Update for 3.1.7

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

.gitattributes

@@ -1,2 +1 @@
 *.pnm -diff -text
-tests/ref/fate/sub-scc eol=crlf

.gitignore

@@ -18,9 +18,6 @@
 *.so.*
 *.swp
 *.ver
-*.version
-*.ptx
-*.ptx.c
 *_g
 \#*
 .\#*
@@ -29,8 +26,8 @@
 /ffmpeg
 /ffplay
 /ffprobe
-/config.asm
-/config.h
+/ffserver
+/config.*
 /coverage.info
 /avversion.h
 /lcov/

.travis.yml

@@ -6,22 +6,18 @@ os:
 addons:
   apt:
     packages:
-      - nasm
+      - yasm
       - diffutils
 compiler:
   - clang
   - gcc
-matrix:
-    exclude:
-        - os: osx
-          compiler: gcc
 cache:
   directories:
     - ffmpeg-samples
 before_install:
   - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
 install:
-  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
+  - if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install yasm; fi
 script:
   - mkdir -p ffmpeg-samples
   - ./configure --samples=ffmpeg-samples --cc=$CC

CONTRIBUTING.md

@@ -1,4 +0,0 @@
-# Note to Github users
-Patches should be submitted to the [ffmpeg-devel mailing list](https://ffmpeg.org/mailman/listinfo/ffmpeg-devel) using `git format-patch` or `git send-email`. Github pull requests should be avoided because they are not part of our review process and **will be ignored**.
-
-See [https://ffmpeg.org/developer.html#Contributing](https://ffmpeg.org/developer.html#Contributing) for more information.

Changelog

@@ -1,345 +1,241 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 4.0.2:
-- avcodec/dvdsub_parser: Allocate input padding
-- avcodec/dvdsub_parser: Init output buf/size
-- avcodec/dirac_dwt_template: Fix signedness regression in interleave()
-- avformat/mov: Simplify last element computation in mov_estimate_video_delay()
-- avformat/mov: Break out of inner loop early in mov_estimate_video_delay()
-- avformat/mov: Eliminate variable buf_size from mov_estimate_video_delay()
-- avformat/mov: remove modulo operations from mov_estimate_video_delay()
-- avformat/movenc: Write version 2 of audio atom if channels is not known
-- swresample/arm: rename labels to fix xcode build error
-- avformat/movenc: Check input sample count
-- avcodec/mjpegdec: Check for odd progressive RGB
-- avcodec/vp8_parser: Do not leave data/size uninitialized
-- avformat/mms: Add missing chunksize check
-- avformat/pva: Check for EOF before retrying in read_part_of_packet()
-- avformat/rmdec: Do not pass mime type in rm_read_multi() to ff_rm_read_mdpr_codecdata()
-- avformat/asfdec_o: Check size_bmp more fully
-- avformat/mxfdec: Fix av_log context
-- avcodec/mpeg4videodec: Check for bitstream end in read_quant_matrix_ext()
-- avcodec/indeo4: Check for end of bitstream in decode_mb_info()
-- avcodec/ac3dec: Check channel_map index
-- avcodec/mpeg4videodec: Remove use of FF_PROFILE_MPEG4_SIMPLE_STUDIO as indicator of studio profile
-- avcodec/shorten: Fix undefined addition in shorten_decode_frame()
-- avcodec/shorten: Fix undefined integer overflow
-- avcodec/jpeg2000dec: Fixes invalid shifts in jpeg2000_decode_packets_po_iteration()
-- avcodec/jpeg2000dec: Check that there are enough bytes for all tiles
-- avformat/movenc: Use mov->fc consistently for av_log()
-- avcodec/mpeg4videodec: Check read profile before setting it
-- avformat/movenc: Do not pass AVCodecParameters in avpriv_request_sample
-- avcodec/ac3_parser: Check init_get_bits8() for failure
-- avformat/movenc: Check that frame_types other than EAC3_FRAME_TYPE_INDEPENDENT have a supported substream id
-- avcodec/dpx: Check elements in 12bps planar path
-- avcodec/escape124: Fix spelling errors in comment
-- avcodec/ra144: Fix integer overflow in ff_eval_refl()
-- avcodec/cscd: Check output buffer size for lzo.
-- avcodec/escape124: Check buf_size against num_superblocks
-- avcodec/h264_parser: Reduce needed history for parsing mb index
-- avcodec/magicyuv: Check bits left in flags&1 branch
-- avcodec/mjpegdec: Check for end of bitstream in ljpeg_decode_rgb_scan()
-- ffmpeg: fix -stream_loop with multiple inputs
-- ffmpeg: factorize input thread creation and destruction
-- avformat/mpegts: parse large PMTs with multiple tables
-- Revert "avcodec/mediacodecdec: wait on first frame after input buffers are full"
-- avcodec/videotoolboxenc: fix invalid session on iOS
-- avcodec/videotoolboxenc: split initialization
-- avcodec/videotoolboxenc: fix mutex/cond leak in error path
+version 3.1.7:
+- avcodec/h264_slice: Clear ref_counts on redundant slices
+- lavf/mov.c: Avoid heap allocation wrap in mov_read_uuid
+- lavf/mov.c: Avoid heap allocation wrap in mov_read_hdlr
+- avcodec/pictordec: Fix logic error
+- ffserver_config: Setup codecpar in add_codec()
+- avcodec/movtextdec: Fix decode_styl() cleanup
+- lavf/matroskadec: fix is_keyframe for early Blocks
+- configure: bump year
+- avcodec/pngdec: Check trns more completely
+- avcodec/interplayvideo: Move parameter change check up
+- avcodec/dca_lbr: Fix off by 1 error in freq check
+- avcodec/mjpegdec: Check for for the bitstream end in mjpeg_decode_scan_progressive_ac()
+- avformat/flacdec: Check avio_read result when reading flac block header.
+- avcodec/utils: correct align value for interplay
+- avcodec/vp56: Check for the bitstream end, pass error codes on
+- avcodec/mjpegdec: Check remaining bitstream in ljpeg_decode_yuv_scan()
+- avcodec/pngdec: Fix off by 1 size in decode_zbuf()
+- avcodec/omx: Do not pass negative value into av_malloc()
+- avformat/avidec: skip odml master index chunks in avi_sync
+- avcodec/mjpegdec: Check for rgb before flipping
+- avutil/random_seed: Reduce the time needed on systems with very low precision clock()
+- avutil/random_seed: Improve get_generic_seed() with higher precision clock()
+- avformat/mp3dec: fix msan warning when verifying mpa header
+- avformat/utils: Print verbose error message if stream count exceeds max_streams
+- avformat/options_table: Set the default maximum number of streams to 1000
+- pgssubdec: reset rle_data_len/rle_remaining_len on allocation error
+- avutil: Add av_image_check_size2()
+- avformat: Add max_streams option
+- avcodec/ffv1enc: Allocate smaller packet if the worst case size cannot be allocated
+- avcodec/mpeg4videodec: Fix undefined shifts in mpeg4_decode_sprite_trajectory()
+- avformat/oggdec: Skip streams in duration correction that did not had their duration set.
+- avcodec/ffv1enc: Fix size of first slice
+- avfilter/vf_hwupload_cuda: Add min/max limits for the 'device' option
+- configure: check for strtoull on msvc
 
-version 4.0.1:
-- avcodec/aacdec_fixed: Fix undefined integer overflow in apply_independent_coupling_fixed()
-- avcodec/dirac_dwt_template: Fix undefined behavior in interleave()
-- avutil/common: Fix undefined behavior in av_clip_uintp2_c()
-- fftools/ffmpeg: Fallback to duration if sample rate is unavailable
-- avformat/mov: Only set pkt->duration to non negative values
-- avcodec/mpeg4videodec: Clear bits_per_raw_sample if it has originated from a previous instance
-- avformat/movenc: fix recognization of cover image streams
-- avformat/movenc: properly handle cover image codecs
-- avcodec/h264_slice: Fix overflow in recovery_frame computation
-- avcodec/h264_ps: Move MAX_LOG2_MAX_FRAME_NUM to header so it can be used in h264_sei
-- avcodec/h264_mc_template: Only prefetch motion if the list is used.
-- avcodec/xwddec: Use ff_set_dimensions()
-- avcodec/wavpack: Fix overflow in adding tail
-- avcodec/shorten: Fix multiple integer overflows
-- avcodec/shorten: Fix undefined shift in fix_bitshift()
-- avcodec/shorten: Fix a negative left shift in shorten_decode_frame()
-- avcodec/shorten: Sanity check nmeans
-- avcodec/shorten: Check non COMM chunk len before skip in decode_aiff_header()
-- avcodec/mjpegdec: Fix integer overflow in ljpeg_decode_rgb_scan()
-- avcodec/truemotion2: Fix overflow in tm2_apply_deltas()
-- avcodec/opus_silk: Change silk_lsf2lpc() slightly toward silk/NLSF2A.c
-- avcodec/amrwbdec: Fix division by 0 in find_hb_gain()
-- avcodec/h263dec: Reinitialize idct context if it has not been setup for the active profile
-- avcodec/idctdsp: Clear idct/idct_add for studio profile
-- avformat/mov: replace a value error by clipping into valid range in mov_read_stsc()
-- avformat/bintext: Reduce detection for random .bin files as it more likely is not a multimedia related file
-- avformat/mov: Break out early if chunk_count is 0 in mov_build_index()
-- avcodec/fic: Avoid some magic numbers related to cursors
-- avcodec/mpeg4video: Detect reference studio streams as studio streams
-- avcodec/mpeg4videodec: Do not corrupt bits_per_raw_sample
-- avcodec/mpeg4videode: Eliminate out of loop VOP startcode reading for studio profile
-- avcodec/g2meet: ask for sample with overflowing RGB
-- avcodec/idctdsp: Transmit studio_profile to init instead of using AVCodecContext profile
-- avcodec/ac3dec: Check that the number of channels with dependant streams is valid
-- avcodec/ac3dec: Fix null pointer dereference in ac3_decode_frame()
-- avcodec/aacdec_fixed: use 64bit to avoid overflow in rounding in apply_dependent_coupling_fixed()
-- oavcodec/aacpsdsp_template: Use unsigned for hs0X to prevent undefined behavior
-- avcodec/g723_1dec: Clip bits2 in both directions
-- avcodec/mpeg4videoenc: Use 64 bit for times in mpeg4_encode_gop_header()
-- avcodec/mlpdec: Only change noise_type if the related fields are valid
-- indeo4: Decode all or nothing of a band header.
-- avcodec/ac3dec: Use frame_size if superframe_size is 0
-- avformat/mov: Only fail for STCO/STSC contradictions if both exist
-- avcodec/dirac_dwt: Fix integer overflow in COMPOSE_DD97iH0 / COMPOSE_DD137iL0
-- avcodec/fic: Check available input space for cursor
-- avcodec/mpeg4videodec: Check bps (VOL header) before VOP for studio profile
-- avcodec/g2meet: Check RGB upper limit
-- avcodec/jpeg2000dec: Fix undefined shift in the jpeg2000_decode_packets_po_iteration() CPRL case
-- avcodec/jpeg2000dec: Skip init for component in CPRL if nothing is to be done
-- avcodec/g2meet: Change order of operations to avoid undefined behavior
-- avcodec/flac_parser: Fix infinite loop
-- avcodec/mpeg4videodec: Split decode_studio_vol_header() out of decode_studiovisualobject()
-- avcodec/mpeg4videodec: Move decode_studiovisualobject() parsing in the branch for visual object parsing
-- avcodec/mpeg4video_parser: Avoid litteral 0x1B6, use named constant instead
-- avcodec/mpeg4video_parser: Fix incorrect spliting of MPEG-4 studio frames
-- avformat/m4vdec: Use the same constant names as libavcodec
-- avformat/m4vdec: Fix detection of raw MPEG-4 ES Studio
-- avcodec/wavpack: Fix integer overflow in DEC_MED() / INC_MED()
-- avcodec/wavpack: Fix integer overflow in wv_unpack_stereo()
-- avcodec/error_resilience: Fix integer overflow in filter181()
-- avcodec/h263dec: Check slice_ret in mspeg4 slice loop
-- avcodec/elsdec: Fix memleaks
-- avcodec/vc1_block: simplify ac_val computation
-- avcodec/ffv1enc: Check that the crc + version combination is supported
-- configure: The eac3_core bitstream filter needs the ac3 parser.
-- configure: fix arm inline asm checks
-- lavf/libssh: translate a read of 0 to EOF
-- ffprobe: fix SEGV when new streams are added
-- avformat/mpegts: fix incorrect indentation
-- avformat/mpegts: initialize section_buf to fix valgrind test failure
-- avformat/mpegts: reindent after last change
-- avformat/mpegts: parse sections with multiple tables
-- avformat/mpegts: clean up whitespace
-- avformat/mpegts: use MAX_SECTION_SIZE instead of hardcoded value
-- avformat/mpegts: skip non-PMT tids earlier
-- avcodec/mediacodecdec: add workaround for buggy amlogic mpeg2 decoder
-- avcodec/mediacodecdec: wait on first frame after input buffers are full
-- avcodec/mediacodecdec: restructure mediacodec_receive_frame
-- avcodec/mediacodec_wrapper: add helper to fetch SDK_INT
-- avcodec/mediacodecdec: refactor pts handling
-- avcodec/mediacodecdec: use AV_TIME_BASE_Q
-- avcodec/mediacodecdec: clarify delay_flush specific code
-- avcodec/videotoolbox: fix decoding of some HEVC videos
-- avcodec/hevc: remove videotoolbox hack
-- avcodec/videotoolbox: split h264/hevc callbacks
-- avcodec/videotoolbox: cleanups
-- avcodec/videotoolbox: fix kVTCouldNotFindVideoDecoderErr trying to decode HEVC on iOS
-- avcodec/videotoolbox: improve logging of decoder errors
-- avcodec/xwddec: fix palette alpha
-- avformat/webm_chunk: always use a static buffer for get_chunk_filename
-- configure: fix configure check for lilv-0
-- avcodec/nvdec_hevc: fix scaling lists
-- avcodec/hevcdec: make ff_hevc_frame_nb_refs take a const pointer
-- lavf/bluray: translate a read of 0 to EOF
-- lavf/dashenc: don't call flush_init_segment before avformat_write_header
-- avdevice/decklink_dec: unref packets on avpacket_queue_put error
-- avcodec/hnm4video: fix palette alpha
-- avcodec/anm: fix palette alpha
-- avformat/qtpalette: parse color table according to the QuickTime file format specs
-- ffplay: Fix realloc_texture when input texture is NULL.
-- hwcontext_vaapi: Fix compilation with libva versions < 1.4.0
-- lavf/qsv: clone the frame which may be managed by framework
-- lavf: make overlay_qsv work based on framesync
-- avformat/segafilm - revert keyframe detection
-- avformat/utils: refactor upstream_stream_timings
-- avformat/utils: ignore outlier durations on subtitle/data streams as well
+version 3.1.6:
+- configure: check for strtoull on msvc
+- http: move chunk handling from http_read_stream() to http_buf_read().
+- http: make length/offset-related variables unsigned.
+- ffserver: Check chunk size
+- Avoid using the term "file" and prefer "url" in some docs and comments
+- avformat/rtmppkt: Check for packet size mismatches
+- zmqsend: Initialize ret to 0
+- avcodec/rawdec: check for side data before checking its size
+- avcodec/flacdec: Fix undefined shift in decode_subframe()
+- avcodec/get_bits: Fix get_sbits_long(0)
+- avformat/ffmdec: Check media type for chunks
+- avcodec/flacdec: Fix signed integer overflow in decode_subframe_fixed()
+- avcodec/flacdsp_template: Fix undefined shift in flac_decorrelate_indep_c
+- avformat/oggparsespeex: Check frames_per_packet and packet_size
+- avformat/utils: Check start/end before computing duration in update_stream_timings()
+- avcodec/flac_parser: Update nb_headers_buffered
+- avformat/idroqdec: Check chunk_size for being too large
+- avformat/utils: Fix type mismatch
+- avformat/mpeg: Adjust vid probe threshold to correct mis-detection
+- avcodec/rv40: Test remaining space in loop of get_dimension()
+- avcodec/ituh263dec: Avoid spending a long time in slice sync
+- avcodec/movtextdec: Add error message for tsmb_size check
+- avcodec/movtextdec: Fix tsmb_size check==0 check
+- avcodec/movtextdec: Fix potential integer overflow
+- avcodec/sunrast: Fix input buffer pointer check
+- avcodec/tscc:  Check side data size before use
+- avcodec/rawdec: Check side data size before use
+- avcodec/msvideo1: Check side data size before use
+- avcodec/qpeg:  Check side data size before use
+- avcodec/qtrle:  Check side data size before use
+- avcodec/msrle:  Check side data size before use
+- avcodec/kmvc:  Check side data size before use
+- avcodec/idcinvideo: Check side data size before use
+- avcodec/cinepak: Check side data size before use
+- avcodec/8bps: Check side data size before use
+- avformat/flvdec: Fix regression losing streams
+- avcodec/dvdsubdec: Fix off by 1 error
+- avformat/isom: Fix old API regression with exporting max bitrate
+- avcodec/dvdsubdec: Fix buf_size check
+- vp9: change order of operations in adapt_prob().
+- avcodec/interplayvideo: Check side data size before use
+- mss2: only use error correction for matching block counts
+- softfloat: decrease MIN_EXP to cover full float range
+- libopusdec: default to stereo for invalid number of channels
+- flvdec: require need_context_update when changing codec id
+- pgssubdec: only set w/h/linesize when allocating data
+- sbgdec: prevent NULL pointer access
+- rmdec: validate block alignment
+- smacker: limit recursion depth of smacker_decode_bigtree
+- mxfdec: fix NULL pointer dereference in mxf_read_packet_old
+- ffmdec: validate codec parameters
+- exr: reindent after previous commit
+- exr: fix out-of-bounds read
+- libschroedingerdec: fix leaking of framewithpts
+- libschroedingerdec: don't produce empty frames
+- softfloat: handle -INT_MAX correctly
+- filmstripdec: correctly check image dimensions
+- pnmdec: make sure v is capped by maxval
+- smvjpegdec: make sure cur_frame is not negative
+- icodec: correctly check avio_read return value
+- dvbsubdec: fix division by zero in compute_default_clut
+- proresdec_lgpl: explicitly check coff[3] against slice_data_size
+- escape124: reject codebook size 0
+- icodec: add ico_read_close to fix leaking ico->images
+- icodec: fix leaking pkt on error
+- mpegts: prevent division by zero
+- matroskadec: fix NULL pointer dereference in webm_dash_manifest_read_header
+- mpegaudio_parser: don't return AVERROR_PATCHWELCOME
+- mxfdec: fix NULL pointer dereference
+- lzf: update pointer p after realloc
+- diracdec: check return code of get_buffer_with_edge
+- ppc: pixblockdsp: do unaligned block accesses correctly again
+- interplayacm: increase bitstream buffer size by AV_INPUT_BUFFER_PADDING_SIZE
+- interplayacm: validate number of channels
+- interplayacm: check for too large b
+- mpeg12dec: unref discarded picture from extradata
+- cavsdec: unref frame before referencing again
+- dcstr: fix division by zero
+- aiff: check block_align in aiff_read_packet
+- rsd: limit number of channels
+- avformat: prevent triggering request_probe assert in ff_read_packet
+- westwood_aud: prevent division by zero
+- astdec: fix division by zero
+- aiffdec: fix division by zero
+- avcodec/avpacket: fix leak on realloc in av_packet_add_side_data()
 
+version 3.1.5:
+- avformat/mxfdec: Check size to avoid integer overflow in mxf_read_utf16_string()
+- avcodec/mpegvideo_enc: Clear mmx state in ff_mpv_reallocate_putbitbuffer()
+- avcodec/utils: Clear MMX state before returning from avcodec_default_execute*()
+- doc/examples/demuxing_decoding: Drop AVFrame->pts use
+- libopenjpegenc: fix out-of-bounds reads when filling the edges
+- libopenjpegenc: stop reusing image data buffer for openjpeg 2
+- configure: fix detection of libopenjpeg
+- doc: fix various typos and grammar errors
+- avformat/utils: Update codec_id before using it in the parser init
+- cmdutils: fix typos
+- lavfi: fix typos
+- lavc: fix typos
+- tools: fix grammar error
+- ffmpeg: remove unused and errorneous AVFrame timestamp check
+- Support for MIPS cpu P6600
+- avutil/mips/generic_macros_msa: rename macro variable which causes segfault for mips r6
 
-version 4.0:
-- Bitstream filters for editing metadata in H.264, HEVC and MPEG-2 streams
-- Dropped support for OpenJPEG versions 2.0 and below. Using OpenJPEG now
-  requires 2.1 (or later) and pkg-config.
-- VDA dropped (use VideoToolbox instead)
-- MagicYUV encoder
-- Raw AMR-NB and AMR-WB demuxers
-- TiVo ty/ty+ demuxer
-- Intel QSV-accelerated MJPEG encoding
-- PCE support for extended channel layouts in the AAC encoder
-- native aptX and aptX HD encoder and decoder
-- Raw aptX and aptX HD muxer and demuxer
-- NVIDIA NVDEC-accelerated H.264, HEVC, MJPEG, MPEG-1/2/4, VC1, VP8/9 hwaccel decoding
-- Intel QSV-accelerated overlay filter
-- mcompand audio filter
-- acontrast audio filter
-- OpenCL overlay filter
-- video mix filter
-- video normalize filter
-- audio lv2 wrapper filter
-- VAAPI MJPEG and VP8 decoding
-- AMD AMF H.264 and HEVC encoders
-- video fillborders filter
-- video setrange filter
-- nsp demuxer
-- support LibreSSL (via libtls)
-- AVX-512/ZMM support added
-- Dropped support for building for Windows XP. The minimum supported Windows
-  version is Windows Vista.
-- deconvolve video filter
-- entropy video filter
-- hilbert audio filter source
-- aiir audio filter
-- aiff: add support for CD-ROM XA ADPCM
-- Removed the ffserver program
-- Removed the ffmenc and ffmdec muxer and demuxer
-- VideoToolbox HEVC encoder and hwaccel
-- VAAPI-accelerated ProcAmp (color balance), denoise and sharpness filters
-- Add android_camera indev
-- codec2 en/decoding via libcodec2
-- muxer/demuxer for raw codec2 files and .c2 files
-- Moved nvidia codec headers into an external repository.
-  They can be found at http://git.videolan.org/?p=ffmpeg/nv-codec-headers.git
-- native SBC encoder and decoder
-- drmeter audio filter
-- hapqa_extract bitstream filter
-- filter_units bitstream filter
-- AV1 Support through libaom
-- E-AC-3 dependent frames support
-- bitstream filter for extracting E-AC-3 core
-- Haivision SRT protocol via libsrt
-- segafilm muxer
-- vfrdet filter
+version 3.1.4:
+- avformat/avidec: Check nb_streams in read_gab2_sub()
+- avformat/avidec: Remove ancient assert
+- avfilter/vf_colorspace: fix range for output colorspace option
+- lavc/mediacodecdec_h264: fix SODB escaping
+- avcodec/nvenc: fix const options for hevc gpu setting
+- avformat/avidec: Fix memleak with dv in avi
+- lavc/movtextdec.c: Avoid infinite loop on invalid data.
+- avcodec/ansi: Check dimensions
+- avcodec/cavsdsp: use av_clip_uint8() for idct
+- avformat/movenc: Check packet in mov_write_single_packet() too
+- avformat/movenc: Factor check_pkt() out
+- avformat/utils: fix timebase error in avformat_seek_file()
+- avcodec/g726: Add missing ADDB output mask
+- avcodec/avpacket: clear side_data_elems
+- avformat/movenc: Check first DTS similar to dts difference
+- avcodec/ccaption_dec: Use simple array instead of AVBuffer
+- avcodec/svq3: Reintroduce slice_type
+- avformat/mov: Fix potential integer overflow in mov_read_keys
+- swscale/swscale_unscaled: Try to fix Rgb16ToPlanarRgb16Wrapper() with slices
+- swscale/swscale_unscaled: Fix packed_16bpc_bswap() with slices
+- avformat/avidec: Fix infinite loop in avi_read_nikon()
+- lavf/utils: Avoid an overflow for huge negative durations.
+- avformat/hls: Fix handling of EXT-X-BYTERANGE streams over 2GB
+- lavc/avpacket: Fix undefined behaviour, do not pass a null pointer to memcpy().
+- lavc/mjpegdec: Do not skip reading quantization tables.
+- cmdutils: fix implicit declaration of SetDllDirectory function
 
+version 3.1.3:
+- examples/demuxing_decoding: convert to codecpar
+- avcodec/exr: Check tile positions
+- avcodec/aacenc: Tighter input checks
+- avformat/wtvdec: Check pointer before use
+- libavcodec/wmalosslessdec: Check the remaining bits
+- avcodec/adpcm: Fix adpcm_ima_wav padding
+- avcodec/svq3: fix slice size check
+- avcodec/diracdec: Check numx/y
+- avcodec/h2645_parse: fix nal size
+- avcodec/h2645_parse: Use get_nalsize() in ff_h2645_packet_split()
+- h2645_parse: only read avc length code at the correct position
+- h2645_parse: don't overread AnnexB NALs within an avc stream
+- avcodec/h264_parser: Factor get_avc_nalsize() out
+- avcodec/cfhd: Increase minimum band dimension to 3
+- avcodec/indeo2: check ctab
+- avformat/swfdec: Fix inflate() error code check
+- avcodec/rawdec: Fix bits_per_coded_sample checks
+- vcodec/h2645_parse: Clear buffer padding
+- avcodec/h2645: Fix NAL unit padding
+- avfilter/drawutils: Fix single plane with alpha
+- cmdutils: check for SetDllDirectory() availability
 
-version 3.4:
-- deflicker video filter
-- doubleweave video filter
-- lumakey video filter
-- pixscope video filter
-- oscilloscope video filter
-- config.log and other configuration files moved into ffbuild/ directory
-- update cuvid/nvenc headers to Video Codec SDK 8.0.14
-- afir audio filter
-- scale_cuda CUDA based video scale filter
-- librsvg support for svg rasterization
-- crossfeed audio filter
-- spec compliant VP9 muxing support in MP4
-- remove the libnut muxer/demuxer wrappers
-- remove the libschroedinger encoder/decoder wrappers
-- surround audio filter
-- sofalizer filter switched to libmysofa
-- Gremlin Digital Video demuxer and decoder
-- headphone audio filter
-- superequalizer audio filter
-- roberts video filter
-- The x86 assembler default switched from yasm to nasm, pass
-  --x86asmexe=yasm to configure to restore the old behavior.
-- additional frame format support for Interplay MVE movies
-- support for decoding through D3D11VA in ffmpeg
-- limiter video filter
-- libvmaf video filter
-- Dolby E decoder and SMPTE 337M demuxer
-- unpremultiply video filter
-- tlut2 video filter
-- floodfill video filter
-- pseudocolor video filter
-- raw G.726 muxer and demuxer, left- and right-justified
-- NewTek NDI input/output device
-- Some video filters with several inputs now use a common set of options:
-  blend, libvmaf, lut3d, overlay, psnr, ssim.
-  They must always be used by name.
-- FITS demuxer and decoder
-- FITS muxer and encoder
-- add --disable-autodetect build switch
-- drop deprecated qtkit input device (use avfoundation instead)
-- despill video filter
-- haas audio filter
-- SUP/PGS subtitle muxer
-- convolve video filter
-- VP9 tile threading support
-- KMS screen grabber
-- CUDA thumbnail filter
-- V4L2 mem2mem HW assisted codecs
-- Rockchip MPP hardware decoding
-- vmafmotion video filter
-- use MIME type "G726" for little-endian G.726, "AAL2-G726" for big-endian G.726
+version 3.1.2:
+- cmdutils: remove the current working directory from the DLL search path on win32
+- avcodec/rawdec: Fix palette handling with changing palettes
+- avcodec/raw: Fix decoding of ilacetest.mov
+- avformat/mov: Enable mp3 parsing if a packet needs it
+- avformat/hls: Use an array instead of stream offset for stream mapping
+- avformat/hls: Sync starting segment across variants on live streams
+- avformat/hls: Fix regression with ranged media segments
+- avcodec/ffv1enc: Fix assertion failure with non zero bits per sample
+- avfilter/af_hdcd: small fix in af_hdcd.c where gain was not being adjusted for "attenuate slowly"
+- avformat/oggdec: Fix integer overflow with invalid pts
+- ffplay: Fix invalid array index
+- avcodec/alacenc: allocate bigger packets (cherry picked from commit 82b84c71b009884c8d041361027718b19922c76d)
+- libavcodec/dnxhd: Enable 12-bit DNxHR support.
+- lavc/vaapi_encode_h26x: Fix a crash if "." is not the decimal separator.
+- jni: Return ENOSYS on unsupported platforms
+- lavu/hwcontext_vaapi: Fix compilation if VA_FOURCC_ABGR is not defined.
+- avcodec/vp9_parser: Check the input frame sizes for being consistent
+- avformat/flvdec: parse keyframe before a\v stream was created add_keyframes_index() when stream created or keyframe parsed
+- avformat/flvdec: splitting add_keyframes_index() out from parse_keyframes_index()
+- libavformat/rtpdec_asf: zero initialize the AVIOContext struct
+- libavutil/opt: Small bugfix in example.
+- libx264: Increase x264 opts character limit to 4096
+- avcodec/h264_parser: Set sps/pps_ref
+- librtmp: Avoid an infiniloop setting connection arguments
+- avformat/oggparsevp8: fix pts calculation on pages ending with an invisible frame
+- lavc/Makefile: Fix standalone compilation of the svq3 decoder.
+- lavf/vplayerdec: Improve auto-detection.
+- lavc/mediacodecdec_h264: properly convert extradata to annex-b
+- Revert "configure: Enable GCC vectorization on ≥4.9 on x86"
 
-
-version 3.3:
-- CrystalHD decoder moved to new decode API
-- add internal ebur128 library, remove external libebur128 dependency
-- Pro-MPEG CoP #3-R2 FEC protocol
-- premultiply video filter
-- Support for spherical videos
-- configure now fails if autodetect-libraries are requested but not found
-- PSD Decoder
-- 16.8 floating point pcm decoder
-- 24.0 floating point pcm decoder
-- Apple Pixlet decoder
-- QDMC audio decoder
-- NewTek SpeedHQ decoder
-- MIDI Sample Dump Standard demuxer
-- readeia608 filter
-- Sample Dump eXchange demuxer
-- abitscope multimedia filter
-- Scenarist Closed Captions demuxer and muxer
-- threshold filter
-- midequalizer filter
-- Optimal Huffman tables for (M)JPEG encoding
-- VAAPI-accelerated MPEG-2 and VP8 encoding
-- FM Screen Capture Codec decoder
-- native Opus encoder
-- ScreenPressor decoder
-- incomplete ClearVideo decoder
-- Intel QSV video scaling and deinterlacing filters
-- Support MOV with multiple sample description tables
-- XPM decoder
-- Removed the legacy X11 screen grabber, use XCB instead
-- MPEG-7 Video Signature filter
-- Removed asyncts filter (use af_aresample instead)
-- Intel QSV-accelerated VP8 video decoding
-- VAAPI-accelerated deinterlacing
-
-
-version 3.2:
-- libopenmpt demuxer
-- tee protocol
-- Changed metadata print option to accept general urls
-- Alias muxer for Ogg Video (.ogv)
-- VP8 in Ogg muxing
-- curves filter doesn't automatically insert points at x=0 and x=1 anymore
-- 16-bit support in curves filter and selectivecolor filter
-- OpenH264 decoder wrapper
-- MediaCodec H.264/HEVC/MPEG-4/VP8/VP9 hwaccel
-- True Audio (TTA) muxer
-- crystalizer audio filter
-- acrusher audio filter
-- bitplanenoise video filter
-- floating point support in als decoder
-- fifo muxer
-- maskedclamp filter
-- hysteresis filter
-- lut2 filter
-- yuvtestsrc filter
-- CUDA CUVID H.263/VP8/VP9/10 bit HEVC (Dithered) Decoding
-- vaguedenoiser filter
-- added threads option per filter instance
-- weave filter
-- gblur filter
-- avgblur filter
-- sobel and prewitt filter
-- MediaCodec HEVC/MPEG-4/VP8/VP9 decoding
-- Meridian Lossless Packing (MLP) / TrueHD encoder
-- Non-Local Means (nlmeans) denoising filter
-- sdl2 output device and ffplay support
-- sdl1 output device and sdl1 support removed
-- extended mov edit list support
-- libfaac encoder removed
-- Matroska muxer now writes CRC32 elements by default in all Level 1 elements
-- sidedata video and asidedata audio filter
-- Changed mapping of rtp MIME type G726 to codec g726le.
-- spec compliant VAAPI/DXVA2 VC-1 decoding of slices in frame-coded images
+version 3.1.1:
+- doc/APIchanges: document the lavu/lavf field moves
+- avformat/avformat: Move new field to the end of AVStream
+- avformat/utils: update deprecated AVStream->codec when the context is updated
+- avutil/frame: Move new field to the end of AVFrame
+- libavcodec/exr : fix decoding piz float file.
+- avformat/mov: Check sample size
+- lavfi: Move new field to the end of AVFilterContext
+- lavfi: Move new field to the end of AVFilterLink
+- ffplay: Fix usage of private lavfi API
+- lavc/mediacodecdec_h264: add missing NAL headers to SPS/PPS buffers
+- lavc/pnm_parser: disable parsing for text based PNMs
 
 
 version 3.1:

LICENSE.md

@@ -17,6 +17,7 @@ Specifically, the GPL parts of FFmpeg are:
     - `libavcodec/x86/flac_dsp_gpl.asm`
     - `libavcodec/x86/idct_mmx.c`
     - `libavfilter/x86/vf_removegrain.asm`
+- the X11 grabber in `libavdevice/x11grab.c`
 - the following building and testing tools
     - `compat/solaris/make_sunver.pl`
     - `doc/t2h.pm`
@@ -25,6 +26,7 @@ Specifically, the GPL parts of FFmpeg are:
     - `tests/checkasm/*`
     - `tests/tiny_ssim.c`
 - the following filters in libavfilter:
+    - `f_ebur128.c`
     - `vf_blackframe.c`
     - `vf_boxblur.c`
     - `vf_colormatrix.c`
@@ -113,6 +115,8 @@ The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
 incompatible with the GPLv2 and v3. To the best of our knowledge, they are
 compatible with the LGPL.
 
