

-@anchor{libmp3lame}
@section libmp3lame

-LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.
+LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper

 Requires the presence of the libmp3lame headers and library during
 configuration. You need to explicitly configure the build with
@code{--enable-libmp3lame}.

-See @ref{libshine} for a fixed-point MP3 encoder, although with a
-lower quality.
-
@subsection Options

 The following options are supported by the libmp3lame wrapper. The
@@ -531,10 +534,6 @@ Enable use of bit reservoir when set to 1. Default value is 1. LAME
 has this enabled by default, but can be overriden by use
@option{--nores} option.

-@item joint_stereo (@emph{-m j})
-Enable the encoder to use (on a frame by frame basis) either L/R
-stereo or mid/side stereo. Default value is 1.
-
@end table

@section libopencore-amrnb
@@ -574,45 +573,9 @@ default value is 0 (disabled).

@end table

-@anchor{libshine}
-@section libshine
-
-Shine Fixed-Point MP3 encoder wrapper.
-
-Shine is a fixed-point MP3 encoder. It has a far better performance on
-platforms without an FPU, e.g. armel CPUs, and some phones and tablets.
-However, as it is more targeted on performance than quality, it is not on par
-with LAME and other production-grade encoders quality-wise. Also, according to
-the project's homepage, this encoder may not be free of bugs as the code was
-written a long time ago and the project was dead for at least 5 years.
-
-This encoder only supports stereo and mono input. This is also CBR-only.
-
-The original project (last updated in early 2007) is at
-@url{http://sourceforge.net/projects/libshine-fxp/}. We only support the
-updated fork by the Savonet/Liquidsoap project at @url{https://github.com/savonet/shine}.
-
-Requires the presence of the libshine headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libshine}.
-
-See also @ref{libmp3lame}.
-
-@subsection Options
-
-The following options are supported by the libshine wrapper. The
-@command{shineenc}-equivalent of the options are listed in parentheses.
-
-@table @option
-@item b (@emph{-b})
-Set bitrate expressed in bits/s for CBR. @command{shineenc} @option{-b} option
-is expressed in kilobits/s.
-
-@end table
-
@section libtwolame

-TwoLAME MP2 encoder wrapper.
+TwoLAME MP2 encoder wrapper

 Requires the presence of the libtwolame headers and library during
 configuration. You need to explicitly configure the build with
@@ -678,7 +641,7 @@ Set MPEG audio original flag when set to 1. The default value is 0
@anchor{libvo-aacenc}
@section libvo-aacenc

-VisualOn AAC encoder.
+VisualOn AAC encoder

 Requires the presence of the libvo-aacenc headers and library during
 configuration. You need to explicitly configure the build with
@@ -702,7 +665,7 @@ Set bit rate in bits/s.

@section libvo-amrwbenc

-VisualOn Adaptive Multi-Rate Wideband encoder.
+VisualOn Adaptive Multi-Rate Wideband encoder

 Requires the presence of the libvo-amrwbenc headers and library during
 configuration. You need to explicitly configure the build with
@@ -811,87 +774,6 @@ respectively. The default is 0 (cutoff disabled).

@end table

-@section libvorbis
-
-libvorbis encoder wrapper.
-
-Requires the presence of the libvorbisenc headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libvorbis}.
-
-@subsection Options
-
-The following options are supported by the libvorbis wrapper. The
-@command{oggenc}-equivalent of the options are listed in parentheses.
-
-To get a more accurate and extensive documentation of the libvorbis
-options, consult the libvorbisenc's and @command{oggenc}'s documentations.
-See @url{http://xiph.org/vorbis/},
-@url{http://wiki.xiph.org/Vorbis-tools}, and oggenc(1).
-
-@table @option
-@item b (@emph{-b})
-Set bitrate expressed in bits/s for ABR. @command{oggenc} @option{-b} is
-expressed in kilobits/s.
-
-@item q (@emph{-q})
-Set constant quality setting for VBR. The value should be a float
-number in the range of -1.0 to 10.0. The higher the value, the better
-the quality. The default value is @samp{3.0}.
-
-This option is valid only using the @command{ffmpeg} command-line tool.
-For library interface users, use @option{global_quality}.
-
-@item cutoff (@emph{--advanced-encode-option lowpass_frequency=N})
-Set cutoff bandwidth in Hz, a value of 0 disables cutoff. @command{oggenc}'s
-related option is expressed in kHz. The default value is @samp{0} (cutoff
-disabled).
-
-@item minrate (@emph{-m})
-Set minimum bitrate expressed in bits/s. @command{oggenc} @option{-m} is
-expressed in kilobits/s.
-
-@item maxrate (@emph{-M})
-Set maximum bitrate expressed in bits/s. @command{oggenc} @option{-M} is
-expressed in kilobits/s. This only has effect on ABR mode.
-
-@item iblock (@emph{--advanced-encode-option impulse_noisetune=N})
-Set noise floor bias for impulse blocks. The value is a float number from
-15.0 to 0.0. A negative bias instructs the encoder to pay special attention
-to the crispness of transients in the encoded audio. The tradeoff for better
-transient response is a higher bitrate.
-
-@end table
-
-@section libwavpack
-
-A wrapper providing WavPack encoding through libwavpack.
-
-Only lossless mode using 32-bit integer samples is supported currently.
-The @option{compression_level} option can be used to control speed vs.
-compression tradeoff, with the values mapped to libwavpack as follows:
-
-@table @option
-
-@item 0
-Fast mode - corresponding to the wavpack @option{-f} option.
-
-@item 1
-Normal (default) settings.
-
-@item 2
-High quality - corresponding to the wavpack @option{-h} option.
-
-@item 3
-Very high quality - corresponding to the wavpack @option{-hh} option.
-
-@item 4-8
-Same as 3, but with extra processing enabled - corresponding to the wavpack
-@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as
-@option{-x6}.
-
-@end table
-
@c man end AUDIO ENCODERS

@chapter Video Encoders
@@ -1061,350 +943,57 @@ For more information about libvpx see:

@section libx264

-x264 H.264/MPEG-4 AVC encoder wrapper.
+H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 format supported through
+libx264.

-This encoder requires the presence of the libx264 headers and library
-during configuration. You need to explicitly configure the build with
+Requires the presence of the libx264 headers and library during
+configuration. You need to explicitly configure the build with
@code{--enable-libx264}.

-libx264 supports an impressive number of features, including 8x8 and
-4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC
-entropy coding, interlacing (MBAFF), lossless mode, psy optimizations
-for detail retention (adaptive quantization, psy-RD, psy-trellis).
-
-Many libx264 encoder options are mapped to FFmpeg global codec
-options, while unique encoder options are provided through private
-options. Additionally the @option{x264opts} and @option{x264-params}
-private options allows to pass a list of key=value tuples as accepted
-by the libx264 @code{x264_param_parse} function.
-
-The x264 project website is at
-@url{http://www.videolan.org/developers/x264.html}.
-
@subsection Options

-The following options are supported by the libx264 wrapper. The
-@command{x264}-equivalent options or values are listed in parentheses
-for easy migration.
-
-To reduce the duplication of documentation, only the private options
-and some others requiring special attention are documented here. For
-the documentation of the undocumented generic options, see
-@ref{codec-options,,the Codec Options chapter}.
-
-To get a more accurate and extensive documentation of the libx264
-options, invoke the command @command{x264 --full-help} or consult
-the libx264 documentation.
-
@table @option
-@item b (@emph{bitrate})
-Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
-expressed in bits/s, while @command{x264}'s @option{bitrate} is in
-kilobits/s.

-@item bf (@emph{bframes})
-
-@item g (@emph{keyint})
-
-@item qmax (@emph{qpmax})
-
-@item qmin (@emph{qpmin})
-
-@item qdiff (@emph{qpstep})
-
-@item qblur (@emph{qblur})
-
-@item qcomp (@emph{qcomp})
-
-@item refs (@emph{ref})
-
-@item sc_threshold (@emph{scenecut})
-
-@item trellis (@emph{trellis})
-
-@item nr  (@emph{nr})
-
-@item me_range (@emph{merange})
-
-@item me_method (@emph{me})
-Set motion estimation method. Possible values in the decreasing order
-of speed:
-
-@table @samp
-@item dia (@emph{dia})
-@item epzs (@emph{dia})
-Diamond search with radius 1 (fastest). @samp{epzs} is an alias for
-@samp{dia}.
-@item hex (@emph{hex})
-Hexagonal search with radius 2.
-@item umh (@emph{umh})
-Uneven multi-hexagon search.
-@item esa (@emph{esa})
-Exhaustive search.
-@item tesa (@emph{tesa})
-Hadamard exhaustive search (slowest).
-@end table
-
-@item subq (@emph{subme})
-
-@item b_strategy (@emph{b-adapt})
-
-@item keyint_min (@emph{min-keyint})
-
-@item coder
-Set entropy encoder. Possible values:
-
-@table @samp
-@item ac
-Enable CABAC.
-
-@item vlc
-Enable CAVLC and disable CABAC. It generates the same effect as
-@command{x264}'s @option{--no-cabac} option.
-@end table
-
-@item cmp
-Set full pixel motion estimation comparation algorithm. Possible values:
-
-@table @samp
-@item chroma
-Enable chroma in motion estimation.
-
-@item sad
-Ignore chroma in motion estimation. It generates the same effect as
-@command{x264}'s @option{--no-chroma-me} option.
-@end table
-
-@item threads (@emph{threads})
-
-@item thread_type
-Set multithreading technique. Possible values:
-
-@table @samp
-@item slice
-Slice-based multithreading. It generates the same effect as
-@command{x264}'s @option{--sliced-threads} option.
-@item frame
-Frame-based multithreading.
-@end table
-
-@item flags
-Set encoding flags. It can be used to disable closed GOP and enable
-open GOP by setting it to @code{-cgop}. The result is similar to
-the behavior of @command{x264}'s @option{--open-gop} option.
-
-@item rc_init_occupancy (@emph{vbv-init})
-
-@item preset (@emph{preset})
+@item preset @var{preset_name}
 Set the encoding preset.

-@item tune (@emph{tune})
-Set tuning of the encoding params.
+@item tune @var{tune_name}
+Tune the encoding params.

-@item profile (@emph{profile})
+@item fastfirstpass @var{bool}
+Use fast settings when encoding first pass, default value is 1.
+
+@item profile @var{profile_name}
 Set profile restrictions.

-@item fastfirstpass
-Enable fast settings when encoding first pass, when set to 1. When set
-to 0, it has the same effect of @command{x264}'s
-@option{--slow-firstpass} option.
+@item level @var{level}
+Specify level (as defined by Annex A).
+Deprecated in favor of @var{x264opts}.

-@item crf (@emph{crf})
-Set the quality for constant quality mode.
+@item passlogfile @var{filename}
+Specify filename for 2 pass stats.
+Deprecated in favor of @var{x264opts} (see @var{stats} libx264 option).

-@item crf_max (@emph{crf-max})
-In CRF mode, prevents VBV from lowering quality beyond this point.
+@item wpredp @var{wpred_type}
+Specify Weighted prediction for P-frames.
+Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).

-@item qp (@emph{qp})
-Set constant quantization rate control method parameter.
+@item x264opts @var{options}
+Allow to set any x264 option, see @code{x264 --fullhelp} for a list.

-@item aq-mode (@emph{aq-mode})
-Set AQ method. Possible values:
-
-@table @samp
-@item none (@emph{0})
-Disabled.
-
-@item variance (@emph{1})
-Variance AQ (complexity mask).
-
-@item autovariance (@emph{2})
-Auto-variance AQ (experimental).
-@end table
-
-@item aq-strength (@emph{aq-strength})
-Set AQ strength, reduce blocking and blurring in flat and textured areas.
-
-@item psy
-Use psychovisual optimizations when set to 1. When set to 0, it has the
-same effect as @command{x264}'s @option{--no-psy} option.
-
-@item psy-rd  (@emph{psy-rd})
-Set strength of psychovisual optimization, in
-@var{psy-rd}:@var{psy-trellis} format.
-
-@item rc-lookahead (@emph{rc-lookahead})
-Set number of frames to look ahead for frametype and ratecontrol.
-
-@item weightb
-Enable weighted prediction for B-frames when set to 1. When set to 0,
-it has the same effect as @command{x264}'s @option{--no-weightb} option.
-
-@item weightp (@emph{weightp})
-Set weighted prediction method for P-frames. Possible values:
-
-@table @samp
-@item none (@emph{0})
-Disabled
-@item simple (@emph{1})
-Enable only weighted refs
-@item smart (@emph{2})
-Enable both weighted refs and duplicates
-@end table
-
-@item ssim (@emph{ssim})
-Enable calculation and printing SSIM stats after the encoding.
-
-@item intra-refresh (@emph{intra-refresh})
-Enable the use of Periodic Intra Refresh instead of IDR frames when set
-to 1.
-
-@item bluray-compat (@emph{bluray-compat})
-Configure the encoder to be compatible with the bluray standard.
-It is a shorthand for setting "bluray-compat=1 force-cfr=1".
-
-@item b-bias (@emph{b-bias})
-Set the influence on how often B-frames are used.
-
-@item b-pyramid (@emph{b-pyramid})
-Set method for keeping of some B-frames as references. Possible values:
-
-@table @samp
-@item none (@emph{none})
-Disabled.
-@item strict (@emph{strict})
-Strictly hierarchical pyramid.
-@item normal (@emph{normal})
-Non-strict (not Blu-ray compatible).
-@end table
-
-@item mixed-refs
-Enable the use of one reference per partition, as opposed to one
-reference per macroblock when set to 1. When set to 0, it has the
-same effect as @command{x264}'s @option{--no-mixed-refs} option.
-
-@item 8x8dct
-Enable adaptive spatial transform (high profile 8x8 transform)
-when set to 1. When set to 0, it has the same effect as
-@command{x264}'s @option{--no-8x8dct} option.
-
-@item fast-pskip
-Enable early SKIP detection on P-frames when set to 1. When set
-to 0, it has the same effect as @command{x264}'s
-@option{--no-fast-pskip} option.
-
-@item aud (@emph{aud})
-Enable use of access unit delimiters when set to 1.
-
-@item mbtree
-Enable use macroblock tree ratecontrol when set to 1. When set
-to 0, it has the same effect as @command{x264}'s
-@option{--no-mbtree} option.
-
-@item deblock (@emph{deblock})
-Set loop filter parameters, in @var{alpha}:@var{beta} form.
-
-@item cplxblur (@emph{cplxblur})
-Set fluctuations reduction in QP (before curve compression).
-
-@item partitions (@emph{partitions})
-Set partitions to consider as a comma-separated list of. Possible
-values in the list:
-
-@table @samp
-@item p8x8
-8x8 P-frame partition.
-@item p4x4
-4x4 P-frame partition.
-@item b8x8
-4x4 B-frame partition.
-@item i8x8
-8x8 I-frame partition.
-@item i4x4
-4x4 I-frame partition.
-(Enabling @samp{p4x4} requires @samp{p8x8} to be enabled. Enabling
-@samp{i8x8} requires adaptive spatial transform (@option{8x8dct}
-option) to be enabled.)
-@item none (@emph{none})
-Do not consider any partitions.
-@item all (@emph{all})
-Consider every partition.
-@end table
-
-@item direct-pred (@emph{direct})
-Set direct MV prediction mode. Possible values:
-
-@table @samp
-@item none (@emph{none})
-Disable MV prediction.
-@item spatial (@emph{spatial})
-Enable spatial predicting.
-@item temporal (@emph{temporal})
-Enable temporal predicting.
-@item auto (@emph{auto})
-Automatically decided.
-@end table
-
-@item slice-max-size (@emph{slice-max-size})
-Set the limit of the size of each slice in bytes. If not specified
-but RTP payload size (@option{ps}) is specified, that is used.
-
-@item stats (@emph{stats})
-Set the file name for multi-pass stats.
-
-@item nal-hrd (@emph{nal-hrd})
-Set signal HRD information (requires @option{vbv-bufsize} to be set).
-Possible values:
-
-@table @samp
-@item none (@emph{none})
-Disable HRD information signaling.
-@item vbr (@emph{vbr})
-Variable bit rate.
-@item cbr (@emph{cbr})
-Constant bit rate (not allowed in MP4 container).
-@end table
-
-@item x264opts (N.A.)
-Set any x264 option, see @command{x264 --fullhelp} for a list.
-
-Argument is a list of @var{key}=@var{value} couples separated by
+@var{options} is a list of @var{key}=@var{value} couples separated by
 ":". In @var{filter} and @var{psy-rd} options that use ":" as a separator
 themselves, use "," instead. They accept it as well since long ago but this
 is kept undocumented for some reason.
+@end table

 For example to specify libx264 encoding options with @command{ffmpeg}:
@example
 ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example

-@item x264-params (N.A.)
-Override the x264 configuration using a :-separated list of key=value
-parameters.
-
-This option is functionally the same as the @option{x264opts}, but is
-duplicated for compability with the Libav fork.
-
-For example to specify libx264 encoding options with @command{ffmpeg}:
-@example
-ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
-cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
-no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
-@end example
-@end table
-
-Encoding ffpresets for common usages are provided so they can be used with the
-general presets system (e.g. passing the @option{pre} option).
+For more information about libx264 and the supported options see:
+@url{http://www.videolan.org/developers/x264.html}

@section libxvid

@@ -1517,136 +1106,6 @@ half pixel and quarter pixel refinement for 8x8 blocks, and rate
 distortion-based search using square pattern.
@end table

-@item lumi_aq
-Enable lumi masking adaptive quantization when set to 1. Default is 0
-(disabled).
-
-@item variance_aq
-Enable variance adaptive quantization when set to 1. Default is 0
-(disabled).
-
-When combined with @option{lumi_aq}, the resulting quality will not
-be better than any of the two specified individually. In other
-words, the resulting quality will be the worse one of the two
-effects.
-
-@item ssim
-Set structural similarity (SSIM) displaying method. Possible values:
-
-@table @samp
-@item off
-Disable displaying of SSIM information.
-
-@item avg
-Output average SSIM at the end of encoding to stdout. The format of
-showing the average SSIM is:
-
-@example
-Average SSIM: %f
-@end example
-
-For users who are not familiar with C, %f means a float number, or
-a decimal (e.g. 0.939232).
-
-@item frame
-Output both per-frame SSIM data during encoding and average SSIM at
-the end of encoding to stdout. The format of per-frame information
-is:
-
-@example
-       SSIM: avg: %1.3f min: %1.3f max: %1.3f
-@end example
-
-For users who are not familiar with C, %1.3f means a float number
-rounded to 3 digits after the dot (e.g. 0.932).
-
@end table

-@item ssim_acc
-Set SSIM accuracy. Valid options are integers within the range of
-0-4, while 0 gives the most accurate result and 4 computes the
-fastest.
-
-@end table
-
-@section png
-
-PNG image encoder.
-
-@subsection Private options
-
-@table @option
-@item dpi @var{integer}
-Set physical density of pixels, in dots per inch, unset by default
-@item dpm @var{integer}
-Set physical density of pixels, in dots per meter, unset by default
-@end table
-
-@section ProRes
-
-Apple ProRes encoder.
-
-FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
-The used encoder can be choosen with the @code{-vcodec} option.
-
-@subsection Private Options for prores-ks
-
-@table @option
-@item profile @var{integer}
-Select the ProRes profile to encode
-@table @samp
-@item proxy
-@item lt
-@item standard
-@item hq
-@item 4444
-@end table
-
-@item quant_mat @var{integer}
-Select quantization matrix.
-@table @samp
-@item auto
-@item default
-@item proxy
-@item lt
-@item standard
-@item hq
-@end table
-If set to @var{auto}, the matrix matching the profile will be picked.
-If not set, the matrix providing the highest quality, @var{default}, will be
-picked.
-
-@item bits_per_mb @var{integer}
-How many bits to allot for coding one macroblock. Different profiles use
-between 200 and 2400 bits per macroblock, the maximum is 8000.
-
-@item mbs_per_slice @var{integer}
-Number of macroblocks in each slice (1-8); the default value (8)
-should be good in almost all situations.
-
-@item vendor @var{string}
-Override the 4-byte vendor ID.
-A custom vendor ID like @var{apl0} would claim the stream was produced by
-the Apple encoder.
-
-@item alpha_bits @var{integer}
-Specify number of bits for alpha component.
-Possible values are @var{0}, @var{8} and @var{16}.
-Use @var{0} to disable alpha plane coding.
-
-@end table
-
-@subsection Speed considerations
-
-In the default mode of operation the encoder has to honor frame constraints
-(i.e. not produc frames with size bigger than requested) while still making
-output picture as good as possible.
-A frame containing a lot of small details is harder to compress and the encoder
-would spend more time searching for appropriate quantizers for each slice.
-
-Setting a higher @option{bits_per_mb} limit will improve the speed.
-
-For the fastest encoding speed set the @option{qscale} parameter (4 is the
-recommended value) and do not set a size constraint.
-
@c man end VIDEO ENCODERS
--- a/doc/eval.texi
+++ b/doc/eval.texi
@@ -0,0 +1,252 @@
+@chapter Expression Evaluation
+@c man begin EXPRESSION EVALUATION
+
+When evaluating an arithmetic expression, FFmpeg uses an internal
+formula evaluator, implemented through the @file{libavutil/eval.h}
+interface.
+
+An expression may contain unary, binary operators, constants, and
+functions.
+
+Two expressions @var{expr1} and @var{expr2} can be combined to form
+another expression "@var{expr1};@var{expr2}".
+@var{expr1} and @var{expr2} are evaluated in turn, and the new
+expression evaluates to the value of @var{expr2}.
+
+The following binary operators are available: @code{+}, @code{-},
+@code{*}, @code{/}, @code{^}.
+
+The following unary operators are available: @code{+}, @code{-}.
+
+The following functions are available:
+@table @option
+@item sinh(x)
+Compute hyperbolic sine of @var{x}.
+
+@item cosh(x)
+Compute hyperbolic cosine of @var{x}.
+
+@item tanh(x)
+Compute hyperbolic tangent of @var{x}.
+
+@item sin(x)
+Compute sine of @var{x}.
+
+@item cos(x)
+Compute cosine of @var{x}.
+
+@item tan(x)
+Compute tangent of @var{x}.
+
+@item atan(x)
+Compute arctangent of @var{x}.
+
+@item asin(x)
+Compute arcsine of @var{x}.
+
+@item acos(x)
+Compute arccosine of @var{x}.
+
+@item exp(x)
+Compute exponential of @var{x} (with base @code{e}, the Euler's number).
+
+@item log(x)
+Compute natural logarithm of @var{x}.
+
+@item abs(x)
+Compute absolute value of @var{x}.
+
+@item squish(x)
+Compute expression @code{1/(1 + exp(4*x))}.
+
+@item gauss(x)
+Compute Gauss function of @var{x}, corresponding to
+@code{exp(-x*x/2) / sqrt(2*PI)}.
+
+@item isinf(x)
+Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
+
+@item isnan(x)
+Return 1.0 if @var{x} is NAN, 0.0 otherwise.
+
+@item mod(x, y)
+Compute the remainder of division of @var{x} by @var{y}.
+
+@item max(x, y)
+Return the maximum between @var{x} and @var{y}.
+
+@item min(x, y)
+Return the maximum between @var{x} and @var{y}.
+
+@item eq(x, y)
+Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
+
+@item gte(x, y)
+Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
+
+@item gt(x, y)
+Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
+
+@item lte(x, y)
+Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
+
+@item lt(x, y)
+Return 1 if @var{x} is lesser than @var{y}, 0 otherwise.
+
+@item st(var, expr)
+Allow to store the value of the expression @var{expr} in an internal
+variable. @var{var} specifies the number of the variable where to
+store the value, and it is a value ranging from 0 to 9. The function
+returns the value stored in the internal variable.
+Note, Variables are currently not shared between expressions.
+
+@item ld(var)
+Allow to load the value of the internal variable with number
+@var{var}, which was previously stored with st(@var{var}, @var{expr}).
+The function returns the loaded value.
+
+@item while(cond, expr)
+Evaluate expression @var{expr} while the expression @var{cond} is
+non-zero, and returns the value of the last @var{expr} evaluation, or
+NAN if @var{cond} was always false.
+
+@item ceil(expr)
+Round the value of expression @var{expr} upwards to the nearest
+integer. For example, "ceil(1.5)" is "2.0".
+
+@item floor(expr)
+Round the value of expression @var{expr} downwards to the nearest
+integer. For example, "floor(-1.5)" is "-2.0".
+
+@item trunc(expr)
+Round the value of expression @var{expr} towards zero to the nearest
+integer. For example, "trunc(-1.5)" is "-1.0".
+
+@item sqrt(expr)
+Compute the square root of @var{expr}. This is equivalent to
+"(@var{expr})^.5".
+
+@item not(expr)
+Return 1.0 if @var{expr} is zero, 0.0 otherwise.
+
+@item pow(x, y)
+Compute the power of @var{x} elevated @var{y}, it is equivalent to
+"(@var{x})^(@var{y})".
+
+@item random(x)
+Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
+internal variable which will be used to save the seed/state.
+
+@item hypot(x, y)
+This function is similar to the C function with the same name; it returns
+"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
+right triangle with sides of length @var{x} and @var{y}, or the distance of the
+point (@var{x}, @var{y}) from the origin.
+
+@item gcd(x, y)
+Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
+@var{y} are 0 or either or both are less than zero then behavior is undefined.
+
+@item if(x, y)
+Evaluate @var{x}, and if the result is non-zero return the result of
+the evaluation of @var{y}, return 0 otherwise.
+
+@item ifnot(x, y)
+Evaluate @var{x}, and if the result is zero return the result of the
+evaluation of @var{y}, return 0 otherwise.
+
+@item taylor(expr, x) taylor(expr, x, id)
+Evaluate a taylor series at x.
+expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
+then 0 is assumed.
+note, when you have the derivatives at y instead of 0
+taylor(expr, x-y) can be used
+When the series does not converge the results are undefined.
+
+@item root(expr, max)
+Finds x where f(x)=0 in the interval 0..max.
+f() must be continuous or the result is undefined.
+@end table
+
+The following constants are available:
+@table @option
+@item PI
+area of the unit disc, approximately 3.14
+@item E
+exp(1) (Euler's number), approximately 2.718
+@item PHI
+golden ratio (1+sqrt(5))/2, approximately 1.618
+@end table
+
+Assuming that an expression is considered "true" if it has a non-zero
+value, note that:
+
+@code{*} works like AND
+
+@code{+} works like OR
+
+and the construct:
+@example
+if A then B else C
+@end example
+is equivalent to
+@example
+if(A,B) + ifnot(A,C)
+@end example
+
+In your C code, you can extend the list of unary and binary functions,
+and define recognized constants, so that they are available for your
+expressions.
+
+The evaluator also recognizes the International System number
+postfixes. If 'i' is appended after the postfix, powers of 2 are used
+instead of powers of 10. The 'B' postfix multiplies the value for 8,
+and can be appended after another postfix or used alone. This allows
+using for example 'KB', 'MiB', 'G' and 'B' as postfix.
+
+Follows the list of available International System postfixes, with
+indication of the corresponding powers of 10 and of 2.
+@table @option
+@item y
+-24 / -80
+@item z
+-21 / -70
+@item a
+-18 / -60
+@item f
+-15 / -50
+@item p
+-12 / -40
+@item n
+-9 / -30
+@item u
+-6 / -20
+@item m
+-3 / -10
+@item c
+-2
+@item d
+-1
+@item h
+2
+@item k
+3 / 10
+@item K
+3 / 10
+@item M
+6 / 20
+@item G
+9 / 30
+@item T
+12 / 40
+@item P
+15 / 40
+@item E
+18 / 50
+@item Z
+21 / 60
+@item Y
+24 / 70
+@end table
+
+@c man end
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -7,7 +7,7 @@ FFMPEG_LIBS=    libavdevice                        \
                libswscale                         \
                libavutil                          \

-CFLAGS += -Wall -g
+CFLAGS += -Wall -O2 -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

@@ -25,7 +25,6 @@ OBJS=$(addsuffix .o,$(EXAMPLES))
 # the following examples make explicit use of the math library
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
-resampling_audio:  LDLIBS += -lm

 .phony: all clean-test clean

--- a/doc/examples/README
+++ b/doc/examples/README
@@ -14,5 +14,5 @@ correctly configured.
 2) Build the examples in-tree

 Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then you can go into
+FFmpeg (no need to make install in any prefix). Then you can go into the
 doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -79,7 +79,7 @@ static int select_channel_layout(AVCodec *codec)
 {
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
-    int best_nb_channels   = 0;
+    int best_nb_channells   = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;
@@ -88,9 +88,9 @@ static int select_channel_layout(AVCodec *codec)
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

-        if (nb_channels > best_nb_channels) {
+        if (nb_channels > best_nb_channells) {
            best_ch_layout    = *p;
-            best_nb_channels = nb_channels;
+            best_nb_channells = nb_channels;
        }
        p++;
    }
--- a/doc/examples/demuxing.c
+++ b/doc/examples/demuxing.c
@@ -47,6 +47,10 @@ static uint8_t *video_dst_data[4] = {NULL};
 static int      video_dst_linesize[4];
 static int video_dst_bufsize;

+static uint8_t **audio_dst_data = NULL;
+static int       audio_dst_linesize;
+static int audio_dst_bufsize;
+
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
 static AVPacket pkt;
@@ -56,7 +60,6 @@ static int audio_frame_count = 0;
 static int decode_packet(int *got_frame, int cached)
 {
    int ret = 0;
-    int decoded = pkt.size;

    if (pkt.stream_index == video_stream_idx) {
        /* decode video frame */
@@ -88,32 +91,37 @@ static int decode_packet(int *got_frame, int cached)
            fprintf(stderr, "Error decoding audio frame\n");
            return ret;
        }
-        /* Some audio decoders decode only part of the packet, and have to be
-         * called again with the remainder of the packet data.
-         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
-         * Also, some decoders might over-read the packet. */
-        decoded = FFMIN(ret, pkt.size);

        if (*got_frame) {
-            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   audio_frame_count++, frame->nb_samples,
                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));

-            /* Write the raw audio data samples of the first plane. This works
-             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
-             * most audio decoders output planar audio, which uses a separate
-             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
-             * In other words, this code will write only the first audio channel
-             * in these cases.
-             * You should use libswresample or libavfilter to convert the frame
-             * to packed data. */
-            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
+            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, frame->channels,
+                                   frame->nb_samples, frame->format, 1);
+            if (ret < 0) {
+                fprintf(stderr, "Could not allocate audio buffer\n");
+                return AVERROR(ENOMEM);
+            }
+
+            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
+            audio_dst_bufsize =
+                av_samples_get_buffer_size(NULL, frame->channels,
+                                           frame->nb_samples, frame->format, 1);
+
+            /* copy audio data to destination buffer:
+             * this is required since rawaudio expects non aligned data */
+            av_samples_copy(audio_dst_data, frame->data, 0, 0,
+                            frame->nb_samples, frame->channels, frame->format);
+
+            /* write to rawaudio file */
+            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
+            av_freep(&audio_dst_data[0]);
        }
    }

-    return decoded;
+    return ret;
 }

 static int open_codec_context(int *stream_idx,
@@ -236,6 +244,8 @@ int main (int argc, char **argv)
    }

    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+        int nb_planes;
+
        audio_stream = fmt_ctx->streams[audio_stream_idx];
        audio_dec_ctx = audio_stream->codec;
        audio_dst_file = fopen(audio_dst_filename, "wb");
@@ -244,6 +254,15 @@ int main (int argc, char **argv)
            ret = 1;
            goto end;
        }
+
+        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
+            audio_dec_ctx->channels : 1;
+        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
+        if (!audio_dst_data) {
+            fprintf(stderr, "Could not allocate audio data buffers\n");
+            ret = AVERROR(ENOMEM);
+            goto end;
+        }
    }

    /* dump input information to stderr */
@@ -273,17 +292,8 @@ int main (int argc, char **argv)
        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);

    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_free_packet(&orig_pkt);
-    }
+    while (av_read_frame(fmt_ctx, &pkt) >= 0)
+        decode_packet(&got_frame, 0);

    /* flush cached frames */
    pkt.data = NULL;
@@ -302,25 +312,13 @@ int main (int argc, char **argv)
    }

    if (audio_stream) {
-        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
-        int n_channels = audio_dec_ctx->channels;
        const char *fmt;

-        if (av_sample_fmt_is_planar(sfmt)) {
-            const char *packed = av_get_sample_fmt_name(sfmt);
-            printf("Warning: the sample format the decoder produced is planar "
-                   "(%s). This example will output the first channel only.\n",
-                   packed ? packed : "?");
-            sfmt = av_get_packed_sample_fmt(sfmt);
-            n_channels = 1;
-        }
-
-        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
+        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
            goto end;
-
        printf("Play the output audio file with the command:\n"
               "ffplay -f %s -ac %d -ar %d %s\n",
-               fmt, n_channels, audio_dec_ctx->sample_rate,
+               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
               audio_dst_filename);
    }

@@ -336,6 +334,7 @@ end:
        fclose(audio_dst_file);
    av_free(frame);
    av_free(video_dst_data[0]);
+    av_free(audio_dst_data);

    return ret < 0;
 }
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -36,10 +36,9 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>

-static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
-static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+const char *filter_descr = "aresample=8000,aconvert=s16:mono";
+const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -71,7 +70,6 @@ static int open_input_file(const char *filename)
    }
    audio_stream_index = ret;
    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);

    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -87,12 +85,11 @@ static int init_filters(const char *filters_descr)
    char args[512];
    int ret;
    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+    AVFilter *abuffersink = avfilter_get_by_name("ffabuffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
-    static const int out_sample_rates[] = { 8000, -1 };
+    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    AVABufferSinkParams *abuffersink_params;
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

@@ -113,34 +110,16 @@ static int init_filters(const char *filters_descr)
    }

    /* buffer audio sink: to terminate the filter chain. */
+    abuffersink_params = av_abuffersink_params_alloc();
+    abuffersink_params->sample_fmts     = sample_fmts;
    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, NULL, filter_graph);
+                                       NULL, abuffersink_params, filter_graph);
+    av_free(abuffersink_params);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
        return ret;
    }

-    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
-                              AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-        return ret;
-    }
-
-    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
-                              AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-        return ret;
-    }
-
-    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
-                              AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-        return ret;
-    }
-
    /* Endpoints for the filter graph. */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
@@ -152,7 +131,7 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
                                    &inputs, &outputs, NULL)) < 0)
        return ret;

@@ -171,10 +150,11 @@ static int init_filters(const char *filters_descr)
    return 0;
 }

-static void print_frame(const AVFrame *frame)
+static void print_samplesref(AVFilterBufferRef *samplesref)
 {
-    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
-    const uint16_t *p     = (uint16_t*)frame->data[0];
+    const AVFilterBufferRefAudioProps *props = samplesref->audio;
+    const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
+    const uint16_t *p     = (uint16_t*)samplesref->data[0];
    const uint16_t *p_end = p + n;

    while (p < p_end) {
@@ -189,14 +169,9 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
+    AVFrame frame;
    int got_frame;

-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
        exit(1);
@@ -213,13 +188,14 @@ int main(int argc, char **argv)

    /* read all packets */
    while (1) {
+        AVFilterBufferRef *samplesref;
        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;

        if (packet.stream_index == audio_stream_index) {
-            avcodec_get_frame_defaults(frame);
+            avcodec_get_frame_defaults(&frame);
            got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
+            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                continue;
@@ -227,20 +203,22 @@ int main(int argc, char **argv)

            if (got_frame) {
                /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
+                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                    break;
                }

                /* pull filtered audio from the filtergraph */
                while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
                    if(ret < 0)
                        goto end;
-                    print_frame(filt_frame);
-                    av_frame_unref(filt_frame);
+                    if (samplesref) {
+                        print_samplesref(samplesref);
+                        avfilter_unref_bufferp(&samplesref);
+                    }
                }
            }
        }
@@ -251,8 +229,6 @@ end:
    if (dec_ctx)
        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
        char buf[1024];
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -85,7 +85,7 @@ static int init_filters(const char *filters_descr)
    char args[512];
    int ret;
    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    AVFilter *buffersink = avfilter_get_by_name("buffersink");
+    AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
@@ -129,7 +129,7 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
                                    &inputs, &outputs, NULL)) < 0)
        return ret;

@@ -138,33 +138,33 @@ static int init_filters(const char *filters_descr)
    return 0;
 }

-static void display_frame(const AVFrame *frame, AVRational time_base)
+static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
 {
    int x, y;
    uint8_t *p0, *p;
    int64_t delay;

-    if (frame->pts != AV_NOPTS_VALUE) {
+    if (picref->pts != AV_NOPTS_VALUE) {
        if (last_pts != AV_NOPTS_VALUE) {
            /* sleep roughly the right amount of time;
             * usleep is in microseconds, just like AV_TIME_BASE. */
-            delay = av_rescale_q(frame->pts - last_pts,
+            delay = av_rescale_q(picref->pts - last_pts,
                                 time_base, AV_TIME_BASE_Q);
            if (delay > 0 && delay < 1000000)
                usleep(delay);
        }
-        last_pts = frame->pts;
+        last_pts = picref->pts;
    }

    /* Trivial ASCII grayscale display. */
-    p0 = frame->data[0];
+    p0 = picref->data[0];
    puts("\033c");
-    for (y = 0; y < frame->height; y++) {
+    for (y = 0; y < picref->video->h; y++) {
        p = p0;
-        for (x = 0; x < frame->width; x++)
+        for (x = 0; x < picref->video->w; x++)
            putchar(" .-+#"[*(p++) / 52]);
        putchar('\n');
-        p0 += frame->linesize[0];
+        p0 += picref->linesize[0];
    }
    fflush(stdout);
 }
@@ -173,14 +173,9 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
+    AVFrame frame;
    int got_frame;

-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
    if (argc != 2) {
        fprintf(stderr, "Usage: %s file\n", argv[0]);
        exit(1);
@@ -197,36 +192,40 @@ int main(int argc, char **argv)

    /* read all packets */
    while (1) {
+        AVFilterBufferRef *picref;
        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;

        if (packet.stream_index == video_stream_index) {
-            avcodec_get_frame_defaults(frame);
+            avcodec_get_frame_defaults(&frame);
            got_frame = 0;
-            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
+            ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                break;
            }

            if (got_frame) {
-                frame->pts = av_frame_get_best_effort_timestamp(frame);
+                frame.pts = av_frame_get_best_effort_timestamp(&frame);

                /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
                    break;
                }

-                /* pull filtered frames from the filtergraph */
+                /* pull filtered pictures from the filtergraph */
                while (1) {
-                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
                    if (ret < 0)
                        goto end;
-                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
-                    av_frame_unref(filt_frame);
+
+                    if (picref) {
+                        display_picref(picref, buffersink_ctx->inputs[0]->time_base);
+                        avfilter_unref_bufferp(&picref);
+                    }
                }
            }
        }
@@ -237,8 +236,6 @@ end:
    if (dec_ctx)
        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
        char buf[1024];
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -34,11 +34,9 @@
 #include <string.h>
 #include <math.h>

-#include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
-#include <libswresample/swresample.h>

 /* 5 seconds stream duration */
 #define STREAM_DURATION   200.0
@@ -48,6 +46,13 @@

 static int sws_flags = SWS_BICUBIC;

+/**************************************************************/
+/* audio output */
+
+static float t, tincr, tincr2;
+static int16_t *samples;
+static int audio_input_frame_size;
+
 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                            enum AVCodecID codec_id)
@@ -73,13 +78,15 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,

    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
+        st->id = 1;
+        c->sample_fmt  = AV_SAMPLE_FMT_S16;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
        c->channels    = 2;
        break;

    case AVMEDIA_TYPE_VIDEO:
+        avcodec_get_context_defaults3(c, *codec);
        c->codec_id = codec_id;

        c->bit_rate = 400000;
@@ -121,17 +128,8 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 /* audio output */

 static float t, tincr, tincr2;
-
-static uint8_t **src_samples_data;
-static int       src_samples_linesize;
-static int       src_nb_samples;
-
-static int max_dst_nb_samples;
-uint8_t **dst_samples_data;
-int       dst_samples_linesize;
-int       dst_samples_size;
-
-struct SwrContext *swr_ctx = NULL;
+static int16_t *samples;
+static int audio_input_frame_size;

 static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
@@ -153,51 +151,17 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    /* increment frequency by 110 Hz per second */
    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

-    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
-        10000 : c->frame_size;
-
-    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
-                                             src_nb_samples, c->sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate source samples\n");
+    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
+        audio_input_frame_size = 10000;
+    else
+        audio_input_frame_size = c->frame_size;
+    samples = av_malloc(audio_input_frame_size *
+                        av_get_bytes_per_sample(c->sample_fmt) *
+                        c->channels);
+    if (!samples) {
+        fprintf(stderr, "Could not allocate audio samples buffer\n");
        exit(1);
    }
-
-    /* create resampler context */
-    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
-        swr_ctx = swr_alloc();
-        if (!swr_ctx) {
-            fprintf(stderr, "Could not allocate resampler context\n");
-            exit(1);
-        }
-
-        /* set options */
-        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
-        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
-        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
-
-        /* initialize the resampling context */
-        if ((ret = swr_init(swr_ctx)) < 0) {
-            fprintf(stderr, "Failed to initialize the resampling context\n");
-            exit(1);
-        }
-    }
-
-    /* compute the number of converted samples: buffering is avoided
-     * ensuring that the output buffer will contain at least all the
-     * converted input samples */
-    max_dst_nb_samples = src_nb_samples;
-    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
-                                             max_dst_nb_samples, c->sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate destination samples\n");
-        exit(1);
-    }
-    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
-                                                  c->sample_fmt, 0);
 }

 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -222,45 +186,18 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet, ret, dst_nb_samples;
+    int got_packet, ret;

    av_init_packet(&pkt);
    c = st->codec;

-    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
-
-    /* convert samples from native format to destination codec format, using the resampler */
-    if (swr_ctx) {
-        /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
-                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
-        if (dst_nb_samples > max_dst_nb_samples) {
-            av_free(dst_samples_data[0]);
-            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
-                                   dst_nb_samples, c->sample_fmt, 0);
-            if (ret < 0)
-                exit(1);
-            max_dst_nb_samples = dst_nb_samples;
-            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
-                                                          c->sample_fmt, 0);
-        }
-
-        /* convert to destination format */
-        ret = swr_convert(swr_ctx,
-                          dst_samples_data, dst_nb_samples,
-                          (const uint8_t **)src_samples_data, src_nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            exit(1);
-        }
-    } else {
-        dst_samples_data[0] = src_samples_data[0];
-        dst_nb_samples = src_nb_samples;
-    }
-
-    frame->nb_samples = dst_nb_samples;
+    get_audio_frame(samples, audio_input_frame_size, c->channels);
+    frame->nb_samples = audio_input_frame_size;
    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             dst_samples_data[0], dst_samples_size, 0);
+                             (uint8_t *)samples,
+                             audio_input_frame_size *
+                             av_get_bytes_per_sample(c->sample_fmt) *
+                             c->channels, 1);

    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
    if (ret < 0) {
@@ -286,8 +223,8 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
    avcodec_close(st->codec);
-    av_free(src_samples_data[0]);
-    av_free(dst_samples_data[0]);
+
+    av_free(samples);
 }

 /**************************************************************/
@@ -406,19 +343,25 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)

        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
-        AVPacket pkt = { 0 };
-        int got_packet;
-        av_init_packet(&pkt);
-
        /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+        AVPacket pkt;
+        int got_output;
+
+        av_init_packet(&pkt);
+        pkt.data = NULL;    // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
            exit(1);
        }
-        /* If size is zero, it means the image was buffered. */

-        if (!ret && got_packet && pkt.size) {
+        /* If size is zero, it means the image was buffered. */
+        if (got_output) {
+            if (c->coded_frame->key_frame)
+                pkt.flags |= AV_PKT_FLAG_KEY;
+
            pkt.stream_index = st->index;

            /* Write the compressed frame to the media file. */
@@ -452,8 +395,8 @@ int main(int argc, char **argv)
    AVFormatContext *oc;
    AVStream *audio_st, *video_st;
    AVCodec *audio_codec, *video_codec;
-    double audio_time, video_time;
-    int ret;
+    double audio_pts, video_pts;
+    int ret, i;

    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();
@@ -525,15 +468,23 @@ int main(int argc, char **argv)
        frame->pts = 0;
    for (;;) {
        /* Compute current audio and video time. */
-        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
-        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+        if (audio_st)
+            audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
+        else
+            audio_pts = 0.0;

-        if ((!audio_st || audio_time >= STREAM_DURATION) &&
-            (!video_st || video_time >= STREAM_DURATION))
+        if (video_st)
+            video_pts = (double)video_st->pts.val * video_st->time_base.num /
+                        video_st->time_base.den;
+        else
+            video_pts = 0.0;
+
+        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
+            (!video_st || video_pts >= STREAM_DURATION))
            break;

        /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
+        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
            write_audio_frame(oc, audio_st);
        } else {
            write_video_frame(oc, video_st);
@@ -553,12 +504,18 @@ int main(int argc, char **argv)
    if (audio_st)
        close_audio(oc, audio_st);

+    /* Free the streams. */
+    for (i = 0; i < oc->nb_streams; i++) {
+        av_freep(&oc->streams[i]->codec);
+        av_freep(&oc->streams[i]);
+    }
+
    if (!(fmt->flags & AVFMT_NOFILE))
        /* Close the output file. */
        avio_close(oc->pb);

    /* free the stream */
-    avformat_free_context(oc);
+    av_free(oc);

    return 0;
 }
--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -78,6 +78,18 @@ void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate,
    }
 }

+int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
+                                    int nb_samples, enum AVSampleFormat sample_fmt, int align)
+{
+    int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
+
+    *data = av_malloc(sizeof(*data) * nb_planes);
+    if (!*data)
+        return AVERROR(ENOMEM);
+    return av_samples_alloc(*data, linesize, nb_channels,
+                            nb_samples, sample_fmt, align);
+}
+
 int main(int argc, char **argv)
 {
    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
@@ -137,8 +149,8 @@ int main(int argc, char **argv)
    /* allocate source and destination samples buffers */

    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
-    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
-                                             src_nb_samples, src_sample_fmt, 0);
+    ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
+                                       src_nb_samples, src_sample_fmt, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate source samples\n");
        goto end;
@@ -152,8 +164,8 @@ int main(int argc, char **argv)

    /* buffer is going to be directly written to a rawaudio file, no alignment */
    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
-    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
-                                             dst_nb_samples, dst_sample_fmt, 0);
+    ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
+                                       dst_nb_samples, dst_sample_fmt, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate destination samples\n");
        goto end;
--- a/doc/examples/scaling_video.c
+++ b/doc/examples/scaling_video.c
@@ -107,7 +107,7 @@ int main(int argc, char **argv)
        goto end;
    }

-    /* buffer is going to be written to rawvideo file, no alignment */
+    /* buffer is going to be written to rawvideo file, no alignmnet */
    if ((ret = av_image_alloc(dst_data, dst_linesize,
                              dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
        fprintf(stderr, "Could not allocate destination image\n");
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -105,7 +105,7 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
 Then you may run:

@example
-ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example

 Notice that @samp{%d} is replaced by the image number.
@@ -118,7 +118,7 @@ the sequence. This is useful if your sequence does not start with
 example will start with @file{img100.jpg}:

@example
-ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example

 If you have large number of pictures to rename, you can use the
@@ -128,7 +128,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.

