update for 0.9.4

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
vorbisdec: Check bark_map_size.
2025-12-06 14:59:59 +01:00 · 2014-03-21 05:53:28 +01:00 · 2014-03-21 04:23:59 +01:00 · 2014-03-21 04:23:59 +01:00 · 2014-03-21 04:23:59 +01:00 · 2014-03-21 04:23:59 +01:00
2162 changed files with 109449 additions and 156973 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -1,17 +1,9 @@
 .config
 .version
-*.a
 *.o
 *.d
-*.def
-*.dll
 *.exe
 *.ho
-*.lib
-*.pc
-*.so
-*.so.*
-*.ver
 *-example
 *-test
 *_g
@@ -23,26 +15,26 @@ config.*
 doc/*.1
 doc/*.html
 doc/*.pod
-doc/fate.txt
 doxy
 ffmpeg
 ffplay
 ffprobe
 ffserver
 avconv
-doc/avoptions_codec.texi
-doc/avoptions_format.texi
-doc/print_options
-doc/examples/decoding_encoding
-doc/examples/filtering_audio
-doc/examples/filtering_video
-doc/examples/metadata
-doc/examples/muxing
 libavcodec/*_tablegen
 libavcodec/*_tables.c
 libavcodec/*_tables.h
 libavcodec/codec_names.h
+libavcodec/libavcodec*
+libavcore/libavcore*
+libavdevice/libavdevice*
+libavfilter/libavfilter*
+libavformat/libavformat*
 libavutil/avconfig.h
+libavutil/libavutil*
+libpostproc/libpostproc*
+libswresample/libswresample*
+libswscale/libswscale*
 tests/audiogen
 tests/base64
 tests/data
@@ -51,11 +43,8 @@ tests/tiny_psnr
 tests/videogen
 tests/vsynth1
 tests/vsynth2
-tools/aviocat
 tools/cws2fws
-tools/ffeval
 tools/graph2dot
-tools/ismindex
 tools/lavfi-showfiltfmts
 tools/pktdumper
 tools/probetest
--- a/COPYING.LGPLv2.1
+++ b/COPYING.LGPLv2.1
@@ -500,3 +500,5 @@ necessary.  Here is a sample; alter the names:
  Ty Coon, President of Vice

 That's all there is to it!
+
+
--- a/102
+++ b/102
@@ -1,97 +1,11 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

-version next:
-
-version 0.11.2:
-
- Several bugs and crashes have been fixed as well as build problems
-  with recent mingw64
-
-
-version 0.11:
-Fixes:CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
-      CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
-      CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
-      CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
-      CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
-      CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
- setfield filter
- CDXL demuxer and decoder
- Apple ProRes encoder
- ffprobe -count_packets and -count_frames options
- Sun Rasterfile Encoder
- ID3v2 attached pictures reading and writing
- WMA Lossless decoder
- bluray protocol
- blackdetect filter
- libutvideo encoder wrapper (--enable-libutvideo)
- swapuv filter
- bbox filter
- XBM encoder and decoder
- RealAudio Lossless decoder
- ZeroCodec decoder
- tile video filter
- Metal Gear Solid: The Twin Snakes demuxer
- OpenEXR image decoder
- removelogo filter
- drop support for ffmpeg without libavfilter
- drawtext video filter: fontconfig support
- ffmpeg -benchmark_all option
- super2xsai filter ported from libmpcodecs
- add libavresample audio conversion library for compatibility
- MicroDVD decoder
- Avid Meridien (AVUI) encoder and decoder
- accept + prefix to -pix_fmt option to disable automatic conversions.
- complete audio filtering in libavfilter and ffmpeg
- add fps filter
- audio split filter
- vorbis parser
- png parser
- audio mix filter
-
-
-version 0.10:
- Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
-         CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
-         CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
-         CVE-2011-3950, CVE-2011-3951, CVE-2011-3952
- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
- SBaGen (SBG) binaural beats script demuxer
- OpenMG Audio muxer
- Timecode extraction in DV and MOV
- thumbnail video filter
- XML output in ffprobe
- asplit audio filter
- tinterlace video filter
- astreamsync audio filter
- amerge audio filter
- ISMV (Smooth Streaming) muxer
- GSM audio parser
- SMJPEG muxer
- XWD encoder and decoder
- Automatic thread count based on detection number of (available) CPU cores
- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
- ffprobe -show_error option
- Avid 1:1 10-bit RGB Packer codec
- v308 Quicktime Uncompressed 4:4:4 encoder and decoder
- yuv4 libquicktime packed 4:2:0 encoder and decoder
- ffprobe -show_frames option
- silencedetect audio filter
- ffprobe -show_program_version, -show_library_versions, -show_versions options
- rv34: frame-level multi-threading
- optimized iMDCT transform on x86 using SSE for for mpegaudiodec
- Improved PGS subtitle decoder
- dumpgraph option to lavfi device
- r210 and r10k encoders
- ffwavesynth decoder
- aviocat tool
- ffeval tool
-
-
 version 0.9:

+- Indeo 4 decoder
+- SMJPEG demuxer
+- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
 - openal input device added
 - boxblur filter added
 - BWF muxer
@@ -214,7 +128,7 @@ easier to use. The changes are:
 - pan audio filter
 - IFF Amiga Continuous Bitmap (ACBM) decoder
 - ass filter
- CRI ADX audio format muxer and demuxer
+- CRI ADX audio format demuxer
 - Playstation Portable PMP format demuxer
 - Microsoft Windows ICO demuxer
 - life source
@@ -223,13 +137,11 @@ easier to use. The changes are:
 - new option: -report
 - Dxtory capture format decoder
 - cellauto source
- Simple segmenting muxer
- Indeo 4 decoder
- SMJPEG demuxer
-
+- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder

 version 0.8:

+
 - many many things we forgot because we rather write code than changelogs
 - WebM support in Matroska de/muxer
 - low overhead Ogg muxing
@@ -793,7 +705,7 @@ version 0.4.5:
 - MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
 - ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
 - Windows porting of file converter
- added MJPEG raw format (input/output)
+- added MJPEG raw format (input/ouput)
 - added JPEG image format support (input/output)


--- a/19
+++ b/19
@@ -31,13 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         = 0.11.2
-
-# 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.
-PROJECT_LOGO           =
+PROJECT_NUMBER         = 0.9.4

 # The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute)
 # base path where the generated documentation will be put.
@@ -631,7 +625,8 @@ EXCLUDE_SYMLINKS       = NO
 # for example use the pattern */test/*

 EXCLUDE_PATTERNS       = *.git \
-                         *.d
+                         *.d \
+                         avconv.c

 # The EXCLUDE_SYMBOLS tag can be used to specify one or more symbol names
 # (namespaces, classes, functions, etc.) that should be excluded from the
@@ -766,7 +761,7 @@ ALPHABETICAL_INDEX     = YES
 # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
 # in which this list will be split (can be a number in the range [1..20])

-COLS_IN_ALPHA_INDEX    = 2
+COLS_IN_ALPHA_INDEX    = 5

 # In case all classes in a project start with a common prefix, all
 # classes will be put under the same header in the alphabetical index.
@@ -800,13 +795,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.

-HTML_HEADER            = doc/doxy/header.html
+HTML_HEADER            =

 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.

-HTML_FOOTER            = doc/doxy/footer.html
+HTML_FOOTER            =

 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -815,7 +810,7 @@ HTML_FOOTER            = doc/doxy/footer.html
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!

-HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
+HTML_STYLESHEET        =

 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
--- a/8
+++ b/8
@@ -20,12 +20,8 @@ Specifically, the GPL parts of FFmpeg are

 There are a handful of files under other licensing terms, namely:

-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
-  licensing details. Specifically note that you must credit the IJG in the
-  documentation accompanying your program if you only distribute executables.
-  You must also indicate any changes including additions and deletions to
-  those three files in the documentation.
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
+  are taken from libjpeg, see the top of the files for licensing details.

 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
--- a/25
+++ b/25
@@ -4,7 +4,7 @@ FFmpeg maintainers
 Below is a list of the people maintaining different parts of the
 FFmpeg code.

-Please try to keep entries where you are the maintainer up to date!
+Please try to keep entries where you are the maintainer upto date!

 Names in () mean that the maintainer currently has no time to maintain the code.
 A CC after the name means that the maintainer prefers to be CC-ed on patches
@@ -87,8 +87,6 @@ Generic Parts:
    bitstream.c, bitstream.h            Michael Niedermayer
  CABAC:
    cabac.h, cabac.c                    Michael Niedermayer
-  codec names:
-    codec_names.sh                      Nicolas George
  DSP utilities:
    dsputils.c, dsputils.h              Michael Niedermayer
  entropy coding:
@@ -142,7 +140,6 @@ Codecs:
  dv.c                                  Roman Shaposhnik
  eacmv*, eaidct*, eat*                 Peter Ross
  ffv1.c                                Michael Niedermayer
-  ffwavesynth.c                         Nicolas George
  flac*                                 Justin Ruggles
  flashsv*                              Benjamin Larsson
  flicvideo.c                           Mike Melanson
@@ -159,12 +156,10 @@ Codecs:
  indeo5*                               Kostya Shishkov
  interplayvideo.c                      Mike Melanson
  ivi*                                  Kostya Shishkov
-  jacosub*                              Clément Bœsch
  jpeg_ls.c                             Kostya Shishkov
  jvdec.c                               Peter Ross
  kmvc.c                                Kostya Shishkov
  lcl*.c                                Roberto Togni, Reimar Doeffinger
-  libcelt_dec.c                         Nicolas George
  libgsm.c                              Michel Bardiaux
  libdirac*                             David Conrad
  libopenjpeg.c                         Jaikrishnan Menon
@@ -172,7 +167,6 @@ Codecs:
  libschroedinger*                      David Conrad
  libspeexdec.c                         Justin Ruggles
  libtheoraenc.c                        David Conrad
-  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
  libxavs.c                             Stefan Gehrer
  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
@@ -245,7 +239,6 @@ Codecs:
  xan.c                                 Mike Melanson
  xl.c                                  Kostya Shishkov
  xvmc.c                                Ivan Kalvachev
-  zerocodec.c                           Derek Buitenhuis
  zmbv*                                 Kostya Shishkov

 Hardware acceleration:
@@ -272,10 +265,6 @@ libavfilter
 ===========

 Video filters:
-  graphdump.c                           Nicolas George
-  af_amerge.c                           Nicolas George
-  af_astreamsync.c                      Nicolas George
-  af_pan.c                              Nicolas George
  vsrc_mandelbrot.c                     Michael Niedermayer
  vf_yadif.c                            Michael Niedermayer

@@ -316,7 +305,6 @@ Muxers/Demuxers:
  ipmovie.c                             Mike Melanson
  img2.c                                Michael Niedermayer
  iss.c                                 Stefan Gehrer
-  jacosub*                              Clément Bœsch
  jvdec.c                               Peter Ross
  libmodplug.c                          Clément Bœsch
  libnut.c                              Oded Shimon
@@ -337,7 +325,6 @@ Muxers/Demuxers:
  msnwc_tcp.c                           Ramiro Polla
  mtv.c                                 Reynaldo H. Verdejo Pinochet
  mxf*                                  Baptiste Coudurier
-  mxfdec.c                              Tomas Härdin
  nsvdec.c                              Francois Revol
  nut.c                                 Michael Niedermayer
  nuv.c                                 Reimar Doeffinger
@@ -357,7 +344,6 @@ Muxers/Demuxers:
  rtpdec_asf.*                          Ronald S. Bultje
  rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
  rtsp.c                                Luca Barbato
-  sbgdec.c                              Nicolas George
  sdp.c                                 Martin Storsjo
  segafilm.c                            Mike Melanson
  siff.c                                Kostya Shishkov
@@ -374,7 +360,6 @@ Muxers/Demuxers:
  wv.c                                  Kostya Shishkov

 Protocols:
-  bluray.c                              Petri Hintukainen
  http.c                                Ronald S. Bultje
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni
@@ -400,8 +385,11 @@ x86                                     Michael Niedermayer
 Releases
 ========

-0.11                                    Michael Niedermayer
-0.10                                    Michael Niedermayer
+0.5                                     *Deprecated/Unmaintained*
+0.6                                     *Deprecated/Unmaintained*
+0.7                                     Michael Niedermayer
+0.8                                     Michael Niedermayer
+0.9                                     Michael Niedermayer



@@ -423,7 +411,6 @@ Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
 Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
--- a/29
+++ b/29
@@ -8,9 +8,9 @@ vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
-vpath %/fate_config.sh.template $(SRC_PATH)

 PROGS-$(CONFIG_FFMPEG)   += ffmpeg
+PROGS-$(CONFIG_AVCONV)   += avconv
 PROGS-$(CONFIG_FFPLAY)   += ffplay
 PROGS-$(CONFIG_FFPROBE)  += ffprobe
 PROGS-$(CONFIG_FFSERVER) += ffserver
@@ -19,11 +19,11 @@ PROGS      := $(PROGS-yes:%=%$(EXESUF))
 INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 OBJS        = $(PROGS-yes:%=%.o) cmdutils.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+HOSTPROGS  := $(TESTTOOLS:%=tests/%)
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws

-BASENAMES   = ffmpeg ffplay ffprobe ffserver
+BASENAMES   = ffmpeg avconv ffplay ffprobe ffserver
 ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
 ALLMANPAGES = $(BASENAMES:%=%.1)
@@ -31,7 +31,6 @@ ALLMANPAGES = $(BASENAMES:%=%.1)
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
-FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_AVCODEC)  += avcodec
 FFLIBS-$(CONFIG_POSTPROC) += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
@@ -39,8 +38,7 @@ FFLIBS-$(CONFIG_SWSCALE)  += swscale

 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
+DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset)

 SKIPHEADERS = cmdutils_common_opts.h

@@ -66,11 +64,9 @@ config.h: .config
 	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null

-SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
-               ARCH_HEADERS BUILT_HEADERS SKIPHEADERS                    \
-               ALTIVEC-OBJS ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS MMI-OBJS \
-               MMX-OBJS NEON-OBJS VIS-OBJS YASM-OBJS                     \
-               OBJS TESTOBJS
+SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
+               ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
+               HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS ARMV6-OBJS TOOLS

 define RESET
 $(1) :=
@@ -81,8 +77,6 @@ define DOSUBDIR
 $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/library.mak
 endef

 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
@@ -126,10 +120,9 @@ install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"

-install-data: $(DATA_FILES) $(EXAMPLES_FILES)
-	$(Q)mkdir -p "$(DATADIR)/examples"
+install-data: $(DATA_FILES)
+	$(Q)mkdir -p "$(DATADIR)"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
-	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"

 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data

@@ -163,8 +156,6 @@ coverage-html: coverage.info
 	$(Q)genhtml -o $@ $<
 	$(Q)touch $@

-check: all alltools checkheaders examples testprogs fate
-
 include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile

@@ -179,5 +170,5 @@ $(sort $(OBJDIRS)):
 # so this saves some time on slow systems.
 .SUFFIXES:

-.PHONY: all all-yes alltools check *clean config install*
+.PHONY: all all-yes alltools *clean config examples install*
 .PHONY: testprogs uninstall*
--- a/8
+++ b/8
@@ -4,15 +4,9 @@ FFmpeg README
 1) Documentation
 ----------------

-* Read the documentation in the doc/ directory in git.
-  You can also view it online at http://ffmpeg.org/documentation.html
+* Read the documentation in the doc/ directory.

 2) Licensing
 ------------

 * See the LICENSE file.
-
-3) Build and Install
--------------------
-
-* See the INSTALL file.
--- a/2
+++ b/2
@@ -1 +1 @@
-0.11.2
+0.9.4
--- a/2
+++ b/2
@@ -1 +1 @@
-0.11.2
+0.9.4
--- a/arch.mak
+++ b/arch.mak
@@ -1,13 +0,0 @@
-OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
-OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_ARMVFP)  += $(ARMVFP-OBJS)  $(ARMVFP-OBJS-yes)
-OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
-
-OBJS-$(HAVE_MMI)     += $(MMI-OBJS)     $(MMI-OBJS-yes)
-
-OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
-
-OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-OBJS-yes)
-
-OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
-OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)
--- a/avconv.c
+++ b/avconv.c
--- a/cmdutils.c
+++ b/cmdutils.c
@@ -32,13 +32,8 @@
 #include "libavformat/avformat.h"
 #include "libavfilter/avfilter.h"
 #include "libavdevice/avdevice.h"
-#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/mathematics.h"
 #include "libavutil/parseutils.h"
@@ -56,20 +51,17 @@
 #endif

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

-const int this_year = 2012;
+static const int this_year = 2014;

 static FILE *report_file;

 void init_opts(void)
 {
 #if CONFIG_SWSCALE
-    sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
-                              NULL, NULL, NULL);
+    sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC, NULL, NULL, NULL);
 #endif
-    swr_opts = swr_alloc();
 }

 void uninit_opts(void)
@@ -78,12 +70,11 @@ void uninit_opts(void)
    sws_freeContext(sws_opts);
    sws_opts = NULL;
 #endif
-    swr_free(&swr_opts);
    av_dict_free(&format_opts);
    av_dict_free(&codec_opts);
 }

-void log_callback_help(void *ptr, int level, const char *fmt, va_list vl)
+void log_callback_help(void* ptr, int level, const char* fmt, va_list vl)
 {
    vfprintf(stdout, fmt, vl);
 }
@@ -102,20 +93,19 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
    fflush(report_file);
 }

-double parse_number_or_die(const char *context, const char *numstr, int type,
-                           double min, double max)
+double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max)
 {
    char *tail;
    const char *error;
    double d = av_strtod(numstr, &tail);
    if (*tail)
-        error = "Expected number for %s but found: %s\n";
+        error= "Expected number for %s but found: %s\n";
    else if (d < min || d > max)
-        error = "The value for %s was %s which is not within %f - %f\n";
-    else if (type == OPT_INT64 && (int64_t)d != d)
-        error = "Expected int64 for %s but found %s\n";
+        error= "The value for %s was %s which is not within %f - %f\n";
+    else if(type == OPT_INT64 && (int64_t)d != d)
+        error= "Expected int64 for %s but found %s\n";
    else if (type == OPT_INT && (int)d != d)
-        error = "Expected int for %s but found %s\n";
+        error= "Expected int for %s but found %s\n";
    else
        return d;
    av_log(NULL, AV_LOG_FATAL, error, context, numstr, min, max);
@@ -123,8 +113,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
    return 0;
 }

-int64_t parse_time_or_die(const char *context, const char *timestr,
-                          int is_duration)
+int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration)
 {
    int64_t us;
    if (av_parse_time(&us, timestr, is_duration) < 0) {
@@ -135,14 +124,13 @@ int64_t parse_time_or_die(const char *context, const char *timestr,
    return us;
 }

-void show_help_options(const OptionDef *options, const char *msg, int mask,
-                       int value)
+void show_help_options(const OptionDef *options, const char *msg, int mask, int value)
 {
    const OptionDef *po;
    int first;

    first = 1;
-    for (po = options; po->name != NULL; po++) {
+    for(po = options; po->name != NULL; po++) {
        char buf[64];
        if ((po->flags & mask) == value) {
            if (first) {
@@ -169,8 +157,7 @@ void show_help_children(const AVClass *class, int flags)
        show_help_children(child, flags);
 }

-static const OptionDef *find_option(const OptionDef *po, const char *name)
-{
+static const OptionDef* find_option(const OptionDef *po, const char *name){
    const char *p = strchr(name, ':');
    int len = p ? p - name : strlen(name);

@@ -217,8 +204,8 @@ static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
        buffsize += WideCharToMultiByte(CP_UTF8, 0, argv_w[i], -1,
                                        NULL, 0, NULL, NULL);

-    win32_argv_utf8 = av_mallocz(sizeof(char *) * (win32_argc + 1) + buffsize);
-    argstr_flat     = (char *)win32_argv_utf8 + sizeof(char *) * (win32_argc + 1);
+    win32_argv_utf8 = av_mallocz(sizeof(char*) * (win32_argc + 1) + buffsize);
+    argstr_flat     = (char*)win32_argv_utf8 + sizeof(char*) * (win32_argc + 1);
    if (win32_argv_utf8 == NULL) {
        LocalFree(argv_w);
        return;
@@ -243,8 +230,8 @@ static inline void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
 }
 #endif /* WIN32 && !__MINGW32CE__ */

-int parse_option(void *optctx, const char *opt, const char *arg,
-                 const OptionDef *options)
+
+int parse_option(void *optctx, const char *opt, const char *arg, const OptionDef *options)
 {
    const OptionDef *po;
    int bool_val = 1;
@@ -255,12 +242,14 @@ int parse_option(void *optctx, const char *opt, const char *arg,
    if (!po->name && opt[0] == 'n' && opt[1] == 'o') {
        /* handle 'no' bool option */
        po = find_option(options, opt + 2);
-        if ((po->name && (po->flags & OPT_BOOL)))
-            bool_val = 0;
+        if (!(po->name && (po->flags & OPT_BOOL)))
+            goto unknown_opt;
+        bool_val = 0;
    }
    if (!po->name)
        po = find_option(options, "default");
    if (!po->name) {
+unknown_opt:
        av_log(NULL, AV_LOG_ERROR, "Unrecognized option '%s'\n", opt);
        return AVERROR(EINVAL);
    }
@@ -271,14 +260,13 @@ int parse_option(void *optctx, const char *opt, const char *arg,

    /* new-style options contain an offset into optctx, old-style address of
     * a global var*/
-    dst = po->flags & (OPT_OFFSET | OPT_SPEC) ? (uint8_t *)optctx + po->u.off
-                                              : po->u.dst_ptr;
+    dst = po->flags & (OPT_OFFSET|OPT_SPEC) ? (uint8_t*)optctx + po->u.off : po->u.dst_ptr;

    if (po->flags & OPT_SPEC) {
        SpecifierOpt **so = dst;
        char *p = strchr(opt, ':');

-        dstcount = (int *)(so + 1);
+        dstcount = (int*)(so + 1);
        *so = grow_array(*so, sizeof(**so), dstcount, *dstcount + 1);
        (*so)[*dstcount - 1].specifier = av_strdup(p ? p + 1 : "");
        dst = &(*so)[*dstcount - 1].u;
@@ -287,25 +275,24 @@ int parse_option(void *optctx, const char *opt, const char *arg,
    if (po->flags & OPT_STRING) {
        char *str;
        str = av_strdup(arg);
-        *(char **)dst = str;
+        *(char**)dst = str;
    } else if (po->flags & OPT_BOOL) {
-        *(int *)dst = bool_val;
+        *(int*)dst = bool_val;
    } else if (po->flags & OPT_INT) {
-        *(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
+        *(int*)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
    } else if (po->flags & OPT_INT64) {
-        *(int64_t *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT64_MIN, INT64_MAX);
+        *(int64_t*)dst = parse_number_or_die(opt, arg, OPT_INT64, INT64_MIN, INT64_MAX);
    } else if (po->flags & OPT_TIME) {
-        *(int64_t *)dst = parse_time_or_die(opt, arg, 1);
+        *(int64_t*)dst = parse_time_or_die(opt, arg, 1);
    } else if (po->flags & OPT_FLOAT) {
-        *(float *)dst = parse_number_or_die(opt, arg, OPT_FLOAT, -INFINITY, INFINITY);
+        *(float*)dst = parse_number_or_die(opt, arg, OPT_FLOAT, -INFINITY, INFINITY);
    } else if (po->flags & OPT_DOUBLE) {
-        *(double *)dst = parse_number_or_die(opt, arg, OPT_DOUBLE, -INFINITY, INFINITY);
+        *(double*)dst = parse_number_or_die(opt, arg, OPT_DOUBLE, -INFINITY, INFINITY);
    } else if (po->u.func_arg) {
-        int ret = po->flags & OPT_FUNC2 ? po->u.func2_arg(optctx, opt, arg)
-                                        : po->u.func_arg(opt, arg);
+        int ret = po->flags & OPT_FUNC2 ? po->u.func2_arg(optctx, opt, arg) :
+                                          po->u.func_arg(opt, arg);
        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR,
-                   "Failed to set value '%s' for option '%s'\n", arg, opt);
+            av_log(NULL, AV_LOG_ERROR, "Failed to set value '%s' for option '%s'\n", arg, opt);
            return ret;
        }
    }
@@ -315,7 +302,7 @@ int parse_option(void *optctx, const char *opt, const char *arg,
 }

 void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
-                   void (*parse_arg_function)(void *, const char*))
+                   void (* parse_arg_function)(void *, const char*))
 {
    const char *opt;
    int optindex, handleoptions = 1, ret;
@@ -345,8 +332,10 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
    }
 }

-int locate_option(int argc, char **argv, const OptionDef *options,
-                  const char *optname)
+/*
+ * Return index of option opt in argv or 0 if not found.
+ */
+static int locate_option(int argc, char **argv, const OptionDef *options, const char *optname)
 {
    const OptionDef *po;
    int i;
@@ -420,27 +409,24 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
 #define FLAGS(o) ((o)->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
 int opt_default(const char *opt, const char *arg)
 {
-    const AVOption *oc, *of, *os, *oswr = NULL;
+    const AVOption *oc, *of, *os;
    char opt_stripped[128];
    const char *p;
-    const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class(), *sc, *swr_class;
+    const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class(), *sc;

    if (!(p = strchr(opt, ':')))
        p = opt + strlen(opt);
    av_strlcpy(opt_stripped, opt, FFMIN(sizeof(opt_stripped), p - opt + 1));

-    if ((oc = 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') &&
-         (oc = av_opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ))))
+    if ((oc = 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') &&
+          (oc = av_opt_find(&cc, opt+1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ))))
        av_dict_set(&codec_opts, opt, arg, FLAGS(oc));
-    if ((of = av_opt_find(&fc, opt, NULL, 0,
-                          AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)))
+    if ((of = av_opt_find(&fc, opt, NULL, 0, AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)))
        av_dict_set(&format_opts, opt, arg, FLAGS(of));
 #if CONFIG_SWSCALE
    sc = sws_get_class();
-    if ((os = av_opt_find(&sc, opt, NULL, 0,
-                          AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+    if ((os = 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);
        if (ret < 0) {
@@ -449,17 +435,8 @@ int opt_default(const char *opt, const char *arg)
        }
    }
 #endif
-    swr_class = swr_get_class();
-    if (!oc && !of && !os && (oswr = 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;
-        }
-    }

-    if (oc || of || os || oswr)
+    if (oc || of || os)
        return 0;
    av_log(NULL, AV_LOG_ERROR, "Unrecognized option '%s'\n", opt);
    return AVERROR_OPTION_NOT_FOUND;
@@ -532,32 +509,6 @@ int opt_report(const char *opt)
    return 0;
 }

-int opt_max_alloc(const char *opt, const char *arg)
-{
-    char *tail;
-    size_t max;
-
-    max = strtol(arg, &tail, 10);
-    if (*tail) {
-        av_log(NULL, AV_LOG_FATAL, "Invalid max_alloc \"%s\".\n", arg);
-        exit_program(1);
-    }
-    av_max_alloc(max);
-    return 0;
-}
-
-int opt_cpuflags(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_codec_debug(const char *opt, const char *arg)
 {
    av_log_set_level(AV_LOG_DEBUG);
@@ -592,15 +543,13 @@ static int warned_cfg = 0;
 #define INDENT        1
 #define SHOW_VERSION  2
 #define SHOW_CONFIG   4
-#define SHOW_COPYRIGHT 8

 #define PRINT_LIB_INFO(libname, LIBNAME, flags, level)                  \
    if (CONFIG_##LIBNAME) {                                             \
        const char *indent = flags & INDENT? "  " : "";                 \
        if (flags & SHOW_VERSION) {                                     \
            unsigned int version = libname##_version();                 \
-            av_log(NULL, level,                                         \
-                   "%slib%-11s %2d.%3d.%3d / %2d.%3d.%3d\n",            \
+            av_log(NULL, level, "%slib%-9s %2d.%3d.%2d / %2d.%3d.%2d\n",\
                   indent, #libname,                                    \
                   LIB##LIBNAME##_VERSION_MAJOR,                        \
                   LIB##LIBNAME##_VERSION_MINOR,                        \
@@ -629,26 +578,8 @@ 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(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)
-{
-    const char *indent = flags & INDENT? "  " : "";
-
-    av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
-    if (flags & SHOW_COPYRIGHT)
-        av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
-               program_birth_year, this_year);
-    av_log(NULL, level, "\n");
-    av_log(NULL, level, "%sbuilt on %s %s with %s %s\n",
-           indent, __DATE__, __TIME__, CC_TYPE, CC_VERSION);
-    av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }

 void show_banner(int argc, char **argv, const OptionDef *options)
@@ -657,14 +588,18 @@ void show_banner(int argc, char **argv, const OptionDef *options)
    if (idx)
        return;

-    print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
+    av_log(NULL, AV_LOG_INFO, "%s version " FFMPEG_VERSION ", Copyright (c) %d-%d the FFmpeg developers\n",
+           program_name, program_birth_year, this_year);
+    av_log(NULL, AV_LOG_INFO, "  built on %s %s with %s %s\n",
+           __DATE__, __TIME__, CC_TYPE, CC_VERSION);
+    av_log(NULL, AV_LOG_INFO, "  configuration: " FFMPEG_CONFIGURATION "\n");
    print_all_libs_info(INDENT|SHOW_CONFIG,  AV_LOG_INFO);
    print_all_libs_info(INDENT|SHOW_VERSION, AV_LOG_INFO);
 }

 int opt_version(const char *opt, const char *arg) {
    av_log_set_callback(log_callback_help);
-    print_program_info (0           , AV_LOG_INFO);
+    printf("%s " FFMPEG_VERSION "\n", program_name);
    print_all_libs_info(SHOW_VERSION, AV_LOG_INFO);
    return 0;
 }
@@ -741,133 +676,137 @@ int opt_license(const char *opt, const char *arg)

 int opt_formats(const char *opt, const char *arg)
 {
-    AVInputFormat *ifmt  = NULL;
-    AVOutputFormat *ofmt = NULL;
+    AVInputFormat *ifmt=NULL;
+    AVOutputFormat *ofmt=NULL;
    const char *last_name;

-    printf("File formats:\n"
-           " D. = Demuxing supported\n"
-           " .E = Muxing supported\n"
-           " --\n");
-    last_name = "000";
-    for (;;) {
-        int decode = 0;
-        int encode = 0;
-        const char *name      = NULL;
-        const char *long_name = NULL;
+    printf(
+        "File formats:\n"
+        " D. = Demuxing supported\n"
+        " .E = Muxing supported\n"
+        " --\n");
+    last_name= "000";
+    for(;;){
+        int decode=0;
+        int encode=0;
+        const char *name=NULL;
+        const char *long_name=NULL;

-        while ((ofmt = av_oformat_next(ofmt))) {
-            if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
-                strcmp(ofmt->name, last_name) > 0) {
-                name      = ofmt->name;
-                long_name = ofmt->long_name;
-                encode    = 1;
+        while((ofmt= av_oformat_next(ofmt))) {
+            if((name == NULL || strcmp(ofmt->name, name)<0) &&
+                strcmp(ofmt->name, last_name)>0){
+                name= ofmt->name;
+                long_name= ofmt->long_name;
+                encode=1;
            }
        }
-        while ((ifmt = av_iformat_next(ifmt))) {
-            if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
-                strcmp(ifmt->name, last_name) > 0) {
-                name      = ifmt->name;
-                long_name = ifmt->long_name;
-                encode    = 0;
+        while((ifmt= av_iformat_next(ifmt))) {
+            if((name == NULL || strcmp(ifmt->name, name)<0) &&
+                strcmp(ifmt->name, last_name)>0){
+                name= ifmt->name;
+                long_name= ifmt->long_name;
+                encode=0;
            }
-            if (name && strcmp(ifmt->name, name) == 0)
-                decode = 1;
+            if(name && strcmp(ifmt->name, name)==0)
+                decode=1;
        }
-        if (name == NULL)
+        if(name==NULL)
            break;
-        last_name = name;
+        last_name= name;

-        printf(" %s%s %-15s %s\n",
-               decode ? "D" : " ",
-               encode ? "E" : " ",
-               name,
+        printf(
+            " %s%s %-15s %s\n",
+            decode ? "D":" ",
+            encode ? "E":" ",
+            name,
            long_name ? long_name:" ");
    }
    return 0;
 }

-static char get_media_type_char(enum AVMediaType type)
-{
-    static const char map[AVMEDIA_TYPE_NB] = {
-        [AVMEDIA_TYPE_VIDEO]      = 'V',
-        [AVMEDIA_TYPE_AUDIO]      = 'A',
-        [AVMEDIA_TYPE_DATA]       = 'D',
-        [AVMEDIA_TYPE_SUBTITLE]   = 'S',
-        [AVMEDIA_TYPE_ATTACHMENT] = 'T',
-    };
-    return type >= 0 && type < AVMEDIA_TYPE_NB && map[type] ? map[type] : '?';
-}
-
 int opt_codecs(const char *opt, const char *arg)
 {
-    AVCodec *p = NULL, *p2;
+    AVCodec *p=NULL, *p2;
    const char *last_name;
-    printf("Codecs:\n"
-           " D..... = Decoding supported\n"
-           " .E.... = Encoding supported\n"
-           " ..V... = Video codec\n"
-           " ..A... = Audio codec\n"
-           " ..S... = Subtitle codec\n"
-           " ...S.. = Supports draw_horiz_band\n"
-           " ....D. = Supports direct rendering method 1\n"
-           " .....T = Supports weird frame truncation\n"
-           " ------\n");
+    printf(
+        "Codecs:\n"
+        " D..... = Decoding supported\n"
+        " .E.... = Encoding supported\n"
+        " ..V... = Video codec\n"
+        " ..A... = Audio codec\n"
+        " ..S... = Subtitle codec\n"
+        " ...S.. = Supports draw_horiz_band\n"
+        " ....D. = Supports direct rendering method 1\n"
+        " .....T = Supports weird frame truncation\n"
+        " ------\n");
    last_name= "000";
-    for (;;) {
-        int decode = 0;
-        int encode = 0;
-        int cap    = 0;
+    for(;;){
+        int decode=0;
+        int encode=0;
+        int cap=0;
+        const char *type_str;

-        p2 = NULL;
-        while ((p = av_codec_next(p))) {
-            if ((p2 == NULL || strcmp(p->name, p2->name) < 0) &&
-                strcmp(p->name, last_name) > 0) {
-                p2 = p;
-                decode = encode = cap = 0;
+        p2=NULL;
+        while((p= av_codec_next(p))) {
+            if((p2==NULL || strcmp(p->name, p2->name)<0) &&
+                strcmp(p->name, last_name)>0){
+                p2= p;
+                decode= encode= cap=0;
            }
-            if (p2 && strcmp(p->name, p2->name) == 0) {
-                if (av_codec_is_decoder(p))
-                    decode = 1;
-                if (av_codec_is_encoder(p))
-                    encode = 1;
+            if(p2 && strcmp(p->name, p2->name)==0){
+                if(p->decode) decode=1;
+                if(p->encode) encode=1;
                cap |= p->capabilities;
            }
        }
-        if (p2 == NULL)
+        if(p2==NULL)
            break;
-        last_name = p2->name;
+        last_name= p2->name;

-        printf(" %s%s%c%s%s%s %-15s %s",
-               decode ? "D" : (/* p2->decoder ? "d" : */ " "),
-               encode ? "E" : " ",
-               get_media_type_char(p2->type),
-               cap & CODEC_CAP_DRAW_HORIZ_BAND ? "S" : " ",
-               cap & CODEC_CAP_DR1 ? "D" : " ",
-               cap & CODEC_CAP_TRUNCATED ? "T" : " ",
-               p2->name,
-               p2->long_name ? p2->long_name : "");
-#if 0
-            if (p2->decoder && decode == 0)
-                printf(" use %s for decoding", p2->decoder->name);
-#endif
+        switch(p2->type) {
+        case AVMEDIA_TYPE_VIDEO:
+            type_str = "V";
+            break;
+        case AVMEDIA_TYPE_AUDIO:
+            type_str = "A";
+            break;
+        case AVMEDIA_TYPE_SUBTITLE:
+            type_str = "S";
+            break;
+        default:
+            type_str = "?";
+            break;
+        }
+        printf(
+            " %s%s%s%s%s%s %-15s %s",
+            decode ? "D": (/*p2->decoder ? "d":*/" "),
+            encode ? "E":" ",
+            type_str,
+            cap & CODEC_CAP_DRAW_HORIZ_BAND ? "S":" ",
+            cap & CODEC_CAP_DR1 ? "D":" ",
+            cap & CODEC_CAP_TRUNCATED ? "T":" ",
+            p2->name,
+            p2->long_name ? p2->long_name : "");
+       /* if(p2->decoder && decode==0)
+            printf(" use %s for decoding", p2->decoder->name);*/
        printf("\n");
    }
    printf("\n");
-    printf("Note, the names of encoders and decoders do not always match, so there are\n"
-           "several cases where the above table shows encoder only or decoder only entries\n"
-           "even though both encoding and decoding are supported. For example, the h263\n"
-           "decoder corresponds to the h263 and h263p encoders, for file formats it is even\n"
-           "worse.\n");
+    printf(
+"Note, the names of encoders and decoders do not always match, so there are\n"
+"several cases where the above table shows encoder only or decoder only entries\n"
+"even though both encoding and decoding are supported. For example, the h263\n"
+"decoder corresponds to the h263 and h263p encoders, for file formats it is even\n"
+"worse.\n");
    return 0;
 }

 int opt_bsfs(const char *opt, const char *arg)
 {
-    AVBitStreamFilter *bsf = NULL;
+    AVBitStreamFilter *bsf=NULL;

    printf("Bitstream filters:\n");
-    while ((bsf = av_bitstream_filter_next(bsf)))
+    while((bsf = av_bitstream_filter_next(bsf)))
        printf("%s\n", bsf->name);
    printf("\n");
    return 0;
@@ -875,47 +814,31 @@ int opt_bsfs(const char *opt, const char *arg)

 int opt_protocols(const char *opt, const char *arg)
 {
-    void *opaque = NULL;
-    const char *name;
+    URLProtocol *up=NULL;

    printf("Supported file protocols:\n"
-           "Input:\n");
-    while ((name = avio_enum_protocols(&opaque, 0)))
-        printf("%s\n", name);
-    printf("Output:\n");
-    while ((name = avio_enum_protocols(&opaque, 1)))
-        printf("%s\n", name);
+           "I.. = Input  supported\n"
+           ".O. = Output supported\n"
+           "..S = Seek   supported\n"
+           "FLAGS NAME\n"
+           "----- \n");
+    while((up = av_protocol_next(up)))
+        printf("%c%c%c   %s\n",
+               up->url_read  ? 'I' : '.',
+               up->url_write ? 'O' : '.',
+               up->url_seek  ? 'S' : '.',
+               up->name);
    return 0;
 }