+The FAAC library is incompatible with all versions of GPL and LGPL.
+
 The NVENC library, while its header file is licensed under the compatible MIT
 license, requires a proprietary binary blob at run time, and is deemed to be
 incompatible with the GPL. We are not certain if it is compatible with the

MAINTAINERS

@@ -29,6 +29,9 @@ ffplay:
 ffprobe:
   ffprobe.c                             Stefano Sabatini
 
+ffserver:
+  ffserver.c                            Reynaldo H. Verdejo Pinochet
+
 Commandline utility code:
   cmdutils.c, cmdutils.h                Michael Niedermayer
 
@@ -39,12 +42,11 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan, Gyan Doshi
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
-API tests                               Ludmila Glinskih
 
 
 Communication
@@ -57,7 +59,6 @@ mailing lists                           Baptiste Coudurier, Lou Logan
 Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
 Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu
-ffmpeg-security                         Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
 
 
 libavutil
@@ -77,7 +78,6 @@ Other:
   eval.c, eval.h                        Michael Niedermayer
   float_dsp                             Loren Merritt
   hash                                  Reimar Doeffinger
-  hwcontext_cuda*                       Timo Rothenpieler
   intfloat*                             Michael Niedermayer
   integer.c, integer.h                  Michael Niedermayer
   lzo                                   Reimar Doeffinger
@@ -113,8 +113,6 @@ Generic Parts:
     lzw.*                               Michael Niedermayer
   floating point AAN DCT:
     faandct.c, faandct.h                Michael Niedermayer
-  Non-power-of-two MDCT:
-    mdct15.c, mdct15.h                  Rostislav Pehlivanov
   Golomb coding:
     golomb.c, golomb.h                  Michael Niedermayer
   motion estimation:
@@ -138,12 +136,10 @@ Codecs:
   8svx.c                                Jaikrishnan Menon
   aacenc*, aaccoder.c                   Rostislav Pehlivanov
   alacenc.c                             Jaikrishnan Menon
-  alsdec.c                              Thilo Borgmann, Umair Khan
-  aptx.c                                Aurelien Jacobs
+  alsdec.c                              Thilo Borgmann
   ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3plus*                           Maxim Poliakovski
-  audiotoolbox*                         Rodger Combs
   bgmc.c, bgmc.h                        Thilo Borgmann
   binkaudio.c                           Peter Ross
   cavs*                                 Stefan Gehrer
@@ -151,16 +147,14 @@ Codecs:
   celp_filters.*                        Vitor Sessak
   cinepak.c                             Roberto Togni
   cinepakenc.c                          Rl / Aetey G.T. AB
-  ccaption_dec.c                        Anshul Maheshwari, Aman Gupta
+  ccaption_dec.c                        Anshul Maheshwari
   cljr                                  Alex Beregszaszi
   cpia.c                                Stephan Hilb
   crystalhd.c                           Philip Langdale
   cscd.c                                Reimar Doeffinger
-  cuviddec.c                            Timo Rothenpieler
-  dca*                                  foo86
+  cuvid.c                               Timo Rothenpieler
   dirac*                                Rostislav Pehlivanov
   dnxhd*                                Baptiste Coudurier
-  dolby_e*                              foo86
   dpcm.c                                Mike Melanson
   dss_sp.c                              Oleksij Rempel
   dv.c                                  Roman Shaposhnik
@@ -168,10 +162,8 @@ Codecs:
   eacmv*, eaidct*, eat*                 Peter Ross
   evrc*                                 Paul B Mahol
   exif.c, exif.h                        Thilo Borgmann
-  exr.c                                 Martin Vignali
   ffv1*                                 Michael Niedermayer
   ffwavesynth.c                         Nicolas George
-  fifo.c                                Jan Sebechlebsky
   flicvideo.c                           Mike Melanson
   g722.c                                Martin Storsjo
   g726.c                                Roman Shaposhnik
@@ -180,7 +172,7 @@ Codecs:
   h263*                                 Michael Niedermayer
   h264*                                 Loren Merritt, Michael Niedermayer
   hap*                                  Tom Butterworth
-  huffyuv*                              Michael Niedermayer
+  huffyuv*                              Michael Niedermayer, Christophe Gisquet
   idcinvideo.c                          Mike Melanson
   interplayvideo.c                      Mike Melanson
   jni*, ffjni*                          Matthieu Bouron
@@ -188,12 +180,12 @@ Codecs:
   jvdec.c                               Peter Ross
   lcl*.c                                Roberto Togni, Reimar Doeffinger
   libcelt_dec.c                         Nicolas George
-  libcodec2.c                           Tomas Härdin
   libdirac*                             David Conrad
   libgsm.c                              Michel Bardiaux
   libkvazaar.c                          Arttu Ylä-Outinen
   libopenjpeg.c                         Jaikrishnan Menon
   libopenjpegenc.c                      Michael Bradshaw
+  libschroedinger*                      David Conrad
   libtheoraenc.c                        David Conrad
   libvorbis.c                           David Conrad
   libvpx*                               James Zern
@@ -203,7 +195,7 @@ Codecs:
   mdec.c                                Michael Niedermayer
   mimic.c                               Ramiro Polla
   mjpeg*.c                              Michael Niedermayer
-  mlp*                                  Ramiro Polla, Jai Luthra
+  mlp*                                  Ramiro Polla
   mmvideo.c                             Peter Ross
   mpeg12.c, mpeg12data.h                Michael Niedermayer
   mpegvideo.c, mpegvideo.h              Michael Niedermayer
@@ -212,15 +204,14 @@ Codecs:
   msrle.c                               Mike Melanson
   msvideo1.c                            Mike Melanson
   nuv.c                                 Reimar Doeffinger
-  nvdec*, nvenc*                        Timo Rothenpieler
-  opus*                                 Rostislav Pehlivanov
+  nvenc*                                Timo Rothenpieler
   paf.*                                 Paul B Mahol
   pcx.c                                 Ivo van Poorten
   pgssubdec.c                           Reimar Doeffinger
   ptx.c                                 Ivo van Poorten
   qcelp*                                Reynaldo H. Verdejo Pinochet
   qdm2.c, qdm2data.h                    Roberto Togni
-  qsv*                                  Mark Thompson
+  qsv*                                  Ivan Uskov
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
@@ -228,12 +219,12 @@ Codecs:
   rpza.c                                Roberto Togni
   rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
   rv10.c                                Michael Niedermayer
+  rv4*                                  Christophe Gisquet
   s3tc*                                 Ivo van Poorten
   smc.c                                 Mike Melanson
   smvjpegdec.c                          Ash Hughes
   snow*                                 Michael Niedermayer, Loren Merritt
   sonic.c                               Alex Beregszaszi
-  speedhq.c                             Steinar H. Gunderson
   srt*                                  Aurelien Jacobs
   sunrast.c                             Ivo van Poorten
   svq3.c                                Michael Niedermayer
@@ -242,10 +233,11 @@ Codecs:
   tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
   ttaenc.c                              Paul B Mahol
   txd.c                                 Ivo van Poorten
-  v4l2_*                                Jorge Ramirez-Ortiz
+  vc1*                                  Christophe Gisquet
   vc2*                                  Rostislav Pehlivanov
   vcr1.c                                Michael Niedermayer
-  videotoolboxenc.c                     Rick Kern, Aman Gupta
+  vda_h264_dec.c                        Xidorn Quan
+  videotoolboxenc.c                     Rick Kern
   vima.c                                Paul B Mahol
   vorbisdec.c                           Denes Balatoni, David Conrad
   vorbisenc.c                           Oded Shimon
@@ -266,13 +258,12 @@ Codecs:
 
 Hardware acceleration:
   crystalhd.c                           Philip Langdale
-  dxva2*                                Hendrik Leppkes, Laurent Aimar, Steve Lhomme
-  d3d11va*                              Steve Lhomme
-  mediacodec*                           Matthieu Bouron, Aman Gupta
+  dxva2*                                Hendrik Leppkes, Laurent Aimar
+  mediacodec*                           Matthieu Bouron
   vaapi*                                Gwenole Beauchesne
   vaapi_encode*                         Mark Thompson
   vdpau*                                Philip Langdale, Carl Eugen Hoyos
-  videotoolbox*                         Rick Kern, Aman Gupta
+  videotoolbox*                         Rick Kern
 
 
 libavdevice
@@ -282,8 +273,7 @@ libavdevice
 
 
   avfoundation.m                        Thilo Borgmann
-  android_camera.c                      Felix Matouschek
-  decklink*                             Marton Balint
+  decklink*                             Deti Fliegl
   dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
   fbdev_enc.c                           Lukasz Marek
   gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
@@ -292,8 +282,8 @@ libavdevice
   libdc1394.c                           Roman Shaposhnik
   opengl_enc.c                          Lukasz Marek
   pulse_audio_enc.c                     Lukasz Marek
+  qtkit.m                               Thilo Borgmann
   sdl                                   Stefano Sabatini
-  sdl2.c                                Josh de Kock
   v4l2.c                                Giorgio Vazzana
   vfwcap.c                              Ramiro Polla
   xv.c                                  Lukasz Marek
@@ -304,8 +294,6 @@ libavfilter
 Generic parts:
   graphdump.c                           Nicolas George
 
-  motion_estimation.c                   Davinder Singh
-
 Filters:
   f_drawgraph.c                         Paul B Mahol
   af_adelay.c                           Paul B Mahol
@@ -320,7 +308,6 @@ Filters:
   af_chorus.c                           Paul B Mahol
   af_compand.c                          Paul B Mahol
   af_firequalizer.c                     Muhammad Faiz
-  af_hdcd.c                             Burt P.
   af_ladspa.c                           Paul B Mahol
   af_loudnorm.c                         Kyle Swanson
   af_pan.c                              Nicolas George
@@ -330,7 +317,6 @@ Filters:
   avf_avectorscope.c                    Paul B Mahol
   avf_showcqt.c                         Muhammad Faiz
   vf_blend.c                            Paul B Mahol
-  vf_bwdif                              Thomas Mundt (CC <thomas.mundt@hr.de>)
   vf_chromakey.c                        Timo Rothenpieler
   vf_colorchannelmixer.c                Paul B Mahol
   vf_colorbalance.c                     Paul B Mahol
@@ -346,11 +332,8 @@ Filters:
   vf_hqx.c                              Clément Bœsch
   vf_idet.c                             Pascal Massimino
   vf_il.c                               Paul B Mahol
-  vf_(t)interlace                       Thomas Mundt (CC <thomas.mundt@hr.de>)
   vf_lenscorrection.c                   Daniel Oberhoff
   vf_mergeplanes.c                      Paul B Mahol
-  vf_mestimate.c                        Davinder Singh
-  vf_minterpolate.c                     Davinder Singh
   vf_neighbor.c                         Paul B Mahol
   vf_psnr.c                             Paul B Mahol
   vf_random.c                           Paul B Mahol
@@ -389,32 +372,26 @@ Muxers/Demuxers:
   astdec.c                              Paul B Mahol
   astenc.c                              James Almer
   avi*                                  Michael Niedermayer
-  avisynth.c                            Stephen Hutchinson
+  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
   avr.c                                 Paul B Mahol
   bink.c                                Peter Ross
-  boadec.c                              Michael Niedermayer
   brstm.c                               Paul B Mahol
   caf*                                  Peter Ross
   cdxl.c                                Paul B Mahol
-  codec2.c                              Tomas Härdin
   crc.c                                 Michael Niedermayer
-  dashdec.c                             Steven Liu
-  dashenc.c                             Karthick Jeyapal
   daud.c                                Reimar Doeffinger
   dss.c                                 Oleksij Rempel
-  dtsdec.c                              foo86
   dtshddec.c                            Paul B Mahol
   dv.c                                  Roman Shaposhnik
   electronicarts.c                      Peter Ross
   epafdec.c                             Paul B Mahol
   ffm*                                  Baptiste Coudurier
   flic.c                                Mike Melanson
-  flvdec.c                              Michael Niedermayer
-  flvenc.c                              Michael Niedermayer, Steven Liu
+  flvdec.c, flvenc.c                    Michael Niedermayer
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
   hls.c                                 Anssi Hannula
-  hlsenc.c                              Christian Suloway, Steven Liu
+  hls encryption (hlsenc.c)             Christian Suloway
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
@@ -424,7 +401,7 @@ Muxers/Demuxers:
   iss.c                                 Stefan Gehrer
   jvdec.c                               Peter Ross
   libmodplug.c                          Clément Bœsch
-  libopenmpt.c                          Josh de Kock
+  libnut.c                              Oded Shimon
   lmlm4.c                               Ivo van Poorten
   lvfdec.c                              Paul B Mahol
   lxfdec.c                              Tomas Härdin
@@ -446,6 +423,7 @@ Muxers/Demuxers:
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
   mxf*                                  Baptiste Coudurier
+  mxfdec.c                              Tomas Härdin
   nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
   nut*                                  Michael Niedermayer
@@ -474,7 +452,6 @@ Muxers/Demuxers:
   rtpdec_vc2hq.*, rtpenc_vc2hq.*        Thomas Volkert
   rtpdec_vp9.c                          Thomas Volkert
   rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
-  s337m.c                               foo86
   sbgdec.c                              Nicolas George
   sdp.c                                 Martin Storsjo
   segafilm.c                            Mike Melanson
@@ -524,7 +501,7 @@ Operating systems / CPU architectures
 =====================================
 
 Alpha                                   Falk Hueffner
-MIPS                                    Manojkumar Bhosale
+MIPS                                    Nedeljko Babic
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
@@ -536,34 +513,6 @@ Sparc                                   Roman Shaposhnik
 OS/2                                    KO Myung-Hun
 
 
-Developers with git write access who are currently not maintaining any specific part
-====================================================================================
-Alex Converse
-Andreas Cadhalpun
-Anuradha Suraparaju
-Ben Littler
-Benjamin Larsson
-Bobby Bingham
-Daniel Verkamp
-Derek Buitenhuis
-Ganesh Ajjanagadde
-Henrik Gramner
-Ivan Uskov
-James Darnley
-Jan Ekström
-Joakim Plate
-Jun Zhao
-Kieran Kunhya
-Kirill Gavrilov
-Martin Storsjö
-Panagiotis Issaris
-Pedro Arthur
-Sebastien Zwickert
-Vittorio Giovara
-wm4
-(this list is incomplete)
-
-
 Releases
 ========
 
@@ -571,6 +520,7 @@ Releases
 2.7                                     Michael Niedermayer
 2.6                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
+2.4                                     Michael Niedermayer
 
 If you want to maintain an older release, please contact us
 
@@ -591,13 +541,11 @@ FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Ganesh Ajjanagadde            C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
-James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
-Nikolay Aleksandrov           8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
 Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
@@ -607,7 +555,6 @@ Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
-Steinar H. Gunderson          C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
 Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83

Makefile

@@ -1,5 +1,5 @@
 MAIN_MAKEFILE=1
-include ffbuild/config.mak
+include config.mak
 
 vpath %.c    $(SRC_PATH)
 vpath %.cpp  $(SRC_PATH)
@@ -11,12 +11,40 @@ vpath %.asm  $(SRC_PATH)
 vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
-vpath %.cu   $(SRC_PATH)
-vpath %.ptx  $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
 
+AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
+AVPROGS-$(CONFIG_FFPLAY)   += ffplay
+AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
+AVPROGS-$(CONFIG_FFSERVER) += ffserver
+
+AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS      += $(AVPROGS)
+
+AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
+ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
+
+OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
+OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
+OBJS-ffmpeg-$(CONFIG_LIBMFX)  += ffmpeg_qsv.o
+OBJS-ffmpeg-$(CONFIG_VAAPI)   += ffmpeg_vaapi.o
+ifndef CONFIG_VIDEOTOOLBOX
+OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_videotoolbox.o
+endif
+OBJS-ffmpeg-$(CONFIG_CUVID)   += ffmpeg_cuvid.o
+OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
+OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
+OBJS-ffserver                 += ffserver_config.o
+
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+TOOLS       = qt-faststart trasher uncoded_frame
+TOOLS-$(CONFIG_ZLIB) += cws2fws
 
 # $(FFLIBS-yes) needs to be in linking order
 FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
@@ -31,45 +59,36 @@ 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 $(SRC_PATH)/doc/examples/README
 
-SKIPHEADERS = compat/w32pthreads.h
+SKIPHEADERS = cmdutils_common_opts.h                                    \
+              compat/w32pthreads.h
 
-# first so "all" becomes default target
-all: all-yes
-
-include $(SRC_PATH)/tools/Makefile
-include $(SRC_PATH)/ffbuild/common.mak
+include $(SRC_PATH)/common.mak
 
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
 
-$(TOOLS): %$(EXESUF): %.o
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
+all: $(AVPROGS)
 
-target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
+$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
 
-tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
+tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
 tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
-tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
 
-CONFIGURABLE_COMPONENTS =                                           \
-    $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
-    $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
-    $(SRC_PATH)/libavformat/protocols.c                             \
-
-config.h: ffbuild/.config
-ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
+config.h: .config
+.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
 	@-tput bold 2>/dev/null
-	@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
+	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
 
-SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
+SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
                HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
                ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
-               ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS                \
+               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
                MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS         \
                MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
 
@@ -84,32 +103,41 @@ SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
 -include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
 -include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
-include $(SRC_PATH)/ffbuild/library.mak
+include $(SRC_PATH)/library.mak
 endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
-include $(SRC_PATH)/fftools/Makefile
 include $(SRC_PATH)/doc/Makefile
-include $(SRC_PATH)/doc/examples/Makefile
 
-libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
+define DOPROG
+OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
+$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
+$$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
+$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
+$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
+-include $$(OBJS-$(1):.o=.d)
+endef
+
+$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
+
+ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
 
 $(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
-ifeq ($(STRIPTYPE),direct)
-	$(STRIP) -o $@ $<
-else
 	$(CP) $< $@
 	$(STRIP) $@
-endif
 
-%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
+%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
 
-VERSION_SH  = $(SRC_PATH)/ffbuild/version.sh
+OBJDIRS += tools
+
+-include $(wildcard tools/*.d)
+
+VERSION_SH  = $(SRC_PATH)/version.sh
 GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 
-.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
+.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
 
 libavutil/ffversion.h .version:
@@ -119,32 +147,44 @@ libavutil/ffversion.h .version:
 # force version.sh to run whenever version might have changed
 -include .version
 
+ifdef AVPROGS
+install: install-progs install-data
+endif
+
 install: install-libs install-headers
 
 install-libs: install-libs-yes
 
-install-data: $(DATA_FILES)
-	$(Q)mkdir -p "$(DATADIR)"
-	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
+install-progs-yes:
+install-progs-$(CONFIG_SHARED): install-libs
 
-uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
+install-progs: install-progs-yes $(AVPROGS)
+	$(Q)mkdir -p "$(BINDIR)"
+	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
+
+install-data: $(DATA_FILES) $(EXAMPLES_FILES)
+	$(Q)mkdir -p "$(DATADIR)/examples"
+	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
+	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
+
+uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
+
+uninstall-progs:
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
 
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 
 clean::
+	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES))
+	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
 	$(RM) -rf coverage.info coverage.info.in lcov
 
-distclean:: clean
-	$(RM) .version avversion.h config.asm config.h mapfile  \
-		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
-		version.h libavutil/ffversion.h libavcodec/codec_names.h \
-		libavcodec/bsf_list.c libavformat/protocol_list.c \
-		libavcodec/codec_list.c libavcodec/parser_list.c \
-		libavformat/muxer_list.c libavformat/demuxer_list.c
+distclean::
+	$(RM) $(DISTCLEANSUFFIXES)
+	$(RM) config.* .config libavutil/avconfig.h .version mapfile avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h libavcodec/bsf_list.c libavformat/protocol_list.c
 ifeq ($(SRC_LINK),src)
 	$(RM) src
 endif
@@ -153,7 +193,6 @@ endif
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
-build: all alltools examples testprogs
 check: all alltools examples testprogs fate
 
 include $(SRC_PATH)/tests/Makefile
@@ -169,5 +208,5 @@ $(sort $(OBJDIRS)):
 # so this saves some time on slow systems.
 .SUFFIXES:
 
-.PHONY: all all-yes alltools build check config testprogs
-.PHONY: *clean install* uninstall*
+.PHONY: all all-yes alltools check *clean config install*
+.PHONY: testprogs uninstall*

README.md

@@ -21,6 +21,8 @@ such as audio, video, subtitles and related metadata.
 * [ffplay](https://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
 * [ffprobe](https://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
   multimedia content.
+* [ffserver](https://ffmpeg.org/ffserver.html) is a multimedia streaming server
+  for live broadcasts.
 * Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
 
 ## Documentation
@@ -43,4 +45,5 @@ GPL. Please refer to the LICENSE file for detailed information.
 
 Patches should be submitted to the ffmpeg-devel mailing list using
 `git format-patch` or `git send-email`. Github pull requests should be
-avoided because they are not part of our review process and will be ignored.
+avoided because they are not part of our review process. Few developers
+follow pull requests so they will likely be ignored.

RELEASE

@@ -1 +1 @@
-4.0.2
+3.1.7

RELEASE_NOTES

@@ -1,13 +1,13 @@
 
-              ┌───────────────────────────────────┐
-              │ RELEASE NOTES for FFmpeg 4.0 "Wu" │
-              └───────────────────────────────────┘
+              ┌────────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 3.1 "Laplace" │
+              └────────────────────────────────────────┘
 
-   The FFmpeg Project proudly presents FFmpeg 4.0 "Wu", about 6
-   months after the release of FFmpeg 3.4.
+   The FFmpeg Project proudly presents FFmpeg 3.1 "Laplace", about 4
+   months after the release of FFmpeg 3.0.
 
    A complete Changelog is available at the root of the project, and the
-   complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
+   complete Git history on http://source.ffmpeg.org.
 
    We hope you will like this release as much as we enjoyed working on it, and
    as usual, if you have any questions about it, or any FFmpeg related topic,

arch.mak

@@ -14,4 +14,4 @@ OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
 
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
-OBJS-$(HAVE_X86ASM)  += $(X86ASM-OBJS)  $(X86ASM-OBJS-yes)
+OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.c

@@ -38,7 +38,6 @@
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
 #include "libpostproc/postprocess.h"
-#include "libavutil/attributes.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
@@ -62,7 +61,7 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
-#ifdef _WIN32
+#if HAVE_SETDLLDIRECTORY
 #include <windows.h>
 #endif
 
@@ -76,12 +75,6 @@ static FILE *report_file;
 static int report_file_level = AV_LOG_DEBUG;
 int hide_banner = 0;
 
-enum show_muxdemuxers {
-    SHOW_DEFAULT,
-    SHOW_DEMUXERS,
-    SHOW_MUXERS,
-};
-
 void init_opts(void)
 {
     av_dict_set(&sws_dict, "flags", "bicubic", 0);
@@ -119,7 +112,7 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
 
 void init_dynload(void)
 {
-#ifdef _WIN32
+#if HAVE_SETDLLDIRECTORY
     /* Calling SetDllDirectory with the empty string (but not NULL) removes the
      * current working directory from the DLL search path as a security pre-caution. */
     SetDllDirectory("");
@@ -232,6 +225,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
  * by default. HAVE_COMMANDLINETOARGVW is true on cygwin, while
  * it doesn't provide the actual command line via GetCommandLineW(). */
 #if HAVE_COMMANDLINETOARGVW && defined(_WIN32)
+#include <windows.h>
 #include <shellapi.h>
 /* Will be leaked on exit */
 static char** win32_argv_utf8 = NULL;
@@ -881,54 +875,28 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
         { "debug"  , AV_LOG_DEBUG   },
         { "trace"  , AV_LOG_TRACE   },
     };
-    const char *token;
     char *tail;
-    int flags = av_log_get_flags();
-    int level = av_log_get_level();
-    int cmd, i = 0;
+    int level;
+    int flags;
+    int i;
 
-    av_assert0(arg);
-    while (*arg) {
-        token = arg;
-        if (*token == '+' || *token == '-') {
-            cmd = *token++;
-        } else {
-            cmd = 0;
-        }
-        if (!i && !cmd) {
-            flags = 0;  /* missing relative prefix, build absolute value */
-        }
-        if (!strncmp(token, "repeat", 6)) {
-            if (cmd == '-') {
-                flags |= AV_LOG_SKIP_REPEATED;
-            } else {
-                flags &= ~AV_LOG_SKIP_REPEATED;
-            }
-            arg = token + 6;
-        } else if (!strncmp(token, "level", 5)) {
-            if (cmd == '-') {
-                flags &= ~AV_LOG_PRINT_LEVEL;
-            } else {
-                flags |= AV_LOG_PRINT_LEVEL;
-            }
-            arg = token + 5;
-        } else {
-            break;
-        }
-        i++;
-    }
-    if (!*arg) {
-        goto end;
-    } else if (*arg == '+') {
-        arg++;
-    } else if (!i) {
-        flags = av_log_get_flags();  /* level value without prefix, reset flags */
-    }
+    flags = av_log_get_flags();
+    tail = strstr(arg, "repeat");
+    if (tail)
+        flags &= ~AV_LOG_SKIP_REPEATED;
+    else
+        flags |= AV_LOG_SKIP_REPEATED;
+
+    av_log_set_flags(flags);
+    if (tail == arg)
+        arg += 6 + (arg[6]=='+');
+    if(tail && !*arg)
+        return 0;
 
     for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) {
         if (!strcmp(log_levels[i].name, arg)) {
-            level = log_levels[i].level;
-            goto end;
+            av_log_set_level(log_levels[i].level);
+            return 0;
         }
     }
 
@@ -940,9 +908,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
             av_log(NULL, AV_LOG_FATAL, "\"%s\"\n", log_levels[i].name);
         exit_program(1);
     }
-
-end:
-    av_log_set_flags(flags);
     av_log_set_level(level);
     return 0;
 }
@@ -1286,12 +1251,10 @@ static int is_device(const AVClass *avclass)
     return AV_IS_INPUT_DEVICE(avclass->category) || AV_IS_OUTPUT_DEVICE(avclass->category);
 }
 
-static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only, int muxdemuxers)
+static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
 {
-    void *ifmt_opaque = NULL;
-    const AVInputFormat *ifmt  = NULL;
-    void *ofmt_opaque = NULL;
-    const AVOutputFormat *ofmt = NULL;
+    AVInputFormat *ifmt  = NULL;
+    AVOutputFormat *ofmt = NULL;
     const char *last_name;
     int is_dev;
 
@@ -1306,35 +1269,29 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
         const char *name      = NULL;
         const char *long_name = NULL;
 
-        if (muxdemuxers !=SHOW_DEMUXERS) {
-            ofmt_opaque = NULL;
-            while ((ofmt = av_muxer_iterate(&ofmt_opaque))) {
-                is_dev = is_device(ofmt->priv_class);
-                if (!is_dev && device_only)
-                    continue;
-                if ((!name || 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))) {
+            is_dev = is_device(ofmt->priv_class);
+            if (!is_dev && device_only)
+                continue;
+            if ((!name || strcmp(ofmt->name, name) < 0) &&
+                strcmp(ofmt->name, last_name) > 0) {
+                name      = ofmt->name;
+                long_name = ofmt->long_name;
+                encode    = 1;
             }
         }
-        if (muxdemuxers != SHOW_MUXERS) {
-            ifmt_opaque = NULL;
-            while ((ifmt = av_demuxer_iterate(&ifmt_opaque))) {
-                is_dev = is_device(ifmt->priv_class);
-                if (!is_dev && device_only)
-                    continue;
-                if ((!name || 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;
+        while ((ifmt = av_iformat_next(ifmt))) {
+            is_dev = is_device(ifmt->priv_class);
+            if (!is_dev && device_only)
+                continue;
+            if ((!name || 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)
             break;
@@ -1351,22 +1308,12 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
 
 int show_formats(void *optctx, const char *opt, const char *arg)
 {
-    return show_formats_devices(optctx, opt, arg, 0, SHOW_DEFAULT);
-}
-
-int show_muxers(void *optctx, const char *opt, const char *arg)
-{
-    return show_formats_devices(optctx, opt, arg, 0, SHOW_MUXERS);
-}
-
-int show_demuxers(void *optctx, const char *opt, const char *arg)
-{
-    return show_formats_devices(optctx, opt, arg, 0, SHOW_DEMUXERS);
+    return show_formats_devices(optctx, opt, arg, 0);
 }
 
 int show_devices(void *optctx, const char *opt, const char *arg)
 {
-    return show_formats_devices(optctx, opt, arg, 1, SHOW_DEFAULT);
+    return show_formats_devices(optctx, opt, arg, 1);
 }
 
 #define PRINT_CODEC_SUPPORTED(codec, field, type, list_name, term, get_name) \
@@ -1632,11 +1579,10 @@ int show_encoders(void *optctx, const char *opt, const char *arg)
 
 int show_bsfs(void *optctx, const char *opt, const char *arg)
 {
-    const AVBitStreamFilter *bsf = NULL;
-    void *opaque = NULL;
+    AVBitStreamFilter *bsf = NULL;
 
     printf("Bitstream filters:\n");
-    while ((bsf = av_bsf_iterate(&opaque)))
+    while ((bsf = av_bitstream_filter_next(bsf)))
         printf("%s\n", bsf->name);
     printf("\n");
     return 0;
@@ -1662,7 +1608,6 @@ int show_filters(void *optctx, const char *opt, const char *arg)
 #if CONFIG_AVFILTER
     const AVFilter *filter = NULL;
     char descr[64], *descr_cur;
-    void *opaque = NULL;
     int i, j;
     const AVFilterPad *pad;
 