@example
-x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+  x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example

 If you want to sequence them by oldest modified first, substitute
@@ -137,7 +137,7 @@ If you want to sequence them by oldest modified first, substitute
 Then run:

@example
-ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example

 The same logic is used for any image format that ffmpeg reads.
@@ -145,7 +145,7 @@ The same logic is used for any image format that ffmpeg reads.
 You can also use @command{cat} to pipe images to ffmpeg:

@example
-cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
+  cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example

@section How do I encode movie to single pictures?
@@ -153,7 +153,7 @@ cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
 Use:

@example
-ffmpeg -i movie.mpg movie%d.jpg
+  ffmpeg -i movie.mpg movie%d.jpg
@end example

 The @file{movie.mpg} used as input will be converted to
@@ -169,7 +169,7 @@ to force the encoding.

 Applying that to the previous example:
@example
-ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+  ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example

 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -227,15 +227,15 @@ then you may use any file that DirectShow can read as input.

 Just create an "input.avs" text file with this single line ...
@example
-DirectShowSource("C:\path to your file\yourfile.asf")
+  DirectShowSource("C:\path to your file\yourfile.asf")
@end example
 ... and then feed that text file to ffmpeg:
@example
-ffmpeg -i input.avs
+  ffmpeg -i input.avs
@end example

-For ANY other help on AviSynth, please visit the
-@uref{http://www.avisynth.org/, AviSynth homepage}.
+For ANY other help on Avisynth, please visit the
+@uref{http://www.avisynth.org/, Avisynth homepage}.

@section How can I join video files?

@@ -294,12 +294,8 @@ your format doesn't support file level concatenation.

@subsection Concatenating using the concat @emph{protocol} (file level)

-FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
-@code{concat}} protocol designed specifically for that, with examples in the
-documentation.
-
 A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
-video by merely concatenating the files containing them.
+video by merely concatenating the files them.

 Hence you may concatenate your multimedia files by first transcoding them to
 these privileged formats, then using the humble @code{cat} command (or the
@@ -393,17 +389,17 @@ Appending @code{:v} to it will do exactly that.
 Use @option{-dumpgraph -} to find out exactly where the channel layout is
 lost.

-Most likely, it is through @code{auto-inserted aresample}. Try to understand
+Most likely, it is through @code{auto-inserted aconvert}. Try to understand
 why the converting filter was needed at that place.

 Just before the output is a likely place, as @option{-f lavfi} currently
 only support packed S16.

-Then insert the correct @code{aformat} explicitly in the filtergraph,
+Then insert the correct @code{aconvert} explicitly in the filter graph,
 specifying the exact format.

@example
-aformat=sample_fmts=s16:channel_layouts=stereo
+aconvert=s16:stereo:packed
@end example

@section Why does FFmpeg not see the subtitles in my VOB file?
@@ -475,10 +471,9 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.

@section Why are the ffmpeg programs devoid of debugging symbols?

-The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
-contain full debug information. Those binaries are stripped to create
-@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
-the *_g versions.
+The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
+information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
+you need the debug information, use the *_g versions.

@section I do not like the LGPL, can I contribute code under the GPL instead?

@@ -498,7 +493,7 @@ An easy way to get the full list of required libraries in dependency order
 is to use @code{pkg-config}.

@example
-c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example

 See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
@@ -522,6 +517,10 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
 You have to create a custom AVIOContext using @code{avio_alloc_context},
 see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.

+@section Where can I find libav* headers for Pascal/Delphi?
+
+see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
+
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?

 see @url{http://www.ffmpeg.org/~michael/}
@@ -534,12 +533,11 @@ In this specific case please look at RFC 4629 to see how it should be done.

@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.

-@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
+r_frame_rate is NOT the average frame rate, it is the smallest frame rate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
-will be 150 (it is the least common multiple).
-If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
+For example, if you have mixed 25 and 30 fps content, then r_frame_rate
+will be 150.

@section Why is @code{make fate} not running all tests?

--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -27,7 +27,7 @@ by visiting this website:

  This is especially recommended for all people contributing source
 code to FFmpeg, as it can be seen if some test on some platform broke
-with their recent contribution. This usually happens on the platforms
+with there recent contribution. This usually happens on the platforms
 the developers could not test on.

  The second part of this document describes how you can run FATE to
@@ -131,12 +131,7 @@ of the server and to accept its host key. This can usually be achieved by
 running your SSH client manually and killing it after you accepted the key.
 The FATE server's fingerprint is:

-@table @option
-@item RSA
-   d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
-@item ECDSA
-   76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
-@end table
+  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92

  If you have problems connecting to the FATE server, it may help to try out
 the @command{ssh} command with one or more @option{-v} options. You should
@@ -153,20 +148,20 @@ the synchronisation of the samples directory.

@table @option
@item fate-rsync
-Download/synchronize sample files to the configured samples directory.
+    Download/synchronize sample files to the configured samples directory.

@item fate-list
-Will list all fate/regression test targets.
+    Will list all fate/regression test targets.

@item fate
-Run the FATE test suite (requires the fate-suite dataset).
+    Run the FATE test suite (requires the fate-suite dataset).
@end table

@section Makefile variables

@table @option
@item V
-Verbosity level, can be set to 0, 1 or 2.
+    Verbosity level, can be set to 0, 1 or 2.
    @itemize
        @item 0: show just the test arguments
        @item 1: show just the command used in the test
@@ -174,28 +169,22 @@ Verbosity level, can be set to 0, 1 or 2.
    @end itemize

@item SAMPLES
-Specify or override the path to the FATE samples at make time, it has a
-meaning only while running the regression tests.
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.

@item THREADS
-Specify how many threads to use while running regression tests, it is
-quite useful to detect thread-related regressions.
-
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
@item THREAD_TYPE
-Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @var{slice+frame}
-
+    Specify which threading strategy test, either @var{slice} or @var{frame},
+    by default @var{slice+frame}
@item CPUFLAGS
-Specify CPU flags.
-
+    Specify CPU flags.
@item TARGET_EXEC
-Specify or override the wrapper used to run the tests.
-The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
-@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-through @command{ssh}.
-
-@item GEN
-Set to @var{1} to generate the missing or mismatched references.
+    Specify or override the wrapper used to run the tests.
+    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
+    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
+    through @command{ssh}.
@end table

@section Examples
--- a/doc/fate_config.sh.template
+++ b/doc/fate_config.sh.template
@@ -4,20 +4,16 @@ samples=                                 # path to samples directory
 workdir=                                 # directory in which to do all the work
 #fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
 comment=                                 # optional description
-build_only=     # set to "yes" for a compile-only instance that skips tests

 # the following are optional and map to configure options
 arch=
 cpu=
 cross_prefix=
-as=
 cc=
-ld=
 target_os=
 sysroot=
 target_exec=
 target_path=
-target_samples=
 extra_cflags=
 extra_ldflags=
 extra_libs=
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
--- a/doc/ffmpeg-devices.texi
+++ b/doc/ffmpeg-devices.texi
@@ -17,7 +17,27 @@ libavdevice library.

@c man end DESCRIPTION

-@include devices.texi
+@chapter Device Options
+@c man begin DEVICE OPTIONS
+
+The libavdevice library provides the same interface as
+libavformat. Namely, an input device is considered like a demuxer, and
+an output device like a muxer, and the interface and generic device
+options are the same provided by libavformat (see the ffmpeg-formats
+manual).
+
+In addition each input or output device may support so-called private
+options, which are specific for that component.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, or by setting the value explicitly in the device
+@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
+for programmatic use.
+
+@c man end DEVICE OPTIONS
+
+@include indevs.texi
+@include outdevs.texi

@chapter See Also

--- a/doc/ffmpeg-formats.texi
+++ b/doc/ffmpeg-formats.texi
@@ -17,7 +17,142 @@ provided by the libavformat library.

@c man end DESCRIPTION

-@include formats.texi
+@chapter Format Options
+@c man begin FORMAT OPTIONS
+
+The libavformat library provides some generic global options, which
+can be set on all the muxers and demuxers. In addition each muxer or
+demuxer may support so-called private options, which are specific for
+that component.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, or by setting the value explicitly in the
+@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
+for programmatic use.
+
+The list of supported options follows:
+
+@table @option
+@item avioflags @var{flags} (@emph{input/output})
+Possible values:
+@table @samp
+@item direct
+Reduce buffering.
+@end table
+
+@item probesize @var{integer} (@emph{input})
+Set probing size in bytes, i.e. the size of the data to analyze to get
+stream information. A higher value will allow to detect more
+information in case it is dispersed into the stream, but will increase
+latency. Must be an integer not lesser than 32. It is 5000000 by default.
+
+@item packetsize @var{integer} (@emph{output})
+Set packet size.
+
+@item fflags @var{flags} (@emph{input/output})
+Set format flags.
+
+Possible values:
+@table @samp
+@item ignidx
+Ignore index.
+@item genpts
+Generate PTS.
+@item nofillin
+Do not fill in missing values that can be exactly calculated.
+@item noparse
+Disable AVParsers, this needs @code{+nofillin} too.
+@item igndts
+Ignore DTS.
+@item discardcorrupt
+Discard corrupted frames.
+@item sortdts
+Try to interleave output packets by DTS.
+@item keepside
+Do not merge side data.
+@item latm
+Enable RTP MP4A-LATM payload.
+@item nobuffer
+Reduce the latency introduced by optional buffering
+@end table
+
+@item analyzeduration @var{integer} (@emph{input})
+Specify how many microseconds are analyzed to estimate duration.
+
+@item cryptokey @var{hexadecimal string} (@emph{input})
+Set decryption key.
+
+@item indexmem @var{integer} (@emph{input})
+Set max memory used for timestamp index (per stream).
+
+@item rtbufsize @var{integer} (@emph{input})
+Set max memory used for buffering real-time frames.
+
+@item fdebug @var{flags} (@emph{input/output})
+Print specific debug info.
+
+Possible values:
+@table @samp
+@item ts
+@end table
+
+@item max_delay @var{integer} (@emph{input/output})
+Set maximum muxing or demuxing delay in microseconds.
+
+@item fpsprobesize @var{integer} (@emph{input})
+Set number of frames used to probe fps.
+
+@item audio_preload @var{integer} (@emph{output})
+Set microseconds by which audio packets should be interleaved earlier.
+
+@item chunk_duration @var{integer} (@emph{output})
+Set microseconds for each chunk.
+
+@item chunk_size @var{integer} (@emph{output})
+Set size in bytes for each chunk.
+
+@item err_detect, f_err_detect @var{flags} (@emph{input})
+Set error detection flags. @code{f_err_detect} is deprecated and
+should be used only via the @command{ffmpeg} tool.
+
+Possible values:
+@table @samp
+@item crccheck
+Verify embedded CRCs.
+@item bitstream
+Detect bitstream specification deviations.
+@item buffer
+Detect improper bitstream length.
+@item explode
+Abort decoding on minor error detection.
+@item careful
+Consider things that violate the spec and have not been seen in the
+wild as errors.
+@item compliant
+Consider all spec non compliancies as errors.
+@item aggressive
+Consider things that a sane encoder should not do as an error.
+@end table
+
+@item use_wallclock_as_timestamps @var{integer} (@emph{input})
+Use wallclock as timestamps.
+
+@item avoid_negative_ts @var{integer} (@emph{output})
+Shift timestamps to make them positive. 1 enables, 0 disables, default
+of -1 enables when required by target format.
+
+@item skip_initial_bytes @var{integer} (@emph{input})
+Set number initial bytes to skip. Default is 0.
+
+@item correct_ts_overflow @var{integer} (@emph{input})
+Correct single timestamp overflows if set to 1. Default is 1.
+@end table
+
+@c man end FORMAT OPTIONS
+
+@include demuxers.texi
+@include muxers.texi
+@include metadata.texi

@chapter See Also

--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -12,14 +12,208 @@
@chapter Description
@c man begin DESCRIPTION

-The FFmpeg resampler provides a high-level interface to the
+The FFmpeg resampler provides an high-level interface to the
 libswresample library audio resampling utilities. In particular it
 allows to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.

@c man end DESCRIPTION

-@include resampler.texi
+@chapter Resampler Options
+@c man begin RESAMPLER OPTIONS
+
+The audio resampler supports the following named options.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, @var{option}=@var{value} for the aresample filter,
+by setting the value explicitly in the
+@code{SwrContext} options or using the @file{libavutil/opt.h} API for
+programmatic use.
+
+@table @option
+
+@item ich, in_channel_count
+Set the number of input channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{in_channel_layout} is set.
+
+@item och, out_channel_count
+Set the number of output channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{out_channel_layout} is set.
+
+@item uch, used_channel_count
+Set the number of used channels. Default value is 0. This option is
+only used for special remapping.
+
+@item isr, in_sample_rate
+Set the input sample rate. Default value is 0.
+
+@item osr, out_sample_rate
+Set the output sample rate. Default value is 0.
+
+@item isf, in_sample_fmt
+Specify the input sample format. It is set by default to @code{none}.
+
+@item osf, out_sample_fmt
+Specify the output sample format. It is set by default to @code{none}.
+
+@item tsf, internal_sample_fmt
+Set the internal sample format. Default value is @code{none}.
+
+@item icl, in_channel_layout
+Set the input channel layout.
+
+@item ocl, out_channel_layout
+Set the output channel layout.
+
+@item clev, center_mix_level
+Set center mix level. It is a value expressed in deciBel, and must be
+inclusively included between -32 and +32.
+
+@item slev, surround_mix_level
+Set surround mix level. It is a value expressed in deciBel, and must
+be inclusively included between -32 and +32.
+
+@item lfe_mix_evel
+Set LFE mix level.
+
+@item rmvol, rematrix_volume
+Set rematrix volume. Default value is 1.0.
+
+@item flags, swr_flags
+Set flags used by the converter. Default value is 0.
+
+It supports the following individual flags:
+@table @option
+@item res
+force resampling
+@end table
+
+@item dither_scale
+Set the dither scale. Default value is 1.
+
+@item dither_method
+Set dither method. Default value is 0.
+
+Supported values:
+@table @samp
+@item rectangular
+select rectangular dither
+@item triangular
+select triangular dither
+@item triangular_hp
+select triangular dither with high pass
+@end table
+
+@item resampler
+Set resampling engine. Default value is swr.
+
+Supported values:
+@table @samp
+@item swr
+select the native SW Resampler; filter options precision and cheby are not
+applicable in this case.
+@item soxr
+select the SoX Resampler (where available); compensation, and filter options
+filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
+case.
+@end table
+
+@item filter_size
+For swr only, set resampling filter size, default value is 32.
+
+@item phase_shift
+For swr only, set resampling phase shift, default value is 10, must be included
+between 0 and 30.
+
+@item linear_interp
+Use Linear Interpolation if set to 1, default value is 0.
+
+@item cutoff
+Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
+value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
+(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
+
+@item precision
+For soxr only, the precision in bits to which the resampled signal will be
+calculated.  The default value of 20 (which, with suitable dithering, is
+appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
+value of 28 gives SoX's 'Very High Quality'.
+
+@item cheby
+For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
+approximation for 'irrational' ratios. Default value is 0.
+
+@item async
+For swr only, simple 1 parameter audio sync to timestamps using stretching,
+squeezing, filling and trimming. Setting this to 1 will enable filling and
+trimming, larger values represent the maximum amount in samples that the data
+may be stretched or squeezed for each second.
+Default value is 0, thus no compensation is applied to make the samples match
+the audio timestamps.
+
+@item min_comp
+For swr only, set the minimum difference between timestamps and audio data (in
+seconds) to trigger stretching/squeezing/filling or trimming of the
+data to make it match the timestamps. The default is that
+stretching/squeezing/filling and trimming is disabled
+(@option{min_comp} = @code{FLT_MAX}).
+
+@item min_hard_comp
+For swr only, set the minimum difference between timestamps and audio data (in
+seconds) to trigger adding/dropping samples to make it match the
+timestamps.  This option effectively is a threshold to select between
+hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
+all compensation is by default disabled through @option{min_comp}.
+The default is 0.1.
+
+@item comp_duration
+For swr only, set duration (in seconds) over which data is stretched/squeezed
+to make it match the timestamps. Must be a non-negative double float value,
+default value is 1.0.
+
+@item max_soft_comp
+For swr only, set maximum factor by which data is stretched/squeezed to make it
+match the timestamps. Must be a non-negative double float value, default value
+is 0.
+
+@item matrix_encoding
+Select matrixed stereo encoding.
+
+It accepts the following values:
+@table @samp
+@item none
+select none
+@item dolby
+select Dolby
+@item dplii
+select Dolby Pro Logic II
+@end table
+
+Default value is @code{none}.
+
+@item filter_type
+For swr only, select resampling filter type. This only affects resampling
+operations.
+
+It accepts the following values:
+@table @samp
+@item cubic
+select cubic
+@item blackman_nuttall
+select Blackman Nuttall Windowed Sinc
+@item kaiser
+select Kaiser Windowed Sinc
+@end table
+
+@item kaiser_beta
+For swr only, set Kaiser Window Beta value. Must be an integer included between
+2 and 16, default value is 9.
+
+@end table
+
+@c man end RESAMPLER OPTIONS

@chapter See Also

--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@@ -12,13 +12,111 @@
@chapter Description
@c man begin DESCRIPTION

-The FFmpeg rescaler provides a high-level interface to the libswscale
+The FFmpeg rescaler provides an high-level interface to the libswscale
 library image conversion utilities. In particular it allows to perform
 image rescaling and pixel format conversion.

@c man end DESCRIPTION

-@include scaler.texi
+@chapter Scaler Options
+@c man begin SCALER OPTIONS
+
+The video scaler supports the following named options.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools. For programmatic use, they can be set explicitly in the
+@code{SwsContext} options or through the @file{libavutil/opt.h} API.
+
+@table @option
+
+@item sws_flags
+Set the scaler flags. This is also used to set the scaling
+algorithm. Only a single algorithm should be selected.
+
+It accepts the following values:
+@table @samp
+@item fast_bilinear
+Select fast bilinear scaling algorithm.
+
+@item bilinear
+Select bilinear scaling algorithm.
+
+@item bicubic
+Select bicubic scaling algorithm.
+
+@item experimental
+Select experimental scaling algorithm.
+
+@item neighbor
+Select nearest neighbor rescaling algorithm.
+
+@item area
+Select averaging area rescaling algorithm.
+
+@item bicubiclin
+Select bicubic scaling algorithm for the luma component, bilinear for
+chroma components.
+
+@item gauss
+Select Gaussian rescaling algorithm.
+
+@item sinc
+Select sinc rescaling algorithm.
+
+@item lanczos
+Select lanczos rescaling algorithm.
+
+@item spline
+Select natural bicubic spline rescaling algorithm.
+
+@item print_info
+Enable printing/debug logging.
+
+@item accurate_rnd
+Enable accurate rounding.
+
+@item full_chroma_int
+Enable full chroma interpolation.
+
+@item full_chroma_inp
+Select full chroma input.
+
+@item bitexact
+Enable bitexact output.
+@end table
+
+@item srcw
+Set source width.
+
+@item srch
+Set source height.
+
+@item dstw
+Set destination width.
+
+@item dsth
+Set destination height.
+
+@item src_format
+Set source pixel format (must be expressed as an integer).
+
+@item dst_format
+Set destination pixel format (must be expressed as an integer).
+
+@item src_range
+Select source range.
+
+@item dst_range
+Select destination range.
+
+@item param0, param1
+Set scaling algorithm parameters. The specified values are specific of
+some scaling algorithms and ignored by others. The specified values
+are floating point number values.
+
+@end table
+
+@c man end SCALER OPTIONS

@chapter See Also

--- a/doc/ffmpeg-utils.texi
+++ b/doc/ffmpeg-utils.texi
@@ -17,7 +17,8 @@ by the libavutil library.

@c man end DESCRIPTION

-@include utils.texi
+@include syntax.texi
+@include eval.texi

@chapter See Also

--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -16,26 +16,26 @@ ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@
@chapter Description
@c man begin DESCRIPTION

-@command{ffmpeg} is a very fast video and audio converter that can also grab from
+ffmpeg is a very fast video and audio converter that can also grab from
 a live audio/video source. It can also convert between arbitrary sample
 rates and resize video on the fly with a high quality polyphase filter.

-@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
+ffmpeg reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
@code{-i} option, and writes to an arbitrary number of output "files", which are
 specified by a plain output filename. Anything found on the command line which
 cannot be interpreted as an option is considered to be an output filename.

-Each input or output file can, in principle, contain any number of streams of
-different types (video/audio/subtitle/attachment/data). The allowed number and/or
-types of streams may be limited by the container format. Selecting which
-streams from which inputs will go into which output is either done automatically
-or with the @code{-map} option (see the Stream selection chapter).
+Each input or output file can in principle contain any number of streams of
+different types (video/audio/subtitle/attachment/data). Allowed number and/or
+types of streams can be limited by the container format. Selecting, which
+streams from which inputs go into output, is done either automatically or with
+the @code{-map} option (see the Stream selection chapter).

 To refer to input files in options, you must use their indices (0-based). E.g.
-the first input file is @code{0}, the second is @code{1}, etc. Similarly, streams
+the first input file is @code{0}, the second is @code{1} etc. Similarly, streams
 within a file are referred to by their indices. E.g. @code{2:3} refers to the
-fourth stream in the third input file. Also see the Stream specifiers chapter.
+fourth stream in the third input file. See also the Stream specifiers chapter.

 As a general rule, options are applied to the next specified
 file. Therefore, order is important, and you can have the same
@@ -50,7 +50,7 @@ options apply ONLY to the next input or output file and are reset between files.

@itemize
@item
-To set the video bitrate of the output file to 64 kbit/s:
+To set the video bitrate of the output file to 64kbit/s:
@example
 ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
@end example
@@ -96,14 +96,14 @@ tracking lowest timestamp on any active input stream.
 Encoded packets are then passed to the decoder (unless streamcopy is selected
 for the stream, see further for a description). The decoder produces
 uncompressed frames (raw video/PCM audio/...) which can be processed further by
-filtering (see next section). After filtering, the frames are passed to the
-encoder, which encodes them and outputs encoded packets. Finally those are
+filtering (see next section). After filtering the frames are passed to the
+encoder, which encodes them and outputs encoded packets again. Finally those are
 passed to the muxer, which writes the encoded packets to the output file.

@section Filtering
 Before encoding, @command{ffmpeg} can process raw audio and video frames using
 filters from the libavfilter library. Several chained filters form a filter
-graph. @command{ffmpeg} distinguishes between two types of filtergraphs:
+graph.  @command{ffmpeg} distinguishes between two types of filtergraphs -
 simple and complex.

@subsection Simple filtergraphs
@@ -139,7 +139,7 @@ only sets timestamps and otherwise passes the frames unchanged.

@subsection Complex filtergraphs
 Complex filtergraphs are those which cannot be described as simply a linear
-processing chain applied to one stream. This is the case, for example, when the graph has
+processing chain applied to one stream. This is the case e.g. when the graph has
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:

@@ -164,11 +164,9 @@ input. They can be represented with the following diagram:
@end example

 Complex filtergraphs are configured with the @option{-filter_complex} option.
-Note that this option is global, since a complex filtergraph, by its nature,
+Note that this option is global, since a complex filtergraph by its nature
 cannot be unambiguously associated with a single stream or file.

-The @option{-lavfi} option is equivalent to @option{-filter_complex}.
-
 A trivial example of a complex filtergraph is the @code{overlay} filter, which
 has two video inputs and one video output, containing one video overlaid on top
 of the other. Its audio counterpart is the @code{amix} filter.
@@ -178,7 +176,7 @@ Stream copy is a mode selected by supplying the @code{copy} parameter to the
@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding
 step for the specified stream, so it does only demuxing and muxing. It is useful
 for changing the container format or modifying container-level metadata. The
-diagram above will, in this case, simplify to this:
+diagram above will in this case simplify to this:

@example
 _______              ______________            ________
@@ -190,7 +188,7 @@ diagram above will, in this case, simplify to this:
@end example

 Since there is no decoding or encoding, it is very fast and there is no quality
-loss. However, it might not work in some cases because of many factors. Applying
+loss. However it might not work in some cases because of many factors. Applying
 filters is obviously also impossible, since filters work on uncompressed data.

@c man end DETAILED DESCRIPTION
@@ -198,14 +196,14 @@ filters is obviously also impossible, since filters work on uncompressed data.
@chapter Stream selection
@c man begin STREAM SELECTION

-By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
+By default ffmpeg includes only one stream of each type (video, audio, subtitle)
 present in the input files and adds them to each output file.  It picks the
-"best" of each based upon the following criteria: for video, it is the stream
-with the highest resolution, for audio, it is the stream with the most channels, for
-subtitles, it is the first subtitle stream. In the case where several streams of
-the same type rate equally, the stream with the lowest index is chosen.
+"best" of each based upon the following criteria; for video it is the stream
+with the highest resolution, for audio the stream with the most channels, for
+subtitle it's the first subtitle stream. In the case where several streams of
+the same type rate equally, the lowest numbered stream is chosen.

-You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For
+You can disable some of those defaults by using @code{-vn/-an/-sn} options. For
 full manual control, use the @code{-map} option, which disables the defaults just
 described.

@@ -214,7 +212,7 @@ described.
@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@@ -222,7 +220,7 @@ described.

@item -f @var{fmt} (@emph{input/output})
 Force input or output file format. The format is normally auto detected for input
-files and guessed from the file extension for output files, so this option is not
+files and guessed from file extension for output files, so this option is not
 needed in most cases.

@item -i @var{filename} (@emph{input})
@@ -232,8 +230,7 @@ input file name
 Overwrite output files without asking.

@item -n (@emph{global})
-Do not overwrite output files, and exit immediately if a specified
-output file already exists.
+Do not overwrite output files but exit if file exists.

@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@@ -259,28 +256,14 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
 Stop writing the output after its duration reaches @var{duration}.
@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.

-to and -t are mutually exclusive and -t has priority.
-
-@item -to @var{position} (@emph{output})
-Stop writing the output at @var{position}.
-@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
-
-to and -t are mutually exclusive and -t has priority.
-
@item -fs @var{limit_size} (@emph{output})
 Set the file size limit, expressed in bytes.

@item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
-@var{position}. Note the in most formats it is not possible to seek exactly, so
-@command{ffmpeg} will seek to the closest seek point before @var{position}.
-When transcoding and @option{-accurate_seek} is enabled (the default), this
-extra segment between the seek point and @var{position} will be decoded and
-discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
-will be preserved.
-
-When used as an output option (before an output filename), decodes but discards
-input until the timestamps reach @var{position}.
+@var{position}. When used as an output option (before an output filename),
+decodes but discards input until the timestamps reach @var{position}. This is
+slower, but more accurate.

@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.

@@ -351,32 +334,18 @@ Stop writing to the stream after @var{framecount} frames.
 Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.

-@anchor{filter_option}
-@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the stream.
-
-@var{filtergraph} is a description of the filtergraph to apply to
-the stream, and must have a single input and a single output of the
-same type of the stream. In the filtergraph, the input is associated
-to the label @code{in}, and the output to the label @code{out}. See
-the ffmpeg-filters manual for more information about the filtergraph
-syntax.
-
-See the @ref{filter_complex_option,,-filter_complex option} if you
-want to create filtergraphs with multiple inputs and/or outputs.
-
-@item -filter_script[:@var{stream_specifier}] @var{filename} (@emph{output,per-stream})
-This option is similar to @option{-filter}, the only difference is that its
-argument is the name of the file from which a filtergraph description is to be
-read.
+@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream})
+@var{filter_graph} is a description of the filter graph to apply to
+the stream. Use @code{-filters} to show all the available filters
+(including also sources and sinks).

+See also the @option{-filter_complex} option if you want to create filter graphs
+with multiple inputs and/or outputs.
@item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
 Specify the preset for matching stream(s).

@item -stats (@emph{global})
-Print encoding progress/statistics. It is on by default, to explicitly
-disable it you need to specify @code{-nostats}.
+Print encoding progress/statistics. On by default.

@item -progress @var{url} (@emph{global})
 Send program-friendly progress information to @var{url}.
@@ -426,11 +395,11 @@ will be used.

 E.g. to extract the first attachment to a file named 'out.ttf':
@example
-ffmpeg -dump_attachment:t:0 out.ttf -i INPUT
+ffmpeg -dump_attachment:t:0 out.ttf INPUT
@end example
 To extract all attachments to files determined by the @code{filename} tag:
@example
-ffmpeg -dump_attachment:t "" -i INPUT
+ffmpeg -dump_attachment:t "" INPUT
@end example

 Technical note -- attachments are implemented as codec extradata, so this
@@ -474,9 +443,20 @@ form @var{num}:@var{den}, where @var{num} and @var{den} are the
 numerator and denominator of the aspect ratio. For example "4:3",
 "16:9", "1.3333", and "1.7777" are valid argument values.

-If used together with @option{-vcodec copy}, it will affect the aspect ratio
-stored at container level, but not the aspect ratio stored in encoded
-frames, if it exists.
+@item -croptop @var{size}
+@item -cropbottom @var{size}
+@item -cropleft @var{size}
+@item -cropright @var{size}
+All the crop options have been removed. Use -vf
+crop=width:height:x:y instead.
+
+@item -padtop @var{size}
+@item -padbottom @var{size}
+@item -padleft @var{size}
+@item -padright @var{size}
+@item -padcolor @var{hex_color}
+All the pad options have been removed. Use -vf
+pad=width:height:x:y:color instead.

@item -vn (@emph{output})
 Disable video recording.
@@ -506,11 +486,12 @@ stream
@item -vlang @var{code}
 Set the ISO 639 language code (3 letters) of the current video stream.

-@item -vf @var{filtergraph} (@emph{output})
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the stream.
+@item -vf @var{filter_graph} (@emph{output})
+@var{filter_graph} is a description of the filter graph to apply to
+the input video.
+Use the option "-filters" to show all the available filters (including
+also sources and sinks).  This is an alias for @code{-filter:v}.

-This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
@end table

@section Advanced Video Options
@@ -523,7 +504,7 @@ If the selected pixel format can not be selected, ffmpeg will print a
 warning and select the best pixel format supported by the encoder.
 If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error
 if the requested pixel format can not be selected, and automatic conversions
-inside filtergraphs are disabled.
+inside filter graphs are disabled.
 If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format
 as the input (or graph output) and automatic conversions are disabled.

@@ -538,6 +519,10 @@ list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.

+@item -deinterlace
+Deinterlace pictures.
+This option is deprecated since the deinterlacing is very low quality.
+Use the yadif filter with @code{-filter:v yadif}.
@item -ilme
 Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
 Use this option if your input file is interlaced and you want
@@ -560,58 +545,12 @@ Force video tag/fourcc. This is an alias for @code{-tag:v}.
 Show QP histogram
@item -vbsf @var{bitstream_filter}
 Deprecated see -bsf
-
@item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
-@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
 Force key frames at the specified timestamps, more precisely at the first
 frames after each specified time.
-
-If the argument is prefixed with @code{expr:}, the string @var{expr}
-is interpreted like an expression and is evaluated for each frame. A
-key frame is forced in case the evaluation is non-zero.
-
-If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
-the time of the beginning of all chapters in the file, shifted by
-@var{delta}, expressed as a time in seconds.
 This option can be useful to ensure that a seek point is present at a
 chapter mark or any other designated place in the output file.
-
-For example, to insert a key frame at 5 minutes, plus key frames 0.1 second
-before the beginning of every chapter:
-@example
-force_key_frames 0:05:00,chapters-0.1
-@end example
-
-The expression in @var{expr} can contain the following constants:
-@table @option
-@item n
-the number of current processed frame, starting from 0
-@item n_forced
-the number of forced frames
-@item prev_forced_n
-the number of the previous forced frame, it is @code{NAN} when no
-keyframe was forced yet
-@item prev_forced_t
-the time of the previous forced frame, it is @code{NAN} when no
-keyframe was forced yet
-@item t
-the time of the current processed frame
-@end table
-
-For example to force a key frame every 5 seconds, you can specify:
-@example
-force_key_frames expr:gte(t,n_forced*5)
-@end example
-
-To force a key frame 5 seconds after the time of the last forced one,
-starting from second 13:
-@example
-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
-@end example
-
-Note that forcing too many keyframes is very harmful for the lookahead
-algorithms of certain encoders: using fixed-GOP options or similar
-would be more efficient.
+The timestamps must be specified in ascending order.

@item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
@@ -642,12 +581,11 @@ Set the audio codec. This is an alias for @code{-codec:a}.
@item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.
-
-@item -af @var{filtergraph} (@emph{output})
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the stream.
-
-This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
+@item -af @var{filter_graph} (@emph{output})
+@var{filter_graph} is a description of the filter graph to apply to
+the input audio.
+Use the option "-filters" to show all the available filters (including
+also sources and sinks).  This is an alias for @code{-filter:a}.
@end table

@section Advanced Audio options:
@@ -657,12 +595,6 @@ This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter optio
 Force audio tag/fourcc. This is an alias for @code{-tag:a}.
@item -absf @var{bitstream_filter}
 Deprecated, see -bsf
-@item -guess_layout_max @var{channels} (@emph{input,per-stream})
-If some input channel layout is not known, try to guess only if it
-corresponds to at most the specified number of channels. For example, 2
-tells to @command{ffmpeg} to recognize 1 channel as mono and 2 channels as
-stereo but not 6 channels as 5.1. The default is to always try to guess. Use
-0 to disable all guessing.
@end table

@section Subtitle options:
@@ -695,9 +627,6 @@ Note that this option will delay the output of all data until the next
 subtitle packet is decoded: it may increase memory consumption and latency a
 lot.

-@item -canvas_size @var{size}
-Set the size of the canvas used to render subtitles.
-
@end table

@section Advanced options
@@ -877,12 +806,13 @@ Dump each input packet to stderr.
 When dumping packets, also dump the payload.
@item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
-or live input stream (e.g. when reading from a file). Should not be used
-with actual grab devices or live input streams (where it can cause packet
-loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
-of the input(s). It is useful for real-time output (e.g. live streaming).
+of the input(s). It is useful for real-time output (e.g. live streaming). If
+your input(s) is coming from some other live streaming source (through HTTP or
+UDP for example) the server might already be in real-time, thus the option will
+likely not be required. On the other hand, this is meaningful if your input(s)
+is a file you are trying to push in real-time.
@item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@@ -901,7 +831,7 @@ Newly added values will have to be specified as strings always.
 Each frame is passed with its timestamp from the demuxer to the muxer.
@item 1, cfr
 Frames will be duplicated and dropped to achieve exactly the requested
-constant frame rate.
+constant framerate.
@item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
@@ -913,10 +843,6 @@ Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
@end table

-Note that the timestamps may be further modified by the muxer, after this.
-For example, in the case that the format option @option{avoid_negative_ts}
-is enabled.
-
 With -map you can select from which stream the timestamps should be
 taken. You can leave either video or audio unchanged and sync the
 remaining stream(s) to the unchanged one.
@@ -926,11 +852,6 @@ Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps
 the parameter is the maximum samples per second by which the audio is changed.
 -async 1 is a special case where only the start of the audio stream is corrected
 without any later correction.
-
-Note that the timestamps may be further modified by the muxer, after this.
-For example, in the case that the format option @option{avoid_negative_ts}
-is enabled.
-
 This option has been deprecated. Use the @code{aresample} audio filter instead.

@item -copyts
@@ -939,8 +860,7 @@ to sanitize them. In particular, do not remove the initial start time
 offset value.

 Note that, depending on the @option{vsync} option or on specific muxer
-processing (e.g. in case the format option @option{avoid_negative_ts}
-is enabled) the output timestamps may mismatch with the input
+processing, the output timestamps may mismatch with the input
 timestamps even when this option is selected.

@item -copytb @var{mode}
@@ -1008,12 +928,11 @@ Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
 ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
@end example

-@anchor{filter_complex_option}
@item -filter_complex @var{filtergraph} (@emph{global})
-Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
+Define a complex filter graph, i.e. one with arbitrary number of inputs and/or
 outputs. For simple graphs -- those with one input and one output of the same
 type -- see the @option{-filter} options. @var{filtergraph} is a description of
-the filtergraph, as described in the ``Filtergraph syntax'' section of the
+the filter graph, as described in the ``Filtergraph syntax'' section of the
 ffmpeg-filters manual.

 Input link labels must refer to input streams using the
@@ -1053,32 +972,8 @@ ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv

 To generate 5 seconds of pure red video using lavfi @code{color} source:
@example
-ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
+ffmpeg -filter_complex 'color=red' -t 5 out.mkv
@end example
-
-@item -lavfi @var{filtergraph} (@emph{global})
-Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
-outputs. Equivalent to @option{-filter_complex}.
-
-@item -filter_complex_script @var{filename} (@emph{global})
-This option is similar to @option{-filter_complex}, the only difference is that
-its argument is the name of the file from which a complex filtergraph
-description is to be read.
-
-@item -accurate_seek (@emph{input})
-This option enables or disables accurate seeking in input files with the
-@option{-ss} option. It is enabled by default, so seeking is accurate when
-transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
-e.g. when copying some streams and transcoding the others.
-
-@item -override_ffserver (@emph{global})
-Overrides the input specifications from ffserver. Using this option you can
-map any input stream to ffserver and control many aspects of the encoding from
-ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
-ffserver.
-The option is intended for cases where features are needed that cannot be
-specified to ffserver but can be to ffmpeg.
-
@end table

 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1136,7 +1031,7 @@ then it will search for the file @file{libvpx-1080p.ffpreset}.

@itemize
@item
-For streaming at very low bitrates, use a low frame rate
+For streaming at very low bitrate application, use a low frame rate
 and a small GOP size. This is especially true for RealVideo where
 the Linux player does not seem to be very fast, so it can miss
 frames. An example is:
@@ -1215,14 +1110,14 @@ standard mixer.
 Grab the X11 display with ffmpeg via

@example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as
 the DISPLAY environment variable.

@example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as the DISPLAY environment
@@ -1381,48 +1276,15 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end itemize
@c man end EXAMPLES

-@include config.texi
-@ifset config-all
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
-@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
@chapter See Also

@ifhtml
-@ifset config-all
-@url{ffmpeg.html,ffmpeg}
-@end ifset
-@ifset config-not-all
-@url{ffmpeg-all.html,ffmpeg-all},
-@end ifset
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@@ -1430,12 +1292,6 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end ifhtml

@ifnothtml
-@ifset config-all
-ffmpeg(1),
-@end ifset
-@ifset config-not-all
-ffmpeg-all(1),
-@end ifset
 ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -24,7 +24,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@@ -73,22 +73,11 @@ Default value is "video", if video is not present or cannot be played
 You can interactively cycle through the available show modes by
 pressing the key @key{w}.

-@item -vf @var{filtergraph}
-Create the filtergraph specified by @var{filtergraph} and use it to
-filter the video stream.
-
-@var{filtergraph} is a description of the filtergraph to apply to
-the stream, and must have a single video input and a single video
-output. In the filtergraph, the input is associated to the label
-@code{in}, and the output to the label @code{out}. See the
-ffmpeg-filters manual for more information about the filtergraph
-syntax.
-
-@item -af @var{filtergraph}
-@var{filtergraph} is a description of the filtergraph to apply to
-the input audio.
+@item -vf @var{filter_graph}
+@var{filter_graph} is a description of the filter graph to apply to
+the input video.
 Use the option "-filters" to show all the available filters (including
-sources and sinks).
+also sources and sinks).

@item -i @var{input_file}
 Read @var{input_file}.
@@ -99,13 +88,9 @@ Read @var{input_file}.
@item -pix_fmt @var{format}
 Set pixel format.
 This option has been deprecated in favor of private options, try -pixel_format.
-
@item -stats
-Print several playback statistics, in particular show the stream
-duration, the codec parameters, the current position in the stream and
-the audio/video synchronisation drift. It is on by default, to
-explicitly disable it you need to specify @code{-nostats}.
-
+Show the stream duration, the codec parameters, the current position in
+the stream and the audio/video synchronisation drift.
@item -bug
 Work around bugs.
@item -fast
@@ -174,16 +159,13 @@ Toggle full screen.
 Pause.

@item a
-Cycle audio channel in the curret program.
+Cycle audio channel.

@item v
 Cycle video channel.

@item t
-Cycle subtitle channel in the current program.
-
-@item c
-Cycle program.
+Cycle subtitle channel.

@item w
 Show audio waves.
@@ -204,48 +186,15 @@ Seek to percentage in file corresponding to fraction of width.

@c man end

-@include config.texi
-@ifset config-all
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
-@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
@chapter See Also

@ifhtml
-@ifset config-all
-@url{ffplay.html,ffplay},
-@end ifset
-@ifset config-not-all
-@url{ffplay-all.html,ffmpeg-all},
-@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@@ -253,12 +202,6 @@ Seek to percentage in file corresponding to fraction of width.
@end ifhtml

@ifnothtml
-@ifset config-all
-ffplay(1),
-@end ifset
-@ifset config-not-all
-ffplay-all(1),
-@end ifset
 ffmpeg(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -44,15 +44,14 @@ name (which may be shared by other sections), and an unique
 name. See the output of @option{sections}.

 Metadata tags stored in the container or in the streams are recognized
-and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
-section.
+and printed in the corresponding "FORMAT" or "STREAM" section.

@c man end

@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@@ -113,7 +112,7 @@ ffprobe -show_packets -select_streams v:1 INPUT
@end example

@item -show_data
-Show payload data, as a hexadecimal and ASCII dump. Coupled with
+Show payload data, as an hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.

@@ -210,18 +209,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".

-@item -show_programs
-Show information about programs and their streams contained in the input
-multimedia stream.
-
-Each media stream information is printed within a dedicated section
-with name "PROGRAM_STREAM".
-
-@item -show_chapters
-Show information about chapters stored in the format.
-
-Each chapter is printed within a dedicated section with name "CHAPTER".
-
@item -count_frames
 Count the number of frames per stream and report it in the
 corresponding stream section.
@@ -230,70 +217,6 @@ corresponding stream section.
 Count the number of packets per stream and report it in the
 corresponding stream section.

-@item -read_intervals @var{read_intervals}
-
-Read only the specified intervals. @var{read_intervals} must be a
-sequence of interval specifications separated by ",".
-@command{ffprobe} will seek to the interval starting point, and will
-continue reading from that.
-
-Each interval is specified by two optional parts, separated by "%".
-
-The first part specifies the interval start position. It is
-interpreted as an abolute position, or as a relative offset from the
-current position if it is preceded by the "+" character. If this first
-part is not specified, no seeking will be performed when reading this
-interval.
-
-The second part specifies the interval end position. It is interpreted
-as an absolute position, or as a relative offset from the current
-position if it is preceded by the "+" character. If the offset
-specification starts with "#", it is interpreted as the number of
-packets to read (not including the flushing packets) from the interval
-start. If no second part is specified, the program will read until the
-end of the input.
-
-Note that seeking is not accurate, thus the actual interval start
-point may be different from the specified position. Also, when an
-interval duration is specified, the absolute end time will be computed
-by adding the duration to the interval start point found by seeking
-the file, rather than to the specified start value.
-
-The formal syntax is given by:
-@example
-@var{INTERVAL}  ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
-@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
-@end example
-
-A few examples follow.
-@itemize
-@item
-Seek to time 10, read packets until 20 seconds after the found seek
-point, then seek to position @code{01:30} (1 minute and thirty
-seconds) and read packets until position @code{01:45}.
-@example
-10%+20,01:30%01:45
-@end example
-
-@item
-Read only 42 packets after seeking to position @code{01:23}:
-@example
-01:23%+#42
-@end example
-
-@item
-Read only the first 20 seconds from the start:
-@example
-%+20
-@end example
-
-@item
-Read from the start until position @code{02:30}:
-@example
-%02:30
-@end example
-@end itemize
-
@item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -351,8 +274,8 @@ keyN=valN
 [/SECTION]
@end example

-Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
-PROGRAM_STREAM section, and are prefixed by the string "TAG:".
+Metadata tags are printed as a line in the corresponding FORMAT or
+STREAM section, and are prefixed by the string "TAG:".