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

    printf("Filters:\n");
 #if CONFIG_AVFILTER
-    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;
-            for (j = 0; 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++) = '|';
-        }
-        *descr_cur = 0;
-        printf("%-16s %-10s %s\n", (*filter)->name, descr, (*filter)->description);
-    }
+    while ((filter = av_filter_next(filter)) && *filter)
+        printf("%-16s %s\n", (*filter)->name, (*filter)->description);
 #endif
    return 0;
 }
@@ -924,14 +847,15 @@ int opt_pix_fmts(const char *opt, const char *arg)
 {
    enum PixelFormat pix_fmt;

-    printf("Pixel formats:\n"
-           "I.... = Supported Input  format for conversion\n"
-           ".O... = Supported Output format for conversion\n"
-           "..H.. = Hardware accelerated format\n"
-           "...P. = Paletted format\n"
-           "....B = Bitstream format\n"
-           "FLAGS NAME            NB_COMPONENTS BITS_PER_PIXEL\n"
-           "-----\n");
+    printf(
+        "Pixel formats:\n"
+        "I.... = Supported Input  format for conversion\n"
+        ".O... = Supported Output format for conversion\n"
+        "..H.. = Hardware accelerated format\n"
+        "...P. = Paletted format\n"
+        "....B = Bitstream format\n"
+        "FLAGS NAME            NB_COMPONENTS BITS_PER_PIXEL\n"
+        "-----\n");

 #if !CONFIG_SWSCALE
 #   define sws_isSupportedInput(x)  0
@@ -981,8 +905,7 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
    FILE *f = fopen(filename, "rb");

    if (!f) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
-               strerror(errno));
+        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename, strerror(errno));
        return AVERROR(errno);
    }
    fseek(f, 0, SEEK_END);
@@ -1013,14 +936,14 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
 }

 FILE *get_preset_file(char *filename, size_t filename_size,
-                      const char *preset_name, int is_path,
-                      const char *codec_name)
+                      const char *preset_name, int is_path, const char *codec_name)
 {
    FILE *f = NULL;
    int i;
-    const char *base[3] = { getenv("FFMPEG_DATADIR"),
-                            getenv("HOME"),
-                            FFMPEG_DATADIR, };
+    const char *base[3]= { getenv("FFMPEG_DATADIR"),
+                           getenv("HOME"),
+                           FFMPEG_DATADIR,
+                         };

    if (is_path) {
        av_strlcpy(filename, preset_name, filename_size);
@@ -1046,14 +969,11 @@ FILE *get_preset_file(char *filename, size_t filename_size,
        for (i = 0; i < 3 && !f; i++) {
            if (!base[i])
                continue;
-            snprintf(filename, filename_size, "%s%s/%s.ffpreset", base[i],
-                     i != 1 ? "" : "/.ffmpeg", preset_name);
+            snprintf(filename, filename_size, "%s%s/%s.ffpreset", base[i], i != 1 ? "" : "/.ffmpeg", preset_name);
            f = fopen(filename, "r");
            if (!f && codec_name) {
                snprintf(filename, filename_size,
-                         "%s%s/%s-%s.ffpreset",
-                         base[i], i != 1 ? "" : "/.ffmpeg", codec_name,
-                         preset_name);
+                         "%s%s/%s-%s.ffpreset", base[i],  i != 1 ? "" : "/.ffmpeg", codec_name, preset_name);
                f = fopen(filename, "r");
            }
        }
@@ -1064,23 +984,22 @@ FILE *get_preset_file(char *filename, size_t filename_size,

 int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
 {
-    if (*spec <= '9' && *spec >= '0') /* opt:index */
+    if (*spec <= '9' && *spec >= '0')                                        /* opt:index */
        return strtol(spec, NULL, 0) == st->index;
-    else if (*spec == 'v' || *spec == 'a' || *spec == 's' || *spec == 'd' ||
-             *spec == 't') { /* opt:[vasdt] */
+    else if (*spec == 'v' || *spec == 'a' || *spec == 's' || *spec == 'd' || *spec == 't') { /* opt:[vasdt] */
        enum AVMediaType type;

        switch (*spec++) {
-        case 'v': type = AVMEDIA_TYPE_VIDEO;      break;
-        case 'a': type = AVMEDIA_TYPE_AUDIO;      break;
-        case 's': type = AVMEDIA_TYPE_SUBTITLE;   break;
-        case 'd': type = AVMEDIA_TYPE_DATA;       break;
+        case 'v': type = AVMEDIA_TYPE_VIDEO;    break;
+        case 'a': type = AVMEDIA_TYPE_AUDIO;    break;
+        case 's': type = AVMEDIA_TYPE_SUBTITLE; break;
+        case 'd': type = AVMEDIA_TYPE_DATA;     break;
        case 't': type = AVMEDIA_TYPE_ATTACHMENT; break;
-        default:  av_assert0(0);
+        default: abort(); // never reached, silence warning
        }
        if (type != st->codec->codec_type)
            return 0;
-        if (*spec++ == ':') { /* possibly followed by :index */
+        if (*spec++ == ':') {                                   /* possibly followed by :index */
            int i, index = strtol(spec, NULL, 0);
            for (i = 0; i < s->nb_streams; i++)
                if (s->streams[i]->codec->codec_type == type && index-- == 0)
@@ -1099,9 +1018,8 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)

            if (*endptr++ == ':') {
                int stream_idx = strtol(endptr, NULL, 0);
-                return stream_idx >= 0 &&
-                    stream_idx < s->programs[i]->nb_stream_indexes &&
-                    st->index == s->programs[i]->stream_index[stream_idx];
+                return (stream_idx >= 0 && stream_idx < s->programs[i]->nb_stream_indexes &&
+                        st->index == s->programs[i]->stream_index[stream_idx]);
            }

            for (j = 0; j < s->programs[i]->nb_stream_indexes; j++)
@@ -1109,12 +1027,6 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
                    return 1;
        }
        return 0;
-    } else if (*spec == '#') {
-        int sid;
-        char *endptr;
-        sid = strtol(spec + 1, &endptr, 0);
-        if (!*endptr)
-            return st->id == sid;
    } else if (!*spec) /* empty specifier, matches everything */
        return 1;

@@ -1122,13 +1034,11 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
    return AVERROR(EINVAL);
 }

-AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
-                                AVFormatContext *s, AVStream *st)
+AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec, AVFormatContext *s, AVStream *st)
 {
    AVDictionary    *ret = NULL;
    AVDictionaryEntry *t = NULL;
-    int            flags = s->oformat ? AV_OPT_FLAG_ENCODING_PARAM
-                                      : AV_OPT_FLAG_DECODING_PARAM;
+    int            flags = s->oformat ? AV_OPT_FLAG_ENCODING_PARAM : AV_OPT_FLAG_DECODING_PARAM;
    char          prefix = 0;
    const AVClass    *cc = avcodec_get_class();

@@ -1136,18 +1046,9 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
        return NULL;

    switch (codec->type) {
-    case AVMEDIA_TYPE_VIDEO:
-        prefix  = 'v';
-        flags  |= AV_OPT_FLAG_VIDEO_PARAM;
-        break;
-    case AVMEDIA_TYPE_AUDIO:
-        prefix  = 'a';
-        flags  |= AV_OPT_FLAG_AUDIO_PARAM;
-        break;
-    case AVMEDIA_TYPE_SUBTITLE:
-        prefix  = 's';
-        flags  |= AV_OPT_FLAG_SUBTITLE_PARAM;
-        break;
+    case AVMEDIA_TYPE_VIDEO:    prefix = 'v'; flags |= AV_OPT_FLAG_VIDEO_PARAM;    break;
+    case AVMEDIA_TYPE_AUDIO:    prefix = 'a'; flags |= AV_OPT_FLAG_AUDIO_PARAM;    break;
+    case AVMEDIA_TYPE_SUBTITLE: prefix = 's'; flags |= AV_OPT_FLAG_SUBTITLE_PARAM; break;
    }

    while (t = av_dict_get(opts, "", t, AV_DICT_IGNORE_SUFFIX)) {
@@ -1162,14 +1063,10 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
            }

        if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
-            (codec && codec->priv_class &&
-             av_opt_find(&codec->priv_class, t->key, NULL, flags,
-                         AV_OPT_SEARCH_FAKE_OBJ)))
+            (codec && codec->priv_class && av_opt_find(&codec->priv_class, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ)))
            av_dict_set(&ret, t->key, t->value, 0);
-        else if (t->key[0] == prefix &&
-                 av_opt_find(&cc, t->key + 1, NULL, flags,
-                             AV_OPT_SEARCH_FAKE_OBJ))
-            av_dict_set(&ret, t->key + 1, t->value, 0);
+        else if (t->key[0] == prefix && av_opt_find(&cc, t->key+1, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ))
+            av_dict_set(&ret, t->key+1, t->value, 0);

        if (p)
            *p = ':';
@@ -1177,8 +1074,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
    return ret;
 }

-AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
-                                           AVDictionary *codec_opts)
+AVDictionary **setup_find_stream_info_opts(AVFormatContext *s, AVDictionary *codec_opts)
 {
    int i;
    AVDictionary **opts;
@@ -1187,13 +1083,11 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
        return NULL;
    opts = av_mallocz(s->nb_streams * sizeof(*opts));
    if (!opts) {
-        av_log(NULL, AV_LOG_ERROR,
-               "Could not alloc memory for stream options.\n");
+        av_log(NULL, AV_LOG_ERROR, "Could not alloc memory for stream options.\n");
        return NULL;
    }
    for (i = 0; i < s->nb_streams; i++)
-        opts[i] = filter_codec_opts(codec_opts, avcodec_find_decoder(s->streams[i]->codec->codec_id),
-                                    s, s->streams[i]);
+        opts[i] = filter_codec_opts(codec_opts, avcodec_find_decoder(s->streams[i]->codec->codec_id), s, s->streams[i]);
    return opts;
 }

--- a/cmdutils.h
+++ b/cmdutils.h
@@ -43,15 +43,9 @@ extern const char program_name[];
 */
 extern const int program_birth_year;

-/**
- * this year, defined by the program for show_banner()
- */
-extern const int this_year;
-
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
-extern struct SwrContext *swr_opts;
 extern AVDictionary *format_opts, *codec_opts;

 /**
@@ -84,10 +78,6 @@ int opt_loglevel(const char *opt, const char *arg);

 int opt_report(const char *opt);

-int opt_max_alloc(const char *opt, const char *arg);
-
-int opt_cpuflags(const char *opt, const char *arg);
-
 int opt_codec_debug(const char *opt, const char *arg);

 /**
@@ -101,15 +91,14 @@ int opt_timelimit(const char *opt, const char *arg);
 * parsed or the corresponding value is invalid.
 *
 * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
 * @param numstr the string to be parsed
 * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
 * string should be parsed
 * @param min the minimum valid accepted value
 * @param max the maximum valid accepted value
 */
-double parse_number_or_die(const char *context, const char *numstr, int type,
-                           double min, double max);
+double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max);

 /**
 * Parse a string specifying a time and return its corresponding
@@ -117,7 +106,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
 * the string cannot be correctly parsed.
 *
 * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
 * @param timestr the string to be parsed
 * @param is_duration a flag which tells how to interpret timestr, if
 * not zero timestr is interpreted as a duration, otherwise as a
@@ -125,8 +114,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
 *
 * @see parse_date()
 */
-int64_t parse_time_or_die(const char *context, const char *timestr,
-                          int is_duration);
+int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);

 typedef struct SpecifierOpt {
    char *specifier;    /**< stream/chapter/program/... specifier */
@@ -172,8 +160,7 @@ typedef struct {
    const char *argname;
 } OptionDef;

-void show_help_options(const OptionDef *options, const char *msg, int mask,
-                       int value);
+void show_help_options(const OptionDef *options, const char *msg, int mask, int value);

 /**
 * Show help for all options with given flags in class and all its
@@ -199,20 +186,13 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
 *
 * @return on success 1 if arg was consumed, 0 otherwise; negative number on error
 */
-int parse_option(void *optctx, const char *opt, const char *arg,
-                 const OptionDef *options);
+int parse_option(void *optctx, const char *opt, const char *arg, const OptionDef *options);

 /**
- * Find the '-loglevel' option in the command line args and apply it.
+ * Find the '-loglevel' option in the commandline args and apply it.
 */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);

-/**
- * Return index of option opt in argv or 0 if not found.
- */
-int locate_option(int argc, char **argv, const OptionDef *options,
-                  const char *optname);
-
 /**
 * Check if the given stream matches a stream specifier.
 *
@@ -234,8 +214,7 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
 * @param st A stream from s for which the options should be filtered.
 * @return a pointer to the created dictionary
 */
-AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
-                                AVFormatContext *s, AVStream *st);
+AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec, AVFormatContext *s, AVStream *st);

 /**
 * Setup AVCodecContext options for avformat_find_stream_info().
@@ -248,8 +227,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
 * @return pointer to the created array of dictionaries, NULL if it
 * cannot be created
 */
-AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
-                                           AVDictionary *codec_opts);
+AVDictionary **setup_find_stream_info_opts(AVFormatContext *s, AVDictionary *codec_opts);

 /**
 * Print an error message to stderr, indicating filename and a human
@@ -358,7 +336,7 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
 * at configuration time or in a "ffpresets" folder along the executable
 * on win32, in that order. If no such file is found and
 * codec_name is defined, then search for a file named
- * codec_name-preset_name.avpreset in the above-mentioned directories.
+ * codec_name-preset_name.ffpreset in the above-mentioned directories.
 *
 * @param filename buffer where the name of the found filename is written
 * @param filename_size size in bytes of the filename buffer
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -14,7 +14,4 @@
    { "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
    { "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
    { "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
-    { "fdebug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
    { "report", 0, {(void*)opt_report}, "generate a report" },
-    { "max_alloc", HAS_ARG, {(void*)opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags", HAS_ARG | OPT_EXPERT, {(void*)opt_cpuflags}, "force specific cpu flags", "flags" },
--- a/common.mak
+++ b/common.mak
@@ -20,7 +20,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif

-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample

 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_PATH)/
@@ -73,7 +73,7 @@ COMPILE_S = $(call COMPILE,AS)
 $(OBJS):
 endif

-include $(SRC_PATH)/arch.mak
+OBJS-$(HAVE_MMX) +=  $(MMX-OBJS-yes)

 OBJS      += $(OBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
@@ -115,6 +115,6 @@ OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))

 CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
-LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
+LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp

 -include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))
--- a/575
+++ b/575
--- a/doc/APIchanges
+++ b/doc/APIchanges
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -8,51 +8,31 @@ HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
              doc/git-howto.html                                        \
              doc/libavfilter.html                                      \
              doc/platform.html                                         \
-              doc/syntax.html                                           \
-
-TXTPAGES    = doc/fate.txt                                              \
-

 DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
-ifdef HAVE_MAKEINFO
-DOCS += $(TXTPAGES)
-endif

 all-$(CONFIG_DOC): documentation

 documentation: $(DOCS)

-TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
-
-doc/%.txt: TAG = TXT
-doc/%.txt: doc/%.texi
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
-
-doc/print_options.o: libavformat/options_table.h libavcodec/options_table.h
-
-GENTEXI  = format codec
-GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
-
-$(GENTEXI): TAG = GENTEXI
-$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
-	$(M)doc/print_options $* > $@
+TEXIDEP = awk '/^@include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)

 doc/%.html: TAG = HTML
-doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
+doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
 	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+	$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<

 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(GENTEXI)
+doc/%.pod: doc/%.texi
 	$(Q)$(TEXIDEP)
-	$(M)$(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
+	$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@

 doc/%.1: TAG = MAN
-doc/%.1: doc/%.pod $(GENTEXI)
+doc/%.1: doc/%.pod
 	$(M)pod2man --section=1 --center=" " --release=" " $< > $@

-$(DOCS): | doc/
+$(DOCS): | doc
+OBJDIRS += doc

 install-progs-$(CONFIG_DOC): install-man

@@ -66,7 +46,7 @@ uninstall-man:
 	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))

 clean::
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
+	$(RM) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)

 -include $(wildcard $(DOCS:%=%.d))

--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -1,11 +1,12 @@
 Release Notes
 =============

-* 0.11 "Happiness"  May, 2012
+* 0.9 "Harmony"  December, 2011


 General notes
 -------------
+
 See the Changelog file for a list of significant changes. Note, there
 are many more new features and bugfixes than whats listed there.

--- a/doc/avconv.texi
+++ b/doc/avconv.texi
--- a/doc/avtools-common-opts.texi
+++ b/doc/avtools-common-opts.texi
@@ -11,7 +11,6 @@ corresponding value to true. They can be set to false by prefixing
 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) does a given option belong to.
@@ -41,8 +40,6 @@ streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
 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 format-specific ID.
@end table
@section Generic options

@@ -121,8 +118,8 @@ Set the logging level used by the library.
 By default the program logs to stderr, if coloring is supported by the
 terminal, colors are used to mark errors and warnings. Log coloring
 can be disabled setting the environment variable
-@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
-the environment variable @env{AV_LOG_FORCE_COLOR}.
+@env{FFMPEG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
+the environment variable @env{FFMPEG_FORCE_COLOR}.
 The use of the environment variable @env{NO_COLOR} is deprecated and
 will be dropped in a following FFmpeg version.

@@ -136,15 +133,6 @@ It also implies @code{-loglevel verbose}.
 Note: setting the environment variable @code{FFREPORT} to any value has the
 same effect.

-@item -cpuflags flags (@emph{global})
-Allows setting and clearing cpu flags. This option is intended
-for testing. Do not use it unless you know what you're doing.
-@example
-ffmpeg -cpuflags -sse+mmx ...
-ffmpeg -cpuflags mmx ...
-ffmpeg -cpuflags 0 ...
-@end example
-
@end table

@section AVOptions
@@ -177,6 +165,3 @@ use @option{-option 0}/@option{-option 1}.

 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.
-
-@include avoptions_codec.texi
-@include avoptions_format.texi
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -23,20 +23,6 @@ Below is a description of the currently available bitstream filters.

@section h264_mp4toannexb

-Convert an H.264 bitstream from length prefixed mode to start code
-prefixed mode (as defined in the Annex B of the ITU-T H.264
-specification).
-
-This is required by some streaming formats, typically the MPEG-2
-transport stream format ("mpegts").
-
-For example to remux an MP4 file containing an H.264 stream to mpegts
-format with @command{ffmpeg}, you can use the command:
-
-@example
-ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
-@end example
-
@section imx_dump_header

@section mjpeg2jpeg
@@ -71,7 +57,7 @@ stream (carrying the AVI1 header ID and lacking a DHT segment) to
 produce fully qualified JPEG images.

@example
-ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
+ffmpeg -i mjpeg-movie.avi -c:v copy -vbsf mjpeg2jpeg frame_%d.jpg
 exiftran -i -9 frame*.jpg
 ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -48,16 +48,3 @@ top-field-first is assumed
@end table

@c man end VIDEO DECODERS
-
-@chapter Audio Decoders
-@c man begin AUDIO DECODERS
-
-@section ffwavesynth
-
-Internal wave synthetizer.
-
-This decoder generates wave patterns according to predefined sequences. Its
-use is purely internal and the format of the data it accepts is not publicly
-documented.
-
-@c man end AUDIO DECODERS
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -75,34 +75,4 @@ 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
-
-SBaGen script demuxer.
-
-This demuxer reads the script language used by SBaGen
-@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
-script looks like that:
-@example
-SE
-a: 300-2.5/3 440+4.5/0
-b: 300-2.5/0 440+4.5/3
-off: -
-NOW      == a
-+0:07:00 == b
-+0:14:00 == a
-+0:21:00 == b
-+0:30:00    off
-@end example
-
-A SBG script can mix absolute and relative timestamps. If the script uses
-either only absolute timestamps (including the script start time) or only
-relative ones, then its layout is fixed, and the conversion is
-straightforward. On the other hand, if the script mixes both kind of
-timestamps, then the @var{NOW} reference for relative timestamps will be
-taken from the current time of day at the time the script is read, and the
-script layout will be frozen according to that reference. That means that if
-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.
-
@c man end INPUT DEVICES
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -14,13 +14,12 @@
@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.
+decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.

@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.
+player. See @file{libavformat/output-example.c} to use it to generate
+audio or video streams.

@end itemize

@@ -188,8 +187,6 @@ the following snippet into your @file{.vimrc}:
 set expandtab
 set shiftwidth=4
 set softtabstop=4
-set cindent
-set cinoptions=(0
 " allow tabs in Makefiles
 autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
 " Trailing whitespace and tabs are forbidden, so highlight them.
@@ -201,16 +198,10 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/

 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@example
-(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode nil)
-               (show-trailing-whitespace t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
+(setq c-default-style "k&r")
+(setq-default c-basic-offset 4)
+(setq-default indent-tabs-mode nil)
+(setq-default show-trailing-whitespace t)
@end example

@section Development Policy
@@ -354,7 +345,7 @@ for us and greatly increases your chances of getting your patch applied.
 Use the patcheck tool of FFmpeg to check your patch.
 The tool is located in the tools directory.

-Run the @ref{Regression tests} before submitting a patch in order to verify
+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
@@ -517,13 +508,12 @@ not related to the comments received during review. Such patches will
 be rejected. Instead, submit significant changes or new features as
 separate patches.

-@anchor{Regression tests}
@section Regression tests

 Before submitting a patch (or committing to the repository), you should at least
 test that you did not break anything.

-Running 'make fate' accomplishes this, please see @url{fate.html} for details.
+Running 'make fate' accomplishes this, please see @file{doc/fate.txt} for details.

 [Of course, some patches may change the results of the regression tests. In
 this case, the reference results of the regression tests shall be modified
--- a/doc/doxy/doxy_stylesheet.css
+++ b/doc/doxy/doxy_stylesheet.css
--- a/doc/doxy/footer.html
+++ b/doc/doxy/footer.html
@@ -1,10 +0,0 @@
- </div>
-
- <div id="footer">
-   Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
- </div>
-
-</div>
-
-</body>
-</html>
--- a/doc/doxy/header.html
+++ b/doc/doxy/header.html
@@ -1,14 +0,0 @@
-<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
-<html xmlns="http://www.w3.org/1999/xhtml">
-<head>
-<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
-<meta http-equiv="X-UA-Compatible" content="IE=9"/>
-<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
-<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
-<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
-</head>
-
-<div id="container">
-
-<div id="body">
- <div>
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -577,7 +577,7 @@ Allow to set any x264 option, see x264 --fullhelp for a list.
 ":".
@end table

-For example to specify libx264 encoding options with @command{ffmpeg}:
+For example to specify libx264 encoding options with @file{ffmpeg}:
@example
 ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example
--- a/doc/eval.texi
+++ b/doc/eval.texi
@@ -98,26 +98,6 @@ 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:
@@ -130,20 +110,19 @@ exp(1) (Euler's number), approximately 2.718
 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:
+Note that:

@code{*} works like AND

@code{+} works like OR

-and the construct:
+thus
@example
 if A then B else C
@end example
 is equivalent to
@example
-if(A,B) + ifnot(A,C)
+A*B + not(A)*C
@end example

 In your C code, you can extend the list of unary and binary functions,
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -1,28 +1,17 @@
-# use pkg-config for getting CFLAGS and LDLIBS
-FFMPEG_LIBS=    libavdevice                        \
-                libavformat                        \
-                libavfilter                        \
-                libavcodec                         \
-                libavresample                      \
-                libswresample                      \
-                libswscale                         \
-                libavutil                          \
+# use pkg-config for getting CFLAGS abd LDFLAGS
+FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
+CFLAGS+=$(shell pkg-config  --cflags $(FFMPEG_LIBS))
+LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))

-CFLAGS += -Wall -O2 -g
-CFLAGS += $(shell pkg-config --cflags $(FFMPEG_LIBS))
-LDLIBS += $(shell pkg-config --libs $(FFMPEG_LIBS))
-
-EXAMPLES=       decoding_encoding                  \
-                filtering_video                    \
-                filtering_audio                    \
-                metadata                           \
-                muxing                             \
+EXAMPLES=decoding_encoding filtering metadata muxing

 OBJS=$(addsuffix .o,$(EXAMPLES))

-# the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
-muxing:            LDLIBS += -lm
+%: %.o
+	$(CC) $< $(LDFLAGS) -o $@
+
+%.o: %.c
+	$(CC) $< $(CFLAGS) -c -o $@

 .phony: all clean

--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -29,13 +29,11 @@
 * format handling
 */

-#include <math.h>
-
-#include <libavutil/imgutils.h>
-#include <libavutil/opt.h>
-#include <libavcodec/avcodec.h>
-#include <libavutil/mathematics.h>
-#include <libavutil/samplefmt.h>
+#include "libavutil/imgutils.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avcodec.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/samplefmt.h"

 #define INBUF_SIZE 4096
 #define AUDIO_INBUF_SIZE 20480
@@ -54,7 +52,7 @@ static void audio_encode_example(const char *filename)
    float t, tincr;
    uint8_t *outbuf;

-    printf("Encode audio file %s\n", filename);
+    printf("Audio encoding\n");

    /* find the MP2 encoder */
    codec = avcodec_find_encoder(CODEC_ID_MP2);
@@ -72,7 +70,7 @@ static void audio_encode_example(const char *filename)
    c->sample_fmt = AV_SAMPLE_FMT_S16;

    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }
@@ -125,7 +123,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)

    av_init_packet(&avpkt);

-    printf("Decode audio file %s\n", filename);
+    printf("Audio decoding\n");

    /* find the mpeg audio decoder */
    codec = avcodec_find_decoder(CODEC_ID_MP2);
@@ -137,7 +135,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
    c = avcodec_alloc_context3(codec);

    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }
@@ -213,13 +211,12 @@ static void video_encode_example(const char *filename, int codec_id)
 {
    AVCodec *codec;
    AVCodecContext *c= NULL;
-    int i, out_size, x, y, outbuf_size;
+    int i, out_size, size, x, y, outbuf_size;
    FILE *f;
    AVFrame *picture;
    uint8_t *outbuf;
-    int had_output=0;

-    printf("Encode video file %s\n", filename);
+    printf("Video encoding\n");

    /* find the mpeg1 video encoder */
    codec = avcodec_find_encoder(codec_id);
@@ -246,7 +243,7 @@ static void video_encode_example(const char *filename, int codec_id)
        av_opt_set(c->priv_data, "preset", "slow", 0);

    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }
@@ -258,7 +255,7 @@ static void video_encode_example(const char *filename, int codec_id)
    }

    /* alloc image and output buffer */
-    outbuf_size = 100000 + 12*c->width*c->height;
+    outbuf_size = 100000;
    outbuf = malloc(outbuf_size);

    /* the image can be allocated by any means and av_image_alloc() is
@@ -287,17 +284,15 @@ static void video_encode_example(const char *filename, int codec_id)

        /* encode the image */
        out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
-        had_output |= out_size;
        printf("encoding frame %3d (size=%5d)\n", i, out_size);
        fwrite(outbuf, 1, out_size, f);
    }

    /* get the delayed frames */
-    for(; out_size || !had_output; i++) {
+    for(; out_size; i++) {
        fflush(stdout);

        out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
-        had_output |= out_size;
        printf("write frame %3d (size=%5d)\n", i, out_size);
        fwrite(outbuf, 1, out_size, f);
    }
@@ -351,7 +346,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);

-    printf("Decode video file %s\n", filename);
+    printf("Video decoding\n");

    /* find the mpeg1 video decoder */
    codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
@@ -371,7 +366,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
       available in the bitstream. */

    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }
@@ -458,6 +453,9 @@ int main(int argc, char **argv)
 {
    const char *filename;

+    /* must be called before using avcodec lib */
+    avcodec_init();
+
    /* register all the codecs */
    avcodec_register_all();

--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -27,13 +27,11 @@
 */

 #define _XOPEN_SOURCE 600 /* for usleep */
-#include <unistd.h>

 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
+#include <libavfilter/vsrc_buffer.h>

 const char *filter_descr = "scale=78:24";

@@ -47,7 +45,7 @@ static int64_t last_pts = AV_NOPTS_VALUE;

 static int open_input_file(const char *filename)
 {
-    int ret;
+    int ret, i;
    AVCodec *dec;

    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
@@ -55,7 +53,7 @@ static int open_input_file(const char *filename)
        return ret;
    }

-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+    if ((ret = av_find_stream_info(fmt_ctx)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
        return ret;
    }
@@ -70,7 +68,7 @@ static int open_input_file(const char *filename)
    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;

    /* init the video decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+    if ((ret = avcodec_open(dec_ctx, dec)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
        return ret;
    }
@@ -120,13 +118,12 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

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

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
        return ret;
-    return 0;
 }

 static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
@@ -198,14 +195,15 @@ int main(int argc, char **argv)
            }

            if (got_frame) {
-                frame.pts = av_frame_get_best_effort_timestamp(&frame);
-
+                if (frame.pts == AV_NOPTS_VALUE)
+                    frame.pts = frame.pkt_dts == AV_NOPTS_VALUE ?
+                        frame.pkt_dts : frame.pkt_pts;
                /* push the decoded frame into the filtergraph */
-                av_vsrc_buffer_add_frame(buffersrc_ctx, &frame, 0);
+                av_vsrc_buffer_add_frame(buffersrc_ctx, &frame);

                /* pull filtered pictures from the filtergraph */
                while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
-                    av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
+                    av_vsink_buffer_get_video_buffer_ref(buffersink_ctx, &picref, 0);
                    if (picref) {
                        display_picref(picref, buffersink_ctx->inputs[0]->time_base);
                        avfilter_unref_buffer(picref);
@@ -218,7 +216,7 @@ end:
    avfilter_graph_free(&filter_graph);
    if (dec_ctx)
        avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
+    av_close_input_file(fmt_ctx);

    if (ret < 0 && ret != AVERROR_EOF) {
        char buf[1024];
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -1,235 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2012 Clément Bœsch
- *
- * 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.
- */
-
-/**
- * @file
- * API example for audio decoding and filtering
- */
-
-#include <unistd.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-
-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;
-AVFilterContext *buffersink_ctx;
-AVFilterContext *buffersrc_ctx;
-AVFilterGraph *filter_graph;
-static int audio_stream_index = -1;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *dec;
-
-    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    /* select the audio stream */
-    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
-        return ret;
-    }
-    audio_stream_index = ret;
-    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-
-    /* init the audio decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-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");
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    const int64_t *chlayouts                = avfilter_all_channel_layouts;
-    AVABufferSinkParams *abuffersink_params;
-    const AVFilterLink *outlink;
-
-    filter_graph = avfilter_graph_alloc();
-
-    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (!dec_ctx->channel_layout)
-        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64,
-             dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
-    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
-                                       args, NULL, filter_graph);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
-    }
-
-    /* buffer audio sink: to terminate the filter chain. */
-    abuffersink_params = av_abuffersink_params_alloc();
-    abuffersink_params->sample_fmts     = sample_fmts;
-    abuffersink_params->channel_layouts = chlayouts;
-    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       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;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
-                                    &inputs, &outputs, NULL)) < 0)
-        return ret;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-
-    /* Print summary of the sink buffer
-     * Note: args buffer is reused to store channel layout string */
-    outlink = buffersink_ctx->inputs[0];
-    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
-    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
-           (int)outlink->sample_rate,
-           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
-           args);
-
-    return 0;
-}
-
-static void print_samplesref(AVFilterBufferRef *samplesref)
-{
-    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) {
-        fputc(*p    & 0xff, stdout);
-        fputc(*p>>8 & 0xff, stdout);
-        p++;
-    }
-    fflush(stdout);
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet;
-    AVFrame frame;
-    int got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
-        exit(1);
-    }
-
-    avcodec_register_all();
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = init_filters(filter_descr)) < 0)
-        goto end;
-
-    /* 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);
-            got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
-            av_free_packet(&packet);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
-                continue;
-            }
-
-            if (got_frame) {
-                /* push the audio data from decoded frame into the filtergraph */
-                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 (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
-                    av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
-                    if (samplesref) {
-                        print_samplesref(samplesref);
-                        avfilter_unref_buffer(samplesref);
-                    }
-                }
-            }
-        }
-    }
-end:
-    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
-
-    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
-        exit(1);
-    }
-
-    exit(0);
-}
--- a/doc/examples/metadata.c
+++ b/doc/examples/metadata.c
@@ -50,6 +50,6 @@ int main (int argc, char **argv)
    while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
        printf("%s=%s\n", tag->key, tag->value);

-    avformat_close_input(&fmt_ctx);
+    avformat_free_context(fmt_ctx);
    return 0;
 }
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -33,9 +33,9 @@
 #include <string.h>
 #include <math.h>

-#include <libavutil/mathematics.h>
-#include <libavformat/avformat.h>
-#include <libswscale/swscale.h>
+#include "libavutil/mathematics.h"
+#include "libavformat/avformat.h"
+#include "libswscale/swscale.h"

 #undef exit

@@ -43,7 +43,7 @@
 #define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
 #define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
-#define STREAM_PIX_FMT    PIX_FMT_YUV420P /* default pix_fmt */
+#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */

 static int sws_flags = SWS_BICUBIC;

@@ -52,6 +52,8 @@ static int sws_flags = SWS_BICUBIC;

 static float t, tincr, tincr2;
 static int16_t *samples;
+static uint8_t *audio_outbuf;
+static int audio_outbuf_size;
 static int audio_input_frame_size;

 /*
@@ -61,16 +63,8 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
    AVCodecContext *c;
    AVStream *st;
-    AVCodec *codec;

-    /* find the audio encoder */
-    codec = avcodec_find_encoder(codec_id);
-    if (!codec) {
-        fprintf(stderr, "codec not found\n");
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, codec);
+    st = avformat_new_stream(oc, NULL);
    if (!st) {
        fprintf(stderr, "Could not alloc stream\n");
        exit(1);
@@ -78,12 +72,14 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
    st->id = 1;

    c = st->codec;
+    c->codec_id = codec_id;
+    c->codec_type = AVMEDIA_TYPE_AUDIO;

    /* put sample parameters */
-    c->sample_fmt  = AV_SAMPLE_FMT_S16;
-    c->bit_rate    = 64000;
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    c->bit_rate = 64000;
    c->sample_rate = 44100;
-    c->channels    = 2;
+    c->channels = 2;

    // some formats want stream headers to be separate
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
@@ -95,32 +91,54 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
 static void open_audio(AVFormatContext *oc, AVStream *st)
 {
    AVCodecContext *c;
+    AVCodec *codec;

    c = st->codec;

+    /* find the audio encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
    /* open it */
-    if (avcodec_open2(c, NULL, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }

    /* init signal generator */
-    t     = 0;
+    t = 0;
    tincr = 2 * M_PI * 110.0 / c->sample_rate;
    /* increment frequency by 110 Hz per second */
    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        audio_input_frame_size = 10000;
-    else
+    audio_outbuf_size = 10000;
+    audio_outbuf = av_malloc(audio_outbuf_size);
+
+    /* ugly hack for PCM codecs (will be removed ASAP with new PCM
+       support to compute the input frame size in samples */
+    if (c->frame_size <= 1) {
+        audio_input_frame_size = audio_outbuf_size / c->channels;
+        switch(st->codec->codec_id) {
+        case CODEC_ID_PCM_S16LE:
+        case CODEC_ID_PCM_S16BE:
+        case CODEC_ID_PCM_U16LE:
+        case CODEC_ID_PCM_U16BE:
+            audio_input_frame_size >>= 1;
+            break;
+        default:
+            break;
+        }
+    } 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);
+    }
+    samples = av_malloc(audio_input_frame_size * 2 * c->channels);
 }

-/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
- * 'nb_channels' channels. */
+/* prepare a 16 bit dummy audio frame of 'frame_size' samples and
+   'nb_channels' channels */
 static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
    int j, i, v;
@@ -129,9 +147,9 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
    q = samples;
    for (j = 0; j < frame_size; j++) {
        v = (int)(sin(t) * 10000);
-        for (i = 0; i < nb_channels; i++)
+        for(i = 0; i < nb_channels; i++)
            *q++ = v;
-        t     += tincr;
+        t += tincr;
        tincr += tincr2;
    }
 }
@@ -139,28 +157,22 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 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;
-
+    AVPacket pkt;
    av_init_packet(&pkt);
+
    c = st->codec;

    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,
-                             (uint8_t *)samples,
-                             audio_input_frame_size *
-                             av_get_bytes_per_sample(c->sample_fmt) *
-                             c->channels, 1);

-    avcodec_encode_audio2(c, &pkt, frame, &got_packet);
-    if (!got_packet)
-        return;
+    pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);

+    if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
+        pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+    pkt.flags |= AV_PKT_FLAG_KEY;
    pkt.stream_index = st->index;
+    pkt.data = audio_outbuf;

-    /* Write the compressed frame to the media file. */
+    /* write the compressed frame in the media file */
    if (av_interleaved_write_frame(oc, &pkt) != 0) {
        fprintf(stderr, "Error while writing audio frame\n");
        exit(1);
@@ -172,6 +184,7 @@ static void close_audio(AVFormatContext *oc, AVStream *st)
    avcodec_close(st->codec);

    av_free(samples);
+    av_free(audio_outbuf);
 }

 /**************************************************************/
@@ -181,21 +194,14 @@ static AVFrame *picture, *tmp_picture;
 static uint8_t *video_outbuf;
 static int frame_count, video_outbuf_size;

-/* Add a video output stream. */
+/* add a video output stream */
 static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
    AVCodecContext *c;
    AVStream *st;
    AVCodec *codec;

-    /* find the video encoder */
-    codec = avcodec_find_encoder(codec_id);
-    if (!codec) {
-        fprintf(stderr, "codec not found\n");
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, codec);
+    st = avformat_new_stream(oc, NULL);
    if (!st) {
        fprintf(stderr, "Could not alloc stream\n");
        exit(1);
@@ -213,30 +219,30 @@ static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)

    c->codec_id = codec_id;

-    /* Put sample parameters. */
+    /* put sample parameters */
    c->bit_rate = 400000;
-    /* Resolution must be a multiple of two. */
-    c->width    = 352;
-    c->height   = 288;
-    /* timebase: This is the fundamental unit of time (in seconds) in terms
-     * of which frame timestamps are represented. For fixed-fps content,
-     * timebase should be 1/framerate and timestamp increments should be
-     * identical to 1. */
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* time base: this is the fundamental unit of time (in seconds) in terms
+       of which frame timestamps are represented. for fixed-fps content,
+       timebase should be 1/framerate and timestamp increments should be
+       identically 1. */
    c->time_base.den = STREAM_FRAME_RATE;
    c->time_base.num = 1;
-    c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
-    c->pix_fmt       = STREAM_PIX_FMT;
+    c->gop_size = 12; /* emit one intra frame every twelve frames at most */
+    c->pix_fmt = STREAM_PIX_FMT;
    if (c->codec_id == CODEC_ID_MPEG2VIDEO) {
        /* just for testing, we also add B frames */
        c->max_b_frames = 2;
    }
-    if (c->codec_id == CODEC_ID_MPEG1VIDEO) {
+    if (c->codec_id == CODEC_ID_MPEG1VIDEO){
        /* Needed to avoid using macroblocks in which some coeffs overflow.
-         * This does not happen with normal video, it just happens here as
-         * the motion of the chroma plane does not match the luma plane. */
-        c->mb_decision = 2;
+           This does not happen with normal video, it just happens here as
+           the motion of the chroma plane does not match the luma plane. */
+        c->mb_decision=2;
    }
-    /* Some formats want stream headers to be separate. */
+    // some formats want stream headers to be separate
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;