@@ -1674,7 +1619,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
            "  V = Video input/output\n"
            "  N = Dynamic number and/or type of input/output\n"
            "  | = Source or sink filter\n");
-    while ((filter = av_filter_iterate(&opaque))) {
+    while ((filter = avfilter_next(filter))) {
         descr_cur = descr;
         for (i = 0; i < 2; i++) {
             if (i) {
@@ -1737,7 +1682,7 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg)
 #endif
 
     while ((pix_desc = av_pix_fmt_desc_next(pix_desc))) {
-        enum AVPixelFormat av_unused pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
+        enum AVPixelFormat pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
         printf("%c%c%c%c%c %-16s       %d            %2d\n",
                sws_isSupportedInput (pix_fmt)              ? 'I' : '.',
                sws_isSupportedOutput(pix_fmt)              ? 'O' : '.',
@@ -1931,22 +1876,6 @@ static void show_help_filter(const char *name)
 }
 #endif
 
-static void show_help_bsf(const char *name)
-{
-    const AVBitStreamFilter *bsf = av_bsf_get_by_name(name);
-
-    if (!bsf) {
-        av_log(NULL, AV_LOG_ERROR, "Unknown bit stream filter '%s'.\n", name);
-        return;
-    }
-
-    printf("Bit stream filter %s\n", bsf->name);
-    PRINT_CODEC_SUPPORTED(bsf, codec_ids, enum AVCodecID, "codecs",
-                          AV_CODEC_ID_NONE, GET_CODEC_NAME);
-    if (bsf->priv_class)
-        show_help_children(bsf->priv_class, AV_OPT_FLAG_BSF_PARAM);
-}
-
 int show_help(void *optctx, const char *opt, const char *arg)
 {
     char *topic, *par;
@@ -1973,8 +1902,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
     } else if (!strcmp(topic, "filter")) {
         show_help_filter(par);
 #endif
-    } else if (!strcmp(topic, "bsf")) {
-        show_help_bsf(par);
     } else {
         show_help_default(topic, par);
     }
@@ -2066,7 +1993,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
         codec            = s->oformat ? avcodec_find_encoder(codec_id)
                                       : avcodec_find_decoder(codec_id);
 
-    switch (st->codecpar->codec_type) {
+    switch (st->codec->codec_type) {
     case AVMEDIA_TYPE_VIDEO:
         prefix  = 'v';
         flags  |= AV_OPT_FLAG_VIDEO_PARAM;
@@ -2124,7 +2051,7 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
         return NULL;
     }
     for (i = 0; i < s->nb_streams; i++)
-        opts[i] = filter_codec_opts(codec_opts, s->streams[i]->codecpar->codec_id,
+        opts[i] = filter_codec_opts(codec_opts, s->streams[i]->codec->codec_id,
                                     s, s->streams[i], NULL);
     return opts;
 }
@@ -2150,10 +2077,18 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
 
 double get_rotation(AVStream *st)
 {
+    AVDictionaryEntry *rotate_tag = av_dict_get(st->metadata, "rotate", NULL, 0);
     uint8_t* displaymatrix = av_stream_get_side_data(st,
                                                      AV_PKT_DATA_DISPLAYMATRIX, NULL);
     double theta = 0;
-    if (displaymatrix)
+
+    if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
+        char *tail;
+        theta = av_strtod(rotate_tag->value, &tail);
+        if (*tail)
+            theta = 0;
+    }
+    if (displaymatrix && !theta)
         theta = -av_display_rotation_get((int32_t*) displaymatrix);
 
     theta -= 360*floor(theta/360 + 0.9/360);

cmdutils.h

@@ -19,8 +19,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef FFTOOLS_CMDUTILS_H
-#define FFTOOLS_CMDUTILS_H
+#ifndef CMDUTILS_H
+#define CMDUTILS_H
 
 #include <stdint.h>
 
@@ -105,6 +105,12 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);
 
+#if CONFIG_OPENCL
+int opt_opencl(void *optctx, const char *opt, const char *arg);
+
+int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
+#endif
+
 /**
  * Limit the execution time.
  */
@@ -149,7 +155,6 @@ typedef struct SpecifierOpt {
         uint8_t *str;
         int        i;
         int64_t  i64;
-        uint64_t ui64;
         float      f;
         double   dbl;
     } u;
@@ -201,47 +206,6 @@ typedef struct OptionDef {
 void show_help_options(const OptionDef *options, const char *msg, int req_flags,
                        int rej_flags, int alt_flags);
 
-#if CONFIG_AVDEVICE
-#define CMDUTILS_COMMON_OPTIONS_AVDEVICE                                                                                \
-    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },                                                  \
-      "list sources of the input device", "device" },                                                                   \
-    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },                                                    \
-      "list sinks of the output device", "device" },                                                                    \
-
-#else
-#define CMDUTILS_COMMON_OPTIONS_AVDEVICE
-#endif
-
-#define CMDUTILS_COMMON_OPTIONS                                                                                         \
-    { "L",           OPT_EXIT,             { .func_arg = show_license },     "show license" },                          \
-    { "h",           OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
-    { "?",           OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
-    { "help",        OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
-    { "-help",       OPT_EXIT,             { .func_arg = show_help },        "show help", "topic" },                    \
-    { "version",     OPT_EXIT,             { .func_arg = show_version },     "show version" },                          \
-    { "buildconf",   OPT_EXIT,             { .func_arg = show_buildconf },   "show build configuration" },              \
-    { "formats",     OPT_EXIT,             { .func_arg = show_formats },     "show available formats" },                \
-    { "muxers",      OPT_EXIT,             { .func_arg = show_muxers },      "show available muxers" },                 \
-    { "demuxers",    OPT_EXIT,             { .func_arg = show_demuxers },    "show available demuxers" },               \
-    { "devices",     OPT_EXIT,             { .func_arg = show_devices },     "show available devices" },                \
-    { "codecs",      OPT_EXIT,             { .func_arg = show_codecs },      "show available codecs" },                 \
-    { "decoders",    OPT_EXIT,             { .func_arg = show_decoders },    "show available decoders" },               \
-    { "encoders",    OPT_EXIT,             { .func_arg = show_encoders },    "show available encoders" },               \
-    { "bsfs",        OPT_EXIT,             { .func_arg = show_bsfs },        "show available bit stream filters" },     \
-    { "protocols",   OPT_EXIT,             { .func_arg = show_protocols },   "show available protocols" },              \
-    { "filters",     OPT_EXIT,             { .func_arg = show_filters },     "show available filters" },                \
-    { "pix_fmts",    OPT_EXIT,             { .func_arg = show_pix_fmts },    "show available pixel formats" },          \
-    { "layouts",     OPT_EXIT,             { .func_arg = show_layouts },     "show standard channel layouts" },         \
-    { "sample_fmts", OPT_EXIT,             { .func_arg = show_sample_fmts }, "show available audio sample formats" },   \
-    { "colors",      OPT_EXIT,             { .func_arg = show_colors },      "show available color names" },            \
-    { "loglevel",    HAS_ARG,              { .func_arg = opt_loglevel },     "set logging level", "loglevel" },         \
-    { "v",           HAS_ARG,              { .func_arg = opt_loglevel },     "set logging level", "loglevel" },         \
-    { "report",      0,                    { (void*)opt_report },            "generate a report" },                     \
-    { "max_alloc",   HAS_ARG,              { .func_arg = opt_max_alloc },    "set maximum size of a single allocated block", "bytes" }, \
-    { "cpuflags",    HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags },     "force specific cpu flags", "flags" },     \
-    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },          \
-    CMDUTILS_COMMON_OPTIONS_AVDEVICE                                                                                    \
-
 /**
  * Show help for all options with given flags in class and all its
  * children.
@@ -477,20 +441,6 @@ int show_license(void *optctx, const char *opt, const char *arg);
  */
 int show_formats(void *optctx, const char *opt, const char *arg);
 
-/**
- * Print a listing containing all the muxers supported by the
- * program (including devices).
- * This option processing function does not utilize the arguments.
- */
-int show_muxers(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the demuxer supported by the
- * program (including devices).
- * This option processing function does not utilize the arguments.
- */
-int show_demuxers(void *optctx, const char *opt, const char *arg);
-
 /**
  * Print a listing containing all the devices supported by the
  * program.
@@ -625,9 +575,6 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
 #define GET_PIX_FMT_NAME(pix_fmt)\
     const char *name = av_get_pix_fmt_name(pix_fmt);
 
-#define GET_CODEC_NAME(id)\
-    const char *name = avcodec_descriptor_get(id)->name;
-
 #define GET_SAMPLE_FMT_NAME(sample_fmt)\
     const char *name = av_get_sample_fmt_name(sample_fmt)
 
@@ -645,4 +592,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
 
 double get_rotation(AVStream *st);
 
-#endif /* FFTOOLS_CMDUTILS_H */
+#endif /* CMDUTILS_H */

cmdutils_common_opts.h

@@ -0,0 +1,35 @@
+    { "L"          , OPT_EXIT, {.func_arg = show_license},      "show license" },
+    { "h"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "?"          , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
+    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
+    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
+    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
+    { "devices"    , OPT_EXIT, {.func_arg = show_devices  },    "show available devices" },
+    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
+    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
+    { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
+    { "bsfs"       , OPT_EXIT, {.func_arg = show_bsfs     },    "show available bit stream filters" },
+    { "protocols"  , OPT_EXIT, {.func_arg = show_protocols},    "show available protocols" },
+    { "filters"    , OPT_EXIT, {.func_arg = show_filters  },    "show available filters" },
+    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
+    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
+    { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
+    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
+    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
+    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
+    { "report"     , 0,        {(void*)opt_report}, "generate a report" },
+    { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
+    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
+    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
+#if CONFIG_OPENCL
+    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
+    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
+#endif
+#if CONFIG_AVDEVICE
+    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
+      "list sources of the input device", "device" },
+    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
+      "list sinks of the output device", "device" },
+#endif

cmdutils_opencl.c

@@ -0,0 +1,278 @@
+/*
+ * Copyright (C) 2013 Lenny Wang
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "libavutil/log.h"
+#include "libavutil/opencl.h"
+#include "libavutil/avstring.h"
+#include "cmdutils.h"
+
+typedef struct {
+    int platform_idx;
+    int device_idx;
+    char device_name[64];
+    int64_t runtime;
+} OpenCLDeviceBenchmark;
+
+const char *ocl_bench_source = AV_OPENCL_KERNEL(
+inline unsigned char clip_uint8(int a)
+{
+    if (a & (~0xFF))
+        return (-a)>>31;
+    else
+        return a;
+}
+
+kernel void unsharp_bench(
+                    global unsigned char *src,
+                    global unsigned char *dst,
+                    global int *mask,
+                    int width,
+                    int height)
+{
+    int i, j, local_idx, lc_idx, sum = 0;
+    int2 thread_idx, block_idx, global_idx, lm_idx;
+    thread_idx.x = get_local_id(0);
+    thread_idx.y = get_local_id(1);
+    block_idx.x = get_group_id(0);
+    block_idx.y = get_group_id(1);
+    global_idx.x = get_global_id(0);
+    global_idx.y = get_global_id(1);
+    local uchar data[32][32];
+    local int lc[128];
+
+    for (i = 0; i <= 1; i++) {
+        lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
+        lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
+        lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
+        for (j = 0; j <= 1; j++) {
+            lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
+            lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
+            lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
+            data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
+        }
+    }
+    local_idx = thread_idx.y*16 + thread_idx.x;
+    if (local_idx < 128)
+        lc[local_idx] = mask[local_idx];
+    barrier(CLK_LOCAL_MEM_FENCE);
+
+    \n#pragma unroll\n
+    for (i = -4; i <= 4; i++) {
+        lm_idx.y = 8 + i + thread_idx.y;
+        \n#pragma unroll\n
+        for (j = -4; j <= 4; j++) {
+            lm_idx.x = 8 + j + thread_idx.x;
+            lc_idx = (i + 4)*8 + j + 4;
+            sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
+        }
+    }
+    int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
+    int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
+    if (global_idx.x < width && global_idx.y < height)
+        dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
+}
+);
+
+#define OCLCHECK(method, ... )                                                 \
+do {                                                                           \
+    status = method(__VA_ARGS__);                                              \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n",                   \
+               av_opencl_errstr(status));                                      \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+#define CREATEBUF(out, flags, size)                                            \
+do {                                                                           \
+    out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n");        \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+static void fill_rand_int(int *data, int n)
+{
+    int i;
+    srand(av_gettime());
+    for (i = 0; i < n; i++)
+        data[i] = rand();
+}
+
+#define OPENCL_NB_ITER 5
+static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
+{
+    int i, arg = 0, width = 1920, height = 1088;
+    int64_t start, ret = 0;
+    cl_int status;
+    size_t kernel_len;
+    char *inbuf;
+    int *mask;
+    int buf_size = width * height * sizeof(char);
+    int mask_size = sizeof(uint32_t) * 128;
+
+    cl_mem cl_mask, cl_inbuf, cl_outbuf;
+    cl_kernel kernel = NULL;
+    cl_program program = NULL;
+    size_t local_work_size_2d[2] = {16, 16};
+    size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
+
+    if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
+        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fill_rand_int((int*)inbuf, buf_size/4);
+    fill_rand_int(mask, mask_size/4);
+
+    CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
+    CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
+    CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
+
+    kernel_len = strlen(ocl_bench_source);
+    program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
+                                        &kernel_len, &status);
+    if (status != CL_SUCCESS || !program) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    kernel = clCreateKernel(program, "unsharp_bench", &status);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
+             buf_size, inbuf, 0, NULL, NULL);
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
+             mask_size, mask, 0, NULL, NULL);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
+
+    start = av_gettime_relative();
+    for (i = 0; i < OPENCL_NB_ITER; i++)
+        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
+                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
+    clFinish(ext_opencl_env->command_queue);
+    ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
+end:
+    if (kernel)
+        clReleaseKernel(kernel);
+    if (program)
+        clReleaseProgram(program);
+    if (cl_inbuf)
+        clReleaseMemObject(cl_inbuf);
+    if (cl_outbuf)
+        clReleaseMemObject(cl_outbuf);
+    if (cl_mask)
+        clReleaseMemObject(cl_mask);
+    av_free(inbuf);
+    av_free(mask);
+    return ret;
+}
+
+static int compare_ocl_device_desc(const void *a, const void *b)
+{
+    const OpenCLDeviceBenchmark* va = (const OpenCLDeviceBenchmark*)a;
+    const OpenCLDeviceBenchmark* vb = (const OpenCLDeviceBenchmark*)b;
+    return FFDIFFSIGN(va->runtime , vb->runtime);
+}
+
+int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
+{
+    int i, j, nb_devices = 0, count = 0;
+    int64_t score = 0;
+    AVOpenCLDeviceList *device_list;
+    AVOpenCLDeviceNode *device_node = NULL;
+    OpenCLDeviceBenchmark *devices = NULL;
+    cl_platform_id platform;
+
+    av_opencl_get_device_list(&device_list);
+    for (i = 0; i < device_list->platform_num; i++)
+        nb_devices += device_list->platform_node[i]->device_num;
+    if (!nb_devices) {
+        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
+        return AVERROR(EINVAL);
+    }
+    if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
+        return AVERROR(ENOMEM);
+    }
+
+    for (i = 0; i < device_list->platform_num; i++) {
+        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
+            device_node = device_list->platform_node[i]->device_node[j];
+            platform = device_list->platform_node[i]->platform_id;
+            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
+            if (score > 0) {
+                devices[count].platform_idx = i;
+                devices[count].device_idx = j;
+                devices[count].runtime = score;
+                av_strlcpy(devices[count].device_name, device_node->device_name,
+                           sizeof(devices[count].device_name));
+                count++;
+            }
+        }
+    }
+    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
+    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
+    for (i = 0; i < count; i++)
+        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
+                devices[i].platform_idx, devices[i].device_idx,
+                devices[i].device_name, devices[i].runtime);
+
+    av_opencl_free_device_list(&device_list);
+    av_free(devices);
+    return 0;
+}
+
+int opt_opencl(void *optctx, const char *opt, const char *arg)
+{
+    char *key, *value;
+    const char *opts = arg;
+    int ret = 0;
+    while (*opts) {
+        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
+        if (ret < 0)
+            return ret;
+        ret = av_opencl_set_option(key, value);
+        if (ret < 0)
+            return ret;
+        if (*opts)
+            opts++;
+    }
+    return ret;
+}

common.mak

@@ -2,12 +2,15 @@
 # common bits used by all libraries
 #
 
-DEFAULT_X86ASMD=.dbg
+# first so "all" becomes default target
+all: all-yes
+
+DEFAULT_YASMD=.dbg
 
 ifeq ($(DBG),1)
-X86ASMD=$(DEFAULT_X86ASMD)
+YASMD=$(DEFAULT_YASMD)
 else
-X86ASMD=
+YASMD=
 endif
 
 ifndef SUBDIR
@@ -15,8 +18,8 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
-SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
+BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
+SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
 
 MSG    = $@
 M      = @$(call ECHO,$(TAG),$@);
@@ -37,8 +40,7 @@ OBJCFLAGS  += $(EOBJCFLAGS)
 OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
-X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
-NVCCFLAGS  += -ptx
+YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
 
 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
@@ -52,9 +54,7 @@ COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
 COMPILE_M = $(call COMPILE,OBJCC)
-COMPILE_X86ASM = $(call COMPILE,X86ASM)
 COMPILE_HOSTC = $(call COMPILE,HOSTCC)
-COMPILE_NVCC = $(call COMPILE,NVCC)
 
 %.o: %.c
 	$(COMPILE_C)
@@ -74,14 +74,6 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
 %_host.o: %.c
 	$(COMPILE_HOSTC)
 
-%$(DEFAULT_X86ASMD).asm: %.asm
-	$(DEPX86ASM) $(X86ASMFLAGS) -M -o $@ $< > $(@:.asm=.d)
-	$(X86ASM) $(X86ASMFLAGS) -e $< | sed '/^%/d;/^$$/d;' > $@
-
-%.o: %.asm
-	$(COMPILE_X86ASM)
-	-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
-
 %.o: %.rc
 	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
 
@@ -91,13 +83,12 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@
 
-%.ptx: %.cu
-	$(COMPILE_NVCC)
+%.ver: %.v
+	$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ | sed -e 's/:/:\
+/' -e 's/; /;\
+/g' > $@
 
-%.ptx.c: %.ptx
-	$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
-
-%.c %.h %.pc %.ver %.version: TAG = GEN
+%.c %.h: TAG = GEN
 
 # Dummy rule to stop make trying to rebuild removed or renamed headers
 %.h:
@@ -111,7 +102,7 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
 $(OBJS):
 endif
 
-include $(SRC_PATH)/ffbuild/arch.mak
+include $(SRC_PATH)/arch.mak
 
 OBJS      += $(OBJS-yes)
 SLIBOBJS  += $(SLIBOBJS-yes)
@@ -119,7 +110,7 @@ FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
-FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=EXTRALIBS-%),$($(lib))) $(EXTRALIBS)
+FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
 
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
 SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
@@ -141,10 +132,8 @@ ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
 HOBJS        = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
-PTXOBJS      = $(filter %.ptx.o,$(OBJS))
-$(HOBJS):     CCFLAGS += $(CFLAGS_HEADERS)
 checkheaders: $(HOBJS)
-.SECONDARY:   $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)
+.SECONDARY:   $(HOBJS:.o=.c)
 
 alltools: $(TOOLS)
 
@@ -152,7 +141,7 @@ $(HOSTOBJS): %.o: %.c
 	$(COMPILE_HOSTC)
 
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
@@ -163,7 +152,8 @@ $(TOOLOBJS): | tools
 
 OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~
+CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ver-sol2 *.ho *.gcno *.gcda *$(DEFAULT_YASMD).asm
+DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
 define RULES
@@ -173,4 +163,4 @@ endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)

compat/atomics/dummy/stdatomic.h

@@ -1,176 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * based on vlc_atomic.h from VLC
- * Copyright (C) 2010 Rémi Denis-Courmont
- */
-
-#ifndef COMPAT_ATOMICS_DUMMY_STDATOMIC_H
-#define COMPAT_ATOMICS_DUMMY_STDATOMIC_H
-
-#include <stdint.h>
-
-#define ATOMIC_FLAG_INIT 0
-
-#define ATOMIC_VAR_INIT(value) (value)
-
-#define atomic_init(obj, value) \
-do {                            \
-    *(obj) = (value);           \
-} while(0)
-
-#define kill_dependency(y) ((void)0)
-
-#define atomic_thread_fence(order) \
-    ((void)0)
-
-#define atomic_signal_fence(order) \
-    ((void)0)
-
-#define atomic_is_lock_free(obj) 0
-
-typedef intptr_t atomic_flag;
-typedef intptr_t atomic_bool;
-typedef intptr_t atomic_char;
-typedef intptr_t atomic_schar;
-typedef intptr_t atomic_uchar;
-typedef intptr_t atomic_short;
-typedef intptr_t atomic_ushort;
-typedef intptr_t atomic_int;
-typedef intptr_t atomic_uint;
-typedef intptr_t atomic_long;
-typedef intptr_t atomic_ulong;
-typedef intptr_t atomic_llong;
-typedef intptr_t atomic_ullong;
-typedef intptr_t atomic_wchar_t;
-typedef intptr_t atomic_int_least8_t;
-typedef intptr_t atomic_uint_least8_t;
-typedef intptr_t atomic_int_least16_t;
-typedef intptr_t atomic_uint_least16_t;
-typedef intptr_t atomic_int_least32_t;
-typedef intptr_t atomic_uint_least32_t;
-typedef intptr_t atomic_int_least64_t;
-typedef intptr_t atomic_uint_least64_t;
-typedef intptr_t atomic_int_fast8_t;
-typedef intptr_t atomic_uint_fast8_t;
-typedef intptr_t atomic_int_fast16_t;
-typedef intptr_t atomic_uint_fast16_t;
-typedef intptr_t atomic_int_fast32_t;
-typedef intptr_t atomic_uint_fast32_t;
-typedef intptr_t atomic_int_fast64_t;
-typedef intptr_t atomic_uint_fast64_t;
-typedef intptr_t atomic_intptr_t;
-typedef intptr_t atomic_uintptr_t;
-typedef intptr_t atomic_size_t;
-typedef intptr_t atomic_ptrdiff_t;
-typedef intptr_t atomic_intmax_t;
-typedef intptr_t atomic_uintmax_t;
-
-#define atomic_store(object, desired)   \
-do {                                    \
-    *(object) = (desired);              \
-} while (0)
-
-#define atomic_store_explicit(object, desired, order) \
-    atomic_store(object, desired)
-
-#define atomic_load(object) \
-    (*(object))
-
-#define atomic_load_explicit(object, order) \
-    atomic_load(object)
-
-static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
-{
-    intptr_t ret = *object;
-    *object = desired;
-    return ret;
-}
-
-#define atomic_exchange_explicit(object, desired, order) \
-    atomic_exchange(object, desired)
-
-static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
-                                                 intptr_t desired)
-{
-    int ret;
-    if (*object == *expected) {
-        *object = desired;
-        ret = 1;
-    } else {
-        *expected = *object;
-        ret = 0;
-    }
-    return ret;
-}
-
-#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak(object, expected, desired) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_weak(object, expected, desired)
-
-#define FETCH_MODIFY(opname, op)                                            \
-static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
-{                                                                    \
-    intptr_t ret;                                                    \
-    ret = *object;                                                   \
-    *object = *object op operand;                                    \
-    return ret;                                                      \
-}
-
-FETCH_MODIFY(add, +)
-FETCH_MODIFY(sub, -)
-FETCH_MODIFY(or,  |)
-FETCH_MODIFY(xor, ^)
-FETCH_MODIFY(and, &)
-
-#undef FETCH_MODIFY
-
-#define atomic_fetch_add_explicit(object, operand, order) \
-    atomic_fetch_add(object, operand)
-
-#define atomic_fetch_sub_explicit(object, operand, order) \
-    atomic_fetch_sub(object, operand)
-
-#define atomic_fetch_or_explicit(object, operand, order) \
-    atomic_fetch_or(object, operand)
-
-#define atomic_fetch_xor_explicit(object, operand, order) \
-    atomic_fetch_xor(object, operand)
-
-#define atomic_fetch_and_explicit(object, operand, order) \
-    atomic_fetch_and(object, operand)
-
-#define atomic_flag_test_and_set(object) \
-    atomic_exchange(object, 1)
-
-#define atomic_flag_test_and_set_explicit(object, order) \
-    atomic_flag_test_and_set(object)
-
-#define atomic_flag_clear(object) \
-    atomic_store(object, 0)
-
-#define atomic_flag_clear_explicit(object, order) \
-    atomic_flag_clear(object)
-
-#endif /* COMPAT_ATOMICS_DUMMY_STDATOMIC_H */

compat/atomics/gcc/stdatomic.h

@@ -1,173 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * based on vlc_atomic.h from VLC
- * Copyright (C) 2010 Rémi Denis-Courmont
- */
-
-#ifndef COMPAT_ATOMICS_GCC_STDATOMIC_H
-#define COMPAT_ATOMICS_GCC_STDATOMIC_H
-
-#include <stddef.h>
-#include <stdint.h>
-
-#define ATOMIC_FLAG_INIT 0
-
-#define ATOMIC_VAR_INIT(value) (value)
-
-#define atomic_init(obj, value) \
-do {                            \
-    *(obj) = (value);           \
-} while(0)
-
-#define kill_dependency(y) ((void)0)
-
-#define atomic_thread_fence(order) \
-    __sync_synchronize()
-
-#define atomic_signal_fence(order) \
-    ((void)0)
-
-#define atomic_is_lock_free(obj) 0
-
-typedef         _Bool      atomic_flag;
-typedef         _Bool      atomic_bool;
-typedef          char      atomic_char;
-typedef   signed char      atomic_schar;
-typedef unsigned char      atomic_uchar;
-typedef          short     atomic_short;
-typedef unsigned short     atomic_ushort;
-typedef          int       atomic_int;
-typedef unsigned int       atomic_uint;
-typedef          long      atomic_long;
-typedef unsigned long      atomic_ulong;
-typedef          long long atomic_llong;
-typedef unsigned long long atomic_ullong;
-typedef          wchar_t   atomic_wchar_t;
-typedef       int_least8_t atomic_int_least8_t;
-typedef      uint_least8_t atomic_uint_least8_t;
-typedef      int_least16_t atomic_int_least16_t;
-typedef     uint_least16_t atomic_uint_least16_t;
-typedef      int_least32_t atomic_int_least32_t;
-typedef     uint_least32_t atomic_uint_least32_t;
-typedef      int_least64_t atomic_int_least64_t;
-typedef     uint_least64_t atomic_uint_least64_t;
-typedef       int_fast8_t atomic_int_fast8_t;
-typedef      uint_fast8_t atomic_uint_fast8_t;
-typedef      int_fast16_t atomic_int_fast16_t;
-typedef     uint_fast16_t atomic_uint_fast16_t;
-typedef      int_fast32_t atomic_int_fast32_t;
-typedef     uint_fast32_t atomic_uint_fast32_t;
-typedef      int_fast64_t atomic_int_fast64_t;
-typedef     uint_fast64_t atomic_uint_fast64_t;
-typedef          intptr_t atomic_intptr_t;
-typedef         uintptr_t atomic_uintptr_t;
-typedef            size_t atomic_size_t;
-typedef         ptrdiff_t atomic_ptrdiff_t;
-typedef          intmax_t atomic_intmax_t;
-typedef         uintmax_t atomic_uintmax_t;
-
-#define atomic_store(object, desired)   \
-do {                                    \
-    *(object) = (desired);              \
-    __sync_synchronize();               \
-} while (0)
-
-#define atomic_store_explicit(object, desired, order) \
-    atomic_store(object, desired)
-
-#define atomic_load(object) \
-    (__sync_synchronize(), *(object))
-
-#define atomic_load_explicit(object, order) \
-    atomic_load(object)
-
-#define atomic_exchange(object, desired)                            \
-({                                                                  \
-    __typeof__(object) _obj = (object);                             \
-    __typeof__(*object) _old;                                       \
-    do                                                              \
-        _old = atomic_load(_obj);                                   \
-    while (!__sync_bool_compare_and_swap(_obj, _old, (desired)));   \
-    _old;                                                           \
-})
-
-#define atomic_exchange_explicit(object, desired, order) \
-    atomic_exchange(object, desired)
-
-#define atomic_compare_exchange_strong(object, expected, desired)   \
-({                                                                  \
-    __typeof__(object) _exp = (expected);                           \
-    __typeof__(*object) _old = *_exp;                               \
-    *_exp = __sync_val_compare_and_swap((object), _old, (desired)); \
-    *_exp == _old;                                                  \
-})
-
-#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak(object, expected, desired) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_weak(object, expected, desired)
-
-#define atomic_fetch_add(object, operand) \
-    __sync_fetch_and_add(object, operand)
-
-#define atomic_fetch_add_explicit(object, operand, order) \
-    atomic_fetch_add(object, operand)
-
-#define atomic_fetch_sub(object, operand) \
-    __sync_fetch_and_sub(object, operand)
-
-#define atomic_fetch_sub_explicit(object, operand, order) \
-    atomic_fetch_sub(object, operand)
-
-#define atomic_fetch_or(object, operand) \
-    __sync_fetch_and_or(object, operand)
-
-#define atomic_fetch_or_explicit(object, operand, order) \
-    atomic_fetch_or(object, operand)
-
-#define atomic_fetch_xor(object, operand) \
-    __sync_fetch_and_xor(object, operand)
-
-#define atomic_fetch_xor_explicit(object, operand, order) \
-    atomic_fetch_xor(object, operand)
-
-#define atomic_fetch_and(object, operand) \
-    __sync_fetch_and_and(object, operand)
-
-#define atomic_fetch_and_explicit(object, operand, order) \
-    atomic_fetch_and(object, operand)
-
-#define atomic_flag_test_and_set(object) \
-    atomic_exchange(object, 1)
-
-#define atomic_flag_test_and_set_explicit(object, order) \
-    atomic_flag_test_and_set(object)
-
-#define atomic_flag_clear(object) \
-    atomic_store(object, 0)
-
-#define atomic_flag_clear_explicit(object, order) \
-    atomic_flag_clear(object)
-
-#endif /* COMPAT_ATOMICS_GCC_STDATOMIC_H */

compat/atomics/pthread/stdatomic.c

@@ -1,39 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * based on vlc_atomic.h from VLC
- * Copyright (C) 2010 Rémi Denis-Courmont
- */
-
-#include <pthread.h>
-#include <stdint.h>
-
-#include "stdatomic.h"
-
-static pthread_mutex_t atomic_lock = PTHREAD_MUTEX_INITIALIZER;
-
-void avpriv_atomic_lock(void)
-{
-    pthread_mutex_lock(&atomic_lock);
-}
-
-void avpriv_atomic_unlock(void)
-{
-    pthread_mutex_unlock(&atomic_lock);
-}