 A description of the accepted options follows.

@@ -564,48 +487,15 @@ DV, GXF and AVI timecodes are available in format metadata
@end itemize
@c man end TIMECODE

-@include config.texi
-@ifset config-all
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
-@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
@chapter See Also

@ifhtml
-@ifset config-all
-@url{ffprobe.html,ffprobe},
-@end ifset
-@ifset config-not-all
-@url{ffprobe-all.html,ffprobe-all},
-@end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
+@url{ffplay.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@@ -613,12 +503,6 @@ DV, GXF and AVI timecodes are available in format metadata
@end ifhtml

@ifnothtml
-@ifset config-all
-ffprobe(1),
-@end ifset
-@ifset config-not-all
-ffprobe-all(1),
-@end ifset
 ffmpeg(1), ffplay(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -11,8 +11,6 @@
            <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
            <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
            <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
            <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
            <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
@@ -80,6 +78,7 @@
      <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
      <xsd:attribute name="top_field_first"        type="xsd:int"   />
      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
+      <xsd:attribute name="reference"              type="xsd:int"   />
    </xsd:complexType>

    <xsd:complexType name="streamsType">
@@ -88,12 +87,6 @@
        </xsd:sequence>
    </xsd:complexType>

-    <xsd:complexType name="programsType">
-        <xsd:sequence>
-            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
    <xsd:complexType name="streamDispositionType">
      <xsd:attribute name="default"          type="xsd:int" use="required" />
      <xsd:attribute name="dub"              type="xsd:int" use="required" />
@@ -110,8 +103,8 @@

    <xsd:complexType name="streamType">
      <xsd:sequence>
-        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
      </xsd:sequence>

      <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -138,7 +131,6 @@
      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
      <xsd:attribute name="sample_rate"      type="xsd:int"/>
      <xsd:attribute name="channels"         type="xsd:int"/>
-      <xsd:attribute name="channel_layout"   type="xsd:string"/>
      <xsd:attribute name="bits_per_sample"  type="xsd:int"/>

      <xsd:attribute name="id"               type="xsd:string"/>
@@ -155,23 +147,6 @@
      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
    </xsd:complexType>

-    <xsd:complexType name="programType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="start_time"           type="xsd:float"/>
-      <xsd:attribute name="start_pts"            type="xsd:long"/>
-      <xsd:attribute name="end_time"             type="xsd:float"/>
-      <xsd:attribute name="end_pts"              type="xsd:long"/>
-      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
-      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
-    </xsd:complexType>
-
    <xsd:complexType name="formatType">
      <xsd:sequence>
        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -179,14 +154,12 @@

      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
      <xsd:attribute name="format_long_name" type="xsd:string"/>
      <xsd:attribute name="start_time"       type="xsd:float"/>
      <xsd:attribute name="duration"         type="xsd:float"/>
      <xsd:attribute name="size"             type="xsd:long"/>
      <xsd:attribute name="bit_rate"         type="xsd:long"/>
-      <xsd:attribute name="probe_score"      type="xsd:int"/>
    </xsd:complexType>

    <xsd:complexType name="tagType">
@@ -209,25 +182,6 @@
      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
    </xsd:complexType>

-    <xsd:complexType name="chaptersType">
-      <xsd:sequence>
-        <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="chapterType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="id"         type="xsd:int" use="required"/>
-      <xsd:attribute name="time_base"  type="xsd:string" use="required"/>
-      <xsd:attribute name="start"      type="xsd:int" use="required"/>
-      <xsd:attribute name="start_time" type="xsd:float"/>
-      <xsd:attribute name="end"        type="xsd:int" use="required"/>
-      <xsd:attribute name="end_time"   type="xsd:float" use="required"/>
-    </xsd:complexType>
-
    <xsd:complexType name="libraryVersionType">
      <xsd:attribute name="name"        type="xsd:string" use="required"/>
      <xsd:attribute name="major"       type="xsd:int"    use="required"/>
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -16,28 +16,34 @@ ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION

-@command{ffserver} is a streaming server for both audio and video. It
-supports several live feeds, streaming from files and time shifting on
-live feeds (you can seek to positions in the past on each live feed,
-provided you specify a big enough feed storage in
-@file{ffserver.conf}).
+ffserver is a streaming server for both audio and video. It supports

-@command{ffserver} receives prerecorded files or FFM streams from some
-@command{ffmpeg} instance as input, then streams them over
-RTP/RTSP/HTTP.
+several live feeds, streaming from files and time shifting on live feeds
+(you can seek to positions in the past on each live feed, provided you
+specify a big enough feed storage in ffserver.conf).

-An @command{ffserver} instance will listen on some port as specified
-in the configuration file. You can launch one or more instances of
-@command{ffmpeg} and send one or more FFM streams to the port where
-ffserver is expecting to receive them. Alternately, you can make
-@command{ffserver} launch such @command{ffmpeg} instances at startup.
+This documentation covers only the streaming aspects of ffserver /
+ffmpeg. All questions about parameters for ffmpeg, codec questions,
+etc. are not covered here. Read @file{ffmpeg.html} for more
+information.

-Input streams are called feeds, and each one is specified by a
-@code{<Feed>} section in the configuration file.
+@section How does it work?
+
+ffserver receives prerecorded files or FFM streams from some ffmpeg
+instance as input, then streams them over RTP/RTSP/HTTP.
+
+An ffserver instance will listen on some port as specified in the
+configuration file. You can launch one or more instances of ffmpeg and
+send one or more FFM streams to the port where ffserver is expecting
+to receive them. Alternately, you can make ffserver launch such ffmpeg
+instances at startup.
+
+Input streams are called feeds, and each one is specified by a <Feed>
+section in the configuration file.

 For each feed you can have different output streams in various
-formats, each one specified by a @code{<Stream>} section in the
-configuration file.
+formats, each one specified by a <Stream> section in the configuration
+file.

@section Status stream

@@ -73,6 +79,14 @@ web server can be used to serve up the files just as well.
 It can stream prerecorded video from .ffm files, though it is somewhat tricky
 to make it work correctly.

+@section What do I need?
+
+I use Linux on a 900 MHz Duron with a cheap Bt848 based TV capture card. I'm
+using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
+I needed some special drivers for my motherboard-based sound card.]
+
+I understand that FreeBSD systems work just fine as well.
+
@section How do I make it work?

 First, build the kit. It *really* helps to have installed LAME first. Then when
@@ -221,7 +235,7 @@ of an infinite movie or a whole movie.

 FFM is version specific, and there is limited compatibility of FFM files
 generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but it is not guaranteed to work.
+ffmpeg/ffserver. It may work but its not guaranteed to work.

 FFM2 is extensible while maintaining compatibility and should work between
 differing versions of tools. FFM2 is the default.
@@ -229,7 +243,7 @@ differing versions of tools. FFM2 is the default.
@chapter Options
@c man begin OPTIONS

-@include fftools-common-opts.texi
+@include avtools-common-opts.texi

@section Main options

@@ -246,49 +260,16 @@ messages to stdout.
@end table
@c man end

-@include config.texi
-@ifset config-all
-@ifset config-avutil
-@include utils.texi
-@end ifset
-@ifset config-avcodec
-@include codecs.texi
-@include bitstream_filters.texi
-@end ifset
-@ifset config-avformat
-@include formats.texi
-@include protocols.texi
-@end ifset
-@ifset config-avdevice
-@include devices.texi
-@end ifset
-@ifset config-swresample
-@include resampler.texi
-@end ifset
-@ifset config-swscale
-@include scaler.texi
-@end ifset
-@ifset config-avfilter
-@include filters.texi
-@end ifset
-@end ifset
-
@chapter See Also

@ifhtml
-@ifset config-all
-@url{ffserver.html,ffserver},
-@end ifset
-@ifset config-not-all
-@url{ffserver-all.html,ffserver-all},
-@end ifset
-the @file{doc/ffserver.conf} example,
+The @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@@ -296,13 +277,7 @@ the @file{doc/ffserver.conf} example,
@end ifhtml

@ifnothtml
-@ifset config-all
-ffserver(1),
-@end ifset
-@ifset config-not-all
-ffserver-all(1),
-@end ifset
-the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
+The @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -15,13 +15,13 @@ Format negotiation
  the list of supported formats.

  For video links, that means pixel format. For audio links, that means
-  channel layout, sample format (the sample packing is implied by the sample
-  format) and sample rate.
+  channel layout, and sample format (the sample packing is implied by the
+  sample format).

  The lists are not just lists, they are references to shared objects. When
  the negotiation mechanism computes the intersection of the formats
-  supported at each end of a link, all references to both lists are replaced
-  with a reference to the intersection. And when a single format is
+  supported at each ends of a link, all references to both lists are
+  replaced with a reference to the intersection. And when a single format is
  eventually chosen for a link amongst the remaining list, again, all
  references to the list are updated.

@@ -29,11 +29,6 @@ Format negotiation
  same format amongst a supported list, all it has to do is use a reference
  to the same list of formats.

-  query_formats can leave some formats unset and return AVERROR(EAGAIN) to
-  cause the negotiation mechanism to try again later. That can be used by
-  filters with complex requirements to use the format negotiated on one link
-  to set the formats supported on another.
-

 Buffer references ownership and permissions
 ===========================================
@@ -73,15 +68,15 @@ Buffer references ownership and permissions

    Here are the (fairly obvious) rules for reference ownership:

-    * A reference received by the filter_frame method (or its start_frame
-      deprecated version) belongs to the corresponding filter.
+    * A reference received by the start_frame or filter_frame method
+      belong to the corresponding filter.

      Special exception: for video references: the reference may be used
      internally for automatic copying and must not be destroyed before
      end_frame; it can be given away to ff_start_frame.

-    * A reference passed to ff_filter_frame (or the deprecated
-      ff_start_frame) is given away and must no longer be used.
+    * A reference passed to ff_start_frame or ff_filter_frame is given
+      away and must no longer be used.

    * A reference created with avfilter_ref_buffer belongs to the code that
      created it.
@@ -95,11 +90,27 @@ Buffer references ownership and permissions
  Link reference fields
  ---------------------

-    The AVFilterLink structure has a few AVFilterBufferRef fields. The
-    cur_buf and out_buf were used with the deprecated
-    start_frame/draw_slice/end_frame API and should no longer be used.
-    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
-    and must not be accessed by filters.
+    The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
+    the rules to handle them:
+
+    * cur_buf is set before the start_frame and filter_frame methods to
+      the same reference given as argument to the methods and belongs to the
+      destination filter of the link. If it has not been cleared after
+      end_frame or filter_frame, libavfilter will automatically destroy
+      the reference; therefore, any filter that needs to keep the reference
+      for longer must set cur_buf to NULL.
+
+    * out_buf belongs to the source filter of the link and can be used to
+      store a reference to the buffer that has been sent to the destination.
+      If it is not NULL after end_frame or filter_frame, libavfilter will
+      automatically destroy the reference.
+
+      If a video input pad does not have a start_frame method, the default
+      method will request a buffer on the first output of the filter, store
+      the reference in out_buf and push a second reference to the output.
+
+    * src_buf, cur_buf_copy and partial_buf are used by libavfilter
+      internally and must not be accessed by filters.

  Reference permissions
  ---------------------
@@ -108,10 +119,8 @@ Buffer references ownership and permissions
    the code that owns the reference is allowed to do to the buffer data.
    Different references for the same buffer can have different permissions.

-    For video filters that implement the deprecated
-    start_frame/draw_slice/end_frame API, the permissions only apply to the
-    parts of the buffer that have already been covered by the draw_slice
-    method.
+    For video filters, the permissions only apply to the parts of the buffer
+    that have already been covered by the draw_slice method.

    The value is a binary OR of the following constants:

@@ -166,13 +175,13 @@ Buffer references ownership and permissions
      WRITE permission.

    * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and must request a buffer
+      scale) need the READ permission on input and and must request a buffer
      with the WRITE permission.

    * Filters that intend to keep a reference after the filtering process
-      is finished (after filter_frame returns) must have the PRESERVE
-      permission on it and remove the WRITE permission if they create a new
-      reference to give it away.
+      is finished (after end_frame or filter_frame returns) must have the
+      PRESERVE permission on it and remove the WRITE permission if they
+      create a new reference to give it away.

    * Filters that intend to modify a reference they have kept after the end
      of the filtering process need the REUSE2 permission and must remove
@@ -189,11 +198,11 @@ Frame scheduling
  Simple filters that output one frame for each input frame should not have
  to worry about it.

-  filter_frame
-  ------------
+  start_frame / filter_frame
+  ----------------------------

-    This method is called when a frame is pushed to the filter's input. It
-    can be called at any time except in a reentrant way.
+    These methods are called when a frame is pushed to the filter's input.
+    They can be called at any time except in a reentrant way.

    If the input frame is enough to produce output, then the filter should
    push the output frames on the output link immediately.
@@ -204,7 +213,7 @@ Frame scheduling
    filter; these buffered frames must be flushed immediately if a new input
    produces new output.

-    (Example: frame rate-doubling filter: filter_frame must (1) flush the
+    (Example: framerate-doubling filter: start_frame must (1) flush the
    second copy of the previous frame, if it is still there, (2) push the
    first copy of the incoming frame, (3) keep the second copy for later.)

@@ -224,8 +233,8 @@ Frame scheduling

    This method is called when a frame is wanted on an output.

-    For an input, it should directly call filter_frame on the corresponding
-    output.
+    For an input, it should directly call start_frame or filter_frame on
+    the corresponding output.

    For a filter, if there are queued frames already ready, one of these
    frames should be pushed. If not, the filter should request a frame on
@@ -246,7 +255,7 @@ Frame scheduling
        }
        while (!frame_pushed) {
            input = input_where_a_frame_is_most_needed();
-            ret = ff_request_frame(input);
+            ret = avfilter_request_frame(input);
            if (ret == AVERROR_EOF) {
                process_eof_on_input();
            } else if (ret < 0) {
@@ -257,14 +266,4 @@ Frame scheduling

    Note that, except for filters that can have queued frames, request_frame
    does not push frames: it requests them to its input, and as a reaction,
-    the filter_frame method will be called and do the work.
-
-Legacy API
-==========
-
-  Until libavfilter 3.23, the filter_frame method was split:
-
-  - for video filters, it was made of start_frame, draw_slice (that could be
-    called several times on distinct parts of the frame) and end_frame;
-
-  - for audio filters, it was called filter_samples.
+    the start_frame / filter_frame method will be called and do the work.
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/doc/formats.texi
@@ -1,188 +0,0 @@
-@chapter Format Options
-@c man begin FORMAT OPTIONS
-
-The libavformat library provides some generic global options, which
-can be set on all the muxers and demuxers. In addition each muxer or
-demuxer may support so-called private options, which are specific for
-that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-The list of supported options follows:
-
-@table @option
-@item avioflags @var{flags} (@emph{input/output})
-Possible values:
-@table @samp
-@item direct
-Reduce buffering.
-@end table
-
-@item probesize @var{integer} (@emph{input})
-Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will allow to detect more
-information in case it is dispersed into the stream, but will increase
-latency. Must be an integer not lesser than 32. It is 5000000 by default.
-
-@item packetsize @var{integer} (@emph{output})
-Set packet size.
-
-@item fflags @var{flags} (@emph{input/output})
-Set format flags.
-
-Possible values:
-@table @samp
-@item ignidx
-Ignore index.
-@item genpts
-Generate PTS.
-@item nofillin
-Do not fill in missing values that can be exactly calculated.
-@item noparse
-Disable AVParsers, this needs @code{+nofillin} too.
-@item igndts
-Ignore DTS.
-@item discardcorrupt
-Discard corrupted frames.
-@item sortdts
-Try to interleave output packets by DTS.
-@item keepside
-Do not merge side data.
-@item latm
-Enable RTP MP4A-LATM payload.
-@item nobuffer
-Reduce the latency introduced by optional buffering
-@end table
-
-@item seek2any @var{integer} (@emph{input})
-Allow seeking to non-keyframes on demuxer level when supported if set to 1.
-Default is 0.
-
-@item analyzeduration @var{integer} (@emph{input})
-Specify how many microseconds are analyzed to probe the input. A
-higher value will allow to detect more accurate information, but will
-increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
-
-@item cryptokey @var{hexadecimal string} (@emph{input})
-Set decryption key.
-
-@item indexmem @var{integer} (@emph{input})
-Set max memory used for timestamp index (per stream).
-
-@item rtbufsize @var{integer} (@emph{input})
-Set max memory used for buffering real-time frames.
-
-@item fdebug @var{flags} (@emph{input/output})
-Print specific debug info.
-
-Possible values:
-@table @samp
-@item ts
-@end table
-
-@item max_delay @var{integer} (@emph{input/output})
-Set maximum muxing or demuxing delay in microseconds.
-
-@item fpsprobesize @var{integer} (@emph{input})
-Set number of frames used to probe fps.
-
-@item audio_preload @var{integer} (@emph{output})
-Set microseconds by which audio packets should be interleaved earlier.
-
-@item chunk_duration @var{integer} (@emph{output})
-Set microseconds for each chunk.
-
-@item chunk_size @var{integer} (@emph{output})
-Set size in bytes for each chunk.
-
-@item err_detect, f_err_detect @var{flags} (@emph{input})
-Set error detection flags. @code{f_err_detect} is deprecated and
-should be used only via the @command{ffmpeg} tool.
-
-Possible values:
-@table @samp
-@item crccheck
-Verify embedded CRCs.
-@item bitstream
-Detect bitstream specification deviations.
-@item buffer
-Detect improper bitstream length.
-@item explode
-Abort decoding on minor error detection.
-@item careful
-Consider things that violate the spec and have not been seen in the
-wild as errors.
-@item compliant
-Consider all spec non compliancies as errors.
-@item aggressive
-Consider things that a sane encoder should not do as an error.
-@end table
-
-@item use_wallclock_as_timestamps @var{integer} (@emph{input})
-Use wallclock as timestamps.
-
-@item avoid_negative_ts @var{integer} (@emph{output})
-Shift timestamps to make them non-negative. A value of 1 enables shifting,
-a value of 0 disables it, the default value of -1 enables shifting
-when required by the target format.
-
-When shifting is enabled, all output timestamps are shifted by the
-same amount. Audio, video, and subtitles desynching and relative
-timestamp differences are preserved compared to how they would have
-been without shifting.
-
-Also note that this affects only leading negative timestamps, and not
-non-monotonic negative timestamps.
-
-@item skip_initial_bytes @var{integer} (@emph{input})
-Set number of bytes to skip before reading header and frames if set to 1.
-Default is 0.
-
-@item correct_ts_overflow @var{integer} (@emph{input})
-Correct single timestamp overflows if set to 1. Default is 1.
-
-@item flush_packets @var{integer} (@emph{output})
-Flush the underlying I/O stream after each packet. Default 1 enables it, and
-has the effect of reducing the latency; 0 disables it and may slightly
-increase performance in some cases.
-@end table
-
-@c man end FORMAT OPTIONS
-
-@anchor{Format stream specifiers}
-@section Format stream specifiers
-
-Format stream specifiers allow selection of one or more streams that
-match specific properties.
-
-Possible forms of stream specifiers are:
-@table @option
-@item @var{stream_index}
-Matches the stream with this index.
-
-@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio,
-'s' for subtitle, 'd' for data, and 't' for attachments. If
-@var{stream_index} is given, then it matches the stream number
-@var{stream_index} of this type. Otherwise, it matches all streams of
-this type.
-
-@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then it matches the stream with number
-@var{stream_index} in the program with the id
-@var{program_id}. Otherwise, it matches all streams in the program.
-
-@item #@var{stream_id}
-Matches the stream by a format-specific ID.
-@end table
-
-The exact semantics of stream specifiers is defined by the
-@code{avformat_match_stream_specifier()} function declared in the
-@file{libavformat/avformat.h} header.
-
-@include demuxers.texi
-@include muxers.texi
-@include metadata.texi
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -94,20 +94,12 @@ Then pass @code{--enable-libtwolame} to configure to enable it.

@section libvpx

-FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
+FFmpeg can make use of the libvpx library for VP8 encoding.

 Go to @url{http://www.webmproject.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libvpx} to configure to
 enable it.

-@section libwavpack
-
-FFmpeg can make use of the libwavpack library for WavPack encoding.
-
-Go to @url{http://www.wavpack.com/} and follow the instructions for
-installing the library. Then pass @code{--enable-libwavpack} to configure to
-enable it.
-
@section x264

 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -133,20 +125,6 @@ Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.

-@section libzvbi
-
-libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
-teletext pages and DVB teletext subtitles.
-
-Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
-installing the library. Then pass @code{--enable-libzvbi} to configure to
-enable it.
-
-@float NOTE
-libzvbi is licensed under the GNU General Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
-you must upgrade FFmpeg's license to GPL in order to use it.
-@end float


@chapter Supported File Formats, Codecs or Features
@@ -174,15 +152,13 @@ library:
    @tab Multimedia format used in game Heart Of Darkness.
@item Apple HTTP Live Streaming @tab   @tab X
@item Artworx Data Format       @tab   @tab X
-@item ADP                       @tab   @tab X
-    @tab Audio format used on the Nintendo Gamecube.
@item AFC                       @tab   @tab X
    @tab Audio format used on the Nintendo Gamecube.
@item ASF                       @tab X @tab X
@item AST                       @tab X @tab X
    @tab Audio format used on the Nintendo Wii.
@item AVI                       @tab X @tab X
-@item AviSynth                  @tab   @tab X
+@item AVISynth                  @tab   @tab X
@item AVR                       @tab   @tab X
    @tab Audio format used on Mac.
@item AVS                       @tab   @tab X
@@ -351,7 +327,7 @@ library:
@item raw Shorten               @tab   @tab X
@item raw TAK                   @tab   @tab X
@item raw TrueHD                @tab X @tab X
-@item raw VC-1                  @tab X @tab X
+@item raw VC-1                  @tab   @tab X
@item raw PCM A-law             @tab X @tab X
@item raw PCM mu-law            @tab X @tab X
@item raw PCM signed 8 bit      @tab X @tab X
@@ -377,13 +353,11 @@ library:
    @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.
@item RealMedia                 @tab X @tab X
@item Redirector                @tab   @tab X
-@item RedSpark                  @tab   @tab X
@item Renderware TeXture Dictionary @tab   @tab X
@item RL2                       @tab   @tab X
    @tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie               @tab   @tab X
@item Lego Mindstorms RSO       @tab X @tab X
-@item RSD                       @tab   @tab X
@item RTMP                      @tab X @tab X
    @tab Output is performed by publishing stream to RTMP server
@item RTP                       @tab X @tab X
@@ -449,6 +423,7 @@ following image formats are supported:
@item .Y.U.V       @tab X @tab X
    @tab one raw file per component
@item animated GIF @tab X @tab X
+    @tab Only uncompressed GIFs are generated.
@item BMP          @tab X @tab X
    @tab Microsoft BMP image
@item PIX          @tab   @tab X
@@ -488,8 +463,6 @@ following image formats are supported:
    @tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa  @tab X @tab X
    @tab Targa (.TGA) image format
-@item WebP         @tab @tab X
-    @tab WebP image format
@item XBM  @tab X @tab X
    @tab X BitMap image format
@item XFace @tab X @tab X
@@ -517,7 +490,6 @@ following image formats are supported:
@item AMV Video              @tab  X  @tab  X
    @tab Used in Chinese MP3 players.
@item ANSI/ASCII art         @tab     @tab  X
-@item Apple Intermediate Codec @tab     @tab  X
@item Apple MJPEG-B          @tab     @tab  X
@item Apple ProRes           @tab  X  @tab  X
@item Apple QuickDraw        @tab     @tab  X
@@ -600,13 +572,12 @@ following image formats are supported:
    @tab Sorenson H.263 used in Flash
@item Forward Uncompressed   @tab     @tab  X
@item Fraps                  @tab     @tab  X
-@item Go2Webinar             @tab     @tab  X
-    @tab fourcc: G2M4
@item H.261                  @tab  X  @tab  X
@item H.263 / H.263-1996     @tab  X  @tab  X
@item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
    @tab encoding supported through external library libx264
+@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration)  @tab  E  @tab  X
@item HuffYUV                @tab  X  @tab  X
@item HuffYUV FFmpeg variant @tab  X  @tab  X
@item IBM Ultimotion         @tab     @tab  X
@@ -659,6 +630,7 @@ following image formats are supported:
@item Motion Pixels video    @tab     @tab  X
@item MPEG-1 video           @tab  X  @tab  X
@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
+@item MPEG-1/2 video (VDPAU acceleration)  @tab     @tab  X
@item MPEG-2 video           @tab  X  @tab  X
@item MPEG-4 part 2          @tab  X  @tab  X
    @tab libxvidcore can be used alternatively for encoding.
@@ -676,8 +648,6 @@ following image formats are supported:
    @tab fourcc: VP60,VP61,VP62
@item VP8                    @tab  E  @tab  X
    @tab fourcc: VP80, encoding supported through external library libvpx
-@item VP9                    @tab  E  @tab  X
-    @tab encoding supported through external library libvpx
@item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
    @tab fourcc: Y216
@item Prores                 @tab     @tab  X
@@ -804,11 +774,9 @@ following image formats are supported:
    @tab Used in some Sega Saturn console games.
@item ADPCM IMA Duck DK4     @tab     @tab  X
    @tab Used in some Sega Saturn console games.
-@item ADPCM IMA Radical      @tab     @tab  X
@item ADPCM Microsoft        @tab  X  @tab  X
@item ADPCM MS IMA           @tab  X  @tab  X
@item ADPCM Nintendo Gamecube AFC  @tab     @tab  X
-@item ADPCM Nintendo Gamecube DTK  @tab     @tab  X
@item ADPCM Nintendo Gamecube THP  @tab     @tab  X
@item ADPCM QT IMA           @tab  X  @tab  X
@item ADPCM SEGA CRI ADX     @tab  X  @tab  X
@@ -827,8 +795,8 @@ following image formats are supported:
@item Amazing Studio PAF Audio @tab     @tab  X
@item Apple lossless audio   @tab  X  @tab  X
    @tab QuickTime fourcc 'alac'
-@item ATRAC1                 @tab     @tab  X
-@item ATRAC3                 @tab     @tab  X
+@item Atrac 1                @tab     @tab  X
+@item Atrac 3                @tab     @tab  X
@item Bink Audio             @tab     @tab  X
    @tab Used in Bink and Smacker files in many games.
@item CELT                   @tab     @tab  E
@@ -851,7 +819,6 @@ following image formats are supported:
@item DSP Group TrueSpeech   @tab     @tab  X
@item DV audio               @tab     @tab  X
@item Enhanced AC-3          @tab  X  @tab  X
-@item EVRC (Enhanced Variable Rate Codec) @tab     @tab  X
@item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
@item G.723.1                @tab X @tab X
@item G.729                  @tab     @tab  X
@@ -868,6 +835,7 @@ following image formats are supported:
@item MLP (Meridian Lossless Packing)  @tab     @tab  X
    @tab Used in DVD-Audio discs.
@item Monkey's Audio         @tab     @tab  X
+    @tab Only versions 3.97-3.99 are supported.
@item MP1 (MPEG audio layer 1)  @tab     @tab IX
@item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
    @tab libtwolame can be used alternatively for encoding.
@@ -922,7 +890,7 @@ following image formats are supported:
@item Sierra VMD audio       @tab     @tab  X
    @tab Used in Sierra VMD files.
@item Smacker audio          @tab     @tab  X
-@item SMPTE 302M AES3 audio  @tab  X  @tab  X
+@item SMPTE 302M AES3 audio  @tab     @tab  X
@item Sonic                  @tab  X  @tab  X
    @tab experimental codec
@item Sonic lossless         @tab  X  @tab  X
@@ -930,7 +898,7 @@ following image formats are supported:
@item Speex                  @tab  E  @tab  E
    @tab supported through external library libspeex
@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
-@item True Audio (TTA)       @tab  X  @tab  X
+@item True Audio (TTA)       @tab     @tab  X
@item TrueHD                 @tab     @tab  X
    @tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor)    @tab     @tab  X
@@ -938,9 +906,7 @@ following image formats are supported:
    @tab Used in LucasArts SMUSH animations.
@item Vorbis                 @tab  E  @tab  X
    @tab A native but very primitive encoder exists.
-@item Voxware MetaSound      @tab     @tab  X
-    @tab imperfect and incomplete support
-@item WavPack                @tab  X  @tab  X
+@item WavPack                @tab     @tab  X
@item Westwood Audio (SND1)  @tab     @tab  X
@item Windows Media Audio 1  @tab  X  @tab  X
@item Windows Media Audio 2  @tab  X  @tab  X
@@ -963,7 +929,6 @@ performance on systems without hardware floating point support).
@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
@item AQTitle          @tab   @tab X @tab   @tab X
@item DVB              @tab X @tab X @tab X @tab X
-@item DVB teletext     @tab   @tab X @tab   @tab E
@item DVD              @tab X @tab X @tab X @tab X
@item JACOsub          @tab X @tab X @tab   @tab X
@item MicroDVD         @tab X @tab X @tab   @tab X
@@ -980,14 +945,12 @@ performance on systems without hardware floating point support).
@item TED Talks captions @tab @tab X @tab   @tab X
@item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
@item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab X @tab X @tab   @tab X
+@item WebVTT           @tab   @tab X @tab   @tab X
@item XSUB             @tab   @tab   @tab X @tab X
@end multitable

@code{X} means that the feature is supported.

-@code{E} means that support is provided through an external library.
-
@section Network Protocols

@multitable @columnfractions .4 .1
@@ -1035,7 +998,7 @@ performance on systems without hardware floating point support).
@item OSS               @tab X      @tab X
@item Pulseaudio        @tab X      @tab
@item SDL               @tab        @tab X
-@item Video4Linux2      @tab X      @tab X
+@item Video4Linux2      @tab X      @tab
@item VfW capture       @tab X      @tab
@item X11 grabbing      @tab X      @tab
@end multitable
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -86,7 +86,7 @@ fail to open.
 Set the video size in the captured video.

@item framerate
-Set the frame rate in the captured video.
+Set the framerate in the captured video.

@item sample_rate
 Set the sample rate (in Hz) of the captured audio.
@@ -485,52 +485,87 @@ For more information about OSS see:

@section pulse

-PulseAudio input device.
+pulseaudio input device.

-To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
+To enable this input device during configuration you need libpulse-simple
+installed in your system.

 The filename to provide to the input device is a source device or the
 string "default"

-To list the PulseAudio source devices and their properties you can invoke
+To list the pulse source devices and their properties you can invoke
 the command @command{pactl list sources}.

-More information about PulseAudio can be found on @url{http://www.pulseaudio.org}.
-
-@subsection Options
-@table @option
-@item server
-Connect to a specific PulseAudio server, specified by an IP address.
-Default server is used when not provided.
-
-@item name
-Specify the application name PulseAudio will use when showing active clients,
-by default it is the @code{LIBAVFORMAT_IDENT} string.
-
-@item stream_name
-Specify the stream name PulseAudio will use when showing active streams,
-by default it is "record".
-
-@item sample_rate
-Specify the samplerate in Hz, by default 48kHz is used.
-
-@item channels
-Specify the channels in use, by default 2 (stereo) is set.
-
-@item frame_size
-Specify the number of bytes per frame, by default it is set to 1024.
-
-@item fragment_size
-Specify the minimal buffering fragment in PulseAudio, it will affect the
-audio latency. By default it is unset.
-@end table
-
-@subsection Examples
-Record a stream from default device:
@example
 ffmpeg -f pulse -i default /tmp/pulse.wav
@end example

+@subsection @var{server} AVOption
+
+The syntax is:
+@example
+-server @var{server name}
+@end example
+
+Connects to a specific server.
+
+@subsection @var{name} AVOption
+
+The syntax is:
+@example
+-name @var{application name}
+@end example
+
+Specify the application name pulse will use when showing active clients,
+by default it is the LIBAVFORMAT_IDENT string
+
+@subsection @var{stream_name} AVOption
+
+The syntax is:
+@example
+-stream_name @var{stream name}
+@end example
+
+Specify the stream name pulse will use when showing active streams,
+by default it is "record"
+
+@subsection @var{sample_rate} AVOption
+
+The syntax is:
+@example
+-sample_rate @var{samplerate}
+@end example
+
+Specify the samplerate in Hz, by default 48kHz is used.
+
+@subsection @var{channels} AVOption
+
+The syntax is:
+@example
+-channels @var{N}
+@end example
+
+Specify the channels in use, by default 2 (stereo) is set.
+
+@subsection @var{frame_size} AVOption
+
+The syntax is:
+@example
+-frame_size @var{bytes}
+@end example
+
+Specify the number of byte per frame, by default it is set to 1024.
+
+@subsection @var{fragment_size} AVOption
+
+The syntax is:
+@example
+-fragment_size @var{bytes}
+@end example
+
+Specify the minimal buffering fragment in pulseaudio, it will affect the
+audio latency. By default it is unset.
+
@section sndio

 sndio input device.
@@ -548,16 +583,10 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example

-@section video4linux2, v4l2
+@section video4linux2

 Video4Linux2 input video device.

-"v4l2" can be used as alias for "video4linux2".
-
-If FFmpeg is built with v4l-utils support (by using the
-@code{--enable-libv4l2} configure option), it is possible to use it with the
-@code{-use_libv4l2} input device option.
-
 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
 (e.g. an USB webcam) is plugged into the system, and has a name of the
@@ -565,10 +594,10 @@ kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
 the device.

 Video4Linux2 devices usually support a limited set of
-@var{width}x@var{height} sizes and frame rates. You can check which are
+@var{width}x@var{height} sizes and framerates. You can check which are
 supported using @command{-list_formats all} for Video4Linux2 devices.
-Some devices, like TV cards, support one or more standards. It is possible
-to list all the supported standards using @command{-list_standards all}.
+
+Some usage examples of the video4linux2 devices with ffmpeg and ffplay:

 The time base for the timestamps is 1 microsecond. Depending on the kernel
 version and configuration, the timestamps may be derived from the real time
@@ -577,94 +606,19 @@ boot time, unaffected by NTP or manual changes to the clock). The
@option{-timestamps abs} or @option{-ts abs} option can be used to force
 conversion into the real time clock.

-Some usage examples of the video4linux2 device with @command{ffmpeg}
-and @command{ffplay}:
-@itemize
-@item
-Grab and show the input of a video4linux2 device:
+Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
+option), it will always be used.
@example
+# Grab and show the input of a video4linux2 device.
 ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
-@end example

-@item
-Grab and record the input of a video4linux2 device, leave the
-frame rate and size as previously set:
-@example
+# Grab and record the input of a video4linux2 device, leave the
+framerate and size as previously set.
 ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
@end example
-@end itemize

-For more information about Video4Linux, check @url{http://linuxtv.org/}.
-
-@subsection Options
-
-@table @option
-@item standard
-Set the standard. Must be the name of a supported standard. To get a
-list of the supported standards, use the @option{list_standards}
-option.
-
-@item channel
-Set the input channel number. Default to -1, which means using the
-previously selected channel.
-
-@item video_size
-Set the video frame size. The argument must be a string in the form
-@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation.
-
-@item pixel_format
-Select the pixel format (only valid for raw video input).
-
-@item input_format
-Set the preferred pixel format (for raw video) or a codec name.
-This option allows to select the input format, when several are
-available.
-
-@item framerate
-Set the preferred video frame rate.
-
-@item list_formats
-List available formats (supported pixel formats, codecs, and frame
-sizes) and exit.
-
-Available values are:
-@table @samp
-@item all
-Show all available (compressed and non-compressed) formats.
-
-@item raw
-Show only raw video (non-compressed) formats.
-
-@item compressed
-Show only compressed formats.
-@end table
-
-@item list_standards
-List supported standards and exit.
-
-Available values are:
-@table @samp
-@item all
-Show all supported standards.
-@end table
-
-@item timestamps, ts
-Set type of timestamps for grabbed frames.
-
-Available values are:
-@table @samp
-@item default
-Use timestamps from the kernel.
-
-@item abs
-Use absolute timestamps (wall clock).
-
-@item mono2abs
-Force conversion from monotonic to absolute timestamps.
-@end table
-
-Default value is @code{default}.
-@end table
+"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
+"video4linux2".

@section vfwcap

@@ -701,12 +655,12 @@ properties of your X11 display (e.g. grep for "name" or "dimensions").

 For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
-ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
@end example

 Grab at position @code{10,20}:
@example
-ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
@end example

@subsection Options
@@ -727,17 +681,17 @@ zero) to the edge of region.

 For example:
@example
-ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
@end example

 To follow only when the mouse pointer reaches within 100 pixels to edge:
@example
-ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
@end example

@item framerate
 Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a frame rate of @code{30000/1001}.
+corresponding to a framerate of @code{30000/1001}.

@item show_region
 Show grabbed region on screen.
@@ -748,12 +702,12 @@ know what is being grabbed if only a portion of the screen is grabbed.

 For example:
@example
-ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
@end example

 With @var{follow_mouse}:
@example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
@end example

@item video_size
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -1,4 +1,4 @@
-FFmpeg's bug/feature request tracker manual
+FFmpeg's bug/patch/feature request tracker manual
 =================================================

 NOTE: This is a draft.
@@ -11,7 +11,7 @@ existing issues can be done through a web interface.

 Issues can be different kinds of things we want to keep track of
 but that do not belong into the source tree itself. This includes
-bug reports, feature requests and license violations. We
+bug reports, patches, feature requests and license violations. We
 might add more items to this list in the future, so feel free to
 propose a new `type of issue' on the ffmpeg-devel mailing list if
 you feel it is worth tracking.
@@ -28,9 +28,6 @@ http(s)://trac.ffmpeg.org

 Type:
 -----
-art
-    Artwork such as photos, music, banners, and logos.
-
 bug / defect
    An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
    prevents it from behaving as intended.
@@ -44,18 +41,20 @@ feature request / enhancement
 license violation
    ticket to keep track of (L)GPL violations of ffmpeg by others

-sponsoring request
-    Developer requests for hardware, software, specifications, money,
-    refunds, etc.
+patch
+    A patch as generated by diff which conforms to the patch submission and
+    development policy.
+

 Priority:
 ---------
 critical
-    Bugs about data loss and security issues.
+    Bugs and patches which deal with data loss and security issues.
    No feature request can be critical.

 important
-    Bugs which make FFmpeg unusable for a significant number of users.
+    Bugs which make FFmpeg unusable for a significant number of users, and
+    patches fixing them.
    Examples here might be completely broken MPEG-4 decoding or a build issue
    on Linux.
    While broken 4xm decoding or a broken OS/2 build would not be important,
@@ -69,7 +68,7 @@ normal


 minor
-    Bugs about things like spelling errors, "mp2" instead of
+    Bugs and patches about things like spelling errors, "mp2" instead of
    "mp3" being shown and such.
    Feature requests about things few people want or which do not make a big
    difference.
@@ -104,13 +103,13 @@ This state implicates that the bug either has been reproduced or that
 reproduction is not needed as the bug is already understood.


-Type/Status:
+Type/Status/Substatus:
 ----------
-*/new
-    Initial state of new bugs and feature requests submitted by
+*/new/new
+    Initial state of new bugs, patches and feature requests submitted by
    users.

-*/open
+*/open/open
    Issues which have been briefly looked at and which did not look outright
    invalid.
    This implicates that no real more detailed state applies yet. Conversely,
@@ -118,7 +117,9 @@ Type/Status:
    looked at.

 */closed/duplicate
-    Bugs or feature requests which are duplicates.
+    Bugs, patches or feature requests which are duplicates.
+    Note that patches dealing with the same thing in a different way are not
+    duplicates.
    Note, if you mark something as duplicate, do not forget setting the
    superseder so bug reports are properly linked.

@@ -133,7 +134,7 @@ Type/Status:
 bug/closed/fixed
    Bugs which have to the best of our knowledge been fixed.

-bug/closed/wontfix
+bug/closed/wont_fix
    Bugs which we will not fix. Possible reasons include legality, high
    complexity for the sake of supporting obscure corner cases, speed loss
    for similarly esoteric purposes, et cetera.
@@ -147,15 +148,33 @@ bug/closed/works_for_me
    reproduction failed - that is the code seems to work correctly to the
    best of our knowledge.

-feature_request/closed/fixed
+patch/open/approved
+    Patches which have been reviewed and approved by a developer.
+    Such patches can be applied anytime by any other developer after some
+    reasonable testing (compile + regression tests + does the patch do
+    what the author claimed).
+
+patch/open/needs_changes
+    Patches which have been reviewed and need changes to be accepted.
+
+patch/closed/applied
+    Patches which have been applied.
+
+patch/closed/rejected
+    Patches which have been rejected.
+
+feature_request/closed/implemented
    Feature requests which have been implemented.

-feature_request/closed/wontfix
+feature_request/closed/wont_implement
    Feature requests which will not be implemented. The reasons here could
    be legal, philosophical or others.

+Note, please do not use type-status-substatus combinations other than the
+above without asking on ffmpeg-dev first!
+
 Note2, if you provide the requested info do not forget to remove the
-needs_more_info resolution.
+needs_more_info substatus.

 Component:
 ----------
--- a/doc/libswresample.texi
+++ b/doc/libswresample.texi
@@ -20,7 +20,7 @@ Specifically, this library performs the following conversions:
@itemize
@item
@emph{Resampling}: is the process of changing the audio rate, for
-example from a high sample rate of 44100Hz to 8000Hz. Audio
+example from an high sample rate of 44100Hz to 8000Hz. Audio
 conversion from high to low sample rate is a lossy process. Several
 resampling options and algorithms are available.

--- a/doc/metadata.texi
+++ b/doc/metadata.texi
@@ -65,20 +65,4 @@ title=chapter \#1
 title=multi\
 line
@end example
-
-By using the ffmetadata muxer and demuxer it is possible to extract
-metadata from an input file to an ffmetadata file, and then transcode
-the file into an output file with the edited ffmetadata file.
-
-Extracting an ffmetadata file with @file{ffmpeg} goes as follows:
-@example
-ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE
-@end example
-
-Reinserting edited metadata information from the FFMETADATAFILE file can
-be done as:
-@example
-ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT
-@end example
-
@c man end METADATA
--- a/doc/mips.txt
+++ b/doc/mips.txt
@@ -47,14 +47,7 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
      float_dsp_mips.c
      libm_mips.h
-* libavcodec/
-      fft_fixed_32.c
-      fft_init_table.c
-      fft_table.h
-      mdct_fixed_32.c
 * libavcodec/mips/
-      aaccoder_mips.c
-      aacpsy_mips.h
      ac3dsp_mips.c
      acelp_filters_mips.c
      acelp_vectors_mips.c
@@ -70,6 +63,5 @@ Files that have MIPS copyright notice in them:
      fft_table.h
      fft_init_table.c
      fmtconvert_mips.c
-      iirfilter_mips.c
      mpegaudiodsp_mips_fixed.c
      mpegaudiodsp_mips_float.c
--- a/doc/multithreading.txt
+++ b/doc/multithreading.txt
@@ -57,11 +57,6 @@ which re-allocates them for other threads.
 Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.

-If there are inter-frame dependencies, so the codec calls
-ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
-frames must then be freed with ff_thread_release_buffer().
-Otherwise leave it at zero and decode directly into the user-supplied frames.
-
 Call ff_thread_report_progress() after some part of the current picture has decoded.
 A good place to put this is where draw_horiz_band() is called - add this if it isn't
 called anywhere, as it's useful too and the implementation is trivial when you're
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -258,15 +258,8 @@ ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example

@table @option
-@item start_number @var{number}
-Start the sequence from @var{number}. Default value is 1. Must be a
-non-negative number.
-
-@item -update @var{number}
-If @var{number} is nonzero, the filename will always be interpreted as just a
-filename, not a pattern, and this file will be continuously overwritten with new
-images.
-
+@item -start_number @var{number}
+Start the sequence from @var{number}.
@end table

 The image muxer supports the .Y.U.V image file format. This format is
@@ -275,90 +268,6 @@ each of the YUV420P components. To read or write this image file format,
 specify the name of the '.Y' file. The muxer will automatically open the
 '.U' and '.V' files as required.