@@ -252,7 +258,7 @@ static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
    picture = avcodec_alloc_frame();
    if (!picture)
        return NULL;
-    size        = avpicture_get_size(pix_fmt, width, height);
+    size = avpicture_get_size(pix_fmt, width, height);
    picture_buf = av_malloc(size);
    if (!picture_buf) {
        av_free(picture);
@@ -265,38 +271,46 @@ static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)

 static void open_video(AVFormatContext *oc, AVStream *st)
 {
+    AVCodec *codec;
    AVCodecContext *c;

    c = st->codec;

+    /* find the video encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
    /* open the codec */
-    if (avcodec_open2(c, NULL, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }

    video_outbuf = NULL;
    if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
-        /* Allocate output buffer. */
-        /* XXX: API change will be done. */
-        /* Buffers passed into lav* can be allocated any way you prefer,
-         * as long as they're aligned enough for the architecture, and
-         * they're freed appropriately (such as using av_free for buffers
-         * allocated with av_malloc). */
+        /* allocate output buffer */
+        /* XXX: API change will be done */
+        /* buffers passed into lav* can be allocated any way you prefer,
+           as long as they're aligned enough for the architecture, and
+           they're freed appropriately (such as using av_free for buffers
+           allocated with av_malloc) */
        video_outbuf_size = 200000;
-        video_outbuf      = av_malloc(video_outbuf_size);
+        video_outbuf = av_malloc(video_outbuf_size);
    }

-    /* Allocate the encoded raw picture. */
+    /* allocate the encoded raw picture */
    picture = alloc_picture(c->pix_fmt, c->width, c->height);
    if (!picture) {
        fprintf(stderr, "Could not allocate picture\n");
        exit(1);
    }

-    /* If the output format is not YUV420P, then a temporary YUV420P
-     * picture is needed too. It is then converted to the required
-     * output format. */
+    /* if the output format is not YUV420P, then a temporary YUV420P
+       picture is needed too. It is then converted to the required
+       output format */
    tmp_picture = NULL;
    if (c->pix_fmt != PIX_FMT_YUV420P) {
        tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
@@ -307,22 +321,23 @@ static void open_video(AVFormatContext *oc, AVStream *st)
    }
 }

-/* Prepare a dummy image. */
-static void fill_yuv_image(AVFrame *pict, int frame_index,
-                           int width, int height)
+/* prepare a dummy image */
+static void fill_yuv_image(AVFrame *pict, int frame_index, int width, int height)
 {
    int x, y, i;

    i = frame_index;

    /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
+    for (y = 0; y < height; y++) {
+        for (x = 0; x < width; x++) {
            pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+        }
+    }

    /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
+    for (y = 0; y < height/2; y++) {
+        for (x = 0; x < width/2; x++) {
            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
        }
@@ -338,13 +353,13 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
    c = st->codec;

    if (frame_count >= STREAM_NB_FRAMES) {
-        /* No more frames to compress. The codec has a latency of a few
-         * frames if using B-frames, so we get the last frames by
-         * passing the same picture again. */
+        /* no more frame to compress. The codec has a latency of a few
+           frames if using B frames, so we get the last frames by
+           passing the same picture again */
    } else {
        if (c->pix_fmt != PIX_FMT_YUV420P) {
            /* as we only generate a YUV420P picture, we must convert it
-             * to the codec pixel format if needed */
+               to the codec pixel format if needed */
            if (img_convert_ctx == NULL) {
                img_convert_ctx = sws_getContext(c->width, c->height,
                                                 PIX_FMT_YUV420P,
@@ -352,8 +367,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
                                                 c->pix_fmt,
                                                 sws_flags, NULL, NULL, NULL);
                if (img_convert_ctx == NULL) {
-                    fprintf(stderr,
-                            "Cannot initialize the conversion context\n");
+                    fprintf(stderr, "Cannot initialize the conversion context\n");
                    exit(1);
                }
            }
@@ -365,38 +379,36 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
        }
    }

+
    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* Raw video case - the API will change slightly in the near
-         * future for that. */
+        /* raw video case. The API will change slightly in the near
+           future for that. */
        AVPacket pkt;
        av_init_packet(&pkt);

-        pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = st->index;
-        pkt.data          = (uint8_t *)picture;
-        pkt.size          = sizeof(AVPicture);
+        pkt.flags |= AV_PKT_FLAG_KEY;
+        pkt.stream_index = st->index;
+        pkt.data = (uint8_t *)picture;
+        pkt.size = sizeof(AVPicture);

        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
        /* encode the image */
-        out_size = avcodec_encode_video(c, video_outbuf,
-                                        video_outbuf_size, picture);
-        /* If size is zero, it means the image was buffered. */
+        out_size = avcodec_encode_video(c, video_outbuf, video_outbuf_size, picture);
+        /* if zero size, it means the image was buffered */
        if (out_size > 0) {
            AVPacket pkt;
            av_init_packet(&pkt);

            if (c->coded_frame->pts != AV_NOPTS_VALUE)
-                pkt.pts = av_rescale_q(c->coded_frame->pts,
-                                       c->time_base, st->time_base);
-            if (c->coded_frame->key_frame)
+                pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+            if(c->coded_frame->key_frame)
                pkt.flags |= AV_PKT_FLAG_KEY;
-
            pkt.stream_index = st->index;
-            pkt.data         = video_outbuf;
-            pkt.size         = out_size;
+            pkt.data = video_outbuf;
+            pkt.size = out_size;

-            /* Write the compressed frame to the media file. */
+            /* write the compressed frame in the media file */
            ret = av_interleaved_write_frame(oc, &pkt);
        } else {
            ret = 0;
@@ -433,7 +445,7 @@ int main(int argc, char **argv)
    double audio_pts, video_pts;
    int i;

-    /* Initialize libavcodec, and register all codecs and formats. */
+    /* initialize libavcodec, and register all codecs and formats */
    av_register_all();

    if (argc != 2) {
@@ -458,8 +470,8 @@ int main(int argc, char **argv)
    }
    fmt = oc->oformat;

-    /* Add the audio and video streams using the default format codecs
-     * and initialize the codecs. */
+    /* add the audio and video streams using the default format codecs
+       and initialize the codecs */
    video_st = NULL;
    audio_st = NULL;
    if (fmt->video_codec != CODEC_ID_NONE) {
@@ -469,15 +481,15 @@ int main(int argc, char **argv)
        audio_st = add_audio_stream(oc, fmt->audio_codec);
    }

-    /* Now that all the parameters are set, we can open the audio and
-     * video codecs and allocate the necessary encode buffers. */
+    av_dump_format(oc, 0, filename, 1);
+
+    /* now that all the parameters are set, we can open the audio and
+       video codecs and allocate the necessary encode buffers */
    if (video_st)
        open_video(oc, video_st);
    if (audio_st)
        open_audio(oc, audio_st);

-    av_dump_format(oc, 0, filename, 1);
-
    /* open the output file, if needed */
    if (!(fmt->flags & AVFMT_NOFILE)) {
        if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
@@ -486,20 +498,18 @@ int main(int argc, char **argv)
        }
    }

-    /* Write the stream header, if any. */
-    avformat_write_header(oc, NULL);
-
+    /* write the stream header, if any */
+    av_write_header(oc);
    picture->pts = 0;
-    for (;;) {
-        /* Compute current audio and video time. */
+    for(;;) {
+        /* compute current audio and video time */
        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 (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
+            video_pts = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
        else
            video_pts = 0.0;

@@ -516,27 +526,28 @@ int main(int argc, char **argv)
        }
    }

-    /* Write the trailer, if any. The trailer must be written before you
-     * close the CodecContexts open when you wrote the header; otherwise
-     * av_write_trailer() may try to use memory that was freed on
-     * av_codec_close(). */
+    /* write the trailer, if any.  the trailer must be written
+     * before you close the CodecContexts open when you wrote the
+     * header; otherwise write_trailer may try to use memory that
+     * was freed on av_codec_close() */
    av_write_trailer(oc);

-    /* Close each codec. */
+    /* close each codec */
    if (video_st)
        close_video(oc, video_st);
    if (audio_st)
        close_audio(oc, audio_st);

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

    /* free the stream */
    av_free(oc);
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -272,44 +272,6 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
@end example

-@section -profile option fails when encoding H.264 video with AAC audio
-
-@command{ffmpeg} prints an error like
-
-@example
-Undefined constant or missing '(' in 'baseline'
-Unable to parse option value "baseline"
-Error setting option profile to value baseline.
-@end example
-
-Short answer: write @option{-profile:v} instead of @option{-profile}.
-
-Long answer: this happens because the @option{-profile} option can apply to both
-video and audio.  Specifically the AAC encoder also defines some profiles, none
-of which are named @var{baseline}.
-
-The solution is to apply the @option{-profile} option to the video stream only
-by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
-Appending @code{:v} to it will do exactly that.
-
-@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
-
-Use @option{-dumpgraph -} to find out exactly where the channel layout is
-lost.
-
-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{aconvert} explicitly in the filter graph,
-specifying the exact format.
-
-@example
-aconvert=s16:stereo:packed
-@end example
-
@chapter Development

@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -415,16 +377,4 @@ wrong if it is larger than the average!
 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?
-
-Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
-or @code{FATE_SAMPLES} environment variable or the @code{--samples}
-@command{configure} option is set to the right path.
-
-@section Why is @code{make fate} not finding the samples?
-
-Do you happen to have a @code{~} character in the samples path to indicate a
-home directory? The value is used in ways where the shell cannot expand it,
-causing FATE to not find files. Just replace @code{~} by the full path.
-
@bye
--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -5,172 +5,131 @@
@center @titlefont{FATE Automated Testing Environment}
@end titlepage

-@node Top
@top

@contents

@chapter Introduction

-  FATE is an extended regression suite on the client-side and a means
-for results aggregation and presentation on the server-side.
+FATE provides a regression testsuite embedded within the FFmpeg build system.
+It can be run locally and optionally configured to send reports to a web
+aggregator and viewer @url{http://fate.ffmpeg.org}.

-  The first part of this document explains how you can use FATE from
-your FFmpeg source directory to test your ffmpeg binary. The second
-part describes how you can run FATE to submit the results to FFmpeg's
-FATE server.
+It is advised to run FATE before submitting patches to the current codebase
+and provide new tests when submitting patches to add additional features.

-  In any way you can have a look at the publicly viewable FATE results
-by visiting this website:
+@chapter Running FATE

-  @url{http://fate.ffmpeg.org/}
+@section Samples and References
+In order to run, FATE needs a large amount of data (samples and references)
+that is provided separately from the actual source distribution.

-  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 there recent contribution. This usually happens on the platforms
-the developers could not test on.
+To inform the build system about the testsuite location, pass
+@option{--samples=<path to the samples>} to @command{configure} or set the
+@var{SAMPLES} Make variable or the @var{FATE_SAMPLES} environment variable
+to a suitable value.

-  The second part of this document describes how you can run FATE to
-submit your results to FFmpeg's FATE server. If you want to submit your
-results be sure to check that your combination of CPU, OS and compiler
-is not already listed on the above mentioned website.
-
-  In the third part you can find a comprehensive listing of FATE makefile
-targets and variables.
-
-
-@chapter Using FATE from your FFmpeg source directory
-
-  If you want to run FATE on your machine you need to have the samples
-in place. You can get the samples via the build target fate-rsync.
-Use this command from the top-level source directory:
+The dataset is available through @command{rsync}, is possible to fetch
+the current sample using the straight rsync command or through a specific
+@ref{Makefile target}.

@example
-make fate-rsync SAMPLES=fate-suite/
-make fate       SAMPLES=fate-suite/
+# rsync -aL rsync://fate.ffmpeg.org/fate-suite/ fate-suite
@end example

-  The above commands set the samples location by passing a makefile
-variable via command line. It is also possible to set the samples
-location at source configuration time by invoking configure with
-`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the SAMPLES makefile
-variable. This is illustrated by the following commands:
-
@example
-./configure --samples=fate-suite/
-make fate-rsync
-make fate
+# make fate-rsync SAMPLES=fate-suite
@end example

-  Yet another way to tell FATE about the location of the sample
-directory is by making sure the environment variable FATE_SAMPLES
-contains the path to your samples directory. This can be achieved
-by e.g. putting that variable in your shell profile or by setting
-it in your interactive session.

-@example
-FATE_SAMPLES=fate-suite/ make fate
-@end example
-
-@float NOTE
-Do not put a '~' character in the samples path to indicate a home
-directory. Because of shell nuances, this will cause FATE to fail.
-@end float
-
-
-@chapter Submitting the results to the FFmpeg result aggregation server
-
-  To submit your results to the server you should run fate through the
-shell script tests/fate.sh from the FFmpeg sources. This script needs
-to be invoked with a configuration file as its first argument.
-
-@example
-tests/fate.sh /path/to/fate_config
-@end example
-
-  A configuration file template with comments describing the individual
-configuration variables can be found at @file{tests/fate_config.sh.template}.
-
-@ifhtml
-  The mentioned configuration template is also available here:
-@verbatiminclude ../tests/fate_config.sh.template
-@end ifhtml
-
-  Create a configuration that suits your needs, based on the configuration
-template. The `slot' configuration variable can be any string that is not
-yet used, but it is suggested that you name it adhering to the following
-pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-itself will be sourced in a shell script, therefore all shell features may
-be used. This enables you to setup the environment as you need it for your
-build.
-
-  For your first test runs the `fate_recv' variable should be empty or
-commented out. This will run everything as normal except that it will omit
-the submission of the results to the server. The following files should be
-present in $workdir as specified in the configuration file:
-
-@itemize
-    @item configure.log
-    @item compile.log
-    @item test.log
-    @item report
-    @item version
-@end itemize
-
-  When you have everything working properly you can create an SSH key and
-send its public part to the FATE server administrator.
-
-  Configure your SSH client to use public key authentication with that key
-when connecting to the FATE server. Also do not forget to check the identity
-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:
-
-  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
-
-  The only thing left is to automate the execution of the fate.sh script and
-the synchronisation of the samples directory.
-
-
-@chapter FATE makefile targets and variables
-
-@section Makefile targets
+@chapter Manual Run
+FATE regression test can be run through @command{make}.
+Specific Makefile targets and Makefile variables are available:

+@anchor{Makefile target}
+@section FATE Makefile targets
@table @option
-@item fate-rsync
-    Download/synchronize sample files to the configured samples directory.
-
@item fate-list
-    Will list all fate/regression test targets.
-
+List all fate/regression test targets.
+@item fate-rsync
+Shortcut to download the fate test samples to the specified testsuite location.
@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
-
+@section Fate Makefile variables
@table @option
@item V
-    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
-        @item 2: show everything
-    @end itemize
-
+Verbosity level, can be set to 0, 1 or 2.
+@table @option
+    @item 0
+    show just the test arguments
+    @item 1
+    show just the command used in the test
+    @item 2
+    show everything
+@end table
@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.
-@item CPUFLAGS
-    Specify CPU flags.
+Specify how many threads to use while running regression tests, it is
+quite useful to detect thread-related regressions.
@end table

-Example:
@example
-make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
+    make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
@end example
+
+@chapter Automated Tests
+In order to automatically testing specific configurations, e.g. multiple
+compilers, @command{tests/fate.sh} is provided.
+
+This shell script builds FFmpeg, runs the regression tests and prepares a
+report that can be sent to @url{fate.ffmpeg.org} or directly examined locally.
+
+@section Testing Profiles
+The configuration file passed to @command{fate.sh} is shell scripts as well.
+
+It must provide at least a @var{slot} identifier, the @var{repo} from
+which fetch the sources, the @var{samples} directory, a @var{workdir} with
+enough space to build and run all the tests.
+Optional submit command @var{fate_recv} and a @var{comment} to describe
+the testing profile are available.
+
+Additional optional parameter to tune the FFmpeg building and reporting process
+can be passed.
+
+@example
+slot=                                   # some unique identifier
+repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
+samples=/path/to/fate/samples
+workdir=                                # directory in which to do all the work
+fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
+comment=                                # optional description
+
+# the following are optional and map to configure options
+arch=
+cpu=
+cross_prefix=
+cc=
+target_os=
+sysroot=
+target_exec=
+target_path=
+extra_cflags=
+extra_ldflags=
+extra_libs=
+extra_conf=     # extra configure options not covered above
+
+#make=          # name of GNU make if not 'make'
+makeopts=       # extra options passed to 'make'
+#tar=           # command to create a tar archive from its arguments on
+                # stdout, defaults to 'tar c'
+@end example
+
+@section Submitting Reports
+In order to send reports you need to create an @command{ssh} key and send it
+to the fate server administrator.
+The current server fingerprint is @var{b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92}
+
--- a/doc/fate.txt
+++ b/doc/fate.txt
@@ -0,0 +1,139 @@
+FATE Automated Testing Environment
+==================================
+
+  FATE is an extended regression suite on the client-side and a means
+for results aggregation and presentation on the server-side.
+
+  The first part of this document explains how you can use FATE from
+your FFmpeg source directory to test your ffmpeg binary. The second
+part describes how you can run FATE to submit the results to FFmpeg's
+FATE server.
+
+  In any way you can have a look at the publicly viewable FATE results
+by visiting this website:
+
+  http://fate.ffmpeg.org/
+
+  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 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
+submit your results to FFmpeg's FATE server. If you want to submit your
+results be sure to check that your combination of CPU, OS and compiler
+is not already listed on the above mentioned website.
+
+  In the third part you can find a comprehensive listing of FATE makefile
+targets and variables.
+
+
+1. Using FATE from your FFmpeg source directory
+-----------------------------------------------
+
+  If you want to run FATE on your machine you need to have the samples
+in place. You can get the samples via the build target fate-rsync.
+Use this command from the top-level source directory:
+
+# make fate-rsync SAMPLES=fate-suite/
+# make fate       SAMPLES=fate-suite/
+
+  The above commands set the samples location by passing a makefile
+variable via command line. It is also possible to set the samples
+location at source configuration time by invoking configure with
+`--samples=<path to the samples directory>'. Afterwards you can
+invoke the makefile targets without setting the SAMPLES makefile
+variable. This is illustrated by the following commands:
+
+# ./configure --samples=fate-suite/
+# make fate-rsync
+# make fate
+
+  Yet another way to tell FATE about the location of the sample
+directory is by making sure the environment variable FATE_SAMPLES
+contains the path to your samples directory. This can be achieved
+by e.g. putting that variable in your shell profile or by setting
+it in your interactive session.
+
+# FATE_SAMPLES=fate-suite/ make fate
+
+NOTE:
+Do not put a '~' character in the samples path to indicate a home
+directory. Because of shell nuances, this will cause FATE to fail.
+
+
+2. Submitting the results to the FFmpeg result aggregation server
+-----------------------------------------------------------------
+
+  To submit your results to the server you should run fate through the
+shell script tests/fate.sh from the FFmpeg sources. This script needs
+to be invoked with a configuration file as its first argument.
+
+# tests/fate.sh /path/to/fate_config
+
+  A configuration file template with comments describing the individual
+configuration variables can be found at tests/fate_config.sh.template .
+
+  Create a configuration that suits your needs, based on the configuration
+template. The `slot' configuration variable can be any string that is not
+yet used, but it is suggested that you name it adhering to the following
+pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
+itself will be sourced in a shell script, therefore all shell features may
+be used. This enables you to setup the environment as you need it for your
+build.
+
+  For your first test runs the `fate_recv' variable should be empty or
+commented out. This will run everything as normal except that it will omit
+the submission of the results to the server. The following files should be
+present in $workdir as specified in the configuration file:
+
+  - configure.log
+  - compile.log
+  - test.log
+  - report
+  - version
+
+  When you have everything working properly you can create an SSH key and
+send its public part to the FATE server administrator.
+
+  Configure your SSH client to use public key authentication with that key
+when connecting to the FATE server. Also do not forget to check the identity
+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:
+
+  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
+
+  The only thing left is to automate the execution of the fate.sh script and
+the synchronisation of the samples directory.
+
+
+3. FATE makefile targets and variables
+--------------------------------------
+
+FATE Makefile targets:
+
+fate-list
+    Will list all fate/regression test targets.
+
+fate
+    Run the FATE test suite (requires the fate-suite dataset).
+
+FATE Makefile variables:
+
+V
+    Verbosity level, can be set to 0, 1 or 2.
+    * 0: show just the test arguments
+    * 1: show just the command used in the test
+    * 2: show everything
+
+SAMPLES
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.
+
+THREADS
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
+
+Example:
+    make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
--- a/doc/ffmpeg-mt-authorship.txt
+++ b/doc/ffmpeg-mt-authorship.txt
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -29,7 +29,7 @@ rates and resize video on the fly with a high quality polyphase filter.
 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
+specified by a plain output filename. Anything found on the commandline 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
@@ -143,7 +143,7 @@ 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.

@item -fs @var{limit_size} (@emph{output})
-Set the file size limit, expressed in bytes.
+Set the file size limit.

@item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
@@ -164,7 +164,7 @@ streams are delayed by @var{offset} seconds.
 Set the recording timestamp in the container.
 The syntax for @var{time} is:
@example
-now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
+now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.m...]]]))[Z|z])
@end example
 If the value is "now" it takes the current time.
 Time is local time unless 'Z' or 'z' is appended, in which case it is
@@ -187,9 +187,9 @@ For example, for setting the title in the output file:
 ffmpeg -i in.avi -metadata title="my title" out.flv
@end example

-To set the language of the first audio stream:
+To set the language of the second stream:
@example
-ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
+ffmpeg -i INPUT -metadata:s:1 language=eng OUTPUT
@end example

@item -target @var{type} (@emph{output})
@@ -224,23 +224,12 @@ codec-dependent.
@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. On by default.

-@item -debug_ts (@emph{global})
-Print timestamp information. It is off by default. This option is
-mostly useful for testing and debugging purposes, and the output
-format may change from one version to another, so it should not be
-employed by portable scripts.
-
-See also the option @code{-fdebug ts}.
-
@item -attach @var{filename} (@emph{output})
 Add an attachment to the output file. This is supported by a few formats
 like Matroska for e.g. fonts used in rendering subtitles. Attachments
@@ -282,10 +271,70 @@ attachments.
@item -vframes @var{number} (@emph{output})
 Set the number of video frames to record. This is an alias for @code{-frames:v}.
@item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
-Set frame rate (Hz value, fraction or abbreviation), (default = 25). For output
-streams implies @code{-vsync cfr}.
+Set frame rate (Hz value, fraction or abbreviation), (default = 25).
@item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
 Set frame size. The format is @samp{wxh} (default - same as source).
+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

@item -aspect[:@var{stream_specifier}] @var{aspect} (@emph{output,per-stream})
 Set the video display aspect ratio specified by @var{aspect}.
@@ -312,7 +361,25 @@ pad=width:height:x:y:color instead.

@item -vn (@emph{output})
 Disable video recording.
-
+@item -bt @var{tolerance}
+Set video bitrate tolerance (in bits, default 4000k).
+Has a minimum value of: (target_bitrate/target_framerate).
+In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
+willing to deviate from the target average bitrate value. This is
+not related to min/max bitrate. Lowering tolerance too much has
+an adverse effect on quality.
+@item -maxrate @var{bitrate}
+Set max video bitrate (in bit/s).
+Requires -bufsize to be set.
+@item -minrate @var{bitrate}
+Set min video bitrate (in bit/s).
+Most useful in setting up a CBR encode:
+@example
+ffmpeg -i myfile.avi -b:v 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
+@end example
+It is of little use elsewise.
+@item -bufsize @var{size}
+Set video buffer verifier buffer size (in bits).
@item -vcodec @var{codec} (@emph{output})
 Set the video codec. This is an alias for @code{-codec:v}.
@item -same_quant
@@ -361,29 +428,202 @@ also sources and sinks).  This is an alias for @code{-filter:v}.
@item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
 Set pixel format. Use @code{-pix_fmts} to show all the supported
 pixel formats.
-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 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.
-
@item -sws_flags @var{flags} (@emph{input/output})
 Set SwScaler flags.
+@item -g @var{gop_size}
+Set the group of pictures size.
+@item -intra
+deprecated, use -g 1
@item -vdt @var{n}
 Discard threshold.
+@item -qmin @var{q}
+minimum video quantizer scale (VBR)
+@item -qmax @var{q}
+maximum video quantizer scale (VBR)
+@item -qdiff @var{q}
+maximum difference between the quantizer scales (VBR)
+@item -qblur @var{blur}
+video quantizer scale blur (VBR) (range 0.0 - 1.0)
+@item -qcomp @var{compression}
+video quantizer scale compression (VBR) (default 0.5).
+Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
+
+@item -lmin @var{lambda}
+minimum video lagrange factor (VBR)
+@item -lmax @var{lambda}
+max video lagrange factor (VBR)
+@item -mblmin @var{lambda}
+minimum macroblock quantizer scale (VBR)
+@item -mblmax @var{lambda}
+maximum macroblock quantizer scale (VBR)
+
+These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
+but you may use the QP2LAMBDA constant to easily convert from 'q' units:
+@example
+ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
+@end example
+
+@item -rc_init_cplx @var{complexity}
+initial complexity for single pass encoding
+@item -b_qfactor @var{factor}
+qp factor between P- and B-frames
+@item -i_qfactor @var{factor}
+qp factor between P- and I-frames
+@item -b_qoffset @var{offset}
+qp offset between P- and B-frames
+@item -i_qoffset @var{offset}
+qp offset between P- and I-frames
+@item -rc_eq @var{equation}
+Set rate control equation (see section "Expression Evaluation")
+(default = @code{tex^qComp}).
+
+When computing the rate control equation expression, besides the
+standard functions defined in the section "Expression Evaluation", the
+following functions are available:
+@table @var
+@item bits2qp(bits)
+@item qp2bits(qp)
+@end table
+
+and the following constants are available:
+@table @var
+@item iTex
+@item pTex
+@item tex
+@item mv
+@item fCode
+@item iCount
+@item mcVar
+@item var
+@item isI
+@item isP
+@item isB
+@item avgQP
+@item qComp
+@item avgIITex
+@item avgPITex
+@item avgPPTex
+@item avgBPTex
+@item avgTex
+@end table

@item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
 Rate control override for specific intervals, formatted as "int,int,int"
 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 -me_method @var{method}
+Set motion estimation method to @var{method}.
+Available methods are (from lowest to best quality):
+@table @samp
+@item zero
+Try just the (0, 0) vector.
+@item phods
+@item log
+@item x1
+@item hex
+@item umh
+@item epzs
+(default method)
+@item full
+exhaustive search (slow and marginally better than epzs)
+@end table
+
+@item -dct_algo @var{algo}
+Set DCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_DCT_AUTO (default)
+@item 1
+FF_DCT_FASTINT
+@item 2
+FF_DCT_INT
+@item 3
+FF_DCT_MMX
+@item 4
+FF_DCT_MLIB
+@item 5
+FF_DCT_ALTIVEC
+@end table
+
+@item -idct_algo @var{algo}
+Set IDCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_IDCT_AUTO (default)
+@item 1
+FF_IDCT_INT
+@item 2
+FF_IDCT_SIMPLE
+@item 3
+FF_IDCT_SIMPLEMMX
+@item 4
+FF_IDCT_LIBMPEG2MMX
+@item 5
+FF_IDCT_PS2
+@item 6
+FF_IDCT_MLIB
+@item 7
+FF_IDCT_ARM
+@item 8
+FF_IDCT_ALTIVEC
+@item 9
+FF_IDCT_SH4
+@item 10
+FF_IDCT_SIMPLEARM
+@end table
+
+@item -er @var{n}
+Set error resilience to @var{n}.
+@table @samp
+@item 1
+FF_ER_CAREFUL (default)
+@item 2
+FF_ER_COMPLIANT
+@item 3
+FF_ER_AGGRESSIVE
+@item 4
+FF_ER_VERY_AGGRESSIVE
+@end table
+
+@item -ec @var{bit_mask}
+Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
+the following values:
+@table @samp
+@item 1
+FF_EC_GUESS_MVS (default = enabled)
+@item 2
+FF_EC_DEBLOCK (default = enabled)
+@end table
+
+@item -bf @var{frames}
+Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
+@item -mbd @var{mode}
+macroblock decision
+@table @samp
+@item 0
+FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in ffmpeg).
+@item 1
+FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
+@item 2
+FF_MB_DECISION_RD: rate distortion
+@end table
+
+@item -4mv
+Use four motion vector by macroblock (MPEG-4 only).
+@item -part
+Use data partitioning (MPEG-4 only).
+@item -bug @var{param}
+Work around encoder bugs that are not auto-detected.
+@item -strict @var{strictness}
+How strictly to follow the standards.
+@item -aic
+Enable Advanced intra coding (h263+).
+@item -umv
+Enable Unlimited Motion Vector (h263+)

@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
@@ -442,11 +682,6 @@ 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{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:
@@ -454,6 +689,28 @@ also sources and sinks).  This is an alias for @code{-filter:a}.
@table @option
@item -atag @var{fourcc/tag} (@emph{output})
 Force audio tag/fourcc. This is an alias for @code{-tag:a}.
+@item -audio_service_type @var{type}
+Set the type of service that the audio stream contains.
+@table @option
+@item ma
+Main Audio Service (default)
+@item ef
+Effects
+@item vi
+Visually Impaired
+@item hi
+Hearing Impaired
+@item di
+Dialogue
+@item co
+Commentary
+@item em
+Emergency
+@item vo
+Voice Over
+@item ka
+Karaoke
+@end table
@item -absf @var{bitstream_filter}
 Deprecated, see -bsf
@end table
@@ -481,7 +738,7 @@ Synchronize read on input.
@section Advanced options

@table @option
-@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
+@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] (@emph{output})

 Designate one or more input streams as a source for the output file. Each input
 stream is identified by the input file index @var{input_file_id} and
@@ -497,10 +754,6 @@ the source for output stream 1, etc.
 A @code{-} character before the stream identifier creates a "negative" mapping.
 It disables matching streams from already created mappings.

-An alternative @var{[linklabel]} form will map outputs from complex filter
-graphs (see the @option{-filter_complex} option) to the output file.
-@var{linklabel} must correspond to a defined output link label in the graph.
-
 For example, to map ALL streams from the first input file to output
@example
 ffmpeg -i INPUT -map 0 output
@@ -538,7 +791,7 @@ Note that using this option disables the default mappings for this output file.

@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
 Map an audio channel from a given input to an output. If
-@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
+@var{output_file_id}.@var{stream_specifier} are not set, the audio channel will
 be mapped on all the audio streams.

 Using "-1" instead of
@@ -560,63 +813,34 @@ The order of the "-map_channel" option specifies the order of the channels in
 the output stream. The output channel layout is guessed from the number of
 channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac"
 in combination of "-map_channel" makes the channel gain levels to be updated if
-input and output channel layouts don't match (for instance two "-map_channel"
-options and "-ac 6").
+channel layouts don't match (for instance two "-map_channel" options and "-ac
+6").

-You can also extract each channel of an input to specific outputs; the following
-command extracts two channels of the @var{INPUT} audio stream (file 0, stream 0)
-to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1} outputs:
+You can also extract each channel of an @var{INPUT} to specific outputs; the
+following command extract each channel of the audio stream (file 0, stream 0)
+to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1}:
@example
 ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1
@end example

-The following example splits the channels of a stereo input into two separate
-streams, which are put into the same output file:
+The following example split the channels of a stereo input into streams:
+
@example
 ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg
@end example

-Note that currently each output stream can only contain channels from a single
-input stream; you can't for example use "-map_channel" to pick multiple input
-audio channels contained in different streams (from the same or different files)
-and merge them into a single output stream. It is therefore not currently
-possible, for example, to turn two separate mono streams into a single stereo
-stream. However splitting a stereo stream into two single channel mono streams
-is possible.
+Note that "-map_channel" is currently limited to the scope of one input for
+each output; you can't for example use it to pick multiple input audio files
+and mix them into one single output.

-If you need this feature, a possible workaround is to use the @emph{amerge}
-filter. For example, if you need to merge a media (here @file{input.mkv}) with 2
-mono audio streams into one single stereo channel audio stream (and keep the
-video stream), you can use the following command:
-@example
-ffmpeg -i input.mkv -f lavfi -i "
-amovie=input.mkv:si=1 [a1];
-amovie=input.mkv:si=2 [a2];
-[a1][a2] amerge" -c:a pcm_s16le -c:v copy output.mkv
-@end example
-
-@item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata})
+@item -map_metadata[:@var{metadata_type}][:@var{index}] @var{infile}[:@var{metadata_type}][:@var{index}] (@emph{output,per-metadata})
 Set metadata information of the next output file from @var{infile}. Note that
 those are file indices (zero-based), not filenames.
-Optional @var{metadata_spec_in/out} parameters specify, which metadata to copy.
-A metadata specifier can have the following forms:
-@table @option
-@item @var{g}
-global metadata, i.e. metadata that applies to the whole file
-
-@item @var{s}[:@var{stream_spec}]
-per-stream metadata. @var{stream_spec} is a stream specifier as described
-in the @ref{Stream specifiers} chapter. In an input metadata specifier, the first
-matching stream is copied from. In an output metadata specifier, all matching
-streams are copied to.
-
-@item @var{c}:@var{chapter_index}
-per-chapter metadata. @var{chapter_index} is the zero-based chapter index.
-
-@item @var{p}:@var{program_index}
-per-program metadata. @var{program_index} is the zero-based program index.
-@end table
-If metadata specifier is omitted, it defaults to global.
+Optional @var{metadata_type} parameters specify, which metadata to copy - (g)lobal
+(i.e. metadata that applies to the whole file), per-(s)tream, per-(c)hapter or
+per-(p)rogram. All metadata specifiers other than global must be followed by the
+stream/chapter/program index. If metadata specifier is omitted, it defaults to
+global.

 By default, global metadata is copied from the first input file,
 per-stream and per-chapter metadata is copied along with streams/chapters. These
@@ -628,14 +852,6 @@ of the output file:
@example
 ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3
@end example
-
-To do the reverse, i.e. copy global metadata to all audio streams:
-@example
-ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv
-@end example
-Note that simple @code{0} would work as well in this example, since global
-metadata is assumed by default.
-
@item -map_chapters @var{input_file_index} (@emph{output})
 Copy chapters from input file with index @var{input_file_index} to the next
 output file. If no chapter mapping is specified, then chapters are copied from
@@ -679,15 +895,14 @@ Show benchmarking information at the end of an encode.
 Shows CPU time used and maximum memory consumption.
 Maximum memory consumption is not supported on all systems,
 it will usually display as 0 if not supported.
-@item -benchmark_all (@emph{global})
-Show benchmarking information during the encode.
-Shows CPU time used in various steps (audio/video encode/decode).
@item -timelimit @var{duration} (@emph{global})
 Exit after ffmpeg has been running for @var{duration} seconds.
@item -dump (@emph{global})
 Dump each input packet to stderr.
@item -hex (@emph{global})
 When dumping packets, also dump the payload.
+@item -ps @var{size}
+Set RTP payload size in bytes.
@item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
@item -loop_input
@@ -698,24 +913,21 @@ This option is deprecated, use -loop 1.
 Repeatedly loop output for formats that support looping such as animated GIF
 (0 will loop the output infinitely).
 This option is deprecated, use -loop.
+@item -threads @var{count}
+Thread count.
@item -vsync @var{parameter}
 Video sync method.
-For compatibility reasons old values can be specified as numbers.
-Newly added values will have to be specified as strings always.

@table @option
-@item 0, passthrough
+@item 0
 Each frame is passed with its timestamp from the demuxer to the muxer.
-@item 1, cfr
+@item 1
 Frames will be duplicated and dropped to achieve exactly the requested
 constant framerate.
-@item 2, vfr
+@item 2
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
-@item drop
-As passthrough but destroys all timestamps, making the muxer generate
-fresh timestamps based on frame-rate.
-@item -1, auto
+@item -1
 Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
@end table
@@ -729,33 +941,10 @@ 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.
-This option has been deprecated. Use the @code{asyncts} audio filter instead.
@item -copyts
 Copy timestamps from input to output.
-@item -copytb @var{mode}
-Specify how to set the encoder timebase when stream copying.  @var{mode} is an
-integer numeric value, and can assume one of the following values:
-
-@table @option
-@item 1
-Use the demuxer timebase.
-
-The time base is copied to the output encoder from the corresponding input
-demuxer. This is sometimes required to avoid non monotonically increasing
-timestamps when copying video streams with variable frame rate.
-
-@item 0
-Use the decoder timebase.
-
-The time base is copied to the output encoder from the corresponding input
-decoder.
-
-@item -1
-Try to make the choice automatically, in order to generate a sane output.
-@end table
-
-Default value is -1.
-
+@item -copytb
+Copy input stream time base from input to output when stream copying.
@item -shortest
 Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
@@ -781,59 +970,14 @@ Set bitstream filters for matching streams. @var{bistream_filters} is
 a comma-separated list of bitstream filters. Use the @code{-bsfs} option
 to get the list of bitstream filters.
@example
-ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
+ffmpeg -i h264.mp4 -c:v copy -vbsf h264_mp4toannexb -an out.h264
@end example
@example
-ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
+ffmpeg -i file.mov -an -vn -sbsf mov2textsub -c:s copy -f rawvideo sub.txt
@end example

@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
 Force a tag/fourcc for matching streams.
-
-@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
-Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
-(or '.') for drop.
-@example
-ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
-@end example
-
-@item -filter_complex @var{filtergraph} (@emph{global})
-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 filter graph, as described in @ref{Filtergraph syntax}.
-
-Input link labels must refer to input streams using the
-@code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map}
-uses). If @var{stream_specifier} matches multiple streams, the first one will be
-used. An unlabeled input will be connected to the first unused input stream of
-the matching type.
-
-Output link labels are referred to with @option{-map}. Unlabeled outputs are
-added to the first output file.
-
-For example, to overlay an image over video
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map
-'[out]' out.mkv
-@end example
-Here @code{[0:v]} refers to the first video stream in the first input file,
-which is linked to the first (main) input of the overlay filter. Similarly the
-first video stream in the second input is linked to the second (overlay) input
-of overlay.
-
-Assuming there is only one video stream in each input file, we can omit input
-labels, so the above is equivalent to
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map
-'[out]' out.mkv
-@end example
-
-Furthermore we can omit the output label and the single output from the filter
-graph will be added to the output file automatically, so we can simply write
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
-@end example
@end table

@section Preset files
@@ -895,7 +1039,7 @@ frame rate or decrease the frame size.
@item
 If your computer is not fast enough, you can speed up the
 compression at the expense of the compression ratio. You can use
-'-me zero' to speed up motion estimation, and '-g 0' to disable
+'-me zero' to speed up motion estimation, and '-intra' to disable
 motion estimation completely (you have only I-frames, which means it
 is about as good as JPEG compression).

@@ -1084,18 +1228,6 @@ composed of three digits padded with zeroes to express the sequence
 number. It is the same syntax supported by the C printf function, but
 only formats accepting a normal integer are suitable.