compat/atomics/pthread/stdatomic.h

@@ -1,197 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * based on vlc_atomic.h from VLC
- * Copyright (C) 2010 Rémi Denis-Courmont
- */
-
-#ifndef COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
-#define COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
-
-#include <stdint.h>
-
-#define ATOMIC_FLAG_INIT 0
-
-#define ATOMIC_VAR_INIT(value) (value)
-
-#define atomic_init(obj, value) \
-do {                            \
-    *(obj) = (value);           \
-} while(0)
-
-#define kill_dependency(y) ((void)0)
-
-#define atomic_signal_fence(order) \
-    ((void)0)
-
-#define atomic_is_lock_free(obj) 0
-
-typedef intptr_t atomic_flag;
-typedef intptr_t atomic_bool;
-typedef intptr_t atomic_char;
-typedef intptr_t atomic_schar;
-typedef intptr_t atomic_uchar;
-typedef intptr_t atomic_short;
-typedef intptr_t atomic_ushort;
-typedef intptr_t atomic_int;
-typedef intptr_t atomic_uint;
-typedef intptr_t atomic_long;
-typedef intptr_t atomic_ulong;
-typedef intptr_t atomic_llong;
-typedef intptr_t atomic_ullong;
-typedef intptr_t atomic_wchar_t;
-typedef intptr_t atomic_int_least8_t;
-typedef intptr_t atomic_uint_least8_t;
-typedef intptr_t atomic_int_least16_t;
-typedef intptr_t atomic_uint_least16_t;
-typedef intptr_t atomic_int_least32_t;
-typedef intptr_t atomic_uint_least32_t;
-typedef intptr_t atomic_int_least64_t;
-typedef intptr_t atomic_uint_least64_t;
-typedef intptr_t atomic_int_fast8_t;
-typedef intptr_t atomic_uint_fast8_t;
-typedef intptr_t atomic_int_fast16_t;
-typedef intptr_t atomic_uint_fast16_t;
-typedef intptr_t atomic_int_fast32_t;
-typedef intptr_t atomic_uint_fast32_t;
-typedef intptr_t atomic_int_fast64_t;
-typedef intptr_t atomic_uint_fast64_t;
-typedef intptr_t atomic_intptr_t;
-typedef intptr_t atomic_uintptr_t;
-typedef intptr_t atomic_size_t;
-typedef intptr_t atomic_ptrdiff_t;
-typedef intptr_t atomic_intmax_t;
-typedef intptr_t atomic_uintmax_t;
-
-void avpriv_atomic_lock(void);
-void avpriv_atomic_unlock(void);
-
-static inline void atomic_thread_fence(int order)
-{
-    avpriv_atomic_lock();
-    avpriv_atomic_unlock();
-}
-
-static inline void atomic_store(intptr_t *object, intptr_t desired)
-{
-    avpriv_atomic_lock();
-    *object = desired;
-    avpriv_atomic_unlock();
-}
-
-#define atomic_store_explicit(object, desired, order) \
-    atomic_store(object, desired)
-
-static inline intptr_t atomic_load(intptr_t *object)
-{
-    intptr_t ret;
-    avpriv_atomic_lock();
-    ret = *object;
-    avpriv_atomic_unlock();
-    return ret;
-}
-
-#define atomic_load_explicit(object, order) \
-    atomic_load(object)
-
-static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
-{
-    intptr_t ret;
-    avpriv_atomic_lock();
-    ret     = *object;
-    *object = desired;
-    avpriv_atomic_unlock();
-    return ret;
-}
-
-#define atomic_exchange_explicit(object, desired, order) \
-    atomic_exchange(object, desired)
-
-static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
-                                                 intptr_t desired)
-{
-    int ret;
-    avpriv_atomic_lock();
-    if (*object == *expected) {
-        ret     = 1;
-        *object = desired;
-    } else {
-        ret = 0;
-        *expected = *object;
-    }
-    avpriv_atomic_unlock();
-    return ret;
-}
-
-#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak(object, expected, desired) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_weak(object, expected, desired)
-
-#define FETCH_MODIFY(opname, op)                                                   \
-static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
-{                                                                                  \
-    intptr_t ret;                                                                  \
-    avpriv_atomic_lock();                                                          \
-    ret = *object;                                                                 \
-    *object = *object op operand;                                                  \
-    avpriv_atomic_unlock();                                                        \
-    return ret;                                                                    \
-}
-
-FETCH_MODIFY(add, +)
-FETCH_MODIFY(sub, -)
-FETCH_MODIFY(or,  |)
-FETCH_MODIFY(xor, ^)
-FETCH_MODIFY(and, &)
-
-#undef FETCH_MODIFY
-
-#define atomic_fetch_add_explicit(object, operand, order) \
-    atomic_fetch_add(object, operand)
-
-#define atomic_fetch_sub_explicit(object, operand, order) \
-    atomic_fetch_sub(object, operand)
-
-#define atomic_fetch_or_explicit(object, operand, order) \
-    atomic_fetch_or(object, operand)
-
-#define atomic_fetch_xor_explicit(object, operand, order) \
-    atomic_fetch_xor(object, operand)
-
-#define atomic_fetch_and_explicit(object, operand, order) \
-    atomic_fetch_and(object, operand)
-
-#define atomic_flag_test_and_set(object) \
-    atomic_exchange(object, 1)
-
-#define atomic_flag_test_and_set_explicit(object, order) \
-    atomic_flag_test_and_set(object)
-
-#define atomic_flag_clear(object) \
-    atomic_store(object, 0)
-
-#define atomic_flag_clear_explicit(object, order) \
-    atomic_flag_clear(object)
-
-#endif /* COMPAT_ATOMICS_PTHREAD_STDATOMIC_H */

compat/atomics/suncc/stdatomic.h

@@ -1,186 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef COMPAT_ATOMICS_SUNCC_STDATOMIC_H
-#define COMPAT_ATOMICS_SUNCC_STDATOMIC_H
-
-#include <atomic.h>
-#include <mbarrier.h>
-#include <stddef.h>
-#include <stdint.h>
-
-#define ATOMIC_FLAG_INIT 0
-
-#define ATOMIC_VAR_INIT(value) (value)
-
-#define atomic_init(obj, value) \
-do {                            \
-    *(obj) = (value);           \
-} while(0)
-
-#define kill_dependency(y) ((void)0)
-
-#define atomic_thread_fence(order) \
-    __machine_rw_barrier();
-
-#define atomic_signal_fence(order) \
-    ((void)0)
-
-#define atomic_is_lock_free(obj) 0
-
-typedef intptr_t atomic_flag;
-typedef intptr_t atomic_bool;
-typedef intptr_t atomic_char;
-typedef intptr_t atomic_schar;
-typedef intptr_t atomic_uchar;
-typedef intptr_t atomic_short;
-typedef intptr_t atomic_ushort;
-typedef intptr_t atomic_int;
-typedef intptr_t atomic_uint;
-typedef intptr_t atomic_long;
-typedef intptr_t atomic_ulong;
-typedef intptr_t atomic_llong;
-typedef intptr_t atomic_ullong;
-typedef intptr_t atomic_wchar_t;
-typedef intptr_t atomic_int_least8_t;
-typedef intptr_t atomic_uint_least8_t;
-typedef intptr_t atomic_int_least16_t;
-typedef intptr_t atomic_uint_least16_t;
-typedef intptr_t atomic_int_least32_t;
-typedef intptr_t atomic_uint_least32_t;
-typedef intptr_t atomic_int_least64_t;
-typedef intptr_t atomic_uint_least64_t;
-typedef intptr_t atomic_int_fast8_t;
-typedef intptr_t atomic_uint_fast8_t;
-typedef intptr_t atomic_int_fast16_t;
-typedef intptr_t atomic_uint_fast16_t;
-typedef intptr_t atomic_int_fast32_t;
-typedef intptr_t atomic_uint_fast32_t;
-typedef intptr_t atomic_int_fast64_t;
-typedef intptr_t atomic_uint_fast64_t;
-typedef intptr_t atomic_intptr_t;
-typedef intptr_t atomic_uintptr_t;
-typedef intptr_t atomic_size_t;
-typedef intptr_t atomic_ptrdiff_t;
-typedef intptr_t atomic_intmax_t;
-typedef intptr_t atomic_uintmax_t;
-
-static inline void atomic_store(intptr_t *object, intptr_t desired)
-{
-    *object = desired;
-    __machine_rw_barrier();
-}
-
-#define atomic_store_explicit(object, desired, order) \
-    atomic_store(object, desired)
-
-static inline intptr_t atomic_load(intptr_t *object)
-{
-    __machine_rw_barrier();
-    return *object;
-}
-
-#define atomic_load_explicit(object, order) \
-    atomic_load(object)
-
-#define atomic_exchange(object, desired) \
-    atomic_swap_ptr(object, desired)
-
-#define atomic_exchange_explicit(object, desired, order) \
-    atomic_exchange(object, desired)
-
-static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
-                                                 intptr_t desired)
-{
-    intptr_t  old = *expected;
-    *expected = (intptr_t)atomic_cas_ptr(object, (void *)old, (void *)desired);
-    return *expected == old;
-}
-
-#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak(object, expected, desired) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_weak(object, expected, desired)
-
-static inline intptr_t atomic_fetch_add(intptr_t *object, intptr_t operand)
-{
-    return atomic_add_ptr_nv(object, operand) - operand;
-}
-
-#define atomic_fetch_sub(object, operand) \
-    atomic_fetch_add(object, -(operand))
-
-static inline intptr_t atomic_fetch_or(intptr_t *object, intptr_t operand)
-{
-    intptr_t old;
-    do {
-        old = atomic_load(object);
-    } while (!atomic_compare_exchange_strong(object, old, old | operand));
-    return old;
-}
-
-static inline intptr_t atomic_fetch_xor(intptr_t *object, intptr_t operand)
-{
-    intptr_t old;
-    do {
-        old = atomic_load(object);
-    } while (!atomic_compare_exchange_strong(object, old, old ^ operand));
-    return old;
-}
-
-static inline intptr_t atomic_fetch_and(intptr_t *object, intptr_t operand)
-{
-    intptr_t old;
-    do {
-        old = atomic_load(object);
-    } while (!atomic_compare_exchange_strong(object, old, old & operand));
-    return old;
-}
-
-#define atomic_fetch_add_explicit(object, operand, order) \
-    atomic_fetch_add(object, operand)
-
-#define atomic_fetch_sub_explicit(object, operand, order) \
-    atomic_fetch_sub(object, operand)
-
-#define atomic_fetch_or_explicit(object, operand, order) \
-    atomic_fetch_or(object, operand)
-
-#define atomic_fetch_xor_explicit(object, operand, order) \
-    atomic_fetch_xor(object, operand)
-
-#define atomic_fetch_and_explicit(object, operand, order) \
-    atomic_fetch_and(object, operand)
-
-#define atomic_flag_test_and_set(object) \
-    atomic_exchange(object, 1)
-
-#define atomic_flag_test_and_set_explicit(object, order) \
-    atomic_flag_test_and_set(object)
-
-#define atomic_flag_clear(object) \
-    atomic_store(object, 0)
-
-#define atomic_flag_clear_explicit(object, order) \
-    atomic_flag_clear(object)
-
-#endif /* COMPAT_ATOMICS_SUNCC_STDATOMIC_H */

compat/atomics/win32/stdatomic.h

@@ -1,181 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef COMPAT_ATOMICS_WIN32_STDATOMIC_H
-#define COMPAT_ATOMICS_WIN32_STDATOMIC_H
-
-#define WIN32_LEAN_AND_MEAN
-#include <stddef.h>
-#include <stdint.h>
-#include <windows.h>
-
-#define ATOMIC_FLAG_INIT 0
-
-#define ATOMIC_VAR_INIT(value) (value)
-
-#define atomic_init(obj, value) \
-do {                            \
-    *(obj) = (value);           \
-} while(0)
-
-#define kill_dependency(y) ((void)0)
-
-#define atomic_thread_fence(order) \
-    MemoryBarrier();
-
-#define atomic_signal_fence(order) \
-    ((void)0)
-
-#define atomic_is_lock_free(obj) 0
-
-typedef intptr_t atomic_flag;
-typedef intptr_t atomic_bool;
-typedef intptr_t atomic_char;
-typedef intptr_t atomic_schar;
-typedef intptr_t atomic_uchar;
-typedef intptr_t atomic_short;
-typedef intptr_t atomic_ushort;
-typedef intptr_t atomic_int;
-typedef intptr_t atomic_uint;
-typedef intptr_t atomic_long;
-typedef intptr_t atomic_ulong;
-typedef intptr_t atomic_llong;
-typedef intptr_t atomic_ullong;
-typedef intptr_t atomic_wchar_t;
-typedef intptr_t atomic_int_least8_t;
-typedef intptr_t atomic_uint_least8_t;
-typedef intptr_t atomic_int_least16_t;
-typedef intptr_t atomic_uint_least16_t;
-typedef intptr_t atomic_int_least32_t;
-typedef intptr_t atomic_uint_least32_t;
-typedef intptr_t atomic_int_least64_t;
-typedef intptr_t atomic_uint_least64_t;
-typedef intptr_t atomic_int_fast8_t;
-typedef intptr_t atomic_uint_fast8_t;
-typedef intptr_t atomic_int_fast16_t;
-typedef intptr_t atomic_uint_fast16_t;
-typedef intptr_t atomic_int_fast32_t;
-typedef intptr_t atomic_uint_fast32_t;
-typedef intptr_t atomic_int_fast64_t;
-typedef intptr_t atomic_uint_fast64_t;
-typedef intptr_t atomic_intptr_t;
-typedef intptr_t atomic_uintptr_t;
-typedef intptr_t atomic_size_t;
-typedef intptr_t atomic_ptrdiff_t;
-typedef intptr_t atomic_intmax_t;
-typedef intptr_t atomic_uintmax_t;
-
-#define atomic_store(object, desired)   \
-do {                                    \
-    *(object) = (desired);              \
-    MemoryBarrier();                    \
-} while (0)
-
-#define atomic_store_explicit(object, desired, order) \
-    atomic_store(object, desired)
-
-#define atomic_load(object) \
-    (MemoryBarrier(), *(object))
-
-#define atomic_load_explicit(object, order) \
-    atomic_load(object)
-
-#define atomic_exchange(object, desired) \
-    InterlockedExchangePointer(object, desired);
-
-#define atomic_exchange_explicit(object, desired, order) \
-    atomic_exchange(object, desired)
-
-static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
-                                                 intptr_t desired)
-{
-    intptr_t old = *expected;
-    *expected = (intptr_t)InterlockedCompareExchangePointer(
-        (PVOID *)object, (PVOID)desired, (PVOID)old);
-    return *expected == old;
-}
-
-#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak(object, expected, desired) \
-    atomic_compare_exchange_strong(object, expected, desired)
-
-#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
-    atomic_compare_exchange_weak(object, expected, desired)
-
-#ifdef _WIN64
-#define atomic_fetch_add(object, operand) \
-    InterlockedExchangeAdd64(object, operand)
-
-#define atomic_fetch_sub(object, operand) \
-    InterlockedExchangeAdd64(object, -(operand))
-
-#define atomic_fetch_or(object, operand) \
-    InterlockedOr64(object, operand)
-
-#define atomic_fetch_xor(object, operand) \
-    InterlockedXor64(object, operand)
-
-#define atomic_fetch_and(object, operand) \
-    InterlockedAnd64(object, operand)
-#else
-#define atomic_fetch_add(object, operand) \
-    InterlockedExchangeAdd(object, operand)
-
-#define atomic_fetch_sub(object, operand) \
-    InterlockedExchangeAdd(object, -(operand))
-
-#define atomic_fetch_or(object, operand) \
-    InterlockedOr(object, operand)
-
-#define atomic_fetch_xor(object, operand) \
-    InterlockedXor(object, operand)
-
-#define atomic_fetch_and(object, operand) \
-    InterlockedAnd(object, operand)
-#endif /* _WIN64 */
-
-#define atomic_fetch_add_explicit(object, operand, order) \
-    atomic_fetch_add(object, operand)
-
-#define atomic_fetch_sub_explicit(object, operand, order) \
-    atomic_fetch_sub(object, operand)
-
-#define atomic_fetch_or_explicit(object, operand, order) \
-    atomic_fetch_or(object, operand)
-
-#define atomic_fetch_xor_explicit(object, operand, order) \
-    atomic_fetch_xor(object, operand)
-
-#define atomic_fetch_and_explicit(object, operand, order) \
-    atomic_fetch_and(object, operand)
-
-#define atomic_flag_test_and_set(object) \
-    atomic_exchange(object, 1)
-
-#define atomic_flag_test_and_set_explicit(object, order) \
-    atomic_flag_test_and_set(object)
-
-#define atomic_flag_clear(object) \
-    atomic_store(object, 0)
-
-#define atomic_flag_clear_explicit(object, order) \
-    atomic_flag_clear(object)
-
-#endif /* COMPAT_ATOMICS_WIN32_STDATOMIC_H */

compat/avisynth/avisynth_c.h

@@ -75,149 +75,54 @@ enum {AVS_PLANAR_Y=1<<0,
       AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
 
   // Colorspace properties.
-enum {
-    AVS_CS_YUVA = 1 << 27,
-    AVS_CS_BGR = 1 << 28,
-    AVS_CS_YUV = 1 << 29,
-    AVS_CS_INTERLEAVED = 1 << 30,
-    AVS_CS_PLANAR = 1 << 31,
+enum {AVS_CS_BGR = 1<<28,
+      AVS_CS_YUV = 1<<29,
+      AVS_CS_INTERLEAVED = 1<<30,
+      AVS_CS_PLANAR = 1<<31,
 
-    AVS_CS_SHIFT_SUB_WIDTH = 0,
-    AVS_CS_SHIFT_SUB_HEIGHT = 8,
-    AVS_CS_SHIFT_SAMPLE_BITS = 16,
+      AVS_CS_SHIFT_SUB_WIDTH   = 0,
+      AVS_CS_SHIFT_SUB_HEIGHT  = 8,
+      AVS_CS_SHIFT_SAMPLE_BITS = 16,
 
-    AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
-    AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
-    AVS_CS_SUB_WIDTH_2 = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
-    AVS_CS_SUB_WIDTH_4 = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
+      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
+      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
+      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
+      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
 
-    AVS_CS_VPLANEFIRST = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
-    AVS_CS_UPLANEFIRST = 1 << 4, // I420
+      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
+      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
 
-    AVS_CS_SUB_HEIGHT_MASK = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
-    AVS_CS_SUB_HEIGHT_1 = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
-    AVS_CS_SUB_HEIGHT_2 = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
-    AVS_CS_SUB_HEIGHT_4 = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
+      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
+      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
+      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
+      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
 
-    AVS_CS_SAMPLE_BITS_MASK = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
-    AVS_CS_SAMPLE_BITS_8 = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
-    AVS_CS_SAMPLE_BITS_10 = 5 << AVS_CS_SHIFT_SAMPLE_BITS,
-    AVS_CS_SAMPLE_BITS_12 = 6 << AVS_CS_SHIFT_SAMPLE_BITS,
-    AVS_CS_SAMPLE_BITS_14 = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
-    AVS_CS_SAMPLE_BITS_16 = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
-    AVS_CS_SAMPLE_BITS_32 = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
-
-    AVS_CS_PLANAR_MASK = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_YUVA | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
-    AVS_CS_PLANAR_FILTER = ~(AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST),
-
-    AVS_CS_RGB_TYPE  = 1 << 0,
-    AVS_CS_RGBA_TYPE = 1 << 1,
-
-    AVS_CS_GENERIC_YUV420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // 4:2:0 planar
-    AVS_CS_GENERIC_YUV422  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // 4:2:2 planar
-    AVS_CS_GENERIC_YUV444  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // 4:4:4 planar
-    AVS_CS_GENERIC_Y       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV,                                             // Y only (4:0:0)
-    AVS_CS_GENERIC_RGBP    = AVS_CS_PLANAR | AVS_CS_BGR | AVS_CS_RGB_TYPE,                                                // planar RGB
-    AVS_CS_GENERIC_RGBAP   = AVS_CS_PLANAR | AVS_CS_BGR | AVS_CS_RGBA_TYPE,                                               // planar RGBA
-    AVS_CS_GENERIC_YUVA420 = AVS_CS_PLANAR | AVS_CS_YUVA | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // 4:2:0:A planar
-    AVS_CS_GENERIC_YUVA422 = AVS_CS_PLANAR | AVS_CS_YUVA | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // 4:2:2:A planar
-    AVS_CS_GENERIC_YUVA444 = AVS_CS_PLANAR | AVS_CS_YUVA | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1 }; // 4:4:4:A planar
+      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
 
+      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
+      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
 
   // Specific colorformats
 enum {
   AVS_CS_UNKNOWN = 0,
-  AVS_CS_BGR24 = AVS_CS_RGB_TYPE  | AVS_CS_BGR | AVS_CS_INTERLEAVED,
-  AVS_CS_BGR32 = AVS_CS_RGBA_TYPE | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
   AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
   //  AVS_CS_YV12  = 1<<3  Reserved
   //  AVS_CS_I420  = 1<<4  Reserved
   AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
 
-  AVS_CS_YV24  = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_8,  // YVU 4:4:4 planar
-  AVS_CS_YV16  = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_8,  // YVU 4:2:2 planar
-  AVS_CS_YV12  = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_8,  // YVU 4:2:0 planar
+  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
+  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
+  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:0 planar
   AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
   AVS_CS_IYUV  = AVS_CS_I420,
   AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
   AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
-  AVS_CS_Y8    = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_8,       // Y   4:0:0 planar
-
-  //-------------------------
-  // AVS16: new planar constants go live! Experimental PF 160613
-  // 10-12-14 bit + planar RGB + BRG48/64 160725
-  AVS_CS_YUV444P10 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_10, // YUV 4:4:4 10bit samples
-  AVS_CS_YUV422P10 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_10, // YUV 4:2:2 10bit samples
-  AVS_CS_YUV420P10 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_10, // YUV 4:2:0 10bit samples
-  AVS_CS_Y10       = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_10,      // Y   4:0:0 10bit samples
-
-  AVS_CS_YUV444P12 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_12, // YUV 4:4:4 12bit samples
-  AVS_CS_YUV422P12 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_12, // YUV 4:2:2 12bit samples
-  AVS_CS_YUV420P12 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_12, // YUV 4:2:0 12bit samples
-  AVS_CS_Y12       = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_12,      // Y   4:0:0 12bit samples
-
-  AVS_CS_YUV444P14 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_14, // YUV 4:4:4 14bit samples
-  AVS_CS_YUV422P14 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_14, // YUV 4:2:2 14bit samples
-  AVS_CS_YUV420P14 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_14, // YUV 4:2:0 14bit samples
-  AVS_CS_Y14       = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_14,      // Y   4:0:0 14bit samples
-
-  AVS_CS_YUV444P16 = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_16, // YUV 4:4:4 16bit samples
-  AVS_CS_YUV422P16 = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_16, // YUV 4:2:2 16bit samples
-  AVS_CS_YUV420P16 = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_16, // YUV 4:2:0 16bit samples
-  AVS_CS_Y16       = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_16,      // Y   4:0:0 16bit samples
-
-  // 32 bit samples (float)
-  AVS_CS_YUV444PS = AVS_CS_GENERIC_YUV444 | AVS_CS_SAMPLE_BITS_32,  // YUV 4:4:4 32bit samples
-  AVS_CS_YUV422PS = AVS_CS_GENERIC_YUV422 | AVS_CS_SAMPLE_BITS_32,  // YUV 4:2:2 32bit samples
-  AVS_CS_YUV420PS = AVS_CS_GENERIC_YUV420 | AVS_CS_SAMPLE_BITS_32,  // YUV 4:2:0 32bit samples
-  AVS_CS_Y32      = AVS_CS_GENERIC_Y | AVS_CS_SAMPLE_BITS_32,       // Y   4:0:0 32bit samples
-
-  // RGB packed
-  AVS_CS_BGR48 = AVS_CS_RGB_TYPE | AVS_CS_BGR | AVS_CS_INTERLEAVED | AVS_CS_SAMPLE_BITS_16,    // BGR 3x16 bit
-  AVS_CS_BGR64 = AVS_CS_RGBA_TYPE | AVS_CS_BGR | AVS_CS_INTERLEAVED | AVS_CS_SAMPLE_BITS_16,    // BGR 4x16 bit
-  // no packed 32 bit (float) support for these legacy types
-
-  // RGB planar
-  AVS_CS_RGBP   = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_8,  // Planar RGB 8 bit samples
-  AVS_CS_RGBP10 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_10, // Planar RGB 10bit samples
-  AVS_CS_RGBP12 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_12, // Planar RGB 12bit samples
-  AVS_CS_RGBP14 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_14, // Planar RGB 14bit samples
-  AVS_CS_RGBP16 = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_16, // Planar RGB 16bit samples
-  AVS_CS_RGBPS  = AVS_CS_GENERIC_RGBP | AVS_CS_SAMPLE_BITS_32, // Planar RGB 32bit samples
-
-  // RGBA planar
-  AVS_CS_RGBAP   = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_8,  // Planar RGBA 8 bit samples
-  AVS_CS_RGBAP10 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_10, // Planar RGBA 10bit samples
-  AVS_CS_RGBAP12 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_12, // Planar RGBA 12bit samples
-  AVS_CS_RGBAP14 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_14, // Planar RGBA 14bit samples
-  AVS_CS_RGBAP16 = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_16, // Planar RGBA 16bit samples
-  AVS_CS_RGBAPS  = AVS_CS_GENERIC_RGBAP | AVS_CS_SAMPLE_BITS_32, // Planar RGBA 32bit samples
-
-  // Planar YUVA
-  AVS_CS_YUVA444    = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_8,  // YUVA 4:4:4 8bit samples
-  AVS_CS_YUVA422    = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_8,  // YUVA 4:2:2 8bit samples
-  AVS_CS_YUVA420    = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_8,  // YUVA 4:2:0 8bit samples
-
-  AVS_CS_YUVA444P10 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_10, // YUVA 4:4:4 10bit samples
-  AVS_CS_YUVA422P10 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_10, // YUVA 4:2:2 10bit samples
-  AVS_CS_YUVA420P10 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_10, // YUVA 4:2:0 10bit samples
-
-  AVS_CS_YUVA444P12 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_12, // YUVA 4:4:4 12bit samples
-  AVS_CS_YUVA422P12 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_12, // YUVA 4:2:2 12bit samples
-  AVS_CS_YUVA420P12 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_12, // YUVA 4:2:0 12bit samples
-
-  AVS_CS_YUVA444P14 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_14, // YUVA 4:4:4 14bit samples
-  AVS_CS_YUVA422P14 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_14, // YUVA 4:2:2 14bit samples
-  AVS_CS_YUVA420P14 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_14, // YUVA 4:2:0 14bit samples
-
-  AVS_CS_YUVA444P16 = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_16, // YUVA 4:4:4 16bit samples
-  AVS_CS_YUVA422P16 = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_16, // YUVA 4:2:2 16bit samples
-  AVS_CS_YUVA420P16 = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_16, // YUVA 4:2:0 16bit samples
-
-  AVS_CS_YUVA444PS  = AVS_CS_GENERIC_YUVA444 | AVS_CS_SAMPLE_BITS_32,  // YUVA 4:4:4 32bit samples
-  AVS_CS_YUVA422PS  = AVS_CS_GENERIC_YUVA422 | AVS_CS_SAMPLE_BITS_32,  // YUVA 4:2:2 32bit samples
-  AVS_CS_YUVA420PS  = AVS_CS_GENERIC_YUVA420 | AVS_CS_SAMPLE_BITS_32,  // YUVA 4:2:0 32bit samples
-
+  AVS_CS_Y8    = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
 };
 
 enum {
@@ -342,10 +247,10 @@ AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
         { return !!(p->pixel_type&AVS_CS_BGR); }
 
 AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
-        { return ((p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24) && ((p->pixel_type & AVS_CS_SAMPLE_BITS_MASK) == AVS_CS_SAMPLE_BITS_8); }
+        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
 
 AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
-       { return ((p->pixel_type&AVS_CS_BGR32)==AVS_CS_BGR32) && ((p->pixel_type & AVS_CS_SAMPLE_BITS_MASK) == AVS_CS_SAMPLE_BITS_8); }
+        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
 
 AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
         { return !!(p->pixel_type&AVS_CS_YUV ); }
@@ -353,10 +258,6 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
         { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
 
-AVSC_API(int, avs_is_rgb48)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_rgb64)(const AVS_VideoInfo * p);
-
 AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
 
 AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
@@ -367,38 +268,6 @@ AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
 
 AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
 
-AVSC_API(int, avs_is_yuv444p16)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuv422p16)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuv420p16)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_y16)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuv444ps)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuv422ps)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuv420ps)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_y32)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_444)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_422)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_420)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_y)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_yuva)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_planar_rgb)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_is_planar_rgba)(const AVS_VideoInfo * p);
-
-
-
 AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
         { return ((p->image_type & property)==property ); }
 
@@ -496,12 +365,6 @@ AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
 }
 #endif
 
-AVSC_API(int, avs_num_components)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_component_size)(const AVS_VideoInfo * p);
-
-AVSC_API(int, avs_bits_per_component)(const AVS_VideoInfo * p);
-
 /////////////////////////////////////////////////////////////////////
 //
 // AVS_VideoFrame
@@ -665,7 +528,7 @@ AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
         { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
 #endif
 AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = (short)size; return v; }
+        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
 
 /////////////////////////////////////////////////////////////////////
 //
@@ -898,28 +761,11 @@ struct AVS_Library {
   AVSC_DECLARE_FUNC(avs_vsprintf);
 
   AVSC_DECLARE_FUNC(avs_get_error);
-  AVSC_DECLARE_FUNC(avs_is_rgb48);
-  AVSC_DECLARE_FUNC(avs_is_rgb64);
   AVSC_DECLARE_FUNC(avs_is_yv24);
   AVSC_DECLARE_FUNC(avs_is_yv16);
   AVSC_DECLARE_FUNC(avs_is_yv12);
   AVSC_DECLARE_FUNC(avs_is_yv411);
   AVSC_DECLARE_FUNC(avs_is_y8);
-  AVSC_DECLARE_FUNC(avs_is_yuv444p16);
-  AVSC_DECLARE_FUNC(avs_is_yuv422p16);
-  AVSC_DECLARE_FUNC(avs_is_yuv420p16);
-  AVSC_DECLARE_FUNC(avs_is_y16);
-  AVSC_DECLARE_FUNC(avs_is_yuv444ps);
-  AVSC_DECLARE_FUNC(avs_is_yuv422ps);
-  AVSC_DECLARE_FUNC(avs_is_yuv420ps);
-  AVSC_DECLARE_FUNC(avs_is_y32);
-  AVSC_DECLARE_FUNC(avs_is_444);
-  AVSC_DECLARE_FUNC(avs_is_422);
-  AVSC_DECLARE_FUNC(avs_is_420);
-  AVSC_DECLARE_FUNC(avs_is_y);
-  AVSC_DECLARE_FUNC(avs_is_yuva);
-  AVSC_DECLARE_FUNC(avs_is_planar_rgb);
-  AVSC_DECLARE_FUNC(avs_is_planar_rgba);
   AVSC_DECLARE_FUNC(avs_is_color_space);
 
   AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
@@ -934,11 +780,6 @@ struct AVS_Library {
   AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
   AVSC_DECLARE_FUNC(avs_is_writable);
   AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
-
-  AVSC_DECLARE_FUNC(avs_num_components);
-  AVSC_DECLARE_FUNC(avs_component_size);
-  AVSC_DECLARE_FUNC(avs_bits_per_component);
-
 };
 
 #undef AVSC_DECLARE_FUNC
@@ -999,28 +840,11 @@ AVSC_INLINE AVS_Library * avs_load_library() {
   AVSC_LOAD_FUNC(avs_vsprintf);
 
   AVSC_LOAD_FUNC(avs_get_error);
-  AVSC_LOAD_FUNC(avs_is_rgb48);
-  AVSC_LOAD_FUNC(avs_is_rgb64);
   AVSC_LOAD_FUNC(avs_is_yv24);
   AVSC_LOAD_FUNC(avs_is_yv16);
   AVSC_LOAD_FUNC(avs_is_yv12);
   AVSC_LOAD_FUNC(avs_is_yv411);
   AVSC_LOAD_FUNC(avs_is_y8);
-  AVSC_LOAD_FUNC(avs_is_yuv444p16);
-  AVSC_LOAD_FUNC(avs_is_yuv422p16);
-  AVSC_LOAD_FUNC(avs_is_yuv420p16);
-  AVSC_LOAD_FUNC(avs_is_y16);
-  AVSC_LOAD_FUNC(avs_is_yuv444ps);
-  AVSC_LOAD_FUNC(avs_is_yuv422ps);
-  AVSC_LOAD_FUNC(avs_is_yuv420ps);
-  AVSC_LOAD_FUNC(avs_is_y32);
-  AVSC_LOAD_FUNC(avs_is_444);
-  AVSC_LOAD_FUNC(avs_is_422);
-  AVSC_LOAD_FUNC(avs_is_420);
-  AVSC_LOAD_FUNC(avs_is_y);
-  AVSC_LOAD_FUNC(avs_is_yuva);
-  AVSC_LOAD_FUNC(avs_is_planar_rgb);
-  AVSC_LOAD_FUNC(avs_is_planar_rgba);
   AVSC_LOAD_FUNC(avs_is_color_space);
 
   AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
@@ -1036,12 +860,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
   AVSC_LOAD_FUNC(avs_is_writable);
   AVSC_LOAD_FUNC(avs_get_write_ptr_p);
 
-  AVSC_LOAD_FUNC(avs_num_components);
-  AVSC_LOAD_FUNC(avs_component_size);
-  AVSC_LOAD_FUNC(avs_bits_per_component);
-
-
-
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC

compat/cuda/dynlink_loader.h

@@ -1,33 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AV_COMPAT_CUDA_DYNLINK_LOADER_H
-#define AV_COMPAT_CUDA_DYNLINK_LOADER_H
-
-#include "libavutil/log.h"
-#include "compat/w32dlfcn.h"
-
-#define FFNV_LOAD_FUNC(path) dlopen((path), RTLD_LAZY)
-#define FFNV_SYM_FUNC(lib, sym) dlsym((lib), (sym))
-#define FFNV_FREE_FUNC(lib) dlclose(lib)
-#define FFNV_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_ERROR, msg,  __VA_ARGS__)
-#define FFNV_DEBUG_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_DEBUG, msg,  __VA_ARGS__)
-
-#include <ffnvcodec/dynlink_loader.h>
-
-#endif

compat/cuda/ptx2c.sh

@@ -1,36 +0,0 @@
-#!/bin/sh
-
-# Copyright (c) 2017, NVIDIA CORPORATION. All rights reserved.
-#
-# 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.
-
-set -e
-
-OUT="$1"
-IN="$2"
-NAME="$(basename "$IN" | sed 's/\..*//')"
-
-printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
-while read LINE
-do
-    printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
-done < "$IN"
-printf ";\n" >> "$OUT"
-
-exit 0

compat/os2threads.h

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2011-2017 KO Myung-Hun <komh@chollian.net>
+ * Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
  *
  * This file is part of FFmpeg.
  *
@@ -46,11 +46,9 @@ typedef struct {
 
 typedef void pthread_attr_t;
 
-typedef _fmutex pthread_mutex_t;
+typedef HMTX pthread_mutex_t;
 typedef void pthread_mutexattr_t;
 
-#define PTHREAD_MUTEX_INITIALIZER _FMUTEX_INITIALIZER
-
 typedef struct {
     HEV event_sem;
     HEV ack_sem;
@@ -100,28 +98,28 @@ static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
 static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
                                                const pthread_mutexattr_t *attr)
 {
-    _fmutex_create(mutex, 0);
+    DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
 
     return 0;
 }
 
 static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
 {
-    _fmutex_close(mutex);
+    DosCloseMutexSem(*(PHMTX)mutex);
 
     return 0;
 }
 
 static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
 {
-    _fmutex_request(mutex, 0);
+    DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
 
     return 0;
 }
 
 static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
 {
-    _fmutex_release(mutex);
+    DosReleaseMutexSem(*(PHMTX)mutex);
 
     return 0;
 }

compat/plan9/head

@@ -0,0 +1,10 @@
+#!/bin/sh
+
+n=10
+
+case "$1" in
+    -n)  n=$2;      shift 2 ;;
+    -n*) n=${1#-n}; shift   ;;
+esac
+
+exec sed ${n}q "$@"

compat/plan9/main.c

@@ -1,7 +1,4 @@
 /*
- * NewTek NDI common code
- * Copyright (c) 2017 Maksym Veremeyenko
- *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
@@ -19,12 +16,19 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#ifndef AVDEVICE_LIBNDI_NEWTEK_COMMON_H
-#define AVDEVICE_LIBNDI_NEWTEK_COMMON_H
-
-#include <Processing.NDI.Lib.h>
-
-#define NDI_TIME_BASE 10000000
-#define NDI_TIME_BASE_Q (AVRational){1, NDI_TIME_BASE}
+int plan9_main(int argc, char **argv);
 
+#undef main
+int main(int argc, char **argv)
+{
+    /* The setfcr() function in lib9 is broken, must use asm. */
+#ifdef __i386
+    short fcr;
+    __asm__ volatile ("fstcw        %0 \n"
+                      "or      $63, %0 \n"
+                      "fldcw        %0 \n"
+                      : "=m"(fcr));
 #endif
+
+    return plan9_main(argc, argv);
+}

compat/plan9/printf

@@ -0,0 +1,2 @@
+#!/bin/sh
+exec awk "BEGIN { for (i = 2; i < ARGC; i++) printf \"$1\", ARGV[i] }" "$@"

compat/strtod.c

@@ -25,9 +25,9 @@
 #include "libavutil/avstring.h"
 #include "libavutil/mathematics.h"
 
-static const char *check_nan_suffix(const char *s)
+static char *check_nan_suffix(char *s)
 {
-    const char *start = s;
+    char *start = s;
 
     if (*s++ != '(')
         return start;
@@ -44,7 +44,7 @@ double strtod(const char *, char **);
 
 double avpriv_strtod(const char *nptr, char **endptr)
 {
-    const char *end;
+    char *end;
     double res;
 
     /* Skip leading spaces */
@@ -81,13 +81,13 @@ double avpriv_strtod(const char *nptr, char **endptr)
                !av_strncasecmp(nptr, "+0x", 3)) {
         /* FIXME this doesn't handle exponents, non-integers (float/double)
          * and numbers too large for long long */
-        res = strtoll(nptr, (char **)&end, 16);
+        res = strtoll(nptr, &end, 16);
     } else {
-        res = strtod(nptr, (char **)&end);
+        res = strtod(nptr, &end);
     }
 
     if (endptr)
-        *endptr = (char *)end;
+        *endptr = end;
 
     return res;
 }

compat/tms470/math.h

@@ -16,15 +16,15 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "config.h"
+#ifndef COMPAT_TMS470_MATH_H
+#define COMPAT_TMS470_MATH_H
 
-#include "libavcodec/aarch64/idct.h"
+#include_next <math.h>
 
-static const struct algo fdct_tab_arch[] = {
-    { 0 }
-};
+#undef INFINITY
+#undef NAN
 
-static const struct algo idct_tab_arch[] = {
-    { "SIMPLE-NEON", ff_simple_idct_neon, FF_IDCT_PERM_PARTTRANS, AV_CPU_FLAG_NEON },
-    { 0 }
-};
+#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
+#define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
+
+#endif /* COMPAT_TMS470_MATH_H */

compat/w32dlfcn.h

@@ -1,94 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef COMPAT_W32DLFCN_H
-#define COMPAT_W32DLFCN_H
-
-#ifdef _WIN32
-#include <windows.h>
-#include "config.h"
-#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
-#include "libavutil/wchar_filename.h"
-#endif
-/**
- * Safe function used to open dynamic libs. This attempts to improve program security
- * by removing the current directory from the dll search path. Only dll's found in the
- * executable or system directory are allowed to be loaded.
- * @param name  The dynamic lib name.
- * @return A handle to the opened lib.
- */
-static inline HMODULE win32_dlopen(const char *name)
-{
-#if _WIN32_WINNT < 0x0602
-    // Need to check if KB2533623 is available
-    if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
-        HMODULE module = NULL;
-        wchar_t *path = NULL, *name_w = NULL;
-        DWORD pathlen;
-        if (utf8towchar(name, &name_w))
-            goto exit;
-        path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
-        // Try local directory first
-        pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
-        pathlen = wcsrchr(path, '\\') - path;
-        if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
-            goto exit;
-        path[pathlen] = '\\';
-        wcscpy(path + pathlen + 1, name_w);
-        module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
-        if (module == NULL) {
-            // Next try System32 directory
-            pathlen = GetSystemDirectoryW(path, MAX_PATH);
-            if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
-                goto exit;
-            path[pathlen] = '\\';
-            wcscpy(path + pathlen + 1, name_w);
-            module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
-        }
-exit:
-        av_free(path);
-        av_free(name_w);
-        return module;
-    }
-#endif
-#ifndef LOAD_LIBRARY_SEARCH_APPLICATION_DIR
-#   define LOAD_LIBRARY_SEARCH_APPLICATION_DIR 0x00000200
-#endif
-#ifndef LOAD_LIBRARY_SEARCH_SYSTEM32
-#   define LOAD_LIBRARY_SEARCH_SYSTEM32        0x00000800
-#endif
-#if HAVE_WINRT
-    wchar_t *name_w = NULL;
-    int ret;
-    if (utf8towchar(name, &name_w))
-        return NULL;
-    ret = LoadPackagedLibrary(name_w, 0);
-    av_free(name_w);
-    return ret;
-#else
-    return LoadLibraryExA(name, NULL, LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32);
-#endif
-}
-#define dlopen(name, flags) win32_dlopen(name)
-#define dlclose FreeLibrary
-#define dlsym GetProcAddress
-#else
-#include <dlfcn.h>
-#endif
-
-#endif /* COMPAT_W32DLFCN_H */

compat/w32pthreads.h

@@ -39,6 +39,11 @@
 #include <windows.h>
 #include <process.h>
 
+#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
+#undef MemoryBarrier
+#define MemoryBarrier __sync_synchronize
+#endif
+
 #include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
@@ -51,19 +56,28 @@ typedef struct pthread_t {
     void *ret;
 } pthread_t;
 
-/* use light weight mutex/condition variable API for Windows Vista and later */
-typedef SRWLOCK pthread_mutex_t;
+/* the conditional variable api for windows 6.0+ uses critical sections and
+ * not mutexes */
+typedef CRITICAL_SECTION pthread_mutex_t;
+
+/* This is the CONDITION_VARIABLE typedef for using Windows' native
+ * conditional variables on kernels 6.0+. */
+#if HAVE_CONDITION_VARIABLE_PTR
 typedef CONDITION_VARIABLE pthread_cond_t;
+#else
+typedef struct pthread_cond_t {
+    void *Ptr;
+} pthread_cond_t;
+#endif
 
-#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
-#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
-
+#if _WIN32_WINNT >= 0x0600
 #define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
 #define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
+#endif
 
 static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
-    pthread_t *h = (pthread_t*)arg;
+    pthread_t *h = arg;
     h->ret = h->func(h->arg);
     return 0;
 }
@@ -100,25 +114,26 @@ static av_unused int pthread_join(pthread_t thread, void **value_ptr)
 
 static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
 {
-    InitializeSRWLock(m);
+    InitializeCriticalSection(m);
     return 0;
 }
 static inline int pthread_mutex_destroy(pthread_mutex_t *m)
 {
-    /* Unlocked SWR locks use no resources */
+    DeleteCriticalSection(m);
     return 0;
 }
 static inline int pthread_mutex_lock(pthread_mutex_t *m)
 {
-    AcquireSRWLockExclusive(m);
+    EnterCriticalSection(m);
     return 0;
 }
 static inline int pthread_mutex_unlock(pthread_mutex_t *m)
 {
-    ReleaseSRWLockExclusive(m);
+    LeaveCriticalSection(m);
     return 0;
 }
 
+#if _WIN32_WINNT >= 0x0600
 typedef INIT_ONCE pthread_once_t;
 #define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
 
@@ -152,7 +167,7 @@ static inline int pthread_cond_broadcast(pthread_cond_t *cond)
 
 static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
-    SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
+    SleepConditionVariableCS(cond, mutex, INFINITE);
     return 0;
 }
 
@@ -162,4 +177,242 @@ static inline int pthread_cond_signal(pthread_cond_t *cond)
     return 0;
 }
 
+#else // _WIN32_WINNT < 0x0600
+
+/* atomic init state of dynamically loaded functions */
+static LONG w32thread_init_state = 0;
+static av_unused void w32thread_init(void);
+
+/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
+ * compatible to the one used in the native API */
+
+typedef union pthread_once_t  {
+    void * Ptr;    ///< For the Windows 6.0+ native functions
+    LONG state;    ///< For the pre-Windows 6.0 compat code
+} pthread_once_t;
+
+#define PTHREAD_ONCE_INIT {0}
+
+/* function pointers to init once API on windows 6.0+ kernels */
+static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
+static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
+
+/* pre-Windows 6.0 compat using a spin-lock */
+static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
+{
+    switch (InterlockedCompareExchange(state, 1, 0)) {
+    /* Initial run */
+    case 0:
+        init_routine();
+        InterlockedExchange(state, 2);
+        break;
+    /* Another thread is running init */
+    case 1:
+        while (1) {
+            MemoryBarrier();
+            if (*state == 2)
+                break;
+            Sleep(0);
+        }
+        break;
+    /* Initialization complete */
+    case 2:
+        break;
+    }
+}
+
+static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
+{
+    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
+
+    /* Use native functions on Windows 6.0+ */
+    if (initonce_begin && initonce_complete) {
+        BOOL pending = FALSE;
+        initonce_begin(once_control, 0, &pending, NULL);
+        if (pending)
+            init_routine();
+        initonce_complete(once_control, 0, NULL);
+        return 0;
+    }
+
+    w32thread_once_fallback(&once_control->state, init_routine);
+    return 0;
+}
+
+/* for pre-Windows 6.0 platforms we need to define and use our own condition
+ * variable and api */
+
+typedef struct  win32_cond_t {
+    pthread_mutex_t mtx_broadcast;
+    pthread_mutex_t mtx_waiter_count;
+    volatile int waiter_count;
+    HANDLE semaphore;
+    HANDLE waiters_done;
+    volatile int is_broadcast;
+} win32_cond_t;
+
+/* function pointers to conditional variable API on windows 6.0+ kernels */
+static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
+static void (WINAPI *cond_init)(pthread_cond_t *cond);
+static void (WINAPI *cond_signal)(pthread_cond_t *cond);
+static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
+                                DWORD milliseconds);
+
+static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
+{
+    win32_cond_t *win32_cond = NULL;
+
+    w32thread_once_fallback(&w32thread_init_state, w32thread_init);
+
+    if (cond_init) {
+        cond_init(cond);
+        return 0;
+    }
+
+    /* non native condition variables */
+    win32_cond = av_mallocz(sizeof(win32_cond_t));
+    if (!win32_cond)
+        return ENOMEM;
+    cond->Ptr = win32_cond;
+    win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
+    if (!win32_cond->semaphore)
+        return ENOMEM;
+    win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
+    if (!win32_cond->waiters_done)
+        return ENOMEM;
+
+    pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
+    pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
+    return 0;
+}
+
+static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
+{
+    win32_cond_t *win32_cond = cond->Ptr;
+    /* native condition variables do not destroy */
+    if (cond_init)
+        return 0;
+
+    /* non native condition variables */
+    CloseHandle(win32_cond->semaphore);
+    CloseHandle(win32_cond->waiters_done);
+    pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
+    pthread_mutex_destroy(&win32_cond->mtx_broadcast);
+    av_freep(&win32_cond);
+    cond->Ptr = NULL;
+    return 0;
+}
+
+static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
+{
+    win32_cond_t *win32_cond = cond->Ptr;
+    int have_waiter;
+
+    if (cond_broadcast) {
+        cond_broadcast(cond);
+        return 0;
+    }
+
+    /* non native condition variables */
+    pthread_mutex_lock(&win32_cond->mtx_broadcast);
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    have_waiter = 0;
+
+    if (win32_cond->waiter_count) {
+        win32_cond->is_broadcast = 1;
+        have_waiter = 1;
+    }
+
+    if (have_waiter) {
+        ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
+        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
+        ResetEvent(win32_cond->waiters_done);
+        win32_cond->is_broadcast = 0;
+    } else
+        pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
+    return 0;
+}
+
+static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+{
+    win32_cond_t *win32_cond = cond->Ptr;
+    int last_waiter;
+    if (cond_wait) {
+        cond_wait(cond, mutex, INFINITE);
+        return 0;
+    }
+
+    /* non native condition variables */
+    pthread_mutex_lock(&win32_cond->mtx_broadcast);
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    win32_cond->waiter_count++;
+    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
+
+    // unlock the external mutex
+    pthread_mutex_unlock(mutex);
+    WaitForSingleObject(win32_cond->semaphore, INFINITE);
+
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    win32_cond->waiter_count--;
+    last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
+    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+
+    if (last_waiter)
+        SetEvent(win32_cond->waiters_done);
+
+    // lock the external mutex
+    return pthread_mutex_lock(mutex);
+}
+
+static av_unused int pthread_cond_signal(pthread_cond_t *cond)
+{
+    win32_cond_t *win32_cond = cond->Ptr;
+    int have_waiter;
+    if (cond_signal) {
+        cond_signal(cond);
+        return 0;
+    }
+
+    pthread_mutex_lock(&win32_cond->mtx_broadcast);
+
+    /* non-native condition variables */
+    pthread_mutex_lock(&win32_cond->mtx_waiter_count);
+    have_waiter = win32_cond->waiter_count;
+    pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
+
+    if (have_waiter) {
+        ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
+        WaitForSingleObject(win32_cond->waiters_done, INFINITE);
+        ResetEvent(win32_cond->waiters_done);
+    }
+
+    pthread_mutex_unlock(&win32_cond->mtx_broadcast);
+    return 0;
+}
+#endif
+
+static av_unused void w32thread_init(void)
+{
+#if _WIN32_WINNT < 0x0600
+    HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
+    /* if one is available, then they should all be available */
+    cond_init      =
+        (void*)GetProcAddress(kernel_dll, "InitializeConditionVariable");
+    cond_broadcast =
+        (void*)GetProcAddress(kernel_dll, "WakeAllConditionVariable");
+    cond_signal    =
+        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
+    cond_wait      =
+        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
+    initonce_begin =
+        (void*)GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
+    initonce_complete =
+        (void*)GetProcAddress(kernel_dll, "InitOnceComplete");
+#endif
+
+}
+
 #endif /* COMPAT_W32PTHREADS_H */

compat/windows/makedef

@@ -45,11 +45,7 @@ libname=$(mktemp -u "library").lib
 
 trap 'rm -f -- $libname' EXIT
 
-if [ -n "$AR" ]; then
-    $AR rcs ${libname} $@ >/dev/null
-else
-    lib -out:${libname} $@ >/dev/null
-fi
+lib -out:${libname} $@ >/dev/null
 if [ $? != 0 ]; then
     echo "Could not create temporary library." >&2
     exit 1
@@ -58,7 +54,23 @@ fi
 IFS='
 '
 
-prefix="$EXTERN_PREFIX"
+# Determine if we're building for x86 or x86_64 and
+# set the symbol prefix accordingly.
+prefix=""
+arch=$(dumpbin -headers ${libname} |
+       tr '\t' ' ' |
+       grep '^ \+.\+machine \+(.\+)' |
+       head -1 |
+       sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
+
+if [ "${arch}" = "x86" ]; then
+    prefix="_"
+else
+    if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
+        echo "Unknown machine type." >&2
+        exit 1
+    fi
+fi
 
 started=0
 regex="none"
@@ -100,19 +112,7 @@ for line in $(cat ${vscript} | tr '\t' ' '); do
 '
 done
 
-if [ -n "$NM" ]; then
-    # Use eval, since NM="nm -g"
-    dump=$(eval "$NM --defined-only -g ${libname}" |
-              grep -v : |
-              grep -v ^$ |
-              cut -d' ' -f3 |
-              sed -e "s/^${prefix}//")
-else
-    dump=$(dumpbin -linkermember:1 ${libname} |
-              sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
-              tail -n +2 |
-              cut -d' ' -f3)
-fi
+dump=$(dumpbin -linkermember:1 ${libname})
 
 rm ${libname}
 
@@ -121,6 +121,9 @@ list=""
 for exp in ${regex}; do
     list="${list}"'
 '$(echo "${dump}" |
+          sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
+          tail -n +2 |
+          cut -d' ' -f3 |
           grep "^${exp}" |
           sed -e 's/^/    /')
 done

doc/APIchanges

@@ -2,450 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.
 
 The last version increases were:
-libavcodec:    2017-10-21
-libavdevice:   2017-10-21
-libavfilter:   2017-10-21
-libavformat:   2017-10-21
-libavresample: 2017-10-21
-libpostproc:   2017-10-21
-libswresample: 2017-10-21
-libswscale:    2017-10-21
-libavutil:     2017-10-21
+libavcodec:    2015-08-28
+libavdevice:   2015-08-28
+libavfilter:   2015-08-28
+libavformat:   2015-08-28
+libavresample: 2015-08-28
+libpostproc:   2015-08-28
+libswresample: 2015-08-28
+libswscale:    2015-08-28
+libavutil:     2015-08-28
 
 
 API changes, most recent first:
 