-@section matroska
-
-Matroska container muxer.
-
-This muxer implements the matroska and webm container specs.
-
-The recognized metadata settings in this muxer are:
-
-@table @option
-
-@item title=@var{title name}
-Name provided to a single track
-@end table
-
-@table @option
-
-@item language=@var{language name}
-Specifies the language of the track in the Matroska languages form
-@end table
-
-@table @option
-
-@item stereo_mode=@var{mode}
-Stereo 3D video layout of two views in a single video track
-@table @option
-@item mono
-video is not stereo
-@item left_right
-Both views are arranged side by side, Left-eye view is on the left
-@item bottom_top
-Both views are arranged in top-bottom orientation, Left-eye view is at bottom
-@item top_bottom
-Both views are arranged in top-bottom orientation, Left-eye view is on top
-@item checkerboard_rl
-Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
-@item checkerboard_lr
-Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
-@item row_interleaved_rl
-Each view is constituted by a row based interleaving, Right-eye view is first row
-@item row_interleaved_lr
-Each view is constituted by a row based interleaving, Left-eye view is first row
-@item col_interleaved_rl
-Both views are arranged in a column based interleaving manner, Right-eye view is first column
-@item col_interleaved_lr
-Both views are arranged in a column based interleaving manner, Left-eye view is first column
-@item anaglyph_cyan_red
-All frames are in anaglyph format viewable through red-cyan filters
-@item right_left
-Both views are arranged side by side, Right-eye view is on the left
-@item anaglyph_green_magenta
-All frames are in anaglyph format viewable through green-magenta filters
-@item block_lr
-Both eyes laced in one Block, Left-eye view is first
-@item block_rl
-Both eyes laced in one Block, Right-eye view is first
-@end table
-@end table
-
-For example a 3D WebM clip can be created using the following command line:
-@example
-ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
-@end example
-
-This muxer supports the following options:
-
-@table @option
-
-@item reserve_index_space
-By default, this muxer writes the index for seeking (called cues in Matroska
-terms) at the end of the file, because it cannot know in advance how much space
-to leave for the index at the beginning of the file. However for some use cases
-- e.g.  streaming where seeking is possible but slow -- it is useful to put the
-index at the beginning of the file.
-
-If this option is set to a non-zero value, the muxer will reserve a given amount
-of space in the file header and then try to write the cues there when the muxing
-finishes. If the available space does not suffice, muxing will fail. A safe size
-for most use cases should be about 50kB per hour of video.
-
-Note that cues are only written if the output is seekable and this option will
-have no effect if it is not.
-
-@end table
-
@anchor{md5}
@section md5

@@ -451,11 +360,9 @@ pair for each track, making it easier to separate tracks.

 This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -movflags faststart
-Run a second pass moving the index (moov atom) to the beginning of the file.
-This operation can take a while, and will not work in various situations such
+Run a second pass moving the moov atom on top of the file. This
+operation can take a while, and will not work in various situations such
 as fragmented output, thus it is not enabled by default.
-@item -movflags rtphint
-Add RTP hinting tracks to the output file.
@end table

 Smooth Streaming content can be pushed in real time to a publishing
@@ -464,42 +371,6 @@ point on IIS with this muxer. Example:
 ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example

-@section mp3
-
-The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
-optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
-not written by default, but may be enabled with the @code{write_id3v1} option.
-
-For seekable output the muxer also writes a Xing frame at the beginning, which
-contains the number of frames in the file. It is useful for computing duration
-of VBR files.
-
-The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
-are supplied to the muxer in form of a video stream with a single packet. There
-can be any number of those streams, each will correspond to a single APIC frame.
-The stream metadata tags @var{title} and @var{comment} map to APIC
-@var{description} and @var{picture type} respectively. See
-@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
-
-Note that the APIC frames must be written at the beginning, so the muxer will
-buffer the audio frames until it gets all the pictures. It is therefore advised
-to provide the pictures as soon as possible to avoid excessive buffering.
-
-Examples:
-
-Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
-@example
-ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
-@end example
-
-To attach a picture to an mp3 file select both the audio and the picture stream
-with @code{map}:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
-@end example
-
@section mpegts

 MPEG transport stream muxer.
@@ -522,40 +393,6 @@ Set the service_id (default 0x0001) also known as program in DVB.
 Set the first PID for PMT (default 0x1000, max 0x1f00).
@item -mpegts_start_pid @var{number}
 Set the first PID for data packets (default 0x0100, max 0x0f00).
-@item -mpegts_m2ts_mode @var{number}
-Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
-@item -muxrate @var{number}
-Set muxrate.
-@item -pes_payload_size @var{number}
-Set minimum PES packet payload in bytes.
-@item -mpegts_flags @var{flags}
-Set flags (see below).
-@item -mpegts_copyts @var{number}
-Preserve original timestamps, if value is set to 1. Default value is -1, which
-results in shifting timestamps so that they start from 0.
-@item -tables_version @var{number}
-Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
-This option allows updating stream structure so that standard consumer may
-detect the change. To do so, reopen output AVFormatContext (in case of API
-usage) or restart ffmpeg instance, cyclically changing tables_version value:
-@example
-ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
-ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
-...
-ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
-ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
-ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
-...
-@end example
-@end table
-
-Option mpegts_flags may take a set of such flags:
-
-@table @option
-@item resend_headers
-Reemit PAT/PMT before writing the next packet.
-@item latm
-Use LATM packetization for AAC.
@end table

 The recognized metadata settings in mpegts muxer are @code{service_provider}
@@ -597,21 +434,69 @@ Alternatively you can write the command as:
 ffmpeg -benchmark -i INPUT -f null -
@end example

-@section ogg
+@section matroska

-Ogg container muxer.
+Matroska container muxer.
+
+This muxer implements the matroska and webm container specs.
+
+The recognized metadata settings in this muxer are:

@table @option
-@item -page_duration @var{duration}
-Preferred page duration, in microseconds. The muxer will attempt to create
-pages that are approximately @var{duration} microseconds long. This allows the
-user to compromise between seek granularity and container overhead. The default
-is 1 second. A value of 0 will fill all segments, making pages as large as
-possible. A value of 1 will effectively use 1 packet-per-page in most
-situations, giving a small seek granularity at the cost of additional container
-overhead.
+
+@item title=@var{title name}
+Name provided to a single track
@end table

+@table @option
+
+@item language=@var{language name}
+Specifies the language of the track in the Matroska languages form
+@end table
+
+@table @option
+
+@item stereo_mode=@var{mode}
+Stereo 3D video layout of two views in a single video track
+@table @option
+@item mono
+video is not stereo
+@item left_right
+Both views are arranged side by side, Left-eye view is on the left
+@item bottom_top
+Both views are arranged in top-bottom orientation, Left-eye view is at bottom
+@item top_bottom
+Both views are arranged in top-bottom orientation, Left-eye view is on top
+@item checkerboard_rl
+Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
+@item checkerboard_lr
+Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
+@item row_interleaved_rl
+Each view is constituted by a row based interleaving, Right-eye view is first row
+@item row_interleaved_lr
+Each view is constituted by a row based interleaving, Left-eye view is first row
+@item col_interleaved_rl
+Both views are arranged in a column based interleaving manner, Right-eye view is first column
+@item col_interleaved_lr
+Both views are arranged in a column based interleaving manner, Left-eye view is first column
+@item anaglyph_cyan_red
+All frames are in anaglyph format viewable through red-cyan filters
+@item right_left
+Both views are arranged side by side, Right-eye view is on the left
+@item anaglyph_green_magenta
+All frames are in anaglyph format viewable through green-magenta filters
+@item block_lr
+Both eyes laced in one Block, Left-eye view is first
+@item block_rl
+Both eyes laced in one Block, Right-eye view is first
+@end table
+@end table
+
+For example a 3D WebM clip can be created using the following command line:
+@example
+ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
+@end example
+
@section segment, stream_segment, ssegment

 Basic stream segmenter.
@@ -648,40 +533,40 @@ Set the reference stream, as specified by the string @var{specifier}.
 If @var{specifier} is set to @code{auto}, the reference is choosen
 automatically. Otherwise it must be a stream specifier (see the ``Stream
 specifiers'' chapter in the ffmpeg manual) which specifies the
-reference stream. The default value is @code{auto}.
+reference stream. The default value is ``auto''.

@item segment_format @var{format}
 Override the inner container format, by default it is guessed by the filename
 extension.
-
@item segment_list @var{name}
 Generate also a listfile named @var{name}. If not specified no
 listfile is generated.
-
@item segment_list_flags @var{flags}
 Set flags affecting the segment list generation.

 It currently supports the following flags:
-@table @samp
+@table @var
@item cache
 Allow caching (only affects M3U8 list files).

@item live
 Allow live-friendly file generation.
+
+This currently only affects M3U8 lists. In particular, write a fake
+EXT-X-TARGETDURATION duration field at the top of the file, based on
+the specified @var{segment_time}.
@end table

-Default value is @code{samp}.
+Default value is @code{cache}.

@item segment_list_size @var{size}
-Update the list file so that it contains at most the last @var{size}
-segments. If 0 the list file will contain all the segments. Default
-value is 0.
-
-@item segment_list_type @var{type}
+Overwrite the listfile once it reaches @var{size} entries. If 0
+the listfile is never overwritten. Default value is 0.
+@item segment_list type @var{type}
 Specify the format for the segment list file.

 The following values are recognized:
-@table @samp
+@table @option
@item flat
 Generate a flat list for the created segments, one segment per line.

@@ -702,36 +587,21 @@ the segment start and end time expressed in seconds.
 A list file with the suffix @code{".csv"} or @code{".ext"} will
 auto-select this format.

-@samp{ext} is deprecated in favor or @samp{csv}.
-
-@item ffconcat
-Generate an ffconcat file for the created segments. The resulting file
-can be read using the FFmpeg @ref{concat} demuxer.
-
-A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will
-auto-select this format.
+@code{ext} is deprecated in favor or @code{csv}.

@item m3u8
-Generate an extended M3U8 file, version 3, compliant with
-@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}.
+Generate an extended M3U8 file, version 4, compliant with
+@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming-08.txt}.

 A list file with the suffix @code{".m3u8"} will auto-select this format.
@end table

 If not specified the type is guessed from the list file name suffix.
-
@item segment_time @var{time}
-Set segment duration to @var{time}, the value must be a duration
-specification. Default value is "2". See also the
-@option{segment_times} option.
-
-Note that splitting may not be accurate, unless you force the
-reference stream key-frames at the given time. See the introductory
-notice and the examples below.
-
+Set segment duration to @var{time}. Default value is "2".
@item segment_time_delta @var{delta}
 Specify the accuracy time when selecting the start time for a
-segment, expressed as a duration specification. Default value is "0".
+segment. Default value is "0".

 When delta is specified a key-frame will start a new segment if its
 PTS satisfies the relation:
@@ -753,8 +623,7 @@ the specified time and the time set by @var{force_key_frames}.

@item segment_times @var{times}
 Specify a list of split points. @var{times} contains a list of comma
-separated duration specifications, in increasing order. See also
-the @option{segment_time} option.
+separated duration specifications, in increasing order.

@item segment_frames @var{frames}
 Specify a list of split video frame numbers. @var{frames} contains a
@@ -775,13 +644,9 @@ Reset timestamps at the begin of each segment, so that each segment
 will start with near-zero timestamps. It is meant to ease the playback
 of the generated segments. May not work with some combinations of
 muxers/codecs. It is set to @code{0} by default.
-
-@item initial_offset @var{offset}
-Specify timestamp offset to apply to the output packet timestamps. The
-argument must be a time duration specification, and defaults to 0.
@end table

-@subsection Examples
+@section Examples

@itemize
@item
@@ -800,9 +665,9 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_ti
@end example

@item
-As the example above, but use the @command{ffmpeg} @option{force_key_frames}
+As the example above, but use the @code{ffmpeg} @var{force_key_frames}
 option to force key frames in the input at the specified location, together
-with the segment option @option{segment_time_delta} to account for
+with the segment option @var{segment_time_delta} to account for
 possible roundings operated when setting key frame times.
@example
 ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
@@ -813,7 +678,7 @@ required.

@item
 Segment the input file by splitting the input file according to the
-frame numbers sequence specified with the @option{segment_frames} option:
+frame numbers sequence specified with the @var{segment_frames} option:
@example
 ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
@end example
@@ -834,75 +699,40 @@ ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
@end example
@end itemize

-@section tee
+@section mp3

-The tee muxer can be used to write the same data to several files or any
-other kind of muxer. It can be used, for example, to both stream a video to
-the network and save it to disk at the same time.
+The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
+optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
+@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
+not written by default, but may be enabled with the @code{write_id3v1} option.

-It is different from specifying several outputs to the @command{ffmpeg}
-command-line tool because the audio and video data will be encoded only once
-with the tee muxer; encoding can be a very expensive process. It is not
-useful when using the libavformat API directly because it is then possible
-to feed the same packets to several muxers directly.
+For seekable output the muxer also writes a Xing frame at the beginning, which
+contains the number of frames in the file. It is useful for computing duration
+of VBR files.

-The slave outputs are specified in the file name given to the muxer,
-separated by '|'. If any of the slave name contains the '|' separator,
-leading or trailing spaces or any special character, it must be
-escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
-manual).
+The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
+are supplied to the muxer in form of a video stream with a single packet. There
+can be any number of those streams, each will correspond to a single APIC frame.
+The stream metadata tags @var{title} and @var{comment} map to APIC
+@var{description} and @var{picture type} respectively. See
+@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.

-Muxer options can be specified for each slave by prepending them as a list of
-@var{key}=@var{value} pairs separated by ':', between square brackets. If
-the options values contain a special character or the ':' separator, they
-must be escaped; note that this is a second level escaping.
+Note that the APIC frames must be written at the beginning, so the muxer will
+buffer the audio frames until it gets all the pictures. It is therefore advised
+to provide the pictures as soon as possible to avoid excessive buffering.

-The following special options are also recognized:
-@table @option
-@item f
-Specify the format name. Useful if it cannot be guessed from the
-output name suffix.
+Examples:

-@item bsfs[/@var{spec}]
-Specify a list of bitstream filters to apply to the specified
-output. It is possible to specify to which streams a given bitstream
-filter applies, by appending a stream specifier to the option
-separated by @code{/}. If the stream specifier is not specified, the
-bistream filters will be applied to all streams in the output.
-
-Several bitstream filters can be specified, separated by ",".
-
-@item select
-Select the streams that should be mapped to the slave output,
-specified by a stream specifier. If not specified, this defaults to
-all the input streams.
-@end table
-
-Some examples follow.
-@itemize
-@item
-Encode something and both archive it in a WebM file and stream it
-as MPEG-TS over UDP (the streams need to be explicitly mapped):
+Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
@example
-ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
-  "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
+ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
@end example

-@item
-Use @command{ffmpeg} to encode the input, and send the output
-to three different destinations. The @code{dump_extra} bitstream
-filter is used to add extradata information to all the output video
-keyframes packets, as requested by the MPEG-TS format. The select
-option is applied to @file{out.aac} in order to make it contain only
-audio packets.
+To attach a picture to an mp3 file select both the audio and the picture stream
+with @code{map}:
@example
-ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
+ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
+-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
@end example
-@end itemize
-
-Note: some codecs may need different options depending on the output format;
-the auto-detection of this can not work with the tee muxer. The main example
-is the @option{global_header} flag.

@c man end MUXERS
--- a/doc/optimization.txt
+++ b/doc/optimization.txt
@@ -148,7 +148,7 @@ Alignment:
 Some instructions on some architectures have strict alignment restrictions,
 for example most SSE/SSE2 instructions on x86.
 The minimum guaranteed alignment is written in the .h files, for example:
-    void (*put_pixels_clamped)(const int16_t *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
+    void (*put_pixels_clamped)(const DCTELEM *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);


 General Tips:
--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -1,7 +1,7 @@
@chapter Output Devices
@c man begin OUTPUT DEVICES

-Output devices are configured elements in FFmpeg that can write
+Output devices are configured elements in FFmpeg which allow to write
 multimedia data to an output device attached to your system.

 When you configure your FFmpeg build, all the supported output devices
@@ -26,7 +26,7 @@ ALSA (Advanced Linux Sound Architecture) output device.

 CACA output device.

-This output device allows to show a video stream in CACA window.
+This output devices allows to show a video stream in CACA window.
 Only one CACA window is allowed per application, so you can
 have only one instance of this output device in an application.

@@ -104,79 +104,15 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
@end example
@end itemize

-@section fbdev
-
-Linux framebuffer output device.
-
-The Linux framebuffer is a graphic hardware-independent abstraction
-layer to show graphics on a computer monitor, typically on the
-console. It is accessed through a file device node, usually
-@file{/dev/fb0}.
-
-For more detailed information read the file
-@file{Documentation/fb/framebuffer.txt} included in the Linux source tree.
-
-@subsection Options
-@table @option
-
-@item xoffset
-@item yoffset
-Set x/y coordinate of top left corner. Default is 0.
-@end table
-
-@subsection Examples
-Play a file on framebuffer device @file{/dev/fb0}.
-Required pixel format depends on current framebuffer settings.
-@example
-ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
-@end example
-
-See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
-
@section oss

 OSS (Open Sound System) output device.

-@section pulse
-
-PulseAudio output device.
-
-To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
-
-More information about PulseAudio can be found on @url{http://www.pulseaudio.org}
-
-@subsection Options
-@table @option
-
-@item server
-Connect to a specific PulseAudio server, specified by an IP address.
-Default server is used when not provided.
-
-@item name
-Specify the application name PulseAudio will use when showing active clients,
-by default it is the @code{LIBAVFORMAT_IDENT} string.
-
-@item stream_name
-Specify the stream name PulseAudio will use when showing active streams,
-by default it is set to the specified output name.
-
-@item device
-Specify the device to use. Default device is used when not provided.
-List of output devices can be obtained with command @command{pactl list sinks}.
-
-@end table
-
-@subsection Examples
-Play a file on default device on default server:
-@example
-ffmpeg  -i INPUT -f pulse "stream name"
-@end example
-
@section sdl

 SDL (Simple DirectMedia Layer) output device.

-This output device allows to show a video stream in an SDL
+This output devices allows to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
 have only one instance of this output device in an application.

@@ -203,10 +139,6 @@ Set the SDL window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
 If not specified it defaults to the size of the input video,
 downscaled according to the aspect ratio.
-
-@item window_fullscreen
-Set fullscreen mode when non-zero value is provided.
-Zero is a default.
@end table

@subsection Examples
@@ -221,69 +153,4 @@ ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL

 sndio audio output device.

-@section xv
-
-XV (XVideo) output device.
-
-This output device allows to show a video stream in a X Window System
-window.
-
-@subsection Options
-
-@table @option
-@item display_name
-Specify the hardware display name, which determines the display and
-communications domain to be used.
-
-The display name or DISPLAY environment variable can be a string in
-the format @var{hostname}[:@var{number}[.@var{screen_number}]].
-
-@var{hostname} specifies the name of the host machine on which the
-display is physically attached. @var{number} specifies the number of
-the display server on that host machine. @var{screen_number} specifies
-the screen to be used on that server.
-
-If unspecified, it defaults to the value of the DISPLAY environment
-variable.
-
-For example, @code{dual-headed:0.1} would specify screen 1 of display
-0 on the machine named ``dual-headed''.
-
-Check the X11 specification for more detailed information about the
-display name format.
-
-@item window_size
-Set the created window size, can be a string of the form
-@var{width}x@var{height} or a video size abbreviation. If not
-specified it defaults to the size of the input video.
-
-@item window_x
-@item window_y
-Set the X and Y window offsets for the created window. They are both
-set to 0 by default. The values may be ignored by the window manager.
-
-@item window_title
-Set the window title, if not specified default to the filename
-specified for the output device.
-@end table
-
-For more information about XVideo see @url{http://www.x.org/}.
-
-@subsection Examples
-
-@itemize
-@item
-Decode, display and encode video input with @command{ffmpeg} at the
-same time:
-@example
-ffmpeg -i INPUT OUTPUT -f xv display
-@end example
-
-@item
-Decode and display the input video to multiple X11 windows:
-@example
-ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
-@end example
-@end itemize
-
@c man end OUTPUT DEVICES
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -106,10 +106,10 @@ libavformat) as DLLs.

@end itemize

-@section Microsoft Visual C++ or Intel C++ Compiler for Windows
+@section Microsoft Visual C++

-FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
-wrapper. For ICL, only the wrapper is used, since ICL supports C99.
+FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and
+wrapper.

 You will need the following prerequisites:

@@ -122,33 +122,28 @@ You will need the following prerequisites:
 you want to run @uref{fate.html, FATE}.
@end itemize

-To set up a proper environment in MSYS, you need to run @code{msys.bat} from
-the Visual Studio or Intel Compiler command prompt.
+To set up a proper MSVC environment in MSYS, you simply need to run
+@code{msys.bat} from the Visual Studio command prompt.

 Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
 somewhere in your @code{PATH}.

 Next, make sure @code{inttypes.h} and any other headers and libs you want to use
-are located in a spot that the compiler can see. Do so by modifying the @code{LIB}
-and @code{INCLUDE} environment variables to include the @strong{Windows} paths to
+are located in a spot that MSVC can see. Do so by modifying the @code{LIB} and
+@code{INCLUDE} environment variables to include the @strong{Windows} paths to
 these directories. Alternatively, you can try and use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options.

 Finally, run:

@example
-For MSVC:
 ./configure --toolchain=msvc
-
-For ICL:
-./configure --toolchain=icl
-
 make
 make install
@end example

 If you wish to compile shared libraries, add @code{--enable-shared} to your
-configure options. Note that due to the way MSVC and ICL handle DLL imports and
+configure options. Note that due to the way MSVC handles DLL imports and
 exports, you cannot compile static and shared libraries at the same time, and
 enabling shared libraries will automatically disable the static ones.

@@ -178,12 +173,7 @@ erroneously included when building FFmpeg.
 can see.
@end enumerate

-@item FFmpeg has been tested with the following on i686 and x86_64:
-@itemize
-@item Visual Studio 2010 Pro and Express
-@item Visual Studio 2012 Pro and Express
-@item Intel Composer XE 2013
-@end itemize
+@item FFmpeg has been tested with Visual Studio 2010 and 2012, Pro and Express.
 Anything else is not officially supported.

@end itemize
@@ -194,7 +184,16 @@ If you plan to link with MSVC-built static libraries, you will need
 to make sure you have @code{Runtime Library} set to
@code{Multi-threaded (/MT)} in your project's settings.

-You will need to define @code{inline} to something MSVC understands:
+FFmpeg headers do not declare global data for Windows DLLs through the usual
+dllexport/dllimport interface. Such data will be exported properly while
+building, but to use them in your MSVC code you will have to edit the
+appropriate headers and mark the data as dllimport. For example, in
+libavutil/pixdesc.h you should have:
+@example
+extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
+@end example
+
+You will also need to define @code{inline} to something MSVC understands:
@example
 #define inline __inline
@end example
--- a/doc/print_options.c
+++ b/doc/print_options.c
@@ -1,20 +1,20 @@
 /*
 * Copyright (c) 2012 Anton Khirnov
 *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
 *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

@@ -27,9 +27,7 @@
 #include <float.h>

 #include "libavformat/avformat.h"
-#include "libavformat/options_table.h"
 #include "libavcodec/avcodec.h"
-#include "libavcodec/options_table.h"
 #include "libavutil/opt.h"

 static void print_usage(void)
@@ -41,9 +39,6 @@ static void print_usage(void)

 static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
 {
-    if (!(o->flags & (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM)))
-        return;
-
    printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
    switch (o->type) {
    case AV_OPT_TYPE_BINARY:   printf("hexadecimal string"); break;
@@ -98,14 +93,18 @@ static void show_opts(const AVOption *opts, int per_stream)

 static void show_format_opts(void)
 {
+#include "libavformat/options_table.h"
+
    printf("@section Format AVOptions\n");
-    show_opts(avformat_options, 0);
+    show_opts(options, 0);
 }

 static void show_codec_opts(void)
 {
+#include "libavcodec/options_table.h"
+
    printf("@section Codec AVOptions\n");
-    show_opts(avcodec_options, 1);
+    show_opts(options, 1);
 }

 int main(int argc, char **argv)
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -1,8 +1,8 @@
@chapter Protocols
@c man begin PROTOCOLS

-Protocols are configured elements in FFmpeg that enable access to
-resources that require specific protocols.
+Protocols are configured elements in FFmpeg which allow to access
+resources which require the use of a particular protocol.

 When you configure your FFmpeg build, all the supported protocols are
 enabled by default. You can list all available ones using the
@@ -49,16 +49,6 @@ Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapte
 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
@end example

-@section cache
-
-Caching wrapper for input stream.
-
-Cache the input stream to temporary file. It brings seeking capability to live streams.
-
-@example
-cache:@var{URL}
-@end example
-
@section concat

 Physical concatenation protocol.
@@ -85,25 +75,6 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 Note that you may need to escape the character "|" which is special for
 many shells.

-@section crypto
-
-AES-encrypted stream reading protocol.
-
-The accepted options are:
-@table @option
-@item key
-Set the AES decryption key binary block from given hexadecimal representation.
-
-@item iv
-Set the AES decryption initialization vector binary block from given hexadecimal representation.
-@end table
-
-Accepted URL formats:
-@example
-crypto:@var{URL}
-crypto+@var{URL}
-@end example
-
@section data

 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
@@ -129,54 +100,6 @@ The ff* tools default to the file protocol, that is a resource
 specified with the name "FILE.mpeg" is interpreted as the URL
 "file:FILE.mpeg".

-This protocol accepts the following options:
-
-@table @option
-@item truncate
-Truncate existing files on write, if set to 1. A value of 0 prevents
-truncating. Default value is 1.
-
-@item blocksize
-Set I/O operation maximum block size, in bytes. Default value is
-@code{INT_MAX}, which results in not limiting the requested block size.
-Setting this value reasonably low improves user termination request reaction
-time, which is valuable for files on slow medium.
-@end table
-
-@section ftp
-
-FTP (File Transfer Protocol).
-
-Allow to read from or write to remote resources using FTP protocol.
-
-Following syntax is required.
-@example
-ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
-@end example
-
-This protocol accepts the following options.
-
-@table @option
-@item timeout
-Set timeout of socket I/O operations used by the underlying low level
-operation. By default it is set to -1, which means that the timeout is
-not specified.
-
-@item ftp-anonymous-password
-Password used when login as anonymous user. Typically an e-mail address
-should be used.
-
-@item ftp-write-seekable
-Control seekability of connection during encoding. If set to 1 the
-resource is supposed to be seekable, if set to 0 it is assumed not
-to be seekable. Default value is 0.
-@end table
-
-NOTE: Protocol can be used as output, but it is recommended to not do
-it, unless special care is taken (tests, customized server configuration
-etc.). Different FTP servers behave in different way during seek
-operation. ff* tools may produce incomplete content due to server limitations.
-
@section gopher

 Gopher protocol.
@@ -205,77 +128,6 @@ m3u8 files.

 HTTP (Hyper Text Transfer Protocol).

-This protocol accepts the following options.
-
-@table @option
-@item seekable
-Control seekability of connection. If set to 1 the resource is
-supposed to be seekable, if set to 0 it is assumed not to be seekable,
-if set to -1 it will try to autodetect if it is seekable. Default
-value is -1.
-
-@item chunked_post
-If set to 1 use chunked transfer-encoding for posts, default is 1.
-
-@item headers
-Set custom HTTP headers, can override built in default headers. The
-value must be a string encoding the headers.
-
-@item content_type
-Force a content type.
-
-@item user-agent
-Override User-Agent header. If not specified the protocol will use a
-string describing the libavformat build.
-
-@item multiple_requests
-Use persistent connections if set to 1. By default it is 0.
-
-@item post_data
-Set custom HTTP post data.
-
-@item timeout
-Set timeout of socket I/O operations used by the underlying low level
-operation. By default it is set to -1, which means that the timeout is
-not specified.
-
-@item mime_type
-Set MIME type.
-
-@item icy
-If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
-supports this, the metadata has to be retrieved by the application by reading
-the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
-The default is 0.
-
-@item icy_metadata_headers
-If the server supports ICY metadata, this contains the ICY specific HTTP reply
-headers, separated with newline characters.
-
-@item icy_metadata_packet
-If the server supports ICY metadata, and @option{icy} was set to 1, this
-contains the last non-empty metadata packet sent by the server.
-
-@item cookies
-Set the cookies to be sent in future requests. The format of each cookie is the
-same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
-delimited by a newline character.
-@end table
-
-@subsection HTTP Cookies
-
-Some HTTP requests will be denied unless cookie values are passed in with the
-request. The @option{cookies} option allows these cookies to be specified. At
-the very least, each cookie must specify a value along with a path and domain.
-HTTP requests that match both the domain and path will automatically include the
-cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
-by a newline.
-
-The required syntax to play a stream specifying a cookie is:
-@example
-ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
-@end example
-
@section mmst

 MMS (Microsoft Media Server) protocol over TCP.
@@ -339,16 +191,6 @@ ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
@end example

-This protocol accepts the following options:
-
-@table @option
-@item blocksize
-Set I/O operation maximum block size, in bytes. Default value is
-@code{INT_MAX}, which results in not limiting the requested block size.
-Setting this value reasonably low improves user termination request reaction
-time, which is valuable if data transmission is slow.
-@end table
-
 Note that some formats (typically MOV), require the output protocol to
 be seekable, so they will fail with the pipe output protocol.

@@ -361,18 +203,12 @@ content across a TCP/IP network.

 The required syntax is:
@example
-rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
+rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
@end example

 The accepted parameters are:
@table @option

-@item username
-An optional username (mostly for publishing).
-
-@item password
-An optional password (mostly for publishing).
-
@item server
 The address of the RTMP server.

@@ -423,8 +259,7 @@ times to construct arbitrary AMF sequences.

@item rtmp_flashver
 Version of the Flash plugin used to run the SWF player. The default
-is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
-<libavformat version>).)
+is LNX 9,0,124,2.

@item rtmp_flush_interval
 Number of packets flushed in the same request (RTMPT only). The default
@@ -474,12 +309,6 @@ For example to read with @command{ffplay} a multimedia resource named
 ffplay rtmp://myserver/vod/sample
@end example

-To publish to a password protected server, passing the playpath and
-app names separately:
-@example
-ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
-@end example
-
@section rtmpe

 Encrypted Real-Time Messaging Protocol.
@@ -520,39 +349,7 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
 for streaming multimedia content within HTTPS requests to traverse
 firewalls.

-@section libssh
-
-Secure File Transfer Protocol via libssh
-
-Allow to read from or write to remote resources using SFTP protocol.
-
-Following syntax is required.
-
-@example
-sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
-@end example
-
-This protocol accepts the following options.
-
-@table @option
-@item timeout
-Set timeout of socket I/O operations used by the underlying low level
-operation. By default it is set to -1, which means that the timeout
-is not specified.
-
-@item truncate
-Truncate existing files on write, if set to 1. A value of 0 prevents
-truncating. Default value is 1.
-
-@end table
-
-Example: Play a file stored on remote server.
-
-@example
-ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
-@end example
-
-@section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
+@section rtmp, rtmpe, rtmps, rtmpt, rtmpte

 Real-Time Messaging Protocol and its variants supported through
 librtmp.
@@ -594,70 +391,7 @@ ffplay "rtmp://myserver/live/mystream live=1"

@section rtp

-Real-time Transport Protocol.
-
-The required syntax for an RTP URL is:
-rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
-
-@var{port} specifies the RTP port to use.
-
-The following URL options are supported:
-
-@table @option
-
-@item ttl=@var{n}
-Set the TTL (Time-To-Live) value (for multicast only).
-
-@item rtcpport=@var{n}
-Set the remote RTCP port to @var{n}.
-
-@item localrtpport=@var{n}
-Set the local RTP port to @var{n}.
-
-@item localrtcpport=@var{n}'
-Set the local RTCP port to @var{n}.
-
-@item pkt_size=@var{n}
-Set max packet size (in bytes) to @var{n}.
-
-@item connect=0|1
-Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
-to 0).
-
-@item sources=@var{ip}[,@var{ip}]
-List allowed source IP addresses.
-
-@item block=@var{ip}[,@var{ip}]
-List disallowed (blocked) source IP addresses.
-
-@item write_to_source=0|1
-Send packets to the source address of the latest received packet (if
-set to 1) or to a default remote address (if set to 0).
-
-@item localport=@var{n}
-Set the local RTP port to @var{n}.
-
-This is a deprecated option. Instead, @option{localrtpport} should be
-used.
-
-@end table
-
-Important notes:
-
-@enumerate
-
-@item
-If @option{rtcpport} is not set the RTCP port will be set to the RTP
-port value plus 1.
-
-@item
-If @option{localrtpport} (the local RTP port) is not set any available
-port will be used for the local RTP and RTCP ports.
-
-@item
-If @option{localrtcpport} (the local RTCP port) is not set it will be
-set to the the local RTP port value plus 1.
-@end enumerate
+Real-Time Protocol.

@section rtsp

@@ -747,11 +481,6 @@ To receive a stream in realtime:
 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@end example

-@table @option
-@item stimeout
-Socket IO timeout in micro seconds.
-@end table
-
@section sap

 Session Announcement Protocol (RFC 2974). This is not technically a
@@ -844,50 +573,6 @@ To play back the first stream announced on one the default IPv6 SAP multicast ad
 ffplay sap://[ff0e::2:7ffe]
@end example

-@section sctp
-
-Stream Control Transmission Protocol.
-
-The accepted URL syntax is:
-@example
-sctp://@var{host}:@var{port}[?@var{options}]
-@end example
-
-The protocol accepts the following options:
-@table @option
-@item listen
-If set to any value, listen for an incoming connection. Outgoing connection is done by default.
-
-@item max_streams
-Set the maximum number of streams. By default no limit is set.
-@end table
-
-@section srtp
-
-Secure Real-time Transport Protocol.
-
-The accepted options are:
-@table @option
-@item srtp_in_suite
-@item srtp_out_suite
-Select input and output encoding suites.
-
-Supported values:
-@table @samp
-@item AES_CM_128_HMAC_SHA1_80
-@item SRTP_AES128_CM_HMAC_SHA1_80
-@item AES_CM_128_HMAC_SHA1_32
-@item SRTP_AES128_CM_HMAC_SHA1_32
-@end table
-
-@item srtp_in_params
-@item srtp_out_params
-Set input and output encoding parameters, which are expressed by a
-base64-encoded representation of a binary block. The first 16 bytes of
-this binary block are used as master key, the following 14 bytes are
-used as master salt.
-@end table
-
@section tcp

 Trasmission Control Protocol.
@@ -916,48 +601,29 @@ ffplay tcp://@var{hostname}:@var{port}

@section tls

-Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
+Transport Layer Security/Secure Sockets Layer

 The required syntax for a TLS/SSL url is:
@example
 tls://@var{hostname}:@var{port}[?@var{options}]
@end example

-The following parameters can be set via command line options
-(or in code via @code{AVOption}s):
-
@table @option

-@item ca_file, cafile=@var{filename}
-A file containing certificate authority (CA) root certificates to treat
-as trusted. If the linked TLS library contains a default this might not
-need to be specified for verification to work, but not all libraries and
-setups have defaults built in.
-The file must be in OpenSSL PEM format.
+@item listen
+Act as a server, listening for an incoming connection.

-@item tls_verify=@var{1|0}
-If enabled, try to verify the peer that we are communicating with.
-Note, if using OpenSSL, this currently only makes sure that the
-peer certificate is signed by one of the root certificates in the CA
-database, but it does not validate that the certificate actually
-matches the host name we are trying to connect to. (With GnuTLS,
-the host name is validated as well.)
+@item cafile=@var{filename}
+Certificate authority file. The file must be in OpenSSL PEM format.

-This is disabled by default since it requires a CA database to be
-provided by the caller in many cases.
+@item cert=@var{filename}
+Certificate file. The file must be in OpenSSL PEM format.

-@item cert_file, cert=@var{filename}
-A file containing a certificate to use in the handshake with the peer.
-(When operating as server, in listen mode, this is more often required
-by the peer, while client certificates only are mandated in certain
-setups.)
+@item key=@var{filename}
+Private key file.

-@item key_file, key=@var{filename}
-A file containing the private key for the certificate.
-
-@item listen=@var{1|0}
-If enabled, listen for connections on the provided port, and assume
-the server role in the handshake instead of the client role.
+@item verify=@var{0|1}
+Verify the peer's certificate.

@end table

@@ -1064,24 +730,4 @@ To receive over UDP from a remote endpoint:
 ffmpeg -i udp://[@var{multicast-address}]:@var{port}
@end example

-@section unix
-
-Unix local socket
-
-The required syntax for a Unix socket URL is:
-
-@example
-unix://@var{filepath}
-@end example
-
-The following parameters can be set via command line options
-(or in code via @code{AVOption}s):
-
-@table @option
-@item timeout
-Timeout in ms.
-@item listen
-Create the Unix socket in listening mode.
-@end table
-
@c man end PROTOCOLS
--- a/doc/resampler.texi
+++ b/doc/resampler.texi
@@ -1,232 +0,0 @@
-@chapter Resampler Options
-@c man begin RESAMPLER OPTIONS
-
-The audio resampler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, @var{option}=@var{value} for the aresample filter,
-by setting the value explicitly in the
-@code{SwrContext} options or using the @file{libavutil/opt.h} API for
-programmatic use.
-
-@table @option
-
-@item ich, in_channel_count
-Set the number of input channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{in_channel_layout} is set.
-
-@item och, out_channel_count
-Set the number of output channels. Default value is 0. Setting this
-value is not mandatory if the corresponding channel layout
-@option{out_channel_layout} is set.
-
-@item uch, used_channel_count
-Set the number of used input channels. Default value is 0. This option is
-only used for special remapping.
-
-@item isr, in_sample_rate
-Set the input sample rate. Default value is 0.
-
-@item osr, out_sample_rate
-Set the output sample rate. Default value is 0.
-
-@item isf, in_sample_fmt
-Specify the input sample format. It is set by default to @code{none}.
-
-@item osf, out_sample_fmt
-Specify the output sample format. It is set by default to @code{none}.
-
-@item tsf, internal_sample_fmt
-Set the internal sample format. Default value is @code{none}.
-This will automatically be chosen when it is not explicitly set.
-
-@item icl, in_channel_layout
-@item ocl, out_channel_layout
-Set the input/output channel layout.
-
-See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
-for the required syntax.
-
-@item clev, center_mix_level
-Set the center mix level. It is a value expressed in deciBel, and must be
-in the interval [-32,32].
-
-@item slev, surround_mix_level
-Set the surround mix level. It is a value expressed in deciBel, and must
-be in the interval [-32,32].
-
-@item lfe_mix_level
-Set LFE mix into non LFE level. It is used when there is a LFE input but no
-LFE output. It is a value expressed in deciBel, and must
-be in the interval [-32,32].
-
-@item rmvol, rematrix_volume
-Set rematrix volume. Default value is 1.0.
-
-@item rematrix_maxval
-Set maximum output value for rematrixing.
-This can be used to prevent clipping vs. preventing volumn reduction
-A value of 1.0 prevents cliping.
-
-@item flags, swr_flags
-Set flags used by the converter. Default value is 0.
-
-It supports the following individual flags:
-@table @option
-@item res
-force resampling, this flag forces resampling to be used even when the
-input and output sample rates match.
-@end table
-
-@item dither_scale
-Set the dither scale. Default value is 1.
-
-@item dither_method
-Set dither method. Default value is 0.
-
-Supported values:
-@table @samp
-@item rectangular
-select rectangular dither
-@item triangular
-select triangular dither
-@item triangular_hp
-select triangular dither with high pass
-@item lipshitz
-select lipshitz noise shaping dither
-@item shibata
-select shibata noise shaping dither
-@item low_shibata
-select low shibata noise shaping dither
-@item high_shibata
-select high shibata noise shaping dither
-@item f_weighted
-select f-weighted noise shaping dither
-@item modified_e_weighted
-select modified-e-weighted noise shaping dither
-@item improved_e_weighted
-select improved-e-weighted noise shaping dither
-
-@end table
-
-@item resampler
-Set resampling engine. Default value is swr.
-
-Supported values:
-@table @samp
-@item swr
-select the native SW Resampler; filter options precision and cheby are not
-applicable in this case.
-@item soxr
-select the SoX Resampler (where available); compensation, and filter options
-filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
-case.
-@end table
-
-@item filter_size
-For swr only, set resampling filter size, default value is 32.
-
-@item phase_shift
-For swr only, set resampling phase shift, default value is 10, and must be in
-the interval [0,30].
-
-@item linear_interp
-Use Linear Interpolation if set to 1, default value is 0.
-
-@item cutoff
-Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
-value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
-(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
-
-@item precision
-For soxr only, the precision in bits to which the resampled signal will be
-calculated.  The default value of 20 (which, with suitable dithering, is
-appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
-value of 28 gives SoX's 'Very High Quality'.
-
-@item cheby
-For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
-approximation for 'irrational' ratios. Default value is 0.
-
-@item async
-For swr only, simple 1 parameter audio sync to timestamps using stretching,
-squeezing, filling and trimming. Setting this to 1 will enable filling and
-trimming, larger values represent the maximum amount in samples that the data
-may be stretched or squeezed for each second.
-Default value is 0, thus no compensation is applied to make the samples match
-the audio timestamps.
-
-@item first_pts
-For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
-This allows for padding/trimming at the start of stream. By default, no
-assumption is made about the first frame's expected pts, so no padding or
-trimming is done. For example, this could be set to 0 to pad the beginning with
-silence if an audio stream starts after the video stream or to trim any samples
-with a negative pts due to encoder delay.
-
-@item min_comp
-For swr only, set the minimum difference between timestamps and audio data (in
-seconds) to trigger stretching/squeezing/filling or trimming of the
-data to make it match the timestamps. The default is that
-stretching/squeezing/filling and trimming is disabled
-(@option{min_comp} = @code{FLT_MAX}).
-
-@item min_hard_comp
-For swr only, set the minimum difference between timestamps and audio data (in
-seconds) to trigger adding/dropping samples to make it match the
-timestamps.  This option effectively is a threshold to select between
-hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
-all compensation is by default disabled through @option{min_comp}.
-The default is 0.1.
-
-@item comp_duration
-For swr only, set duration (in seconds) over which data is stretched/squeezed
-to make it match the timestamps. Must be a non-negative double float value,
-default value is 1.0.
-
-@item max_soft_comp
-For swr only, set maximum factor by which data is stretched/squeezed to make it
-match the timestamps. Must be a non-negative double float value, default value
-is 0.
-
-@item matrix_encoding
-Select matrixed stereo encoding.
-
-It accepts the following values:
-@table @samp
-@item none
-select none
-@item dolby
-select Dolby
-@item dplii
-select Dolby Pro Logic II
-@end table
-
-Default value is @code{none}.
-
-@item filter_type
-For swr only, select resampling filter type. This only affects resampling
-operations.
-
-It accepts the following values:
-@table @samp
-@item cubic
-select cubic
-@item blackman_nuttall
-select Blackman Nuttall Windowed Sinc
-@item kaiser
-select Kaiser Windowed Sinc
-@end table
-
-@item kaiser_beta
-For swr only, set Kaiser Window Beta value. Must be an integer in the
-interval [2,16], default value is 9.
-
-@item output_sample_bits
-For swr only, set number of used output sample bits for dithering. Must be an integer in the
-interval [0,64], default value is 0, which means it's not used.
-
-@end table
-
-@c man end RESAMPLER OPTIONS
--- a/doc/scaler.texi
+++ b/doc/scaler.texi
@@ -1,119 +0,0 @@
-@anchor{scaler_options}
-@chapter Scaler Options
-@c man begin SCALER OPTIONS
-
-The video scaler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools. For programmatic use, they can be set explicitly in the
-@code{SwsContext} options or through the @file{libavutil/opt.h} API.
-
-@table @option
-
-@anchor{sws_flags}
-@item sws_flags
-Set the scaler flags. This is also used to set the scaling
-algorithm. Only a single algorithm should be selected.
-
-It accepts the following values:
-@table @samp
-@item fast_bilinear
-Select fast bilinear scaling algorithm.
-
-@item bilinear
-Select bilinear scaling algorithm.
-
-@item bicubic
-Select bicubic scaling algorithm.
-
-@item experimental
-Select experimental scaling algorithm.
-
-@item neighbor
-Select nearest neighbor rescaling algorithm.
-
-@item area
-Select averaging area rescaling algorithm.
-
-@item bicubiclin
-Select bicubic scaling algorithm for the luma component, bilinear for
-chroma components.
-
-@item gauss
-Select Gaussian rescaling algorithm.
-
-@item sinc
-Select sinc rescaling algorithm.
-
-@item lanczos
-Select lanczos rescaling algorithm.
-
-@item spline
-Select natural bicubic spline rescaling algorithm.
-
-@item print_info
-Enable printing/debug logging.
-
-@item accurate_rnd
-Enable accurate rounding.
-
-@item full_chroma_int
-Enable full chroma interpolation.
-
-@item full_chroma_inp
-Select full chroma input.
-
-@item bitexact
-Enable bitexact output.
-@end table
-
-@item srcw
-Set source width.
-
-@item srch
-Set source height.
-
-@item dstw
-Set destination width.
-
-@item dsth
-Set destination height.
-
-@item src_format
-Set source pixel format (must be expressed as an integer).
-
-@item dst_format
-Set destination pixel format (must be expressed as an integer).
-
-@item src_range
-Select source range.
-
-@item dst_range
-Select destination range.
-
-@item param0, param1
-Set scaling algorithm parameters. The specified values are specific of
-some scaling algorithms and ignored by others. The specified values
-are floating point number values.
-
-@item sws_dither
-Set the dithering algorithm. Accepts one of the following
-values. Default value is @samp{auto}.
-
-@table @samp
-@item auto
-automatic choice
-
-@item none
-no dithering
-
-@item bayer
-bayer dither
-
-@item ed
-error diffusion dither
-@end table
-
-@end table
-
-@c man end SCALER OPTIONS
--- a/doc/snow.txt
+++ b/doc/snow.txt
@@ -50,10 +50,8 @@ header:
        temporal_decomposition_count    u   header_state
        spatial_decomposition_count     u   header_state
        colorspace_type                 u   header_state
-        if (nb_planes > 2) {
-            chroma_h_shift              u   header_state
-            chroma_v_shift              u   header_state
-        }
+        chroma_h_shift                  u   header_state
+        chroma_v_shift                  u   header_state
        spatial_scalability             b   header_state
        max_ref_frames-1                u   header_state
        qlogs
@@ -61,7 +59,7 @@ header:
    if(!keyframe){
        update_mc                       b   header_state
        if(update_mc){
-            for(plane=0; plane<nb_plane_types; plane++){
+            for(plane=0; plane<2; plane++){
                diag_mc                 b   header_state
                htaps/2-1               u   header_state
                for(i= p->htaps/2; i; i--)
@@ -82,7 +80,7 @@ header:
    block_max_depth                     s   header_state

 qlogs:
-    for(plane=0; plane<nb_plane_types; plane++){
+    for(plane=0; plane<2; plane++){
        quant_table[plane][0][0]        s   header_state
        for(level=0; level < spatial_decomposition_count; level++){
            quant_table[plane][level][1]s   header_state
@@ -133,10 +131,8 @@ block(level):

 residual:
    residual2(luma)
-    if (nb_planes > 2) {
-        residual2(chroma_cr)
-        residual2(chroma_cb)
-    }
+    residual2(chroma_cr)
+    residual2(chroma_cb)

 residual2:
    for(level=0; level<spatial_decomposition_count; level++){
@@ -150,7 +146,7 @@ residual2:
 subband:
    FIXME