-When importing an image sequence, -i also supports expanding shell-like
-wildcard patterns (globbing) internally. To lower the chance of interfering
-with your actual file names and the shell's glob expansion, you are required
-to activate glob meta characters by prefixing them with a single @code{%}
-character, like in @code{foo-%*.jpeg}, @code{foo-%?%?%?.jpeg} or
-@code{foo-00%[234%]%*.jpeg}.
-If your filename actually contains a character sequence of a @code{%} character
-followed by a glob character, you must double the @code{%} character to escape
-it. Imagine your files begin with @code{%?-foo-}, then you could use a glob
-pattern like @code{%%?-foo-%*.jpeg}. For input patterns that could be both a
-printf or a glob pattern, ffmpeg will assume it is a glob pattern.
-
@item
 You can put many streams of the same type in the output:

@@ -1106,23 +1238,9 @@ ffmpeg -i test1.avi -i test2.avi -map 0.3 -map 0.2 -map 0.1 -map 0.0 -c copy tes
 The resulting output file @file{test12.avi} will contain first four streams from
 the input file in reverse order.

-@item
-To force CBR video output:
-@example
-ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
-@end example
-
-@item
-The four options lmin, lmax, mblmin and mblmax use 'lambda' units,
-but you may use the QP2LAMBDA constant to easily convert from 'q' units:
-@example
-ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
-@end example
-
@end itemize
@c man end EXAMPLES

-@include syntax.texi
@include eval.texi
@include decoders.texi
@include encoders.texi
--- a/doc/ffmpeg.txt
+++ b/doc/ffmpeg.txt
@@ -29,7 +29,7 @@
                                                                          \                        /                                   :
                                                                    +======\======================/======+      ^                      :
                                                          ------> 0 |   : source_index  : st-:---        |      :                      :
-                    OutputFile output_files[]           /           +------------------------------------+      :                      :
+                    OuputFile output_files[]            /           +------------------------------------+      :                      :
                                                       /          1 |   :               :    :           |      :                      :
                 ^  +------+------------+-----+       /             +------------------------------------+      :                      :
                 :  |      : ost_index -:-----:------/            2 |   :               :    :           |      :                      :
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -178,7 +178,6 @@ Seek to percentage in file corresponding to fraction of width.

@c man end

-@include syntax.texi
@include eval.texi
@include decoders.texi
@include demuxers.texi
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -94,11 +94,6 @@ For example for printing the output in JSON format, specify:
 For more details on the available output printing formats, see the
 Writers section below.

-@item -show_error
-Show information about the error found when trying to probe the input.
-
-The error information is printed within a section with name "ERROR".
-
@item -show_format
 Show information about the container format of the input multimedia
 stream.
@@ -106,11 +101,6 @@ stream.
 All the container format information is printed within a section with
 name "FORMAT".

-@item -show_format_entry @var{name}
-Like @option{-show_format}, but only prints the specified entry of the
-container format information, rather than all. This option may be given more
-than once, then all specified entries will be shown.
-
@item -show_packets
 Show information about each packet contained in the input multimedia
 stream.
@@ -118,13 +108,6 @@ stream.
 The information for each single packet is printed within a dedicated
 section with name "PACKET".

-@item -show_frames
-Show information about each frame contained in the input multimedia
-stream.
-
-The information for each single frame is printed within a dedicated
-section with name "FRAME".
-
@item -show_streams
 Show information about each media stream contained in the input
 multimedia stream.
@@ -132,37 +115,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".

-@item -count_frames
-Count the number of frames per stream and report it in the
-corresponding stream section.
-
-@item -count_packets
-Count the number of packets per stream and report it in the
-corresponding stream section.
-
-@item -show_private_data, -private
-Show private data, that is data depending on the format of the
-particular shown element.
-This option is enabled by default, but you may need to disable it
-for specific uses, for example when creating XSD-compliant XML output.
-
-@item -show_program_version
-Show information related to program version.
-
-Version information is printed within a section with name
-"PROGRAM_VERSION".
-
-@item -show_library_versions
-Show information related to library versions.
-
-Version information for each library is printed within a section with
-name "LIBRARY_VERSION".
-
-@item -show_versions
-Show information related to program and library versions. This is the
-equivalent of setting both @option{-show_program_version} and
-@option{-show_library_versions} options.
-
@item -i @var{input_file}
 Read @var{input_file}.

@@ -172,7 +124,7 @@ Read @var{input_file}.
@chapter Writers
@c man begin WRITERS

-A writer defines the output format adopted by @command{ffprobe}, and will be
+A writer defines the output format adopted by @file{ffprobe}, and will be
 used for printing all the parts of the output.

 A writer may accept one or more arguments, which specify the options to
@@ -195,22 +147,6 @@ keyN=valN
 Metadata tags are printed as a line in the corresponding FORMAT or
 STREAM section, and are prefixed by the string "TAG:".

-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-A description of the accepted options follows.
-
-@table @option
-
-@item nokey, nk
-If set to 1 specify not to print the key of each field. Default value
-is 0.
-
-@item noprint_wrappers, nw
-If set to 1 specify not to print the section header and footer.
-Default value is 0.
-@end table
-
@section compact
 Compact format.

@@ -274,81 +210,10 @@ JSON based format.

 Each section is printed using JSON notation.

-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-The description of the accepted options follows.
-
-@table @option
-
-@item compact, c
-If set to 1 enable compact output, that is each section will be
-printed on a single line. Default value is 0.
-@end table
-
 For more information about JSON, see @url{http://www.json.org/}.

-@section xml
-XML based format.
-
-The XML output is described in the XML schema description file
-@file{ffprobe.xsd} installed in the FFmpeg datadir.
-
-An updated version of the schema can be retrieved at the url
-@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
-latest schema committed into the FFmpeg development source code tree.
-
-Note that the output issued will be compliant to the
-@file{ffprobe.xsd} schema only when no special global output options
-(@option{unit}, @option{prefix}, @option{byte_binary_prefix},
-@option{sexagesimal} etc.) are specified.
-
-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-The description of the accepted options follows.
-
-@table @option
-
-@item fully_qualified, q
-If set to 1 specify if the output should be fully qualified. Default
-value is 0.
-This is required for generating an XML file which can be validated
-through an XSD file.
-
-@item xsd_compliant, x
-If set to 1 perform more checks for ensuring that the output is XSD
-compliant. Default value is 0.
-This option automatically sets @option{fully_qualified} to 1.
-@end table
-
-For more information about the XML format, see
-@url{http://www.w3.org/XML/}.
@c man end WRITERS

-@chapter Timecode
-@c man begin TIMECODE
-
-@command{ffprobe} supports Timecode extraction:
-
-@itemize
-
-@item
-MPEG1/2 timecode is extracted from the GOP, and is available in the video
-stream details (@option{-show_streams}, see @var{timecode}).
-
-@item
-MOV timecode is extracted from tmcd track, so is available in the tmcd
-stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
-
-@item
-DV and GXF timecodes are available in format metadata
-(@option{-show_format}, see @var{TAG:timecode}).
-
-@end itemize
-@c man end TIMECODE
-
-@include syntax.texi
@include decoders.texi
@include demuxers.texi
@include protocols.texi
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -1,167 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-
-<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
-    targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
-    xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
-
-    <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
-
-    <xsd:complexType name="ffprobeType">
-        <xsd:sequence>
-            <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="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" />
-            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetsType">
-        <xsd:sequence>
-            <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="framesType">
-        <xsd:sequence>
-            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetType">
-      <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
-      <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
-      <xsd:attribute name="pts"           type="xsd:long"  />
-      <xsd:attribute name="pts_time"      type="xsd:float" />
-      <xsd:attribute name="dts"           type="xsd:long"  />
-      <xsd:attribute name="dts_time"      type="xsd:float" />
-      <xsd:attribute name="duration"      type="xsd:long"  />
-      <xsd:attribute name="duration_time" type="xsd:float" />
-      <xsd:attribute name="size"          type="xsd:long" use="required" />
-      <xsd:attribute name="pos"           type="xsd:long"  />
-      <xsd:attribute name="flags"         type="xsd:string" use="required" />
-    </xsd:complexType>
-
-    <xsd:complexType name="frameType">
-      <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
-      <xsd:attribute name="pts"           type="xsd:long" />
-      <xsd:attribute name="pts_time"      type="xsd:float"/>
-      <xsd:attribute name="pkt_pts"       type="xsd:long" />
-      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
-      <xsd:attribute name="pkt_dts"       type="xsd:long" />
-      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="pkt_pos"       type="xsd:long" />
-
-      <!-- audio attributes -->
-      <xsd:attribute name="sample_fmt"             type="xsd:string"/>
-      <xsd:attribute name="nb_samples"             type="xsd:long"  />
-
-      <!-- video attributes -->
-      <xsd:attribute name="width"                  type="xsd:long"  />
-      <xsd:attribute name="height"                 type="xsd:long"  />
-      <xsd:attribute name="pix_fmt"                type="xsd:string"/>
-      <xsd:attribute name="sample_aspect_ratio"    type="xsd:string"/>
-      <xsd:attribute name="pict_type"              type="xsd:string"/>
-      <xsd:attribute name="coded_picture_number"   type="xsd:long"  />
-      <xsd:attribute name="display_picture_number" type="xsd:long"  />
-      <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">
-        <xsd:sequence>
-            <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="streamType">
-      <xsd:attribute name="index"            type="xsd:int" use="required"/>
-      <xsd:attribute name="codec_name"       type="xsd:string" />
-      <xsd:attribute name="codec_long_name"  type="xsd:string" />
-      <xsd:attribute name="codec_type"       type="xsd:string" />
-      <xsd:attribute name="codec_time_base"  type="xsd:string" use="required"/>
-      <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
-      <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
-
-      <!-- video attributes -->
-      <xsd:attribute name="width"                type="xsd:int"/>
-      <xsd:attribute name="height"               type="xsd:int"/>
-      <xsd:attribute name="has_b_frames"         type="xsd:int"/>
-      <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
-      <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
-      <xsd:attribute name="pix_fmt"              type="xsd:string"/>
-      <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="timecode"             type="xsd:string"/>
-
-      <!-- audio attributes -->
-      <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="bits_per_sample"  type="xsd:int"/>
-
-      <xsd:attribute name="id"               type="xsd:string"/>
-      <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
-      <xsd:attribute name="avg_frame_rate"   type="xsd:string" use="required"/>
-      <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
-      <xsd:attribute name="start_time"       type="xsd:float"/>
-      <xsd:attribute name="duration"         type="xsd:float"/>
-      <xsd:attribute name="bit_rate"         type="xsd:int"/>
-      <xsd:attribute name="nb_frames"        type="xsd:int"/>
-      <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
-      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="formatType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
-      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
-      <xsd:attribute name="format_long_name" type="xsd:string" use="required"/>
-      <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:complexType>
-
-    <xsd:complexType name="tagType">
-      <xsd:attribute name="key"   type="xsd:string" use="required"/>
-      <xsd:attribute name="value" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="errorType">
-      <xsd:attribute name="code"   type="xsd:int"    use="required"/>
-      <xsd:attribute name="string" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="programVersionType">
-      <xsd:attribute name="version"          type="xsd:string" use="required"/>
-      <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
-      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
-      <xsd:attribute name="configuration"    type="xsd:string" 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"/>
-      <xsd:attribute name="minor"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="micro"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="version"     type="xsd:int"    use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="libraryVersionsType">
-        <xsd:sequence>
-          <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-</xsd:schema>
--- a/doc/ffserver.conf
+++ b/doc/ffserver.conf
@@ -373,3 +373,5 @@ ACL allow 192.168.0.0 192.168.255.255
 <Redirect index.html>
 URL http://www.ffmpeg.org/
 </Redirect>
+
+
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -147,7 +147,7 @@ that only captures in stereo and also requires that one channel be flipped.
 If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
 starting ffmpeg.

-@subsection The audio and video lose sync after a while.
+@subsection The audio and video loose sync after a while.

 Yes, they do.

--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -1,116 +0,0 @@
-Filter design
-=============
-
-This document explains guidelines that should be observed (or ignored with
-good reason) when writing filters for libavfilter.
-
-In this document, the word “frame” indicates either a video frame or a group
-of audio samples, as stored in an AVFilterBuffer structure.
-
-
-Format negotiation
-==================
-
-  The query_formats method should set, for each input and each output links,
-  the list supported formats.
-
-  For video links, that means pixel format. For audio links, that means
-  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 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.
-
-  That means that if a filter requires that its input and output have the
-  same format amongst a supported list, all it have to do is use a reference
-  to the same list of formats.
-
-
-Buffer references ownership and permissions
-===========================================
-
-  TODO
-
-
-Frame scheduling
-================
-
-  The purpose of these rules is to ensure that frames flow in the filter
-  graph without getting stuck and accumulating somewhere.
-
-  Simple filters that output one frame for each input frame should not have
-  to worry about it.
-
-  start_frame / filter_samples
-  ----------------------------
-
-    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.
-
-    As an exception to the previous rule, if the input frame is enough to
-    produce several output frames, then the filter needs output only at
-    least one per link. The additional frames can be left buffered in the
-    filter; these buffered frames must be flushed immediately if a new input
-    produces new output.
-
-    (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.)
-
-    If the input frame is not enough to produce output, the filter must not
-    call request_frame to get more. It must just process the frame or queue
-    it. The task of requesting more frames is left to the filter's
-    request_frame method or the application.
-
-    If a filter has several inputs, the filter must be ready for frames
-    arriving randomly on any input. Therefore, any filter with several input
-    will most likely require some kind of queuing mechanism. It is perfectly
-    acceptable to have a limited queue and to drop frames when the inputs
-    are too unbalanced.
-
-  request_frame
-  -------------
-
-    This method is called when a frame is wanted on an output.
-
-    For an input, it should directly call start_frame or filter_samples 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
-    one of its input, repeatedly until at least one frame has been pushed.
-
-    Return values:
-    if request_frame could produce a frame, it should return 0;
-    if it could not for temporary reasons, it should return AVERROR(EAGAIN);
-    if it could not because there are no more frames, it should return
-    AVERROR_EOF.
-
-    The typical implementation of request_frame for a filter with several
-    inputs will look like that:
-
-        if (frames_queued) {
-            push_one_frame();
-            return 0;
-        }
-        while (!frame_pushed) {
-            input = input_where_a_frame_is_most_needed();
-            ret = avfilter_request_frame(input);
-            if (ret == AVERROR_EOF) {
-                process_eof_on_input();
-            } else if (ret < 0) {
-                return ret;
-            }
-        }
-        return 0;
-
-    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 start_frame / filter_samples method will be called and do the work.
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -13,8 +13,7 @@

 FFmpeg can be hooked up with a number of external libraries to add support
 for more formats. None of them are used by default, their use has to be
-explicitly requested by passing the appropriate flags to
-@command{./configure}.
+explicitly requested by passing the appropriate flags to @file{./configure}.

@section OpenJPEG

@@ -95,7 +94,7 @@ details), you must upgrade FFmpeg's license to GPL in order to use it.



-@chapter Supported File Formats, Codecs or Features
+@chapter Supported File Formats and Codecs

 You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.

@@ -135,7 +134,7 @@ library:
@item Brute Force & Ignorance   @tab   @tab X
    @tab Used in the game Flash Traffic: City of Angels.
@item BWF                       @tab X @tab X
-@item CRI ADX                   @tab X @tab X
+@item CRI ADX                   @tab   @tab X
    @tab Audio-only format used in console video games.
@item Discworld II BMV          @tab   @tab X
@item Interplay C93             @tab   @tab X
@@ -144,8 +143,6 @@ library:
    @tab Multimedia format used by Delphine Software games.
@item CD+G                      @tab   @tab X
    @tab Video format used by CD+G karaoke disks
-@item Commodore CDXL            @tab   @tab X
-    @tab Amiga CD video format
@item Core Audio Format         @tab X @tab X
    @tab Apple Core Audio Format
@item CRC testing format        @tab X @tab
@@ -211,7 +208,6 @@ library:
@item MAXIS XA                  @tab   @tab X
    @tab Used in Sim City 3000; file extension .xa.
@item MD Studio                 @tab   @tab X
-@item Metal Gear Solid: The Twin Snakes @tab @tab X
@item Mobotix .mxg              @tab   @tab X
@item Monkey's Audio            @tab   @tab X
@item Motion Pixels MVI         @tab   @tab X
@@ -311,7 +307,6 @@ library:
@item RTP                       @tab X @tab X
@item RTSP                      @tab X @tab X
@item SAP                       @tab X @tab X
-@item SBG                       @tab   @tab X
@item SDP                       @tab   @tab X
@item Sega FILM/CPK             @tab   @tab X
    @tab Used in many Sega Saturn console games.
@@ -321,9 +316,8 @@ library:
    @tab Used in Sierra CD-ROM games.
@item Smacker                   @tab   @tab X
    @tab Multimedia format used by many games.
-@item SMJPEG                    @tab X @tab X
-    @tab Used in certain Loki game ports.
-@item Sony OpenMG (OMA)         @tab X @tab X
+@item SMJPEG                    @tab   @tab X
+@item Sony OpenMG (OMA)         @tab   @tab X
    @tab Audio format used in Sony Sonic Stage and Sony Vegas.
@item Sony PlayStation STR      @tab   @tab X
@item Sony Wave64 (W64)         @tab   @tab X
@@ -353,7 +347,6 @@ library:
@item eXtended BINary text (XBIN) @tab @tab X
@item YUV4MPEG pipe             @tab X @tab X
@item Psygnosis YOP             @tab   @tab X
-@item ZeroCodec Lossless Video  @tab   @tab X
@end multitable

@code{X} means that encoding (resp. decoding) is supported.
@@ -373,8 +366,6 @@ following image formats are supported:
    @tab Microsoft BMP image
@item DPX          @tab X @tab X
    @tab Digital Picture Exchange
-@item EXR          @tab   @tab X
-    @tab OpenEXR
@item JPEG         @tab X @tab X
    @tab Progressive JPEG is not supported.
@item JPEG 2000    @tab X @tab X
@@ -400,16 +391,12 @@ following image formats are supported:
    @tab V.Flash PTX format
@item SGI          @tab X @tab X
    @tab SGI RGB image format
-@item Sun Rasterfile  @tab X @tab X
+@item Sun Rasterfile  @tab   @tab X
    @tab Sun RAS image format
@item TIFF         @tab X @tab X
    @tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa  @tab X @tab X
    @tab Targa (.TGA) image format
-@item XBM  @tab X @tab X
-    @tab X BitMap image format
-@item XWD  @tab X @tab X
-    @tab X Window Dump image format
@end multitable

@code{X} means that encoding (resp. decoding) is supported.
@@ -449,12 +436,8 @@ following image formats are supported:
@item Autodesk Animator Flic video  @tab     @tab  X
@item Autodesk RLE           @tab     @tab  X
    @tab fourcc: AASC
-@item Avid 1:1 10-bit RGB Packer  @tab  X  @tab  X
-    @tab fourcc: AVrp
@item AVS (Audio Video Standard) video  @tab     @tab  X
    @tab Video encoding used by the Creature Shock game.
-@item AYUV                   @tab  X  @tab  X
-    @tab Microsoft uncompressed packed 4:4:4:4
@item Beam Software VB       @tab     @tab  X
@item Bethesda VID video     @tab     @tab  X
    @tab Used in some games from Bethesda Softworks.
@@ -469,8 +452,6 @@ following image formats are supported:
    @tab fourcc: CSCD
@item CD+G                   @tab     @tab  X
    @tab Video codec for CD+G karaoke disks
-@item CDXL                   @tab     @tab  X
-    @tab Amiga CD video codec
@item Chinese AVS video      @tab  E  @tab  X
    @tab AVS1-P2, JiZhun profile, encoding through external library libxavs
@item Delphine Software International CIN video  @tab     @tab  X
@@ -483,7 +464,7 @@ following image formats are supported:
@item DFA                    @tab     @tab  X
    @tab Codec used in Chronomaster game.
@item Dirac                  @tab  E  @tab  X
-    @tab supported through external library libschroedinger
+    @tab supported through external libdirac/libschroedinger libraries
@item Deluxe Paint Animation @tab     @tab  X
@item DNxHD                  @tab   X @tab  X
    @tab aka SMPTE VC3
@@ -504,13 +485,12 @@ following image formats are supported:
@item Escape 124             @tab     @tab  X
@item Escape 130             @tab     @tab  X
@item FFmpeg video codec #1  @tab  X  @tab  X
-    @tab lossless codec (fourcc: FFV1)
+    @tab experimental lossless codec (fourcc: FFV1)
@item Flash Screen Video v1  @tab  X  @tab  X
    @tab fourcc: FSV1
@item Flash Screen Video v2  @tab  X  @tab  X
@item Flash Video (FLV)      @tab  X  @tab  X
    @tab Sorenson H.263 used in Flash
-@item Forward Uncompressed   @tab     @tab  X
@item Fraps                  @tab     @tab  X
@item H.261                  @tab  X  @tab  X
@item H.263 / H.263-1996     @tab  X  @tab  X
@@ -591,8 +571,8 @@ following image formats are supported:
    @tab fourcc: 'smc '
@item QuickTime video (RPZA) @tab     @tab  X
    @tab fourcc: rpza
-@item R10K AJA Kona 10-bit RGB Codec     @tab  X  @tab  X
-@item R210 Quicktime Uncompressed RGB 10-bit     @tab  X  @tab  X
+@item R10K AJA Kona 10-bit RGB Codec     @tab     @tab  X
+@item R210 Quicktime Uncompressed RGB 10-bit     @tab     @tab  X
@item Raw Video              @tab  X  @tab  X
@item RealVideo 1.0          @tab  X  @tab  X
@item RealVideo 2.0          @tab  X  @tab  X
@@ -624,10 +604,8 @@ following image formats are supported:
@item Tiertex Limited SEQ video  @tab     @tab  X
    @tab Codec used in DOS CD-ROM FlashBack game.
@item Ut Video               @tab     @tab  X
-@item v210 QuickTime uncompressed 4:2:2 10-bit     @tab  X  @tab  X
-@item v308 QuickTime uncompressed 4:4:4            @tab  X  @tab  X
-@item v408 QuickTime uncompressed 4:4:4:4          @tab  X  @tab  X
-@item v410 QuickTime uncompressed 4:4:4 10-bit     @tab  X  @tab  X
+@item V210 Quicktime Uncompressed 4:2:2 10-bit     @tab  X  @tab  X
+@item v410 Quicktime Uncompressed 4:4:4 10-bit     @tab  X  @tab  X
@item VBLE Lossless Codec    @tab     @tab  X
@item VMware Screen Codec / VMware Video  @tab     @tab  X
    @tab Codec used in videos captured by VMware.
@@ -645,8 +623,6 @@ following image formats are supported:
@item WMV7                   @tab  X  @tab  X
@item YAMAHA SMAF            @tab  X  @tab  X
@item Psygnosis YOP Video    @tab     @tab  X
-@item yuv4                   @tab  X  @tab  X
-    @tab libquicktime uncompressed packed 4:2:0
@item ZLIB                   @tab  X  @tab  X
    @tab part of LCL, encoder experimental
@item Zip Motion Blocks Video  @tab   X @tab  X
@@ -796,7 +772,6 @@ following image formats are supported:
    @tab Real 28800 bit/s codec
@item RealAudio 3.0 (dnet)   @tab IX  @tab  X
    @tab Real low bitrate AC-3 codec
-@item RealAudio Lossless     @tab     @tab  X
@item RealAudio SIPR / ACELP.NET @tab     @tab  X
@item Shorten                @tab     @tab  X
@item Sierra VMD audio       @tab     @tab  X
@@ -819,7 +794,6 @@ following image formats are supported:
@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
-@item Windows Media Audio Lossless @tab  @tab  X
@item Windows Media Audio Pro @tab    @tab  X
@item Windows Media Audio Voice @tab  @tab  X
@end multitable
@@ -838,8 +812,7 @@ performance on systems without hardware floating point support).
@item SSA/ASS      @tab X @tab X @tab X @tab X
@item DVB          @tab X @tab X @tab X @tab X
@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
+@item MicroDVD     @tab X @tab X @tab   @tab
@item PGS          @tab   @tab   @tab   @tab X
@item SubRip (SRT) @tab X @tab X @tab X @tab X
@item XSUB         @tab   @tab   @tab X @tab X
@@ -872,12 +845,11 @@ performance on systems without hardware floating point support).
@item ALSA              @tab X      @tab X
@item BKTR              @tab X      @tab
@item DV1394            @tab X      @tab
-@item Linux framebuffer @tab X      @tab
@item JACK              @tab X      @tab
-@item LIBCDIO           @tab X
@item LIBDC1394         @tab X      @tab
@item OSS               @tab X      @tab X
@item Pulseaudio        @tab X      @tab
+@item Video4Linux       @tab X      @tab
@item Video4Linux2      @tab X      @tab
@item VfW capture       @tab X      @tab
@item X11 grabbing      @tab X      @tab
@@ -885,15 +857,4 @@ performance on systems without hardware floating point support).

@code{X} means that input/output is supported.

-@section Timecode
-
-@multitable @columnfractions .4 .1 .1
-@item Codec/format      @tab Read   @tab Write
-@item DV                @tab X      @tab X
-@item GXF               @tab X      @tab X
-@item MOV               @tab X      @tab
-@item MPEG1/2           @tab X      @tab X
-@item MXF               @tab X      @tab X
-@end multitable
-
@bye
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -65,14 +65,6 @@ git clone git@@source.ffmpeg.org:ffmpeg <target>
 This will put the FFmpeg sources into the directory @var{<target>} and let
 you push back your changes to the remote repository.

-Make sure that you do not have Windows line endings in your checkouts,
-otherwise you may experience spurious compilation failures. One way to
-achieve this is to run
-
-@example
-git config --global core.autocrlf false
-@end example
-

@section Updating the source tree to the latest revision

--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -59,7 +59,7 @@ BSD video input device.

 Windows DirectShow input device.

-DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.
+DirectShow support is enabled when FFmpeg is built with mingw-w64.
 Currently only audio and video devices are supported.

 Multiple devices may be opened as separate inputs, but they may also be
@@ -196,12 +196,12 @@ device.
 Once you have created one or more JACK readable clients, you need to
 connect them to one or more JACK writable clients.

-To connect or disconnect JACK clients you can use the @command{jack_connect}
-and @command{jack_disconnect} programs, or do it through a graphical interface,
-for example with @command{qjackctl}.
+To connect or disconnect JACK clients you can use the
+@file{jack_connect} and @file{jack_disconnect} programs, or do it
+through a graphical interface, for example with @file{qjackctl}.

 To list the JACK clients and their properties you can invoke the command
-@command{jack_lsp}.
+@file{jack_lsp}.

 Follows an example which shows how to capture a JACK readable client
 with @command{ffmpeg}.
@@ -260,7 +260,7 @@ device.

@itemize
@item
-Create a color video stream and play it back with @command{ffplay}:
+Create a color video stream and play it back with @file{ffplay}:
@example
 ffplay -f lavfi -graph "color=pink [out0]" dummy
@end example
@@ -280,14 +280,14 @@ ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [ou

@item
 Read an audio stream from a file using the amovie source and play it
-back with @command{ffplay}:
+back with @file{ffplay}:
@example
 ffplay -f lavfi "amovie=test.wav"
@end example

@item
 Read an audio stream and a video stream and play it back with
-@command{ffplay}:
+@file{ffplay}:
@example
 ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
@end example
@@ -380,7 +380,7 @@ $ ffmpeg -f openal -i '' out.ogg
@end example

 Capture from two devices simultaneously, writing to two different files,
-within the same @command{ffmpeg} command:
+within the same @file{ffmpeg} command:
@example
 $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
@end example
@@ -415,7 +415,7 @@ The filename to provide to the input device is a source device or the
 string "default"

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

@example
 ffmpeg -f pulse -i default /tmp/pulse.wav
@@ -504,9 +504,9 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example

-@section video4linux2
+@section video4linux and video4linux2

-Video4Linux2 input video device.
+Video4Linux and Video4Linux2 input video devices.

 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
@@ -514,28 +514,36 @@ systems tend to automatically create such nodes when the device
 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 framerates. You can check which are
-supported using @command{-list_formats all} for Video4Linux2 devices.
+Video4Linux and Video4Linux2 devices only support a limited set of
+@var{width}x@var{height} sizes and frame rates. You can check which are
+supported for example with the command @file{dov4l} for Video4Linux
+devices and the command @file{v4l-info} for Video4Linux2 devices.

-Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
+If the size for the device is set to 0x0, the input device will
+try to auto-detect the size to use.
+Only for the video4linux2 device, if the frame rate is set to 0/0 the
+input device will use the frame rate value already set in the driver.

-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
-clock (origin at the Unix Epoch) or the monotonic clock (origin usually at
-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.
+Video4Linux support is deprecated since Linux 2.6.30, and will be
+dropped in later versions.

 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

-# 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
+Follow some usage examples of the video4linux devices with the ff*
+tools.
+@example
+# Grab and show the input of a video4linux device, frame rate is set
+# to the default of 25/1.
+ffplay -s 320x240 -f video4linux /dev/video0
+
+# Grab and show the input of a video4linux2 device, auto-adjust size.
+ffplay -f video4linux2 /dev/video0
+
+# Grab and record the input of a video4linux2 device, auto-adjust size,
+# frame rate value defaults to 0/0 so it is read from the video4linux2
+# driver.
+ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
@end example

 "v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
@@ -571,7 +579,7 @@ default to 0.

 Check the X11 documentation (e.g. man X) for more detailed information.

-Use the @command{dpyinfo} program for getting basic information about the
+Use the @file{dpyinfo} program for getting basic information about the
 properties of your X11 display (e.g. grep for "name" or "dimensions").

 For example to grab from @file{:0.0} using @command{ffmpeg}:
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -24,7 +24,7 @@ a mail for every change to every issue.
 The subscription URL for the ffmpeg-trac list is:
 http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-http(s)://ffmpeg.org/trac/ffmpeg
+http(s)://trac.ffmpeg.org

 Type:
 -----
--- a/doc/libavfilter.texi
+++ b/doc/libavfilter.texi
@@ -43,13 +43,13 @@ The result will be that in output the top half of the video is mirrored
 onto the bottom half.

 Video filters are loaded using the @var{-vf} option passed to
-@command{ffmpeg} or to @command{ffplay}. Filters in the same linear
-chain are separated by commas. In our example, @var{split, fifo,
-overlay} are in one linear chain, and @var{fifo, crop, vflip} are in
-another. The points where the linear chains join are labeled by names
-enclosed in square brackets. In our example, that is @var{[T1]} and
-@var{[T2]}. The magic labels @var{[in]} and @var{[out]} are the points
-where video is input and output.
+ffmpeg or to ffplay. Filters in the same linear chain are separated by
+commas. In our example, @var{split, fifo, overlay} are in one linear
+chain, and @var{fifo, crop, vflip} are in another. The points where
+the linear chains join are labeled by names enclosed in square
+brackets. In our example, that is @var{[T1]} and @var{[T2]}. The magic
+labels @var{[in]} and @var{[out]} are the points where video is input
+and output.

 Some filters take in input a list of parameters: they are specified
 after the filter name and an equal sign, and are separated each other
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -56,37 +56,31 @@ See also the @ref{framecrc} muxer.
@anchor{framecrc}
@section framecrc

-Per-packet CRC (Cyclic Redundancy Check) testing format.
+Per-frame CRC (Cyclic Redundancy Check) testing format.

-This muxer computes and prints the Adler-32 CRC for each audio
-and video packet. By default audio frames are converted to signed
+This muxer computes and prints the Adler-32 CRC for each decoded audio
+and video frame. By default audio frames are converted to signed
 16-bit raw audio and video frames to raw video before computing the
 CRC.

 The output of the muxer consists of a line for each audio and video
-packet of the form:
-@example
-@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC}
-@end example
+frame of the form: @var{stream_index}, @var{frame_dts},
+@var{frame_size}, 0x@var{CRC}, where @var{CRC} is a hexadecimal
+number 0-padded to 8 digits containing the CRC of the decoded frame.

-@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
-CRC of the packet.
-
-For example to compute the CRC of the audio and video frames in
-@file{INPUT}, converted to raw audio and video packets, and store it
-in the file @file{out.crc}:
+For example to compute the CRC of each decoded frame in the input, and
+store it in the file @file{out.crc}:
@example
 ffmpeg -i INPUT -f framecrc out.crc
@end example

-To print the information to stdout, use the command:
+You can print the CRC of each decoded frame to stdout with the command:
@example
 ffmpeg -i INPUT -f framecrc -
@end example

-With @command{ffmpeg}, you can select the output format to which the
-audio and video frames are encoded before computing the CRC for each
-packet by specifying the audio and video codec. For example, to
+You can select the output format of each frame with @command{ffmpeg} by
+specifying the audio and video codec and format. For example, to
 compute the CRC of each decoded input audio frame converted to PCM
 unsigned 8-bit and of each decoded input video frame converted to
 MPEG-2 video, use the command:
@@ -96,40 +90,6 @@ ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -

 See also the @ref{crc} muxer.

-@anchor{framemd5}
-@section framemd5
-
-Per-packet MD5 testing format.
-
-This muxer computes and prints the MD5 hash for each audio
-and video packet. By default audio frames are converted to signed
-16-bit raw audio and video frames to raw video before computing the
-hash.
-
-The output of the muxer consists of a line for each audio and video
-packet of the form:
-@example
-@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5}
-@end example
-
-@var{MD5} is a hexadecimal number representing the computed MD5 hash
-for the packet.
-
-For example to compute the MD5 of the audio and video frames in
-@file{INPUT}, converted to raw audio and video packets, and store it
-in the file @file{out.md5}:
-@example
-ffmpeg -i INPUT -f framemd5 out.md5
-@end example
-
-To print the information to stdout, use the command:
-@example
-ffmpeg -i INPUT -f framemd5 -
-@end example
-
-See also the @ref{md5} muxer.
-
-@anchor{image2}
@section image2

 Image file muxer.
@@ -187,104 +147,18 @@ 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.

-@anchor{md5}
-@section md5
+@section mov

-MD5 testing format.
+MOV / MP4 muxer

-This muxer computes and prints the MD5 hash of all the input audio
-and video frames. By default audio frames are converted to signed
-16-bit raw audio and video frames to raw video before computing the
-hash.
-
-The output of the muxer consists of a single line of the form:
-MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing
-the computed MD5 hash.
-
-For example to compute the MD5 hash of the input converted to raw
-audio and video, and store it in the file @file{out.md5}:
-@example
-ffmpeg -i INPUT -f md5 out.md5
-@end example
-
-You can print the MD5 to stdout with the command:
-@example
-ffmpeg -i INPUT -f md5 -
-@end example
-
-See also the @ref{framemd5} muxer.
-
-@section MOV/MP4/ISMV
-
-The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
-file has all the metadata about all packets stored in one location
-(written at the end of the file, it can be moved to the start for
-better playback using the @command{qt-faststart} tool). A fragmented
-file consists of a number of fragments, where packets and metadata
-about these packets are stored together. Writing a fragmented
-file has the advantage that the file is decodable even if the
-writing is interrupted (while a normal MOV/MP4 is undecodable if
-it is not properly finished), and it requires less memory when writing
-very long files (since writing normal MOV/MP4 files stores info about
-every single packet in memory until the file is closed). The downside
-is that it is less compatible with other applications.
-
-Fragmentation is enabled by setting one of the AVOptions that define
-how to cut the file into fragments:
+The muxer options are:

@table @option
@item -moov_size @var{bytes}
 Reserves space for the moov atom at the beginning of the file instead of placing the
 moov atom at the end. If the space reserved is insufficient, muxing will fail.
-@item -movflags frag_keyframe
-Start a new fragment at each video keyframe.
-@item -frag_duration @var{duration}
-Create fragments that are @var{duration} microseconds long.
-@item -frag_size @var{size}
-Create fragments that contain up to @var{size} bytes of payload data.
-@item -movflags frag_custom
-Allow the caller to manually choose when to cut fragments, by
-calling @code{av_write_frame(ctx, NULL)} to write a fragment with
-the packets written so far. (This is only useful with other
-applications integrating libavformat, not from @command{ffmpeg}.)
-@item -min_frag_duration @var{duration}
-Don't create fragments that are shorter than @var{duration} microseconds long.
@end table

-If more than one condition is specified, fragments are cut when
-one of the specified conditions is fulfilled. The exception to this is
-@code{-min_frag_duration}, which has to be fulfilled for any of the other
-conditions to apply.
-
-Additionally, the way the output file is written can be adjusted
-through a few other options:
-
-@table @option
-@item -movflags empty_moov
-Write an initial moov atom directly at the start of the file, without
-describing any samples in it. Generally, an mdat/moov pair is written
-at the start of the file, as a normal MOV/MP4 file, containing only
-a short portion of the file. With this option set, there is no initial
-mdat atom, and the moov atom only describes the tracks but has
-a zero duration.
-
-Files written with this option set do not work in QuickTime.
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -movflags separate_moof
-Write a separate moof (movie fragment) atom for each track. Normally,
-packets for all tracks are written in a moof atom (which is slightly
-more efficient), but with this option set, the muxer writes one moof/mdat
-pair for each track, making it easier to separate tracks.
-
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@end table
-
-Smooth Streaming content can be pushed in real time to a publishing
-point on IIS with this muxer. Example:
-@example
-ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
-@end example
-
@section mpegts

 MPEG transport stream muxer.
@@ -411,71 +285,4 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
@end example

-@section segment
-
-Basic stream segmenter.
-
-The segmenter muxer outputs streams to a number of separate files of nearly
-fixed duration. Output filename pattern can be set in a fashion similar to
-@ref{image2}.
-
-Every segment starts with a video keyframe, if a video stream is present.
-The segment muxer works best with a single constant frame rate video.
-
-Optionally it can generate a flat list of the created segments, one segment
-per line.
-
-@table @option
-@item segment_format @var{format}
-Override the inner container format, by default it is guessed by the filename
-extension.
-@item segment_time @var{t}
-Set segment duration to @var{t} seconds.
-@item segment_list @var{name}
-Generate also a listfile named @var{name}.
-@item segment_list_size @var{size}
-Overwrite the listfile once it reaches @var{size} entries.
-@item segment_wrap @var{limit}
-Wrap around segment index once it reaches @var{limit}.
-@end table
-
-@example
-ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
-@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
-
-Attach a picture to an mp3:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover"
-metadata:s:v comment="Cover (Front)" out.mp3
-@end example
-
@c man end MUXERS
--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -60,7 +60,7 @@ If not specified it defaults to the size of the input video.

@subsection Examples

-The following command shows the @command{ffmpeg} output is an
+The following command shows the @file{ffmpeg} output is an
 SDL window, forcing its size to the qcif format:
@example
 ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -27,11 +27,11 @@ to configure.
@section BSD

 BSD make will not build FFmpeg, you need to install and use GNU Make
-(@command{gmake}).
+(@file{gmake}).

@section (Open)Solaris

-GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}),
+GNU Make is required to build FFmpeg, so you have to invoke (@file{gmake}),
 standard Solaris Make will not work. When building with a non-c99 front-end
 (gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
 or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
@@ -89,7 +89,7 @@ section and the FAQ.