--------- 8< --------- FFmpeg 4.0 was cut here -------- 8< ---------
-
-2018-04-03 - d6fc031caf - lavu 56.13.100 - pixdesc.h
-  Deprecate AV_PIX_FMT_FLAG_PSEUDOPAL and make allocating a pseudo palette
-  optional for API users (see AV_PIX_FMT_FLAG_PSEUDOPAL doxygen for details).
-
-2018-04-01 - 860086ee16 - lavc 58.17.100 - avcodec.h
-  Add av_packet_make_refcounted().
-
-2018-04-01 - f1805d160d - lavfi 7.14.100 - avfilter.h
-  Deprecate use of avfilter_register(), avfilter_register_all(),
-  avfilter_next(). Add av_filter_iterate().
-
-2018-03-25 - b7d0d912ef - lavc 58.16.100 - avcodec.h
-  Add FF_SUB_CHARENC_MODE_IGNORE.
-
-2018-03-23 - db2a7c947e - lavu 56.12.100 - encryption_info.h
-  Add AVEncryptionInitInfo and AVEncryptionInfo structures to hold new side-data
-  for encryption info.
-
-2018-03-21 - f14ca60001 - lavc 58.15.100 - avcodec.h
-  Add av_packet_make_writable().
-
-2018-03-18 - 4b86ac27a0 - lavu 56.11.100 - frame.h
-  Add AV_FRAME_DATA_QP_TABLE_PROPERTIES and AV_FRAME_DATA_QP_TABLE_DATA.
-
-2018-03-15 - e0e72539cf - lavu 56.10.100 - opt.h
-  Add AV_OPT_FLAG_BSF_PARAM
-
-2018-03-07 - 950170bd3b - lavu 56.9.100 - crc.h
-  Add AV_CRC_8_EBU crc variant.
-
-2018-03-07 - 2a0eb86857 - lavc 58.14.100 - mediacodec.h
-  Change the default behavior of avcodec_flush() on mediacodec
-  video decoders. To restore the previous behavior, use the new
-  delay_flush=1 option.
-
-2018-03-01 - 6731f60598 - lavu 56.8.100 - frame.h
-  Add av_frame_new_side_data_from_buf().
-
-2018-02-15 - 8a8d0b319a
-  Change av_ripemd_update(), av_murmur3_update() and av_hash_update() length
-  parameter type to size_t at next major bump.
-
-2018-02-12 - bcab11a1a2 - lavfi 7.12.100 - avfilter.h
-  Add AVFilterContext.extra_hw_frames.
-
-2018-02-12 - d23fff0d8a - lavc 58.11.100 - avcodec.h
-  Add AVCodecContext.extra_hw_frames.
-
-2018-02-06 - 0694d87024 - lavf 58.9.100 - avformat.h
-  Deprecate use of av_register_input_format(), av_register_output_format(),
-  av_register_all(), av_iformat_next(), av_oformat_next().
-  Add av_demuxer_iterate(), and av_muxer_iterate().
-
-2018-02-06 - 36c85d6e77 - lavc 58.10.100 - avcodec.h
-  Deprecate use of avcodec_register(), avcodec_register_all(),
-  av_codec_next(), av_register_codec_parser(), and av_parser_next().
-  Add av_codec_iterate() and av_parser_iterate().
-
-2018-02-04 - ff46124b0d - lavf 58.8.100 - avformat.h
-  Deprecate the current names of the RTSP "timeout", "stimeout", "user-agent"
-  options. Introduce "listen_timeout" as replacement for the current "timeout"
-  option, and "user_agent" as replacement for "user-agent". Once the deprecation
-  is over, the old "timeout" option will be removed, and "stimeout" will be
-  renamed to "stimeout" (the "timeout" option will essentially change semantics).
-
-2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
-  Deprecate AVFormatContext filename field which had limited length, use the
-  new dynamically allocated url field instead.
-
-2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
-  Add url field to AVFormatContext and add ff_format_set_url helper function.
-
-2018-01-27 - 6194d7e564 - lavf 58.6.100 - avformat.h
-  Add AVFMTCTX_UNSEEKABLE (for HLS demuxer).
-
-2018-01-23 - 9f07cf7c00 - lavu 56.9.100 - aes_ctr.h
-  Add method to set the 16-byte IV.
-
-2018-01-16 - 631c56a8e4 - lavf 58.5.100 - avformat.h
-  Explicitly make avformat_network_init() and avformat_network_deinit() optional.
-  If these are not called, network initialization and deinitialization is
-  automatic, and unlike in older versions, fully supported, unless libavformat
-  is linked to ancient GnuTLS and OpenSSL.
-
-2018-01-16 - 6512ff72f9 - lavf 58.4.100 - avformat.h
-  Deprecate AVStream.recommended_encoder_configuration. It was useful only for
-  FFserver, which has been removed.
-
-2018-01-05 - 798dcf2432 - lavfi 7.11.101 - avfilter.h
-  Deprecate avfilter_link_get_channels(). Use av_buffersink_get_channels().
-
-2017-01-04 - c29038f304 - lavr 4.0.0 - avresample.h
-  Deprecate the entire library. Merged years ago to provide compatibility
-  with Libav, it remained unmaintained by the FFmpeg project and duplicated
-  functionality provided by libswresample.
-
-  In order to improve consistency and reduce attack surface, it has been deprecated.
-  Users of this library are asked to migrate to libswresample, which, as well as
-  providing more functionality, is faster and has higher accuracy.
-
-2017-12-26 - a04c2c707d - lavc 58.9.100 - avcodec.h
-  Deprecate av_lockmgr_register(). You need to build FFmpeg with threading
-  support enabled to get basic thread-safety (which is the default build
-  configuration).
-
-2017-12-24 - 8b81eabe57 - lavu 56.7.100 - cpu.h
-  AVX-512 flags added.
-
-2017-12-16 - 8bf4e6d3ce - lavc 58.8.100 - avcodec.h
-  The MediaCodec decoders now support AVCodecContext.hw_device_ctx.
-
-2017-12-16 - e4d9f05ca7 - lavu 56.6.100 - hwcontext.h hwcontext_mediacodec.h
-  Add AV_HWDEVICE_TYPE_MEDIACODEC and a new installed header with
-  MediaCodec-specific hwcontext definitions.
-
-2017-12-14 - b945fed629 - lavc 58.7.100 - avcodec.h
-  Add AV_CODEC_CAP_HARDWARE, AV_CODEC_CAP_HYBRID, and AVCodec.wrapper_name,
-  and mark all AVCodecs accordingly.
-
-2017-11-29 - d268094f88 - lavu 56.4.100 / 56.7.0 - stereo3d.h
-  Add view field to AVStereo3D structure and AVStereo3DView enum.
-
-2017-11-26 - 3a71bcc213 - lavc 58.6.100 - avcodec.h
-  Add const to AVCodecContext.hwaccel.
-
-2017-11-26 - 3536a3efb9 - lavc 58.5.100 - avcodec.h
-  Deprecate user visibility of the AVHWAccel structure and the functions
-  av_register_hwaccel() and av_hwaccel_next().
-
-2017-11-26 - 24cc0a53e9 - lavc 58.4.100 - avcodec.h
-  Add AVCodecHWConfig and avcodec_get_hw_config().
-
-2017-11-22 - 3650cb2dfa - lavu 56.3.100 - opencl.h
-  Remove experimental OpenCL API (av_opencl_*).
-
-2017-11-22 - b25d8ef0a7 - lavu 56.2.100 - hwcontext.h hwcontext_opencl.h
-  Add AV_HWDEVICE_TYPE_OPENCL and a new installed header with
-  OpenCL-specific hwcontext definitions.
-
-2017-11-22 - a050f56c09 - lavu 56.1.100 - pixfmt.h
-  Add AV_PIX_FMT_OPENCL.
-
-2017-11-11 - 48e4eda11d - lavc 58.3.100 - avcodec.h
-  Add avcodec_get_hw_frames_parameters().
-
--------- 8< --------- FFmpeg 3.4 was cut here -------- 8< ---------
-
-2017-09-28 - b6cf66ae1c - lavc 57.106.104 - avcodec.h
-  Add AV_PKT_DATA_A53_CC packet side data, to export closed captions
-
-2017-09-27 - 7aa6b8a68f - lavu 55.77.101 / lavu 55.31.1 - frame.h
-  Allow passing the value of 0 (meaning "automatic") as the required alignment
-  to av_frame_get_buffer().
-
-2017-09-27 - 522f877086 - lavu 55.77.100 / lavu 55.31.0 - cpu.h
-  Add av_cpu_max_align() for querying maximum required data alignment.
-
-2017-09-26 - b1cf151c4d - lavc 57.106.102 - avcodec.h
-  Deprecate AVCodecContext.refcounted_frames. This was useful for deprecated
-  API only (avcodec_decode_video2/avcodec_decode_audio4). The new decode APIs
-  (avcodec_send_packet/avcodec_receive_frame) always work with reference
-  counted frames.
-
-2017-09-21 - 6f15f1cdc8 - lavu 55.76.100 / 56.6.0 - pixdesc.h
-  Add av_color_range_from_name(), av_color_primaries_from_name(),
-  av_color_transfer_from_name(), av_color_space_from_name(), and
-  av_chroma_location_from_name().
-
-2017-09-13 - 82342cead1 - lavc 57.106.100 - avcodec.h
-  Add AV_PKT_FLAG_TRUSTED.
-
-2017-09-13 - 9cb23cd9fe - lavu 55.75.100 - hwcontext.h hwcontext_drm.h
-  Add AV_HWDEVICE_TYPE_DRM and implementation.
-
-2017-09-08 - 5ba2aef6ec - lavfi 6.103.100 - buffersrc.h
-  Add av_buffersrc_close().
-
-2017-09-04 - 6cadbb16e9 - lavc 57.105.100 - avcodec.h
-  Add AV_HWACCEL_CODEC_CAP_EXPERIMENTAL, replacing the deprecated
-  HWACCEL_CODEC_CAP_EXPERIMENTAL flag.
-
-2017-09-01 - 5d76674756 - lavf 57.81.100 - avio.h
-  Add avio_read_partial().
-
-2017-09-01 - xxxxxxx - lavf 57.80.100 / 57.11.0 - avio.h
-  Add avio_context_free(). From now on it must be used for freeing AVIOContext.
-
-2017-08-08 - 1460408703 - lavu 55.74.100 - pixdesc.h
-  Add AV_PIX_FMT_FLAG_FLOAT pixel format flag.
-
-2017-08-08 - 463b81de2b - lavu 55.72.100 - imgutils.h
-  Add av_image_fill_black().
-
-2017-08-08 - caa12027ba - lavu 55.71.100 - frame.h
-  Add av_frame_apply_cropping().
-
-2017-07-25 - 24de4fddca - lavu 55.69.100 - frame.h
-  Add AV_FRAME_DATA_ICC_PROFILE side data type.
-
-2017-06-27 - 70143a3954 - lavc 57.100.100 - avcodec.h
-  DXVA2 and D3D11 hardware accelerated decoding now supports the new hwaccel API,
-  which can create the decoder context and allocate hardware frame automatically.
-  See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx. For D3D11,
-  the new AV_PIX_FMT_D3D11 pixfmt must be used with the new API.
-
-2017-06-27 - 3303511f33 - lavu 56.67.100 - hwcontext.h
-  Add AV_HWDEVICE_TYPE_D3D11VA and AV_PIX_FMT_D3D11.
-
-2017-06-24 - 09891c5391 - lavf 57.75.100 - avio.h
-  Add AVIO_DATA_MARKER_FLUSH_POINT to signal preferred flush points to aviobuf.
-
-2017-06-14 - d59c6a3aeb - lavu 55.66.100 - hwcontext.h
-  av_hwframe_ctx_create_derived() now takes some AV_HWFRAME_MAP_* combination
-  as its flags argument (which was previously unused).
-
-2017-06-14 - 49ae8a5e87 - lavc 57.99.100 - avcodec.h
-  Add AV_HWACCEL_FLAG_ALLOW_PROFILE_MISMATCH.
-
-2017-06-14 - 0b1794a43e - lavu 55.65.100 - hwcontext.h
-  Add AV_HWDEVICE_TYPE_NONE, av_hwdevice_find_type_by_name(),
-  av_hwdevice_get_type_name() and av_hwdevice_iterate_types().
-
-2017-06-14 - b22172f6f3 - lavu 55.64.100 - hwcontext.h
-  Add av_hwdevice_ctx_create_derived().
-
-2017-05-15 - 532b23f079 - lavc 57.96.100 - avcodec.h
-  VideoToolbox hardware-accelerated decoding now supports the new hwaccel API,
-  which can create the decoder context and allocate hardware frames automatically.
-  See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx.
-
-2017-05-15 - 532b23f079 - lavu 57.63.100 - hwcontext.h
-  Add AV_HWDEVICE_TYPE_VIDEOTOOLBOX and implementation.
-
-2017-05-08 - f089e02fa2 - lavc 57.95.100 / 57.31.0 - avcodec.h
-  Add AVCodecContext.apply_cropping to control whether cropping
-  is handled by libavcodec or the caller.
-
-2017-05-08 - a47bd5d77e - lavu 55.62.100 / 55.30.0 - frame.h
-  Add AVFrame.crop_left/right/top/bottom fields for attaching cropping
-  information to video frames.
-
-2017-xx-xx - xxxxxxxxxx
-  Change av_sha_update(), av_sha512_update() and av_md5_sum()/av_md5_update() length
-  parameter type to size_t at next major bump.
-
-2017-05-05 - c0f17a905f - lavc 57.94.100 - avcodec.h
-  The cuvid decoders now support AVCodecContext.hw_device_ctx, which removes
-  the requirement to set an incomplete AVCodecContext.hw_frames_ctx only to
-  set the Cuda device handle.
-
-2017-04-11 - 8378466507 - lavu 55.61.100 - avstring.h
-  Add av_strireplace().
-
-2016-04-06 - 157e57a181 - lavc 57.92.100 - avcodec.h
-  Add AV_PKT_DATA_CONTENT_LIGHT_LEVEL packet side data.
-
-2016-04-06 - b378f5bd64 - lavu 55.60.100 - mastering_display_metadata.h
-  Add AV_FRAME_DATA_CONTENT_LIGHT_LEVEL value, av_content_light_metadata_alloc()
-  and av_content_light_metadata_create_side_data() API, and AVContentLightMetadata
-  type to export content light level video properties.
-
-2017-03-31 - 9033e8723c - lavu 55.57.100 - spherical.h
-  Add av_spherical_projection_name().
-  Add av_spherical_from_name().
-
-2017-03-30 - 4cda23f1f1 - lavu 55.53.100 / 55.27.0 - hwcontext.h
-  Add av_hwframe_map() and associated AV_HWFRAME_MAP_* flags.
-  Add av_hwframe_ctx_create_derived().
-
-2017-03-29 - bfdcdd6d82 - lavu 55.52.100 - avutil.h
-  add av_fourcc_make_string() function and av_fourcc2str() macro to replace
-  av_get_codec_tag_string() from lavc.
-
-2017-03-27 - ddef3d902f - lavf 57.68.100 - avformat.h
-  Deprecate that demuxers export the stream rotation angle in AVStream.metadata
-  (via an entry named "rotate"). Use av_stream_get_side_data() with
-  AV_PKT_DATA_DISPLAYMATRIX instead, and read the rotation angle with
-  av_display_rotation_get(). The same is done for muxing. Instead of adding a
-  "rotate" entry to AVStream.metadata, AV_PKT_DATA_DISPLAYMATRIX side data has
-  to be added to the AVStream.
-
-2017-03-23 - 7e4ba776a2 - lavc 57.85.101 - avcodec.h
-  vdpau hardware accelerated decoding now supports the new hwaccel API, which
-  can create the decoder context and allocate hardware frame automatically.
-  See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx.
-
-2017-03-23 - 156bd8278f - lavc 57.85.100 - avcodec.h
-  Add AVCodecContext.hwaccel_flags field. This will control some hwaccels at
-  a later point.
-
-2017-03-21 - fc9f14c7de - lavf 57.67.100 / 57.08.0 - avio.h
-  Add AVIO_SEEKABLE_TIME flag.
-
-2017-03-21 - d682ae70b4 - lavf 57.66.105, lavc 57.83.101 - avformat.h, avcodec.h
-  Deprecate AVFMT_FLAG_KEEP_SIDE_DATA. It will be ignored after the next major
-  bump, and libavformat will behave as if it were always set.
-  Deprecate av_packet_merge_side_data() and av_packet_split_side_data().
-
-2016-03-20 - 8200b16a9c - lavu 55.50.100 / 55.21.0 - imgutils.h
-  Add av_image_copy_uc_from(), a version of av_image_copy() for copying
-  from GPU mapped memory.
-
-2017-03-20 - 9c2436e - lavu 55.49.100 - pixdesc.h
-  Add AV_PIX_FMT_FLAG_BAYER pixel format flag.
-
-2017-03-18 - 3796fb2692 - lavfi 6.77.100 - avfilter.h
-  Deprecate AVFilterGraph.resample_lavr_opts
-  It's never been used by avfilter nor passed to anything.
-
-2017-02-10 - 1b7ffddb3a - lavu 55.48.100 / 55.33.0 - spherical.h
-  Add AV_SPHERICAL_EQUIRECTANGULAR_TILE, av_spherical_tile_bounds(),
-  and projection-specific properties (bound_left, bound_top, bound_right,
-  bound_bottom, padding) to AVSphericalMapping.
-
-2017-03-02 - ade7c1a232 - lavc 57.81.104 - videotoolbox.h
-  AVVideotoolboxContext.cv_pix_fmt_type can now be set to 0 to output the
-  native decoder format. (The default value is not changed.)
-
-2017-03-02 - 554bc4eea8 - lavu 55.47.101, lavc 57.81.102, lavf 57.66.103
-  Remove requirement to use AVOption or accessors to access certain fields
-  in AVFrame, AVCodecContext, and AVFormatContext that were previously
-  documented as "no direct access" allowed.
-
-2017-02-13 - c1a5fca06f - lavc 57.80.100 - avcodec.h
-  Add AVCodecContext.hw_device_ctx.
-
-2017-02-11 - e3af49b14b - lavu 55.47.100 - frame.h
-  Add AVFrame.opaque_ref.
-
-2017-01-31 - 2eab48177d - lavu 55.46.100 / 55.20.0 - cpu.h
-  Add AV_CPU_FLAG_SSSE3SLOW.
-
-2017-01-24 - c4618f842a - lavu 55.45.100 - channel_layout.h
-  Add av_get_extended_channel_layout()
-
-2017-01-22 - 76c5a69e26 - lavu 55.44.100 - lfg.h
-  Add av_lfg_init_from_data().
-
-2017-01-17 - 2a4a8653b6 - lavc 57.74.100 - vaapi.h
-  Deprecate struct vaapi_context and the vaapi.h installed header.
-  Callers should set AVCodecContext.hw_frames_ctx instead.
-
-2017-01-12 - dbe9dbed31 - lavfi 6.69.100 - buffersink.h
-  Add av_buffersink_get_*() functions.
-
-2017-01-06 - 9488032e10 - lavf 57.62.100 - avio.h
-  Add avio_get_dyn_buf()
-
-2016-12-10 - f542b152aa - lavu 55.43.100 - imgutils.h
-  Add av_image_check_size2()
-
-2016-12-07 - e7a6f8c972 - lavc 57.67.100 / 57.29.0 - avcodec.h
-  Add AV_PKT_DATA_SPHERICAL packet side data to export AVSphericalMapping
-  information from containers.
-
-2016-12-07 - 8f58ecc344 - lavu 55.42.100 / 55.30.0 - spherical.h
-  Add AV_FRAME_DATA_SPHERICAL value, av_spherical_alloc() API and
-  AVSphericalMapping type to export and describe spherical video properties.
-
-2016-11-18 - 2ab50647ff - lavf 57.58.100 - avformat.h
-  Add av_stream_add_side_data().
-
-2016-11-13 - 775a8477b7 - lavu 55.39.100 - hwcontext_vaapi.h
-  Add AV_VAAPI_DRIVER_QUIRK_ATTRIB_MEMTYPE.
-
-2016-11-13 - a8d51bb424 - lavu 55.38.100 - hwcontext_vaapi.h
-  Add driver quirks field to VAAPI-specific hwdevice and enum with
-  members AV_VAAPI_DRIVER_QUIRK_* to represent its values.
-
-2016-11-10 - 638b216d4f - lavu 55.36.100 - pixfmt.h
-  Add AV_PIX_FMT_GRAY12(LE/BE).
-
--------- 8< --------- FFmpeg 3.2 was cut here -------- 8< ---------
-
-2016-10-24 - 73ead47 - lavf 57.55.100 - avformat.h
-  Add AV_DISPOSITION_TIMED_THUMBNAILS
-
-2016-10-24 - a246fef - lavf 57.54.100 - avformat.h
-  Add avformat_init_output() and AVSTREAM_INIT_IN_ macros
-
-2016-10-22 - f5495c9 - lavu 55.33.100 - avassert.h
-  Add av_assert0_fpu() / av_assert2_fpu()
-
-2016-10-07 - 3f9137c / 32c8359 - lavc 57.61.100 / 57.24.0 - avcodec.h
-  Decoders now export the frame timestamp as AVFrame.pts. It was
-  previously exported as AVFrame.pkt_pts, which is now deprecated.
-
-  Note: When decoding, AVFrame.pts uses the stream/packet timebase,
-  and not the codec timebase.
-
-2016-09-28 - eba0414 - lavu 55.32.100 / 55.16.0 - hwcontext.h hwcontext_qsv.h
-  Add AV_HWDEVICE_TYPE_QSV and a new installed header with QSV-specific
-  hwcontext definitions.
-
-2016-09-26 - 32c25f0 - lavc 57.59.100 / 57.23.0 - avcodec.h
-  AVCodecContext.hw_frames_ctx now may be used by decoders.
-
-2016-09-27 - f0b6f72 - lavf 57.51.100 - avformat.h
-  Add av_stream_get_codec_timebase()
-
-2016-09-27 - 23c0779 - lswr 2.2.100 - swresample.h
-  Add swr_build_matrix().
-
-2016-09-23 - 30d3e36 - lavc 57.58.100 - avcodec.h
-  Add AV_CODEC_CAP_AVOID_PROBING codec capability flag.
-
-2016-09-14 - ae1dd0c - lavf 57.49.100 - avformat.h
-  Add avformat_transfer_internal_stream_timing_info helper to help with stream
-  copy.
-
-2016-08-29 - 4493390 - lavfi 6.58.100 - avfilter.h
-  Add AVFilterContext.nb_threads.
-
-2016-08-15 - c3c4c72 - lavc 57.53.100 - avcodec.h
-  Add trailing_padding to AVCodecContext to match the corresponding
-  field in AVCodecParameters.
-
-2016-08-15 - b746ed7 - lavc 57.52.100 - avcodec.h
-  Add a new API for chained BSF filters and passthrough (null) BSF --
-  av_bsf_list_alloc(), av_bsf_list_free(), av_bsf_list_append(),
-  av_bsf_list_append2(), av_bsf_list_finalize(), av_bsf_list_parse_str()
-  and av_bsf_get_null_filter().
-
-2016-08-04 - 82a33c8 - lavf 57.46.100 - avformat.h
-  Add av_get_frame_filename2()
-
-2016-07-09 - 775389f / 90f469a - lavc 57.50.100 / 57.20.0 - avcodec.h
-  Add FF_PROFILE_H264_MULTIVIEW_HIGH and FF_PROFILE_H264_STEREO_HIGH.
-
 2016-06-30 - c1c7e0ab - lavf 57.41.100 - avformat.h
   Moved codecpar field from AVStream to the end of the struct, so that
   the following private fields are in the same location as in FFmpeg 3.0 (lavf 57.25.100).
@@ -889,7 +458,7 @@ API changes, most recent first:
   Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
   dictionary types being set as options.
 
-2014-08-13 - afbd4b7e09 - lavf 56.01.0 - avformat.h
+2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
   Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
   the user when events happen in the file/stream.
 
@@ -906,7 +475,7 @@ API changes, most recent first:
 2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
   Add avio_feof() and deprecate url_feof().
 
-2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
+2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
   sws_getContext is not going to be removed in the future.
 
 2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h

doc/Makefile

@@ -24,7 +24,6 @@ HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
               doc/fate.html                                             \
               doc/general.html                                          \
               doc/git-howto.html                                        \
-              doc/mailing-list-faq.html                                 \
               doc/nut.html                                              \
               doc/platform.html                                         \
 
@@ -37,6 +36,30 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
+DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
+DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
+DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
+DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
+DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
+DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
+DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
+DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
+DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
+DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
+DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
+DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
+DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
+DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
+ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
+
+DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+PROGS              += $(DOC_EXAMPLES)
+
 all-$(CONFIG_DOC): doc
 
 doc: documentation
@@ -44,6 +67,8 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)
 
+examples: $(DOC_EXAMPLES)
+
 TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
 
 doc/%.txt: TAG = TXT
@@ -96,9 +121,11 @@ doc/%.3: doc/%.pod $(GENTEXI)
 	$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
 
 $(DOCS) doc/doxy/html: | doc/
+$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
+OBJDIRS += doc/examples
 
-DOXY_INPUT      = $(INSTHEADERS)
-DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak
+DOXY_INPUT      = $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)
+DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT))
 
 doc/doxy/html: TAG = DOXY
 doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
@@ -144,7 +171,11 @@ clean:: docclean
 distclean:: docclean
 	$(RM) doc/config.texi
 
-docclean::
+examplesclean:
+	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
+	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
+
+docclean: examplesclean
 	$(RM) $(CLEANSUFFIXES:%=doc/%)
 	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html

doc/bitstream_filters.texi

@@ -18,7 +18,7 @@ comma-separated list of filters, whose parameters follow the filter
 name after a '='.
 
 @example
-ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
+ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
 @end example
 
 Below is a description of the currently available bitstream filters,
@@ -26,46 +26,38 @@ with their parameters, if any.
 
 @section aac_adtstoasc
 
-Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
-bitstream.
+Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
+bitstream filter.
 
 This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
 ADTS header and removes the ADTS header.
 
-This filter is required for example when copying an AAC stream from a
-raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
-to MOV/MP4 files and related formats such as 3GP or M4A. Please note
-that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
+This is required for example when copying an AAC stream from a raw
+ADTS AAC container to a FLV or a MOV/MP4 file.
 
 @section chomp
 
 Remove zero padding at the end of a packet.
 
-@section dca_core
-
-Extract the core from a DCA/DTS stream, dropping extensions such as
-DTS-HD.
-
 @section dump_extra
 
 Add extradata to the beginning of the filtered packets.
 
-@table @option
-@item freq
 The additional argument specifies which packets should be filtered.
 It accepts the values:
 @table @samp
+@item a
+add extradata to all key packets, but only if @var{local_header} is
+set in the @option{flags2} codec context field
+
 @item k
-@item keyframe
 add extradata to all key packets
 
 @item e
-@item all
 add extradata to all packets
 @end table
-@end table
 
-If not specified it is assumed @samp{e}.
+If not specified it is assumed @samp{k}.
 
 For example the following @command{ffmpeg} command forces a global
 header (thus disabling individual packet headers) in the H.264 packets
@@ -75,147 +67,9 @@ the header stored in extradata to the key packets:
 ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
 @end example
 
-@section eac3_core
+@section dca_core
 
-Extract the core from a E-AC-3 stream, dropping extra channels.
-
-@section extract_extradata
-
-Extract the in-band extradata.
-
-Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers,
-or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part
-of the bitstream containing the coded frames) or "out of band" (e.g. on the
-container level). This latter form is called "extradata" in FFmpeg terminology.
-
-This bitstream filter detects the in-band headers and makes them available as
-extradata.
-
-@table @option
-@item remove
-When this option is enabled, the long-term headers are removed from the
-bitstream after extraction.
-@end table
-
-@section filter_units
-
-Remove units with types in or not in a given set from the stream.
-
-@table @option
-@item pass_types
-List of unit types or ranges of unit types to pass through while removing
-all others.  This is specified as a '|'-separated list of unit type values
-or ranges of values with '-'.
-
-@item remove_types
-Identical to @option{pass_types}, except the units in the given set
-removed and all others passed through.
-@end table
-
-Extradata is unchanged by this transformation, but note that if the stream
-contains inline parameter sets then the output may be unusable if they are
-removed.
-
-For example, to remove all non-VCL NAL units from an H.264 stream:
-@example
-ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
-@end example
-
-To remove all AUDs, SEI and filler from an H.265 stream:
-@example
-ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
-@end example
-
-@section hapqa_extract
-
-Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
-
-@table @option
-@item texture
-Specifies the texture to keep.
-
-@table @option
-@item color
-@item alpha
-@end table
-
-@end table
-
-Convert HAPQA to HAPQ
-@example
-ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
-@end example
-
-Convert HAPQA to HAPAlphaOnly
-@example
-ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
-@end example
-
-@section h264_metadata
-
-Modify metadata embedded in an H.264 stream.
-
-@table @option
-@item aud
-Insert or remove AUD NAL units in all access units of the stream.
-
-@table @samp
-@item insert
-@item remove
-@end table
-
-@item sample_aspect_ratio
-Set the sample aspect ratio of the stream in the VUI parameters.
-
-@item video_format
-@item video_full_range_flag
-Set the video format in the stream (see H.264 section E.2.1 and
-table E-2).
-
-@item colour_primaries
-@item transfer_characteristics
-@item matrix_coefficients
-Set the colour description in the stream (see H.264 section E.2.1
-and tables E-3, E-4 and E-5).
-
-@item chroma_sample_loc_type
-Set the chroma sample location in the stream (see H.264 section
-E.2.1 and figure E-1).
-
-@item tick_rate
-Set the tick rate (num_units_in_tick / time_scale) in the VUI
-parameters.  This is the smallest time unit representable in the
-stream, and in many cases represents the field rate of the stream
-(double the frame rate).
-@item fixed_frame_rate_flag
-Set whether the stream has fixed framerate - typically this indicates
-that the framerate is exactly half the tick rate, but the exact
-meaning is dependent on interlacing and the picture structure (see
-H.264 section E.2.1 and table E-6).
-
-@item crop_left
-@item crop_right
-@item crop_top
-@item crop_bottom
-Set the frame cropping offsets in the SPS.  These values will replace
-the current ones if the stream is already cropped.
-
-These fields are set in pixels.  Note that some sizes may not be
-representable if the chroma is subsampled or the stream is interlaced
-(see H.264 section 7.4.2.1.1).
-
-@item sei_user_data
-Insert a string as SEI unregistered user data.  The argument must
-be of the form @emph{UUID+string}, where the UUID is as hex digits
-possibly separated by hyphens, and the string can be anything.
-
-For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
-insert the string ``hello'' associated with the given UUID.
-
-@item delete_filler
-Deletes both filler NAL units and filler SEI messages.
-
-@end table
+Extract DCA core from DTS-HD streams.
 
 @section h264_mp4toannexb
 
@@ -224,7 +78,7 @@ 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 (muxer @code{mpegts}).
+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:
@@ -233,92 +87,6 @@ format with @command{ffmpeg}, you can use the command:
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
 @end example
 
-Please note that this filter is auto-inserted for MPEG-TS (muxer
-@code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
-
-@section h264_redundant_pps
-
-This applies a specific fixup to some Blu-ray streams which contain
-redundant PPSs modifying irrelevant parameters of the stream which
-confuse other transformations which require correct extradata.
-
-A new single global PPS is created, and all of the redundant PPSs
-within the stream are removed.
-
-@section hevc_metadata
-
-Modify metadata embedded in an HEVC stream.
-
-@table @option
-@item aud
-Insert or remove AUD NAL units in all access units of the stream.
-
-@table @samp
-@item insert
-@item remove
-@end table
-
-@item sample_aspect_ratio
-Set the sample aspect ratio in the stream in the VUI parameters.
-
-@item video_format
-@item video_full_range_flag
-Set the video format in the stream (see H.265 section E.3.1 and
-table E.2).
-
-@item colour_primaries
-@item transfer_characteristics
-@item matrix_coefficients
-Set the colour description in the stream (see H.265 section E.3.1
-and tables E.3, E.4 and E.5).
-
-@item chroma_sample_loc_type
-Set the chroma sample location in the stream (see H.265 section
-E.3.1 and figure E.1).
-
-@item tick_rate
-Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
-time_scale).  Combined with @option{num_ticks_poc_diff_one}, this can
-set a constant framerate in the stream.  Note that it is likely to be
-overridden by container parameters when the stream is in a container.
-
-@item num_ticks_poc_diff_one
-Set poc_proportional_to_timing_flag in VPS and VUI and use this value
-to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and
-E.3.1).  Ignored if @option{tick_rate} is not also set.
-
-@item crop_left
-@item crop_right
-@item crop_top
-@item crop_bottom
-Set the conformance window cropping offsets in the SPS.  These values
-will replace the current ones if the stream is already cropped.
-
-These fields are set in pixels.  Note that some sizes may not be
-representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
-
-@end table
-
-@section hevc_mp4toannexb
-
-Convert an HEVC/H.265 bitstream from length prefixed mode to start code
-prefixed mode (as defined in the Annex B of the ITU-T H.265
-specification).
-
-This is required by some streaming formats, typically the MPEG-2
-transport stream format (muxer @code{mpegts}).
-
-For example to remux an MP4 file containing an HEVC stream to mpegts
-format with @command{ffmpeg}, you can use the command:
-
-@example
-ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
-@end example
-
-Please note that this filter is auto-inserted for MPEG-TS (muxer
-@code{mpegts}) and raw HEVC/H.265 (muxer @code{h265} or
-@code{hevc}) output formats.
-
 @section imxdump
 
 Modifies the bitstream to fit in MOV and to be usable by the Final Cut
@@ -369,58 +137,11 @@ exiftran -i -9 frame*.jpg
 ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 @end example
 
-@section mjpegadump
+@section mjpega_dump_header
 
-Add an MJPEG A header to the bitstream, to enable decoding by
-Quicktime.
+@section movsub
 
-@anchor{mov2textsub}
-@section mov2textsub
-
-Extract a representable text file from MOV subtitles, stripping the
-metadata header from each subtitle packet.
-
-See also the @ref{text2movsub} filter.
-
-@section mp3decomp
-
-Decompress non-standard compressed MP3 audio headers.
-
-@section mpeg2_metadata
-
-Modify metadata embedded in an MPEG-2 stream.
-
-@table @option
-@item display_aspect_ratio
-Set the display aspect ratio in the stream.
-
-The following fixed values are supported:
-@table @option
-@item 4/3
-@item 16/9
-@item 221/100
-@end table
-Any other value will result in square pixels being signalled instead
-(see H.262 section 6.3.3 and table 6-3).
-
-@item frame_rate
-Set the frame rate in the stream.  This is constructed from a table
-of known values combined with a small multiplier and divisor - if
-the supplied value is not exactly representable, the nearest
-representable value will be used instead (see H.262 section 6.3.3
-and table 6-4).
-
-@item video_format
-Set the video format in the stream (see H.262 section 6.3.6 and
-table 6-6).
-
-@item colour_primaries
-@item transfer_characteristics
-@item matrix_coefficients
-Set the colour description in the stream (see H.262 section 6.3.6
-and tables 6-7, 6-8 and 6-9).
-
-@end table
+@section mp3_header_decompress
 
 @section mpeg4_unpack_bframes
 
@@ -444,82 +165,20 @@ ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
 
 @section noise
 
-Damages the contents of packets or simply drops them without damaging the
-container. Can be used for fuzzing or testing error resilience/concealment.
+Damages the contents of packets without damaging the container. Can be
+used for fuzzing or testing error resilience/concealment.
 
 Parameters:
-@table @option
-@item amount
 A numeral string, whose value is related to how often output bytes will
 be modified. Therefore, values below or equal to 0 are forbidden, and
 the lower the more frequent bytes will be modified, with 1 meaning
 every byte is modified.
-@item dropamount
-A numeral string, whose value is related to how often packets will be dropped.
-Therefore, values below or equal to 0 are forbidden, and the lower the more
-frequent packets will be dropped, with 1 meaning every packet is dropped.
-@end table
 
-The following example applies the modification to every byte but does not drop
-any packets.
 @example
 ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
 @end example
-
-@section null
-This bitstream filter passes the packets through unchanged.
+applies the modification to every byte.
 
 @section remove_extra
 
-Remove extradata from packets.
-
-It accepts the following parameter:
-@table @option
-@item freq
-Set which frame types to remove extradata from.
-
-@table @samp
-@item k
-Remove extradata from non-keyframes only.
-
-@item keyframe
-Remove extradata from keyframes only.
-
-@item e, all
-Remove extradata from all frames.
-
-@end table
-@end table
-
-@anchor{text2movsub}
-@section text2movsub
-
-Convert text subtitles to MOV subtitles (as used by the @code{mov_text}
-codec) with metadata headers.
-
-See also the @ref{mov2textsub} filter.
-
-@section trace_headers
-
-Log trace output containing all syntax elements in the coded stream
-headers (everything above the level of individual coded blocks).
-This can be useful for debugging low-level stream issues.
-
-Supports H.264, H.265 and MPEG-2.
-
-@section vp9_superframe
-
-Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
-fixes merging of split/segmented VP9 streams where the alt-ref frame
-was split from its visible counterpart.
-
-@section vp9_superframe_split
-
-Split VP9 superframes into single frames.
-
-@section vp9_raw_reorder
-
-Given a VP9 stream with correct timestamps but possibly out of order,
-insert additional show-existing-frame packets to correct the ordering.
-
 @c man end BITSTREAM FILTERS

doc/build_system.txt

@@ -45,9 +45,6 @@ libswscale/swscale-test
 config
     Reconfigure the project with the current configuration.
 
-tools/target_dec_<decoder>_fuzzer
-    Build fuzzer to fuzz the specified decoder.
-
 
 Useful standard make commands:
 make -t <target>

doc/codecs.texi

@@ -44,6 +44,12 @@ Use 1/4 pel motion compensation.
 Use loop filter.
 @item qscale
 Use fixed qscale.
+@item gmc
+Use gmc.
+@item mv0
+Always try a mb with mv=<0,0>.
+@item input_preserved
+
 @item pass1
 Use internal 2pass ratecontrol in first pass mode.
 @item pass2
@@ -56,6 +62,8 @@ Do not draw edges.
 Set error[?] variables during encoding.
 @item truncated
 
+@item naq
+Normalize adaptive quantization.
 @item ildct
 Use interlaced DCT.
 @item low_delay
@@ -130,8 +138,7 @@ Set audio sampling rate (in Hz).
 Set number of audio channels.
 
 @item cutoff @var{integer} (@emph{encoding,audio})
-Set cutoff bandwidth. (Supported only by selected encoders, see
-their respective documentation sections.)
+Set cutoff bandwidth.
 
 @item frame_size @var{integer} (@emph{encoding,audio})
 Set audio frame size.
@@ -467,6 +474,8 @@ rate control
 macroblock (MB) type
 @item qp
 per-block quantization parameter (QP)
+@item mv
+motion vector
 @item dct_coeff
 
 @item green_metadata
@@ -476,12 +485,18 @@ display complexity metadata for the upcoming frame, GoP or for a given duration.
 
 @item startcode
 
+@item pts
+
 @item er
 error recognition
 @item mmco
 memory management control operations (H.264)
 @item bugs
 
+@item vis_qp
+visualize quantization parameter (QP), lower QP are tinted greener
+@item vis_mb_type
+visualize block types
 @item buffers
 picture buffer allocations
 @item thread_ops
@@ -490,6 +505,21 @@ threading operations
 skip motion compensation
 @end table
 
+@item vismv @var{integer} (@emph{decoding,video})
+Visualize motion vectors (MVs).
+
+This option is deprecated, see the codecview filter instead.
+
+Possible values:
+@table @samp
+@item pf
+forward predicted MVs of P-frames
+@item bf
+forward predicted MVs of B-frames
+@item bb
+backward predicted MVs of B-frames
+@end table
+
 @item cmp @var{integer} (@emph{encoding,video})
 Set full pel me compare function.
 
@@ -726,6 +756,8 @@ Set context model.
 
 @item slice_flags @var{integer}
 
+@item xvmc_acceleration @var{integer}
+
 @item mbd @var{integer} (@emph{encoding,video})
 Set macroblock decision algorithm (high quality mode).
 
@@ -1017,34 +1049,7 @@ Possible values:
 @item rc_max_vbv_use @var{float} (@emph{encoding,video})
 @item rc_min_vbv_use @var{float} (@emph{encoding,video})
 @item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})
-
 @item color_primaries @var{integer} (@emph{decoding/encoding,video})
-Possible values:
-@table @samp
-@item bt709
-BT.709
-@item bt470m
-BT.470 M
-@item bt470bg
-BT.470 BG
-@item smpte170m
-SMPTE 170 M
-@item smpte240m
-SMPTE 240 M
-@item film
-Film
-@item bt2020
-BT.2020
-@item smpte428
-@item smpte428_1
-SMPTE ST 428-1
-@item smpte431
-SMPTE 431-2
-@item smpte432
-SMPTE 432-1
-@item jedec-p22
-JEDEC P22
-@end table
 
 @item color_trc @var{integer} (@emph{decoding/encoding,video})
 Possible values:
@@ -1055,98 +1060,35 @@ BT.709
 BT.470 M
 @item gamma28
 BT.470 BG
-@item smpte170m
-SMPTE 170 M
-@item smpte240m
-SMPTE 240 M
 @item linear
-Linear
+SMPTE 170 M
 @item log
-@item log100
-Log
+SMPTE 240 M
 @item log_sqrt
-@item log316
-Log square root
+Linear
 @item iec61966_2_4
-@item iec61966-2-4
-IEC 61966-2-4
+Log
 @item bt1361
-@item bt1361e
-BT.1361
+Log square root
 @item iec61966_2_1
-@item iec61966-2-1
-IEC 61966-2-1
-@item bt2020_10
+IEC 61966-2-4
 @item bt2020_10bit
-BT.2020 - 10 bit
-@item bt2020_12
+BT.1361
 @item bt2020_12bit
-BT.2020 - 12 bit
+IEC 61966-2-1
 @item smpte2084
-SMPTE ST 2084
-@item smpte428
+BT.2020 - 10 bit
 @item smpte428_1
-SMPTE ST 428-1
-@item arib-std-b67
-ARIB STD-B67
+BT.2020 - 12 bit
 @end table
 
 @item colorspace @var{integer} (@emph{decoding/encoding,video})
-Possible values:
-@table @samp
-@item rgb
-RGB
-@item bt709
-BT.709
-@item fcc
-FCC
-@item bt470bg
-BT.470 BG
-@item smpte170m
-SMPTE 170 M
-@item smpte240m
-SMPTE 240 M
-@item ycocg
-YCOCG
-@item bt2020nc
-@item bt2020_ncl
-BT.2020 NCL
-@item bt2020c
-@item bt2020_cl
-BT.2020 CL
-@item smpte2085
-SMPTE 2085
-@end table
 
 @item color_range @var{integer} (@emph{decoding/encoding,video})
 If used as input parameter, it serves as a hint to the decoder, which
 color_range the input has.
-Possible values:
-@table @samp
-@item tv
-@item mpeg
-MPEG (219*2^(n-8))
-@item pc
-@item jpeg
-JPEG (2^n-1)
-@end table
 
 @item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
-Possible values:
-@table @samp
-@item left
-
-@item center
-
-@item topleft
-
-@item top
-
-@item bottomleft
-
-@item bottom
-
-@end table
 
 @item log_level_offset @var{integer}
 Set the log level offset.
@@ -1225,7 +1167,7 @@ Interlaced video, top coded first, bottom displayed first
 Interlaced video, bottom coded first, top displayed first
 @end table
 
-@item skip_alpha @var{bool} (@emph{decoding,video})
+@item skip_alpha @var{integer} (@emph{decoding,video})
 Set to 1 to disable processing alpha (transparency). This works like the
 @samp{gray} flag in the @option{flags} option which skips chroma information
 instead of alpha. Default is 0.
@@ -1242,20 +1184,6 @@ ffprobe -dump_separator "
                           "  -i ~/videos/matrixbench_mpeg2.mpg
 @end example
 
-@item max_pixels @var{integer} (@emph{decoding/encoding,video})
-Maximum number of pixels per image. This value can be used to avoid out of
-memory failures due to large images.
-
-@item apply_cropping @var{bool} (@emph{decoding,video})
-Enable cropping if cropping parameters are multiples of the required
-alignment for the left and top parameters. If the alignment is not met the
-cropping will be partially applied to maintain alignment.
-Default is 1 (enabled).
-Note: The required alignment depends on if @code{AV_CODEC_FLAG_UNALIGNED} is set and the
-CPU. @code{AV_CODEC_FLAG_UNALIGNED} cannot be changed from the command line. Also hardware
-decoders will not apply left/top Cropping.
-
-
 @end table
 
 @c man end CODEC OPTIONS

doc/decoders.texi

@@ -25,6 +25,13 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.
 