-nb_plane_types = gray ? 1 : 2;
+

 Tag description:
 ----------------
@@ -172,11 +168,7 @@ spatial_decomposition_count
    FIXME

 colorspace_type
-    0   unspecified YcbCr
-    1   Gray
-    2   Gray + Alpha
-    3   GBR
-    4   GBRA
+    0
    this MUST NOT change within a bitstream

 chroma_h_shift
--- a/doc/swresample.txt
+++ b/doc/swresample.txt
@@ -32,9 +32,9 @@ Special Converter         v
                        Output

 Planar/Packed conversion is done when needed during sample format conversion.
-Every step can be skipped without memcpy when it is not needed.
+Every step can be skipped without memcpy when its not needed.
 Either Resampling and Rematrixing can be performed first depending on which
-way it is faster.
+way its faster.
 The Buffers are needed for resampling due to resamplng being a process that
 requires future and past data, it thus also introduces inevitably a delay when
 used.
--- a/doc/syntax.texi
+++ b/doc/syntax.texi
@@ -0,0 +1,230 @@
+@chapter Syntax
+@c man begin SYNTAX
+
+This section documents the syntax and formats employed by the FFmpeg
+libraries and tools.
+
+@anchor{quoting_and_escaping}
+@section Quoting and escaping
+
+FFmpeg adopts the following quoting and escaping mechanism, unless
+explicitly specified. The following rules are applied:
+
+@itemize
+@item
+@code{'} and @code{\} are special characters (respectively used for
+quoting and escaping). In addition to them, there might be other
+special characters depending on the specific syntax where the escaping
+and quoting are employed.
+
+@item
+A special character is escaped by prefixing it with a '\'.
+
+@item
+All characters enclosed between '' are included literally in the
+parsed string. The quote character @code{'} itself cannot be quoted,
+so you may need to close the quote and escape it.
+
+@item
+Leading and trailing whitespaces, unless escaped or quoted, are
+removed from the parsed string.
+@end itemize
+
+Note that you may need to add a second level of escaping when using
+the command line or a script, which depends on the syntax of the
+adopted shell language.
+
+The function @code{av_get_token} defined in
+@file{libavutil/avstring.h} can be used to parse a token quoted or
+escaped according to the rules defined above.
+
+The tool @file{tools/ffescape} in the FFmpeg source tree can be used
+to automatically quote or escape a string in a script.
+
+@subsection Examples
+
+@itemize
+@item
+Escape the string @code{Crime d'Amour} containing the @code{'} special
+character:
+@example
+Crime d\'Amour
+@end example
+
+@item
+The string above contains a quote, so the @code{'} needs to be escaped
+when quoting it:
+@example
+'Crime d'\''Amour'
+@end example
+
+@item
+Include leading or trailing whitespaces using quoting:
+@example
+'  this string starts and ends with whitespaces  '
+@end example
+
+@item
+Escaping and quoting can be mixed together:
+@example
+' The string '\'string\'' is a string '
+@end example
+
+@item
+To include a literal @code{\} you can use either escaping or quoting:
+@example
+'c:\foo' can be written as c:\\foo
+@end example
+@end itemize
+
+@anchor{date syntax}
+@section Date
+
+The accepted syntax is:
+@example
+[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
+now
+@end example
+
+If the value is "now" it takes the current time.
+
+Time is local time unless Z is appended, in which case it is
+interpreted as UTC.
+If the year-month-day part is not specified it takes the current
+year-month-day.
+
+@anchor{time duration syntax}
+@section Time duration
+
+The accepted syntax is:
+@example
+[-]HH:MM:SS[.m...]
+[-]S+[.m...]
+@end example
+
+@var{HH} expresses the number of hours, @var{MM} the number a of minutes
+and @var{SS} the number of seconds.
+
+@anchor{video size syntax}
+@section Video size
+Specify the size of the sourced video, it may be a string of the form
+@var{width}x@var{height}, or the name of a size abbreviation.
+
+The following abbreviations are recognized:
+@table @samp
+@item sqcif
+128x96
+@item qcif
+176x144
+@item cif
+352x288
+@item 4cif
+704x576
+@item 16cif
+1408x1152
+@item qqvga
+160x120
+@item qvga
+320x240
+@item vga
+640x480
+@item svga
+800x600
+@item xga
+1024x768
+@item uxga
+1600x1200
+@item qxga
+2048x1536
+@item sxga
+1280x1024
+@item qsxga
+2560x2048
+@item hsxga
+5120x4096
+@item wvga
+852x480
+@item wxga
+1366x768
+@item wsxga
+1600x1024
+@item wuxga
+1920x1200
+@item woxga
+2560x1600
+@item wqsxga
+3200x2048
+@item wquxga
+3840x2400
+@item whsxga
+6400x4096
+@item whuxga
+7680x4800
+@item cga
+320x200
+@item ega
+640x350
+@item hd480
+852x480
+@item hd720
+1280x720
+@item hd1080
+1920x1080
+@end table
+
+@anchor{video rate syntax}
+@section Video rate
+
+Specify the frame rate of a video, expressed as the number of frames
+generated per second. It has to be a string in the format
+@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
+number or a valid video frame rate abbreviation.
+
+The following abbreviations are recognized:
+@table @samp
+@item ntsc
+30000/1001
+@item pal
+25/1
+@item qntsc
+30000/1
+@item qpal
+25/1
+@item sntsc
+30000/1
+@item spal
+25/1
+@item film
+24/1
+@item ntsc-film
+24000/1
+@end table
+
+@anchor{ratio syntax}
+@section Ratio
+
+A ratio can be expressed as an expression, or in the form
+@var{numerator}:@var{denominator}.
+
+Note that a ratio with infinite (1/0) or negative value is
+considered valid, so you should check on the returned value if you
+want to exclude those values.
+
+The undefined value can be expressed using the "0:0" string.
+
+@anchor{color syntax}
+@section Color
+
+It can be the name of a color (case insensitive match) or a
+[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
+representing the alpha component.
+
+The alpha component may be a string composed by "0x" followed by an
+hexadecimal number or a decimal number between 0.0 and 1.0, which
+represents the opacity value (0x00/0.0 means completely transparent,
+0xff/1.0 completely opaque).
+If the alpha component is not specified then 0xff is assumed.
+
+The string "random" will result in a random color.
+
+@c man end SYNTAX
--- a/doc/t2h.init
+++ b/doc/t2h.init
@@ -17,7 +17,6 @@ my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
 </head>
 <body>
 <div id="container">
-<div id="body">
 EOT

 $PRE_BODY_CLOSE = '</div></div>';
@@ -33,7 +32,7 @@ sub FFmpeg_print_page_foot($$)
        T2H_DEFAULT_program_string() : program_string();
    print $fh '<footer class="footer pagination-right">' . "\n";
    print $fh '<span class="label label-info">' . $program_string;
-    print $fh "</span></footer></div></div></body>\n";
+    print $fh "</span></footer></div>\n";
 }

 $float = \&FFmpeg_float;
@@ -69,7 +68,7 @@ $print_page_head = \&FFmpeg_print_page_head;
 sub FFmpeg_print_page_head($$)
 {
    my $fh = shift;
-    my $longtitle = "$Texi2HTML::THISDOC{'fulltitle_no_texi'}";
+    my $longtitle = "$Texi2HTML::THISDOC{'title_no_texi'}";
    $longtitle .= ": $Texi2HTML::NO_TEXI{'This'}" if exists $Texi2HTML::NO_TEXI{'This'};
    my $description = $DOCUMENT_DESCRIPTION;
    $description = $longtitle if (!defined($description));
@@ -93,6 +92,8 @@ $Texi2HTML::THISDOC{program_authors}

 $description
 <meta name="keywords" content="$longtitle">
+<meta name="resource-type" content="document">
+<meta name="distribution" content="global">
 <meta name="Generator" content="$Texi2HTML::THISDOC{program}">
 $encoding
 $CSS_LINES
--- a/doc/utils.texi
+++ b/doc/utils.texi
--- a/ffmpeg.c
+++ b/ffmpeg.c
--- a/ffmpeg.h
+++ b/ffmpeg.h
@@ -37,10 +37,10 @@
 #include "libavcodec/avcodec.h"

 #include "libavfilter/avfilter.h"
+#include "libavfilter/avfiltergraph.h"

 #include "libavutil/avutil.h"
 #include "libavutil/dict.h"
-#include "libavutil/eval.h"
 #include "libavutil/fifo.h"
 #include "libavutil/pixfmt.h"
 #include "libavutil/rational.h"
@@ -51,7 +51,6 @@
 #define VSYNC_PASSTHROUGH 0
 #define VSYNC_CFR         1
 #define VSYNC_VFR         2
-#define VSYNC_VSCFR       0xfe
 #define VSYNC_DROP        0xff

 #define MAX_STREAMS 1024    /* arbitrary sanity check value */
@@ -94,7 +93,6 @@ typedef struct OptionsContext {
    /* input options */
    int64_t input_ts_offset;
    int rate_emu;
-    int accurate_seek;

    SpecifierOpt *ts_scale;
    int        nb_ts_scale;
@@ -115,7 +113,6 @@ typedef struct OptionsContext {
    int chapters_input_file;

    int64_t recording_time;
-    int64_t stop_time;
    uint64_t limit_filesize;
    float mux_preload;
    float mux_max_delay;
@@ -166,22 +163,14 @@ typedef struct OptionsContext {
    int        nb_copy_prior_start;
    SpecifierOpt *filters;
    int        nb_filters;
-    SpecifierOpt *filter_scripts;
-    int        nb_filter_scripts;
    SpecifierOpt *reinit_filters;
    int        nb_reinit_filters;
    SpecifierOpt *fix_sub_duration;
    int        nb_fix_sub_duration;
-    SpecifierOpt *canvas_sizes;
-    int        nb_canvas_sizes;
    SpecifierOpt *pass;
    int        nb_pass;
    SpecifierOpt *passlogfiles;
    int        nb_passlogfiles;
-    SpecifierOpt *guess_layout_max;
-    int        nb_guess_layout_max;
-    SpecifierOpt *apad;
-    int        nb_apad;
 } OptionsContext;

 typedef struct InputFilter {
@@ -206,7 +195,6 @@ typedef struct FilterGraph {
    const char    *graph_desc;

    AVFilterGraph *graph;
-    int reconfiguration;

    InputFilter   **inputs;
    int          nb_inputs;
@@ -221,7 +209,6 @@ typedef struct InputStream {
    int decoding_needed;     /* true if the packets must be decoded in 'raw_fifo' */
    AVCodec *dec;
    AVFrame *decoded_frame;
-    AVFrame *filter_frame; /* a ref of decoded_frame, to be sent to filters */

    int64_t       start;     /* time when read started */
    /* predicted dts of the next packet read for this stream or (when there are
@@ -242,7 +229,6 @@ typedef struct InputStream {
    AVDictionary *opts;
    AVRational framerate;               /* framerate forced with -r */
    int top_field_first;
-    int guess_layout_max;

    int resample_height;
    int resample_width;
@@ -263,10 +249,12 @@ typedef struct InputStream {
    struct sub2video {
        int64_t last_pts;
        int64_t end_pts;
-        AVFrame *frame;
+        AVFilterBufferRef *ref;
        int w, h;
    } sub2video;

+    /* a pool of free buffers for decoded data */
+    FrameBuffer *buffer_pool;
    int dr1;

    /* decoded data from this stream goes into all those filters
@@ -282,16 +270,11 @@ typedef struct InputFile {
    int eof_reached;      /* true if eof reached */
    int eagain;           /* true if last read attempt returned EAGAIN */
    int ist_index;        /* index of first stream in input_streams */
-    int64_t input_ts_offset;
    int64_t ts_offset;
-    int64_t last_ts;
-    int64_t start_time;   /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
-    int64_t recording_time;
    int nb_streams;       /* number of stream that ffmpeg is aware of; may be different
                             from ctx.nb_streams if new streams appear during av_read_frame() */
    int nb_streams_warn;  /* number of streams that the user was warned of */
    int rate_emu;
-    int accurate_seek;

 #if HAVE_PTHREADS
    pthread_t thread;           /* thread reading from this file */
@@ -303,17 +286,6 @@ typedef struct InputFile {
 #endif
 } InputFile;

-enum forced_keyframes_const {
-    FKF_N,
-    FKF_N_FORCED,
-    FKF_PREV_FORCED_N,
-    FKF_PREV_FORCED_T,
-    FKF_T,
-    FKF_NB
-};
-
-extern const char *const forced_keyframes_const_names[];
-
 typedef struct OutputStream {
    int file_index;          /* file index */
    int index;               /* stream index in the output file */
@@ -328,8 +300,6 @@ typedef struct OutputStream {
    /* pts of the first frame encoded for this stream, used for limiting
     * recording time */
    int64_t first_pts;
-    /* dts of the last packet sent to the muxer */
-    int64_t last_mux_dts;
    AVBitStreamFilterContext *bitstream_filters;
    AVCodec *enc;
    int64_t max_frames;
@@ -340,15 +310,13 @@ typedef struct OutputStream {
    int force_fps;
    int top_field_first;

-    AVRational frame_aspect_ratio;
+    float frame_aspect_ratio;

    /* forced key frames */
    int64_t *forced_kf_pts;
    int forced_kf_count;
    int forced_kf_index;
    char *forced_keyframes;
-    AVExpr *forced_keyframes_pexpr;
-    double forced_keyframes_expr_const_values[FKF_NB];

    /* audio only */
    int audio_channels_map[SWR_CH_MAX];  /* list of the channels id to pick from the source stream */
@@ -361,10 +329,10 @@ typedef struct OutputStream {
    char *avfilter;

    int64_t sws_flags;
+    int64_t swr_filter_type;
+    int64_t swr_dither_method;
+    double swr_dither_scale;
    AVDictionary *opts;
-    AVDictionary *swr_opts;
-    AVDictionary *resample_opts;
-    char *apad;
    int finished;        /* no more packets should be written for this stream */
    int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
    int stream_copy;
@@ -422,7 +390,6 @@ extern int qp_hist;
 extern int stdin_interaction;
 extern int frame_bits_per_raw_sample;
 extern AVIOContext *progress_avio;
-extern float max_error_rate;

 extern const AVIOInterruptCB int_cb;

--- a/ffmpeg_filter.c
+++ b/ffmpeg_filter.c
@@ -21,10 +21,9 @@
 #include "ffmpeg.h"

 #include "libavfilter/avfilter.h"
+#include "libavfilter/avfiltergraph.h"
 #include "libavfilter/buffersink.h"

-#include "libavresample/avresample.h"
-
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
@@ -92,11 +91,6 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)

 static char *choose_pix_fmts(OutputStream *ost)
 {
-    AVDictionaryEntry *strict_dict = av_dict_get(ost->opts, "strict", NULL, 0);
-    if (strict_dict)
-        // used by choose_pixel_fmt() and below
-        av_opt_set(ost->st->codec, "strict", strict_dict->value, 0);
-
     if (ost->keep_pix_fmt) {
        if (ost->filter)
            avfilter_graph_set_auto_convert(ost->filter->graph->graph,
@@ -114,7 +108,7 @@ static char *choose_pix_fmts(OutputStream *ost)
        int len;

        if (avio_open_dyn_buf(&s) < 0)
-            exit_program(1);
+            exit(1);

        p = ost->enc->pix_fmts;
        if (ost->st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
@@ -128,7 +122,7 @@ static char *choose_pix_fmts(OutputStream *ost)

        for (; *p != AV_PIX_FMT_NONE; p++) {
            const char *name = av_get_pix_fmt_name(*p);
-            avio_printf(s, "%s|", name);
+            avio_printf(s, "%s:", name);
        }
        len = avio_close_dyn_buf(s, &ret);
        ret[len - 1] = 0;
@@ -139,24 +133,24 @@ static char *choose_pix_fmts(OutputStream *ost)

 /* Define a function for building a string containing a list of
 * allowed formats. */
-#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name)           \
+#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name, separator)\
 static char *choose_ ## var ## s(OutputStream *ost)                            \
 {                                                                              \
    if (ost->st->codec->var != none) {                                         \
        get_name(ost->st->codec->var);                                         \
        return av_strdup(name);                                                \
-    } else if (ost->enc && ost->enc->supported_list) {                         \
+    } else if (ost->enc->supported_list) {                                     \
        const type *p;                                                         \
        AVIOContext *s = NULL;                                                 \
        uint8_t *ret;                                                          \
        int len;                                                               \
                                                                               \
        if (avio_open_dyn_buf(&s) < 0)                                         \
-            exit_program(1);                                                           \
+            exit(1);                                                           \
                                                                               \
        for (p = ost->enc->supported_list; *p != none; p++) {                  \
            get_name(*p);                                                      \
-            avio_printf(s, "%s|", name);                                       \
+            avio_printf(s, "%s" separator, name);                              \
        }                                                                      \
        len = avio_close_dyn_buf(s, &ret);                                     \
        ret[len - 1] = 0;                                                      \
@@ -166,28 +160,28 @@ static char *choose_ ## var ## s(OutputStream *ost)                            \
 }

 // DEF_CHOOSE_FORMAT(enum AVPixelFormat, pix_fmt, pix_fmts, AV_PIX_FMT_NONE,
-//                   GET_PIX_FMT_NAME)
+//                   GET_PIX_FMT_NAME, ":")

 DEF_CHOOSE_FORMAT(enum AVSampleFormat, sample_fmt, sample_fmts,
-                  AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME)
+                  AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME, ",")

 DEF_CHOOSE_FORMAT(int, sample_rate, supported_samplerates, 0,
-                  GET_SAMPLE_RATE_NAME)
+                  GET_SAMPLE_RATE_NAME, ",")

 DEF_CHOOSE_FORMAT(uint64_t, channel_layout, channel_layouts, 0,
-                  GET_CH_LAYOUT_NAME)
+                  GET_CH_LAYOUT_NAME, ",")

 FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost)
 {
    FilterGraph *fg = av_mallocz(sizeof(*fg));

    if (!fg)
-        exit_program(1);
+        exit(1);
    fg->index = nb_filtergraphs;

    GROW_ARRAY(fg->outputs, fg->nb_outputs);
    if (!(fg->outputs[0] = av_mallocz(sizeof(*fg->outputs[0]))))
-        exit_program(1);
+        exit(1);
    fg->outputs[0]->ost   = ost;
    fg->outputs[0]->graph = fg;

@@ -195,7 +189,7 @@ FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost)

    GROW_ARRAY(fg->inputs, fg->nb_inputs);
    if (!(fg->inputs[0] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit_program(1);
+        exit(1);
    fg->inputs[0]->ist   = ist;
    fg->inputs[0]->graph = fg;

@@ -218,7 +212,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
    if (type != AVMEDIA_TYPE_VIDEO && type != AVMEDIA_TYPE_AUDIO) {
        av_log(NULL, AV_LOG_FATAL, "Only video and audio filters supported "
               "currently.\n");
-        exit_program(1);
+        exit(1);
    }

    if (in->name) {
@@ -230,7 +224,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
        if (file_idx < 0 || file_idx >= nb_input_files) {
            av_log(NULL, AV_LOG_FATAL, "Invalid file index %d in filtergraph description %s.\n",
                   file_idx, fg->graph_desc);
-            exit_program(1);
+            exit(1);
        }
        s = input_files[file_idx]->ctx;

@@ -248,7 +242,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
        if (!st) {
            av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s "
                   "matches no streams.\n", p, fg->graph_desc);
-            exit_program(1);
+            exit(1);
        }
        ist = input_streams[input_files[file_idx]->ist_index + st->index];
    } else {
@@ -262,7 +256,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
            av_log(NULL, AV_LOG_FATAL, "Cannot find a matching stream for "
                   "unlabeled input pad %d on filter %s\n", in->pad_idx,
                   in->filter_ctx->name);
-            exit_program(1);
+            exit(1);
        }
    }
    av_assert0(ist);
@@ -273,7 +267,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)

    GROW_ARRAY(fg->inputs, fg->nb_inputs);
    if (!(fg->inputs[fg->nb_inputs - 1] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit_program(1);
+        exit(1);
    fg->inputs[fg->nb_inputs - 1]->ist   = ist;
    fg->inputs[fg->nb_inputs - 1]->graph = fg;

@@ -281,72 +275,22 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
    ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
 }

-static int insert_trim(int64_t start_time, int64_t duration,
-                       AVFilterContext **last_filter, int *pad_idx,
-                       const char *filter_name)
-{
-    AVFilterGraph *graph = (*last_filter)->graph;
-    AVFilterContext *ctx;
-    const AVFilter *trim;
-    enum AVMediaType type = avfilter_pad_get_type((*last_filter)->output_pads, *pad_idx);
-    const char *name = (type == AVMEDIA_TYPE_VIDEO) ? "trim" : "atrim";
-    int ret = 0;
-
-    if (duration == INT64_MAX && start_time == AV_NOPTS_VALUE)
-        return 0;
-
-    trim = avfilter_get_by_name(name);
-    if (!trim) {
-        av_log(NULL, AV_LOG_ERROR, "%s filter not present, cannot limit "
-               "recording time.\n", name);
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    ctx = avfilter_graph_alloc_filter(graph, trim, filter_name);
-    if (!ctx)
-        return AVERROR(ENOMEM);
-
-    if (duration != INT64_MAX) {
-        ret = av_opt_set_int(ctx, "durationi", duration,
-                                AV_OPT_SEARCH_CHILDREN);
-    }
-    if (ret >= 0 && start_time != AV_NOPTS_VALUE) {
-        ret = av_opt_set_int(ctx, "starti", start_time,
-                                AV_OPT_SEARCH_CHILDREN);
-    }
-    if (ret < 0) {
-        av_log(ctx, AV_LOG_ERROR, "Error configuring the %s filter", name);
-        return ret;
-    }
-
-    ret = avfilter_init_str(ctx, NULL);
-    if (ret < 0)
-        return ret;
-
-    ret = avfilter_link(*last_filter, *pad_idx, ctx, 0);
-    if (ret < 0)
-        return ret;
-
-    *last_filter = ctx;
-    *pad_idx     = 0;
-    return 0;
-}
-
 static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
 {
    char *pix_fmts;
    OutputStream *ost = ofilter->ost;
-    OutputFile    *of = output_files[ost->file_index];
    AVCodecContext *codec = ost->st->codec;
    AVFilterContext *last_filter = out->filter_ctx;
    int pad_idx = out->pad_idx;
    int ret;
    char name[255];
+    AVBufferSinkParams *buffersink_params = av_buffersink_params_alloc();

    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("buffersink"),
+                                       avfilter_get_by_name("ffbuffersink"),
                                       name, NULL, NULL, fg->graph);
+    av_freep(&buffersink_params);

    if (ret < 0)
        return ret;
@@ -355,7 +299,7 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
        char args[255];
        AVFilterContext *filter;

-        snprintf(args, sizeof(args), "%d:%d:0x%X",
+        snprintf(args, sizeof(args), "%d:%d:flags=0x%X",
                 codec->width,
                 codec->height,
                 (unsigned)ost->sws_flags);
@@ -375,18 +319,17 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
        AVFilterContext *filter;
        snprintf(name, sizeof(name), "pixel format for output stream %d:%d",
                 ost->file_index, ost->index);
-        ret = avfilter_graph_create_filter(&filter,
+        if ((ret = avfilter_graph_create_filter(&filter,
                                                avfilter_get_by_name("format"),
                                                "format", pix_fmts, NULL,
-                                                fg->graph);
-        av_freep(&pix_fmts);
-        if (ret < 0)
+                                                fg->graph)) < 0)
            return ret;
        if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
            return ret;

        last_filter = filter;
        pad_idx     = 0;
+        av_freep(&pix_fmts);
    }

    if (ost->frame_rate.num && 0) {
@@ -409,14 +352,6 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
        pad_idx = 0;
    }

-    snprintf(name, sizeof(name), "trim for output stream %d:%d",
-             ost->file_index, ost->index);
-    ret = insert_trim(of->start_time, of->recording_time,
-                      &last_filter, &pad_idx, name);
-    if (ret < 0)
-        return ret;
-
-
    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
        return ret;

@@ -426,7 +361,6 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
 static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
 {
    OutputStream *ost = ofilter->ost;
-    OutputFile    *of = output_files[ost->file_index];
    AVCodecContext *codec  = ost->st->codec;
    AVFilterContext *last_filter = out->filter_ctx;
    int pad_idx = out->pad_idx;
@@ -434,14 +368,13 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
    char name[255];
    int ret;

+
    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("abuffersink"),
+                                       avfilter_get_by_name("ffabuffersink"),
                                       name, NULL, NULL, fg->graph);
    if (ret < 0)
        return ret;
-    if ((ret = av_opt_set_int(ofilter->filter, "all_channel_counts", 1, AV_OPT_SEARCH_CHILDREN)) < 0)
-        return ret;

 #define AUTO_INSERT_FILTER(opt_name, filter_name, arg) do {                 \
    AVFilterContext *filt_ctx;                                              \
@@ -524,27 +457,6 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
        AUTO_INSERT_FILTER("-vol", "volume", args);
    }

-    if (ost->apad && of->shortest) {
-        char args[256];
-        int i;
-
-        for (i=0; i<of->ctx->nb_streams; i++)
-            if (of->ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
-                break;
-
-        if (i<of->ctx->nb_streams) {
-            snprintf(args, sizeof(args), "%s", ost->apad);
-            AUTO_INSERT_FILTER("-apad", "apad", args);
-        }
-    }
-
-    snprintf(name, sizeof(name), "trim for output stream %d:%d",
-             ost->file_index, ost->index);
-    ret = insert_trim(of->start_time, of->recording_time,
-                      &last_filter, &pad_idx, name);
-    if (ret < 0)
-        return ret;
-
    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
        return ret;

@@ -555,11 +467,11 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
 {                                                                  \
    AVFilterContext *ctx = inout->filter_ctx;                      \
    AVFilterPad *pads = in ? ctx->input_pads  : ctx->output_pads;  \
-    int       nb_pads = in ? ctx->nb_inputs   : ctx->nb_outputs;   \
+    int       nb_pads = in ? ctx->input_count : ctx->output_count; \
    AVIOContext *pb;                                               \
                                                                   \
    if (avio_open_dyn_buf(&pb) < 0)                                \
-        exit_program(1);                                           \
+        exit(1);                                                   \
                                                                   \
    avio_printf(pb, "%s", ctx->filter->name);                      \
    if (nb_pads > 1)                                               \
@@ -583,7 +495,9 @@ int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOu
 static int sub2video_prepare(InputStream *ist)
 {
    AVFormatContext *avf = input_files[ist->file_index]->ctx;
-    int i, w, h;
+    int i, ret, w, h;
+    uint8_t *image[4];
+    int linesize[4];

    /* Compute the size of the canvas for the subtitles stream.
       If the subtitles codec has set a size, use it. Otherwise use the
@@ -608,37 +522,47 @@ static int sub2video_prepare(InputStream *ist)

    /* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the
       palettes for all rectangles are identical or compatible */
-    ist->resample_pix_fmt = ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32;
+    ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32;

-    ist->sub2video.frame = av_frame_alloc();
-    if (!ist->sub2video.frame)
+    ret = av_image_alloc(image, linesize, w, h, AV_PIX_FMT_RGB32, 32);
+    if (ret < 0)
+        return ret;
+    memset(image[0], 0, h * linesize[0]);
+    ist->sub2video.ref = avfilter_get_video_buffer_ref_from_arrays(
+            image, linesize, AV_PERM_READ | AV_PERM_PRESERVE,
+            w, h, AV_PIX_FMT_RGB32);
+    if (!ist->sub2video.ref) {
+        av_free(image[0]);
        return AVERROR(ENOMEM);
+    }
    return 0;
 }

 static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                        AVFilterInOut *in)
 {
-    AVFilterContext *last_filter;
-    const AVFilter *buffer_filt = avfilter_get_by_name("buffer");
+    AVFilterContext *first_filter = in->filter_ctx;
+    AVFilter *filter = avfilter_get_by_name("buffer");
    InputStream *ist = ifilter->ist;
-    InputFile     *f = input_files[ist->file_index];
    AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) :
                                         ist->st->time_base;
-    AVRational fr = ist->framerate;
+    AVRational fr = ist->framerate.num ? ist->framerate :
+                                         ist->st->r_frame_rate;
    AVRational sar;
    AVBPrint args;
    char name[255];
-    int ret, pad_idx = 0;
+    int pad_idx = in->pad_idx;
+    int ret;

-    if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot connect video filter to audio input\n");
-        return AVERROR(EINVAL);
+    if (!ist->framerate.num && ist->st->codec->ticks_per_frame>1) {
+        AVRational codec_fr = av_inv_q(ist->st->codec->time_base);
+        AVRational   avg_fr = ist->st->avg_frame_rate;
+        codec_fr.den *= ist->st->codec->ticks_per_frame;
+        if (   codec_fr.num>0 && codec_fr.den>0 && av_q2d(codec_fr) < av_q2d(fr)*0.7
+            && fabs(1.0 - av_q2d(av_div_q(avg_fr, fr)))>0.1)
+            fr = codec_fr;
    }

-    if (!fr.num)
-        fr = av_guess_frame_rate(input_files[ist->file_index]->ctx, ist->st, NULL);
-
    if (ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
        ret = sub2video_prepare(ist);
        if (ret < 0)
@@ -662,10 +586,9 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
             ist->file_index, ist->st->index);

-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, buffer_filt, name,
+    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter, name,
                                            args.str, NULL, fg->graph)) < 0)
        return ret;
-    last_filter = ifilter->filter;

    if (ist->framerate.num) {
        AVFilterContext *setpts;
@@ -678,37 +601,14 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                                fg->graph)) < 0)
            return ret;

-        if ((ret = avfilter_link(last_filter, 0, setpts, 0)) < 0)
+        if ((ret = avfilter_link(setpts, 0, first_filter, pad_idx)) < 0)
            return ret;

-        last_filter = setpts;
+        first_filter = setpts;
+        pad_idx = 0;
    }

-    if (do_deinterlace) {
-        AVFilterContext *yadif;
-
-        snprintf(name, sizeof(name), "deinterlace input from stream %d:%d",
-                 ist->file_index, ist->st->index);
-        if ((ret = avfilter_graph_create_filter(&yadif,
-                                                avfilter_get_by_name("yadif"),
-                                                name, "", NULL,
-                                                fg->graph)) < 0)
-            return ret;
-
-        if ((ret = avfilter_link(last_filter, 0, yadif, 0)) < 0)
-            return ret;
-
-        last_filter = yadif;
-    }
-
-    snprintf(name, sizeof(name), "trim for input stream %d:%d",
-             ist->file_index, ist->st->index);
-    ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
-                      AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
-    if (ret < 0)
-        return ret;
-
-    if ((ret = avfilter_link(last_filter, 0, in->filter_ctx, in->pad_idx)) < 0)
+    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
        return ret;
    return 0;
 }
@@ -716,37 +616,26 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
 static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
                                        AVFilterInOut *in)
 {
-    AVFilterContext *last_filter;
-    const AVFilter *abuffer_filt = avfilter_get_by_name("abuffer");
+    AVFilterContext *first_filter = in->filter_ctx;
+    AVFilter *filter = avfilter_get_by_name("abuffer");
    InputStream *ist = ifilter->ist;
-    InputFile     *f = input_files[ist->file_index];
-    AVBPrint args;
-    char name[255];
-    int ret, pad_idx = 0;
+    int pad_idx = in->pad_idx;
+    char args[255], name[255];
+    int ret;

-    if (ist->st->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot connect audio filter to non audio input\n");
-        return AVERROR(EINVAL);
-    }
-
-    av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
-    av_bprintf(&args, "time_base=%d/%d:sample_rate=%d:sample_fmt=%s",
+    snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s"
+             ":channel_layout=0x%"PRIx64,
             1, ist->st->codec->sample_rate,
             ist->st->codec->sample_rate,
-             av_get_sample_fmt_name(ist->st->codec->sample_fmt));
-    if (ist->st->codec->channel_layout)
-        av_bprintf(&args, ":channel_layout=0x%"PRIx64,
-                   ist->st->codec->channel_layout);
-    else
-        av_bprintf(&args, ":channels=%d", ist->st->codec->channels);
+             av_get_sample_fmt_name(ist->st->codec->sample_fmt),
+             ist->st->codec->channel_layout);
    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
             ist->file_index, ist->st->index);

-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, abuffer_filt,
-                                            name, args.str, NULL,
+    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter,
+                                            name, args, NULL,
                                            fg->graph)) < 0)
        return ret;
-    last_filter = ifilter->filter;

 #define AUTO_INSERT_FILTER_INPUT(opt_name, filter_name, arg) do {                 \
    AVFilterContext *filt_ctx;                                              \
@@ -762,11 +651,11 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
    if (ret < 0)                                                            \
        return ret;                                                         \
                                                                            \
-    ret = avfilter_link(last_filter, 0, filt_ctx, 0);                       \
+    ret = avfilter_link(filt_ctx, 0, first_filter, pad_idx);                \
    if (ret < 0)                                                            \
        return ret;                                                         \
                                                                            \
-    last_filter = filt_ctx;                                                 \
+    first_filter = filt_ctx;                                                  \
 } while (0)

    if (audio_sync_method > 0) {
@@ -775,8 +664,6 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
        av_strlcatf(args, sizeof(args), "async=%d", audio_sync_method);
        if (audio_drift_threshold != 0.1)
            av_strlcatf(args, sizeof(args), ":min_hard_comp=%f", audio_drift_threshold);
-        if (!fg->reconfiguration)
-            av_strlcatf(args, sizeof(args), ":first_pts=0");
        AUTO_INSERT_FILTER_INPUT("-async", "aresample", args);
    }

@@ -802,15 +689,7 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
        snprintf(args, sizeof(args), "%f", audio_volume / 256.);
        AUTO_INSERT_FILTER_INPUT("-vol", "volume", args);
    }
-
-    snprintf(name, sizeof(name), "trim for input stream %d:%d",
-             ist->file_index, ist->st->index);
-    ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
-                      AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
-    if (ret < 0)
-        return ret;
-
-    if ((ret = avfilter_link(last_filter, 0, in->filter_ctx, in->pad_idx)) < 0)
+    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
        return ret;

    return 0;
@@ -842,33 +721,20 @@ int configure_filtergraph(FilterGraph *fg)

    if (simple) {
        OutputStream *ost = fg->outputs[0]->ost;
-        char args[512];
-        AVDictionaryEntry *e = NULL;
-
+        char args[255];
        snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
        fg->graph->scale_sws_opts = av_strdup(args);

        args[0] = 0;
-        while ((e = av_dict_get(ost->swr_opts, "", e,
-                                AV_DICT_IGNORE_SUFFIX))) {
-            av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value);
-        }
+        if (ost->swr_filter_type != SWR_FILTER_TYPE_KAISER)
+            av_strlcatf(args, sizeof(args), "filter_type=%d:", (int)ost->swr_filter_type);
+        if (ost->swr_dither_method)
+            av_strlcatf(args, sizeof(args), "dither_method=%d:", (int)ost->swr_dither_method);
+        if (ost->swr_dither_scale != 1.0)
+            av_strlcatf(args, sizeof(args), "dither_scale=%f:", ost->swr_dither_scale);
        if (strlen(args))
            args[strlen(args)-1] = 0;
        av_opt_set(fg->graph, "aresample_swr_opts", args, 0);
-
-        args[0] = '\0';
-        while ((e = av_dict_get(fg->outputs[0]->ost->resample_opts, "", e,
-                                AV_DICT_IGNORE_SUFFIX))) {
-            av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value);
-        }
-        if (strlen(args))
-            args[strlen(args) - 1] = '\0';
-        fg->graph->resample_lavr_opts = av_strdup(args);
-
-        e = av_dict_get(ost->opts, "threads", NULL, 0);
-        if (e)
-            av_opt_set(fg->graph, "threads", e->value, 0);
    }

    if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
@@ -902,7 +768,7 @@ int configure_filtergraph(FilterGraph *fg)
        for (cur = outputs; cur;) {
            GROW_ARRAY(fg->outputs, fg->nb_outputs);
            if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
-                exit_program(1);
+                exit(1);
            fg->outputs[fg->nb_outputs - 1]->graph   = fg;
            fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
            cur = cur->next;
@@ -910,7 +776,6 @@ int configure_filtergraph(FilterGraph *fg)
        }
    }

-    fg->reconfiguration = 1;
    return 0;
 }

--- a/ffmpeg_opt.c
+++ b/ffmpeg_opt.c
--- a/ffplay.c
+++ b/ffplay.c
--- a/ffprobe.c
+++ b/ffprobe.c
@@ -37,9 +37,7 @@
 #include "libavutil/pixdesc.h"
 #include "libavutil/dict.h"
 #include "libavutil/libm.h"
-#include "libavutil/parseutils.h"
 #include "libavutil/timecode.h"
-#include "libavutil/timestamp.h"
 #include "libavdevice/avdevice.h"
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
@@ -54,12 +52,10 @@ static int do_count_frames = 0;
 static int do_count_packets = 0;
 static int do_read_frames  = 0;
 static int do_read_packets = 0;
-static int do_show_chapters = 0;
 static int do_show_error   = 0;
 static int do_show_format  = 0;
 static int do_show_frames  = 0;
 static int do_show_packets = 0;
-static int do_show_programs = 0;
 static int do_show_streams = 0;
 static int do_show_stream_disposition = 0;
 static int do_show_data    = 0;
@@ -75,17 +71,6 @@ static int show_private_data            = 1;
 static char *print_format;
 static char *stream_specifier;

-typedef struct {
-    int id;             ///< identifier
-    int64_t start, end; ///< start, end in second/AV_TIME_BASE units
-    int has_start, has_end;
-    int start_is_offset, end_is_offset;
-    int duration_frames;
-} ReadInterval;
-
-static ReadInterval *read_intervals;
-static int read_intervals_nb = 0;
-
 /* section structure definition */

 #define SECTION_MAX_NB_CHILDREN 10
@@ -108,9 +93,6 @@ struct section {

 typedef enum {
    SECTION_ID_NONE = -1,
-    SECTION_ID_CHAPTER,
-    SECTION_ID_CHAPTER_TAGS,
-    SECTION_ID_CHAPTERS,
    SECTION_ID_ERROR,
    SECTION_ID_FORMAT,
    SECTION_ID_FORMAT_TAGS,
@@ -122,14 +104,7 @@ typedef enum {
    SECTION_ID_PACKET,
    SECTION_ID_PACKETS,
    SECTION_ID_PACKETS_AND_FRAMES,
-    SECTION_ID_PROGRAM_STREAM_DISPOSITION,
-    SECTION_ID_PROGRAM_STREAM_TAGS,
-    SECTION_ID_PROGRAM,
-    SECTION_ID_PROGRAM_STREAMS,
-    SECTION_ID_PROGRAM_STREAM,
-    SECTION_ID_PROGRAM_TAGS,
    SECTION_ID_PROGRAM_VERSION,
-    SECTION_ID_PROGRAMS,
    SECTION_ID_ROOT,
    SECTION_ID_STREAM,
    SECTION_ID_STREAM_DISPOSITION,
@@ -138,9 +113,6 @@ typedef enum {
 } SectionID;

 static struct section sections[] = {
-    [SECTION_ID_CHAPTERS] =           { SECTION_ID_CHAPTERS, "chapters", SECTION_FLAG_IS_ARRAY, { SECTION_ID_CHAPTER, -1 } },
-    [SECTION_ID_CHAPTER] =            { SECTION_ID_CHAPTER, "chapter", 0, { SECTION_ID_CHAPTER_TAGS, -1 } },
-    [SECTION_ID_CHAPTER_TAGS] =       { SECTION_ID_CHAPTER_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "chapter_tags" },
    [SECTION_ID_ERROR] =              { SECTION_ID_ERROR, "error", 0, { -1 } },
    [SECTION_ID_FORMAT] =             { SECTION_ID_FORMAT, "format", 0, { SECTION_ID_FORMAT_TAGS, -1 } },
    [SECTION_ID_FORMAT_TAGS] =        { SECTION_ID_FORMAT_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "format_tags" },
@@ -152,17 +124,10 @@ static struct section sections[] = {
    [SECTION_ID_PACKETS] =            { SECTION_ID_PACKETS, "packets", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
    [SECTION_ID_PACKETS_AND_FRAMES] = { SECTION_ID_PACKETS_AND_FRAMES, "packets_and_frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
    [SECTION_ID_PACKET] =             { SECTION_ID_PACKET, "packet", 0, { -1 } },
-    [SECTION_ID_PROGRAM_STREAM_DISPOSITION] = { SECTION_ID_PROGRAM_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "program_stream_disposition" },
-    [SECTION_ID_PROGRAM_STREAM_TAGS] =        { SECTION_ID_PROGRAM_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "program_stream_tags" },
-    [SECTION_ID_PROGRAM] =                    { SECTION_ID_PROGRAM, "program", 0, { SECTION_ID_PROGRAM_TAGS, SECTION_ID_PROGRAM_STREAMS, -1 } },
-    [SECTION_ID_PROGRAM_STREAMS] =            { SECTION_ID_PROGRAM_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PROGRAM_STREAM, -1 }, .unique_name = "program_streams" },
-    [SECTION_ID_PROGRAM_STREAM] =             { SECTION_ID_PROGRAM_STREAM, "stream", 0, { SECTION_ID_PROGRAM_STREAM_DISPOSITION, SECTION_ID_PROGRAM_STREAM_TAGS, -1 }, .unique_name = "program_stream" },
-    [SECTION_ID_PROGRAM_TAGS] =               { SECTION_ID_PROGRAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "program_tags" },
    [SECTION_ID_PROGRAM_VERSION] =    { SECTION_ID_PROGRAM_VERSION, "program_version", 0, { -1 } },
-    [SECTION_ID_PROGRAMS] =                   { SECTION_ID_PROGRAMS, "programs", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PROGRAM, -1 } },
    [SECTION_ID_ROOT] =               { SECTION_ID_ROOT, "root", SECTION_FLAG_IS_WRAPPER,
-                                        { SECTION_ID_CHAPTERS, SECTION_ID_FORMAT, SECTION_ID_FRAMES, SECTION_ID_PROGRAMS, SECTION_ID_STREAMS,
-                                          SECTION_ID_PACKETS, SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS, -1} },
+                                        { SECTION_ID_FORMAT, SECTION_ID_FRAMES, SECTION_ID_STREAMS, SECTION_ID_PACKETS,
+                                          SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS, -1} },
    [SECTION_ID_STREAMS] =            { SECTION_ID_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM, -1 } },
    [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
    [SECTION_ID_STREAM_DISPOSITION] = { SECTION_ID_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_disposition" },
@@ -187,7 +152,7 @@ static uint64_t *nb_streams_packets;
 static uint64_t *nb_streams_frames;
 static int *selected_streams;

-static void ffprobe_cleanup(int ret)
+static void exit_program(void)
 {
    int i;
    for (i = 0; i < FF_ARRAY_ELEMS(sections); i++)
@@ -335,7 +300,7 @@ static int writer_open(WriterContext **wctx, const Writer *writer, const char *a
 {
    int i, ret = 0;