 FFmpeg does not build out-of-the-box with the packages the automated MinGW
 installer provides. It also requires coreutils to be installed and many other
-packages updated to the latest version. The minimum versions for some packages
+packages updated to the latest version. The minimum version for some packages
 are listed below:

@itemize
@@ -109,11 +109,14 @@ Notes:
@item Building natively using MSYS can be sped up by disabling implicit rules
 in the Makefile by calling @code{make -r} instead of plain @code{make}. This
 speed up is close to non-existent for normal one-off builds and is only
-noticeable when running make for a second time (for example during
+noticeable when running make for a second time (for example in
@code{make install}).

@item In order to compile FFplay, you must have the MinGW development library
-of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed.
+of @uref{http://www.libsdl.org/, SDL}.
+Edit the @file{bin/sdl-config} script so that it points to the correct prefix
+where SDL was installed. Verify that @file{sdl-config} can be launched from
+the MSYS command line.

@item By using @code{./configure --enable-shared} when configuring FFmpeg,
 you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
@@ -137,7 +140,7 @@ you might have to modify the procedures slightly.

@subsection Using static libraries

-Assuming you have just built and installed FFmpeg in @file{/usr/local}:
+Assuming you have just built and installed FFmpeg in @file{/usr/local}.

@enumerate

@@ -232,8 +235,6 @@ make install
 Your install path (@file{/usr/local/} by default) should now have the
 necessary DLL and LIB files under the @file{bin} directory.

-@end enumerate
-
 Alternatively, build the libraries with a cross compiler, according to
 the instructions below in @ref{Cross compilation for Windows with Linux}.

@@ -272,7 +273,7 @@ To create import libraries that work with the @code{/OPT:REF} option

@enumerate

-@item Open @emph{Visual Studio 2005 Command Prompt}.
+@item Open @file{Visual Studio 2005 Command Prompt}.

 Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
 which sets up the environment variables for the Visual C++ tools
@@ -282,14 +283,17 @@ which sets up the environment variables for the Visual C++ tools
@item Enter the @file{bin} directory where the created LIB and DLL files
 are stored.

-@item Generate new import libraries with @command{lib.exe}:
+@item Generate new import libraries with @file{lib.exe}:

@example
-lib /machine:i386 /def:..\lib\foo-version.def  /out:foo.lib
+lib /machine:i386 /def:..\lib\avcodec-53.def  /out:avcodec.lib
+lib /machine:i386 /def:..\lib\avdevice-53.def /out:avdevice.lib
+lib /machine:i386 /def:..\lib\avfilter-2.def  /out:avfilter.lib
+lib /machine:i386 /def:..\lib\avformat-53.def /out:avformat.lib
+lib /machine:i386 /def:..\lib\avutil-51.def   /out:avutil.lib
+lib /machine:i386 /def:..\lib\swscale-2.def   /out:swscale.lib
@end example

-Replace @code{foo-version} and @code{foo} with the respective library names.
-
@end enumerate

@anchor{Cross compilation for Windows with Linux}
@@ -331,8 +335,8 @@ Then run

 to make a static build.

-To build shared libraries add a special compiler flag to work around current
-@code{gcc4-core} package bugs in addition to the normal configure flags:
+The current @code{gcc4-core} package is buggy and needs this flag to build
+shared libraries:

@example
 ./configure --enable-shared --disable-static --extra-cflags=-fno-reorder-functions
@@ -348,12 +352,16 @@ These library packages are only available from
@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:

@example
-yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
-libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
+yasm, libSDL-devel, libdirac-devel, libfaac-devel, libaacplus-devel, libgsm-devel,
+libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel,
+libxvidcore-devel
@end example

-The recommendation for x264 is to build it from source, as it evolves too
-quickly for Cygwin Ports to be up to date.
+The recommendation for libnut and x264 is to build them from source by
+yourself, as they evolve too quickly for Cygwin Ports to be up to date.
+
+Cygwin 1.7.x has IPv6 support. You can add IPv6 to Cygwin 1.5.x by means
+of the @code{libgetaddrinfo-devel} package, available at Cygwin Ports.

@section Crosscompilation for Windows under Cygwin

--- a/doc/print_options.c
+++ b/doc/print_options.c
@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2012 Anton Khirnov
- *
- * This file is part of Libav.
- *
- * 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.
- *
- * 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * generate texinfo manpages for avoptions
- */
-
-#include <stddef.h>
-#include <string.h>
-#include <float.h>
-
-#include "libavformat/avformat.h"
-#include "libavcodec/avcodec.h"
-#include "libavutil/opt.h"
-
-static void print_usage(void)
-{
-    fprintf(stderr, "Usage: enum_options type\n"
-            "type: format codec\n");
-    exit(1);
-}
-
-static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
-{
-    printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
-    switch (o->type) {
-    case AV_OPT_TYPE_BINARY:   printf("hexadecimal string"); break;
-    case AV_OPT_TYPE_STRING:   printf("string");             break;
-    case AV_OPT_TYPE_INT:
-    case AV_OPT_TYPE_INT64:    printf("integer");            break;
-    case AV_OPT_TYPE_FLOAT:
-    case AV_OPT_TYPE_DOUBLE:   printf("float");              break;
-    case AV_OPT_TYPE_RATIONAL: printf("rational number");    break;
-    case AV_OPT_TYPE_FLAGS:    printf("flags");              break;
-    default:                   printf("value");              break;
-    }
-    printf("} (@emph{");
-
-    if (o->flags & AV_OPT_FLAG_DECODING_PARAM) {
-        printf("input");
-        if (o->flags & AV_OPT_FLAG_ENCODING_PARAM)
-            printf("/");
-    }
-    if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) printf("output");
-    if (o->flags & AV_OPT_FLAG_AUDIO_PARAM)    printf(",audio");
-    if (o->flags & AV_OPT_FLAG_VIDEO_PARAM)    printf(",video");
-    if (o->flags & AV_OPT_FLAG_SUBTITLE_PARAM) printf(",subtitles");
-
-    printf("})\n");
-    if (o->help)
-        printf("%s\n", o->help);
-
-    if (o->unit) {
-        const AVOption *u;
-        printf("\nPossible values:\n@table @samp\n");
-
-        for (u = opts; u->name; u++) {
-            if (u->type == AV_OPT_TYPE_CONST && u->unit && !strcmp(u->unit, o->unit))
-                printf("@item %s\n%s\n", u->name, u->help ? u->help : "");
-        }
-        printf("@end table\n");
-    }
-}
-
-static void show_opts(const AVOption *opts, int per_stream)
-{
-    const AVOption *o;
-
-    printf("@table @option\n");
-    for (o = opts; o->name; o++) {
-        if (o->type != AV_OPT_TYPE_CONST)
-            print_option(opts, o, per_stream);
-    }
-    printf("@end table\n");
-}
-
-static void show_format_opts(void)
-{
-#include "libavformat/options_table.h"
-
-    printf("@section Format AVOptions\n");
-    show_opts(options, 0);
-}
-
-static void show_codec_opts(void)
-{
-#include "libavcodec/options_table.h"
-
-    printf("@section Codec AVOptions\n");
-    show_opts(options, 1);
-}
-
-int main(int argc, char **argv)
-{
-    if (argc < 2)
-        print_usage();
-
-    printf("@c DO NOT EDIT THIS FILE!\n"
-           "@c It was generated by print_options.\n\n");
-    if (!strcmp(argv[1], "format"))
-        show_format_opts();
-    else if (!strcmp(argv[1], "codec"))
-        show_codec_opts();
-    else
-        print_usage();
-
-    return 0;
-}
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -19,34 +19,20 @@ supported protocols.

 A description of the currently available protocols follows.

-@section bluray
+@section applehttp

-Read BluRay playlist.
+Read Apple HTTP Live Streaming compliant segmented stream as
+a uniform one. The M3U8 playlists describing the segments can be
+remote HTTP resources or local files, accessed using the standard
+file protocol.
+HTTP is default, specific protocol can be declared by specifying
+"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
+is either "file" or "http".

-The accepted options are:
-@table @option
-
-@item angle
-BluRay angle
-
-@item chapter
-Start chapter (1...N)
-
-@item playlist
-Playlist to read (BDMV/PLAYLIST/?????.mpls)
-
-@end table
-
-Examples:
-
-Read longest playlist from BluRay mounted to /mnt/bluray:
@example
-bluray:/mnt/bluray
-@end example
-
-Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
-@example
-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
+applehttp://host/path/to/remote/resource.m3u8
+applehttp+http://host/path/to/remote/resource.m3u8
+applehttp+file://path/to/local/resource.m3u8
@end example

@section concat
@@ -66,7 +52,7 @@ resource to be concatenated, each one possibly specifying a distinct
 protocol.

 For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
+@file{split2.mpeg}, @file{split3.mpeg} with @file{ffplay} use the
 command:
@example
 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@@ -95,26 +81,6 @@ specified with the name "FILE.mpeg" is interpreted as the URL

 Gopher protocol.

-@section hls
-
-Read Apple HTTP Live Streaming compliant segmented stream as
-a uniform one. The M3U8 playlists describing the segments can be
-remote HTTP resources or local files, accessed using the standard
-file protocol.
-The nested protocol is declared by specifying
-"+@var{proto}" after the hls URI scheme name, where @var{proto}
-is either "file" or "http".
-
-@example
-hls+http://host/path/to/remote/resource.m3u8
-hls+file://path/to/local/resource.m3u8
-@end example
-
-Using this protocol is discouraged - the hls demuxer should work
-just as well (if not, please report the issues) and is more complete.
-To use the hls demuxer instead, simply use the direct URLs to the
-m3u8 files.
-
@section http

 HTTP (Hyper Text Transfer Protocol).
@@ -189,8 +155,8 @@ be seekable, so they will fail with the pipe output protocol.

 Real-Time Messaging Protocol.

-The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
-content across a TCP/IP network.
+The Real-Time Messaging Protocol (RTMP) is used for streaming
+multimedia content across a TCP/IP network.

 The required syntax is:
@example
@@ -217,7 +183,7 @@ application specified in @var{app}, may be prefixed by "mp4:".

@end table

-For example to read with @command{ffplay} a multimedia resource named
+For example to read with @file{ffplay} a multimedia resource named
 "sample" from the application "vod" from an RTMP server "myserver":
@example
 ffplay rtmp://myserver/vod/sample
@@ -258,7 +224,7 @@ For example, to stream a file in real-time to an RTMP server using
 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
@end example

-To play the same stream using @command{ffplay}:
+To play the same stream using @file{ffplay}:
@example
 ffplay "rtmp://myserver/live/mystream live=1"
@end example
@@ -283,7 +249,7 @@ The required syntax for a RTSP url is:
 rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example

-The following options (set on the @command{ffmpeg}/@command{ffplay} command
+The following options (set on the @command{ffmpeg}/@file{ffplay} command
 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
 are supported:

@@ -318,11 +284,11 @@ Accept packets only from negotiated peer address and port.
@end table

 When receiving data over UDP, the demuxer tries to reorder received packets
-(since they may arrive out of order, or packets may get lost totally). This
-can be disabled by setting the maximum demuxing delay to zero (via
-the @code{max_delay} field of AVFormatContext).
+(since they may arrive out of order, or packets may get lost totally). In
+order for this to be enabled, a maximum delay must be specified in the
+@code{max_delay} field of AVFormatContext.

-When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
+When watching multi-bitrate Real-RTSP streams with @file{ffplay}, the
 streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.
@@ -399,13 +365,13 @@ To broadcast a stream on the local subnet, for watching in VLC:
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example

-Similarly, for watching in @command{ffplay}:
+Similarly, for watching in ffplay:

@example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
@end example

-And for watching in @command{ffplay}, over IPv6:
+And for watching in ffplay, over IPv6:

@example
 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
--- a/doc/soc.txt
+++ b/doc/soc.txt
@@ -18,7 +18,7 @@ essential that changes to their codebase are publicly visible, clean and
 easy reviewable that again leads us to:
 * use of a revision control system like git
 * separation of cosmetic from non-cosmetic changes (this is almost entirely
-  ignored by mentors and students in soc 2006 which might lead to a surprise
+  ignored by mentors and students in soc 2006 which might lead to a suprise
  when the code will be reviewed at the end before a possible inclusion in
  FFmpeg, individual changes were generally not reviewable due to cosmetics).
 * frequent commits, so that comments can be provided early
--- a/doc/swresample.txt
+++ b/doc/swresample.txt
@@ -32,7 +32,7 @@ Special Converter         v
                        Output

 Planar/Packed convertion is done when needed during sample format convertion
-Every step can be skipped without memcpy when its not needed.
+Every step can be skiped without memcpy when its not needed.
 Either Resampling and Rematrixing can be performed first depending on which
 way its faster.
 The Buffers are needed for resampling due to resamplng being a process that
--- a/doc/swscale.txt
+++ b/doc/swscale.txt
@@ -96,3 +96,4 @@ would benefit from it.
 Also, as already hinted at, initFilter() accepts an optional convolutional
 filter as input that can be used for contrast, saturation, blur, sharpening
 shift, chroma vs. luma shift, ...
+
--- a/doc/syntax.texi
+++ b/doc/syntax.texi
@@ -1,158 +0,0 @@
-@chapter Syntax
-@c man begin SYNTAX
-
-When evaluating specific formats, FFmpeg uses internal library parsing
-functions, shared by the tools. This section documents the syntax of
-some of these formats.
-
-@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/texi2pod.pl
+++ b/doc/texi2pod.pl
@@ -36,7 +36,7 @@ $shift = "";
 %defs = ();
 $fnno = 1;
 $inf = "";
-@ibase = ();
+$ibase = "";

 while ($_ = shift) {
    if (/^-D(.*)$/) {
@@ -52,8 +52,6 @@ while ($_ = shift) {
        die "flags may only contain letters, digits, hyphens, dashes and underscores\n"
            unless $flag =~ /^[a-zA-Z0-9_-]+$/;
        $defs{$flag} = $value;
-    } elsif (/^-I(.*)$/) {
-        push @ibase, $1 ne "" ? $1 : shift;
    } elsif (/^-/) {
        usage();
    } else {
@@ -63,12 +61,10 @@ while ($_ = shift) {
    }
 }

-push @ibase, ".";
-
 if (defined $in) {
    $inf = gensym();
    open($inf, "<$in") or die "opening \"$in\": $!\n";
-    push @ibase, $1 if $in =~ m|^(.+)/[^/]+$|;
+    $ibase = $1 if $in =~ m|^(.+)/[^/]+$|;
 } else {
    $inf = \*STDIN;
 }
@@ -78,7 +74,7 @@ if (defined $out) {
 }

 while(defined $inf) {
-INF: while(<$inf>) {
+while(<$inf>) {
    # Certain commands are discarded without further processing.
    /^\@(?:
         [a-z]+index            # @*index: useful only in complete manual
@@ -108,10 +104,11 @@ INF: while(<$inf>) {
        push @instack, $inf;
        $inf = gensym();

-        for (@ibase) {
-            open($inf, "<" . $_ . "/" . $1) and next INF;
-        }
-        die "cannot open $1: $!\n";
+        # Try cwd and $ibase.
+        open($inf, "<" . $1)
+            or open($inf, "<" . $ibase . "/" . $1)
+                or die "cannot open $1 or $ibase/$1: $!\n";
+        next;
    };

    # Look for blocks surrounded by @c man begin SECTION ... @c man end.
--- a/doc/viterbi.txt
+++ b/doc/viterbi.txt
@@ -107,3 +107,4 @@ one with score 3)

 Author: Michael niedermayer
 Copyright LGPL
+
--- a/ffmpeg.c
+++ b/ffmpeg.c
--- a/ffplay.c
+++ b/ffplay.c
--- a/ffpresets/libvpx-1080p.ffpreset
+++ b/ffpresets/libvpx-1080p.ffpreset
@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=1
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3
--- a/ffpresets/libvpx-1080p50_60.ffpreset
+++ b/ffpresets/libvpx-1080p50_60.ffpreset
@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=25
+quality=good
+speed=0
+profile=1
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3
--- a/ffpresets/libvpx-360p.ffpreset
+++ b/ffpresets/libvpx-360p.ffpreset
@@ -0,0 +1,16 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=0
+qmax=63
+qmin=0
+vb=768k
+
+#ignored unless using -pass 2
+maxrate=1.5M
+minrate=40k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3
--- a/ffpresets/libvpx-720p.ffpreset
+++ b/ffpresets/libvpx-720p.ffpreset
@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=0
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3
--- a/ffpresets/libvpx-720p50_60.ffpreset
+++ b/ffpresets/libvpx-720p50_60.ffpreset
@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=25
+quality=good
+speed=0
+profile=0
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3
--- a/ffprobe.c
+++ b/ffprobe.c
--- a/ffserver.c
+++ b/ffserver.c
@@ -1,4 +1,5 @@
 /*
+ * Multiple format streaming server
 * Copyright (c) 2000, 2001, 2002 Fabrice Bellard
 *
 * This file is part of FFmpeg.
@@ -18,11 +19,6 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

-/**
- * @file
- * multiple format streaming server based on the FFmpeg libraries
- */
-
 #include "config.h"
 #if !HAVE_CLOSESOCKET
 #define closesocket close
@@ -30,16 +26,13 @@
 #include <string.h>
 #include <stdlib.h>
 #include "libavformat/avformat.h"
-// FIXME those are internal headers, ffserver _really_ shouldn't use them
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
 #include "libavformat/rtpdec.h"
 #include "libavformat/rtsp.h"
+// XXX for ffio_open_dyn_packet_buffer, to be removed
 #include "libavformat/avio_internal.h"
-#include "libavformat/internal.h"
-#include "libavformat/url.h"
-
 #include "libavutil/avstring.h"
 #include "libavutil/lfg.h"
 #include "libavutil/dict.h"
@@ -339,7 +332,8 @@ static int resolve_host(struct in_addr *sin_addr, const char *hostname)
    if (!ff_inet_aton(hostname, sin_addr)) {
 #if HAVE_GETADDRINFO
        struct addrinfo *ai, *cur;
-        struct addrinfo hints = { 0 };
+        struct addrinfo hints;
+        memset(&hints, 0, sizeof(hints));
        hints.ai_family = AF_INET;
        if (getaddrinfo(hostname, NULL, &hints, &ai))
            return -1;
@@ -481,7 +475,7 @@ static void start_children(FFStream *feed)
                    slash++;
                strcpy(slash, "ffmpeg");

-                http_log("Launch command line: ");
+                http_log("Launch commandline: ");
                http_log("%s ", pathname);
                for (i = 1; feed->child_argv[i] && feed->child_argv[i][0]; i++)
                    http_log("%s ", feed->child_argv[i]);
@@ -501,10 +495,7 @@ static void start_children(FFStream *feed)
                }

                /* This is needed to make relative pathnames work */
-                if (chdir(my_program_dir) < 0) {
-                    http_log("chdir failed\n");
-                    exit(1);
-                }
+                chdir(my_program_dir);

                signal(SIGPIPE, SIG_DFL);

@@ -858,7 +849,7 @@ static void close_connection(HTTPContext *c)
            if (st->codec->codec)
                avcodec_close(st->codec);
        }
-        avformat_close_input(&c->fmt_in);
+        av_close_input_file(c->fmt_in);
    }

    /* free RTP output streams if any */
@@ -876,7 +867,7 @@ static void close_connection(HTTPContext *c)
        }
        h = c->rtp_handles[i];
        if (h)
-            ffurl_close(h);
+            url_close(h);
    }

    ctx = &c->fmt_ctx;
@@ -1871,7 +1862,7 @@ static int http_parse_request(HTTPContext *c)

 static void fmt_bytecount(AVIOContext *pb, int64_t count)
 {
-    static const char suffix[] = " kMGTP";
+    static const char *suffix = " kMGTP";
    const char *s;

    for (s = suffix; count >= 100000 && s[1]; count /= 1000, s++);
@@ -2121,18 +2112,33 @@ static void compute_status(HTTPContext *c)
    c->buffer_end = c->pb_buffer + len;
 }

+/* check if the parser needs to be opened for stream i */
+static void open_parser(AVFormatContext *s, int i)
+{
+    AVStream *st = s->streams[i];
+    AVCodec *codec;
+
+    if (!st->codec->codec) {
+        codec = avcodec_find_decoder(st->codec->codec_id);
+        if (codec && (codec->capabilities & CODEC_CAP_PARSE_ONLY)) {
+            st->codec->parse_only = 1;
+            if (avcodec_open2(st->codec, codec, NULL) < 0)
+                st->codec->parse_only = 0;
+        }
+    }
+}
+
 static int open_input_stream(HTTPContext *c, const char *info)
 {
    char buf[128];
    char input_filename[1024];
    AVFormatContext *s = NULL;
-    int buf_size, i, ret;
+    int i, ret;
    int64_t stream_pos;

    /* find file name */
    if (c->stream->feed) {
        strcpy(input_filename, c->stream->feed->feed_filename);
-        buf_size = FFM_PACKET_SIZE;
        /* compute position (absolute time) */
        if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
            if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0)
@@ -2144,7 +2150,6 @@ static int open_input_stream(HTTPContext *c, const char *info)
            stream_pos = av_gettime() - c->stream->prebuffer * (int64_t)1000;
    } else {
        strcpy(input_filename, c->stream->feed_filename);
-        buf_size = 0;
        /* compute position (relative time) */
        if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
            if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0)
@@ -2160,18 +2165,18 @@ static int open_input_stream(HTTPContext *c, const char *info)
        http_log("could not open %s: %d\n", input_filename, ret);
        return -1;
    }
-
-    /* set buffer size */
-    if (buf_size > 0) ffio_set_buf_size(s->pb, buf_size);
-
    s->flags |= AVFMT_FLAG_GENPTS;
    c->fmt_in = s;
    if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) {
        http_log("Could not find stream info '%s'\n", input_filename);
-        avformat_close_input(&s);
+        av_close_input_file(s);
        return -1;
    }

+    /* open each parser */
+    for(i=0;i<s->nb_streams;i++)
+        open_parser(s, i);
+
    /* choose stream as clock source (we favorize video stream if
       present) for packet sending */
    c->pts_stream_index = 0;
@@ -2263,6 +2268,7 @@ static int http_prepare_data(HTTPContext *c)
         * Default value from FFmpeg
         * Try to set it use configuration option
         */
+        c->fmt_ctx.preload   = (int)(0.5*AV_TIME_BASE);
        c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);

        if (avformat_write_header(&c->fmt_ctx, NULL) < 0) {
@@ -2305,7 +2311,8 @@ static int http_prepare_data(HTTPContext *c)
                    return 0;
                } else {
                    if (c->stream->loop) {
-                        avformat_close_input(&c->fmt_in);
+                        av_close_input_file(c->fmt_in);
+                        c->fmt_in = NULL;
                        if (open_input_stream(c, "") < 0)
                            goto no_loop;
                        goto redo;
@@ -2381,7 +2388,7 @@ static int http_prepare_data(HTTPContext *c)
                        if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP)
                            max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
                        else
-                            max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
+                            max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
                        ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
                    } else {
                        ret = avio_open_dyn_buf(&ctx->pb);
@@ -2534,8 +2541,8 @@ static int http_send_data(HTTPContext *c)
                } else {
                    /* send RTP packet directly in UDP */
                    c->buffer_ptr += 4;
-                    ffurl_write(c->rtp_handles[c->packet_stream_index],
-                                c->buffer_ptr, len);
+                    url_write(c->rtp_handles[c->packet_stream_index],
+                              c->buffer_ptr, len);
                    c->buffer_ptr += len;
                    /* here we continue as we can send several packets per 10 ms slot */
                }
@@ -2585,11 +2592,8 @@ static int http_start_receive_data(HTTPContext *c)
    if (c->stream->truncate) {
        /* truncate feed file */
        ffm_write_write_index(c->feed_fd, FFM_PACKET_SIZE);
+        ftruncate(c->feed_fd, FFM_PACKET_SIZE);
        http_log("Truncating feed file '%s'\n", c->stream->feed_filename);
-        if (ftruncate(c->feed_fd, FFM_PACKET_SIZE) < 0) {
-            http_log("Error truncating feed file: %s\n", strerror(errno));
-            return -1;
-        }
    } else {
        if ((c->stream->feed_write_index = ffm_read_write_index(fd)) < 0) {
            http_log("Error reading write index from feed file: %s\n", strerror(errno));
@@ -2732,7 +2736,7 @@ static int http_receive_data(HTTPContext *c)

            /* Now we have the actual streams */
            if (s->nb_streams != feed->nb_streams) {
-                avformat_close_input(&s);
+                av_close_input_stream(s);
                av_free(pb);
                http_log("Feed '%s' stream number does not match registered feed\n",
                         c->stream->feed_filename);
@@ -2745,7 +2749,7 @@ static int http_receive_data(HTTPContext *c)
                avcodec_copy_context(fst->codec, st->codec);
            }

-            avformat_close_input(&s);
+            av_close_input_stream(s);
            av_free(pb);
        }
        c->buffer_ptr = c->buffer;
@@ -2837,7 +2841,7 @@ static int rtsp_parse_request(HTTPContext *c)
    char protocol[32];
    char line[1024];
    int len;
-    RTSPMessageHeader header1 = { 0 }, *header = &header1;
+    RTSPMessageHeader header1, *header = &header1;

    c->buffer_ptr[0] = '\0';
    p = c->buffer;
@@ -2863,6 +2867,7 @@ static int rtsp_parse_request(HTTPContext *c)
    }

    /* parse each header line */
+    memset(header, 0, sizeof(*header));
    /* skip to next line */
    while (*p != '\n' && *p != '\0')
        p++;
@@ -3420,10 +3425,10 @@ static int rtp_new_av_stream(HTTPContext *c,
                     "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
        }

-        if (ffurl_open(&h, ctx->filename, AVIO_FLAG_WRITE, NULL, NULL) < 0)
+        if (url_open(&h, ctx->filename, AVIO_FLAG_WRITE) < 0)
            goto fail;
        c->rtp_handles[stream_index] = h;
-        max_packet_size = h->max_packet_size;
+        max_packet_size = url_get_max_packet_size(h);
        break;
    case RTSP_LOWER_TRANSPORT_TCP:
        /* RTP/TCP case */
@@ -3446,7 +3451,7 @@ static int rtp_new_av_stream(HTTPContext *c,
    if (avformat_write_header(ctx, NULL) < 0) {
    fail:
        if (h)
-            ffurl_close(h);
+            url_close(h);
        av_free(ctx);
        return -1;
    }
@@ -3483,7 +3488,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
    }
    fst->priv_data = av_mallocz(sizeof(FeedData));
    fst->index = stream->nb_streams;
-    avpriv_set_pts_info(fst, 33, 1, 90000);
+    av_set_pts_info(fst, 33, 1, 90000);
    fst->sample_aspect_ratio = codec->sample_aspect_ratio;
    stream->streams[stream->nb_streams++] = fst;
    return fst;
@@ -3624,7 +3629,7 @@ static void build_file_streams(void)
                if (avformat_find_stream_info(infile, NULL) < 0) {
                    http_log("Could not find codec parameters from '%s'\n",
                             stream->feed_filename);
-                    avformat_close_input(&infile);
+                    av_close_input_file(infile);
                    goto fail;
                }
                extract_mpeg4_header(infile);
@@ -3632,7 +3637,7 @@ static void build_file_streams(void)
                for(i=0;i<infile->nb_streams;i++)
                    add_av_stream1(stream, infile->streams[i]->codec, 1);

-                avformat_close_input(&infile);
+                av_close_input_file(infile);
            }
        }
    }
@@ -3669,8 +3674,6 @@ static void build_feed_streams(void)
            int matches = 0;

            if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) >= 0) {
-                /* set buffer size */
-                ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
                /* Now see if it matches */
                if (s->nb_streams == feed->nb_streams) {
                    matches = 1;
@@ -3724,7 +3727,7 @@ static void build_feed_streams(void)
                    http_log("Deleting feed file '%s' as stream counts differ (%d != %d)\n",
                        feed->feed_filename, s->nb_streams, feed->nb_streams);

-                avformat_close_input(&s);
+                av_close_input_file(s);
            } else
                http_log("Deleting feed file '%s' as it appears to be corrupt\n",
                        feed->feed_filename);
@@ -4230,8 +4233,8 @@ static int parse_ffconfig(const char *filename)
                }

                stream->fmt = ffserver_guess_format(NULL, stream->filename, NULL);
-                avcodec_get_context_defaults3(&video_enc, NULL);
-                avcodec_get_context_defaults3(&audio_enc, NULL);
+                avcodec_get_context_defaults2(&video_enc, AVMEDIA_TYPE_VIDEO);
+                avcodec_get_context_defaults2(&audio_enc, AVMEDIA_TYPE_AUDIO);

                audio_id = CODEC_ID_NONE;
                video_id = CODEC_ID_NONE;
@@ -4665,7 +4668,7 @@ static const OptionDef options[] = {

 int main(int argc, char **argv)
 {
-    struct sigaction sigact = { { 0 } };
+    struct sigaction sigact;

    parse_loglevel(argc, argv, options);
    av_register_all();
@@ -4683,6 +4686,7 @@ int main(int argc, char **argv)

    av_lfg_init(&random_state, av_get_random_seed());

+    memset(&sigact, 0, sizeof(sigact));
    sigact.sa_handler = handle_child_exit;
    sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART;
    sigaction(SIGCHLD, &sigact, 0);
--- a/libavcodec/4xm.c
+++ b/libavcodec/4xm.c
--- a/libavcodec/8bps.c
+++ b/libavcodec/8bps.c
@@ -38,20 +38,20 @@
 #include "avcodec.h"


-static const enum PixelFormat pixfmt_rgb24[] = {
-    PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE };
+static const enum PixelFormat pixfmt_rgb24[] = {PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE};

 /*
 * Decoder context
 */
 typedef struct EightBpsContext {
-    AVCodecContext *avctx;
-    AVFrame pic;

-    unsigned char planes;
-    unsigned char planemap[4];
+        AVCodecContext *avctx;
+        AVFrame pic;

-    uint32_t pal[256];
+        unsigned char planes;
+        unsigned char planemap[4];
+
+        uint32_t pal[256];
 } EightBpsContext;


@@ -60,90 +60,87 @@ typedef struct EightBpsContext {
 * Decode a frame
 *
 */
-static int decode_frame(AVCodecContext *avctx, void *data,
-                        int *data_size, AVPacket *avpkt)
+static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt)
 {
-    const uint8_t *buf = avpkt->data;
-    int buf_size       = avpkt->size;
-    EightBpsContext * const c = avctx->priv_data;
-    const unsigned char *encoded = buf;
-    unsigned char *pixptr, *pixptr_end;
-    unsigned int height = avctx->height; // Real image height
-    unsigned int dlen, p, row;
-    const unsigned char *lp, *dp;
-    unsigned char count;
-    unsigned int planes     = c->planes;
-    unsigned char *planemap = c->planemap;
+        const uint8_t *buf = avpkt->data;
+        int buf_size = avpkt->size;
+        EightBpsContext * const c = avctx->priv_data;
+        const unsigned char *encoded = buf;
+        unsigned char *pixptr, *pixptr_end;
+        unsigned int height = avctx->height; // Real image height
+        unsigned int dlen, p, row;
+        const unsigned char *lp, *dp;
+        unsigned char count;
+        unsigned int planes = c->planes;
+        unsigned char *planemap = c->planemap;

-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
+        if(c->pic.data[0])
+                avctx->release_buffer(avctx, &c->pic);

-    c->pic.reference    = 0;
-    c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
-    if (avctx->get_buffer(avctx, &c->pic) < 0){
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-        return -1;
-    }
+        c->pic.reference = 0;
+        c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
+        if(avctx->get_buffer(avctx, &c->pic) < 0){
+                av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+                return -1;
+        }

-    /* Set data pointer after line lengths */
-    dp = encoded + planes * (height << 1);
+        /* Set data pointer after line lengths */
+        dp = encoded + planes * (height << 1);

-    for (p = 0; p < planes; p++) {
-        /* Lines length pointer for this plane */
-        lp = encoded + p * (height << 1);
+        for (p = 0; p < planes; p++) {
+                /* Lines length pointer for this plane */
+                lp = encoded + p * (height << 1);

-        /* Decode a plane */
-        for (row = 0; row < height; row++) {
-            pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
-            pixptr_end = pixptr + c->pic.linesize[0];
-            dlen = av_be2ne16(*(const unsigned short *)(lp + row * 2));
-            /* Decode a row of this plane */
-            while (dlen > 0) {
-                if (dp + 1 >= buf + buf_size)
-                    return -1;
-                if ((count = *dp++) <= 127) {
-                    count++;
-                    dlen -= count + 1;
-                    if (pixptr + count * planes > pixptr_end)
-                        break;
-                    if (dp + count > buf + buf_size)
-                        return -1;
-                    while (count--) {
-                        *pixptr = *dp++;
-                        pixptr += planes;
-                    }
-                } else {
-                    count = 257 - count;
-                    if (pixptr + count * planes > pixptr_end)
-                        break;
-                    while (count--) {
-                        *pixptr = *dp;
-                        pixptr += planes;
-                    }
-                    dp++;
-                    dlen -= 2;
+                /* Decode a plane */
+                for(row = 0; row < height; row++) {
+                        pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
+                        pixptr_end = pixptr + c->pic.linesize[0];
+                        dlen = av_be2ne16(*(const unsigned short *)(lp+row*2));
+                        /* Decode a row of this plane */
+                        while(dlen > 0) {
+                                if(dp + 1 >= buf+buf_size) return -1;
+                                if ((count = *dp++) <= 127) {
+                                        count++;
+                                        dlen -= count + 1;
+                                        if (pixptr + count * planes > pixptr_end)
+                                            break;
+                                        if(dp + count > buf+buf_size) return -1;
+                                        while(count--) {
+                                                *pixptr = *dp++;
+                                                pixptr += planes;
+                                        }
+                                } else {
+                                        count = 257 - count;
+                                        if (pixptr + count * planes > pixptr_end)
+                                            break;
+                                        while(count--) {
+                                                *pixptr = *dp;
+                                                pixptr += planes;
+                                        }
+                                        dp++;
+                                        dlen -= 2;
+                                }
+                        }
                }
-            }
-        }
-    }
-
-    if (avctx->bits_per_coded_sample <= 8) {
-        const uint8_t *pal = av_packet_get_side_data(avpkt,
-                                                     AV_PKT_DATA_PALETTE,
-                                                     NULL);
-        if (pal) {
-            c->pic.palette_has_changed = 1;
-            memcpy(c->pal, pal, AVPALETTE_SIZE);
        }

-        memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
-    }
+        if (avctx->bits_per_coded_sample <= 8) {
+                const uint8_t *pal = av_packet_get_side_data(avpkt,
+                                                             AV_PKT_DATA_PALETTE,
+                                                             NULL);
+                if (pal) {
+                        c->pic.palette_has_changed = 1;
+                        memcpy(c->pal, pal, AVPALETTE_SIZE);
+                }

-    *data_size = sizeof(AVFrame);
-    *(AVFrame*)data = c->pic;
+                memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
+        }

-    /* always report that the buffer was completely consumed */
-    return buf_size;
+        *data_size = sizeof(AVFrame);
+        *(AVFrame*)data = c->pic;
+
+        /* always report that the buffer was completely consumed */
+        return buf_size;
 }


@@ -154,47 +151,47 @@ static int decode_frame(AVCodecContext *avctx, void *data,
 */
 static av_cold int decode_init(AVCodecContext *avctx)
 {
-    EightBpsContext * const c = avctx->priv_data;
+        EightBpsContext * const c = avctx->priv_data;

-    c->avctx       = avctx;
-    c->pic.data[0] = NULL;
+        c->avctx = avctx;

-    avcodec_get_frame_defaults(&c->pic);
-    switch (avctx->bits_per_coded_sample) {
-    case 8:
-        avctx->pix_fmt = PIX_FMT_PAL8;
-        c->planes      = 1;
-        c->planemap[0] = 0; // 1st plane is palette indexes
-        break;
-    case 24:
-        avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
-        c->planes      = 3;
-        c->planemap[0] = 2; // 1st plane is red
-        c->planemap[1] = 1; // 2nd plane is green
-        c->planemap[2] = 0; // 3rd plane is blue
-        break;
-    case 32:
-        avctx->pix_fmt = PIX_FMT_RGB32;
-        c->planes      = 4;
+        avcodec_get_frame_defaults(&c->pic);
+        c->pic.data[0] = NULL;
+
+        switch (avctx->bits_per_coded_sample) {
+                case 8:
+                        avctx->pix_fmt = PIX_FMT_PAL8;
+                        c->planes = 1;
+                        c->planemap[0] = 0; // 1st plane is palette indexes
+                        break;
+                case 24:
+                        avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
+                        c->planes = 3;
+                        c->planemap[0] = 2; // 1st plane is red
+                        c->planemap[1] = 1; // 2nd plane is green
+                        c->planemap[2] = 0; // 3rd plane is blue
+                        break;
+                case 32:
+                        avctx->pix_fmt = PIX_FMT_RGB32;
+                        c->planes = 4;
 #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
+                        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
+                        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 -1;
-    }
+                        break;
+                default:
+                        av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n", avctx->bits_per_coded_sample);
+                        return -1;
+        }

-    return 0;
+  return 0;
 }


@@ -207,12 +204,12 @@ static av_cold int decode_init(AVCodecContext *avctx)
 */
 static av_cold int decode_end(AVCodecContext *avctx)
 {
-    EightBpsContext * const c = avctx->priv_data;
+        EightBpsContext * const c = avctx->priv_data;

-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
+        if (c->pic.data[0])
+                avctx->release_buffer(avctx, &c->pic);

-    return 0;
+        return 0;
 }


--- a/libavcodec/8svx.c
+++ b/libavcodec/8svx.c
@@ -47,7 +47,7 @@ typedef struct EightSvxContext {
    /* buffer used to store the whole audio decoded/interleaved chunk,
     * which is sent with the first packet */
    uint8_t *samples;
-    int64_t samples_size;
+    size_t samples_size;
    int samples_idx;
 } EightSvxContext;

@@ -106,11 +106,12 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
 {
    EightSvxContext *esc = avctx->priv_data;
    int n, out_data_size, ret;
+    uint8_t *out_date;
    uint8_t *src, *dst;

    /* decode and interleave the first packet */
    if (!esc->samples && avpkt) {
-        uint8_t *deinterleaved_samples, *p = NULL;
+        uint8_t *deinterleaved_samples;

        esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
            avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
@@ -129,7 +130,6 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
            }
            if (!(deinterleaved_samples = av_mallocz(n)))
                return AVERROR(ENOMEM);
-            p = deinterleaved_samples;

            /* the uncompressed starting value is contained in the first byte */
            if (avctx->channels == 2) {
@@ -146,7 +146,6 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
            interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
        else
            memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
-        av_freep(&p);
    }

    /* get output buffer */
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -3,14 +3,7 @@ include $(SUBDIR)../config.mak
 NAME = avcodec
 FFLIBS = avutil

-HEADERS = avcodec.h                                                     \
-          avfft.h                                                       \
-          dxva2.h                                                       \
-          vaapi.h                                                       \
-          vda.h                                                         \
-          vdpau.h                                                       \
-          version.h                                                     \
-          xvmc.h                                                        \
+HEADERS = avcodec.h avfft.h dxva2.h opt.h vaapi.h vda.h vdpau.h version.h xvmc.h