+@section hevc
+
+HEVC / H.265 decoder.
+
+Note: the @option{skip_loop_filter} option has effect only at level
+@code{all}.
+
 @section rawvideo
 
 Raw video decoder.
@@ -102,7 +109,7 @@ correctly by using lavc's old buggy lpc logic for decoding.
 
 @section ffwavesynth
 
-Internal wave synthesizer.
+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
@@ -268,11 +275,11 @@ Y offset of generated bitmaps, default is 0.
 Chops leading and trailing spaces and removes empty lines from the generated
 text. This option is useful for teletext based subtitles where empty spaces may
 be present at the start or at the end of the lines or empty lines may be
-present between the subtitle lines because of double-sized teletext characters.
+present between the subtitle lines because of double-sized teletext charactes.
 Default value is 1.
 @item txt_duration
 Sets the display duration of the decoded teletext pages or subtitles in
-milliseconds. Default value is 30000 which is 30 seconds.
+miliseconds. Default value is 30000 which is 30 seconds.
 @item txt_transparent
 Force transparent background of the generated teletext bitmaps. Default value
 is 0 which means an opaque background.

doc/demuxers.texi

@@ -13,9 +13,8 @@ You can disable all the demuxers using the configure option
 the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
 with the option @code{--disable-demuxer=@var{DEMUXER}}.
 
-The option @code{-demuxers} of the ff* tools will display the list of
-enabled demuxers. Use @code{-formats} to view a combined list of
-enabled demuxers and muxers.
+The option @code{-formats} of the ff* tools will display the list of
+enabled demuxers.
 
 The description of some of the currently available demuxers follows.
 
@@ -244,33 +243,30 @@ file subdir/file-2.wav
 @end example
 @end itemize
 
-@section dash
-
-Dynamic Adaptive Streaming over HTTP demuxer.
-
-This demuxer presents all AVStreams found in the manifest.
-By setting the discard flags on AVStreams the caller can decide
-which streams to actually receive.
-Each stream mirrors the @code{id} and @code{bandwidth} properties from the
-@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
-
-@section flv, live_flv
+@section flv
 
 Adobe Flash Video Format demuxer.
 
-This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
-
-@example
-ffmpeg -f flv -i myfile.flv ...
-ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
-@end example
-
+This demuxer is used to demux FLV files and RTMP network streams.
 
 @table @option
 @item -flv_metadata @var{bool}
 Allocate the streams according to the onMetaData array content.
 @end table
 
+@section libgme
+
+The Game Music Emu library is a collection of video game music file emulators.
+
+See @url{http://code.google.com/p/game-music-emu/} for more information.
+
+Some files have multiple tracks. The demuxer will pick the first track by
+default. The @option{track_index} option can be used to select a different
+track. Track indexes start at 0. The demuxer exports the number of tracks as
+@var{tracks} meta data entry.
+
+For very large files, the @option{max_size} option may have to be adjusted.
+
 @section gif
 
 Animated GIF demuxer.
@@ -310,32 +306,6 @@ used to end the output video at the length of the shortest input file,
 which in this case is @file{input.mp4} as the GIF in this example loops
 infinitely.
 
-@section hls
-
-HLS demuxer
-
-It accepts the following options:
-
-@table @option
-@item live_start_index
-segment index to start live streams at (negative values are from the end).
-
-@item allowed_extensions
-',' separated list of file extensions that hls is allowed to access.
-
-@item max_reload
-Maximum number of times a insufficient list is attempted to be reloaded.
-Default value is 1000.
-
-@item http_persistent
-Use persistent HTTP connections. Applicable only for HTTP streams.
-Enabled by default.
-
-@item http_multiple
-Use multiple HTTP connections for downloading HTTP segments.
-Enabled by default for HTTP/1.1 servers.
-@end table
-
 @section image2
 
 Image file demuxer.
@@ -471,46 +441,6 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
 @end example
 @end itemize
 
-@section libgme
-
-The Game Music Emu library is a collection of video game music file emulators.
-
-See @url{http://code.google.com/p/game-music-emu/} for more information.
-
-Some files have multiple tracks. The demuxer will pick the first track by
-default. The @option{track_index} option can be used to select a different
-track. Track indexes start at 0. The demuxer exports the number of tracks as
-@var{tracks} meta data entry.
-
-For very large files, the @option{max_size} option may have to be adjusted.
-
-@section libopenmpt
-
-libopenmpt based module demuxer
-
-See @url{https://lib.openmpt.org/libopenmpt/} for more information.
-
-Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
-option.
-
-It accepts the following options:
-
-@table @option
-@item subsong
-Set the subsong index. This can be either  'all', 'auto', or the index of the
-subsong. Subsong indexes start at 0. The default is 'auto'.
-
-The default value is to let libopenmpt choose.
-
-@item layout
-Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
-The default value is STEREO.
-
-@item sample_rate
-Set the sample rate for libopenmpt to output.
-Range is from 1000 to INT_MAX. The value default is 48000.
-@end table
-
 @section mov/mp4/3gp/QuickTime
 
 QuickTime / MP4 demuxer.

doc/developer.texi

@@ -10,7 +10,9 @@
 
 @contents
 
-@chapter Notes for external developers
+@chapter Developers Guide
+
+@section Notes for external developers
 
 This document is mostly useful for internal FFmpeg developers.
 External developers who need to use the API in their application should
@@ -28,13 +30,15 @@ For more detailed legal information about the use of FFmpeg in
 external programs read the @file{LICENSE} file in the source tree and
 consult @url{https://ffmpeg.org/legal.html}.
 
-@chapter Contributing
+@section Contributing
 
-There are 2 ways by which code gets into FFmpeg:
+There are 3 ways by which code gets into FFmpeg.
 @itemize @bullet
-@item Submitting patches to the ffmpeg-devel mailing list.
+@item Submitting patches to the main developer mailing list.
       See @ref{Submitting patches} for details.
 @item Directly committing changes to the main tree.
+@item Committing changes to a git clone, for example on github.com or
+      gitorious.org. And asking us to merge these changes.
 @end itemize
 
 Whichever way, changes should be reviewed by the maintainer of the code
@@ -43,9 +47,9 @@ The developer making the commit and the author are responsible for their changes
 and should try to fix issues their commit causes.
 
 @anchor{Coding Rules}
-@chapter Coding Rules
+@section Coding Rules
 
-@section Code formatting conventions
+@subsection Code formatting conventions
 
 There are the following guidelines regarding the indentation in files:
 
@@ -70,7 +74,7 @@ The presentation is one inspired by 'indent -i4 -kr -nut'.
 The main priority in FFmpeg is simplicity and small code size in order to
 minimize the bug count.
 
-@section Comments
+@subsection Comments
 Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
 can be generated automatically. All nontrivial functions should have a comment
 above them explaining what the function does, even if it is just one sentence.
@@ -110,7 +114,7 @@ int myfunc(int my_parameter)
 ...
 @end example
 
-@section C language features
+@subsection C language features
 
 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
@@ -127,11 +131,6 @@ designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
 
 @item
 compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
-
-@item
-Implementation defined behavior for signed integers is assumed to match the
-expected behavior for two's complement. Non representable values in integer
-casts are binary truncated. Shift right of signed values uses sign extension.
 @end itemize
 
 These features are supported by all compilers we care about, so we will not
@@ -156,7 +155,7 @@ mixing statements and declarations;
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
 @end itemize
 
-@section Naming conventions
+@subsection Naming conventions
 All names should be composed with underscores (_), not CamelCase. For example,
 @samp{avfilter_get_video_buffer} is an acceptable function name and
 @samp{AVFilterGetVideo} is not. The exception from this are type names, like
@@ -180,7 +179,7 @@ e.g. @samp{ff_w64_demuxer}.
 @item
 For variables and functions visible outside of file scope, used internally
 across multiple libraries, use @code{avpriv_} as prefix, for example,
-@samp{avpriv_report_missing_feature}.
+@samp{avpriv_aac_parse_header}.
 
 @item
 Each library has its own prefix for public symbols, in addition to the
@@ -200,7 +199,7 @@ letter as they are reserved by the C standard. Names starting with @code{_}
 are reserved at the file level and may not be used for externally visible
 symbols. If in doubt, just avoid names starting with @code{_} altogether.
 
-@section Miscellaneous conventions
+@subsection Miscellaneous conventions
 
 @itemize @bullet
 @item
@@ -212,7 +211,7 @@ Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
 @end itemize
 
-@section Editor configuration
+@subsection Editor configuration
 In order to configure Vim to follow FFmpeg formatting conventions, paste
 the following snippet into your @file{.vimrc}:
 @example
@@ -245,10 +244,10 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 (setq c-default-style "ffmpeg")
 @end lisp
 
-@chapter Development Policy
+@section Development Policy
 
-@section Patches/Committing
-@subheading Licenses for patches must be compatible with FFmpeg.
+@enumerate
+@item
 Contributions should be licensed under the
 @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
 including an "or any later version" clause, or, if you prefer
@@ -261,15 +260,15 @@ preferred.
 If you add a new file, give it a proper license header. Do not copy and
 paste it from a random place, use an existing file as template.
 
-@subheading You must not commit code which breaks FFmpeg!
-This means unfinished code which is enabled and breaks compilation,
-or compiles but does not work/breaks the regression tests. Code which
-is unfinished but disabled may be permitted under-circumstances, like
-missing samples or an implementation with a small subset of features.
-Always check the mailing list for any reviewers with issues and test
-FATE before you push.
+@item
+You must not commit code which breaks FFmpeg! (Meaning unfinished but
+enabled code which breaks compilation or compiles but does not work or
+breaks the regression tests)
+You can commit unfinished stuff (for testing etc), but it must be disabled
+(#ifdef etc) by default so it does not interfere with other developers'
+work.
 
-@subheading Keep the main commit message short with an extended description below.
+@item
 The commit message should have a short first line in the form of
 a @samp{topic: short description} as a header, separated by a newline
 from the body consisting of an explanation of why the change is necessary.
@@ -277,24 +276,30 @@ If the commit fixes a known bug on the bug tracker, the commit message
 should include its bug ID. Referring to the issue on the bug tracker does
 not exempt you from writing an excerpt of the bug in the commit message.
 
-@subheading Testing must be adequate but not excessive.
-If it works for you, others, and passes FATE then it should be OK to commit
-it, provided it fits the other committing criteria. You should not worry about
-over-testing things. If your code has problems (portability, triggers
-compiler bugs, unusual environment etc) they will be reported and eventually
-fixed.
+@item
+You do not have to over-test things. If it works for you, and you think it
+should work for others, then commit. If your code has problems
+(portability, triggers compiler bugs, unusual environment etc) they will be
+reported and eventually fixed.
 
-@subheading Do not commit unrelated changes together.
-They should be split them into self-contained pieces. Also do not forget
-that if part B depends on part A, but A does not depend on B, then A can
-and should be committed first and separate from B. Keeping changes well
-split into self-contained parts makes reviewing and understanding them on
-the commit log mailing list easier. This also helps in case of debugging
-later on.
+@item
+Do not commit unrelated changes together, split them into self-contained
+pieces. Also do not forget that if part B depends on part A, but A does not
+depend on B, then A can and should be committed first and separate from B.
+Keeping changes well split into self-contained parts makes reviewing and
+understanding them on the commit log mailing list easier. This also helps
+in case of debugging later on.
 Also if you have doubts about splitting or not splitting, do not hesitate to
 ask/discuss it on the developer mailing list.
 
-@subheading Ask before you change the build system (configure, etc).
+@item
+Do not change behavior of the programs (renaming options etc) or public
+API or ABI without first discussing it on the ffmpeg-devel mailing list.
+Do not remove functionality from the code. Just improve!
+
+Note: Redundant code can be removed.
+
+@item
 Do not commit changes to the build system (Makefiles, configure script)
 which change behavior, defaults etc, without asking first. The same
 applies to compiler warning fixes, trivial looking fixes and to code
@@ -303,7 +308,7 @@ the way we do. Send your changes as patches to the ffmpeg-devel mailing
 list, and if the code maintainers say OK, you may commit. This does not
 apply to files you wrote and/or maintain.
 
-@subheading Cosmetic changes should be kept in separate patches.
+@item
 We refuse source indentation and other cosmetic changes if they are mixed
 with functional changes, such commits will be rejected and removed. Every
 developer has his own indentation style, you should not change it. Of course
@@ -317,7 +322,7 @@ NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
 then either do NOT change the indentation of the inner part within (do not
 move it to the right)! or do so in a separate commit
 
-@subheading Commit messages should always be filled out properly.
+@item
 Always fill out the commit log message. Describe in a few lines what you
 changed and why. You can refer to mailing list postings if you fix a
 particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
@@ -329,31 +334,47 @@ area changed: Short 1 line description
 details describing what and why and giving references.
 @end example
 
-@subheading Credit the author of the patch.
+@item
 Make sure the author of the commit is set correctly. (see git commit --author)
 If you apply a patch, send an
 answer to ffmpeg-devel (or wherever you got the patch from) saying that
 you applied the patch.
 
-@subheading Complex patches should refer to discussion surrounding them.
+@item
 When applying patches that have been discussed (at length) on the mailing
 list, reference the thread in the log message.
 
-@subheading Always wait long enough before pushing changes
+@item
 Do NOT commit to code actively maintained by others without permission.
-Send a patch to ffmpeg-devel. If no one answers within a reasonable
-time-frame (12h for build failures and security fixes, 3 days small changes,
+Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+timeframe (12h for build failures and security fixes, 3 days small changes,
 1 week for big patches) then commit your patch if you think it is OK.
 Also note, the maintainer can simply ask for more time to review!
 
-@section Code
-@subheading API/ABI changes should be discussed before they are made.
-Do not change behavior of the programs (renaming options etc) or public
-API or ABI without first discussing it on the ffmpeg-devel mailing list.
-Do not remove widely used functionality or features (redundant code can be removed).
+@item
+Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+are sent there and reviewed by all the other developers. Bugs and possible
+improvements or general questions regarding commits are discussed there. We
+expect you to react if problems with your code are uncovered.
 
-@subheading Remember to check if you need to bump versions for libav*.
-Depending on the change, you may need to change the version integer.
+@item
+Update the documentation if you change behavior or add features. If you are
+unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+maintainer(s) will review and commit your stuff.
+
+@item
+Try to keep important discussions and requests (also) on the public
+developer mailing list, so that all developers can benefit from them.
+
+@item
+Never write to unallocated memory, never write over the end of arrays,
+always check values read from some untrusted source before using them
+as array index or other risky things.
+
+@item
+Remember to check if you need to bump versions for the specific libav*
+parts (libavutil, libavcodec, libavformat) you are changing. You need
+to change the version integer.
 Incrementing the first component means no backward compatibility to
 previous versions (e.g. removal of a function from the public API).
 Incrementing the second component means backward compatible change
@@ -363,7 +384,7 @@ Incrementing the third component means a noteworthy binary compatible
 change (e.g. encoder bug fix that matters for the decoder). The third
 component always starts at 100 to distinguish FFmpeg from Libav.
 
-@subheading Warnings for correct code may be disabled if there is no other option.
+@item
 Compiler warnings indicate potential bugs or code with bad style. If a type of
 warning always points to correct and clean code, that warning should
 be disabled, not the code changed.
@@ -372,54 +393,17 @@ If it is a bug, the bug has to be fixed. If it is not, the code should
 be changed to not generate a warning unless that causes a slowdown
 or obfuscates the code.
 
-@subheading Check untrusted input properly.
-Never write to unallocated memory, never write over the end of arrays,
-always check values read from some untrusted source before using them
-as array index or other risky things.
-
-@section Documentation/Other
-@subheading Subscribe to the ffmpeg-devel mailing list.
-It is important to be subscribed to the
-@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
-mailing list. Almost any non-trivial patch is to be sent there for review.
-Other developers may have comments about your contribution. We expect you see
-those comments, and to improve it if requested. (N.B. Experienced committers
-have other channels, and may sometimes skip review for trivial fixes.) Also,
-discussion here about bug fixes and FFmpeg improvements by other developers may
-be helpful information for you. Finally, by being a list subscriber, your
-contribution will be posted immediately to the list, without the moderation
-hold which messages from non-subscribers experience.
-
-However, it is more important to the project that we receive your patch than
-that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
-want to subscribe and discuss the patch, then please do send it to the list
-anyway.
-
-@subheading Subscribe to the ffmpeg-cvslog mailing list.
-Diffs of all commits are sent to the
-@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
-mailing list. Some developers read this list to review all code base changes
-from all sources. Subscribing to this list is not mandatory.
-
-@subheading Keep the documentation up to date.
-Update the documentation if you change behavior or add features. If you are
-unsure how best to do this, send a patch to ffmpeg-devel, the documentation
-maintainer(s) will review and commit your stuff.
-
-@subheading Important discussions should be accessible to all.
-Try to keep important discussions and requests (also) on the public
-developer mailing list, so that all developers can benefit from them.
-
-@subheading Check your entries in MAINTAINERS.
+@item
 Make sure that no parts of the codebase that you maintain are missing from the
 @file{MAINTAINERS} file. If something that you want to maintain is missing add it with
 your name after it.
 If at some point you no longer want to maintain some code, then please help in
 finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
+@end enumerate
 
 We think our rules are not too hard. If you have comments, contact us.
 
-@chapter Code of conduct
+@section Code of conduct
 
 Be friendly and respectful towards others and third parties.
 Treat others the way you yourself want to be treated.
@@ -449,7 +433,7 @@ Finally, keep in mind the immortal words of Bill and Ted,
 "Be excellent to each other."
 
 @anchor{Submitting patches}
-@chapter Submitting patches
+@section Submitting patches
 
 First, read the @ref{Coding Rules} above if you did not yet, in particular
 the rules regarding patch submission.
@@ -482,11 +466,7 @@ Patches should be posted to the
 mailing list. Use @code{git send-email} when possible since it will properly
 send patches without requiring extra care. If you cannot, then send patches
 as base64-encoded attachments, so your patch is not trashed during
-transmission. Also ensure the correct mime type is used
-(text/x-diff or text/x-patch or at least text/plain) and that only one
-patch is inline or attached per mail.
-You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
-likely was wrong.
+transmission.
 
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
@@ -498,7 +478,7 @@ Give us a few days to react. But if some time passes without reaction,
 send a reminder by email. Your patch should eventually be dealt with.
 
 
-@chapter New codecs or formats checklist
+@section New codecs or formats checklist
 
 @enumerate
 @item
@@ -550,7 +530,7 @@ Did you make sure it compiles standalone, i.e. with
 @end enumerate
 
 
-@chapter Patch submission checklist
+@section patch submission checklist
 
 @enumerate
 @item
@@ -560,9 +540,9 @@ Does @code{make fate} pass with the patch applied?
 Was the patch generated with git format-patch or send-email?
 
 @item
-Did you sign-off your patch? (@code{git commit -s})
-See @uref{https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst, Sign your work} for the meaning
-of @dfn{sign-off}.
+Did you sign off your patch? (git commit -s)
+See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+of sign off.
 
 @item
 Did you provide a clear git commit log message?
@@ -663,7 +643,7 @@ Test your code with valgrind and or Address Sanitizer to ensure it's free
 of leaks, out of array accesses, etc.
 @end enumerate
 
-@chapter Patch review process
+@section Patch review process
 
 All patches posted to ffmpeg-devel will be reviewed, unless they contain a
 clear note that the patch is not for the git master branch.
@@ -694,7 +674,7 @@ to be reviewed, please consider helping to review other patches, that is a great
 way to get everyone's patches reviewed sooner.
 
 @anchor{Regression tests}
-@chapter 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.
@@ -705,7 +685,7 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
 this case, the reference results of the regression tests shall be modified
 accordingly].
 
-@section Adding files to the fate-suite dataset
+@subsection Adding files to the fate-suite dataset
 
 When there is no muxer or encoder available to generate test media for a
 specific test then the media has to be included in the fate-suite.
@@ -716,7 +696,7 @@ Once you have a working fate test and fate sample, provide in the commit
 message or introductory message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.
 
-@section Visualizing Test Coverage
+@subsection Visualizing Test Coverage
 
 The FFmpeg build system allows visualizing the test coverage in an easy
 manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
@@ -743,7 +723,7 @@ You can use the command @code{make lcov-reset} to reset the coverage
 measurements. You will need to rerun @code{make lcov} after running a
 new test.
 
-@section Using Valgrind
+@subsection Using Valgrind
 
 The configure script provides a shortcut for using valgrind to spot bugs
 related to memory handling. Just add the option
@@ -757,7 +737,7 @@ In case you need finer control over how valgrind is invoked, use the
 your configure line instead.
 
 @anchor{Release process}
-@chapter Release process
+@section Release process
 
 FFmpeg maintains a set of @strong{release branches}, which are the
 recommended deliverable for system integrators and distributors (such as
@@ -789,7 +769,7 @@ adjustments to the symbol versioning file. Please discuss such changes
 on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
 
 @anchor{Criteria for Point Releases}
-@section Criteria for Point Releases
+@subsection Criteria for Point Releases
 
 Changes that match the following criteria are valid candidates for
 inclusion into a point release:
@@ -813,7 +793,7 @@ point releases of the same release branch.
 The order for checking the rules is (1 OR 2 OR 3) AND 4.
 
 
-@section Release Checklist
+@subsection Release Checklist
 
 The release process involves the following steps:
 

doc/encoders.texi

@@ -61,9 +61,9 @@ Two loop searching (TLS) method.
 
 This method first sets quantizers depending on band thresholds and then tries
 to find an optimal combination by adding or subtracting a specific value from
-all quantizers and adjusting some individual quantizer a little.  Will tune
-itself based on whether @option{aac_is}, @option{aac_ms} and @option{aac_pns}
-are enabled.
+all quantizers and adjusting some individual quantizer a little.
+Will tune itself based on whether aac_is/aac_ms/aac_pns are enabled.
+This is the default choice for a coder.
 
 @item anmr
 Average noise to mask ratio (ANMR) trellis-based solution.
@@ -76,27 +76,27 @@ Not currently recommended.
 @item fast
 Constant quantizer method.
 
-Uses a cheaper version of twoloop algorithm that doesn't try to do as many
-clever adjustments. Worse with low bitrates (less than 64kbps), but is better
-and much faster at higher bitrates.
-This is the default choice for a coder
+This method sets a constant quantizer for all bands. This is the fastest of all
+the methods and has no rate control or support for @option{aac_is} or
+@option{aac_pns}.
+Not recommended.
 
 @end table
 
 @item aac_ms
-Sets mid/side coding mode. The default value of "auto" will automatically use
+Sets mid/side coding mode. The default value of auto will automatically use
 M/S with bands which will benefit from such coding. Can be forced for all bands
 using the value "enable", which is mainly useful for debugging or disabled using
 "disable".
 
 @item aac_is
 Sets intensity stereo coding tool usage. By default, it's enabled and will
-automatically toggle IS for similar pairs of stereo bands if it's beneficial.
+automatically toggle IS for similar pairs of stereo bands if it's benefitial.
 Can be disabled for debugging by setting the value to "disable".
 
 @item aac_pns
 Uses perceptual noise substitution to replace low entropy high frequency bands
-with imperceptible white noise during the decoding process. By default, it's
+with imperceivable white noise during the decoding process. By default, it's
 enabled, but can be disabled for debugging purposes by using "disable".
 
 @item aac_tns
@@ -130,20 +130,20 @@ The default, AAC "Low-complexity" profile. Is the most compatible and produces
 decent quality.
 
 @item mpeg2_aac_low
-Equivalent to @code{-profile:a aac_low -aac_pns 0}. PNS was introduced with the
-MPEG4 specifications.
+Equivalent to -profile:a aac_low -aac_pns 0. PNS was introduced with the MPEG4
+specifications.
 
 @item aac_ltp
-Long term prediction profile, is enabled by and will enable the @option{aac_ltp}
-option. Introduced in MPEG4.
+Long term prediction profile, is enabled by and will enable the aac_ltp option.
+Introduced in MPEG4.
 
 @item aac_main
-Main-type prediction profile, is enabled by and will enable the @option{aac_pred}
-option. Introduced in MPEG2.
+Main-type prediction profile, is enabled by and will enable the aac_pred option.
+Introduced in MPEG2.
 
-@end table
 If this option is unspecified it is set to @samp{aac_low}.
 @end table
+@end table
 
 @section ac3 and ac3_fixed
 
@@ -487,10 +487,6 @@ is an optional AC-3 feature that increases quality by selectively encoding
 the left/right channels as mid/side. This option is enabled by default, and it
 is highly recommended that it be left as enabled except for testing purposes.
 
-@item cutoff @var{frequency}
-Set lowpass cutoff frequency. If unspecified, the encoder selects a default
-determined by various other encoding parameters.
-
 @end table
 
 @subsection Floating-Point-Only AC-3 Encoding Options
@@ -548,8 +544,7 @@ The following options are supported by FFmpeg's flac encoder.
 @table @option
 @item compression_level
 Sets the compression level, which chooses defaults for many other options
-if they are not set explicitly. Valid values are from 0 to 12, 5 is the
-default.
+if they are not set explicitly.
 
 @item frame_size
 Sets the size of the frames in samples per channel.
@@ -598,7 +593,7 @@ Channel mode
 @item auto
 The mode is chosen automatically for each frame
 @item indep
-Channels are independently coded
+Chanels are independently coded
 @item left_side
 @item right_side
 @item mid_side
@@ -616,27 +611,111 @@ and slightly improves compression.
 
 @end table
 
-@anchor{opusenc}
-@section opus
+@anchor{libfaac}
+@section libfaac
 
-Opus encoder.
+libfaac AAC (Advanced Audio Coding) encoder wrapper.
 
-This is a native FFmpeg encoder for the Opus format. Currently its in development and
-only implements the CELT part of the codec. Its quality is usually worse and at best
-is equal to the libopus encoder.
+This encoder is of much lower quality and is more unstable than any other AAC
+encoders, so it's highly recommended to instead use other encoders, like
+@ref{aacenc,,the native FFmpeg AAC encoder}.
+
+This encoder also requires the presence of the libfaac headers and library
+during configuration. You need to explicitly configure the build with
+@code{--enable-libfaac --enable-nonfree}.
 
 @subsection Options
 
-@table @option
-@item b
-Set bit rate in bits/s. If unspecified it uses the number of channels and the layout
-to make a good guess.
+The following shared FFmpeg codec options are recognized.
 
-@item opus_delay
-Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly
-decrease quality.
+The following options are supported by the libfaac wrapper. The
+@command{faac}-equivalent of the options are listed in parentheses.
+
+@table @option
+@item b (@emph{-b})
+Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
+is not explicitly specified, it is automatically set to a suitable
+value depending on the selected profile. @command{faac} bitrate is
+expressed in kilobits/s.
+
+Note that libfaac does not support CBR (Constant Bit Rate) but only
+ABR (Average Bit Rate).
+
+If VBR mode is enabled this option is ignored.
+
+@item ar (@emph{-R})
+Set audio sampling rate (in Hz).
+
+@item ac (@emph{-c})
+Set the number of audio channels.
+
+@item cutoff (@emph{-C})
+Set cutoff frequency. If not specified (or explicitly set to 0) it
+will use a value automatically computed by the library. Default value
+is 0.
+
+@item profile
+Set audio profile.
+
+The following profiles are recognized:
+@table @samp
+@item aac_main
+Main AAC (Main)
+
+@item aac_low
+Low Complexity AAC (LC)
+
+@item aac_ssr
+Scalable Sample Rate (SSR)
+
+@item aac_ltp
+Long Term Prediction (LTP)
 @end table
 
+If not specified it is set to @samp{aac_low}.
+
+@item flags +qscale
+Set constant quality VBR (Variable Bit Rate) mode.
+
+@item global_quality
+Set quality in VBR mode as an integer number of lambda units.
+
+Only relevant when VBR mode is enabled with @code{flags +qscale}.  The
+value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
+and used to set the quality value used by libfaac. A reasonable range
+for the option value in QP units is [10-500], the higher the value the
+higher the quality.
+
+@item q (@emph{-q})
+Enable VBR mode when set to a non-negative value, and set constant
+quality value as a double floating point value in QP units.
+
+The value sets the quality value used by libfaac. A reasonable range
+for the option value is [10-500], the higher the value the higher the
+quality.
+
+This option is valid only using the @command{ffmpeg} command-line
+tool. For library interface users, use @option{global_quality}.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
+container:
+@example
+ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
+@end example
+
+@item
+Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
+LTP AAC profile:
+@example
+ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
+@end example
+@end itemize
+
 @anchor{libfdk-aac-enc}
 @section libfdk_aac
 
@@ -842,10 +921,6 @@ Set algorithm quality. Valid arguments are integers in the 0-9 range,
 with 0 meaning highest quality but slowest, and 9 meaning fastest
 while producing the worst quality.
 
-@item cutoff (@emph{--lowpass})
-Set lowpass cutoff frequency. If unspecified, the encoder dynamically
-adjusts the cutoff.
-
 @item reservoir
 Enable use of bit reservoir when set to 1. Default value is 1. LAME
 has this enabled by default, but can be overridden by use
@@ -899,95 +974,6 @@ default value is 0 (disabled).
 