-    if (!(*wctx = av_mallocz(sizeof(WriterContext)))) {
+    if (!(*wctx = av_malloc(sizeof(WriterContext)))) {
        ret = AVERROR(ENOMEM);
        goto fail;
    }
@@ -703,8 +668,6 @@ typedef struct CompactContext {
    char *escape_mode_str;
    const char * (*escape_str)(AVBPrint *dst, const char *src, const char sep, void *log_ctx);
    int nested_section[SECTION_MAX_NB_LEVELS];
-    int has_nested_elems[SECTION_MAX_NB_LEVELS];
-    int terminate_line[SECTION_MAX_NB_LEVELS];
 } CompactContext;

 #undef OFFSET
@@ -752,28 +715,18 @@ static void compact_print_section_header(WriterContext *wctx)
    const struct section *section = wctx->section[wctx->level];
    const struct section *parent_section = wctx->level ?
        wctx->section[wctx->level-1] : NULL;
-    compact->terminate_line[wctx->level] = 1;
-    compact->has_nested_elems[wctx->level] = 0;

    av_bprint_clear(&wctx->section_pbuf[wctx->level]);
-    if (!(section->flags & SECTION_FLAG_IS_ARRAY) && parent_section &&
+    if (parent_section &&
        !(parent_section->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY))) {
        compact->nested_section[wctx->level] = 1;
-        compact->has_nested_elems[wctx->level-1] = 1;
        av_bprintf(&wctx->section_pbuf[wctx->level], "%s%s:",
                   wctx->section_pbuf[wctx->level-1].str,
                   (char *)av_x_if_null(section->element_name, section->name));
        wctx->nb_item[wctx->level] = wctx->nb_item[wctx->level-1];
-    } else {
-        if (parent_section && compact->has_nested_elems[wctx->level-1] &&
-            (section->flags & SECTION_FLAG_IS_ARRAY)) {
-            compact->terminate_line[wctx->level-1] = 0;
-            printf("\n");
-        }
-        if (compact->print_section &&
-            !(section->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY)))
-            printf("%s%c", section->name, compact->item_sep);
-    }
+    } else if (compact->print_section &&
+        !(section->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY)))
+       printf("%s%c", section->name, compact->item_sep);
 }

 static void compact_print_section_footer(WriterContext *wctx)
@@ -781,7 +734,6 @@ static void compact_print_section_footer(WriterContext *wctx)
    CompactContext *compact = wctx->priv;

    if (!compact->nested_section[wctx->level] &&
-        compact->terminate_line[wctx->level] &&
        !(wctx->section[wctx->level]->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY)))
        printf("\n");
 }
@@ -1520,11 +1472,11 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
    print_time("pkt_pts_time",          frame->pkt_pts, &stream->time_base);
    print_ts  ("pkt_dts",               frame->pkt_dts);
    print_time("pkt_dts_time",          frame->pkt_dts, &stream->time_base);
-    print_duration_ts  ("pkt_duration",      av_frame_get_pkt_duration(frame));
-    print_duration_time("pkt_duration_time", av_frame_get_pkt_duration(frame), &stream->time_base);
-    if (av_frame_get_pkt_pos (frame) != -1) print_fmt    ("pkt_pos", "%"PRId64, av_frame_get_pkt_pos(frame));
+    print_duration_ts  ("pkt_duration",      frame->pkt_duration);
+    print_duration_time("pkt_duration_time", frame->pkt_duration, &stream->time_base);
+    if (frame->pkt_pos != -1) print_fmt    ("pkt_pos", "%"PRId64, frame->pkt_pos);
    else                      print_str_opt("pkt_pos", "N/A");
-    if (av_frame_get_pkt_size(frame) != -1) print_fmt    ("pkt_size", "%d", av_frame_get_pkt_size(frame));
+    if (frame->pkt_size != -1) print_fmt    ("pkt_size", "%d", av_frame_get_pkt_size(frame));
    else                       print_str_opt("pkt_size", "N/A");

    switch (stream->codec->codec_type) {
@@ -1548,6 +1500,7 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
        print_int("interlaced_frame",       frame->interlaced_frame);
        print_int("top_field_first",        frame->top_field_first);
        print_int("repeat_pict",            frame->repeat_pict);
+        print_int("reference",              frame->reference);
        break;

    case AVMEDIA_TYPE_AUDIO:
@@ -1606,93 +1559,16 @@ static av_always_inline int process_frame(WriterContext *w,
    return got_frame;
 }

-static void log_read_interval(const ReadInterval *interval, void *log_ctx, int log_level)
-{
-    av_log(log_ctx, log_level, "id:%d", interval->id);
-
-    if (interval->has_start) {
-        av_log(log_ctx, log_level, " start:%s%s", interval->start_is_offset ? "+" : "",
-               av_ts2timestr(interval->start, &AV_TIME_BASE_Q));
-    } else {
-        av_log(log_ctx, log_level, " start:N/A");
-    }
-
-    if (interval->has_end) {
-        av_log(log_ctx, log_level, " end:%s", interval->end_is_offset ? "+" : "");
-        if (interval->duration_frames)
-            av_log(log_ctx, log_level, "#%"PRId64, interval->end);
-        else
-            av_log(log_ctx, log_level, "%s", av_ts2timestr(interval->end, &AV_TIME_BASE_Q));
-    } else {
-        av_log(log_ctx, log_level, " end:N/A");
-    }
-
-    av_log(log_ctx, log_level, "\n");
-}
-
-static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
-                                 const ReadInterval *interval, int64_t *cur_ts)
+static void read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
 {
    AVPacket pkt, pkt1;
    AVFrame frame;
-    int ret = 0, i = 0, frame_count = 0;
-    int64_t start = -INT64_MAX, end = interval->end;
-    int has_start = 0, has_end = interval->has_end && !interval->end_is_offset;
+    int i = 0;

    av_init_packet(&pkt);

-    av_log(NULL, AV_LOG_VERBOSE, "Processing read interval ");
-    log_read_interval(interval, NULL, AV_LOG_VERBOSE);
-
-    if (interval->has_start) {
-        int64_t target;
-        if (interval->start_is_offset) {
-            if (*cur_ts == AV_NOPTS_VALUE) {
-                av_log(NULL, AV_LOG_ERROR,
-                       "Could not seek to relative position since current "
-                       "timestamp is not defined\n");
-                ret = AVERROR(EINVAL);
-                goto end;
-            }
-            target = *cur_ts + interval->start;
-        } else {
-            target = interval->start;
-        }
-
-        av_log(NULL, AV_LOG_VERBOSE, "Seeking to read interval start point %s\n",
-               av_ts2timestr(target, &AV_TIME_BASE_Q));
-        if ((ret = avformat_seek_file(fmt_ctx, -1, -INT64_MAX, target, INT64_MAX, 0)) < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Could not seek to position %"PRId64": %s\n",
-                   interval->start, av_err2str(ret));
-            goto end;
-        }
-    }
-
    while (!av_read_frame(fmt_ctx, &pkt)) {
        if (selected_streams[pkt.stream_index]) {
-            AVRational tb = fmt_ctx->streams[pkt.stream_index]->time_base;
-
-            if (pkt.pts != AV_NOPTS_VALUE)
-                *cur_ts = av_rescale_q(pkt.pts, tb, AV_TIME_BASE_Q);
-
-            if (!has_start && *cur_ts != AV_NOPTS_VALUE) {
-                start = *cur_ts;
-                has_start = 1;
-            }
-
-            if (has_start && !has_end && interval->end_is_offset) {
-                end = start + interval->end;
-                has_end = 1;
-            }
-
-            if (interval->end_is_offset && interval->duration_frames) {
-                if (frame_count >= interval->end)
-                    break;
-            } else if (has_end && *cur_ts != AV_NOPTS_VALUE && *cur_ts >= end) {
-                break;
-            }
-
-            frame_count++;
            if (do_read_packets) {
                if (do_show_packets)
                    show_packet(w, fmt_ctx, &pkt, i++);
@@ -1714,33 +1590,9 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
        if (do_read_frames)
            while (process_frame(w, fmt_ctx, &frame, &pkt) > 0);
    }
-
-end:
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Could not read packets in interval ");
-        log_read_interval(interval, NULL, AV_LOG_ERROR);
-    }
-    return ret;
 }

-static void read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
-{
-    int i, ret = 0;
-    int64_t cur_ts = fmt_ctx->start_time;
-
-    if (read_intervals_nb == 0) {
-        ReadInterval interval = (ReadInterval) { .has_start = 0, .has_end = 0 };
-        ret = read_interval_packets(w, fmt_ctx, &interval, &cur_ts);
-    } else {
-        for (i = 0; i < read_intervals_nb; i++) {
-            ret = read_interval_packets(w, fmt_ctx, &read_intervals[i], &cur_ts);
-            if (ret < 0)
-                break;
-        }
-    }
-}
-
-static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx, int in_program)
+static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx)
 {
    AVStream *stream = fmt_ctx->streams[stream_idx];
    AVCodecContext *dec_ctx;
@@ -1752,7 +1604,7 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i

    av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);

-    writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM : SECTION_ID_STREAM);
+    writer_print_section_header(w, SECTION_ID_STREAM);

    print_int("index", stream->index);

@@ -1823,28 +1675,8 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
            else   print_str_opt("sample_fmt", "unknown");
            print_val("sample_rate",     dec_ctx->sample_rate, unit_hertz_str);
            print_int("channels",        dec_ctx->channels);
-
-            if (dec_ctx->channel_layout) {
-                av_bprint_clear(&pbuf);
-                av_bprint_channel_layout(&pbuf, dec_ctx->channels, dec_ctx->channel_layout);
-                print_str    ("channel_layout", pbuf.str);
-            } else {
-                print_str_opt("channel_layout", "unknown");
-            }
-
            print_int("bits_per_sample", av_get_bits_per_sample(dec_ctx->codec_id));
            break;
-
-        case AVMEDIA_TYPE_SUBTITLE:
-            if (dec_ctx->width)
-                print_int("width",       dec_ctx->width);
-            else
-                print_str_opt("width",   "N/A");
-            if (dec_ctx->height)
-                print_int("height",      dec_ctx->height);
-            else
-                print_str_opt("height",  "N/A");
-            break;
        }
    } else {
        print_str_opt("codec_type", "unknown");
@@ -1888,7 +1720,7 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
    } while (0)

    if (do_show_stream_disposition) {
-    writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM_DISPOSITION : SECTION_ID_STREAM_DISPOSITION);
+    writer_print_section_header(w, SECTION_ID_STREAM_DISPOSITION);
    PRINT_DISPOSITION(DEFAULT,          "default");
    PRINT_DISPOSITION(DUB,              "dub");
    PRINT_DISPOSITION(ORIGINAL,         "original");
@@ -1903,7 +1735,7 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
    writer_print_section_footer(w);
    }

-    show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
+    show_tags(w, stream->metadata, SECTION_ID_STREAM_TAGS);

    writer_print_section_footer(w);
    av_bprint_finalize(&pbuf, NULL);
@@ -1916,68 +1748,7 @@ static void show_streams(WriterContext *w, AVFormatContext *fmt_ctx)
    writer_print_section_header(w, SECTION_ID_STREAMS);
    for (i = 0; i < fmt_ctx->nb_streams; i++)
        if (selected_streams[i])
-            show_stream(w, fmt_ctx, i, 0);
-    writer_print_section_footer(w);
-}
-
-static void show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *program)
-{
-    int i;
-
-    writer_print_section_header(w, SECTION_ID_PROGRAM);
-    print_int("program_id", program->id);
-    print_int("program_num", program->program_num);
-    print_int("nb_streams", program->nb_stream_indexes);
-    print_int("pmt_pid", program->pmt_pid);
-    print_int("pcr_pid", program->pcr_pid);
-    print_ts("start_pts", program->start_time);
-    print_time("start_time", program->start_time, &AV_TIME_BASE_Q);
-    print_ts("end_pts", program->end_time);
-    print_time("end_time", program->end_time, &AV_TIME_BASE_Q);
-    show_tags(w, program->metadata, SECTION_ID_PROGRAM_TAGS);
-
-    writer_print_section_header(w, SECTION_ID_PROGRAM_STREAMS);
-    for (i = 0; i < program->nb_stream_indexes; i++) {
-        if (selected_streams[program->stream_index[i]])
-            show_stream(w, fmt_ctx, program->stream_index[i], 1);
-    }
-    writer_print_section_footer(w);
-
-    writer_print_section_footer(w);
-}
-
-static void show_programs(WriterContext *w, AVFormatContext *fmt_ctx)
-{
-    int i;
-
-    writer_print_section_header(w, SECTION_ID_PROGRAMS);
-    for (i = 0; i < fmt_ctx->nb_programs; i++) {
-        AVProgram *program = fmt_ctx->programs[i];
-        if (!program)
-            continue;
-        show_program(w, fmt_ctx, program);
-    }
-    writer_print_section_footer(w);
-}
-
-static void show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
-{
-    int i;
-
-    writer_print_section_header(w, SECTION_ID_CHAPTERS);
-    for (i = 0; i < fmt_ctx->nb_chapters; i++) {
-        AVChapter *chapter = fmt_ctx->chapters[i];
-
-        writer_print_section_header(w, SECTION_ID_CHAPTER);
-        print_int("id", chapter->id);
-        print_q  ("time_base", chapter->time_base, '/');
-        print_int("start", chapter->start);
-        print_time("start_time", chapter->start, &chapter->time_base);
-        print_int("end", chapter->end);
-        print_time("end_time", chapter->end, &chapter->time_base);
-        show_tags(w, chapter->metadata, SECTION_ID_CHAPTER_TAGS);
-        writer_print_section_footer(w);
-    }
+            show_stream(w, fmt_ctx, i);
    writer_print_section_footer(w);
 }

@@ -1989,7 +1760,6 @@ static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
    writer_print_section_header(w, SECTION_ID_FORMAT);
    print_str("filename",         fmt_ctx->filename);
    print_int("nb_streams",       fmt_ctx->nb_streams);
-    print_int("nb_programs",      fmt_ctx->nb_programs);
    print_str("format_name",      fmt_ctx->iformat->name);
    if (!do_bitexact) {
        if (fmt_ctx->iformat->long_name) print_str    ("format_long_name", fmt_ctx->iformat->long_name);
@@ -2001,7 +1771,6 @@ static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
    else           print_str_opt("size", "N/A");
    if (fmt_ctx->bit_rate > 0) print_val    ("bit_rate", fmt_ctx->bit_rate, unit_bit_per_second_str);
    else                       print_str_opt("bit_rate", "N/A");
-    print_int("probe_score", av_format_get_probe_score(fmt_ctx));
    show_tags(w, fmt_ctx->metadata, SECTION_ID_FORMAT_TAGS);

    writer_print_section_footer(w);
@@ -2024,10 +1793,9 @@ static void show_error(WriterContext *w, int err)

 static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
 {
-    int err, i, orig_nb_streams;
+    int err, i;
    AVFormatContext *fmt_ctx = NULL;
    AVDictionaryEntry *t;
-    AVDictionary **opts;

    if ((err = avformat_open_input(&fmt_ctx, filename,
                                   iformat, &format_opts)) < 0) {
@@ -2039,17 +1807,12 @@ static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
        return AVERROR_OPTION_NOT_FOUND;
    }

-    /* fill the streams in the format context */
-    opts = setup_find_stream_info_opts(fmt_ctx, codec_opts);
-    orig_nb_streams = fmt_ctx->nb_streams;

-    if ((err = avformat_find_stream_info(fmt_ctx, opts)) < 0) {
+    /* fill the streams in the format context */
+    if ((err = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
        print_error(filename, err);
        return err;
    }
-    for (i = 0; i < orig_nb_streams; i++)
-        av_dict_free(&opts[i]);
-    av_freep(&opts);

    av_dump_format(fmt_ctx, 0, filename, 0);

@@ -2059,25 +1822,16 @@ static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
        AVCodec *codec;

        if (stream->codec->codec_id == AV_CODEC_ID_PROBE) {
-            av_log(NULL, AV_LOG_WARNING,
+            av_log(NULL, AV_LOG_ERROR,
                   "Failed to probe codec for input stream %d\n",
                    stream->index);
        } else if (!(codec = avcodec_find_decoder(stream->codec->codec_id))) {
-            av_log(NULL, AV_LOG_WARNING,
+            av_log(NULL, AV_LOG_ERROR,
                    "Unsupported codec with id %d for input stream %d\n",
                    stream->codec->codec_id, stream->index);
-        } else {
-            AVDictionary *opts = filter_codec_opts(codec_opts, stream->codec->codec_id,
-                                                   fmt_ctx, stream, codec);
-            if (avcodec_open2(stream->codec, codec, &opts) < 0) {
-                av_log(NULL, AV_LOG_WARNING, "Could not open codec for input stream %d\n",
-                       stream->index);
-            }
-            if ((t = av_dict_get(opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
-                av_log(NULL, AV_LOG_ERROR, "Option %s for input stream %d not found\n",
-                       t->key, stream->index);
-                return AVERROR_OPTION_NOT_FOUND;
-            }
+        } else if (avcodec_open2(stream->codec, codec, NULL) < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Error while opening codec for input stream %d\n",
+                   stream->index);
        }
    }

@@ -2108,57 +1862,50 @@ static int probe_file(WriterContext *wctx, const char *filename)
    do_read_packets = do_show_packets || do_count_packets;

    ret = open_input_file(&fmt_ctx, filename);
-    if (ret < 0)
-        return ret;
+    if (ret >= 0) {
+        nb_streams_frames  = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_frames));
+        nb_streams_packets = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_packets));
+        selected_streams   = av_calloc(fmt_ctx->nb_streams, sizeof(*selected_streams));

-    nb_streams_frames  = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_frames));
-    nb_streams_packets = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_packets));
-    selected_streams   = av_calloc(fmt_ctx->nb_streams, sizeof(*selected_streams));
-
-    for (i = 0; i < fmt_ctx->nb_streams; i++) {
-        if (stream_specifier) {
-            ret = avformat_match_stream_specifier(fmt_ctx,
-                                                  fmt_ctx->streams[i],
-                                                  stream_specifier);
-            if (ret < 0)
-                goto end;
-            else
-                selected_streams[i] = ret;
-            ret = 0;
-        } else {
-            selected_streams[i] = 1;
+        for (i = 0; i < fmt_ctx->nb_streams; i++) {
+            if (stream_specifier) {
+                ret = avformat_match_stream_specifier(fmt_ctx,
+                                                      fmt_ctx->streams[i],
+                                                      stream_specifier);
+                if (ret < 0)
+                    goto end;
+                else
+                    selected_streams[i] = ret;
+            } else {
+                selected_streams[i] = 1;
+            }
        }
+
+        if (do_read_frames || do_read_packets) {
+            if (do_show_frames && do_show_packets &&
+                wctx->writer->flags & WRITER_FLAG_PUT_PACKETS_AND_FRAMES_IN_SAME_CHAPTER)
+                section_id = SECTION_ID_PACKETS_AND_FRAMES;
+            else if (do_show_packets && !do_show_frames)
+                section_id = SECTION_ID_PACKETS;
+            else // (!do_show_packets && do_show_frames)
+                section_id = SECTION_ID_FRAMES;
+            if (do_show_frames || do_show_packets)
+                writer_print_section_header(wctx, section_id);
+            read_packets(wctx, fmt_ctx);
+            if (do_show_frames || do_show_packets)
+                writer_print_section_footer(wctx);
+        }
+        if (do_show_streams)
+            show_streams(wctx, fmt_ctx);
+        if (do_show_format)
+            show_format(wctx, fmt_ctx);
+
+    end:
+        close_input_file(&fmt_ctx);
+        av_freep(&nb_streams_frames);
+        av_freep(&nb_streams_packets);
+        av_freep(&selected_streams);
    }
-
-    if (do_read_frames || do_read_packets) {
-        if (do_show_frames && do_show_packets &&
-            wctx->writer->flags & WRITER_FLAG_PUT_PACKETS_AND_FRAMES_IN_SAME_CHAPTER)
-            section_id = SECTION_ID_PACKETS_AND_FRAMES;
-        else if (do_show_packets && !do_show_frames)
-            section_id = SECTION_ID_PACKETS;
-        else // (!do_show_packets && do_show_frames)
-            section_id = SECTION_ID_FRAMES;
-        if (do_show_frames || do_show_packets)
-            writer_print_section_header(wctx, section_id);
-        read_packets(wctx, fmt_ctx);
-        if (do_show_frames || do_show_packets)
-            writer_print_section_footer(wctx);
-    }
-    if (do_show_programs)
-        show_programs(wctx, fmt_ctx);
-    if (do_show_streams)
-        show_streams(wctx, fmt_ctx);
-    if (do_show_chapters)
-        show_chapters(wctx, fmt_ctx);
-    if (do_show_format)
-        show_format(wctx, fmt_ctx);
-
-end:
-    close_input_file(&fmt_ctx);
-    av_freep(&nb_streams_frames);
-    av_freep(&nb_streams_packets);
-    av_freep(&selected_streams);
-
    return ret;
 }

@@ -2329,7 +2076,7 @@ static void opt_input_file(void *optctx, const char *arg)
        av_log(NULL, AV_LOG_ERROR,
                "Argument '%s' provided as input filename, but '%s' was already specified.\n",
                arg, input_filename);
-        exit_program(1);
+        exit(1);
    }
    if (!strcmp(arg, "-"))
        arg = "pipe:";
@@ -2352,145 +2099,6 @@ void show_help_default(const char *opt, const char *arg)
    show_help_children(avformat_get_class(), AV_OPT_FLAG_DECODING_PARAM);
 }

-/**
- * Parse interval specification, according to the format:
- * INTERVAL ::= [START|+START_OFFSET][%[END|+END_OFFSET]]
- * INTERVALS ::= INTERVAL[,INTERVALS]
-*/
-static int parse_read_interval(const char *interval_spec,
-                               ReadInterval *interval)
-{
-    int ret = 0;
-    char *next, *p, *spec = av_strdup(interval_spec);
-    if (!spec)
-        return AVERROR(ENOMEM);
-
-    if (!*spec) {
-        av_log(NULL, AV_LOG_ERROR, "Invalid empty interval specification\n");
-        ret = AVERROR(EINVAL);
-        goto end;
-    }
-
-    p = spec;
-    next = strchr(spec, '%');
-    if (next)
-        *next++ = 0;
-
-    /* parse first part */
-    if (*p) {
-        interval->has_start = 1;
-
-        if (*p == '+') {
-            interval->start_is_offset = 1;
-            p++;
-        } else {
-            interval->start_is_offset = 0;
-        }
-
-        ret = av_parse_time(&interval->start, p, 1);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Invalid interval start specification '%s'\n", p);
-            goto end;
-        }
-    } else {
-        interval->has_start = 0;
-    }
-
-    /* parse second part */
-    p = next;
-    if (p && *p) {
-        int64_t us;
-        interval->has_end = 1;
-
-        if (*p == '+') {
-            interval->end_is_offset = 1;
-            p++;
-        } else {
-            interval->end_is_offset = 0;
-        }
-
-        if (interval->end_is_offset && *p == '#') {
-            long long int lli;
-            char *tail;
-            interval->duration_frames = 1;
-            p++;
-            lli = strtoll(p, &tail, 10);
-            if (*tail || lli < 0) {
-                av_log(NULL, AV_LOG_ERROR,
-                       "Invalid or negative value '%s' for duration number of frames\n", p);
-                goto end;
-            }
-            interval->end = lli;
-        } else {
-            ret = av_parse_time(&us, p, 1);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Invalid interval end/duration specification '%s'\n", p);
-                goto end;
-            }
-            interval->end = us;
-        }
-    } else {
-        interval->has_end = 0;
-    }
-
-end:
-    av_free(spec);
-    return ret;
-}
-
-static int parse_read_intervals(const char *intervals_spec)
-{
-    int ret, n, i;
-    char *p, *spec = av_strdup(intervals_spec);
-    if (!spec)
-        return AVERROR(ENOMEM);
-
-    /* preparse specification, get number of intervals */
-    for (n = 0, p = spec; *p; p++)
-        if (*p == ',')
-            n++;
-    n++;
-
-    read_intervals = av_malloc(n * sizeof(*read_intervals));
-    if (!read_intervals) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    read_intervals_nb = n;
-
-    /* parse intervals */
-    p = spec;
-    for (i = 0; p; i++) {
-        char *next;
-
-        av_assert0(i < read_intervals_nb);
-        next = strchr(p, ',');
-        if (next)
-            *next++ = 0;
-
-        read_intervals[i].id = i;
-        ret = parse_read_interval(p, &read_intervals[i]);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Error parsing read interval #%d '%s'\n",
-                   i, p);
-            goto end;
-        }
-        av_log(NULL, AV_LOG_VERBOSE, "Parsed log interval ");
-        log_read_interval(&read_intervals[i], NULL, AV_LOG_VERBOSE);
-        p = next;
-    }
-    av_assert0(i == read_intervals_nb);
-
-end:
-    av_free(spec);
-    return ret;
-}
-
-static int opt_read_intervals(void *optctx, const char *opt, const char *arg)
-{
-    return parse_read_intervals(arg);
-}
-
 static int opt_pretty(void *optctx, const char *opt, const char *arg)
 {
    show_value_unit              = 1;
@@ -2543,7 +2151,6 @@ static int opt_show_versions(const char *opt, const char *arg)
        return 0;                                                       \
    }

-DEFINE_OPT_SHOW_SECTION(chapters,         CHAPTERS);
 DEFINE_OPT_SHOW_SECTION(error,            ERROR);
 DEFINE_OPT_SHOW_SECTION(format,           FORMAT);
 DEFINE_OPT_SHOW_SECTION(frames,           FRAMES);
@@ -2551,7 +2158,6 @@ DEFINE_OPT_SHOW_SECTION(library_versions, LIBRARY_VERSIONS);
 DEFINE_OPT_SHOW_SECTION(packets,          PACKETS);
 DEFINE_OPT_SHOW_SECTION(program_version,  PROGRAM_VERSION);
 DEFINE_OPT_SHOW_SECTION(streams,          STREAMS);
-DEFINE_OPT_SHOW_SECTION(programs,         PROGRAMS);

 static const OptionDef real_options[] = {
 #include "cmdutils_common_opts.h"
@@ -2578,9 +2184,7 @@ static const OptionDef real_options[] = {
    { "show_entries", HAS_ARG, {.func_arg = opt_show_entries},
      "show a set of specified entries", "entry_list" },
    { "show_packets", 0, {(void*)&opt_show_packets}, "show packets info" },
-    { "show_programs", 0, {(void*)&opt_show_programs}, "show programs info" },
    { "show_streams", 0, {(void*)&opt_show_streams}, "show streams info" },
-    { "show_chapters", 0, {(void*)&opt_show_chapters}, "show chapters info" },
    { "count_frames", OPT_BOOL, {(void*)&do_count_frames}, "count the number of frames per stream" },
    { "count_packets", OPT_BOOL, {(void*)&do_count_packets}, "count the number of packets per stream" },
    { "show_program_version",  0, {(void*)&opt_show_program_version},  "show ffprobe version" },
@@ -2589,7 +2193,6 @@ static const OptionDef real_options[] = {
    { "show_private_data", OPT_BOOL, {(void*)&show_private_data}, "show private data" },
    { "private",           OPT_BOOL, {(void*)&show_private_data}, "same as show_private_data" },
    { "bitexact", OPT_BOOL, {&do_bitexact}, "force bitexact output" },
-    { "read_intervals", HAS_ARG, {.func_arg = opt_read_intervals}, "set read intervals", "read_intervals" },
    { "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {.func_arg = opt_default}, "generic catch all option", "" },
    { "i", HAS_ARG, {.func_arg = opt_input_file_i}, "read specified file", "input_file"},
    { NULL, },
@@ -2621,7 +2224,7 @@ int main(int argc, char **argv)
    int ret, i;

    av_log_set_flags(AV_LOG_SKIP_REPEATED);
-    register_exit(ffprobe_cleanup);
+    atexit(exit_program);

    options = real_options;
    parse_loglevel(argc, argv, options);
@@ -2636,17 +2239,14 @@ int main(int argc, char **argv)
    parse_options(NULL, argc, argv, options, opt_input_file);

    /* mark things to show, based on -show_entries */
-    SET_DO_SHOW(CHAPTERS, chapters);
    SET_DO_SHOW(ERROR, error);
    SET_DO_SHOW(FORMAT, format);
    SET_DO_SHOW(FRAMES, frames);
    SET_DO_SHOW(LIBRARY_VERSIONS, library_versions);
    SET_DO_SHOW(PACKETS, packets);
    SET_DO_SHOW(PROGRAM_VERSION, program_version);
-    SET_DO_SHOW(PROGRAMS, programs);
    SET_DO_SHOW(STREAMS, streams);
    SET_DO_SHOW(STREAM_DISPOSITION, stream_disposition);
-    SET_DO_SHOW(PROGRAM_STREAM_DISPOSITION, stream_disposition);

    if (do_bitexact && (do_show_program_version || do_show_library_versions)) {
        av_log(NULL, AV_LOG_ERROR,
@@ -2684,7 +2284,7 @@ int main(int argc, char **argv)
            ffprobe_show_library_versions(wctx);

        if (!input_filename &&
-            ((do_show_format || do_show_programs || do_show_streams || do_show_chapters || do_show_packets || do_show_error) ||
+            ((do_show_format || do_show_streams || do_show_packets || do_show_error) ||
             (!do_show_program_version && !do_show_library_versions))) {
            show_usage();
            av_log(NULL, AV_LOG_ERROR, "You have to specify one input file.\n");
@@ -2702,7 +2302,6 @@ int main(int argc, char **argv)

 end:
    av_freep(&print_format);
-    av_freep(&read_intervals);

    uninit_opts();
    for (i = 0; i < FF_ARRAY_ELEMS(sections); i++)
@@ -2710,5 +2309,5 @@ end:

    avformat_network_deinit();

-    return ret < 0;
+    return ret;
 }
--- a/ffserver.c
+++ b/ffserver.c
@@ -36,7 +36,6 @@
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
 #include "libavformat/rtpdec.h"
-#include "libavformat/rtpproto.h"
 #include "libavformat/rtsp.h"
 #include "libavformat/avio_internal.h"
 #include "libavformat/internal.h"
@@ -48,7 +47,6 @@
 #include "libavutil/dict.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/mathematics.h"
-#include "libavutil/pixdesc.h"
 #include "libavutil/random_seed.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/opt.h"
@@ -65,6 +63,9 @@
 #include <time.h>
 #include <sys/wait.h>
 #include <signal.h>
+#if HAVE_DLFCN_H
+#include <dlfcn.h>
+#endif

 #include "cmdutils.h"

@@ -307,7 +308,7 @@ static int rtp_new_av_stream(HTTPContext *c,

 static const char *my_program_name;

-static const char *config_filename;
+static const char *config_filename = "/etc/ffserver.conf";

 static int ffserver_debug;
 static int no_launch;
@@ -403,14 +404,14 @@ static int resolve_host(struct in_addr *sin_addr, const char *hostname)
    return 0;
 }

-static char *ctime1(char *buf2, int buf_size)
+static char *ctime1(char *buf2)
 {
    time_t ti;
    char *p;

    ti = time(NULL);
    p = ctime(&ti);
-    av_strlcpy(buf2, p, buf_size);
+    strcpy(buf2, p);
    p = buf2 + strlen(p) - 1;
    if (*p == '\n')
        *p = '\0';
@@ -423,7 +424,7 @@ static void http_vlog(const char *fmt, va_list vargs)
    if (logfile) {
        if (print_prefix) {
            char buf[32];
-            ctime1(buf, sizeof(buf));
+            ctime1(buf);
            fprintf(logfile, "%s ", buf);
        }
        print_prefix = strstr(fmt, "\n") != NULL;
@@ -1132,7 +1133,7 @@ static int extract_rates(char *rates, int ratelen, const char *request)
        if (av_strncasecmp(p, "Pragma:", 7) == 0) {
            const char *q = p + 7;

-            while (*q && *q != '\n' && av_isspace(*q))
+            while (*q && *q != '\n' && isspace(*q))
                q++;

            if (av_strncasecmp(q, "stream-switch-entry=", 20) == 0) {
@@ -1154,7 +1155,7 @@ static int extract_rates(char *rates, int ratelen, const char *request)
                    if (stream_no < ratelen && stream_no >= 0)
                        rates[stream_no] = rate_no;

-                    while (*q && *q != '\n' && !av_isspace(*q))
+                    while (*q && *q != '\n' && !isspace(*q))
                        q++;
                }

@@ -1265,7 +1266,7 @@ static void get_word(char *buf, int buf_size, const char **pp)
    p = *pp;
    skip_spaces(&p);
    q = buf;
-    while (!av_isspace(*p) && *p != '\0') {
+    while (!isspace(*p) && *p != '\0') {
        if ((q - buf) < buf_size - 1)
            *q++ = *p;
        p++;
@@ -1282,7 +1283,7 @@ static void get_arg(char *buf, int buf_size, const char **pp)
    int quote;

    p = *pp;
-    while (av_isspace(*p)) p++;
+    while (isspace(*p)) p++;
    q = buf;
    quote = 0;
    if (*p == '\"' || *p == '\'')
@@ -1292,7 +1293,7 @@ static void get_arg(char *buf, int buf_size, const char **pp)
            if (*p == quote)
                break;
        } else {
-            if (av_isspace(*p))
+            if (isspace(*p))
                break;
        }
        if (*p == '\0')
@@ -1396,7 +1397,7 @@ static IPAddressACL* parse_dynamic_acl(FFStream *stream, HTTPContext *c)
            break;
        line_num++;
        p = line;
-        while (av_isspace(*p))
+        while (isspace(*p))
            p++;
        if (*p == '\0' || *p == '#')
            continue;
@@ -1547,7 +1548,7 @@ static int http_parse_request(HTTPContext *c)
    for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
        if (av_strncasecmp(p, "User-Agent:", 11) == 0) {
            useragent = p + 11;
-            if (*useragent && *useragent != '\n' && av_isspace(*useragent))
+            if (*useragent && *useragent != '\n' && isspace(*useragent))
                useragent++;
            break;
        }
@@ -1675,7 +1676,7 @@ static int http_parse_request(HTTPContext *c)
            char *eoh;
            char hostbuf[260];

-            while (av_isspace(*hostinfo))
+            while (isspace(*hostinfo))
                hostinfo++;

            eoh = strchr(hostinfo, '\n');
@@ -3914,7 +3915,7 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
        av->rc_buffer_aggressivity = 1.0;

        if (!av->rc_eq)
-            av->rc_eq = av_strdup("tex^qComp");
+            av->rc_eq = "tex^qComp";
        if (!av->i_quant_factor)
            av->i_quant_factor = -0.8;
        if (!av->b_quant_factor)
@@ -3962,6 +3963,32 @@ static enum AVCodecID opt_video_codec(const char *arg)
    return p->id;
 }

+/* simplistic plugin support */
+
+#if HAVE_DLOPEN
+static void load_module(const char *filename)
+{
+    void *dll;
+    void (*init_func)(void);
+    dll = dlopen(filename, RTLD_NOW);
+    if (!dll) {
+        fprintf(stderr, "Could not load module '%s' - %s\n",
+                filename, dlerror());
+        return;
+    }
+
+    init_func = dlsym(dll, "ffserver_module_init");
+    if (!init_func) {
+        fprintf(stderr,
+                "%s: init function 'ffserver_module_init()' not found\n",
+                filename);
+        dlclose(dll);
+    }
+
+    init_func();
+}
+#endif
+
 static int ffserver_opt_default(const char *opt, const char *arg,
                       AVCodecContext *avctx, int type)
 {
@@ -4082,7 +4109,7 @@ static int parse_ffconfig(const char *filename)
            break;
        line_num++;
        p = line;
-        while (av_isspace(*p))
+        while (isspace(*p))
            p++;
        if (*p == '\0' || *p == '#')
            continue;
@@ -4133,7 +4160,7 @@ static int parse_ffconfig(const char *filename)
        } else if (!av_strcasecmp(cmd, "MaxBandwidth")) {
            int64_t llval;
            get_arg(arg, sizeof(arg), &p);
-            llval = strtoll(arg, NULL, 10);
+            llval = atoll(arg);
            if (llval < 10 || llval > 10000000) {
                ERROR("Invalid MaxBandwidth: %s\n", arg);
            } else
@@ -4219,7 +4246,7 @@ static int parse_ffconfig(const char *filename)
                get_arg(arg, sizeof(arg), &p);
                p1 = arg;
                fsize = strtod(p1, &p1);
-                switch(av_toupper(*p1)) {
+                switch(toupper(*p1)) {
                case 'K':
                    fsize *= 1024;
                    break;
@@ -4435,14 +4462,6 @@ static int parse_ffconfig(const char *filename)
                    video_enc.time_base.den = frame_rate.num;
                }
            }
-        } else if (!av_strcasecmp(cmd, "PixelFormat")) {
-            get_arg(arg, sizeof(arg), &p);
-            if (stream) {
-                video_enc.pix_fmt = av_get_pix_fmt(arg);
-                if (video_enc.pix_fmt == AV_PIX_FMT_NONE) {
-                    ERROR("Unknown pixel format: %s\n", arg);
-                }
-            }
        } else if (!av_strcasecmp(cmd, "VideoGopSize")) {
            get_arg(arg, sizeof(arg), &p);
            if (stream)
@@ -4535,6 +4554,14 @@ static int parse_ffconfig(const char *filename)
                    ERROR("VideoQMin out of range\n");
                }
            }
+        } else if (!av_strcasecmp(cmd, "LumaElim")) {
+            get_arg(arg, sizeof(arg), &p);
+            if (stream)
+                video_enc.luma_elim_threshold = atoi(arg);
+        } else if (!av_strcasecmp(cmd, "ChromaElim")) {
+            get_arg(arg, sizeof(arg), &p);
+            if (stream)
+                video_enc.chroma_elim_threshold = atoi(arg);
        } else if (!av_strcasecmp(cmd, "LumiMask")) {
            get_arg(arg, sizeof(arg), &p);
            if (stream)
@@ -4626,7 +4653,12 @@ static int parse_ffconfig(const char *filename)
                redirect = NULL;
            }
        } else if (!av_strcasecmp(cmd, "LoadModule")) {
-            ERROR("Loadable modules no longer supported\n");
+            get_arg(arg, sizeof(arg), &p);
+#if HAVE_DLOPEN
+            load_module(arg);
+#else
+            ERROR("Module support not compiled into this version: '%s'\n", arg);
+#endif
        } else {
            ERROR("Incorrect keyword: '%s'\n", cmd);
        }
@@ -4691,8 +4723,6 @@ int main(int argc, char **argv)
 {
    struct sigaction sigact = { { 0 } };

-    config_filename = av_strdup("/etc/ffserver.conf");
-
    parse_loglevel(argc, argv, options);
    av_register_all();
    avformat_network_init();
@@ -4715,7 +4745,6 @@ int main(int argc, char **argv)
        fprintf(stderr, "Incorrect config file - exiting.\n");
        exit(1);
    }
-    av_freep(&config_filename);

    /* open log file if needed */
    if (logfilename[0] != '\0') {
--- a/libavcodec/012v.c
+++ b/libavcodec/012v.c
@@ -26,12 +26,18 @@

 static av_cold int zero12v_decode_init(AVCodecContext *avctx)
 {
-    avctx->pix_fmt             = AV_PIX_FMT_YUV422P16;
+    avctx->pix_fmt             = PIX_FMT_YUV422P16;
    avctx->bits_per_raw_sample = 10;

-    if (avctx->codec_tag == MKTAG('a', '1', '2', 'v'))
-        avpriv_request_sample(avctx, "transparency");
+    avctx->coded_frame = avcodec_alloc_frame();
+    if (!avctx->coded_frame)
+        return AVERROR(ENOMEM);

+    if (avctx->codec_tag == MKTAG('a', '1', '2', 'v'))
+        av_log_ask_for_sample(avctx, "Samples with actual transparency needed\n");
+
+    avctx->coded_frame->pict_type = AV_PICTURE_TYPE_I;
+    avctx->coded_frame->key_frame = 1;
    return 0;
 }

@@ -40,33 +46,28 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
 {
    int line = 0, ret;
    const int width = avctx->width;
-    AVFrame *pic = data;
+    AVFrame *pic = avctx->coded_frame;
    uint16_t *y, *u, *v;
    const uint8_t *line_end, *src = avpkt->data;
    int stride = avctx->width * 8 / 3;

+    if (pic->data[0])
+        avctx->release_buffer(avctx, pic);
+
    if (width == 1) {
        av_log(avctx, AV_LOG_ERROR, "Width 1 not supported.\n");
        return AVERROR_INVALIDDATA;
    }
-
-    if (   avctx->codec_tag == MKTAG('0', '1', '2', 'v')
-        && avpkt->size % avctx->height == 0
-        && avpkt->size / avctx->height * 3 >= width * 8)
-        stride = avpkt->size / avctx->height;
-
    if (avpkt->size < avctx->height * stride) {
        av_log(avctx, AV_LOG_ERROR, "Packet too small: %d instead of %d\n",
               avpkt->size, avctx->height * stride);
        return AVERROR_INVALIDDATA;
    }

-    if ((ret = ff_get_buffer(avctx, pic, 0)) < 0)
+    pic->reference = 0;
+    if ((ret = ff_get_buffer(avctx, pic)) < 0)
        return ret;

-    pic->pict_type = AV_PICTURE_TYPE_I;
-    pic->key_frame = 1;
-
    y = (uint16_t *)pic->data[0];
    u = (uint16_t *)pic->data[1];
    v = (uint16_t *)pic->data[2];
@@ -144,16 +145,28 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
    }

    *got_frame = 1;
+    *(AVFrame*)data= *avctx->coded_frame;

    return avpkt->size;
 }

+static av_cold int zero12v_decode_close(AVCodecContext *avctx)
+{
+    AVFrame *pic = avctx->coded_frame;
+    if (pic->data[0])
+        avctx->release_buffer(avctx, pic);
+    av_freep(&avctx->coded_frame);
+
+    return 0;
+}
+
 AVCodec ff_zero12v_decoder = {
    .name           = "012v",
-    .long_name      = NULL_IF_CONFIG_SMALL("Uncompressed 4:2:2 10-bit"),
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_012V,
    .init           = zero12v_decode_init,
+    .close          = zero12v_decode_close,
    .decode         = zero12v_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("Uncompressed 4:2:2 10-bit"),
 };
--- a/libavcodec/4xm.c
+++ b/libavcodec/4xm.c
@@ -24,8 +24,6 @@
 * 4XM codec.
 */

-#include "libavutil/avassert.h"
-#include "libavutil/frame.h"
 #include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "bytestream.h"
@@ -33,6 +31,7 @@
 #include "get_bits.h"
 #include "internal.h"

+#include "libavutil/avassert.h"

 #define BLOCK_TYPE_VLC_BITS 5
 #define ACDC_VLC_BITS 9
@@ -131,7 +130,7 @@ typedef struct CFrameBuffer {
 typedef struct FourXContext {
    AVCodecContext *avctx;
    DSPContext dsp;
-    AVFrame *current_picture, *last_picture;
+    AVFrame current_picture, last_picture;
    GetBitContext pre_gb;          ///< ac/dc prefix
    GetBitContext gb;
    GetByteContext g;
@@ -139,7 +138,7 @@ typedef struct FourXContext {
    int mv[256];
    VLC pre_vlc;
    int last_dc;
-    DECLARE_ALIGNED(16, int16_t, block)[6][64];
+    DECLARE_ALIGNED(16, DCTELEM, block)[6][64];
    void *bitstream_buffer;
    unsigned int bitstream_buffer_size;
    int version;
@@ -154,7 +153,7 @@ typedef struct FourXContext {

 #define MULTIPLY(var, const) (((var) * (const)) >> 16)

-static void idct(int16_t block[64])
+static void idct(DCTELEM block[64])
 {
    int tmp0, tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7;
    int tmp10, tmp11, tmp12, tmp13;
@@ -256,15 +255,15 @@ static av_cold void init_vlcs(FourXContext *f)
    }
 }

-static void init_mv(FourXContext *f, int linesize)
+static void init_mv(FourXContext *f)
 {
    int i;

    for (i = 0; i < 256; i++) {
        if (f->version > 1)
-            f->mv[i] = mv[i][0] + mv[i][1] * linesize / 2;
+            f->mv[i] = mv[i][0] + mv[i][1] * f->current_picture.linesize[0] / 2;
        else
-            f->mv[i] = (i & 15) - 8 + ((i >> 4) - 8) * linesize / 2;
+            f->mv[i] = (i & 15) - 8 + ((i >> 4) - 8) * f->current_picture.linesize[0] / 2;
    }
 }

@@ -341,28 +340,70 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
    int code        = get_vlc2(&f->gb,
                               block_type_vlc[1 - (f->version > 1)][index].table,
                               BLOCK_TYPE_VLC_BITS, 1);
-    uint16_t *start = (uint16_t *)f->last_picture->data[0];
+    uint16_t *start = (uint16_t *)f->last_picture.data[0];
    uint16_t *end   = start + stride * (f->avctx->height - h + 1) - (1 << log2w);
    int ret;
-    int scale   = 1;
-    unsigned dc = 0;

    av_assert0(code >= 0 && code <= 6 && log2w >= 0);