 OBJS = allcodecs.o                                                      \
       audioconvert.o                                                   \
@@ -46,7 +39,6 @@ OBJS-$(CONFIG_GOLOMB)                  += golomb.o
 OBJS-$(CONFIG_H264DSP)                 += h264dsp.o h264idct.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
 OBJS-$(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
@@ -64,13 +56,11 @@ OBJS-$(CONFIG_VDPAU)                   += vdpau.o
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
 OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \
-                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp.o aacpsdsp.o
+                                          aacadtsdec.o mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                          aacpsy.o aactab.o      \
                                          psymodel.o iirfilter.o \
-                                          mpeg4audio.o kbdwin.o  \
-                                          audio_frame_queue.o
+                                          mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
@@ -101,18 +91,12 @@ OBJS-$(CONFIG_ATRAC1_DECODER)          += atrac1.o atrac.o
 OBJS-$(CONFIG_ATRAC3_DECODER)          += atrac3.o atrac.o
 OBJS-$(CONFIG_AURA_DECODER)            += cyuv.o
 OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
-OBJS-$(CONFIG_AVRP_DECODER)            += r210dec.o
-OBJS-$(CONFIG_AVRP_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_AVS_DECODER)             += avs.o
-OBJS-$(CONFIG_AVUI_DECODER)            += avuidec.o
-OBJS-$(CONFIG_AVUI_ENCODER)            += avuienc.o
-OBJS-$(CONFIG_AYUV_DECODER)            += v408dec.o
-OBJS-$(CONFIG_AYUV_ENCODER)            += v408enc.o
 OBJS-$(CONFIG_BETHSOFTVID_DECODER)     += bethsoftvideo.o
 OBJS-$(CONFIG_BFI_DECODER)             += bfi.o
 OBJS-$(CONFIG_BINK_DECODER)            += bink.o binkdsp.o
-OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER)   += binkaudio.o wma.o wma_common.o
-OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER)  += binkaudio.o wma.o wma_common.o
+OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER)   += binkaudio.o wma.o
+OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER)  += binkaudio.o wma.o
 OBJS-$(CONFIG_BINTEXT_DECODER)         += bintext.o cga_data.o
 OBJS-$(CONFIG_BMP_DECODER)             += bmp.o msrledec.o
 OBJS-$(CONFIG_BMP_ENCODER)             += bmpenc.o
@@ -122,15 +106,13 @@ OBJS-$(CONFIG_C93_DECODER)             += c93.o
 OBJS-$(CONFIG_CAVS_DECODER)            += cavs.o cavsdec.o cavsdsp.o \
                                          mpeg12data.o mpegvideo.o
 OBJS-$(CONFIG_CDGRAPHICS_DECODER)      += cdgraphics.o
-OBJS-$(CONFIG_CDXL_DECODER)            += cdxl.o
 OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
 OBJS-$(CONFIG_CLJR_DECODER)            += cljr.o
 OBJS-$(CONFIG_CLJR_ENCODER)            += cljr.o
 OBJS-$(CONFIG_COOK_DECODER)            += cook.o
 OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
-OBJS-$(CONFIG_DCA_DECODER)             += dca.o synth_filter.o dcadsp.o \
-                                          dca_parser.o
+OBJS-$(CONFIG_DCA_DECODER)             += dca.o synth_filter.o dcadsp.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
                                          dirac_arith.o mpeg12data.o dwt.o
@@ -148,8 +130,8 @@ OBJS-$(CONFIG_DVBSUB_DECODER)          += dvbsubdec.o
 OBJS-$(CONFIG_DVBSUB_ENCODER)          += dvbsub.o
 OBJS-$(CONFIG_DVDSUB_DECODER)          += dvdsubdec.o
 OBJS-$(CONFIG_DVDSUB_ENCODER)          += dvdsubenc.o
-OBJS-$(CONFIG_DVVIDEO_DECODER)         += dvdec.o dv.o dvdata.o dv_profile.o
-OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o dv_profile.o
+OBJS-$(CONFIG_DVVIDEO_DECODER)         += dv.o dvdata.o
+OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o
 OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
 OBJS-$(CONFIG_DXTORY_DECODER)          += dxtory.o
 OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o eac3_data.o
@@ -170,13 +152,11 @@ OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER)    += 8svx.o
 OBJS-$(CONFIG_EIGHTSVX_RAW_DECODER)    += 8svx.o
 OBJS-$(CONFIG_ESCAPE124_DECODER)       += escape124.o
 OBJS-$(CONFIG_ESCAPE130_DECODER)       += escape130.o
-OBJS-$(CONFIG_EXR_DECODER)             += exr.o
 OBJS-$(CONFIG_FFV1_DECODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFVHUFF_DECODER)         += huffyuv.o
 OBJS-$(CONFIG_FFVHUFF_ENCODER)         += huffyuv.o
-OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
-OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o
+OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
 OBJS-$(CONFIG_FLASHSV_ENCODER)         += flashsvenc.o
@@ -188,15 +168,15 @@ OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
                                          celp_filters.o celp_math.o
-OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
+OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o
 OBJS-$(CONFIG_G729_DECODER)            += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
 OBJS-$(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 h261.o h261data.o  \
+OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o \
                                          mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o h261data.o  \
+OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o             \
                                          mpegvideo_enc.o motion_est.o \
                                          ratecontrol.o mpeg12data.o   \
                                          mpegvideo.o
@@ -231,7 +211,6 @@ OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi_common.o ivi_dsp.o
 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-$(CONFIG_JPEG2000_DECODER)        += j2kdec.o mqcdec.o mqc.o j2k.o j2k_dwt.o
 OBJS-$(CONFIG_JPEG2000_ENCODER)        += j2kenc.o mqcenc.o mqc.o j2k.o j2k_dwt.o
 OBJS-$(CONFIG_JPEGLS_DECODER)          += jpeglsdec.o jpegls.o \
@@ -250,7 +229,6 @@ OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
 OBJS-$(CONFIG_MDEC_DECODER)            += mdec.o mpeg12.o mpeg12data.o \
                                          mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o           \
@@ -314,22 +292,19 @@ OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)     += vaapi_mpeg4.o
 OBJS-$(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 \
-                                          h263dec.o h263.o ituh263dec.o        \
-                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
                                          h263.o ituh263dec.o mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o        \
-                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
 OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o
 OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
-OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o \
-                                          audio_frame_queue.o
+OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
 OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
 OBJS-$(CONFIG_PAM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PAM_ENCODER)             += pamenc.o pnm.o
@@ -343,15 +318,13 @@ OBJS-$(CONFIG_PGMYUV_DECODER)          += pnmdec.o pnm.o
 OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o pnm.o
 OBJS-$(CONFIG_PGSSUB_DECODER)          += pgssubdec.o
 OBJS-$(CONFIG_PICTOR_DECODER)          += pictordec.o cga_data.o
-OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o pngdsp.o
+OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o
 OBJS-$(CONFIG_PNG_ENCODER)             += png.o pngenc.o
 OBJS-$(CONFIG_PPM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
-OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o
-OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
-OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_ANATOLIY_ENCODER) += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_KOSTYA_ENCODER)   += proresenc_kostya.o proresdata.o proresdsp.o
+OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o
+OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o
+OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
 OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o celp_math.o         \
                                          celp_filters.o acelp_vectors.o \
@@ -364,14 +337,10 @@ OBJS-$(CONFIG_QPEG_DECODER)            += qpeg.o
 OBJS-$(CONFIG_QTRLE_DECODER)           += qtrle.o
 OBJS-$(CONFIG_QTRLE_ENCODER)           += qtrleenc.o
 OBJS-$(CONFIG_R10K_DECODER)            += r210dec.o
-OBJS-$(CONFIG_R10K_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_R210_DECODER)            += r210dec.o
-OBJS-$(CONFIG_R210_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_RA_144_DECODER)          += ra144dec.o ra144.o celp_filters.o
-OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o \
-                                          audio_frame_queue.o
+OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_math.o celp_filters.o
-OBJS-$(CONFIG_RALF_DECODER)            += ralf.o
 OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
 OBJS-$(CONFIG_RAWVIDEO_ENCODER)        += rawenc.o
 OBJS-$(CONFIG_RL2_DECODER)             += rl2.o
@@ -413,7 +382,6 @@ OBJS-$(CONFIG_SP5X_DECODER)            += sp5xdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_SRT_DECODER)             += srtdec.o ass.o
 OBJS-$(CONFIG_SRT_ENCODER)             += srtenc.o ass_split.o
 OBJS-$(CONFIG_SUNRAST_DECODER)         += sunrast.o
-OBJS-$(CONFIG_SUNRAST_ENCODER)         += sunrastenc.o
 OBJS-$(CONFIG_SVQ1_DECODER)            += svq1dec.o svq1.o h263.o \
                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_SVQ1_ENCODER)            += svq1enc.o svq1.o    \
@@ -446,10 +414,6 @@ OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
 OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideo.o
 OBJS-$(CONFIG_V210_DECODER)            += v210dec.o
 OBJS-$(CONFIG_V210_ENCODER)            += v210enc.o
-OBJS-$(CONFIG_V308_DECODER)            += v308dec.o
-OBJS-$(CONFIG_V308_ENCODER)            += v308enc.o
-OBJS-$(CONFIG_V408_DECODER)            += v408dec.o
-OBJS-$(CONFIG_V408_ENCODER)            += v408enc.o
 OBJS-$(CONFIG_V410_DECODER)            += v410dec.o
 OBJS-$(CONFIG_V410_ENCODER)            += v410enc.o
 OBJS-$(CONFIG_V210X_DECODER)           += v210x.o
@@ -477,12 +441,12 @@ OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
 OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
-OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
-OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o wma_common.o
-OBJS-$(CONFIG_WMAV1_DECODER)           += wmadec.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
+OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma.o
+OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o
+OBJS-$(CONFIG_WMAV1_DECODER)           += wmadec.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o aactab.o
 OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                          celp_math.o celp_filters.o \
                                          acelp_vectors.o acelp_filters.o
@@ -491,7 +455,7 @@ OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o        \
                                          msmpeg4.o msmpeg4data.o \
                                          intrax8.o intrax8dsp.o
 OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
-                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o \
+                                          msmpeg4.o msmpeg4data.o \
                                          mpeg4videodec.o ituh263dec.o h263dec.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
@@ -499,19 +463,12 @@ OBJS-$(CONFIG_XAN_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_XAN_WC3_DECODER)         += xan.o
 OBJS-$(CONFIG_XAN_WC4_DECODER)         += xxan.o
 OBJS-$(CONFIG_XBIN_DECODER)            += bintext.o cga_data.o
-OBJS-$(CONFIG_XBM_DECODER)             += xbmdec.o
-OBJS-$(CONFIG_XBM_ENCODER)             += xbmenc.o
 OBJS-$(CONFIG_XL_DECODER)              += xl.o
 OBJS-$(CONFIG_XSUB_DECODER)            += xsubdec.o
 OBJS-$(CONFIG_XSUB_ENCODER)            += xsubenc.o
-OBJS-$(CONFIG_XWD_DECODER)             += xwddec.o
-OBJS-$(CONFIG_XWD_ENCODER)             += xwdenc.o
 OBJS-$(CONFIG_Y41P_DECODER)            += y41pdec.o
 OBJS-$(CONFIG_Y41P_ENCODER)            += y41penc.o
 OBJS-$(CONFIG_YOP_DECODER)             += yop.o
-OBJS-$(CONFIG_YUV4_DECODER)            += yuv4dec.o
-OBJS-$(CONFIG_YUV4_ENCODER)            += yuv4enc.o
-OBJS-$(CONFIG_ZEROCODEC_DECODER)       += zerocodec.o
 OBJS-$(CONFIG_ZLIB_DECODER)            += lcldec.o
 OBJS-$(CONFIG_ZLIB_ENCODER)            += lclenc.o
 OBJS-$(CONFIG_ZMBV_DECODER)            += zmbv.o
@@ -570,20 +527,19 @@ OBJS-$(CONFIG_PCM_ZORK_DECODER)           += pcm.o

 OBJS-$(CONFIG_ADPCM_4XM_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_ADX_DECODER)          += adxdec.o adx.o
-OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o adx.o
+OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o
 OBJS-$(CONFIG_ADPCM_CT_DECODER)           += adpcm.o adpcm_data.o
-OBJS-$(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
-OBJS-$(CONFIG_ADPCM_EA_R2_DECODER)        += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_R3_DECODER)        += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER)       += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_EA_DECODER)           += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER)  += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_R1_DECODER)        += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_R2_DECODER)        += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_R3_DECODER)        += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER)       += adpcm.o
 OBJS-$(CONFIG_ADPCM_G722_DECODER)         += g722.o g722dec.o
 OBJS-$(CONFIG_ADPCM_G722_ENCODER)         += g722.o g722enc.o
 OBJS-$(CONFIG_ADPCM_G726_DECODER)         += g726.o
 OBJS-$(CONFIG_ADPCM_G726_ENCODER)         += g726.o
 OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER)  += adpcm.o adpcm_data.o
@@ -597,13 +553,13 @@ OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER)      += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_MS_DECODER)           += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_MS_ENCODER)           += adpcmenc.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER)      += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER)      += adpcm.o
+OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER)      += adpcm.o
+OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER)      += adpcm.o
 OBJS-$(CONFIG_ADPCM_SWF_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_SWF_ENCODER)          += adpcmenc.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_THP_DECODER)          += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_XA_DECODER)           += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_THP_DECODER)          += adpcm.o
+OBJS-$(CONFIG_ADPCM_XA_DECODER)           += adpcm.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o

@@ -611,10 +567,9 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_ADTS_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
 OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_DV_DEMUXER)              += dv_profile.o
-OBJS-$(CONFIG_DV_MUXER)                += dv_profile.o timecode.o
-OBJS-$(CONFIG_FLAC_DEMUXER)            += flacdec.o flacdata.o flac.o vorbis_data.o \
-                                          vorbis_parser.o xiph.o
+OBJS-$(CONFIG_DV_DEMUXER)              += dvdata.o
+OBJS-$(CONFIG_DV_MUXER)                += dvdata.o timecode.o
+OBJS-$(CONFIG_FLAC_DEMUXER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_MUXER)              += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLV_DEMUXER)             += mpeg4audio.o
 OBJS-$(CONFIG_GXF_DEMUXER)             += mpeg12data.o
@@ -626,17 +581,15 @@ OBJS-$(CONFIG_MATROSKA_DEMUXER)        += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MATROSKA_MUXER)          += xiph.o mpeg4audio.o \
                                          flacdec.o flacdata.o flac.o \
                                          mpegaudiodata.o vorbis_data.o
-OBJS-$(CONFIG_MP2_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
 OBJS-$(CONFIG_MP3_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
-OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o ac3tab.o timecode.o
+OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MOV_MUXER)               += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGTS_MUXER)            += mpegvideo.o mpeg4audio.o
 OBJS-$(CONFIG_MPEGTS_DEMUXER)          += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MXF_MUXER)               += timecode.o
 OBJS-$(CONFIG_NUT_MUXER)               += mpegaudiodata.o
 OBJS-$(CONFIG_OGG_DEMUXER)             += flacdec.o flacdata.o flac.o \
-                                          dirac.o mpeg12data.o vorbis_parser.o \
-                                          xiph.o vorbis_data.o
+                                          dirac.o mpeg12data.o vorbis_data.o
 OBJS-$(CONFIG_OGG_MUXER)               += xiph.o flacdec.o flacdata.o flac.o \
                                          vorbis_data.o
 OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o mpegvideo.o xiph.o
@@ -649,40 +602,38 @@ OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 # external codec libraries
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
-OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBDIRAC_DECODER)           += libdiracdec.o
+OBJS-$(CONFIG_LIBDIRAC_ENCODER)           += libdiracenc.o libdirac_libschro.o
+OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_ENCODER)          += libgsm.o
-OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o \
-                                             audio_frame_queue.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o \
-                                             audio_frame_queue.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o \
-                                             audio_frame_queue.o
+OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o
+OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o
+OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENJPEG_DECODER)        += libopenjpegdec.o
 OBJS-$(CONFIG_LIBOPENJPEG_ENCODER)        += libopenjpegenc.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER)    += libschroedingerdec.o \
-                                             libschroedinger.o
+                                             libschroedinger.o    \
+                                             libdirac_libschro.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER)    += libschroedingerenc.o \
-                                             libschroedinger.o
+                                             libschroedinger.o    \
+                                             libdirac_libschro.o
 OBJS-$(CONFIG_LIBSPEEX_DECODER)           += libspeexdec.o
-OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o
 OBJS-$(CONFIG_LIBSTAGEFRIGHT_H264_DECODER)+= libstagefright.o
 OBJS-$(CONFIG_LIBTHEORA_ENCODER)          += libtheoraenc.o
-OBJS-$(CONFIG_LIBUTVIDEO_DECODER)         += libutvideodec.o
-OBJS-$(CONFIG_LIBUTVIDEO_ENCODER)         += libutvideoenc.o
-OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)       += libvo-aacenc.o mpeg4audio.o \
-                                             audio_frame_queue.o
+OBJS-$(CONFIG_LIBUTVIDEO_DECODER)         += libutvideo.o
+OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)       += libvo-aacenc.o mpeg4audio.o
 OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER)     += libvo-amrwbenc.o
-OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbis.o audio_frame_queue.o \
-                                             vorbis_data.o vorbis_parser.o
+OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbis.o vorbis_data.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_LIBXVID)                    += libxvidff.o libxvid_rc.o

 # parsers
 OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o \
@@ -691,7 +642,6 @@ OBJS-$(CONFIG_AC3_PARSER)              += ac3_parser.o ac3tab.o \
                                          aac_ac3_parser.o
 OBJS-$(CONFIG_ADX_PARSER)              += adx_parser.o adx.o
 OBJS-$(CONFIG_CAVSVIDEO_PARSER)        += cavs_parser.o
-OBJS-$(CONFIG_COOK_PARSER)             += cook_parser.o
 OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o
 OBJS-$(CONFIG_DIRAC_PARSER)            += dirac_parser.o
 OBJS-$(CONFIG_DNXHD_PARSER)            += dnxhd_parser.o
@@ -699,7 +649,6 @@ OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                          vorbis_data.o
-OBJS-$(CONFIG_GSM_PARSER)              += gsm_parser.o
 OBJS-$(CONFIG_H261_PARSER)             += h261_parser.o
 OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
 OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o h264.o            \
@@ -715,7 +664,6 @@ OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
                                          mpegvideo.o error_resilience.o \
                                          mpeg4videodec.o mpeg4video.o \
                                          ituh263dec.o h263dec.o
-OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                          mpegaudiodecheader.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGVIDEO_PARSER)        += mpegvideo_parser.o    \
@@ -727,7 +675,6 @@ OBJS-$(CONFIG_RV40_PARSER)             += rv34_parser.o
 OBJS-$(CONFIG_VC1_PARSER)              += vc1_parser.o vc1.o vc1data.o \
                                          msmpeg4.o msmpeg4data.o mpeg4video.o \
                                          h263.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_VORBIS_PARSER)           += vorbis_parser.o xiph.o
 OBJS-$(CONFIG_VP3_PARSER)              += vp3_parser.o
 OBJS-$(CONFIG_VP8_PARSER)              += vp8_parser.o

@@ -753,55 +700,45 @@ OBJS-$(HAVE_PTHREADS)                  += pthread.o
 OBJS-$(HAVE_W32THREADS)                += pthread.o
 OBJS-$(HAVE_OS2THREADS)                += pthread.o

+OBJS-$(CONFIG_MLIB)                    += mlib/dsputil_mlib.o           \
+
 # inverse.o contains the ff_inverse table definition, which is used by
 # the FASTDIV macro (from libavutil); since referencing the external
 # table has a negative effect on performance, copy it in libavcodec as
 # well.
 OBJS-$(!CONFIG_SMALL)                  += inverse.o

+-include $(SRC_PATH)/$(SUBDIR)$(ARCH)/Makefile
+
 SKIPHEADERS                            += %_tablegen.h                  \
                                          %_tables.h                    \
                                          aac_tablegen_decl.h           \
-                                          codec_names.h                 \
                                          fft-internal.h                \
                                          tableprint.h                  \
-                                          $(ARCH)/vp56_arith.h          \
-
+                                          $(ARCH)/vp56_arith.h
 SKIPHEADERS-$(CONFIG_DXVA2)            += dxva2.h dxva2_internal.h
+SKIPHEADERS-$(CONFIG_LIBDIRAC)         += libdirac.h
 SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
-SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
-SKIPHEADERS-$(CONFIG_MPEG_XVMC_DECODER) += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
-SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_internal.h
+SKIPHEADERS-$(CONFIG_VDA)              += vda_internal.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h
-SKIPHEADERS-$(HAVE_OS2THREADS)         += os2threads.h
+SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
 SKIPHEADERS-$(HAVE_W32THREADS)         += w32pthreads.h

-TESTPROGS = cabac                                                       \
-            dct                                                         \
-            fft                                                         \
-            fft-fixed                                                   \
-            golomb                                                      \
-            iirfilter                                                   \
-            rangecoder                                                  \
-            snowenc                                                     \
-
+TESTPROGS = cabac dct fft fft-fixed h264 iirfilter rangecoder snowenc
 TESTPROGS-$(HAVE_MMX) += motion
 TESTOBJS = dctref.o

-HOSTPROGS = aac_tablegen                                                \
-            aacps_tablegen                                              \
-            cbrt_tablegen                                               \
-            cos_tablegen                                                \
-            dv_tablegen                                                 \
-            motionpixels_tablegen                                       \
-            mpegaudio_tablegen                                          \
-            pcm_tablegen                                                \
-            qdm2_tablegen                                               \
-            sinewin_tablegen                                            \
+HOSTPROGS = aac_tablegen aacps_tablegen cbrt_tablegen cos_tablegen      \
+            dv_tablegen motionpixels_tablegen mpegaudio_tablegen        \
+            pcm_tablegen qdm2_tablegen sinewin_tablegen
+
+DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86

 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)

+include $(SRC_PATH)/subdir.mak
+
 $(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o

 TRIG_TABLES  = cos cos_fixed sin
--- a/libavcodec/a64enc.h
+++ b/libavcodec/a64enc.h
@@ -50,9 +50,6 @@ typedef struct A64Context {
    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;

 #endif /* AVCODEC_A64ENC_H */
--- a/libavcodec/a64multienc.c
+++ b/libavcodec/a64multienc.c
@@ -28,7 +28,6 @@
 #include "a64colors.h"
 #include "a64tables.h"
 #include "elbg.h"
-#include "internal.h"
 #include "libavutil/intreadwrite.h"

 #define DITHERSTEPS   8
@@ -222,8 +221,6 @@ static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
    if (!avctx->codec_tag)
         avctx->codec_tag = AV_RL32("a64m");

-    c->next_pts = AV_NOPTS_VALUE;
-
    return 0;
 }

@@ -242,19 +239,19 @@ static void a64_compress_colram(unsigned char *buf, int *charmap, uint8_t *colra
    }
 }

-static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
-                                 const AVFrame *pict, int *got_packet)
+static int a64multi_encode_frame(AVCodecContext *avctx, unsigned char *buf,
+                                 int buf_size, void *data)
 {
    A64Context *c = avctx->priv_data;
-    AVFrame *const p = &c->picture;
+    AVFrame *pict = data;
+    AVFrame *const p = (AVFrame *) & c->picture;

    int frame;
    int x, y;
    int b_height;
    int b_width;

-    int req_size, ret;
-    uint8_t *buf = NULL;
+    int req_size;

    int *charmap     = c->mc_charmap;
    uint8_t *colram  = c->mc_colram;
@@ -277,7 +274,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
    }

    /* no data, means end encoding asap */
-    if (!pict) {
+    if (!data) {
        /* all done, end encoding */
        if (!c->mc_lifetime) return 0;
        /* no more frames in queue, prepare to flush remaining frames */
@@ -295,8 +292,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
            p->key_frame = 1;
            to_meta_with_crop(avctx, p, meta + 32000 * c->mc_frame_counter);
            c->mc_frame_counter++;
-            if (c->next_pts == AV_NOPTS_VALUE)
-                c->next_pts = pict->pts;
            /* lifetime is not reached so wait for next frame first */
            return 0;
        }
@@ -307,11 +302,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
        req_size = 0;
        /* any frames to encode? */
        if (c->mc_lifetime) {
-            req_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
-            if ((ret = ff_alloc_packet2(avctx, pkt, req_size)) < 0)
-                return ret;
-            buf = pkt->data;
-
            /* calc optimal new charset + charmaps */
            ff_init_elbg(meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
            ff_do_elbg  (meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
@@ -320,12 +310,15 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
            render_charset(avctx, charset, colram);

            /* copy charset to buf */
-            memcpy(buf, charset, charset_size);
+            memcpy(buf,charset, charset_size);

            /* advance pointers */
            buf      += charset_size;
            charset  += charset_size;
+            req_size += charset_size;
        }
+        /* no charset so clean buf */
+        else memset(buf, 0, charset_size);

        /* write x frames to buf */
        for (frame = 0; frame < c->mc_lifetime; frame++) {
@@ -358,12 +351,11 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
        /* reset counter */
        c->mc_frame_counter = 0;

-        pkt->pts = pkt->dts = c->next_pts;
-        c->next_pts         = AV_NOPTS_VALUE;
-
-        pkt->size   = req_size;
-        pkt->flags |= AV_PKT_FLAG_KEY;
-        *got_packet = !!req_size;
+        if (req_size > buf_size) {
+            av_log(avctx, AV_LOG_ERROR, "buf size too small (need %d, got %d)\n", req_size, buf_size);
+            return -1;
+        }
+        return req_size;
    }
    return 0;
 }
@@ -374,7 +366,7 @@ AVCodec ff_a64multi_encoder = {
    .id             = CODEC_ID_A64_MULTI,
    .priv_data_size = sizeof(A64Context),
    .init           = a64multi_init_encoder,
-    .encode2        = a64multi_encode_frame,
+    .encode         = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
@@ -387,7 +379,7 @@ AVCodec ff_a64multi5_encoder = {
    .id             = CODEC_ID_A64_MULTI5,
    .priv_data_size = sizeof(A64Context),
    .init           = a64multi_init_encoder,
-    .encode2        = a64multi_encode_frame,
+    .encode         = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -84,7 +84,6 @@ enum BandType {
 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)

 enum ChannelPosition {
-    AAC_CHANNEL_OFF   = 0,
    AAC_CHANNEL_FRONT = 1,
    AAC_CHANNEL_SIDE  = 2,
    AAC_CHANNEL_BACK  = 3,
@@ -112,15 +111,6 @@ enum OCStatus {
    OC_LOCKED,      ///< Output configuration locked in place
 };

-typedef struct {
-    MPEG4AudioConfig m4ac;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int layout_map_tags;
-    int channels;
-    uint64_t channel_layout;
-    enum OCStatus status;
-} OutputConfiguration;
-
 /**
 * Predictor State
 */
@@ -263,6 +253,8 @@ typedef struct {
    AVCodecContext *avctx;
    AVFrame frame;

+    MPEG4AudioConfig m4ac;
+
    int is_saved;                 ///< Set if elements have stored overlap from previous frame.
    DynamicRangeControl che_drc;

@@ -270,6 +262,9 @@ typedef struct {
     * @name Channel element related data
     * @{
     */
+    enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
+                                                   *   first index as the first 4 raw data block types
+                                                   */
    ChannelElement          *che[4][MAX_ELEM_ID];
    ChannelElement  *tag_che_map[4][MAX_ELEM_ID];
    int tags_mapped;
@@ -304,7 +299,7 @@ typedef struct {

    DECLARE_ALIGNED(32, float, temp)[128];

-    OutputConfiguration oc[2];
+    enum OCStatus output_configured;
    int warned_num_aac_frames;
 } AACContext;

--- a/libavcodec/aac_ac3_parser.c
+++ b/libavcodec/aac_ac3_parser.c
@@ -93,7 +93,7 @@ get_next:
            avctx->channels = s->channels;
            avctx->channel_layout = s->channel_layout;
        }
-        s1->duration = s->samples;
+        avctx->frame_size = s->samples;
        avctx->audio_service_type = s->service_type;
    }

--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -110,15 +110,14 @@ static av_always_inline float quantize_and_encode_band_cost_template(
                                int *bits, int BT_ZERO, int BT_UNSIGNED,
                                int BT_PAIR, int BT_ESC)
 {
-    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
-    const float Q   = ff_aac_pow2sf_tab [q_idx];
-    const float Q34 = ff_aac_pow34sf_tab[q_idx];
-    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float IQ = ff_aac_pow2sf_tab[POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float  Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
    const float CLIPPED_ESCAPE = 165140.0f*IQ;
    int i, j;
    float cost = 0;
    const int dim = BT_PAIR ? 2 : 4;
    int resbits = 0;
+    const float  Q34 = sqrtf(Q * sqrtf(Q));
    const int range  = aac_cb_range[cb];
    const int maxval = aac_cb_maxval[cb];
    int off;
@@ -421,7 +420,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
    const int run_esc  = (1 << run_bits) - 1;
    int idx, ppos, count;
    int stackrun[120], stackcb[120], stack_len;
-    float next_minbits = INFINITY;
+    float next_minrd = INFINITY;
    int next_mincb = 0;

    abs_pow34_v(s->scoefs, sce->coeffs, 1024);
@@ -435,7 +434,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
        size = sce->ics.swb_sizes[swb];
        if (sce->zeroes[win*16 + swb]) {
            float cost_stay_here = path[swb][0].cost;
-            float cost_get_here  = next_minbits + run_bits + 4;
+            float cost_get_here  = next_minrd + run_bits + 4;
            if (   run_value_bits[sce->ics.num_windows == 8][path[swb][0].run]
                != run_value_bits[sce->ics.num_windows == 8][path[swb][0].run+1])
                cost_stay_here += run_bits;
@@ -448,7 +447,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                path[swb+1][0].cost     = cost_stay_here;
                path[swb+1][0].run      = path[swb][0].run + 1;
            }
-            next_minbits = path[swb+1][0].cost;
+            next_minrd = path[swb+1][0].cost;
            next_mincb = 0;
            for (cb = 1; cb < 12; cb++) {
                path[swb+1][cb].cost = 61450;
@@ -456,10 +455,10 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                path[swb+1][cb].run = 0;
            }
        } else {
-            float minbits = next_minbits;
+            float minrd = next_minrd;
            int mincb = next_mincb;
            int startcb = sce->band_type[win*16+swb];
-            next_minbits = INFINITY;
+            next_minrd = INFINITY;
            next_mincb = 0;
            for (cb = 0; cb < startcb; cb++) {
                path[swb+1][cb].cost = 61450;
@@ -468,15 +467,15 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
            }
            for (cb = startcb; cb < 12; cb++) {
                float cost_stay_here, cost_get_here;
-                float bits = 0.0f;
+                float rd = 0.0f;
                for (w = 0; w < group_len; w++) {
-                    bits += quantize_band_cost(s, sce->coeffs + start + w*128,
-                                               s->scoefs + start + w*128, size,
-                                               sce->sf_idx[(win+w)*16+swb], cb,
-                                               0, INFINITY, NULL);
+                    rd += quantize_band_cost(s, sce->coeffs + start + w*128,
+                                             s->scoefs + start + w*128, size,
+                                             sce->sf_idx[(win+w)*16+swb], cb,
+                                             0, INFINITY, NULL);
                }
-                cost_stay_here = path[swb][cb].cost + bits;
-                cost_get_here  = minbits            + bits + run_bits + 4;
+                cost_stay_here = path[swb][cb].cost + rd;
+                cost_get_here  = minrd              + rd + run_bits + 4;
                if (   run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
                    != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
                    cost_stay_here += run_bits;
@@ -489,8 +488,8 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                    path[swb+1][cb].cost     = cost_stay_here;
                    path[swb+1][cb].run      = path[swb][cb].run + 1;
                }
-                if (path[swb+1][cb].cost < next_minbits) {
-                    next_minbits = path[swb+1][cb].cost;
+                if (path[swb+1][cb].cost < next_minrd) {
+                    next_minrd = path[swb+1][cb].cost;
                    next_mincb = cb;
                }
            }
@@ -714,17 +713,15 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
 {
    int start = 0, i, w, w2, g;
    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels;
-    float dists[128] = { 0 }, uplims[128];
+    float dists[128], uplims[128];
    float maxvals[128];
    int fflag, minscaler;
    int its  = 0;
    int allz = 0;
    float minthr = INFINITY;

-    // for values above this the decoder might end up in an endless loop
-    // due to always having more bits than what can be encoded.
-    destbits = FFMIN(destbits, 5800);
    //XXX: some heuristic to determine initial quantizers will reduce search time
+    memset(dists, 0, sizeof(dists));
    //determine zero bands and upper limits
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        for (g = 0;  g < sce->ics.num_swb; g++) {
@@ -878,7 +875,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
    } else {
        for (w = 0; w < 8; w++) {
            const float *coeffs = sce->coeffs + w*128;
-            curband = start = 0;
+            start = 0;
            for (i = 0; i < 128; i++) {
                if (i - start >= sce->ics.swb_sizes[curband]) {
                    start += sce->ics.swb_sizes[curband];
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
--- a/libavcodec/aacdectab.h
+++ b/libavcodec/aacdectab.h
@@ -80,14 +80,14 @@ static const float * const tns_tmp2_map[4] = {

 static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };

-static const uint8_t aac_channel_layout_map[7][5][3] = {
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_FRONT }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+static const uint8_t aac_channel_layout_map[7][5][2] = {
+    { { TYPE_SCE, 0 }, },
+    { { TYPE_CPE, 0 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_SCE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_CPE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 2 }, { TYPE_CPE, 1 }, },
 };

 static const uint64_t aac_channel_layout[8] = {
@@ -97,7 +97,7 @@ static const uint64_t aac_channel_layout[8] = {
    AV_CH_LAYOUT_4POINT0,
    AV_CH_LAYOUT_5POINT0_BACK,
    AV_CH_LAYOUT_5POINT1_BACK,
-    AV_CH_LAYOUT_7POINT1_WIDE_BACK,
+    AV_CH_LAYOUT_7POINT1_WIDE,
    0,
 };

--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -34,7 +34,6 @@
 #include "avcodec.h"
 #include "put_bits.h"
 #include "dsputil.h"
-#include "internal.h"
 #include "mpeg4audio.h"
 #include "kbdwin.h"
 #include "sinewin.h"
@@ -47,14 +46,6 @@

 #define AAC_MAX_CHANNELS 6

-#define ERROR_IF(cond, ...) \
-    if (cond) { \
-        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
-        return AVERROR(EINVAL); \
-    }
-
-float ff_aac_pow34sf_tab[428];
-
 static const uint8_t swb_size_1024_96[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
@@ -144,10 +135,7 @@ static const uint8_t aac_chan_configs[6][5] = {
 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
 };

-/**
- * Table to remap channels from libavcodec's default order to AAC order.
- */
-static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+static const uint8_t channel_maps[][AAC_MAX_CHANNELS] = {
    { 0 },
    { 0, 1 },
    { 2, 0, 1 },
@@ -168,7 +156,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
    init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
    put_bits(&pb, 5, 2); //object type - AAC-LC
    put_bits(&pb, 4, s->samplerate_index); //sample rate index
-    put_bits(&pb, 4, s->channels);
+    put_bits(&pb, 4, avctx->channels);
    //GASpecificConfig
    put_bits(&pb, 1, 0); //frame length - 1024 samples
    put_bits(&pb, 1, 0); //does not depend on core coder
@@ -181,81 +169,117 @@ static void put_audio_specific_config(AVCodecContext *avctx)
    flush_put_bits(&pb);
 }

-#define WINDOW_FUNC(type) \
-static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio)
-
-WINDOW_FUNC(only_long)
-{
-    const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    float *out = sce->ret;
-
-    dsp->vector_fmul        (out,        audio,        lwindow, 1024);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
-}
-
-WINDOW_FUNC(long_start)
-{
-    const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
-    float *out = sce->ret;
-
-    dsp->vector_fmul(out, audio, lwindow, 1024);
-    memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
-    memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
-}
-
-WINDOW_FUNC(long_stop)
-{
-    const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
-    float *out = sce->ret;
-
-    memset(out, 0, sizeof(out[0]) * 448);
-    dsp->vector_fmul(out + 448, audio + 448, swindow, 128);
-    memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
-}
-
-WINDOW_FUNC(eight_short)
-{
-    const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
-    const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
-    const float *in = audio + 448;
-    float *out = sce->ret;
-    int w;
-
-    for (w = 0; w < 8; w++) {
-        dsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
-        out += 128;
-        in  += 128;
-        dsp->vector_fmul_reverse(out, in, swindow, 128);
-        out += 128;
-    }
-}
-
-static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = {
-    [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
-    [LONG_START_SEQUENCE]  = apply_long_start_window,
-    [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
-    [LONG_STOP_SEQUENCE]   = apply_long_stop_window
-};
-
-static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
-                                  float *audio)
+static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
+    AACEncContext *s = avctx->priv_data;
    int i;
+    const uint8_t *sizes[2];
+    uint8_t grouping[AAC_MAX_CHANNELS];
+    int lengths[2];
+
+    avctx->frame_size = 1024;
+
+    for (i = 0; i < 16; i++)
+        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
+            break;
+    if (i == 16) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
+        return -1;
+    }
+    if (avctx->channels > AAC_MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
+        return -1;
+    }
+    if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
+        return -1;
+    }
+    if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
+        av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
+        return -1;
+    }
+    s->samplerate_index = i;
+
+    dsputil_init(&s->dsp, avctx);
+    ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
+    ff_mdct_init(&s->mdct128,   8, 0, 1.0);
+    // window init
+    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
+    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
+    ff_init_ff_sine_windows(10);
+    ff_init_ff_sine_windows(7);
+
+    s->chan_map           = aac_chan_configs[avctx->channels-1];
+    s->samples            = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
+    s->cpe                = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]);
+    avctx->extradata      = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE);
+    avctx->extradata_size = 5;
+    put_audio_specific_config(avctx);
+
+    sizes[0]   = swb_size_1024[i];
+    sizes[1]   = swb_size_128[i];
+    lengths[0] = ff_aac_num_swb_1024[i];
+    lengths[1] = ff_aac_num_swb_128[i];
+    for (i = 0; i < s->chan_map[0]; i++)
+        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
+    ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
+    s->psypp = ff_psy_preprocess_init(avctx);
+    s->coder = &ff_aac_coders[s->options.aac_coder];
+
+    s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+
+    ff_aac_tableinit();
+
+    return 0;
+}
+
+static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
+                                  SingleChannelElement *sce, short *audio)
+{
+    int i, k;
+    const int chans = avctx->channels;
+    const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+    const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+    const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
    float *output = sce->ret;