-    if (code == 1) {
+    if (code == 0) {
+        if (bytestream2_get_bytes_left(&f->g) < 1) {
+            av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
+            return AVERROR_INVALIDDATA;
+        }
+        src += f->mv[bytestream2_get_byteu(&f->g)];
+        if (start > src || src > end) {
+            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
+            return AVERROR_INVALIDDATA;
+        }
+        mcdc(dst, src, log2w, h, stride, 1, 0);
+    } else if (code == 1) {
        log2h--;
        if ((ret = decode_p_block(f, dst, src, log2w, log2h, stride)) < 0)
            return ret;
-        return decode_p_block(f, dst + (stride << log2h),
-                              src + (stride << log2h),
-                              log2w, log2h, stride);
+        if ((ret = decode_p_block(f, dst + (stride << log2h),
+                                  src + (stride << log2h),
+                                  log2w, log2h, stride)) < 0)
+            return ret;
    } else if (code == 2) {
        log2w--;
        if ((ret = decode_p_block(f, dst , src, log2w, log2h, stride)) < 0)
            return ret;
-        return decode_p_block(f, dst + (1 << log2w),
-                              src + (1 << log2w),
-                              log2w, log2h, stride);
+        if ((ret = decode_p_block(f, dst + (1 << log2w),
+                                  src + (1 << log2w),
+                                  log2w, log2h, stride)) < 0)
+            return ret;
+    } else if (code == 3 && f->version < 2) {
+        if (start > src || src > end) {
+            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
+            return AVERROR_INVALIDDATA;
+        }
+        mcdc(dst, src, log2w, h, stride, 1, 0);
+    } else if (code == 4) {
+        if (bytestream2_get_bytes_left(&f->g) < 1) {
+            av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
+            return AVERROR_INVALIDDATA;
+        }
+        src += f->mv[bytestream2_get_byteu(&f->g)];
+        if (start > src || src > end) {
+            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
+            return AVERROR_INVALIDDATA;
+        }
+        if (bytestream2_get_bytes_left(&f->g2) < 2){
+            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
+            return AVERROR_INVALIDDATA;
+        }
+        mcdc(dst, src, log2w, h, stride, 1, bytestream2_get_le16u(&f->g2));
+    } else if (code == 5) {
+        if (bytestream2_get_bytes_left(&f->g2) < 2) {
+            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
+            return AVERROR_INVALIDDATA;
+        }
+        if (start > src || src > end) {
+            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
+            return AVERROR_INVALIDDATA;
+        }
+        mcdc(dst, src, log2w, h, stride, 0, bytestream2_get_le16u(&f->g2));
    } else if (code == 6) {
        if (bytestream2_get_bytes_left(&f->g2) < 4) {
            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
@@ -375,73 +416,37 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
            dst[0]      = bytestream2_get_le16u(&f->g2);
            dst[stride] = bytestream2_get_le16u(&f->g2);
        }
-        return 0;
    }
-
-    if ((code&3)==0 && bytestream2_get_bytes_left(&f->g) < 1) {
-        av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
-        return AVERROR_INVALIDDATA;
-    }
-
-    if (code == 0) {
-        src  += f->mv[bytestream2_get_byte(&f->g)];
-    } else if (code == 3 && f->version >= 2) {
-        return 0;
-    } else if (code == 4) {
-        src  += f->mv[bytestream2_get_byte(&f->g)];
-        if (bytestream2_get_bytes_left(&f->g2) < 2){
-            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
-            return AVERROR_INVALIDDATA;
-        }
-        dc    = bytestream2_get_le16(&f->g2);
-    } else if (code == 5) {
-        if (bytestream2_get_bytes_left(&f->g2) < 2){
-            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
-            return AVERROR_INVALIDDATA;
-        }
-        av_assert0(start <= src && src <= end);
-        scale = 0;
-        dc    = bytestream2_get_le16(&f->g2);
-    }
-
-    if (start > src || src > end) {
-        av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
-        return AVERROR_INVALIDDATA;
-    }
-
-    mcdc(dst, src, log2w, h, stride, scale, dc);
-
    return 0;
 }

-static int decode_p_frame(FourXContext *f, AVFrame *frame,
-                          const uint8_t *buf, int length)
+static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length)
 {
    int x, y;
    const int width  = f->avctx->width;
    const int height = f->avctx->height;
-    uint16_t *dst    = (uint16_t *)frame->data[0];
-    const int stride =             frame->linesize[0] >> 1;
-    uint16_t *src;
+    uint16_t *src    = (uint16_t *)f->last_picture.data[0];
+    uint16_t *dst    = (uint16_t *)f->current_picture.data[0];
+    const int stride =             f->current_picture.linesize[0] >> 1;
    unsigned int bitstream_size, bytestream_size, wordstream_size, extra,
                 bytestream_offset, wordstream_offset;
    int ret;

-    if (!f->last_picture->data[0]) {
-        if ((ret = ff_get_buffer(f->avctx, f->last_picture,
-                                 AV_GET_BUFFER_FLAG_REF)) < 0) {
+    if (!f->last_picture.data[0]) {
+        if ((ret = ff_get_buffer(f->avctx, &f->last_picture)) < 0) {
+            av_log(f->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
            return ret;
        }
        for (y=0; y<f->avctx->height; y++)
-            memset(f->last_picture->data[0] + y*f->last_picture->linesize[0], 0, 2*f->avctx->width);
+            memset(f->last_picture.data[0] + y*f->last_picture.linesize[0], 0, 2*f->avctx->width);
    }

-    src = (uint16_t *)f->last_picture->data[0];
-
    if (f->version > 1) {
+        if (length < 20)
+            return AVERROR_INVALIDDATA;
        extra           = 20;
        if (length < extra)
-            return AVERROR_INVALIDDATA;
+            return -1;
        bitstream_size  = AV_RL32(buf + 8);
        wordstream_size = AV_RL32(buf + 12);
        bytestream_size = AV_RL32(buf + 16);
@@ -458,7 +463,7 @@ static int decode_p_frame(FourXContext *f, AVFrame *frame,
        extra > length - bytestream_size - bitstream_size - wordstream_size) {
        av_log(f->avctx, AV_LOG_ERROR, "lengths %d %d %d %d\n", bitstream_size, bytestream_size, wordstream_size,
        bitstream_size+ bytestream_size+ wordstream_size - length);
-        return AVERROR_INVALIDDATA;
+        return -1;
    }

    av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size,
@@ -478,7 +483,7 @@ static int decode_p_frame(FourXContext *f, AVFrame *frame,
    bytestream2_init(&f->g, buf + bytestream_offset,
                     length - bytestream_offset);

-    init_mv(f, frame->linesize[0]);
+    init_mv(f);

    for (y = 0; y < height; y += 8) {
        for (x = 0; x < width; x += 8)
@@ -495,7 +500,7 @@ static int decode_p_frame(FourXContext *f, AVFrame *frame,
 * decode block and dequantize.
 * Note this is almost identical to MJPEG.
 */
-static int decode_i_block(FourXContext *f, int16_t *block)
+static int decode_i_block(FourXContext *f, DCTELEM *block)
 {
    int code, i, j, level, val;

@@ -506,10 +511,8 @@ static int decode_i_block(FourXContext *f, int16_t *block)

    /* DC coef */
    val = get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3);
-    if (val >> 4) {
+    if (val >> 4)
        av_log(f->avctx, AV_LOG_ERROR, "error dc run != 0\n");
-        return AVERROR_INVALIDDATA;
-    }

    if (val)
        val = get_xbits(&f->gb, val);
@@ -527,12 +530,7 @@ static int decode_i_block(FourXContext *f, int16_t *block)
        if (code == 0xf0) {
            i += 16;
        } else {
-            if (code & 0xf) {
-                level = get_xbits(&f->gb, code & 0xf);
-            } else {
-                av_log(f->avctx, AV_LOG_ERROR, "0 coeff\n");
-                return AVERROR_INVALIDDATA;
-            }
+            level = get_xbits(&f->gb, code & 0xf);
            i    += code >> 4;
            if (i >= 64) {
                av_log(f->avctx, AV_LOG_ERROR, "run %d oveflow\n", i);
@@ -550,12 +548,12 @@ static int decode_i_block(FourXContext *f, int16_t *block)
    return 0;
 }

-static inline void idct_put(FourXContext *f, AVFrame *frame, int x, int y)
+static inline void idct_put(FourXContext *f, int x, int y)
 {
-    int16_t (*block)[64] = f->block;
-    int stride           = frame->linesize[0] >> 1;
+    DCTELEM (*block)[64] = f->block;
+    int stride           = f->current_picture.linesize[0] >> 1;
    int i;
-    uint16_t *dst = ((uint16_t*)frame->data[0]) + y * stride + x;
+    uint16_t *dst = ((uint16_t*)f->current_picture.data[0]) + y * stride + x;

    for (i = 0; i < 4; i++) {
        block[i][0] += 0x80 * 8 * 8;
@@ -573,7 +571,7 @@ static inline void idct_put(FourXContext *f, AVFrame *frame, int x, int y)
     * cr = (-1b - 4g + 5r) / 14 */
    for (y = 0; y < 8; y++) {
        for (x = 0; x < 8; x++) {
-            int16_t *temp = block[(x >> 2) + 2 * (y >> 2)] +
+            DCTELEM *temp = block[(x >> 2) + 2 * (y >> 2)] +
                            2 * (x & 3) + 2 * 8 * (y & 3); // FIXME optimize
            int cb = block[4][x + 8 * y];
            int cr = block[5][x + 8 * y];
@@ -598,14 +596,13 @@ static inline void idct_put(FourXContext *f, AVFrame *frame, int x, int y)

 static int decode_i_mb(FourXContext *f)
 {
-    int ret;
    int i;

    f->dsp.clear_blocks(f->block[0]);

    for (i = 0; i < 6; i++)
-        if ((ret = decode_i_block(f, f->block[i])) < 0)
-            return ret;
+        if (decode_i_block(f, f->block[i]) < 0)
+            return -1;

    return 0;
 }
@@ -649,11 +646,6 @@ static const uint8_t *read_huffman_tables(FourXContext *f,
    while ((ptr - buf) & 3)
        ptr++; // 4byte align

-    if (ptr > ptr_end) {
-        av_log(f->avctx, AV_LOG_ERROR, "ptr overflow in read_huffman_tables\n");
-        return NULL;
-    }
-
    for (j = 257; j < 512; j++) {
        int min_freq[2] = { 256 * 256, 256 * 256 };
        int smallest[2] = { 0, 0 };
@@ -715,14 +707,14 @@ static int mix(int c0, int c1)
    return red / 3 * 1024 + green / 3 * 32 + blue / 3;
 }

-static int decode_i2_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, int length)
+static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
 {
    int x, y, x2, y2;
    const int width  = f->avctx->width;
    const int height = f->avctx->height;
    const int mbs    = (FFALIGN(width, 16) >> 4) * (FFALIGN(height, 16) >> 4);
-    uint16_t *dst    = (uint16_t*)frame->data[0];
-    const int stride =            frame->linesize[0]>>1;
+    uint16_t *dst    = (uint16_t*)f->current_picture.data[0];
+    const int stride =            f->current_picture.linesize[0]>>1;
    const uint8_t *buf_end = buf + length;
    GetByteContext g3;

@@ -764,9 +756,9 @@ static int decode_i2_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf,
    return 0;
 }

-static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, int length)
+static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length)
 {
-    int x, y, ret;
+    int x, y;
    const int width  = f->avctx->width;
    const int height = f->avctx->height;
    const unsigned int bitstream_size = AV_RL32(buf);
@@ -788,7 +780,7 @@ static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, i
        || prestream_size > (1 << 26)) {
        av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d %d\n",
               prestream_size, bitstream_size, length);
-        return AVERROR_INVALIDDATA;
+        return -1;
    }

    prestream = read_huffman_tables(f, prestream, prestream_size);
@@ -797,8 +789,6 @@ static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, i
        return AVERROR_INVALIDDATA;
    }

-    av_assert0(prestream <= buf + length);
-
    init_get_bits(&f->gb, buf + 4, 8 * bitstream_size);

    prestream_size = length + buf - prestream;
@@ -817,10 +807,10 @@ static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, i

    for (y = 0; y < height; y += 16) {
        for (x = 0; x < width; x += 16) {
-            if ((ret = decode_i_mb(f)) < 0)
-                return ret;
+            if (decode_i_mb(f) < 0)
+                return -1;

-            idct_put(f, frame, x, y);
+            idct_put(f, x, y);
        }
    }

@@ -837,7 +827,8 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    int buf_size          = avpkt->size;
    FourXContext *const f = avctx->priv_data;
    AVFrame *picture      = data;
-    int i, frame_4cc, frame_size, ret;
+    AVFrame *p, temp;
+    int i, frame_4cc, frame_size;

    if (buf_size < 20)
        return AVERROR_INVALIDDATA;
@@ -897,7 +888,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
        // explicit check needed as memcpy below might not catch a NULL
        if (!cfrm->data) {
            av_log(f->avctx, AV_LOG_ERROR, "realloc failure\n");
-            return AVERROR(ENOMEM);
+            return -1;
        }

        memcpy(cfrm->data + cfrm->size, buf + 20, data_size);
@@ -923,31 +914,39 @@ static int decode_frame(AVCodecContext *avctx, void *data,
        frame_size = buf_size - 12;
    }

-    FFSWAP(AVFrame*, f->current_picture, f->last_picture);
+    temp               = f->current_picture;
+    f->current_picture = f->last_picture;
+    f->last_picture    = temp;
+
+    p                  = &f->current_picture;
+    avctx->coded_frame = p;

    // alternatively we would have to use our own buffer management
    avctx->flags |= CODEC_FLAG_EMU_EDGE;

-    if ((ret = ff_reget_buffer(avctx, f->current_picture)) < 0)
-        return ret;
+    p->reference= 3;
+    if (avctx->reget_buffer(avctx, p) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
+        return -1;
+    }

    if (frame_4cc == AV_RL32("ifr2")) {
-        f->current_picture->pict_type = AV_PICTURE_TYPE_I;
-        if ((ret = decode_i2_frame(f, f->current_picture, buf - 4, frame_size + 4)) < 0) {
+        p->pict_type= AV_PICTURE_TYPE_I;
+        if (decode_i2_frame(f, buf - 4, frame_size + 4) < 0) {
            av_log(f->avctx, AV_LOG_ERROR, "decode i2 frame failed\n");
-            return ret;
+            return -1;
        }
    } else if (frame_4cc == AV_RL32("ifrm")) {
-        f->current_picture->pict_type = AV_PICTURE_TYPE_I;
-        if ((ret = decode_i_frame(f, f->current_picture, buf, frame_size)) < 0) {
+        p->pict_type= AV_PICTURE_TYPE_I;
+        if (decode_i_frame(f, buf, frame_size) < 0) {
            av_log(f->avctx, AV_LOG_ERROR, "decode i frame failed\n");
-            return ret;
+            return -1;
        }
    } else if (frame_4cc == AV_RL32("pfrm") || frame_4cc == AV_RL32("pfr2")) {
-        f->current_picture->pict_type = AV_PICTURE_TYPE_P;
-        if ((ret = decode_p_frame(f, f->current_picture, buf, frame_size)) < 0) {
+        p->pict_type = AV_PICTURE_TYPE_P;
+        if (decode_p_frame(f, buf, frame_size) < 0) {
            av_log(f->avctx, AV_LOG_ERROR, "decode p frame failed\n");
-            return ret;
+            return -1;
        }
    } else if (frame_4cc == AV_RL32("snd_")) {
        av_log(avctx, AV_LOG_ERROR, "ignoring snd_ chunk length:%d\n",
@@ -957,10 +956,9 @@ static int decode_frame(AVCodecContext *avctx, void *data,
               buf_size);
    }

-    f->current_picture->key_frame = f->current_picture->pict_type == AV_PICTURE_TYPE_I;
+    p->key_frame = p->pict_type == AV_PICTURE_TYPE_I;

-    if ((ret = av_frame_ref(picture, f->current_picture)) < 0)
-        return ret;
+    *picture   = *p;
    *got_frame = 1;

    emms_c();
@@ -968,6 +966,16 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    return buf_size;
 }

+
+static av_cold void common_init(AVCodecContext *avctx)
+{
+    FourXContext * const f = avctx->priv_data;
+
+    ff_dsputil_init(&f->dsp, avctx);
+
+    f->avctx = avctx;
+}
+
 static av_cold int decode_init(AVCodecContext *avctx)
 {
    FourXContext * const f = avctx->priv_data;
@@ -981,9 +989,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
        return AVERROR_INVALIDDATA;
    }

+    avcodec_get_frame_defaults(&f->current_picture);
+    avcodec_get_frame_defaults(&f->last_picture);
    f->version = AV_RL32(avctx->extradata) >> 16;
-    ff_dsputil_init(&f->dsp, avctx);
-    f->avctx = avctx;
+    common_init(avctx);
    init_vlcs(f);

    if (f->version > 2)
@@ -991,11 +1000,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
    else
        avctx->pix_fmt = AV_PIX_FMT_BGR555;

-    f->current_picture = av_frame_alloc();
-    f->last_picture    = av_frame_alloc();
-    if (!f->current_picture  || !f->last_picture)
-        return AVERROR(ENOMEM);
-
    return 0;
 }

@@ -1012,15 +1016,16 @@ static av_cold int decode_end(AVCodecContext *avctx)
        f->cfrm[i].allocated_size = 0;
    }
    ff_free_vlc(&f->pre_vlc);
-    av_frame_free(&f->current_picture);
-    av_frame_free(&f->last_picture);
+    if (f->current_picture.data[0])
+        avctx->release_buffer(avctx, &f->current_picture);
+    if (f->last_picture.data[0])
+        avctx->release_buffer(avctx, &f->last_picture);

    return 0;
 }

 AVCodec ff_fourxm_decoder = {
    .name           = "4xm",
-    .long_name      = NULL_IF_CONFIG_SMALL("4X Movie"),
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_4XM,
    .priv_data_size = sizeof(FourXContext),
@@ -1028,4 +1033,5 @@ AVCodec ff_fourxm_decoder = {
    .close          = decode_end,
    .decode         = decode_frame,
    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("4X Movie"),
 };
--- a/libavcodec/8bps.c
+++ b/libavcodec/8bps.c
@@ -46,6 +46,7 @@ static const enum AVPixelFormat pixfmt_rgb24[] = {

 typedef struct EightBpsContext {
    AVCodecContext *avctx;
+    AVFrame pic;

    unsigned char planes;
    unsigned char planemap[4];
@@ -56,7 +57,6 @@ typedef struct EightBpsContext {
 static int decode_frame(AVCodecContext *avctx, void *data,
                        int *got_frame, AVPacket *avpkt)
 {
-    AVFrame *frame = data;
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    EightBpsContext * const c = avctx->priv_data;
@@ -68,10 +68,16 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    unsigned char count;
    unsigned int planes     = c->planes;
    unsigned char *planemap = c->planemap;
-    int ret;

-    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
-        return ret;
+    if (c->pic.data[0])
+        avctx->release_buffer(avctx, &c->pic);
+
+    c->pic.reference    = 0;
+    c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
+    if (ff_get_buffer(avctx, &c->pic) < 0){
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return -1;
+    }

    ep = encoded + buf_size;

@@ -84,29 +90,29 @@ static int decode_frame(AVCodecContext *avctx, void *data,

        /* Decode a plane */
        for (row = 0; row < height; row++) {
-            pixptr = frame->data[0] + row * frame->linesize[0] + planemap[p];
-            pixptr_end = pixptr + frame->linesize[0];
+            pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
+            pixptr_end = pixptr + c->pic.linesize[0];
            if (ep - lp < row * 2 + 2)
                return AVERROR_INVALIDDATA;
            dlen = av_be2ne16(*(const unsigned short *)(lp + row * 2));
            /* Decode a row of this plane */
            while (dlen > 0) {
                if (ep - dp <= 1)
-                    return AVERROR_INVALIDDATA;
+                    return -1;
                if ((count = *dp++) <= 127) {
                    count++;
                    dlen -= count + 1;
-                    if (pixptr_end - pixptr < count * planes)
+                    if (pixptr + count * planes > pixptr_end)
                        break;
                    if (ep - dp < count)
-                        return AVERROR_INVALIDDATA;
+                        return -1;
                    while (count--) {
                        *pixptr = *dp++;
                        pixptr += planes;
                    }
                } else {
                    count = 257 - count;
-                    if (pixptr_end - pixptr < count * planes)
+                    if (pixptr + count * planes > pixptr_end)
                        break;
                    while (count--) {
                        *pixptr = *dp;
@@ -124,14 +130,15 @@ static int decode_frame(AVCodecContext *avctx, void *data,
                                                     AV_PKT_DATA_PALETTE,
                                                     NULL);
        if (pal) {
-            frame->palette_has_changed = 1;
+            c->pic.palette_has_changed = 1;
            memcpy(c->pal, pal, AVPALETTE_SIZE);
        }

-        memcpy (frame->data[1], c->pal, AVPALETTE_SIZE);
+        memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
    }

    *got_frame = 1;
+    *(AVFrame*)data = c->pic;

    /* always report that the buffer was completely consumed */
    return buf_size;
@@ -142,7 +149,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
    EightBpsContext * const c = avctx->priv_data;

    c->avctx       = avctx;
+    c->pic.data[0] = NULL;

+    avcodec_get_frame_defaults(&c->pic);
    switch (avctx->bits_per_coded_sample) {
    case 8:
        avctx->pix_fmt = AV_PIX_FMT_PAL8;
@@ -159,30 +168,45 @@ static av_cold int decode_init(AVCodecContext *avctx)
    case 32:
        avctx->pix_fmt = AV_PIX_FMT_RGB32;
        c->planes      = 4;
-        /* handle planemap setup later for decoding rgb24 data as rbg32 */
+#if HAVE_BIGENDIAN
+        c->planemap[0] = 1; // 1st plane is red
+        c->planemap[1] = 2; // 2nd plane is green
+        c->planemap[2] = 3; // 3rd plane is blue
+        c->planemap[3] = 0; // 4th plane is alpha
+#else
+        c->planemap[0] = 2; // 1st plane is red
+        c->planemap[1] = 1; // 2nd plane is green
+        c->planemap[2] = 0; // 3rd plane is blue
+        c->planemap[3] = 3; // 4th plane is alpha
+#endif
        break;
    default:
        av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n",
               avctx->bits_per_coded_sample);
-        return AVERROR_INVALIDDATA;
+        return -1;
    }

-    if (avctx->pix_fmt == AV_PIX_FMT_RGB32) {
-        c->planemap[0] = HAVE_BIGENDIAN ? 1 : 2; // 1st plane is red
-        c->planemap[1] = HAVE_BIGENDIAN ? 2 : 1; // 2nd plane is green
-        c->planemap[2] = HAVE_BIGENDIAN ? 3 : 0; // 3rd plane is blue
-        c->planemap[3] = HAVE_BIGENDIAN ? 0 : 3; // 4th plane is alpha
-    }
+    return 0;
+}
+
+static av_cold int decode_end(AVCodecContext *avctx)
+{
+    EightBpsContext * const c = avctx->priv_data;
+
+    if (c->pic.data[0])
+        avctx->release_buffer(avctx, &c->pic);
+
    return 0;
 }

 AVCodec ff_eightbps_decoder = {
    .name           = "8bps",
-    .long_name      = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_8BPS,
    .priv_data_size = sizeof(EightBpsContext),
    .init           = decode_init,
+    .close          = decode_end,
    .decode         = decode_frame,
    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
 };
--- a/libavcodec/8svx.c
+++ b/libavcodec/8svx.c
@@ -44,6 +44,7 @@

 /** decoder context */
 typedef struct EightSvxContext {
+    AVFrame frame;
    uint8_t fib_acc[2];
    const int8_t *table;

@@ -87,7 +88,6 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
                                 int *got_frame_ptr, AVPacket *avpkt)
 {
    EightSvxContext *esc = avctx->priv_data;
-    AVFrame *frame       = data;
    int buf_size;
    int ch, ret;
    int hdr_size = 2;
@@ -135,18 +135,21 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
    }

    /* get output buffer */
-    frame->nb_samples = buf_size * 2;
-    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+    esc->frame.nb_samples = buf_size * 2;
+    if ((ret = ff_get_buffer(avctx, &esc->frame)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
+    }

    for (ch = 0; ch < avctx->channels; ch++) {
-        delta_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
+        delta_decode(esc->frame.data[ch], &esc->data[ch][esc->data_idx],
                     buf_size, &esc->fib_acc[ch], esc->table);
    }

    esc->data_idx += buf_size;

-    *got_frame_ptr = 1;
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = esc->frame;

    return ((avctx->frame_number == 0)*hdr_size + buf_size)*avctx->channels;
 }
@@ -169,6 +172,9 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
    }
    avctx->sample_fmt = AV_SAMPLE_FMT_U8P;

+    avcodec_get_frame_defaults(&esc->frame);
+    avctx->coded_frame = &esc->frame;
+
    return 0;
 }

@@ -187,7 +193,6 @@ static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
 #if CONFIG_EIGHTSVX_FIB_DECODER
 AVCodec ff_eightsvx_fib_decoder = {
  .name           = "8svx_fib",
-  .long_name      = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
  .type           = AVMEDIA_TYPE_AUDIO,
  .id             = AV_CODEC_ID_8SVX_FIB,
  .priv_data_size = sizeof (EightSvxContext),
@@ -195,6 +200,7 @@ AVCodec ff_eightsvx_fib_decoder = {
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
  .capabilities   = CODEC_CAP_DR1,
+  .long_name      = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
  .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                    AV_SAMPLE_FMT_NONE },
 };
@@ -202,7 +208,6 @@ AVCodec ff_eightsvx_fib_decoder = {
 #if CONFIG_EIGHTSVX_EXP_DECODER
 AVCodec ff_eightsvx_exp_decoder = {
  .name           = "8svx_exp",
-  .long_name      = NULL_IF_CONFIG_SMALL("8SVX exponential"),
  .type           = AVMEDIA_TYPE_AUDIO,
  .id             = AV_CODEC_ID_8SVX_EXP,
  .priv_data_size = sizeof (EightSvxContext),
@@ -210,6 +215,7 @@ AVCodec ff_eightsvx_exp_decoder = {
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
  .capabilities   = CODEC_CAP_DR1,
+  .long_name      = NULL_IF_CONFIG_SMALL("8SVX exponential"),
  .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                    AV_SAMPLE_FMT_NONE },
 };
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -20,8 +20,11 @@ OBJS = allcodecs.o                                                      \
       bitstream.o                                                      \
       bitstream_filter.o                                               \
       codec_desc.o                                                     \
+       dsputil.o                                                        \
+       faanidct.o                                                       \
       fmtconvert.o                                                     \
       imgconvert.o                                                     \
+       jrevdct.o                                                        \
       mathtables.o                                                     \
       options.o                                                        \
       parser.o                                                         \
@@ -29,37 +32,30 @@ OBJS = allcodecs.o                                                      \
       rawdec.o                                                         \
       resample.o                                                       \
       resample2.o                                                      \
+       simple_idct.o                                                    \
       utils.o                                                          \

-OBJS-$(HAVE_MSVCRT) += file_open.o
-
 # parts needed for many different codecs
 OBJS-$(CONFIG_AANDCTTABLES)            += aandcttab.o
 OBJS-$(CONFIG_AC3DSP)                  += ac3dsp.o
 OBJS-$(CONFIG_AUDIO_FRAME_QUEUE)       += audio_frame_queue.o
 OBJS-$(CONFIG_CRYSTALHD)               += crystalhd.o
 OBJS-$(CONFIG_DCT)                     += dct.o dct32_fixed.o dct32_float.o
+OBJS-$(CONFIG_DWT)                     += dwt.o snow.o
 OBJS-$(CONFIG_DXVA2)                   += dxva2.o
-OBJS-$(CONFIG_DSPUTIL)                 += dsputil.o faanidct.o          \
-                                          simple_idct.o jrevdct.o
 OBJS-$(CONFIG_ENCODERS)                += faandct.o jfdctfst.o jfdctint.o
 OBJS-$(CONFIG_ERROR_RESILIENCE)        += error_resilience.o
-OBJS-$(CONFIG_EXIF)                    += exif.o tiff_common.o
 FFT-OBJS-$(CONFIG_HARDCODED_TABLES)    += cos_tables.o cos_fixed_tables.o
 OBJS-$(CONFIG_FFT)                     += avfft.o fft_fixed.o fft_float.o \
-                                          fft_fixed_32.o fft_init_table.o \
                                          $(FFT-OBJS-yes)
 OBJS-$(CONFIG_GOLOMB)                  += golomb.o
-OBJS-$(CONFIG_H264CHROMA)              += h264chroma.o
 OBJS-$(CONFIG_H264DSP)                 += h264dsp.o h264idct.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
-OBJS-$(CONFIG_H264QPEL)                += h264qpel.o
-OBJS-$(CONFIG_HPELDSP)                 += hpeldsp.o
 OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
 OBJS-$(CONFIG_LSP)                     += lsp.o
-OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o mdct_fixed_32.o
+OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o
 OBJS-$(CONFIG_MPEGAUDIO)               += mpegaudio.o mpegaudiodata.o   \
                                          mpegaudiodecheader.o
 OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
@@ -79,7 +75,7 @@ OBJS-$(CONFIG_VDPAU)                   += vdpau.o
 OBJS-$(CONFIG_VIDEODSP)                += videodsp.o
 OBJS-$(CONFIG_VP3DSP)                  += vp3dsp.o

-# decoders/encoders
+# decoders/encoders/hardware accelerators
 OBJS-$(CONFIG_ZERO12V_DECODER)         += 012v.o
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
@@ -95,7 +91,6 @@ OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
                                          ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
-OBJS-$(CONFIG_AIC_DECODER)             += aic.o
 OBJS-$(CONFIG_ALAC_DECODER)            += alac.o alac_data.o
 OBJS-$(CONFIG_ALAC_ENCODER)            += alacenc.o alac_data.o
 OBJS-$(CONFIG_ALS_DECODER)             += alsdec.o bgmc.o mpeg4audio.o
@@ -115,8 +110,6 @@ OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpeg.o           \
 OBJS-$(CONFIG_ANM_DECODER)             += anm.o
 OBJS-$(CONFIG_ANSI_DECODER)            += ansi.o cga_data.o
 OBJS-$(CONFIG_APE_DECODER)             += apedec.o
-OBJS-$(CONFIG_SSA_DECODER)             += assdec.o ass.o ass_split.o
-OBJS-$(CONFIG_SSA_ENCODER)             += assenc.o ass.o
 OBJS-$(CONFIG_ASS_DECODER)             += assdec.o ass.o ass_split.o
 OBJS-$(CONFIG_ASS_ENCODER)             += assenc.o ass.o
 OBJS-$(CONFIG_ASV1_DECODER)            += asvdec.o asv.o mpeg12data.o
@@ -162,10 +155,10 @@ OBJS-$(CONFIG_CPIA_DECODER)            += cpia.o
 OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
 OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
-                                          synth_filter.o
+                                          dca_parser.o synth_filter.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o dca.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
-                                          dirac_arith.o mpeg12data.o dirac_dwt.o
+                                          dirac_arith.o mpeg12data.o dwt.o
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
 OBJS-$(CONFIG_DNXHD_DECODER)           += dnxhddec.o dnxhddata.o
 OBJS-$(CONFIG_DNXHD_ENCODER)           += dnxhdenc.o dnxhddata.o
@@ -188,14 +181,13 @@ OBJS-$(CONFIG_EAMAD_DECODER)           += eamad.o eaidct.o mpeg12.o \
                                          mpeg12data.o
 OBJS-$(CONFIG_EATGQ_DECODER)           += eatgq.o eaidct.o
 OBJS-$(CONFIG_EATGV_DECODER)           += eatgv.o
-OBJS-$(CONFIG_EATQI_DECODER)           += eatqi.o eaidct.o mpeg12dec.o  \
-                                          mpeg12.o mpeg12data.o
+OBJS-$(CONFIG_EATQI_DECODER)           += eatqi.o eaidct.o mpeg12.o \
+                                          mpeg12data.o
 OBJS-$(CONFIG_EIGHTBPS_DECODER)        += 8bps.o
 OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER)    += 8svx.o
 OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER)    += 8svx.o
 OBJS-$(CONFIG_ESCAPE124_DECODER)       += escape124.o
 OBJS-$(CONFIG_ESCAPE130_DECODER)       += escape130.o
-OBJS-$(CONFIG_EVRC_DECODER)            += evrcdec.o acelp_vectors.o lsp.o
 OBJS-$(CONFIG_EXR_DECODER)             += exr.o
 OBJS-$(CONFIG_FFV1_DECODER)            += ffv1dec.o ffv1.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1enc.o ffv1.o
@@ -212,7 +204,6 @@ OBJS-$(CONFIG_FLIC_DECODER)            += flicvideo.o
 OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
-OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o mjpeg.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
                                          celp_filters.o celp_math.o
 OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
@@ -221,22 +212,22 @@ OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o
 OBJS-$(CONFIG_GIF_ENCODER)             += gif.o lzwenc.o
 OBJS-$(CONFIG_GSM_DECODER)             += gsmdec.o gsmdec_data.o msgsmdec.o
 OBJS-$(CONFIG_GSM_MS_DECODER)          += gsmdec.o gsmdec_data.o msgsmdec.o
-OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261data.o h261.o
-OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261data.o h261.o
+OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o h261data.o
+OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o h261data.o
 OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o ituh263dec.o        \
                                          mpeg4video.o mpeg4videodec.o flvdec.o\
                                          intelh263dec.o
+OBJS-$(CONFIG_H263_VAAPI_HWACCEL)      += vaapi_mpeg4.o
 OBJS-$(CONFIG_H263_ENCODER)            += mpeg4videoenc.o mpeg4video.o  \
                                          h263.o ituh263enc.o flvenc.o
 OBJS-$(CONFIG_H264_DECODER)            += h264.o                               \
                                          h264_loopfilter.o h264_direct.o      \
                                          cabac.o h264_sei.o h264_ps.o         \
                                          h264_refs.o h264_cavlc.o h264_cabac.o
+OBJS-$(CONFIG_H264_DXVA2_HWACCEL)      += dxva2_h264.o
+OBJS-$(CONFIG_H264_VAAPI_HWACCEL)      += vaapi_h264.o
+OBJS-$(CONFIG_H264_VDA_HWACCEL)        += vda_h264.o
 OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
-OBJS-$(CONFIG_HEVC_DECODER)            += hevc.o hevc_mvs.o hevc_ps.o hevc_sei.o \
-                                          hevc_cabac.o hevc_refs.o hevcpred.o    \
-                                          hevcdsp.o hevc_filter.o cabac.o
-
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IAC_DECODER)             += imc.o
@@ -252,10 +243,8 @@ OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi_common.o ivi_dsp.o
 OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
 OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
 OBJS-$(CONFIG_JACOSUB_DECODER)         += jacosubdec.o ass.o
-OBJS-$(CONFIG_JPEG2000_ENCODER)        += j2kenc.o mqcenc.o mqc.o jpeg2000.o \
-                                          jpeg2000dwt.o
-OBJS-$(CONFIG_JPEG2000_DECODER)        += jpeg2000dec.o jpeg2000.o      \
-                                          jpeg2000dwt.o mqcdec.o mqc.o
+OBJS-$(CONFIG_JPEG2000_DECODER)        += j2kdec.o mqcdec.o mqc.o j2k.o j2k_dwt.o
+OBJS-$(CONFIG_JPEG2000_ENCODER)        += j2kenc.o mqcenc.o mqc.o j2k.o j2k_dwt.o
 OBJS-$(CONFIG_JPEGLS_DECODER)          += jpeglsdec.o jpegls.o \
                                          mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_JPEGLS_ENCODER)          += jpeglsenc.o jpegls.o
@@ -268,8 +257,6 @@ OBJS-$(CONFIG_LOCO_DECODER)            += loco.o
 OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
 OBJS-$(CONFIG_MDEC_DECODER)            += mdec.o mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_METASOUND_DECODER)       += metasound.o metasound_data.o \
-                                          twinvq.o
 OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
@@ -297,23 +284,27 @@ OBJS-$(CONFIG_MPC8_DECODER)            += mpc8.o mpc.o
 OBJS-$(CONFIG_MPEGVIDEO_DECODER)       += mpeg12.o mpeg12data.o \
                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MPEG_XVMC_DECODER)       += mpegvideo_xvmc.o
-OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
-OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
+OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12.o mpeg12data.o
+OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
+                                          timecode.o
+OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)     += dxva2_mpeg2.o
+OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)     += vaapi_mpeg2.o
+OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
                                          timecode.o
+OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)     += vaapi_mpeg4.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
-OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o \
-                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263.o
-OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o \
+                                          h263dec.o h263.o ituh263dec.o        \
                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263.o
+                                          h263dec.o h263.o ituh263dec.o        \
+                                          mpeg4videodec.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
 OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
 OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o mss12.o
@@ -347,11 +338,11 @@ OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o pngdsp.o
 OBJS-$(CONFIG_PNG_ENCODER)             += png.o pngenc.o
 OBJS-$(CONFIG_PPM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
-OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o proresdata.o
+OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o
 OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
 OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_AW_ENCODER)       += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_KS_ENCODER)       += proresenc_kostya.o proresdata.o proresdsp.o
+OBJS-$(CONFIG_PRORES_ANATOLIY_ENCODER) += proresenc_anatoliy.o
+OBJS-$(CONFIG_PRORES_KOSTYA_ENCODER)   += proresenc_kostya.o proresdata.o proresdsp.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
 OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o                     \
                                          celp_filters.o acelp_vectors.o \
@@ -386,7 +377,6 @@ OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o rv34dsp.o
 OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv34dsp.o rv40dsp.o
 OBJS-$(CONFIG_SAMI_DECODER)            += samidec.o ass.o
 OBJS-$(CONFIG_S302M_DECODER)           += s302m.o
-OBJS-$(CONFIG_S302M_ENCODER)           += s302menc.o
 OBJS-$(CONFIG_SANM_DECODER)            += sanm.o
 OBJS-$(CONFIG_SGI_DECODER)             += sgidec.o
 OBJS-$(CONFIG_SGI_ENCODER)             += sgienc.o rle.o
@@ -399,9 +389,8 @@ OBJS-$(CONFIG_SIPR_DECODER)            += sipr.o acelp_pitch_delay.o \
 OBJS-$(CONFIG_SMACKAUD_DECODER)        += smacker.o
 OBJS-$(CONFIG_SMACKER_DECODER)         += smacker.o
 OBJS-$(CONFIG_SMC_DECODER)             += smc.o
-OBJS-$(CONFIG_SMVJPEG_DECODER)         += smvjpegdec.o
-OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o snow_dwt.o
-OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o snow_dwt.o             \
+OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o
+OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o              \
                                          h263.o ituh263enc.o
 OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
@@ -431,7 +420,7 @@ OBJS-$(CONFIG_TARGA_Y216_DECODER)      += targa_y216dec.o
 OBJS-$(CONFIG_THEORA_DECODER)          += xiph.o
 OBJS-$(CONFIG_THP_DECODER)             += mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
-OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
+OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o
 OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o tiff_data.o
 OBJS-$(CONFIG_TMV_DECODER)             += tmv.o cga_data.o
 OBJS-$(CONFIG_TRUEHD_DECODER)          += mlpdec.o mlpdsp.o
@@ -440,9 +429,8 @@ OBJS-$(CONFIG_TRUEMOTION2_DECODER)     += truemotion2.o
 OBJS-$(CONFIG_TRUESPEECH_DECODER)      += truespeech.o
 OBJS-$(CONFIG_TSCC_DECODER)            += tscc.o msrledec.o
 OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
-OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o
-OBJS-$(CONFIG_TTA_ENCODER)             += ttaenc.o ttadata.o
-OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvqdec.o twinvq.o
+OBJS-$(CONFIG_TTA_DECODER)             += tta.o
+OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvq.o
 OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
 OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
 OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideodec.o utvideo.o
@@ -459,13 +447,16 @@ OBJS-$(CONFIG_V210X_DECODER)           += v210x.o
 OBJS-$(CONFIG_VB_DECODER)              += vb.o
 OBJS-$(CONFIG_VBLE_DECODER)            += vble.o
 OBJS-$(CONFIG_VC1_DECODER)             += vc1dec.o vc1.o vc1data.o vc1dsp.o \
-                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o \
-                                          intrax8.o intrax8dsp.o wmv2dsp.o
+                                          msmpeg4.o msmpeg4data.o           \
+                                          intrax8.o intrax8dsp.o
+OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)       += dxva2_vc1.o
+OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)       += vaapi_vc1.o
 OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
+OBJS-$(CONFIG_VCR1_ENCODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMNC_DECODER)            += vmnc.o
-OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbisdsp.o vorbis.o \
+OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbis.o \
                                          vorbis_data.o xiph.o
 OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbisenc.o vorbis.o \
                                          vorbis_data.o
@@ -475,13 +466,9 @@ OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56dsp.o \
 OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
                                          vp6dsp.o vp56rac.o
 OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
-OBJS-$(CONFIG_VP9_DECODER)             += vp9.o vp9dsp.o vp56rac.o
 OBJS-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
-OBJS-$(CONFIG_WAVPACK_ENCODER)         += wavpackenc.o
-OBJS-$(CONFIG_WEBP_DECODER)            += vp8.o vp8dsp.o vp56rac.o
-OBJS-$(CONFIG_WEBP_DECODER)            += webp.o
 OBJS-$(CONFIG_WEBVTT_DECODER)          += webvttdec.o
 OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
 OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o wma_common.o
@@ -492,12 +479,13 @@ OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
 OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                          celp_filters.o \
                                          acelp_vectors.o acelp_filters.o
-OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4dec.o msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2dsp.o \
-                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o \
+OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o        \
+                                          msmpeg4.o msmpeg4data.o \
                                          intrax8.o intrax8dsp.o
-OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2dsp.o \
-                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o
+OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
+                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o \
+                                          mpeg4videodec.o ituh263dec.o h263dec.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
 OBJS-$(CONFIG_XAN_DPCM_DECODER)        += dpcm.o
@@ -527,8 +515,8 @@ OBJS-$(CONFIG_ZMBV_ENCODER)            += zmbvenc.o
 # (AD)PCM decoders/encoders
 OBJS-$(CONFIG_PCM_ALAW_DECODER)           += pcm.o
 OBJS-$(CONFIG_PCM_ALAW_ENCODER)           += pcm.o
-OBJS-$(CONFIG_PCM_BLURAY_DECODER)         += pcm-bluray.o
-OBJS-$(CONFIG_PCM_DVD_DECODER)            += pcm-dvd.o
+OBJS-$(CONFIG_PCM_BLURAY_DECODER)         += pcm-mpeg.o
+OBJS-$(CONFIG_PCM_DVD_DECODER)            += pcm.o
 OBJS-$(CONFIG_PCM_F32BE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_F32BE_ENCODER)          += pcm.o
 OBJS-$(CONFIG_PCM_F32LE_DECODER)          += pcm.o
@@ -587,7 +575,6 @@ OBJS-$(CONFIG_ADPCM_ADX_DECODER)          += adxdec.o adx.o
 OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o adx.o
 OBJS-$(CONFIG_ADPCM_AFC_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_CT_DECODER)           += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_DTK_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_DECODER)           += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER)  += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_R1_DECODER)        += adpcm.o adpcm_data.o
@@ -608,7 +595,6 @@ OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_OKI_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER)       += adpcmenc.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_RAD_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER)   += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER)      += adpcmenc.o adpcm_data.o
@@ -626,23 +612,6 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o

-# hardware accelerators
-OBJS-$(CONFIG_H263_VAAPI_HWACCEL)         += vaapi_mpeg4.o vaapi_mpeg.o
-OBJS-$(CONFIG_H263_VDPAU_HWACCEL)         += vdpau_mpeg4.o
-OBJS-$(CONFIG_H264_DXVA2_HWACCEL)         += dxva2_h264.o
-OBJS-$(CONFIG_H264_VAAPI_HWACCEL)         += vaapi_h264.o
-OBJS-$(CONFIG_H264_VDA_HWACCEL)           += vda_h264.o
-OBJS-$(CONFIG_H264_VDPAU_HWACCEL)         += vdpau_h264.o
-OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)        += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)        += dxva2_mpeg2.o
-OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o vaapi_mpeg.o
-OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)        += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o vaapi_mpeg.o
-OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)        += vdpau_mpeg4.o
-OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)          += dxva2_vc1.o
-OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)          += vaapi_vc1.o vaapi_mpeg.o
-OBJS-$(CONFIG_VC1_VDPAU_HWACCEL)          += vdpau_vc1.o
-
 # libavformat dependencies
 OBJS-$(CONFIG_ADTS_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
@@ -690,7 +659,6 @@ OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
 OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
-OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
 OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
@@ -712,7 +680,6 @@ OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER)    += libschroedingerdec.o \
                                             libschroedinger.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER)    += libschroedingerenc.o \
                                             libschroedinger.o
-OBJS-$(CONFIG_LIBSHINE_ENCODER)           += libshine.o
 OBJS-$(CONFIG_LIBSPEEX_DECODER)           += libspeexdec.o
 OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o
 OBJS-$(CONFIG_LIBSTAGEFRIGHT_H264_DECODER)+= libstagefright.o
@@ -725,15 +692,11 @@ OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER)     += libvo-amrwbenc.o
 OBJS-$(CONFIG_LIBVORBIS_DECODER)          += libvorbisdec.o
 OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbisenc.o \
                                             vorbis_data.o vorbis_parser.o xiph.o
-OBJS-$(CONFIG_LIBVPX_VP8_DECODER)         += libvpxdec.o
-OBJS-$(CONFIG_LIBVPX_VP8_ENCODER)         += libvpxenc.o
-OBJS-$(CONFIG_LIBVPX_VP9_DECODER)         += libvpxdec.o
-OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o
-OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.o
+OBJS-$(CONFIG_LIBVPX_DECODER)             += libvpxdec.o
+OBJS-$(CONFIG_LIBVPX_ENCODER)             += libvpxenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
 OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o
-OBJS-$(CONFIG_LIBZVBI_TELETEXT_DECODER)   += libzvbi-teletextdec.o