-    apply_window[sce->ics.window_sequence[0]](&s->dsp, sce, audio);
-
-    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
+    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+        memcpy(output, sce->saved, sizeof(float)*1024);
+        if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
+            memset(output, 0, sizeof(output[0]) * 448);
+            for (i = 448; i < 576; i++)
+                output[i] = sce->saved[i] * pwindow[i - 448];
+            for (i = 576; i < 704; i++)
+                output[i] = sce->saved[i];
+        }
+        if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
+            for (i = 0; i < 1024; i++) {
+                output[i+1024]         = audio[i * chans] * lwindow[1024 - i - 1];
+                sce->saved[i] = audio[i * chans] * lwindow[i];
+            }
+        } else {
+            for (i = 0; i < 448; i++)
+                output[i+1024]         = audio[i * chans];
+            for (; i < 576; i++)
+                output[i+1024]         = audio[i * chans] * swindow[576 - i - 1];
+            memset(output+1024+576, 0, sizeof(output[0]) * 448);
+            for (i = 0; i < 1024; i++)
+                sce->saved[i] = audio[i * chans];
+        }
        s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
-    else
-        for (i = 0; i < 1024; i += 128)
-            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
-    memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
+    } else {
+        for (k = 0; k < 1024; k += 128) {
+            for (i = 448 + k; i < 448 + k + 256; i++)
+                output[i - 448 - k] = (i < 1024)
+                                         ? sce->saved[i]
+                                         : audio[(i-1024)*chans];
+            s->dsp.vector_fmul        (output,     output, k ?  swindow : pwindow, 128);
+            s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
+            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
+        }
+        for (i = 0; i < 1024; i++)
+            sce->saved[i] = audio[i * chans];
+    }
 }

 /**
@@ -464,69 +488,57 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
    put_bits(&s->pb, 3, TYPE_FIL);
    put_bits(&s->pb, 4, FFMIN(namelen, 15));
    if (namelen >= 15)
-        put_bits(&s->pb, 8, namelen - 14);
+        put_bits(&s->pb, 8, namelen - 16);
    put_bits(&s->pb, 4, 0); //extension type - filler
-    padbits = -put_bits_count(&s->pb) & 7;
+    padbits = 8 - (put_bits_count(&s->pb) & 7);
    avpriv_align_put_bits(&s->pb);
    for (i = 0; i < namelen - 2; i++)
        put_bits(&s->pb, 8, name[i]);
    put_bits(&s->pb, 12 - padbits, 0);
 }

-/*
- * Deinterleave input samples.
- * Channels are reordered from libavcodec's default order to AAC order.
- */
-static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
-{
-    int ch, i;
-    const int sinc = s->channels;
-    const uint8_t *channel_map = aac_chan_maps[sinc - 1];
-
-    /* deinterleave and remap input samples */
-    for (ch = 0; ch < sinc; ch++) {
-        /* copy last 1024 samples of previous frame to the start of the current frame */
-        memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
-
-        /* deinterleave */
-        i = 2048;
-        if (frame) {
-            const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];
-            for (; i < 2048 + frame->nb_samples; i++) {
-                s->planar_samples[ch][i] = *sptr;
-                sptr += sinc;
-            }
-        }
-        memset(&s->planar_samples[ch][i], 0,
-               (3072 - i) * sizeof(s->planar_samples[0][0]));
-    }
-}
-
-static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                            const AVFrame *frame, int *got_packet_ptr)
+static int aac_encode_frame(AVCodecContext *avctx,
+                            uint8_t *frame, int buf_size, void *data)
 {
    AACEncContext *s = avctx->priv_data;
-    float **samples = s->planar_samples, *samples2, *la, *overlap;
+    int16_t *samples = s->samples, *samples2, *la;
    ChannelElement *cpe;
-    int i, ch, w, g, chans, tag, start_ch, ret;
+    int i, ch, w, g, chans, tag, start_ch;
    int chan_el_counter[4];
    FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

-    if (s->last_frame == 2)
+    if (s->last_frame)
        return 0;
-
-    /* add current frame to queue */
-    if (frame) {
-        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
-            return ret;
+    if (data) {
+        if (!s->psypp) {
+            if (avctx->channels <= 2) {
+                memcpy(s->samples + 1024 * avctx->channels, data,
+                       1024 * avctx->channels * sizeof(s->samples[0]));
+            } else {
+                for (i = 0; i < 1024; i++)
+                    for (ch = 0; ch < avctx->channels; ch++)
+                        s->samples[(i + 1024) * avctx->channels + ch] =
+                            ((int16_t*)data)[i * avctx->channels +
+                                             channel_maps[avctx->channels-1][ch]];
+            }
+        } else {
+            start_ch = 0;
+            samples2 = s->samples + 1024 * avctx->channels;
+            for (i = 0; i < s->chan_map[0]; i++) {
+                tag = s->chan_map[i+1];
+                chans = tag == TYPE_CPE ? 2 : 1;
+                ff_psy_preprocess(s->psypp,
+                                  (uint16_t*)data + channel_maps[avctx->channels-1][start_ch],
+                                  samples2 + start_ch, start_ch, chans);
+                start_ch += chans;
+            }
+        }
    }
-
-    deinterleave_input_samples(s, frame);
-    if (s->psypp)
-        ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
-
-    if (!avctx->frame_number)
+    if (!avctx->frame_number) {
+        memcpy(s->samples, s->samples + 1024 * avctx->channels,
+               1024 * avctx->channels * sizeof(s->samples[0]));
        return 0;
+    }

    start_ch = 0;
    for (i = 0; i < s->chan_map[0]; i++) {
@@ -537,10 +549,9 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
        for (ch = 0; ch < chans; ch++) {
            IndividualChannelStream *ics = &cpe->ch[ch].ics;
            int cur_channel = start_ch + ch;
-            overlap  = &samples[cur_channel][0];
-            samples2 = overlap + 1024;
-            la       = samples2 + (448+64);
-            if (!frame)
+            samples2 = samples + cur_channel;
+            la       = samples2 + (448+64) * avctx->channels;
+            if (!data)
                la = NULL;
            if (tag == TYPE_LFE) {
                wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
@@ -567,19 +578,13 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            for (w = 0; w < ics->num_windows; w++)
                ics->group_len[w] = wi[ch].grouping[w];

-            apply_window_and_mdct(s, &cpe->ch[ch], overlap);
+            apply_window_and_mdct(avctx, s, &cpe->ch[ch], samples2);
        }
        start_ch += chans;
    }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 768 * s->channels))) {
-        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
-        return ret;
-    }
    do {
        int frame_bits;
-
-        init_put_bits(&s->pb, avpkt->data, avpkt->size);
-
+        init_put_bits(&s->pb, frame, buf_size*8);
        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
            put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
        start_ch = 0;
@@ -639,8 +644,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
        }

        frame_bits = put_bits_count(&s->pb);
-        if (frame_bits <= 6144 * s->channels - 3) {
-            s->psy.bitres.bits = frame_bits / s->channels;
+        if (frame_bits <= 6144 * avctx->channels - 3) {
+            s->psy.bitres.bits = frame_bits / avctx->channels;
            break;
        }

@@ -659,15 +664,11 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
        s->lambda = FFMIN(s->lambda, 65536.f);
    }

-    if (!frame)
-        s->last_frame++;
-
-    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
-                       &avpkt->duration);
-
-    avpkt->size = put_bits_count(&s->pb) >> 3;
-    *got_packet_ptr = 1;
-    return 0;
+    if (!data)
+        s->last_frame = 1;
+    memcpy(s->samples, s->samples + 1024 * avctx->channels,
+           1024 * avctx->channels * sizeof(s->samples[0]));
+    return put_bits_count(&s->pb)>>3;
 }

 static av_cold int aac_encode_end(AVCodecContext *avctx)
@@ -677,122 +678,12 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
    ff_mdct_end(&s->mdct1024);
    ff_mdct_end(&s->mdct128);
    ff_psy_end(&s->psy);
-    if (s->psypp)
-        ff_psy_preprocess_end(s->psypp);
-    av_freep(&s->buffer.samples);
+    ff_psy_preprocess_end(s->psypp);
+    av_freep(&s->samples);
    av_freep(&s->cpe);
-    ff_af_queue_close(&s->afq);
-#if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avctx->coded_frame);
-#endif
    return 0;
 }

-static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
-{
-    int ret = 0;
-
-    ff_dsputil_init(&s->dsp, avctx);
-
-    // window init
-    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
-    ff_init_ff_sine_windows(10);
-    ff_init_ff_sine_windows(7);
-
-    if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
-        return ret;
-    if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))
-        return ret;
-
-    return 0;
-}
-
-static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
-{
-    int ch;
-    FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
-
-    for(ch = 0; ch < s->channels; ch++)
-        s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
-
-#if FF_API_OLD_ENCODE_AUDIO
-    if (!(avctx->coded_frame = avcodec_alloc_frame()))
-        goto alloc_fail;
-#endif
-
-    return 0;
-alloc_fail:
-    return AVERROR(ENOMEM);
-}
-
-static av_cold int aac_encode_init(AVCodecContext *avctx)
-{
-    AACEncContext *s = avctx->priv_data;
-    int i, ret = 0;
-    const uint8_t *sizes[2];
-    uint8_t grouping[AAC_MAX_CHANNELS];
-    int lengths[2];
-
-    avctx->frame_size = 1024;
-
-    for (i = 0; i < 16; i++)
-        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
-            break;
-
-    s->channels = avctx->channels;
-
-    ERROR_IF(i == 16,
-             "Unsupported sample rate %d\n", avctx->sample_rate);
-    ERROR_IF(s->channels > AAC_MAX_CHANNELS,
-             "Unsupported number of channels: %d\n", s->channels);
-    ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
-             "Unsupported profile %d\n", avctx->profile);
-    ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
-             "Too many bits per frame requested\n");
-
-    s->samplerate_index = i;
-
-    s->chan_map = aac_chan_configs[s->channels-1];
-
-    if (ret = dsp_init(avctx, s))
-        goto fail;
-
-    if (ret = alloc_buffers(avctx, s))
-        goto fail;
-
-    avctx->extradata_size = 5;
-    put_audio_specific_config(avctx);
-
-    sizes[0]   = swb_size_1024[i];
-    sizes[1]   = swb_size_128[i];
-    lengths[0] = ff_aac_num_swb_1024[i];
-    lengths[1] = ff_aac_num_swb_128[i];
-    for (i = 0; i < s->chan_map[0]; i++)
-        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
-    if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
-        goto fail;
-    s->psypp = ff_psy_preprocess_init(avctx);
-    s->coder = &ff_aac_coders[s->options.aac_coder];
-
-    s->lambda = avctx->global_quality ? avctx->global_quality : 120;
-
-    ff_aac_tableinit();
-
-    for (i = 0; i < 428; i++)
-        ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
-
-    avctx->delay = 1024;
-    ff_af_queue_init(avctx, &s->afq);
-
-    return 0;
-fail:
-    aac_encode_end(avctx);
-    return ret;
-}
-
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
    {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
@@ -816,13 +707,10 @@ AVCodec ff_aac_encoder = {
    .id             = CODEC_ID_AAC,
    .priv_data_size = sizeof(AACEncContext),
    .init           = aac_encode_init,
-    .encode2        = aac_encode_frame,
+    .encode         = aac_encode_frame,
    .close          = aac_encode_end,
-    .supported_samplerates = avpriv_mpeg4audio_sample_rates,
-    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
-                      CODEC_CAP_EXPERIMENTAL,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
-                                                     AV_SAMPLE_FMT_NONE },
-    .long_name      = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
-    .priv_class     = &aacenc_class,
+    .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
+    .priv_class = &aacenc_class,
 };
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -27,7 +27,7 @@
 #include "dsputil.h"

 #include "aac.h"
-#include "audio_frame_queue.h"
+
 #include "psymodel.h"

 #define AAC_CODER_NB 4
@@ -61,10 +61,9 @@ typedef struct AACEncContext {
    FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
    FFTContext mdct128;                          ///< short (128 samples) frame transform context
    DSPContext  dsp;
-    float *planar_samples[6];                    ///< saved preprocessed input
+    int16_t *samples;                            ///< saved preprocessed input

    int samplerate_index;                        ///< MPEG-4 samplerate index
-    int channels;                                ///< channel count
    const uint8_t *chan_map;                     ///< channel configuration map

    ChannelElement *cpe;                         ///< channel elements
@@ -74,15 +73,8 @@ typedef struct AACEncContext {
    int cur_channel;
    int last_frame;
    float lambda;
-    AudioFrameQueue afq;
    DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
    DECLARE_ALIGNED(32, float, scoefs)[1024];    ///< scaled coefficients
-
-    struct {
-        float *samples;
-    } buffer;
 } AACEncContext;

-extern float ff_aac_pow34sf_tab[428];
-
 #endif /* AVCODEC_AACENC_H */
--- a/libavcodec/aacps.c
+++ b/libavcodec/aacps.c
@@ -27,7 +27,6 @@
 #include "aacps.h"
 #include "aacps_tablegen.h"
 #include "aacpsdata.c"
-#include "dsputil.h"

 #define PS_BASELINE 0  ///< Operate in Baseline PS mode
                       ///< Baseline implies 10 or 20 stereo bands,
@@ -224,7 +223,7 @@ int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps
            cnt -= 2 + ps_read_extension_data(gb, ps, ps_extension_id);
        }
        if (cnt < 0) {
-            av_log(avctx, AV_LOG_ERROR, "ps extension overflow %d\n", cnt);
+            av_log(avctx, AV_LOG_ERROR, "ps extension overflow %d", cnt);
            goto err;
        }
        skip_bits(gb, cnt);
@@ -276,16 +275,12 @@ int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps
 err:
    ps->start = 0;
    skip_bits_long(gb_host, bits_left);
-    memset(ps->iid_par, 0, sizeof(ps->iid_par));
-    memset(ps->icc_par, 0, sizeof(ps->icc_par));
-    memset(ps->ipd_par, 0, sizeof(ps->ipd_par));
-    memset(ps->opd_par, 0, sizeof(ps->opd_par));
    return bits_left;
 }

 /** Split one subband into 2 subsubbands with a symmetric real filter.
 * The filter must have its non-center even coefficients equal to zero. */
-static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8], int len, int reverse)
+static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[7], int len, int reverse)
 {
    int i, j;
    for (i = 0; i < len; i++, in++) {
@@ -305,14 +300,26 @@ static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8
 }

 /** Split one subband into 6 subsubbands with a complex filter */
-static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int len)
+static void hybrid6_cx(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int len)
 {
-    int i;
+    int i, j, ssb;
    int N = 8;
-    LOCAL_ALIGNED_16(float, temp, [8], [2]);
+    float temp[8][2];

    for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(temp, in, filter, 1, N);
+        for (ssb = 0; ssb < N; ssb++) {
+            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
+            for (j = 0; j < 6; j++) {
+                float in0_re = in[j][0];
+                float in0_im = in[j][1];
+                float in1_re = in[12-j][0];
+                float in1_im = in[12-j][1];
+                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
+                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
+            }
+            temp[ssb][0] = sum_re;
+            temp[ssb][1] = sum_im;
+        }
        out[0][i][0] = temp[6][0];
        out[0][i][1] = temp[6][1];
        out[1][i][0] = temp[7][0];
@@ -328,18 +335,28 @@ static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], c
    }
 }

-static void hybrid4_8_12_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int N, int len)
+static void hybrid4_8_12_cx(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int N, int len)
 {
-    int i;
+    int i, j, ssb;

    for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(out[0] + i, in, filter, 32, N);
+        for (ssb = 0; ssb < N; ssb++) {
+            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
+            for (j = 0; j < 6; j++) {
+                float in0_re = in[j][0];
+                float in0_im = in[j][1];
+                float in1_re = in[12-j][0];
+                float in1_im = in[12-j][1];
+                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
+                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
+            }
+            out[ssb][i][0] = sum_re;
+            out[ssb][i][1] = sum_im;
+        }
    }
 }

-static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
-                            float in[5][44][2], float L[2][38][64],
-                            int is34, int len)
+static void hybrid_analysis(float out[91][32][2], float in[5][44][2], float L[2][38][64], int is34, int len)
 {
    int i, j;
    for (i = 0; i < 5; i++) {
@@ -349,17 +366,27 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
        }
    }
    if (is34) {
-        hybrid4_8_12_cx(dsp, in[0], out,    f34_0_12, 12, len);
-        hybrid4_8_12_cx(dsp, in[1], out+12, f34_1_8,   8, len);
-        hybrid4_8_12_cx(dsp, in[2], out+20, f34_2_4,   4, len);
-        hybrid4_8_12_cx(dsp, in[3], out+24, f34_2_4,   4, len);
-        hybrid4_8_12_cx(dsp, in[4], out+28, f34_2_4,   4, len);
-        dsp->hybrid_analysis_ileave(out + 27, L, 5, len);
+        hybrid4_8_12_cx(in[0], out,    f34_0_12, 12, len);
+        hybrid4_8_12_cx(in[1], out+12, f34_1_8,   8, len);
+        hybrid4_8_12_cx(in[2], out+20, f34_2_4,   4, len);
+        hybrid4_8_12_cx(in[3], out+24, f34_2_4,   4, len);
+        hybrid4_8_12_cx(in[4], out+28, f34_2_4,   4, len);
+        for (i = 0; i < 59; i++) {
+            for (j = 0; j < len; j++) {
+                out[i+32][j][0] = L[0][j][i+5];
+                out[i+32][j][1] = L[1][j][i+5];
+            }
+        }
    } else {
-        hybrid6_cx(dsp, in[0], out, f20_0_8, len);
+        hybrid6_cx(in[0], out, f20_0_8, len);
        hybrid2_re(in[1], out+6, g1_Q2, len, 1);
        hybrid2_re(in[2], out+8, g1_Q2, len, 0);
-        dsp->hybrid_analysis_ileave(out + 7, L, 3, len);
+        for (i = 0; i < 61; i++) {
+            for (j = 0; j < len; j++) {
+                out[i+10][j][0] = L[0][j][i+3];
+                out[i+10][j][1] = L[1][j][i+3];
+            }
+        }
    }
    //update in_buf
    for (i = 0; i < 5; i++) {
@@ -367,8 +394,7 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
    }
 }

-static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
-                             float in[91][32][2], int is34, int len)
+static void hybrid_synthesis(float out[2][38][64], float in[91][32][2], int is34, int len)
 {
    int i, n;
    if (is34) {
@@ -392,7 +418,12 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
                out[1][n][4] += in[28+i][n][1];
            }
        }
-        dsp->hybrid_synthesis_deint(out, in + 27, 5, len);
+        for (i = 0; i < 59; i++) {
+            for (n = 0; n < len; n++) {
+                out[0][n][i+5] = in[i+32][n][0];
+                out[1][n][i+5] = in[i+32][n][1];
+            }
+        }
    } else {
        for (n = 0; n < len; n++) {
            out[0][n][0] = in[0][n][0] + in[1][n][0] + in[2][n][0] +
@@ -404,7 +435,12 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
            out[0][n][2] = in[8][n][0] + in[9][n][0];
            out[1][n][2] = in[8][n][1] + in[9][n][1];
        }
-        dsp->hybrid_synthesis_deint(out, in + 7, 3, len);
+        for (i = 0; i < 61; i++) {
+            for (n = 0; n < len; n++) {
+                out[0][n][i+3] = in[i+10][n][0];
+                out[1][n][i+3] = in[i+10][n][1];
+            }
+        }
    }
 }

@@ -608,8 +644,8 @@ static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])

 static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[32][2], int is34)
 {
-    LOCAL_ALIGNED_16(float, power, [34], [PS_QMF_TIME_SLOTS]);
-    LOCAL_ALIGNED_16(float, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
+    float power[34][PS_QMF_TIME_SLOTS] = {{0}};
+    float transient_gain[34][PS_QMF_TIME_SLOTS];
    float *peak_decay_nrg = ps->peak_decay_nrg;
    float *power_smooth = ps->power_smooth;
    float *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
@@ -621,8 +657,10 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
    const float a_smooth          = 0.25f; ///< Smoothing coefficient
    int i, k, m, n;
    int n0 = 0, nL = 32;
-
-    memset(power, 0, 34 * sizeof(*power));
+    static const int link_delay[] = { 3, 4, 5 };
+    static const float a[] = { 0.65143905753106f,
+                               0.56471812200776f,
+                               0.48954165955695f };

    if (is34 != ps->is34bands_old) {
        memset(ps->peak_decay_nrg,         0, sizeof(ps->peak_decay_nrg));
@@ -632,9 +670,11 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
        memset(ps->ap_delay,               0, sizeof(ps->ap_delay));
    }

-    for (k = 0; k < NR_BANDS[is34]; k++) {
-        int i = k_to_i[k];
-        ps->dsp.add_squares(power[i], s[k], nL - n0);
+    for (n = n0; n < nL; n++) {
+        for (k = 0; k < NR_BANDS[is34]; k++) {
+            int i = k_to_i[k];
+            power[i][n] += s[k][n][0] * s[k][n][0] + s[k][n][1] * s[k][n][1];
+        }
    }

    //Transient detection
@@ -662,31 +702,54 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
    for (k = 0; k < NR_ALLPASS_BANDS[is34]; k++) {
        int b = k_to_i[k];
        float g_decay_slope = 1.f - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
+        float ag[PS_AP_LINKS];
        g_decay_slope = av_clipf(g_decay_slope, 0.f, 1.f);
        memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
        memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
        for (m = 0; m < PS_AP_LINKS; m++) {
            memcpy(ap_delay[k][m],   ap_delay[k][m]+numQMFSlots,           5*sizeof(ap_delay[k][m][0]));
+            ag[m] = a[m] * g_decay_slope;
+        }
+        for (n = n0; n < nL; n++) {
+            float in_re = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][0] -
+                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][1];
+            float in_im = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][1] +
+                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][0];
+            for (m = 0; m < PS_AP_LINKS; m++) {
+                float a_re                = ag[m] * in_re;
+                float a_im                = ag[m] * in_im;
+                float link_delay_re       = ap_delay[k][m][n+5-link_delay[m]][0];
+                float link_delay_im       = ap_delay[k][m][n+5-link_delay[m]][1];
+                float fractional_delay_re = Q_fract_allpass[is34][k][m][0];
+                float fractional_delay_im = Q_fract_allpass[is34][k][m][1];
+                ap_delay[k][m][n+5][0] = in_re;
+                ap_delay[k][m][n+5][1] = in_im;
+                in_re = link_delay_re * fractional_delay_re - link_delay_im * fractional_delay_im - a_re;
+                in_im = link_delay_re * fractional_delay_im + link_delay_im * fractional_delay_re - a_im;
+                ap_delay[k][m][n+5][0] += ag[m] * in_re;
+                ap_delay[k][m][n+5][1] += ag[m] * in_im;
+            }
+            out[k][n][0] = transient_gain[b][n] * in_re;
+            out[k][n][1] = transient_gain[b][n] * in_im;
        }
-        ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
-                            phi_fract[is34][k], Q_fract_allpass[is34][k],
-                            transient_gain[b], g_decay_slope, nL - n0);
    }
    for (; k < SHORT_DELAY_BAND[is34]; k++) {
-        int i = k_to_i[k];
        memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
        memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
-        //H = delay 14
-        ps->dsp.mul_pair_single(out[k], delay[k] + PS_MAX_DELAY - 14,
-                                transient_gain[i], nL - n0);
+        for (n = n0; n < nL; n++) {
+            //H = delay 14
+            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][0];
+            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][1];
+        }
    }
    for (; k < NR_BANDS[is34]; k++) {
-        int i = k_to_i[k];
        memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
        memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
-        //H = delay 1
-        ps->dsp.mul_pair_single(out[k], delay[k] + PS_MAX_DELAY - 1,
-                                transient_gain[i], nL - n0);
+        for (n = n0; n < nL; n++) {
+            //H = delay 1
+            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][0];
+            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][1];
+        }
    }
 }

@@ -730,7 +793,7 @@ static void remap20(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],

 static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2], int is34)
 {
-    int e, b, k;
+    int e, b, k, n;

    float (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
    float (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
@@ -842,52 +905,78 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
            H22[0][e+1][b] = h22;
        }
        for (k = 0; k < NR_BANDS[is34]; k++) {
-            float h[2][4];
-            float h_step[2][4];
+            float h11r, h12r, h21r, h22r;
+            float h11i, h12i, h21i, h22i;
+            float h11r_step, h12r_step, h21r_step, h22r_step;
+            float h11i_step, h12i_step, h21i_step, h22i_step;
            int start = ps->border_position[e];
            int stop  = ps->border_position[e+1];
            float width = 1.f / (stop - start);
            b = k_to_i[k];
-            h[0][0] = H11[0][e][b];
-            h[0][1] = H12[0][e][b];
-            h[0][2] = H21[0][e][b];
-            h[0][3] = H22[0][e][b];
+            h11r = H11[0][e][b];
+            h12r = H12[0][e][b];
+            h21r = H21[0][e][b];
+            h22r = H22[0][e][b];
            if (!PS_BASELINE && ps->enable_ipdopd) {
            //Is this necessary? ps_04_new seems unchanged
            if ((is34 && k <= 13 && k >= 9) || (!is34 && k <= 1)) {
-                h[1][0] = -H11[1][e][b];
-                h[1][1] = -H12[1][e][b];
-                h[1][2] = -H21[1][e][b];
-                h[1][3] = -H22[1][e][b];
+                h11i = -H11[1][e][b];
+                h12i = -H12[1][e][b];
+                h21i = -H21[1][e][b];
+                h22i = -H22[1][e][b];
            } else {
-                h[1][0] = H11[1][e][b];
-                h[1][1] = H12[1][e][b];
-                h[1][2] = H21[1][e][b];
-                h[1][3] = H22[1][e][b];
+                h11i = H11[1][e][b];
+                h12i = H12[1][e][b];
+                h21i = H21[1][e][b];
+                h22i = H22[1][e][b];
            }
            }
            //Interpolation
-            h_step[0][0] = (H11[0][e+1][b] - h[0][0]) * width;
-            h_step[0][1] = (H12[0][e+1][b] - h[0][1]) * width;
-            h_step[0][2] = (H21[0][e+1][b] - h[0][2]) * width;
-            h_step[0][3] = (H22[0][e+1][b] - h[0][3]) * width;
+            h11r_step = (H11[0][e+1][b] - h11r) * width;
+            h12r_step = (H12[0][e+1][b] - h12r) * width;
+            h21r_step = (H21[0][e+1][b] - h21r) * width;
+            h22r_step = (H22[0][e+1][b] - h22r) * width;
            if (!PS_BASELINE && ps->enable_ipdopd) {
-                h_step[1][0] = (H11[1][e+1][b] - h[1][0]) * width;
-                h_step[1][1] = (H12[1][e+1][b] - h[1][1]) * width;
-                h_step[1][2] = (H21[1][e+1][b] - h[1][2]) * width;
-                h_step[1][3] = (H22[1][e+1][b] - h[1][3]) * width;
+                h11i_step = (H11[1][e+1][b] - h11i) * width;
+                h12i_step = (H12[1][e+1][b] - h12i) * width;
+                h21i_step = (H21[1][e+1][b] - h21i) * width;
+                h22i_step = (H22[1][e+1][b] - h22i) * width;
+            }
+            for (n = start + 1; n <= stop; n++) {
+                //l is s, r is d
+                float l_re = l[k][n][0];
+                float l_im = l[k][n][1];
+                float r_re = r[k][n][0];
+                float r_im = r[k][n][1];
+                h11r += h11r_step;
+                h12r += h12r_step;
+                h21r += h21r_step;
+                h22r += h22r_step;
+                if (!PS_BASELINE && ps->enable_ipdopd) {
+                    h11i += h11i_step;
+                    h12i += h12i_step;
+                    h21i += h21i_step;
+                    h22i += h22i_step;
+
+                    l[k][n][0] = h11r*l_re + h21r*r_re - h11i*l_im - h21i*r_im;
+                    l[k][n][1] = h11r*l_im + h21r*r_im + h11i*l_re + h21i*r_re;
+                    r[k][n][0] = h12r*l_re + h22r*r_re - h12i*l_im - h22i*r_im;
+                    r[k][n][1] = h12r*l_im + h22r*r_im + h12i*l_re + h22i*r_re;
+                } else {
+                    l[k][n][0] = h11r*l_re + h21r*r_re;
+                    l[k][n][1] = h11r*l_im + h21r*r_im;
+                    r[k][n][0] = h12r*l_re + h22r*r_re;
+                    r[k][n][1] = h12r*l_im + h22r*r_im;
+                }
            }
-            ps->dsp.stereo_interpolate[!PS_BASELINE && ps->enable_ipdopd](
-                l[k] + start + 1, r[k] + start + 1,
-                h, h_step, stop - start);
        }
    }
 }

 int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top)
 {
-    LOCAL_ALIGNED_16(float, Lbuf, [91], [32][2]);
-    LOCAL_ALIGNED_16(float, Rbuf, [91], [32][2]);
+    float Lbuf[91][32][2];
+    float Rbuf[91][32][2];
    const int len = 32;
    int is34 = ps->is34bands;

@@ -896,11 +985,11 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
    if (top < NR_ALLPASS_BANDS[is34])
        memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));

-    hybrid_analysis(&ps->dsp, Lbuf, ps->in_buf, L, is34, len);
+    hybrid_analysis(Lbuf, ps->in_buf, L, is34, len);
    decorrelation(ps, Rbuf, Lbuf, is34);
    stereo_processing(ps, Lbuf, Rbuf, is34);
-    hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
-    hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);
+    hybrid_synthesis(L, Lbuf, is34, len);
+    hybrid_synthesis(R, Rbuf, is34, len);

    return 0;
 }
@@ -948,5 +1037,4 @@ av_cold void ff_ps_init(void) {

 av_cold void ff_ps_ctx_init(PSContext *ps)
 {
-    ff_psdsp_init(&ps->dsp);
 }
--- a/libavcodec/aacps.h
+++ b/libavcodec/aacps.h
@@ -24,7 +24,6 @@

 #include <stdint.h>

-#include "aacpsdsp.h"
 #include "avcodec.h"
 #include "get_bits.h"

@@ -61,19 +60,18 @@ typedef struct {
    int    is34bands;
    int    is34bands_old;

-    DECLARE_ALIGNED(16, float, in_buf)[5][44][2];
-    DECLARE_ALIGNED(16, float, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
-    DECLARE_ALIGNED(16, float, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
-    DECLARE_ALIGNED(16, float, peak_decay_nrg)[34];
-    DECLARE_ALIGNED(16, float, power_smooth)[34];
-    DECLARE_ALIGNED(16, float, peak_decay_diff_smooth)[34];
-    DECLARE_ALIGNED(16, float, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  in_buf[5][44][2];
+    float  delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
+    float  ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
+    float  peak_decay_nrg[34];
+    float  power_smooth[34];
+    float  peak_decay_diff_smooth[34];
+    float  H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
    int8_t opd_hist[PS_MAX_NR_IIDICC];
    int8_t ipd_hist[PS_MAX_NR_IIDICC];
-    PSDSPContext dsp;
 } PSContext;

 void ff_ps_init(void);
--- a/libavcodec/aacps_tablegen.c
+++ b/libavcodec/aacps_tablegen.c
@@ -69,23 +69,23 @@ int main(void)
    write_float_3d_array(HB, 46, 8, 4);
    printf("};\n");

-    printf("static const DECLARE_ALIGNED(16, float, f20_0_8)[8][8][2] = {\n");
-    write_float_3d_array(f20_0_8, 8, 8, 2);
+    printf("static const float f20_0_8[8][7][2] = {\n");
+    write_float_3d_array(f20_0_8, 8, 7, 2);
    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2] = {\n");
-    write_float_3d_array(f34_0_12, 12, 8, 2);
+    printf("static const float f34_0_12[12][7][2] = {\n");
+    write_float_3d_array(f34_0_12, 12, 7, 2);
    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_1_8)[8][8][2] = {\n");
-    write_float_3d_array(f34_1_8, 8, 8, 2);
+    printf("static const float f34_1_8[8][7][2] = {\n");
+    write_float_3d_array(f34_1_8, 8, 7, 2);
    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_2_4)[4][8][2] = {\n");
-    write_float_3d_array(f34_2_4, 4, 8, 2);
+    printf("static const float f34_2_4[4][7][2] = {\n");
+    write_float_3d_array(f34_2_4, 4, 7, 2);
    printf("};\n");

-    printf("static const DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
+    printf("static const float Q_fract_allpass[2][50][3][2] = {\n");
    write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, phi_fract)[2][50][2] = {\n");
+    printf("static const float phi_fract[2][50][2] = {\n");
    write_float_3d_array(phi_fract, 2, 50, 2);
    printf("};\n");

--- a/libavcodec/aacps_tablegen.h
+++ b/libavcodec/aacps_tablegen.h
@@ -31,7 +31,6 @@
 #else
 #include "libavutil/common.h"
 #include "libavutil/mathematics.h"
-#include "libavutil/mem.h"
 #define NR_ALLPASS_BANDS20 30
 #define NR_ALLPASS_BANDS34 50
 #define PS_AP_LINKS 3
@@ -39,12 +38,12 @@ static float pd_re_smooth[8*8*8];
 static float pd_im_smooth[8*8*8];
 static float HA[46][8][4];
 static float HB[46][8][4];
-static DECLARE_ALIGNED(16, float, f20_0_8) [ 8][8][2];
-static DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2];
-static DECLARE_ALIGNED(16, float, f34_1_8) [ 8][8][2];
-static DECLARE_ALIGNED(16, float, f34_2_4) [ 4][8][2];
-static DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2];
-static DECLARE_ALIGNED(16, float, phi_fract)[2][50][2];
+static float f20_0_8 [ 8][7][2];
+static float f34_0_12[12][7][2];
+static float f34_1_8 [ 8][7][2];
+static float f34_2_4 [ 4][7][2];
+static float Q_fract_allpass[2][50][3][2];
+static float phi_fract[2][50][2];

 static const float g0_Q8[] = {
    0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
@@ -66,7 +65,7 @@ static const float g2_Q4[] = {
     0.16486303567403f,  0.23279856662996f, 0.25f
 };

-static void make_filters_from_proto(float (*filter)[8][2], const float *proto, int bands)
+static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
 {
    int q, n;
    for (q = 0; q < bands; q++) {
--- a/libavcodec/aacpsdsp.c
+++ b/libavcodec/aacpsdsp.c
@@ -1,214 +0,0 @@
-/*
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of Libav.
- *
- * 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.
- *
- * 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "config.h"
-#include "libavutil/attributes.h"
-#include "aacpsdsp.h"
-
-static void ps_add_squares_c(float *dst, const float (*src)[2], int n)
-{
-    int i;
-    for (i = 0; i < n; i++)
-        dst[i] += src[i][0] * src[i][0] + src[i][1] * src[i][1];
-}
-
-static void ps_mul_pair_single_c(float (*dst)[2], float (*src0)[2], float *src1,
-                                 int n)
-{
-    int i;
-    for (i = 0; i < n; i++) {
-        dst[i][0] = src0[i][0] * src1[i];
-        dst[i][1] = src0[i][1] * src1[i];
-    }
-}
-
-static void ps_hybrid_analysis_c(float (*out)[2], float (*in)[2],
-                                 const float (*filter)[8][2],
-                                 int stride, int n)
-{
-    int i, j;
-
-    for (i = 0; i < n; i++) {
-        float sum_re = filter[i][6][0] * in[6][0];
-        float sum_im = filter[i][6][0] * in[6][1];
-
-        for (j = 0; j < 6; j++) {
-            float in0_re = in[j][0];
-            float in0_im = in[j][1];
-            float in1_re = in[12-j][0];
-            float in1_im = in[12-j][1];
-            sum_re += filter[i][j][0] * (in0_re + in1_re) -
-                      filter[i][j][1] * (in0_im - in1_im);
-            sum_im += filter[i][j][0] * (in0_im + in1_im) +
-                      filter[i][j][1] * (in0_re - in1_re);
-        }
-        out[i * stride][0] = sum_re;
-        out[i * stride][1] = sum_im;
-    }
-}
-
-static void ps_hybrid_analysis_ileave_c(float (*out)[32][2], float L[2][38][64],
-                                        int i, int len)
-{
-    int j;
-
-    for (; i < 64; i++) {
-        for (j = 0; j < len; j++) {
-            out[i][j][0] = L[0][j][i];
-            out[i][j][1] = L[1][j][i];
-        }
-    }
-}
-
-static void ps_hybrid_synthesis_deint_c(float out[2][38][64],
-                                        float (*in)[32][2],
-                                        int i, int len)
-{
-    int n;
-
-    for (; i < 64; i++) {
-        for (n = 0; n < len; n++) {
-            out[0][n][i] = in[i][n][0];
-            out[1][n][i] = in[i][n][1];
-        }
-    }
-}
-
-static void ps_decorrelate_c(float (*out)[2], float (*delay)[2],
-                             float (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
-                             const float phi_fract[2], float (*Q_fract)[2],
-                             const float *transient_gain,
-                             float g_decay_slope,
-                             int len)
-{
-    static const float a[] = { 0.65143905753106f,
-                               0.56471812200776f,
-                               0.48954165955695f };
-    float ag[PS_AP_LINKS];
-    int m, n;
-
-    for (m = 0; m < PS_AP_LINKS; m++)
-        ag[m] = a[m] * g_decay_slope;
-
-    for (n = 0; n < len; n++) {
-        float in_re = delay[n][0] * phi_fract[0] - delay[n][1] * phi_fract[1];
-        float in_im = delay[n][0] * phi_fract[1] + delay[n][1] * phi_fract[0];
-        for (m = 0; m < PS_AP_LINKS; m++) {
-            float a_re                = ag[m] * in_re;
-            float a_im                = ag[m] * in_im;
-            float link_delay_re       = ap_delay[m][n+2-m][0];
-            float link_delay_im       = ap_delay[m][n+2-m][1];
-            float fractional_delay_re = Q_fract[m][0];
-            float fractional_delay_im = Q_fract[m][1];
-            float apd_re = in_re;
-            float apd_im = in_im;
-            in_re = link_delay_re * fractional_delay_re -
-                    link_delay_im * fractional_delay_im - a_re;
-            in_im = link_delay_re * fractional_delay_im +
-                    link_delay_im * fractional_delay_re - a_im;
-            ap_delay[m][n+5][0] = apd_re + ag[m] * in_re;
-            ap_delay[m][n+5][1] = apd_im + ag[m] * in_im;
-        }
-        out[n][0] = transient_gain[n] * in_re;
-        out[n][1] = transient_gain[n] * in_im;
-    }
-}
-
-static void ps_stereo_interpolate_c(float (*l)[2], float (*r)[2],
-                                    float h[2][4], float h_step[2][4],
-                                    int len)
-{
-    float h0 = h[0][0];
-    float h1 = h[0][1];
-    float h2 = h[0][2];
-    float h3 = h[0][3];
-    float hs0 = h_step[0][0];
-    float hs1 = h_step[0][1];
-    float hs2 = h_step[0][2];
-    float hs3 = h_step[0][3];
-    int n;
-
-    for (n = 0; n < len; n++) {
-        //l is s, r is d
-        float l_re = l[n][0];
-        float l_im = l[n][1];
-        float r_re = r[n][0];
-        float r_im = r[n][1];
-        h0 += hs0;
-        h1 += hs1;
-        h2 += hs2;
-        h3 += hs3;
-        l[n][0] = h0 * l_re + h2 * r_re;
-        l[n][1] = h0 * l_im + h2 * r_im;
-        r[n][0] = h1 * l_re + h3 * r_re;
-        r[n][1] = h1 * l_im + h3 * r_im;
-    }
-}
-
-static void ps_stereo_interpolate_ipdopd_c(float (*l)[2], float (*r)[2],
-                                           float h[2][4], float h_step[2][4],
-                                           int len)
-{
-    float h00  = h[0][0],      h10  = h[1][0];
-    float h01  = h[0][1],      h11  = h[1][1];
-    float h02  = h[0][2],      h12  = h[1][2];
-    float h03  = h[0][3],      h13  = h[1][3];
-    float hs00 = h_step[0][0], hs10 = h_step[1][0];
-    float hs01 = h_step[0][1], hs11 = h_step[1][1];
-    float hs02 = h_step[0][2], hs12 = h_step[1][2];
-    float hs03 = h_step[0][3], hs13 = h_step[1][3];
-    int n;
-
-    for (n = 0; n < len; n++) {
-        //l is s, r is d
-        float l_re = l[n][0];
-        float l_im = l[n][1];
-        float r_re = r[n][0];
-        float r_im = r[n][1];
-        h00 += hs00;
-        h01 += hs01;
-        h02 += hs02;
-        h03 += hs03;
-        h10 += hs10;
-        h11 += hs11;
-        h12 += hs12;
-        h13 += hs13;
-
-        l[n][0] = h00 * l_re + h02 * r_re - h10 * l_im - h12 * r_im;
-        l[n][1] = h00 * l_im + h02 * r_im + h10 * l_re + h12 * r_re;
-        r[n][0] = h01 * l_re + h03 * r_re - h11 * l_im - h13 * r_im;
-        r[n][1] = h01 * l_im + h03 * r_im + h11 * l_re + h13 * r_re;
-    }
-}
-
-av_cold void ff_psdsp_init(PSDSPContext *s)
-{
-    s->add_squares            = ps_add_squares_c;
-    s->mul_pair_single        = ps_mul_pair_single_c;
-    s->hybrid_analysis        = ps_hybrid_analysis_c;
-    s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
-    s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
-    s->decorrelate            = ps_decorrelate_c;
-    s->stereo_interpolate[0]  = ps_stereo_interpolate_c;
-    s->stereo_interpolate[1]  = ps_stereo_interpolate_ipdopd_c;
-
-    if (ARCH_ARM)
-        ff_psdsp_init_arm(s);
-}
--- a/libavcodec/aacpsdsp.h
+++ b/libavcodec/aacpsdsp.h
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2012 Mans Rullgard
- *
- * This file is part of Libav.
- *
- * 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.
- *
- * 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef LIBAVCODEC_AACPSDSP_H
-#define LIBAVCODEC_AACPSDSP_H
-
-#define PS_QMF_TIME_SLOTS 32
-#define PS_AP_LINKS 3
-#define PS_MAX_AP_DELAY 5
-
-typedef struct PSDSPContext {
-    void (*add_squares)(float *dst, const float (*src)[2], int n);
-    void (*mul_pair_single)(float (*dst)[2], float (*src0)[2], float *src1,
-                            int n);
-    void (*hybrid_analysis)(float (*out)[2], float (*in)[2],
-                            const float (*filter)[8][2],
-                            int stride, int n);
-    void (*hybrid_analysis_ileave)(float (*out)[32][2], float L[2][38][64],
-                                   int i, int len);
-    void (*hybrid_synthesis_deint)(float out[2][38][64], float (*in)[32][2],
-                                   int i, int len);
-    void (*decorrelate)(float (*out)[2], float (*delay)[2],
-                        float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
-                        const float phi_fract[2], float (*Q_fract)[2],
-                        const float *transient_gain,
-                        float g_decay_slope,
-                        int len);
-    void (*stereo_interpolate[2])(float (*l)[2], float (*r)[2],
-                                  float h[2][4], float h_step[2][4],
-                                  int len);
-} PSDSPContext;
-
-void ff_psdsp_init(PSDSPContext *s);
-void ff_psdsp_init_arm(PSDSPContext *s);
-
-#endif /* LIBAVCODEC_AACPSDSP_H */
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -216,7 +216,7 @@ static const float psy_fir_coeffs[] = {
 };