 # parsers
 OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o \
@@ -748,9 +711,7 @@ OBJS-$(CONFIG_COOK_PARSER)             += cook_parser.o
 OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o dca.o
 OBJS-$(CONFIG_DIRAC_PARSER)            += dirac_parser.o
 OBJS-$(CONFIG_DNXHD_PARSER)            += dnxhd_parser.o
-OBJS-$(CONFIG_DPX_PARSER)              += dpx_parser.o
 OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
-OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                          vorbis_data.o
@@ -762,7 +723,6 @@ OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o h264.o            \
                                          h264_refs.o h264_sei.o h264_direct.o \
                                          h264_loopfilter.o h264_cabac.o \
                                          h264_cavlc.o h264_ps.o
-OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
@@ -802,11 +762,9 @@ OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF)       += remove_extradata_bsf.o
 OBJS-$(CONFIG_TEXT2MOVSUB_BSF)            += movsub_bsf.o

 # thread libraries
-OBJS-$(HAVE_PTHREADS)                  += pthread.o
-OBJS-$(HAVE_W32THREADS)                += pthread.o
-OBJS-$(HAVE_OS2THREADS)                += pthread.o
-
-OBJS-$(CONFIG_FRAME_THREAD_ENCODER)    += frame_thread_encoder.o
+OBJS-$(HAVE_PTHREADS)                  += pthread.o frame_thread_encoder.o
+OBJS-$(HAVE_W32THREADS)                += pthread.o frame_thread_encoder.o
+OBJS-$(HAVE_OS2THREADS)                += pthread.o frame_thread_encoder.o

 SKIPHEADERS                            += %_tablegen.h                  \
                                          %_tables.h                    \
@@ -822,19 +780,20 @@ SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
 SKIPHEADERS-$(CONFIG_MPEG_XVMC_DECODER) += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
 SKIPHEADERS-$(CONFIG_VDA)              += vda.h
-SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h vdpau_internal.h
+SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h
+SKIPHEADERS-$(HAVE_OS2THREADS)         += os2threads.h
+SKIPHEADERS-$(HAVE_W32THREADS)         += w32pthreads.h

 TESTPROGS = cabac                                                       \
+            dct                                                         \
            fft                                                         \
            fft-fixed                                                   \
-            fft-fixed32                                                 \
            golomb                                                      \
            iirfilter                                                   \
            imgconvert                                                  \
            rangecoder                                                  \
            snowenc                                                     \

-TESTPROGS-$(CONFIG_DCT) += dct
 TESTPROGS-$(HAVE_MMX) += motion
 TESTOBJS = dctref.o

--- a/libavcodec/vaapi_mpeg.c
+++ b/libavcodec/vaapi_mpeg.c
@@ -1,8 +1,6 @@
 /*
- * Video Acceleration API (video decoding)
- * HW decode acceleration for MPEG-2, MPEG-4, H.264 and VC-1
- *
- * Copyright (C) 2013 Anton Khirnov
+ * a64 video encoder - basic headers
+ * Copyright (c) 2009 Tobias Bindhammer
 *
 * This file is part of FFmpeg.
 *
@@ -21,28 +19,40 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

+/**
+ * @file
+ * a64 video encoder - basic headers
+ */
+
+#ifndef AVCODEC_A64ENC_H
+#define AVCODEC_A64ENC_H
+
+#include "libavutil/lfg.h"
 #include "avcodec.h"
-#include "vaapi_internal.h"

-int ff_vaapi_mpeg_end_frame(AVCodecContext *avctx)
-{
-    struct vaapi_context * const vactx = avctx->hwaccel_context;
-    MpegEncContext *s = avctx->priv_data;
-    int ret;
+#define C64XRES 320
+#define C64YRES 200

-    ret = ff_vaapi_commit_slices(vactx);
-    if (ret < 0)
-        goto finish;
+typedef struct A64Context {
+    /* general variables */
+    AVFrame picture;

-    ret = ff_vaapi_render_picture(vactx,
-                                  ff_vaapi_get_surface_id(s->current_picture_ptr));
-    if (ret < 0)
-        goto finish;
+    /* variables for multicolor modes */
+    AVLFG randctx;
+    int mc_lifetime;
+    int mc_use_5col;
+    unsigned mc_frame_counter;
+    int *mc_meta_charset;
+    int *mc_charmap;
+    int *mc_best_cb;
+    int mc_luma_vals[5];
+    uint8_t *mc_charset;
+    uint8_t *mc_colram;
+    uint8_t *mc_palette;
+    int mc_pal_size;

-    ff_mpeg_draw_horiz_band(s, 0, s->avctx->height);
-
-finish:
-    ff_vaapi_common_end_frame(avctx);
-    return ret;
-}
+    /* pts of the next packet that will be output */
+    int64_t next_pts;
+} A64Context;

+#endif /* AVCODEC_A64ENC_H */
--- a/libavcodec/a64multienc.c
+++ b/libavcodec/a64multienc.c
@@ -24,6 +24,7 @@
 * a64 video encoder - multicolor modes
 */

+#include "a64enc.h"
 #include "a64colors.h"
 #include "a64tables.h"
 #include "elbg.h"
@@ -36,31 +37,6 @@
 #define INTERLACED    1
 #define CROP_SCREENS  1

-#define C64XRES 320
-#define C64YRES 200
-
-typedef struct A64Context {
-    /* general variables */
-    AVFrame picture;
-
-    /* variables for multicolor modes */
-    AVLFG randctx;
-    int mc_lifetime;
-    int mc_use_5col;
-    unsigned mc_frame_counter;
-    int *mc_meta_charset;
-    int *mc_charmap;
-    int *mc_best_cb;
-    int mc_luma_vals[5];
-    uint8_t *mc_charset;
-    uint8_t *mc_colram;
-    uint8_t *mc_palette;
-    int mc_pal_size;
-
-    /* pts of the next packet that will be output */
-    int64_t next_pts;
-} A64Context;
-
 /* gray gradient */
 static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};

@@ -396,7 +372,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
 #if CONFIG_A64MULTI_ENCODER
 AVCodec ff_a64multi_encoder = {
    .name           = "a64multi",
-    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_A64_MULTI,
    .priv_data_size = sizeof(A64Context),
@@ -404,13 +379,13 @@ AVCodec ff_a64multi_encoder = {
    .encode2        = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
+    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
    .capabilities   = CODEC_CAP_DELAY,
 };
 #endif
 #if CONFIG_A64MULTI5_ENCODER
 AVCodec ff_a64multi5_encoder = {
    .name           = "a64multi5",
-    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_A64_MULTI5,
    .priv_data_size = sizeof(A64Context),
@@ -418,6 +393,7 @@ AVCodec ff_a64multi5_encoder = {
    .encode2        = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
+    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),
    .capabilities   = CODEC_CAP_DELAY,
 };
 #endif
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -32,6 +32,7 @@

 #include "libavutil/float_dsp.h"
 #include "avcodec.h"
+#include "dsputil.h"
 #include "fft.h"
 #include "mpeg4audio.h"
 #include "sbr.h"
@@ -157,7 +158,7 @@ typedef struct LongTermPrediction {
 typedef struct IndividualChannelStream {
    uint8_t max_sfb;            ///< number of scalefactor bands per group
    enum WindowSequence window_sequence[2];
-    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sine window.
+    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
    int num_window_groups;
    uint8_t group_len[8];
    LongTermPrediction ltp;
@@ -234,7 +235,7 @@ typedef struct SingleChannelElement {
    int sf_idx[128];                                ///< scalefactor indices (used by encoder)
    uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
    DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT
-    DECLARE_ALIGNED(32, float,   saved)[1536];      ///< overlap
+    DECLARE_ALIGNED(32, float,   saved)[1024];      ///< overlap
    DECLARE_ALIGNED(32, float,   ret_buf)[2048];    ///< PCM output buffer
    DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
    PredictorState predictor_state[MAX_PREDICTORS];
@@ -259,10 +260,10 @@ typedef struct ChannelElement {
 /**
 * main AAC context
 */
-struct AACContext {
+typedef struct AACContext {
    AVClass        *class;
    AVCodecContext *avctx;
-    AVFrame *frame;
+    AVFrame frame;

    int is_saved;                 ///< Set if elements have stored overlap from previous frame.
    DynamicRangeControl che_drc;
@@ -290,8 +291,8 @@ struct AACContext {
     */
    FFTContext mdct;
    FFTContext mdct_small;
-    FFTContext mdct_ld;
    FFTContext mdct_ltp;
+    DSPContext dsp;
    FmtConvertContext fmt_conv;
    AVFloatDSPContext fdsp;
    int random_state;
@@ -317,18 +318,6 @@ struct AACContext {

    OutputConfiguration oc[2];
    int warned_num_aac_frames;
-
-    /* aacdec functions pointers */
-    void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
-    void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
-    void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
-                      IndividualChannelStream *ics, int decode);
-    void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
-                                   float *in, IndividualChannelStream *ics);
-    void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
-
-};
-
-void ff_aacdec_init_mips(AACContext *c);
+} AACContext;

 #endif /* AVCODEC_AAC_H */
--- a/libavcodec/aac_adtstoasc_bsf.c
+++ b/libavcodec/aac_adtstoasc_bsf.c
@@ -61,8 +61,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
    }

    if (!hdr.crc_absent && hdr.num_aac_frames > 1) {
-        avpriv_report_missing_feature(avctx,
-                                      "Multiple RDBs per frame with CRC");
+        av_log_missing_feature(avctx, "Multiple RDBs per frame with CRC", 0);
        return AVERROR_PATCHWELCOME;
    }

@@ -75,10 +74,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
        if (!hdr.chan_config) {
            init_get_bits(&gb, buf, buf_size * 8);
            if (get_bits(&gb, 3) != 5) {
-                avpriv_report_missing_feature(avctx,
-                                              "PCE-based channel configuration "
-                                              "without PCE as first syntax "
-                                              "element");
+                av_log_missing_feature(avctx, "PCE based channel configuration, where the PCE is not the first syntax element", 0);
                return AVERROR_PATCHWELCOME;
            }
            init_put_bits(&pb, pce_data, MAX_PCE_SIZE);
@@ -112,7 +108,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
 }

 AVBitStreamFilter ff_aac_adtstoasc_bsf = {
-    .name           = "aac_adtstoasc",
-    .priv_data_size = sizeof(AACBSFContext),
-    .filter         = aac_adtstoasc_filter,
+    "aac_adtstoasc",
+    sizeof(AACBSFContext),
+    aac_adtstoasc_filter,
 };
--- a/libavcodec/aac_tablegen.h
+++ b/libavcodec/aac_tablegen.h
@@ -35,7 +35,7 @@ void ff_aac_tableinit(void)
 {
    int i;
    for (i = 0; i < 428; i++)
-        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
+        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.);
 }
 #endif /* CONFIG_HARDCODED_TABLES */

--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -1113,25 +1113,25 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
 }

 AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
-    [AAC_CODER_FAAC] = {
+    {
        search_for_quantizers_faac,
        encode_window_bands_info,
        quantize_and_encode_band,
        search_for_ms,
    },
-    [AAC_CODER_ANMR] = {
+    {
        search_for_quantizers_anmr,
        encode_window_bands_info,
        quantize_and_encode_band,
        search_for_ms,
    },
-    [AAC_CODER_TWOLOOP] = {
+    {
        search_for_quantizers_twoloop,
        codebook_trellis_rate,
        quantize_and_encode_band,
        search_for_ms,
    },
-    [AAC_CODER_FAST] = {
+    {
        search_for_quantizers_fast,
        encode_window_bands_info,
        quantize_and_encode_band,
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -2,7 +2,6 @@
 * AAC decoder
 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
- * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
 *
 * AAC LATM decoder
 * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
@@ -85,6 +84,7 @@
 #include "avcodec.h"
 #include "internal.h"
 #include "get_bits.h"
+#include "dsputil.h"
 #include "fft.h"
 #include "fmtconvert.h"
 #include "lpc.h"
@@ -108,8 +108,6 @@

 #if ARCH_ARM
 #   include "arm/aac.h"
-#elif ARCH_MIPS
-#   include "mips/aacdec_mips.h"
 #endif

 static VLC vlc_scalefactors;
@@ -193,15 +191,16 @@ static int frame_configure_elements(AVCodecContext *avctx)
    }

    /* get output buffer */
-    av_frame_unref(ac->frame);
-    ac->frame->nb_samples = 2048;
-    if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
+    ac->frame.nb_samples = 2048;
+    if ((ret = ff_get_buffer(avctx, &ac->frame)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
+    }

    /* map output channel pointers to AVFrame data */
    for (ch = 0; ch < avctx->channels; ch++) {
        if (ac->output_element[ch])
-            ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch];
+            ac->output_element[ch]->ret = (float *)ac->frame.extended_data[ch];
    }

    return 0;
@@ -483,8 +482,8 @@ static int output_configure(AACContext *ac,

    memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
    if (layout) avctx->channel_layout = layout;
-                            ac->oc[1].channel_layout = layout;
-    avctx->channels       = ac->oc[1].channels       = channels;
+    ac->oc[1].channel_layout = layout;
+    avctx->channels = ac->oc[1].channels = channels;
    ac->oc[1].status = oc_type;

    if (get_new_frame) {
@@ -746,12 +745,12 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
                                     MPEG4AudioConfig *m4ac,
                                     int channel_config)
 {
-    int extension_flag, ret, ep_config, res_flags;
+    int extension_flag, ret;
    uint8_t layout_map[MAX_ELEM_ID*4][3];
    int tags = 0;

    if (get_bits1(gb)) { // frameLengthFlag
-        avpriv_request_sample(avctx, "960/120 MDCT window");
+        av_log_missing_feature(avctx, "960/120 MDCT window", 1);
        return AVERROR_PATCHWELCOME;
    }

@@ -792,90 +791,14 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
        case AOT_ER_AAC_LTP:
        case AOT_ER_AAC_SCALABLE:
        case AOT_ER_AAC_LD:
-            res_flags = get_bits(gb, 3);
-            if (res_flags) {
-                avpriv_report_missing_feature(avctx,
-                                              "AAC data resilience (flags %x)",
-                                              res_flags);
-                return AVERROR_PATCHWELCOME;
-            }
+            skip_bits(gb, 3);      /* aacSectionDataResilienceFlag
+                                    * aacScalefactorDataResilienceFlag
+                                    * aacSpectralDataResilienceFlag
+                                    */
            break;
        }
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
    }
-    switch (m4ac->object_type) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_SCALABLE:
-    case AOT_ER_AAC_LD:
-        ep_config = get_bits(gb, 2);
-        if (ep_config) {
-            avpriv_report_missing_feature(avctx,
-                                          "epConfig %d", ep_config);
-            return AVERROR_PATCHWELCOME;
-        }
-    }
-    return 0;
-}
-
-static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
-                                     GetBitContext *gb,
-                                     MPEG4AudioConfig *m4ac,
-                                     int channel_config)
-{
-    int ret, ep_config, res_flags;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int tags = 0;
-    const int ELDEXT_TERM = 0;
-
-    m4ac->ps  = 0;
-    m4ac->sbr = 0;
-
-    if (get_bits1(gb)) { // frameLengthFlag
-        avpriv_request_sample(avctx, "960/120 MDCT window");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    res_flags = get_bits(gb, 3);
-    if (res_flags) {
-        avpriv_report_missing_feature(avctx,
-                                      "AAC data resilience (flags %x)",
-                                      res_flags);
-        return AVERROR_PATCHWELCOME;
-    }
-
-    if (get_bits1(gb)) { // ldSbrPresentFlag
-        avpriv_report_missing_feature(avctx,
-                                      "Low Delay SBR");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    while (get_bits(gb, 4) != ELDEXT_TERM) {
-        int len = get_bits(gb, 4);
-        if (len == 15)
-            len += get_bits(gb, 8);
-        if (len == 15 + 255)
-            len += get_bits(gb, 16);
-        if (get_bits_left(gb) < len * 8 + 4) {
-            av_log(ac->avctx, AV_LOG_ERROR, overread_err);
-            return AVERROR_INVALIDDATA;
-        }
-        skip_bits_long(gb, 8 * len);
-    }
-
-    if ((ret = set_default_channel_config(avctx, layout_map,
-                                          &tags, channel_config)))
-        return ret;
-
-    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
-        return ret;
-
-    ep_config = get_bits(gb, 2);
-    if (ep_config) {
-        avpriv_report_missing_feature(avctx,
-                                      "epConfig %d", ep_config);
-        return AVERROR_PATCHWELCOME;
-    }
    return 0;
 }

@@ -902,7 +825,7 @@ static int decode_audio_specific_config(AACContext *ac,

    av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
    for (i = 0; i < bit_size >> 3; i++)
-        av_dlog(avctx, "%02x ", data[i]);
+         av_dlog(avctx, "%02x ", data[i]);
    av_dlog(avctx, "\n");

    if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
@@ -917,13 +840,6 @@ static int decode_audio_specific_config(AACContext *ac,
               m4ac->sampling_index);
        return AVERROR_INVALIDDATA;
    }
-    if (m4ac->object_type == AOT_ER_AAC_LD &&
-        (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
-        av_log(avctx, AV_LOG_ERROR,
-               "invalid low delay sampling rate index %d\n",
-               m4ac->sampling_index);
-        return AVERROR_INVALIDDATA;
-    }

    skip_bits_long(&gb, i);

@@ -931,22 +847,15 @@ static int decode_audio_specific_config(AACContext *ac,
    case AOT_AAC_MAIN:
    case AOT_AAC_LC:
    case AOT_AAC_LTP:
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LD:
        if ((ret = decode_ga_specific_config(ac, avctx, &gb,
                                            m4ac, m4ac->chan_config)) < 0)
            return ret;
        break;
-    case AOT_ER_AAC_ELD:
-        if ((ret = decode_eld_specific_config(ac, avctx, &gb,
-                                              m4ac, m4ac->chan_config)) < 0)
-            return ret;
-        break;
    default:
-        avpriv_report_missing_feature(avctx,
-                                      "Audio object type %s%d",
-                                      m4ac->sbr == 1 ? "SBR+" : "",
-                                      m4ac->object_type);
+        av_log(avctx, AV_LOG_ERROR,
+               "Audio object type %s%d is not supported.\n",
+               m4ac->sbr == 1 ? "SBR+" : "",
+               m4ac->object_type);
        return AVERROR(ENOSYS);
    }

@@ -1020,8 +929,6 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
                                    sizeof(ff_aac_spectral_codes[num][0]), \
        size);

-static void aacdec_init(AACContext *ac);
-
 static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
    AACContext *ac = avctx->priv_data;
@@ -1030,8 +937,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
    ac->avctx = avctx;
    ac->oc[1].m4ac.sample_rate = avctx->sample_rate;

-    aacdec_init(ac);
-
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

    if (avctx->extradata_size > 0) {
@@ -1089,6 +994,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)

    ff_aac_sbr_init();

+    ff_dsputil_init(&ac->dsp, avctx);
    ff_fmt_convert_init(&ac->fmt_conv, avctx);
    avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

@@ -1107,19 +1013,19 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
                    352);

    ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
-    ff_mdct_init(&ac->mdct_ld,    10, 1, 1.0 / (32768.0 * 512.0));
    ff_mdct_init(&ac->mdct_small,  8, 1, 1.0 / (32768.0 * 128.0));
    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0 * 32768.0);
    // window initialization
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_long_512,  4.0, 512);
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
    ff_init_ff_sine_windows(10);
-    ff_init_ff_sine_windows( 9);
    ff_init_ff_sine_windows( 7);

    cbrt_tableinit();

+    avcodec_get_frame_defaults(&ac->frame);
+    avctx->coded_frame = &ac->frame;
+
    return 0;
 }

@@ -1182,25 +1088,14 @@ static void decode_ltp(LongTermPrediction *ltp,
 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                           GetBitContext *gb)
 {
-    int aot = ac->oc[1].m4ac.object_type;
-    if (aot != AOT_ER_AAC_ELD) {
-        if (get_bits1(gb)) {
-            av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
-            return AVERROR_INVALIDDATA;
-        }
-        ics->window_sequence[1] = ics->window_sequence[0];
-        ics->window_sequence[0] = get_bits(gb, 2);
-        if (aot == AOT_ER_AAC_LD &&
-            ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
-                   "window sequence %d found.\n", ics->window_sequence[0]);
-            ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
-            return AVERROR_INVALIDDATA;
-        }
-        ics->use_kb_window[1]   = ics->use_kb_window[0];
-        ics->use_kb_window[0]   = get_bits1(gb);
+    if (get_bits1(gb)) {
+        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+        return AVERROR_INVALIDDATA;
    }
+    ics->window_sequence[1] = ics->window_sequence[0];
+    ics->window_sequence[0] = get_bits(gb, 2);
+    ics->use_kb_window[1]   = ics->use_kb_window[0];
+    ics->use_kb_window[0]   = get_bits1(gb);
    ics->num_window_groups  = 1;
    ics->group_len[0]       = 1;
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
@@ -1222,36 +1117,21 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
    } else {
        ics->max_sfb               = get_bits(gb, 6);
        ics->num_windows           = 1;
-        if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
-            ics->swb_offset        =     ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
-            ics->num_swb           =    ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
-            if (!ics->num_swb || !ics->swb_offset)
-                return AVERROR_BUG;
-        } else {
-            ics->swb_offset        =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
-            ics->num_swb           =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
-        }
+        ics->swb_offset            =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
+        ics->num_swb               =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
-        if (aot != AOT_ER_AAC_ELD) {
-            ics->predictor_present     = get_bits1(gb);
-            ics->predictor_reset_group = 0;
-        }
+        ics->predictor_present     = get_bits1(gb);
+        ics->predictor_reset_group = 0;
        if (ics->predictor_present) {
-            if (aot == AOT_AAC_MAIN) {
+            if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
                if (decode_prediction(ac, ics, gb)) {
                    goto fail;
                }
-            } else if (aot == AOT_AAC_LC ||
-                       aot == AOT_ER_AAC_LC) {
+            } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {
                av_log(ac->avctx, AV_LOG_ERROR,
                       "Prediction is not allowed in AAC-LC.\n");
                goto fail;
            } else {
-                if (aot == AOT_ER_AAC_LD) {
-                    av_log(ac->avctx, AV_LOG_ERROR,
-                           "LTP in ER AAC LD not yet implemented.\n");
-                    return AVERROR_PATCHWELCOME;
-                }
                if ((ics->ltp.present = get_bits(gb, 1)))
                    decode_ltp(&ics->ltp, gb, ics->max_sfb);
            }
@@ -1344,17 +1224,17 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
            int run_end = band_type_run_end[idx];
            if (band_type[idx] == ZERO_BT) {
                for (; i < run_end; i++, idx++)
-                    sf[idx] = 0.0;
+                    sf[idx] = 0.;
            } else if ((band_type[idx] == INTENSITY_BT) ||
                       (band_type[idx] == INTENSITY_BT2)) {
                for (; i < run_end; i++, idx++) {
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                    clipped_offset = av_clip(offset[2], -155, 100);
                    if (offset[2] != clipped_offset) {
-                        avpriv_request_sample(ac->avctx,
-                                              "If you heard an audible artifact, there may be a bug in the decoder. "
-                                              "Clipped intensity stereo position (%d -> %d)",
-                                              offset[2], clipped_offset);
+                        av_log_ask_for_sample(ac->avctx, "Intensity stereo "
+                                "position clipped (%d -> %d).\nIf you heard an "
+                                "audible artifact, there may be a bug in the "
+                                "decoder. ", offset[2], clipped_offset);
                    }
                    sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
                }
@@ -1366,10 +1246,10 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                    clipped_offset = av_clip(offset[1], -100, 155);
                    if (offset[1] != clipped_offset) {
-                        avpriv_request_sample(ac->avctx,
-                                              "If you heard an audible artifact, there may be a bug in the decoder. "
-                                              "Clipped noise gain (%d -> %d)",
-                                              offset[1], clipped_offset);
+                        av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
+                                "(%d -> %d).\nIf you heard an audible "
+                                "artifact, there may be a bug in the decoder. ",
+                                offset[1], clipped_offset);
                    }
                    sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
                }
@@ -1594,7 +1474,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
                        cfo[k] = ac->random_state;
                    }

-                    band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
+                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
                    scale = sf[idx] / sqrtf(band_energy);
                    ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
                }
@@ -1872,15 +1752,9 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
    TemporalNoiseShaping    *tns = &sce->tns;
    IndividualChannelStream *ics = &sce->ics;
    float *out = sce->coeffs;
-    int global_gain, eld_syntax, er_syntax, pulse_present = 0;
+    int global_gain, pulse_present = 0;
    int ret;

-    eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
-    er_syntax  = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
-
    /* This assignment is to silence a GCC warning about the variable being used
     * uninitialized when in fact it always is.
     */
@@ -1902,7 +1776,7 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,

    pulse_present = 0;
    if (!scale_flag) {
-        if (!eld_syntax && (pulse_present = get_bits1(gb))) {
+        if ((pulse_present = get_bits1(gb))) {
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
                av_log(ac->avctx, AV_LOG_ERROR,
                       "Pulse tool not allowed in eight short sequence.\n");
@@ -1914,19 +1788,12 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
                return AVERROR_INVALIDDATA;
            }
        }
-        tns->present = get_bits1(gb);
-        if (tns->present && !er_syntax)
-            if (decode_tns(ac, tns, gb, ics) < 0)
-                return AVERROR_INVALIDDATA;
-        if (!eld_syntax && get_bits1(gb)) {
-            avpriv_request_sample(ac->avctx, "SSR");
+        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
+            return AVERROR_INVALIDDATA;
+        if (get_bits1(gb)) {
+            av_log_missing_feature(ac->avctx, "SSR", 1);
            return AVERROR_PATCHWELCOME;
        }
-        // I see no textual basis in the spec for this occuring after SSR gain
-        // control, but this is what both reference and real implmentations do
-        if (tns->present && er_syntax)
-            if (decode_tns(ac, tns, gb, ics) < 0)
-                return AVERROR_INVALIDDATA;
    }

    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
@@ -1955,9 +1822,9 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
                cpe->ch[0].band_type[idx] < NOISE_BT &&
                cpe->ch[1].band_type[idx] < NOISE_BT) {
                for (group = 0; group < ics->group_len[g]; group++) {
-                    ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
-                                               ch1 + group * 128 + offsets[i],
-                                               offsets[i+1] - offsets[i]);
+                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
+                                              ch1 + group * 128 + offsets[i],
+                                              offsets[i+1] - offsets[i]);
                }
            }
        }
@@ -2018,9 +1885,8 @@ static void apply_intensity_stereo(AACContext *ac,
 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
 {
    int i, ret, common_window, ms_present = 0;
-    int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;

-    common_window = eld_syntax || get_bits1(gb);
+    common_window = get_bits1(gb);
    if (common_window) {
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
            return AVERROR_INVALIDDATA;
@@ -2102,7 +1968,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
        int idx  = 0;
        int cge  = 1;
        int gain = 0;
-        float gain_cache = 1.0;
+        float gain_cache = 1.;
        if (c) {
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
@@ -2361,9 +2227,9 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,
        ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
    }
    if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
-        ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+        ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
    } else {
-        ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+        ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
        memset(in + 1024 + 576, 0, 448 * sizeof(float));
    }
    ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
@@ -2389,10 +2255,10 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
            predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
        memset(&predTime[i], 0, (2048 - i) * sizeof(float));

-        ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
+        windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);

        if (sce->tns.present)
-            ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
+            apply_tns(predFreq, &sce->tns, &sce->ics, 0);

        for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
            if (ltp->used[sfb])
@@ -2416,17 +2282,17 @@ static void update_ltp(AACContext *ac, SingleChannelElement *sce)
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
        memcpy(saved_ltp,       saved, 512 * sizeof(float));
        memset(saved_ltp + 576, 0,     448 * sizeof(float));
-        ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
+        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
        for (i = 0; i < 64; i++)
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
        memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
        memset(saved_ltp + 576, 0,                  448 * sizeof(float));
-        ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
+        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
        for (i = 0; i < 64; i++)
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
    } else { // LONG_STOP or ONLY_LONG
-        ac->fdsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
+        ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
        for (i = 0; i < 512; i++)
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
    }
@@ -2467,113 +2333,38 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
     */
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
-        ac->fdsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
+        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
    } else {
-        memcpy(                         out,               saved,            448 * sizeof(float));
+        memcpy(                        out,               saved,            448 * sizeof(float));

        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-            ac->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
-            ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
-            ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
-            ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
-            ac->fdsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
-            memcpy(                     out + 448 + 4*128, temp, 64 * sizeof(float));
+            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
+            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
+            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
+            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
+            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
+            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
        } else {
-            ac->fdsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
-            memcpy(                     out + 576,         buf + 64,         448 * sizeof(float));
+            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
+            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
        }
    }

    // buffer update
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-        memcpy(                     saved,       temp + 64,         64 * sizeof(float));
-        ac->fdsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
-        ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
-        ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
-        memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
+        memcpy(                    saved,       temp + 64,         64 * sizeof(float));
+        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
+        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
+        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
+        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
-        memcpy(                     saved,       buf + 512,        448 * sizeof(float));
-        memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
+        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
+        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
    } else { // LONG_STOP or ONLY_LONG
-        memcpy(                     saved,       buf + 512,        512 * sizeof(float));
+        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
    }
 }

-static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
-{
-    IndividualChannelStream *ics = &sce->ics;
-    float *in    = sce->coeffs;
-    float *out   = sce->ret;
-    float *saved = sce->saved;
-    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_512 : ff_sine_512;
-    float *buf  = ac->buf_mdct;
-
-    // imdct
-    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
-
-    // window overlapping
-    ac->fdsp.vector_fmul_window(out, saved, buf, lwindow_prev, 256);
-
-    // buffer update
-    memcpy(saved, buf + 256, 256 * sizeof(float));
-}
-
-static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
-{
-    float *in    = sce->coeffs;
-    float *out   = sce->ret;
-    float *saved = sce->saved;
-    const float *const window = ff_aac_eld_window;
-    float *buf  = ac->buf_mdct;
-    int i;
-    const int n  = 512;
-    const int n2 = n >> 1;
-    const int n4 = n >> 2;
-
-    // Inverse transform, mapped to the conventional IMDCT by
-    // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
-    // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
-    // Audio, Language and Image Processing, 2008. ICALIP 2008. International Conference on
-    // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
-    for (i = 0; i < n2; i+=2) {
-        float temp;
-        temp =  in[i    ]; in[i    ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
-        temp = -in[i + 1]; in[i + 1] =  in[n - 2 - i]; in[n - 2 - i] = temp;
-    }
-    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
-    for (i = 0; i < n; i+=2) {
-        buf[i] = -buf[i];
-    }
-    // Like with the regular IMDCT at this point we still have the middle half
-    // of a transform but with even symmetry on the left and odd symmetry on
-    // the right
-
-    // window overlapping
-    // The spec says to use samples [0..511] but the reference decoder uses
-    // samples [128..639].
-    for (i = n4; i < n2; i ++) {
-        out[i - n4] =    buf[n2 - 1 - i]       * window[i       - n4] +
-                       saved[      i + n2]     * window[i +   n - n4] +
-                      -saved[  n + n2 - 1 - i] * window[i + 2*n - n4] +
-                      -saved[2*n + n2 + i]     * window[i + 3*n - n4];
-    }
-    for (i = 0; i < n2; i ++) {
-        out[n4 + i] =    buf[i]               * window[i + n2       - n4] +
-                      -saved[      n - 1 - i] * window[i + n2 +   n - n4] +
-                      -saved[  n + i]         * window[i + n2 + 2*n - n4] +
-                       saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
-    }
-    for (i = 0; i < n4; i ++) {
-        out[n2 + n4 + i] =    buf[      i + n2]     * window[i +   n - n4] +
-                           -saved[      n2 - 1 - i] * window[i + 2*n - n4] +
-                           -saved[  n + n2 + i]     * window[i + 3*n - n4];
-    }
-
-    // buffer update
-    memmove(saved + n, saved, 2 * n * sizeof(float));
-    memcpy( saved,       buf,     n * sizeof(float));
-}
-
 /**
 * Apply dependent channel coupling (applied before IMDCT).
 *
@@ -2670,17 +2461,6 @@ static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
 static void spectral_to_sample(AACContext *ac)
 {
    int i, type;
-    void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
-    switch (ac->oc[1].m4ac.object_type) {
-    case AOT_ER_AAC_LD:
-        imdct_and_window = imdct_and_windowing_ld;
-        break;
-    case AOT_ER_AAC_ELD:
-        imdct_and_window = imdct_and_windowing_eld;
-        break;
-    default:
-        imdct_and_window = ac->imdct_and_windowing;
-    }
    for (type = 3; type >= 0; type--) {
        for (i = 0; i < MAX_ELEM_ID; i++) {
            ChannelElement *che = ac->che[type][i];
@@ -2690,25 +2470,25 @@ static void spectral_to_sample(AACContext *ac)
                if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
                    if (che->ch[0].ics.predictor_present) {
                        if (che->ch[0].ics.ltp.present)
-                            ac->apply_ltp(ac, &che->ch[0]);
+                            apply_ltp(ac, &che->ch[0]);
                        if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
-                            ac->apply_ltp(ac, &che->ch[1]);
+                            apply_ltp(ac, &che->ch[1]);
                    }
                }
                if (che->ch[0].tns.present)
-                    ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
+                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
                if (che->ch[1].tns.present)
-                    ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
+                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
                if (type <= TYPE_CPE)
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
-                    imdct_and_window(ac, &che->ch[0]);
+                    imdct_and_windowing(ac, &che->ch[0]);
                    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
-                        ac->update_ltp(ac, &che->ch[0]);
+                        update_ltp(ac, &che->ch[0]);
                    if (type == TYPE_CPE) {
-                        imdct_and_window(ac, &che->ch[1]);
+                        imdct_and_windowing(ac, &che->ch[1]);
                        if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
-                            ac->update_ltp(ac, &che->ch[1]);
+                            update_ltp(ac, &che->ch[1]);
                    }
                    if (ac->oc[1].m4ac.sbr > 0) {
                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
@@ -2726,29 +2506,25 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
    int size;
    AACADTSHeaderInfo hdr_info;
    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int layout_map_tags, ret;
+    int layout_map_tags;

    size = avpriv_aac_parse_header(gb, &hdr_info);
    if (size > 0) {
        if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
            // This is 2 for "VLB " audio in NSV files.
            // See samples/nsv/vlb_audio.
-            avpriv_report_missing_feature(ac->avctx,
-                                          "More than one AAC RDB per ADTS frame");
+            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame", 0);
            ac->warned_num_aac_frames = 1;
        }
        push_output_configuration(ac);
        if (hdr_info.chan_config) {
            ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
-            if ((ret = set_default_channel_config(ac->avctx,
-                                                  layout_map,
-                                                  &layout_map_tags,
-                                                  hdr_info.chan_config)) < 0)
-                return ret;
-            if ((ret = output_configure(ac, layout_map, layout_map_tags,
-                                        FFMAX(ac->oc[1].status,
-                                              OC_TRIAL_FRAME), 0)) < 0)
-                return ret;
+            if (set_default_channel_config(ac->avctx, layout_map,
+                    &layout_map_tags, hdr_info.chan_config))
+                return -7;
+            if (output_configure(ac, layout_map, layout_map_tags,
+                                 FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))
+                return -7;
        } else {
            ac->oc[1].m4ac.chan_config = 0;
            /**
@@ -2782,66 +2558,6 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
    return size;
 }

-static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
-                               int *got_frame_ptr, GetBitContext *gb)
-{
-    AACContext *ac = avctx->priv_data;
-    ChannelElement *che;
-    int err, i;
-    int samples = 1024;
-    int chan_config = ac->oc[1].m4ac.chan_config;
-    int aot = ac->oc[1].m4ac.object_type;
-
-    if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
-        samples >>= 1;
-
-    ac->frame = data;
-
-    if ((err = frame_configure_elements(avctx)) < 0)
-        return err;
-
-    ac->tags_mapped = 0;
-
-    if (chan_config < 0 || chan_config >= 8) {
-        avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
-                              ac->oc[1].m4ac.chan_config);
-        return AVERROR_INVALIDDATA;
-    }
-    for (i = 0; i < tags_per_config[chan_config]; i++) {
-        const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
-        const int elem_id   = aac_channel_layout_map[chan_config-1][i][1];
-        if (!(che=get_che(ac, elem_type, elem_id))) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "channel element %d.%d is not allocated\n",
-                   elem_type, elem_id);
-            return AVERROR_INVALIDDATA;
-        }
-        if (aot != AOT_ER_AAC_ELD)
-            skip_bits(gb, 4);
-        switch (elem_type) {
-        case TYPE_SCE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            break;
-        case TYPE_CPE:
-            err = decode_cpe(ac, gb, che);
-            break;
-        case TYPE_LFE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            break;
-        }
-        if (err < 0)
-            return err;
-    }
-
-    spectral_to_sample(ac);
-
-    ac->frame->nb_samples = samples;
-    *got_frame_ptr = 1;
-
-    skip_bits_long(gb, get_bits_left(gb));
-    return 0;
-}
-
 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
                                int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
 {
@@ -2852,22 +2568,23 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
    int samples = 0, multiplier, audio_found = 0, pce_found = 0;
    int is_dmono, sce_count = 0;

-    ac->frame = data;
-
    if (show_bits(gb, 12) == 0xfff) {
-        if ((err = parse_adts_frame_header(ac, gb)) < 0) {
+        if (parse_adts_frame_header(ac, gb) < 0) {
            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+            err = -1;
            goto fail;
        }
        if (ac->oc[1].m4ac.sampling_index > 12) {
            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
-            err = AVERROR_INVALIDDATA;
+            err = -1;
            goto fail;
        }
    }

-    if ((err = frame_configure_elements(avctx)) < 0)
+    if (frame_configure_elements(avctx) < 0) {
+        err = -1;
        goto fail;
+    }

    ac->tags_mapped = 0;
    // parse
@@ -2878,7 +2595,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
            if (!(che=get_che(ac, elem_type, elem_id))) {
                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
                       elem_type, elem_id);
-                err = AVERROR_INVALIDDATA;
+                err = -1;
                goto fail;
            }
            samples = 1024;
@@ -2922,6 +2639,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
            if (pce_found) {
                av_log(avctx, AV_LOG_ERROR,
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
+                pop_output_configuration(ac);
            } else {
                err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
                if (!err)
@@ -2936,7 +2654,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
                elem_id += get_bits(gb, 8) - 1;
            if (get_bits_left(gb) < 8 * elem_id) {
                    av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
-                    err = AVERROR_INVALIDDATA;
+                    err = -1;
                    goto fail;
            }
            while (elem_id > 0)
@@ -2945,7 +2663,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
            break;

        default:
-            err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
+            err = -1; /* should not happen, but keeps compiler happy */
            break;
        }

@@ -2957,7 +2675,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,

        if (get_bits_left(gb) < 3) {
            av_log(avctx, AV_LOG_ERROR, overread_err);
-            err = AVERROR_INVALIDDATA;
+            err = -1;
            goto fail;
        }
    }
@@ -2970,10 +2688,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
    is_dmono = ac->dmono_mode && sce_count == 2 &&
               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);

-    if (samples)
-        ac->frame->nb_samples = samples;
-    else
-        av_frame_unref(ac->frame);
+    if (samples) {
+        ac->frame.nb_samples = samples;
+        *(AVFrame *)data = ac->frame;
+    }
    *got_frame_ptr = !!samples;

    if (is_dmono) {
@@ -2991,7 +2709,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,

    if (multiplier) {
        int side_size;
-        const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
+        uint32_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
        if (side && side_size>=4)
            AV_WL32(side, 2*AV_RL32(side));
    }
@@ -3043,23 +2761,10 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
    if (ac->force_dmono_mode >= 0)
        ac->dmono_mode = ac->force_dmono_mode;

-    if (INT_MAX / 8 <= buf_size)
-        return AVERROR_INVALIDDATA;
-
    if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
        return err;

-    switch (ac->oc[1].m4ac.object_type) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_LD:
-    case AOT_ER_AAC_ELD:
-        err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
-        break;
-    default:
-        err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
-    }
-    if (err < 0)
+    if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt)) < 0)
        return err;

    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@@ -3085,7 +2790,6 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)

    ff_mdct_end(&ac->mdct);
    ff_mdct_end(&ac->mdct_small);
-    ff_mdct_end(&ac->mdct_ld);
    ff_mdct_end(&ac->mdct_ltp);
    return 0;
 }
@@ -3095,7 +2799,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)

 struct LATMContext {
    AACContext aac_ctx;     ///< containing AACContext
-    int initialized;        ///< initialized after a valid extradata was seen
+    int initialized;        ///< initilized after a valid extradata was seen

    // parser data
    int audio_mux_version_A; ///< LATM syntax version
@@ -3127,8 +2831,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
        asclen         = get_bits_left(gb);

    if (config_start_bit % 8) {
-        avpriv_request_sample(latmctx->aac_ctx.avctx,
-                              "Non-byte-aligned audio-specific config");
+        av_log_missing_feature(latmctx->aac_ctx.avctx,
+                               "Non-byte-aligned audio-specific config", 1);
        return AVERROR_PATCHWELCOME;
    }
    if (asclen <= 0)
@@ -3147,7 +2851,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
        if(latmctx->initialized) {
            av_log(avctx, AV_LOG_INFO, "audio config changed\n");
        } else {
-            av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
+            av_log(avctx, AV_LOG_INFO, "initializing latmctx\n");
        }
        latmctx->initialized = 0;

@@ -3187,7 +2891,8 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
        skip_bits(gb, 6);                       // numSubFrames
        // numPrograms
        if (get_bits(gb, 4)) {                  // numPrograms
-            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
+            av_log_missing_feature(latmctx->aac_ctx.avctx,
+                                   "Multiple programs", 1);
            return AVERROR_PATCHWELCOME;
        }

@@ -3195,7 +2900,8 @@ static int read_stream_mux_config(struct LATMContext *latmctx,

        // for each layer (which there is only one in DVB)
        if (get_bits(gb, 3)) {                   // numLayer
-            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
+            av_log_missing_feature(latmctx->aac_ctx.avctx,
+                                   "Multiple layers", 1);
            return AVERROR_PATCHWELCOME;
        }

@@ -3306,7 +3012,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
    int                 muxlength, err;
    GetBitContext       gb;

-    if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
+    if ((err = init_get_bits(&gb, avpkt->data, avpkt->size * 8)) < 0)
        return err;

    // check for LOAS sync word
@@ -3361,17 +3067,6 @@ static av_cold int latm_decode_init(AVCodecContext *avctx)
    return ret;
 }

-static void aacdec_init(AACContext *c)
-{
-    c->imdct_and_windowing                      = imdct_and_windowing;
-    c->apply_ltp                                = apply_ltp;
-    c->apply_tns                                = apply_tns;
-    c->windowing_and_mdct_ltp                   = windowing_and_mdct_ltp;
-    c->update_ltp                               = update_ltp;
-
-    if(ARCH_MIPS)
-        ff_aacdec_init_mips(c);
-}
 /**
 * AVOptions for Japanese DTV specific extensions (ADTS only)
 */
@@ -3398,13 +3093,13 @@ static const AVClass aac_decoder_class = {

 AVCodec ff_aac_decoder = {
    .name            = "aac",
-    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
    .type            = AVMEDIA_TYPE_AUDIO,
    .id              = AV_CODEC_ID_AAC,
    .priv_data_size  = sizeof(AACContext),
    .init            = aac_decode_init,
    .close           = aac_decode_close,
    .decode          = aac_decode_frame,
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
    .sample_fmts     = (const enum AVSampleFormat[]) {
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
    },
@@ -3421,13 +3116,13 @@ AVCodec ff_aac_decoder = {
 */
 AVCodec ff_aac_latm_decoder = {
    .name            = "aac_latm",
-    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
    .type            = AVMEDIA_TYPE_AUDIO,
    .id              = AV_CODEC_ID_AAC_LATM,
    .priv_data_size  = sizeof(struct LATMContext),
    .init            = latm_decode_init,
    .close           = aac_decode_close,
    .decode          = latm_decode_frame,
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
    .sample_fmts     = (const enum AVSampleFormat[]) {
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
    },
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .1.1
 .1.7