 /**
- * Calculate the ABR attack threshold from the above LAME psymodel table.
+ * calculates the attack threshold for ABR from the above table for the LAME psy model
 */
 static float lame_calc_attack_threshold(int bitrate)
 {
@@ -389,8 +389,9 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
    AacPsyChannel *pch  = &pctx->ch[channel];
    uint8_t grouping     = 0;
    int next_type        = pch->next_window_seq;
-    FFPsyWindowInfo wi  = { { 0 } };
+    FFPsyWindowInfo wi;

+    memset(&wi, 0, sizeof(wi));
    if (la) {
        float s[8], v;
        int switch_to_eight = 0;
@@ -399,7 +400,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
        int stay_short = 0;
        for (i = 0; i < 8; i++) {
            for (j = 0; j < 128; j++) {
-                v = iir_filter(la[i*128+j], pch->iir_state);
+                v = iir_filter(la[(i*128+j)*ctx->avctx->channels], pch->iir_state);
                sum += v*v;
            }
            s[i]  = sum;
@@ -775,8 +776,9 @@ static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int u
    ctx->next_window_seq = blocktype;
 }

-static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
-                                       const float *la, int channel, int prev_type)
+static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
+                                       const int16_t *audio, const int16_t *la,
+                                       int channel, int prev_type)
 {
    AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
    AacPsyChannel *pch  = &pctx->ch[channel];
@@ -784,28 +786,29 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
    int uselongblock = 1;
    int attacks[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
    int i;
-    FFPsyWindowInfo wi = { { 0 } };
+    FFPsyWindowInfo wi;

+    memset(&wi, 0, sizeof(wi));
    if (la) {
        float hpfsmpl[AAC_BLOCK_SIZE_LONG];
        float const *pf = hpfsmpl;
        float attack_intensity[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
        float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
        float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
-        const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN);
+        int chans = ctx->avctx->channels;
+        const int16_t *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN) * chans;
        int j, att_sum = 0;

        /* LAME comment: apply high pass filter of fs/4 */
        for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
            float sum1, sum2;
-            sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
+            sum1 = firbuf[(i + ((PSY_LAME_FIR_LEN - 1) / 2)) * chans];
            sum2 = 0.0;
            for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
-                sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
-                sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
+                sum1 += psy_fir_coeffs[j] * (firbuf[(i + j) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j) * chans]);
+                sum2 += psy_fir_coeffs[j + 1] * (firbuf[(i + j + 1) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j - 1) * chans]);
            }
-            /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768. Tuning this for normalized floats would be difficult. */
-            hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
+            hpfsmpl[i] = sum1 + sum2;
        }

        /* Calculate the energies of each sub-shortblock */
@@ -820,15 +823,16 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
            float const *const pfe = pf + AAC_BLOCK_SIZE_LONG / (AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS);
            float p = 1.0f;
            for (; pf < pfe; pf++)
-                p = FFMAX(p, fabsf(*pf));
+                if (p < fabsf(*pf))
+                    p = fabsf(*pf);
            pch->prev_energy_subshort[i] = energy_subshort[i + PSY_LAME_NUM_SUBBLOCKS] = p;
            energy_short[1 + i / PSY_LAME_NUM_SUBBLOCKS] += p;
-            /* NOTE: The indexes below are [i + 3 - 2] in the LAME source.
-             *       Obviously the 3 and 2 have some significance, or this would be just [i + 1]
-             *       (which is what we use here). What the 3 stands for is ambiguous, as it is both
-             *       number of short blocks, and the number of sub-short blocks.
-             *       It seems that LAME is comparing each sub-block to sub-block + 1 in the
-             *       previous block.
+            /* FIXME: The indexes below are [i + 3 - 2] in the LAME source.
+             *          Obviously the 3 and 2 have some significance, or this would be just [i + 1]
+             *          (which is what we use here). What the 3 stands for is ambigious, as it is both
+             *          number of short blocks, and the number of sub-short blocks.
+             *          It seems that LAME is comparing each sub-block to sub-block + 1 in the
+             *          previous block.
             */
            if (p > energy_subshort[i + 1])
                p = p / energy_subshort[i + 1];
--- a/libavcodec/aacsbr.c
+++ b/libavcodec/aacsbr.c
@@ -32,7 +32,6 @@
 #include "aacsbrdata.h"
 #include "fft.h"
 #include "aacps.h"
-#include "sbrdsp.h"
 #include "libavutil/libm.h"
 #include "libavutil/avassert.h"

@@ -129,24 +128,13 @@ av_cold void ff_aac_sbr_init(void)
    ff_ps_init();
 }

-/** Places SBR in pure upsampling mode. */
-static void sbr_turnoff(SpectralBandReplication *sbr) {
-    sbr->start = 0;
-    // Init defults used in pure upsampling mode
-    sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
-    sbr->m[1] = 0;
-    // Reset values for first SBR header
-    sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
-    memset(&sbr->spectrum_params, -1, sizeof(SpectrumParameters));
-}
-
 av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
 {
    float mdct_scale;
    if(sbr->mdct.mdct_bits)
        return;
-    sbr->kx[0] = sbr->kx[1];
-    sbr_turnoff(sbr);
+    sbr->kx[0] = sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
+    sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
    sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
    sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
    /* SBR requires samples to be scaled to +/-32768.0 to work correctly.
@@ -156,7 +144,6 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
    ff_mdct_init(&sbr->mdct,     7, 1, 1.0 / (64 * mdct_scale));
    ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
    ff_ps_ctx_init(&sbr->ps);
-    ff_sbrdsp_init(&sbr->dsp);
 }

 av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
@@ -918,7 +905,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
 {
    switch (bs_extension_id) {
    case EXTENSION_ID_PS:
-        if (!ac->oc[1].m4ac.ps) {
+        if (!ac->m4ac.ps) {
            av_log(ac->avctx, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
            skip_bits_long(gb, *num_bits_left); // bs_fill_bits
            *num_bits_left = 0;
@@ -933,9 +920,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
        }
        break;
    default:
-        // some files contain 0-padding
-        if (bs_extension_id || *num_bits_left > 16 || show_bits(gb, *num_bits_left))
-            av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
+        av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
        skip_bits_long(gb, *num_bits_left); // bs_fill_bits
        *num_bits_left = 0;
        break;
@@ -1011,18 +996,18 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,

    if (id_aac == TYPE_SCE || id_aac == TYPE_CCE) {
        if (read_sbr_single_channel_element(ac, sbr, gb)) {
-            sbr_turnoff(sbr);
+            sbr->start = 0;
            return get_bits_count(gb) - cnt;
        }
    } else if (id_aac == TYPE_CPE) {
        if (read_sbr_channel_pair_element(ac, sbr, gb)) {
-            sbr_turnoff(sbr);
+            sbr->start = 0;
            return get_bits_count(gb) - cnt;
        }
    } else {
        av_log(ac->avctx, AV_LOG_ERROR,
            "Invalid bitstream - cannot apply SBR to element type %d\n", id_aac);
-        sbr_turnoff(sbr);
+        sbr->start = 0;
        return get_bits_count(gb) - cnt;
    }
    if (get_bits1(gb)) { // bs_extended_data
@@ -1054,7 +1039,7 @@ static void sbr_reset(AACContext *ac, SpectralBandReplication *sbr)
    if (err < 0) {
        av_log(ac->avctx, AV_LOG_ERROR,
               "SBR reset failed. Switching SBR to pure upsampling mode.\n");
-        sbr_turnoff(sbr);
+        sbr->start = 0;
    }
 }

@@ -1077,9 +1062,9 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
    sbr->reset = 0;

    if (!sbr->sample_rate)
-        sbr->sample_rate = 2 * ac->oc[1].m4ac.sample_rate; //TODO use the nominal sample rate for arbitrary sample rate support
-    if (!ac->oc[1].m4ac.ext_sample_rate)
-        ac->oc[1].m4ac.ext_sample_rate = 2 * ac->oc[1].m4ac.sample_rate;
+        sbr->sample_rate = 2 * ac->m4ac.sample_rate; //TODO use the nominal sample rate for arbitrary sample rate support
+    if (!ac->m4ac.ext_sample_rate)
+        ac->m4ac.ext_sample_rate = 2 * ac->m4ac.sample_rate;

    if (crc) {
        skip_bits(gb, 10); // bs_sbr_crc_bits; TODO - implement CRC check
@@ -1089,7 +1074,6 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
    //Save some state from the previous frame.
    sbr->kx[0] = sbr->kx[1];
    sbr->m[0] = sbr->m[1];
-    sbr->kx_and_m_pushed = 1;

    num_sbr_bits++;
    if (get_bits1(gb)) // bs_header_flag
@@ -1159,21 +1143,33 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
 * @param   x       pointer to the beginning of the first sample window
 * @param   W       array of complex-valued samples split into subbands
 */
-static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct,
-                             SBRDSPContext *sbrdsp, const float *in, float *x,
+static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in, float *x,
                             float z[320], float W[2][32][32][2])
 {
-    int i;
+    int i, k;
    memcpy(W[0], W[1], sizeof(W[0]));
    memcpy(x    , x+1024, (320-32)*sizeof(x[0]));
    memcpy(x+288, in,         1024*sizeof(x[0]));
    for (i = 0; i < 32; i++) { // numTimeSlots*RATE = 16*2 as 960 sample frames
                               // are not supported
        dsp->vector_fmul_reverse(z, sbr_qmf_window_ds, x, 320);
-        sbrdsp->sum64x5(z);
-        sbrdsp->qmf_pre_shuffle(z);
+        for (k = 0; k < 64; k++) {
+            float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
+            z[k] = f;
+        }
+        //Shuffle to IMDCT
+        z[64] = z[0];
+        for (k = 1; k < 32; k++) {
+            z[64+2*k-1] =  z[   k];
+            z[64+2*k  ] = -z[64-k];
+        }
+        z[64+63] = z[32];
+
        mdct->imdct_half(mdct, z, z+64);
-        sbrdsp->qmf_post_shuffle(W[1][i], z);
+        for (k = 0; k < 32; k++) {
+            W[1][i][k][0] = -z[63-k];
+            W[1][i][k][1] = z[k];
+        }
        x += 32;
    }
 }
@@ -1183,22 +1179,20 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct,
 * (14496-3 sp04 p206)
 */
 static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
-                              SBRDSPContext *sbrdsp,
                              float *out, float X[2][38][64],
                              float mdct_buf[2][64],
                              float *v0, int *v_off, const unsigned int div)
 {
    int i, n;
    const float *sbr_qmf_window = div ? sbr_qmf_window_ds : sbr_qmf_window_us;
-    const int step = 128 >> div;
    float *v;
    for (i = 0; i < 32; i++) {
-        if (*v_off < step) {
+        if (*v_off < 128 >> div) {
            int saved_samples = (1280 - 128) >> div;
            memcpy(&v0[SBR_SYNTHESIS_BUF_SIZE - saved_samples], v0, saved_samples * sizeof(float));
-            *v_off = SBR_SYNTHESIS_BUF_SIZE - saved_samples - step;
+            *v_off = SBR_SYNTHESIS_BUF_SIZE - saved_samples - (128 >> div);
        } else {
-            *v_off -= step;
+            *v_off -= 128 >> div;
        }
        v = v0 + *v_off;
        if (div) {
@@ -1207,12 +1201,20 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
                X[0][i][32+n] =  X[1][i][31-n];
            }
            mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
-            sbrdsp->qmf_deint_neg(v, mdct_buf[0]);
+            for (n = 0; n < 32; n++) {
+                v[     n] =  mdct_buf[0][63 - 2*n];
+                v[63 - n] = -mdct_buf[0][62 - 2*n];
+            }
        } else {
-            sbrdsp->neg_odd_64(X[1][i]);
+            for (n = 1; n < 64; n+=2) {
+                X[1][i][n] = -X[1][i][n];
+            }
            mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
            mdct->imdct_half(mdct, mdct_buf[1], X[1][i]);
-            sbrdsp->qmf_deint_bfly(v, mdct_buf[1], mdct_buf[0]);
+            for (n = 0; n < 64; n++) {
+                v[      n] = -mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+                v[127 - n] =  mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+            }
        }
        dsp->vector_fmul_add(out, v                , sbr_qmf_window               , zero64, 64 >> div);
        dsp->vector_fmul_add(out, v + ( 192 >> div), sbr_qmf_window + ( 64 >> div), out   , 64 >> div);
@@ -1228,20 +1230,45 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
    }
 }

+static void autocorrelate(const float x[40][2], float phi[3][2][2], int lag)
+{
+    int i;
+    float real_sum = 0.0f;
+    float imag_sum = 0.0f;
+    if (lag) {
+        for (i = 1; i < 38; i++) {
+            real_sum += x[i][0] * x[i+lag][0] + x[i][1] * x[i+lag][1];
+            imag_sum += x[i][0] * x[i+lag][1] - x[i][1] * x[i+lag][0];
+        }
+        phi[2-lag][1][0] = real_sum + x[ 0][0] * x[lag][0] + x[ 0][1] * x[lag][1];
+        phi[2-lag][1][1] = imag_sum + x[ 0][0] * x[lag][1] - x[ 0][1] * x[lag][0];
+        if (lag == 1) {
+            phi[0][0][0] = real_sum + x[38][0] * x[39][0] + x[38][1] * x[39][1];
+            phi[0][0][1] = imag_sum + x[38][0] * x[39][1] - x[38][1] * x[39][0];
+        }
+    } else {
+        for (i = 1; i < 38; i++) {
+            real_sum += x[i][0] * x[i][0] + x[i][1] * x[i][1];
+        }
+        phi[2][1][0] = real_sum + x[ 0][0] * x[ 0][0] + x[ 0][1] * x[ 0][1];
+        phi[1][0][0] = real_sum + x[38][0] * x[38][0] + x[38][1] * x[38][1];
+    }
+}
+
 /** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
 * (14496-3 sp04 p214)
 * Warning: This routine does not seem numerically stable.
 */
-static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
-                                  float (*alpha0)[2], float (*alpha1)[2],
+static void sbr_hf_inverse_filter(float (*alpha0)[2], float (*alpha1)[2],
                                  const float X_low[32][40][2], int k0)
 {
    int k;
    for (k = 0; k < k0; k++) {
-        LOCAL_ALIGNED_16(float, phi, [3], [2][2]);
-        float dk;
+        float phi[3][2][2], dk;

-        dsp->autocorrelate(X_low[k], phi);
+        autocorrelate(X_low[k], phi, 0);
+        autocorrelate(X_low[k], phi, 1);
+        autocorrelate(X_low[k], phi, 2);

        dk =  phi[2][1][0] * phi[1][0][0] -
             (phi[1][1][0] * phi[1][1][0] + phi[1][1][1] * phi[1][1][1]) / 1.000001f;
@@ -1337,11 +1364,12 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
                      const float bw_array[5], const uint8_t *t_env,
                      int bs_num_env)
 {
-    int j, x;
+    int i, j, x;
    int g = 0;
    int k = sbr->kx[1];
    for (j = 0; j < sbr->num_patches; j++) {
        for (x = 0; x < sbr->patch_num_subbands[j]; x++, k++) {
+            float alpha[4];
            const int p = sbr->patch_start_subband[j] + x;
            while (g <= sbr->n_q && k >= sbr->f_tablenoise[g])
                g++;
@@ -1353,10 +1381,26 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
                return -1;
            }

-            sbr->dsp.hf_gen(X_high[k] + ENVELOPE_ADJUSTMENT_OFFSET,
-                            X_low[p]  + ENVELOPE_ADJUSTMENT_OFFSET,
-                            alpha0[p], alpha1[p], bw_array[g],
-                            2 * t_env[0], 2 * t_env[bs_num_env]);
+            alpha[0] = alpha1[p][0] * bw_array[g] * bw_array[g];
+            alpha[1] = alpha1[p][1] * bw_array[g] * bw_array[g];
+            alpha[2] = alpha0[p][0] * bw_array[g];
+            alpha[3] = alpha0[p][1] * bw_array[g];
+
+            for (i = 2 * t_env[0]; i < 2 * t_env[bs_num_env]; i++) {
+                const int idx = i + ENVELOPE_ADJUSTMENT_OFFSET;
+                X_high[k][idx][0] =
+                    X_low[p][idx - 2][0] * alpha[0] -
+                    X_low[p][idx - 2][1] * alpha[1] +
+                    X_low[p][idx - 1][0] * alpha[2] -
+                    X_low[p][idx - 1][1] * alpha[3] +
+                    X_low[p][idx][0];
+                X_high[k][idx][1] =
+                    X_low[p][idx - 2][1] * alpha[0] +
+                    X_low[p][idx - 2][0] * alpha[1] +
+                    X_low[p][idx - 1][1] * alpha[2] +
+                    X_low[p][idx - 1][0] * alpha[3] +
+                    X_low[p][idx][1];
+            }
        }
    }
    if (k < sbr->m[1] + sbr->kx[1])
@@ -1367,8 +1411,8 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,

 /// Generate the subband filtered lowband
 static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
-                     const float Y0[38][64][2], const float Y1[38][64][2],
-                     const float X_low[32][40][2], int ch)
+                     const float X_low[32][40][2], const float Y[2][38][64][2],
+                     int ch)
 {
    int k, i;
    const int i_f = 32;
@@ -1382,8 +1426,8 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
    }
    for (; k < sbr->kx[0] + sbr->m[0]; k++) {
        for (i = 0; i < i_Temp; i++) {
-            X[0][i][k] = Y0[i + i_f][k][0];
-            X[1][i][k] = Y0[i + i_f][k][1];
+            X[0][i][k] = Y[0][i + i_f][k][0];
+            X[1][i][k] = Y[0][i + i_f][k][1];
        }
    }

@@ -1395,8 +1439,8 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
    }
    for (; k < sbr->kx[1] + sbr->m[1]; k++) {
        for (i = i_Temp; i < i_f; i++) {
-            X[0][i][k] = Y1[i][k][0];
-            X[1][i][k] = Y1[i][k][1];
+            X[0][i][k] = Y[1][i][k][0];
+            X[1][i][k] = Y[1][i][k][1];
        }
    }
    return 0;
@@ -1405,7 +1449,7 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
 /** High Frequency Adjustment (14496-3 sp04 p217) and Mapping
 * (14496-3 sp04 p217)
 */
-static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
+static void sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
                        SBRData *ch_data, int e_a[2])
 {
    int e, i, m;
@@ -1416,12 +1460,7 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
        uint16_t *table = ch_data->bs_freq_res[e + 1] ? sbr->f_tablehigh : sbr->f_tablelow;
        int k;

-        if (sbr->kx[1] != table[0]) {
-            av_log(ac->avctx, AV_LOG_ERROR, "kx != f_table{high,low}[0]. "
-                   "Derived frequency tables were not regenerated.\n");
-            sbr_turnoff(sbr);
-            return AVERROR_BUG;
-        }
+        av_assert0(sbr->kx[1] <= table[0]);
        for (i = 0; i < ilim; i++)
            for (m = table[i]; m < table[i + 1]; m++)
                sbr->e_origmapped[e][m - sbr->kx[1]] = ch_data->env_facs[e+1][i];
@@ -1456,15 +1495,13 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
    }

    memcpy(ch_data->s_indexmapped[0], ch_data->s_indexmapped[ch_data->bs_num_env], sizeof(ch_data->s_indexmapped[0]));
-    return 0;
 }

 /// Estimation of current envelope (14496-3 sp04 p218)
 static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
                             SpectralBandReplication *sbr, SBRData *ch_data)
 {
-    int e, m;
-    int kx1 = sbr->kx[1];
+    int e, i, m;

    if (sbr->bs_interpol_freq) {
        for (e = 0; e < ch_data->bs_num_env; e++) {
@@ -1473,7 +1510,12 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
            int iub = ch_data->t_env[e + 1] * 2 + ENVELOPE_ADJUSTMENT_OFFSET;

            for (m = 0; m < sbr->m[1]; m++) {
-                float sum = sbr->dsp.sum_square(X_high[m+kx1] + ilb, iub - ilb);
+                float sum = 0.0f;
+
+                for (i = ilb; i < iub; i++) {
+                    sum += X_high[m + sbr->kx[1]][i][0] * X_high[m + sbr->kx[1]][i][0] +
+                           X_high[m + sbr->kx[1]][i][1] * X_high[m + sbr->kx[1]][i][1];
+                }
                e_curr[e][m] = sum * recip_env_size;
            }
        }
@@ -1491,11 +1533,14 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
                const int den = env_size * (table[p + 1] - table[p]);

                for (k = table[p]; k < table[p + 1]; k++) {
-                    sum += sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb);
+                    for (i = ilb; i < iub; i++) {
+                        sum += X_high[k][i][0] * X_high[k][i][0] +
+                               X_high[k][i][1] * X_high[k][i][1];
+                    }
                }
                sum /= den;
                for (k = table[p]; k < table[p + 1]; k++) {
-                    e_curr[e][k - kx1] = sum;
+                    e_curr[e][k - sbr->kx[1]] = sum;
                }
            }
        }
@@ -1562,8 +1607,7 @@ static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
 }

 /// Assembling HF Signals (14496-3 sp04 p220)
-static void sbr_hf_assemble(float Y1[38][64][2],
-                            const float X_high[64][40][2],
+static void sbr_hf_assemble(float Y[2][38][64][2], const float X_high[64][40][2],
                            SpectralBandReplication *sbr, SBRData *ch_data,
                            const int e_a[2])
 {
@@ -1585,6 +1629,7 @@ static void sbr_hf_assemble(float Y1[38][64][2],
    float (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
    int indexnoise = ch_data->f_indexnoise;
    int indexsine  = ch_data->f_indexsine;
+    memcpy(Y[0], Y[1], sizeof(Y[0]));

    if (sbr->reset) {
        for (i = 0; i < h_SL; i++) {
@@ -1606,44 +1651,63 @@ static void sbr_hf_assemble(float Y1[38][64][2],
    for (e = 0; e < ch_data->bs_num_env; e++) {
        for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
            int phi_sign = (1 - 2*(kx & 1));
-            LOCAL_ALIGNED_16(float, g_filt_tab, [48]);
-            LOCAL_ALIGNED_16(float, q_filt_tab, [48]);
-            float *g_filt, *q_filt;

            if (h_SL && e != e_a[0] && e != e_a[1]) {
-                g_filt = g_filt_tab;
-                q_filt = q_filt_tab;
                for (m = 0; m < m_max; m++) {
                    const int idx1 = i + h_SL;
-                    g_filt[m] = 0.0f;
-                    q_filt[m] = 0.0f;
-                    for (j = 0; j <= h_SL; j++) {
-                        g_filt[m] += g_temp[idx1 - j][m] * h_smooth[j];
-                        q_filt[m] += q_temp[idx1 - j][m] * h_smooth[j];
-                    }
+                    float g_filt = 0.0f;
+                    for (j = 0; j <= h_SL; j++)
+                        g_filt += g_temp[idx1 - j][m] * h_smooth[j];
+                    Y[1][i][m + kx][0] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][0] * g_filt;
+                    Y[1][i][m + kx][1] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][1] * g_filt;
                }
            } else {
-                g_filt = g_temp[i + h_SL];
-                q_filt = q_temp[i];
+                for (m = 0; m < m_max; m++) {
+                    const float g_filt = g_temp[i + h_SL][m];
+                    Y[1][i][m + kx][0] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][0] * g_filt;
+                    Y[1][i][m + kx][1] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][1] * g_filt;
+                }
            }

-            sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
-                               i + ENVELOPE_ADJUSTMENT_OFFSET);
-
            if (e != e_a[0] && e != e_a[1]) {
-                sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
-                                                   q_filt, indexnoise,
-                                                   kx, m_max);
-            } else {
                for (m = 0; m < m_max; m++) {
-                    Y1[i][m + kx][0] +=
+                    indexnoise = (indexnoise + 1) & 0x1ff;
+                    if (sbr->s_m[e][m]) {
+                        Y[1][i][m + kx][0] +=
+                            sbr->s_m[e][m] * phi[0][indexsine];
+                        Y[1][i][m + kx][1] +=
+                            sbr->s_m[e][m] * (phi[1][indexsine] * phi_sign);
+                    } else {
+                        float q_filt;
+                        if (h_SL) {
+                            const int idx1 = i + h_SL;
+                            q_filt = 0.0f;
+                            for (j = 0; j <= h_SL; j++)
+                                q_filt += q_temp[idx1 - j][m] * h_smooth[j];
+                        } else {
+                            q_filt = q_temp[i][m];
+                        }
+                        Y[1][i][m + kx][0] +=
+                            q_filt * sbr_noise_table[indexnoise][0];
+                        Y[1][i][m + kx][1] +=
+                            q_filt * sbr_noise_table[indexnoise][1];
+                    }
+                    phi_sign = -phi_sign;
+                }
+            } else {
+                indexnoise = (indexnoise + m_max) & 0x1ff;
+                for (m = 0; m < m_max; m++) {
+                    Y[1][i][m + kx][0] +=
                        sbr->s_m[e][m] * phi[0][indexsine];
-                    Y1[i][m + kx][1] +=
+                    Y[1][i][m + kx][1] +=
                        sbr->s_m[e][m] * (phi[1][indexsine] * phi_sign);
                    phi_sign = -phi_sign;
                }
            }
-            indexnoise = (indexnoise + m_max) & 0x1ff;
            indexsine = (indexsine + 1) & 3;
        }
    }
@@ -1654,54 +1718,39 @@ static void sbr_hf_assemble(float Y1[38][64][2],
 void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
                  float* L, float* R)
 {
-    int downsampled = ac->oc[1].m4ac.ext_sample_rate < sbr->sample_rate;
+    int downsampled = ac->m4ac.ext_sample_rate < sbr->sample_rate;
    int ch;
    int nch = (id_aac == TYPE_CPE) ? 2 : 1;
-    int err;
-
-    if (!sbr->kx_and_m_pushed) {
-        sbr->kx[0] = sbr->kx[1];
-        sbr->m[0] = sbr->m[1];
-    } else {
-        sbr->kx_and_m_pushed = 0;
-    }

    if (sbr->start) {
        sbr_dequant(sbr, id_aac);
    }
    for (ch = 0; ch < nch; ch++) {
        /* decode channel */
-        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
+        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
                         (float*)sbr->qmf_filter_scratch,
                         sbr->data[ch].W);
        sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W);
-        sbr->data[ch].Ypos ^= 1;
        if (sbr->start) {
-            sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1, sbr->X_low, sbr->k[0]);
+            sbr_hf_inverse_filter(sbr->alpha0, sbr->alpha1, sbr->X_low, sbr->k[0]);
            sbr_chirp(sbr, &sbr->data[ch]);
            sbr_hf_gen(ac, sbr, sbr->X_high, sbr->X_low, sbr->alpha0, sbr->alpha1,
                       sbr->data[ch].bw_array, sbr->data[ch].t_env,
                       sbr->data[ch].bs_num_env);

            // hf_adj
-            err = sbr_mapping(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
-            if (!err) {
-                sbr_env_estimate(sbr->e_curr, sbr->X_high, sbr, &sbr->data[ch]);
-                sbr_gain_calc(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
-                sbr_hf_assemble(sbr->data[ch].Y[sbr->data[ch].Ypos],
-                                sbr->X_high, sbr, &sbr->data[ch],
-                                sbr->data[ch].e_a);
-            }
+            sbr_mapping(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
+            sbr_env_estimate(sbr->e_curr, sbr->X_high, sbr, &sbr->data[ch]);
+            sbr_gain_calc(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
+            sbr_hf_assemble(sbr->data[ch].Y, sbr->X_high, sbr, &sbr->data[ch],
+                            sbr->data[ch].e_a);
        }

        /* synthesis */
-        sbr_x_gen(sbr, sbr->X[ch],
-                  sbr->data[ch].Y[1-sbr->data[ch].Ypos],
-                  sbr->data[ch].Y[  sbr->data[ch].Ypos],
-                  sbr->X_low, ch);
+        sbr_x_gen(sbr, sbr->X[ch], sbr->X_low, sbr->data[ch].Y, ch);
    }

-    if (ac->oc[1].m4ac.ps == 1) {
+    if (ac->m4ac.ps == 1) {
        if (sbr->ps.start) {
            ff_ps_apply(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
        } else {
@@ -1710,12 +1759,12 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
        nch = 2;
    }

-    sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, &sbr->dsp, L, sbr->X[0], sbr->qmf_filter_scratch,
+    sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, L, sbr->X[0], sbr->qmf_filter_scratch,
                      sbr->data[0].synthesis_filterbank_samples,
                      &sbr->data[0].synthesis_filterbank_samples_offset,
                      downsampled);
    if (nch == 2)
-        sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, &sbr->dsp, R, sbr->X[1], sbr->qmf_filter_scratch,
+        sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, R, sbr->X[1], sbr->qmf_filter_scratch,
                          sbr->data[1].synthesis_filterbank_samples,
                          &sbr->data[1].synthesis_filterbank_samples_offset,
                          downsampled);
--- a/libavcodec/aacsbrdata.h
+++ b/libavcodec/aacsbrdata.h
@@ -267,8 +267,8 @@ static const int8_t sbr_offset[6][16] = {
 };

 ///< window coefficients for analysis/synthesis QMF banks
-static DECLARE_ALIGNED(32, float, sbr_qmf_window_ds)[320];
-static DECLARE_ALIGNED(32, float, sbr_qmf_window_us)[640] = {
+static DECLARE_ALIGNED(16, float, sbr_qmf_window_ds)[320];
+static DECLARE_ALIGNED(16, float, sbr_qmf_window_us)[640] = {
     0.0000000000, -0.0005525286, -0.0005617692, -0.0004947518,
    -0.0004875227, -0.0004893791, -0.0005040714, -0.0005226564,
    -0.0005466565, -0.0005677802, -0.0005870930, -0.0006132747,
@@ -352,8 +352,7 @@ static DECLARE_ALIGNED(32, float, sbr_qmf_window_us)[640] = {
     0.8537385600,
 };

-/* First two entries repeated at end to simplify SIMD implementations. */
-const DECLARE_ALIGNED(16, float, ff_sbr_noise_table)[][2] = {
+static const float sbr_noise_table[512][2] = {
 {-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},
 { 0.14130051758487, -0.95090983575689}, {-0.47005496701697, -0.37340549728647},
 { 0.80705063769351,  0.29653668284408}, {-0.38981478896926,  0.89572605717087},
@@ -610,7 +609,6 @@ const DECLARE_ALIGNED(16, float, ff_sbr_noise_table)[][2] = {
 {-0.93412041758744,  0.41374052024363}, { 0.96063943315511,  0.93116709541280},
 { 0.97534253457837,  0.86150930812689}, { 0.99642466504163,  0.70190043427512},
 {-0.94705089665984, -0.29580042814306}, { 0.91599807087376, -0.98147830385781},
-{-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},
 };

 #endif /* AVCODEC_AACSBRDATA_H */
--- a/libavcodec/aactab.c
+++ b/libavcodec/aactab.c
@@ -33,8 +33,8 @@

 #include <stdint.h>

-DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, float,  ff_aac_kbd_short_128)[128];
+DECLARE_ALIGNED(16, float,  ff_aac_kbd_long_1024)[1024];
+DECLARE_ALIGNED(16, float,  ff_aac_kbd_short_128)[128];

 const uint8_t ff_aac_num_swb_1024[] = {
    41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40, 40
--- a/libavcodec/aactab.h
+++ b/libavcodec/aactab.h
@@ -44,8 +44,8 @@
 /* @name window coefficients
 * @{
 */
-DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_short_128)[128];
+DECLARE_ALIGNED(16, extern float,  ff_aac_kbd_long_1024)[1024];
+DECLARE_ALIGNED(16, extern float,  ff_aac_kbd_short_128)[128];
 // @}

 /* @name number of scalefactor window bands for long and short transform windows respectively
--- a/libavcodec/aasc.c
+++ b/libavcodec/aasc.c
@@ -34,26 +34,23 @@

 typedef struct AascContext {
    AVCodecContext *avctx;
-    GetByteContext gb;
    AVFrame frame;
 } AascContext;

+#define FETCH_NEXT_STREAM_BYTE() \
+    if (stream_ptr >= buf_size) \
+    { \
+      av_log(s->avctx, AV_LOG_ERROR, " AASC: stream ptr just went out of bounds (fetch)\n"); \
+      break; \
+    } \
+    stream_byte = buf[stream_ptr++];
+
 static av_cold int aasc_decode_init(AVCodecContext *avctx)
 {
    AascContext *s = avctx->priv_data;

    s->avctx = avctx;
-    switch (avctx->bits_per_coded_sample) {
-    case 16:
-        avctx->pix_fmt = PIX_FMT_RGB555;
-        break;
-    case 24:
-        avctx->pix_fmt = PIX_FMT_BGR24;
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Unsupported bit depth: %d\n", avctx->bits_per_coded_sample);
-        return -1;
-    }
+    avctx->pix_fmt = PIX_FMT_BGR24;
    avcodec_get_frame_defaults(&s->frame);

    return 0;
@@ -66,7 +63,7 @@ static int aasc_decode_frame(AVCodecContext *avctx,
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    AascContext *s = avctx->priv_data;
-    int compr, i, stride, psize;
+    int compr, i, stride;

    s->frame.reference = 3;
    s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
@@ -78,39 +75,26 @@ static int aasc_decode_frame(AVCodecContext *avctx,
    compr = AV_RL32(buf);
    buf += 4;
    buf_size -= 4;
-    psize = avctx->bits_per_coded_sample / 8;
-    switch (avctx->codec_tag) {
-    case MKTAG('A', 'A', 'S', '4'):
-        bytestream2_init(&s->gb, buf - 4, buf_size + 4);
-        ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, &s->gb);
-        break;
-    case MKTAG('A', 'A', 'S', 'C'):
    switch(compr){
    case 0:
-        stride = (avctx->width * psize + psize) & ~psize;
+        stride = (avctx->width * 3 + 3) & ~3;
        for(i = avctx->height - 1; i >= 0; i--){
-            if(avctx->width * psize > buf_size){
+            if(avctx->width*3 > buf_size){
                av_log(avctx, AV_LOG_ERROR, "Next line is beyond buffer bounds\n");
                break;
            }
-            memcpy(s->frame.data[0] + i*s->frame.linesize[0], buf, avctx->width * psize);
+            memcpy(s->frame.data[0] + i*s->frame.linesize[0], buf, avctx->width*3);
            buf += stride;
            buf_size -= stride;
        }
        break;
    case 1:
-        bytestream2_init(&s->gb, buf, buf_size);
-        ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, &s->gb);
+        ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, buf - 4, buf_size + 4);
        break;
    default:
        av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
        return -1;
    }
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Unknown FourCC: %X\n", avctx->codec_tag);
-        return -1;
-    }

    *data_size = sizeof(AVFrame);
    *(AVFrame*)data = s->frame;
@@ -139,5 +123,5 @@ AVCodec ff_aasc_decoder = {
    .close          = aasc_decode_end,
    .decode         = aasc_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
+    .long_name = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
 };
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .11.2
 .9.4