 @end table
 
-@section libopus
-
-libopus Opus Interactive Audio Codec encoder wrapper.
-
-Requires the presence of the libopus headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libopus}.
-
-@subsection Option Mapping
-
-Most libopus options are modelled after the @command{opusenc} utility from
-opus-tools. The following is an option mapping chart describing options
-supported by the libopus wrapper, and their @command{opusenc}-equivalent
-in parentheses.
-
-@table @option
-
-@item b (@emph{bitrate})
-Set the bit rate in bits/s.  FFmpeg's @option{b} option is
-expressed in bits/s, while @command{opusenc}'s @option{bitrate} in
-kilobits/s.
-
-@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr})
-Set VBR mode. The FFmpeg @option{vbr} option has the following
-valid arguments, with the @command{opusenc} equivalent options
-in parentheses:
-
-@table @samp
-@item off (@emph{hard-cbr})
-Use constant bit rate encoding.
-
-@item on (@emph{vbr})
-Use variable bit rate encoding (the default).
-
-@item constrained (@emph{cvbr})
-Use constrained variable bit rate encoding.
-@end table
-
-@item compression_level (@emph{comp})
-Set encoding algorithm complexity. Valid options are integers in
-the 0-10 range. 0 gives the fastest encodes but lower quality, while 10
-gives the highest quality but slowest encoding. The default is 10.
-
-@item frame_duration (@emph{framesize})
-Set maximum frame size, or duration of a frame in milliseconds. The
-argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
-frame sizes achieve lower latency but less quality at a given bitrate.
-Sizes greater than 20ms are only interesting at fairly low bitrates.
-The default is 20ms.
-
-@item packet_loss (@emph{expect-loss})
-Set expected packet loss percentage. The default is 0.
-
-@item application (N.A.)
-Set intended application type. Valid options are listed below:
-
-@table @samp
-@item voip
-Favor improved speech intelligibility.
-@item audio
-Favor faithfulness to the input (the default).
-@item lowdelay
-Restrict to only the lowest delay modes.
-@end table
-
-@item cutoff (N.A.)
-Set cutoff bandwidth in Hz. The argument must be exactly one of the
-following: 4000, 6000, 8000, 12000, or 20000, corresponding to
-narrowband, mediumband, wideband, super wideband, and fullband
-respectively. The default is 0 (cutoff disabled).
-
-@item mapping_family (@emph{mapping_family})
-Set channel mapping family to be used by the encoder. The default value of -1
-uses mapping family 0 for mono and stereo inputs, and mapping family 1
-otherwise. The default also disables the surround masking and LFE bandwidth
-optimzations in libopus, and requires that the input contains 8 channels or
-fewer.
-
-Other values include 0 for mono and stereo, 1 for surround sound with masking
-and LFE bandwidth optimizations, and 255 for independent streams with an
-unspecified channel layout.
-
-@item apply_phase_inv (N.A.) (requires libopus >= 1.2)
-If set to 0, disables the use of phase inversion for intensity stereo,
-improving the quality of mono downmixes, but slightly reducing normal stereo
-quality. The default is 1 (phase inversion enabled).
-
-@end table
-
 @anchor{libshine}
 @section libshine
 
@@ -1127,6 +1113,79 @@ default value is 0 (disabled).
 
 @end table
 
+@section libopus
+
+libopus Opus Interactive Audio Codec encoder wrapper.
+
+Requires the presence of the libopus headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopus}.
+
+@subsection Option Mapping
+
+Most libopus options are modelled after the @command{opusenc} utility from
+opus-tools. The following is an option mapping chart describing options
+supported by the libopus wrapper, and their @command{opusenc}-equivalent
+in parentheses.
+
+@table @option
+
+@item b (@emph{bitrate})
+Set the bit rate in bits/s.  FFmpeg's @option{b} option is
+expressed in bits/s, while @command{opusenc}'s @option{bitrate} in
+kilobits/s.
+
+@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr})
+Set VBR mode. The FFmpeg @option{vbr} option has the following
+valid arguments, with the @command{opusenc} equivalent options
+in parentheses:
+
+@table @samp
+@item off (@emph{hard-cbr})
+Use constant bit rate encoding.
+
+@item on (@emph{vbr})
+Use variable bit rate encoding (the default).
+
+@item constrained (@emph{cvbr})
+Use constrained variable bit rate encoding.
+@end table
+
+@item compression_level (@emph{comp})
+Set encoding algorithm complexity. Valid options are integers in
+the 0-10 range. 0 gives the fastest encodes but lower quality, while 10
+gives the highest quality but slowest encoding. The default is 10.
+
+@item frame_duration (@emph{framesize})
+Set maximum frame size, or duration of a frame in milliseconds. The
+argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
+frame sizes achieve lower latency but less quality at a given bitrate.
+Sizes greater than 20ms are only interesting at fairly low bitrates.
+The default is 20ms.
+
+@item packet_loss (@emph{expect-loss})
+Set expected packet loss percentage. The default is 0.
+
+@item application (N.A.)
+Set intended application type. Valid options are listed below:
+
+@table @samp
+@item voip
+Favor improved speech intelligibility.
+@item audio
+Favor faithfulness to the input (the default).
+@item lowdelay
+Restrict to only the lowest delay modes.
+@end table
+
+@item cutoff (N.A.)
+Set cutoff bandwidth in Hz. The argument must be exactly one of the
+following: 4000, 6000, 8000, 12000, or 20000, corresponding to
+narrowband, mediumband, wideband, super wideband, and fullband
+respectively. The default is 0 (cutoff disabled).
+
+@end table
+
 @section libvorbis
 
 libvorbis encoder wrapper.
@@ -1226,27 +1285,6 @@ Same as @samp{3}, but with extra processing enabled.
 @end table
 @end table
 
-@anchor{mjpegenc}
-@section mjpeg
-
-Motion JPEG encoder.
-
-@subsection Options
-
-@table @option
-@item huffman
-Set the huffman encoding strategy. Possible values:
-
-@table @samp
-@item default
-Use the default huffman tables. This is the default strategy.
-
-@item optimal
-Compute and use optimal huffman tables.
-
-@end table
-@end table
-
 @anchor{wavpackenc}
 @section wavpack
 
@@ -1316,81 +1354,6 @@ disabled
 A description of some of the currently available video encoders
 follows.
 
-@section Hap
-
-Vidvox Hap video encoder.
-
-@subsection Options
-
-@table @option
-@item format @var{integer}
-Specifies the Hap format to encode.
-
-@table @option
-@item hap
-@item hap_alpha
-@item hap_q
-@end table
-
-Default value is @option{hap}.
-
-@item chunks @var{integer}
-Specifies the number of chunks to split frames into, between 1 and 64. This
-permits multithreaded decoding of large frames, potentially at the cost of
-data-rate. The encoder may modify this value to divide frames evenly.
-
-Default value is @var{1}.
-
-@item compressor @var{integer}
-Specifies the second-stage compressor to use. If set to @option{none},
-@option{chunks} will be limited to 1, as chunked uncompressed frames offer no
-benefit.
-
-@table @option
-@item none
-@item snappy
-@end table
-
-Default value is @option{snappy}.
-
-@end table
-
-@section jpeg2000
-
-The native jpeg 2000 encoder is lossy by default, the @code{-q:v}
-option can be used to set the encoding quality. Lossless encoding
-can be selected with @code{-pred 1}.
-
-@subsection Options
-
-@table @option
-@item format
-Can be set to either @code{j2k} or @code{jp2} (the default) that
-makes it possible to store non-rgb pix_fmts.
-
-@end table
-
-@section libkvazaar
-
-Kvazaar H.265/HEVC encoder.
-
-Requires the presence of the libkvazaar headers and library during
-configuration. You need to explicitly configure the build with
-@option{--enable-libkvazaar}.
-
-@subsection Options
-
-@table @option
-
-@item b
-Set target video bitrate in bit/s and enable rate control.
-
-@item kvazaar-params
-Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
-by commas (,). See kvazaar documentation for a list of options.
-
-@end table
-
 @section libopenh264
 
 Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.
@@ -1427,7 +1390,7 @@ is 0. This is only used when @option{slice_mode} is set to
 @samp{fixed}.
 
 @item slice_mode
-Set slice mode. Can assume one of the following possible values:
+Set slice mode. Can assume one of the follwing possible values:
 
 @table @samp
 @item fixed
@@ -1457,6 +1420,30 @@ Set maximum NAL size in bytes.
 Allow skipping frames to hit the target bitrate if set to 1.
 @end table
 
+@section jpeg2000
+
+The native jpeg 2000 encoder is lossy by default, the @code{-q:v}
+option can be used to set the encoding quality. Lossless encoding
+can be selected with @code{-pred 1}.
+
+@subsection Options
+
+@table @option
+@item format
+Can be set to either @code{j2k} or @code{jp2} (the default) that
+makes it possible to store non-rgb pix_fmts.
+
+@end table
+
+@section snow
+
+@subsection Options
+
+@table @option
+@item iterative_dia_size
+dia size for the iterative motion estimation
+@end table
+
 @section libtheora
 
 libtheora Theora encoder wrapper.
@@ -1670,7 +1657,7 @@ option to 2.
 Enable frame parallel decodability features.
 @item aq-mode
 Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
-cyclic refresh, 4: equator360).
+cyclic refresh).
 @item colorspace @emph{color-space}
 Set input color space. The VP9 bitstream supports signaling the following
 colorspaces:
@@ -1683,15 +1670,6 @@ colorspaces:
 @item @samp{smpte240m} @emph{smpte240}
 @item @samp{bt2020_ncl} @emph{bt2020}
 @end table
-@item row-mt @var{boolean}
-Enable row based multi-threading.
-@item tune-content
-Set content type: default (0), screen (1), film (2).
-@item corpus-complexity
-Corpus VBR mode is a variant of standard VBR where the complexity distribution
-midpoint is passed in rather than calculated for a specific clip or chunk.
-
-The valid range is [0, 10000]. 0 (default) uses standard VBR.
 @end table
 
 @end table
@@ -1699,6 +1677,7 @@ The valid range is [0, 10000]. 0 (default) uses standard VBR.
 For more information about libvpx see:
 @url{http://www.webmproject.org/}
 
+
 @section libwebp
 
 libwebp WebP Image encoder wrapper
@@ -1804,7 +1783,7 @@ the documentation of the undocumented generic options, see
 @ref{codec-options,,the Codec Options chapter}.
 
 To get a more accurate and extensive documentation of the libx264
-options, invoke the command @command{x264 --fullhelp} or consult
+options, invoke the command @command{x264 --full-help} or consult
 the libx264 documentation.
 
 @table @option
@@ -1865,10 +1844,6 @@ Exhaustive search.
 Hadamard exhaustive search (slowest).
 @end table
 
-@item forced-idr
-Normally, when forcing a I-frame type, the encoder can select any type
-of I-frame. This option forces it to choose an IDR-frame.
-
 @item subq (@emph{subme})
 Sub-pixel motion estimation method.
 
@@ -1891,7 +1866,7 @@ Enable CAVLC and disable CABAC. It generates the same effect as
 @end table
 
 @item cmp
-Set full pixel motion estimation comparison algorithm. Possible values:
+Set full pixel motion estimation comparation algorithm. Possible values:
 
 @table @samp
 @item chroma
@@ -2117,12 +2092,12 @@ is kept undocumented for some reason.
 
 For example to specify libx264 encoding options with @command{ffmpeg}:
 @example
-ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
+ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
 @end example
 
 @item a53cc @var{boolean}
 Import closed captions (which must be ATSC compatible format) into output.
-Only the mpeg2 and h264 decoders provide these. Default is 1 (on).
+Only the mpeg2 and h264 decoders provide these. Default is 0 (off).
 
 @item x264-params (N.A.)
 Override the x264 configuration using a :-separated list of key=value
@@ -2159,16 +2134,6 @@ Set the x265 preset.
 @item tune
 Set the x265 tune parameter.
 
-@item profile
-Set profile restrictions.
-
-@item crf
-Set the quality for constant quality mode.
-
-@item forced-idr
-Normally, when forcing a I-frame type, the encoder can select any type
-of I-frame. This option forces it to choose an IDR-frame.
-
 @item x265-params
 Set x265 options using a list of @var{key}=@var{value} couples separated
 by ":". See @command{x265 --help} for a list of options.
@@ -2365,11 +2330,6 @@ Never write it.
 @itemx always
 Always write it.
 @end table
-@item video_format @var{integer}
-Specifies the video_format written into the sequence display extension
-indicating the source of the video pictures. The default is @samp{unspecified},
-can be @samp{component}, @samp{pal}, @samp{ntsc}, @samp{secam} or @samp{mac}.
-For maximum compatibility, use @samp{component}.
 @end table
 
 @section png
@@ -2403,7 +2363,6 @@ Select the ProRes profile to encode
 @item standard
 @item hq
 @item 4444
-@item 4444xq
 @end table
 
 @item quant_mat @var{integer}
@@ -2453,6 +2412,27 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
 For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.
 
+@section libkvazaar
+
+Kvazaar H.265/HEVC encoder.
+
+Requires the presence of the libkvazaar headers and library during
+configuration. You need to explicitly configure the build with
+@option{--enable-libkvazaar}.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set target video bitrate in bit/s and enable rate control.
+
+@item kvazaar-params
+Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
+by commas (,). See kvazaar documentation for a list of options.
+
+@end table
+
 @section QSV encoders
 
 The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)
@@ -2541,126 +2521,6 @@ encoder use CAVLC instead of CABAC.
 
 @end itemize
 
-@section snow
-
-@subsection Options
-
-@table @option
-@item iterative_dia_size
-dia size for the iterative motion estimation
-@end table
-
-@section VAAPI encoders
-
-Wrappers for hardware encoders accessible via VAAPI.
-
-These encoders only accept input in VAAPI hardware surfaces.  If you have input
-in software frames, use the @option{hwupload} filter to upload them to the GPU.
-
-The following standard libavcodec options are used:
-@itemize
-@item
-@option{g} / @option{gop_size}
-@item
-@option{bf} / @option{max_b_frames}
-@item
-@option{profile}
-@item
-@option{level}
-@item
-@option{b} / @option{bit_rate}
-@item
-@option{maxrate} / @option{rc_max_rate}
-@item
-@option{bufsize} / @option{rc_buffer_size}
-@item
-@option{rc_init_occupancy} / @option{rc_initial_buffer_occupancy}
-@item
-@option{compression_level}
-
-Speed / quality tradeoff: higher values are faster / worse quality.
-@item
-@option{q} / @option{global_quality}
-
-Size / quality tradeoff: higher values are smaller / worse quality.
-@item
-@option{qmin}
-(only: @option{qmax} is not supported)
-@item
-@option{i_qfactor} / @option{i_quant_factor}
-@item
-@option{i_qoffset} / @option{i_quant_offset}
-@item
-@option{b_qfactor} / @option{b_quant_factor}
-@item
-@option{b_qoffset} / @option{b_quant_offset}
-@end itemize
-
-@table @option
-
-@item h264_vaapi
-@option{profile} sets the value of @emph{profile_idc} and the @emph{constraint_set*_flag}s.
-@option{level} sets the value of @emph{level_idc}.
-
-@table @option
-@item low_power
-Use low-power encoding mode.
-@item coder
-Set entropy encoder (default is @emph{cabac}).  Possible values:
-
-@table @samp
-@item ac
-@item cabac
-Use CABAC.
-
-@item vlc
-@item cavlc
-Use CAVLC.
-@end table
-@end table
-
-@item hevc_vaapi
-@option{profile} and @option{level} set the values of
-@emph{general_profile_idc} and @emph{general_level_idc} respectively.
-
-@item mjpeg_vaapi
-Always encodes using the standard quantisation and huffman tables -
-@option{global_quality} scales the standard quantisation table (range 1-100).
-
-@item mpeg2_vaapi
-@option{profile} and @option{level} set the value of @emph{profile_and_level_indication}.
-
-No rate control is supported.
-
-@item vp8_vaapi
-B-frames are not supported.
-
-@option{global_quality} sets the @emph{q_idx} used for non-key frames (range 0-127).
-
-@table @option
-@item loop_filter_level
-@item loop_filter_sharpness
-Manually set the loop filter parameters.
-@end table
-
-@item vp9_vaapi
-@option{global_quality} sets the @emph{q_idx} used for P-frames (range 0-255).
-
-@table @option
-@item loop_filter_level
-@item loop_filter_sharpness
-Manually set the loop filter parameters.
-@end table
-
-B-frames are supported, but the output stream is always in encode order rather than display
-order.  If B-frames are enabled, it may be necessary to use the @option{vp9_raw_reorder}
-bitstream filter to modify the output stream to display frames in the correct order.
-
-Only normal frames are produced - the @option{vp9_superframe} bitstream filter may be
-required to produce a stream usable with all decoders.
-
-@end table
-
 @section vc2
 
 SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at

doc/examples/.gitignore

@@ -1,20 +1,14 @@
 /avio_dir_cmd
 /avio_reading
-/decode_audio
-/decode_video
+/decoding_encoding
 /demuxing_decoding
-/encode_audio
-/encode_video
 /extract_mvs
 /filter_audio
 /filtering_audio
 /filtering_video
-/http_multiclient
-/hw_decode
 /metadata
 /muxing
 /pc-uninstalled
-/qsvdec
 /remuxing
 /resampling_audio
 /scaling_video

doc/examples/Makefile

@@ -1,64 +1,46 @@
-EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
-EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)      += decode_audio
-EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)      += decode_video
-EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE)      += encode_audio
-EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE)      += encode_video
-EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
-EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
-EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
-EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
-EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE)  += http_multiclient
-EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE)         += hw_decode
-EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
-EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
-EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
-EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE)      += vaapi_encode
-EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE)   += vaapi_transcode
+# use pkg-config for getting CFLAGS and LDLIBS
+FFMPEG_LIBS=    libavdevice                        \
+                libavformat                        \
+                libavfilter                        \
+                libavcodec                         \
+                libswresample                      \
+                libswscale                         \
+                libavutil                          \
 
-EXAMPLES       := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-EXAMPLES_G     := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
-ALL_EXAMPLES   := $(EXAMPLES) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_EXAMPLES_G := $(EXAMPLES_G) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
-PROGS          += $(EXAMPLES)
+CFLAGS += -Wall -g
+CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
+LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLE_MAKEFILE := $(SRC_PATH)/doc/examples/Makefile
-EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/README $(EXAMPLE_MAKEFILE)
+EXAMPLES=       avio_dir_cmd                       \
+                avio_reading                       \
+                decoding_encoding                  \
+                demuxing_decoding                  \
+                extract_mvs                        \
+                filtering_video                    \
+                filtering_audio                    \
+                http_multiclient                   \
+                metadata                           \
+                muxing                             \
+                remuxing                           \
+                resampling_audio                   \
+                scaling_video                      \
+                transcode_aac                      \
+                transcoding                        \
 
-$(foreach P,$(EXAMPLES),$(eval OBJS-$(P:%$(PROGSSUF)$(EXESUF)=%) = $(P:%$(PROGSSUF)$(EXESUF)=%).o))
-$(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
+OBJS=$(addsuffix .o,$(EXAMPLES))
 
-examples: $(EXAMPLES)
+# the following examples make explicit use of the math library
+avcodec:           LDLIBS += -lm
+decoding_encoding: LDLIBS += -lm
+muxing:            LDLIBS += -lm
+resampling_audio:  LDLIBS += -lm
 
-$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
-OBJDIRS += doc/examples
+.phony: all clean-test clean
 
-DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
+all: $(OBJS) $(EXAMPLES)
 
-install: install-examples
+clean-test:
+	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
 
-install-examples: $(EXAMPLES_FILES)
-	$(Q)mkdir -p "$(DATADIR)/examples"
-	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
-	$(INSTALL) -m 644 $(EXAMPLE_MAKEFILE:%=%.example) "$(DATADIR)/examples/Makefile"
-
-uninstall: uninstall-examples
-
-uninstall-examples:
-	$(RM) -r "$(DATADIR)/examples"
-
-examplesclean:
-	$(RM) $(ALL_EXAMPLES) $(ALL_EXAMPLES_G)
-	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
-
-docclean:: examplesclean
-
--include $(wildcard $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.d))
-
-.PHONY: examples
+clean: clean-test
+	$(RM) $(EXAMPLES) $(OBJS)

doc/examples/Makefile.example

@@ -1,50 +0,0 @@
-# use pkg-config for getting CFLAGS and LDLIBS
-FFMPEG_LIBS=    libavdevice                        \
-                libavformat                        \
-                libavfilter                        \
-                libavcodec                         \
-                libswresample                      \
-                libswscale                         \
-                libavutil                          \
-
-CFLAGS += -Wall -g
-CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
-LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
-
-EXAMPLES=       avio_dir_cmd                       \
-                avio_reading                       \
-                decode_audio                       \
-                decode_video                       \
-                demuxing_decoding                  \
-                encode_audio                       \
-                encode_video                       \
-                extract_mvs                        \
-                filtering_video                    \
-                filtering_audio                    \
-                http_multiclient                   \
-                hw_decode                          \
-                metadata                           \
-                muxing                             \
-                remuxing                           \
-                resampling_audio                   \
-                scaling_video                      \
-                transcode_aac                      \
-                transcoding                        \
-
-OBJS=$(addsuffix .o,$(EXAMPLES))
-
-# the following examples make explicit use of the math library
-avcodec:           LDLIBS += -lm
-encode_audio:      LDLIBS += -lm
-muxing:            LDLIBS += -lm
-resampling_audio:  LDLIBS += -lm
-
-.phony: all clean-test clean
-
-all: $(OBJS) $(EXAMPLES)
-
-clean-test:
-	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
-
-clean: clean-test
-	$(RM) $(EXAMPLES) $(OBJS)

doc/examples/avio_dir_cmd.c

@@ -143,6 +143,8 @@ int main(int argc, char *argv[])
         return 1;
     }
 
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
     avformat_network_init();
 
     op = argv[1];

doc/examples/avio_reading.c

@@ -44,8 +44,6 @@ static int read_packet(void *opaque, uint8_t *buf, int buf_size)
     struct buffer_data *bd = (struct buffer_data *)opaque;
     buf_size = FFMIN(buf_size, bd->size);
 
-    if (!buf_size)
-        return AVERROR_EOF;
     printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
 
     /* copy internal buffer data to buf */
@@ -74,6 +72,9 @@ int main(int argc, char *argv[])
     }
     input_filename = argv[1];
 
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+
     /* slurp file content into buffer */
     ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
     if (ret < 0)

doc/examples/decode_audio.c

@@ -1,184 +0,0 @@
-/*
- * Copyright (c) 2001 Fabrice Bellard
- *
- * 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
- * audio decoding with libavcodec API example
- *
- * @example decode_audio.c
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <libavutil/frame.h>
-#include <libavutil/mem.h>
-
-#include <libavcodec/avcodec.h>
-
-#define AUDIO_INBUF_SIZE 20480
-#define AUDIO_REFILL_THRESH 4096
-
-static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
-                   FILE *outfile)
-{
-    int i, ch;
-    int ret, data_size;
-
-    /* send the packet with the compressed data to the decoder */
-    ret = avcodec_send_packet(dec_ctx, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error submitting the packet to the decoder\n");
-        exit(1);
-    }
-
-    /* read all the output frames (in general there may be any number of them */
-    while (ret >= 0) {
-        ret = avcodec_receive_frame(dec_ctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            return;
-        else if (ret < 0) {
-            fprintf(stderr, "Error during decoding\n");
-            exit(1);
-        }
-        data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
-        if (data_size < 0) {
-            /* This should not occur, checking just for paranoia */
-            fprintf(stderr, "Failed to calculate data size\n");
-            exit(1);
-        }
-        for (i = 0; i < frame->nb_samples; i++)
-            for (ch = 0; ch < dec_ctx->channels; ch++)
-                fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
-    }
-}
-
-int main(int argc, char **argv)
-{
-    const char *outfilename, *filename;
-    const AVCodec *codec;
-    AVCodecContext *c= NULL;
-    AVCodecParserContext *parser = NULL;
-    int len, ret;
-    FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
-    uint8_t *data;
-    size_t   data_size;
-    AVPacket *pkt;
-    AVFrame *decoded_frame = NULL;
-
-    if (argc <= 2) {
-        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
-        exit(0);
-    }
-    filename    = argv[1];
-    outfilename = argv[2];
-
-    pkt = av_packet_alloc();
-
-    /* find the MPEG audio decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
-    if (!codec) {
-        fprintf(stderr, "Codec not found\n");
-        exit(1);
-    }
-
-    parser = av_parser_init(codec->id);
-    if (!parser) {
-        fprintf(stderr, "Parser not found\n");
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "rb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-    outfile = fopen(outfilename, "wb");
-    if (!outfile) {
-        av_free(c);
-        exit(1);
-    }
-
-    /* decode until eof */
-    data      = inbuf;
-    data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
-
-    while (data_size > 0) {
-        if (!decoded_frame) {
-            if (!(decoded_frame = av_frame_alloc())) {
-                fprintf(stderr, "Could not allocate audio frame\n");
-                exit(1);
-            }
-        }
-
-        ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
-                               data, data_size,
-                               AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
-        if (ret < 0) {
-            fprintf(stderr, "Error while parsing\n");
-            exit(1);
-        }
-        data      += ret;
-        data_size -= ret;
-
-        if (pkt->size)
-            decode(c, pkt, decoded_frame, outfile);
-
-        if (data_size < AUDIO_REFILL_THRESH) {
-            memmove(inbuf, data, data_size);
-            data = inbuf;
-            len = fread(data + data_size, 1,
-                        AUDIO_INBUF_SIZE - data_size, f);
-            if (len > 0)
-                data_size += len;
-        }
-    }
-
-    /* flush the decoder */
-    pkt->data = NULL;
-    pkt->size = 0;
-    decode(c, pkt, decoded_frame, outfile);
-
-    fclose(outfile);
-    fclose(f);
-
-    avcodec_free_context(&c);
-    av_parser_close(parser);
-    av_frame_free(&decoded_frame);
-    av_packet_free(&pkt);
-
-    return 0;
-}

doc/examples/decode_video.c

@@ -1,186 +0,0 @@
-/*
- * Copyright (c) 2001 Fabrice Bellard
- *
- * 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
- * video decoding with libavcodec API example
- *
- * @example decode_video.c
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <libavcodec/avcodec.h>
-
-#define INBUF_SIZE 4096
-
-static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
-                     char *filename)
-{
-    FILE *f;
-    int i;
-
-    f = fopen(filename,"w");
-    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
-    for (i = 0; i < ysize; i++)
-        fwrite(buf + i * wrap, 1, xsize, f);
-    fclose(f);
-}
-
-static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
-                   const char *filename)
-{
-    char buf[1024];
-    int ret;
-
-    ret = avcodec_send_packet(dec_ctx, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error sending a packet for decoding\n");
-        exit(1);
-    }
-
-    while (ret >= 0) {
-        ret = avcodec_receive_frame(dec_ctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            return;
-        else if (ret < 0) {
-            fprintf(stderr, "Error during decoding\n");
-            exit(1);
-        }
-
-        printf("saving frame %3d\n", dec_ctx->frame_number);
-        fflush(stdout);
-
-        /* the picture is allocated by the decoder. no need to
-           free it */
-        snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
-        pgm_save(frame->data[0], frame->linesize[0],
-                 frame->width, frame->height, buf);
-    }
-}
-
-int main(int argc, char **argv)
-{
-    const char *filename, *outfilename;
-    const AVCodec *codec;
-    AVCodecParserContext *parser;
-    AVCodecContext *c= NULL;
-    FILE *f;
-    AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
-    uint8_t *data;
-    size_t   data_size;
-    int ret;
-    AVPacket *pkt;
-
-    if (argc <= 2) {
-        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
-        exit(0);
-    }
-    filename    = argv[1];
-    outfilename = argv[2];
-
-    pkt = av_packet_alloc();
-    if (!pkt)
-        exit(1);
-
-    /* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
-    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
-
-    /* find the MPEG-1 video decoder */
-    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
-    if (!codec) {
-        fprintf(stderr, "Codec not found\n");
-        exit(1);
-    }
-
-    parser = av_parser_init(codec->id);
-    if (!parser) {
-        fprintf(stderr, "parser not found\n");
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
-
-    /* For some codecs, such as msmpeg4 and mpeg4, width and height
-       MUST be initialized there because this information is not
-       available in the bitstream. */
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "rb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-
-    while (!feof(f)) {
-        /* read raw data from the input file */
-        data_size = fread(inbuf, 1, INBUF_SIZE, f);
-        if (!data_size)
-            break;
-
-        /* use the parser to split the data into frames */
-        data = inbuf;
-        while (data_size > 0) {
-            ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
-                                   data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
-            if (ret < 0) {
-                fprintf(stderr, "Error while parsing\n");
-                exit(1);
-            }
-            data      += ret;
-            data_size -= ret;
-
-            if (pkt->size)
-                decode(c, frame, pkt, outfilename);
-        }
-    }
-
-    /* flush the decoder */
-    decode(c, frame, NULL, outfilename);
-
-    fclose(f);
-
-    av_parser_close(parser);
-    avcodec_free_context(&c);
-    av_frame_free(&frame);
-    av_packet_free(&pkt);
-
-    return 0;
-}

doc/examples/decoding_encoding.c

@@ -0,0 +1,665 @@
+/*
+ * Copyright (c) 2001 Fabrice Bellard
+ *
+ * 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
+ * libavcodec API use example.
+ *
+ * @example decoding_encoding.c
+ * Note that libavcodec only handles codecs (MPEG, MPEG-4, etc...),
+ * not file formats (AVI, VOB, MP4, MOV, MKV, MXF, FLV, MPEG-TS, MPEG-PS, etc...).
+ * See library 'libavformat' for the format handling
+ */
+
+#include <math.h>
+
+#include <libavutil/opt.h>
+#include <libavcodec/avcodec.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/common.h>
+#include <libavutil/imgutils.h>
+#include <libavutil/mathematics.h>
+#include <libavutil/samplefmt.h>
+
+#define INBUF_SIZE 4096
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
+
+/* check that a given sample format is supported by the encoder */
+static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
+{
+    const enum AVSampleFormat *p = codec->sample_fmts;
+
+    while (*p != AV_SAMPLE_FMT_NONE) {
+        if (*p == sample_fmt)
+            return 1;
+        p++;
+    }
+    return 0;
+}
+
+/* just pick the highest supported samplerate */
+static int select_sample_rate(AVCodec *codec)
+{
+    const int *p;
+    int best_samplerate = 0;
+
+    if (!codec->supported_samplerates)
+        return 44100;
+
+    p = codec->supported_samplerates;
+    while (*p) {
+        best_samplerate = FFMAX(*p, best_samplerate);
+        p++;
+    }
+    return best_samplerate;
+}
+
+/* select layout with the highest channel count */
+static int select_channel_layout(AVCodec *codec)
+{
+    const uint64_t *p;
+    uint64_t best_ch_layout = 0;
+    int best_nb_channels   = 0;
+
+    if (!codec->channel_layouts)
+        return AV_CH_LAYOUT_STEREO;
+
+    p = codec->channel_layouts;
+    while (*p) {
+        int nb_channels = av_get_channel_layout_nb_channels(*p);
+
+        if (nb_channels > best_nb_channels) {
+            best_ch_layout    = *p;
+            best_nb_channels = nb_channels;
+        }
+        p++;
+    }
+    return best_ch_layout;
+}
+
+/*
+ * Audio encoding example
+ */
+static void audio_encode_example(const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    AVFrame *frame;
+    AVPacket pkt;
+    int i, j, k, ret, got_output;
+    int buffer_size;
+    FILE *f;
+    uint16_t *samples;
+    float t, tincr;
+
+    printf("Encode audio file %s\n", filename);
+
+    /* find the MP2 encoder */
+    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate audio codec context\n");
+        exit(1);
+    }
+
+    /* put sample parameters */
+    c->bit_rate = 64000;
+
+    /* check that the encoder supports s16 pcm input */
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (!check_sample_fmt(codec, c->sample_fmt)) {
+        fprintf(stderr, "Encoder does not support sample format %s",
+                av_get_sample_fmt_name(c->sample_fmt));
+        exit(1);
+    }
+
+    /* select other audio parameters supported by the encoder */
+    c->sample_rate    = select_sample_rate(codec);
+    c->channel_layout = select_channel_layout(codec);
+    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    /* frame containing input raw audio */
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate audio frame\n");
+        exit(1);
+    }
+
+    frame->nb_samples     = c->frame_size;
+    frame->format         = c->sample_fmt;
+    frame->channel_layout = c->channel_layout;
+
+    /* the codec gives us the frame size, in samples,
+     * we calculate the size of the samples buffer in bytes */
+    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
+                                             c->sample_fmt, 0);
+    if (buffer_size < 0) {
+        fprintf(stderr, "Could not get sample buffer size\n");
+        exit(1);
+    }
+    samples = av_malloc(buffer_size);
+    if (!samples) {
+        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
+                buffer_size);
+        exit(1);
+    }
+    /* setup the data pointers in the AVFrame */
+    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                                   (const uint8_t*)samples, buffer_size, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not setup audio frame\n");
+        exit(1);
+    }
+
+    /* encode a single tone sound */
+    t = 0;
+    tincr = 2 * M_PI * 440.0 / c->sample_rate;
+    for (i = 0; i < 200; i++) {
+        av_init_packet(&pkt);
+        pkt.data = NULL; // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        for (j = 0; j < c->frame_size; j++) {
+            samples[2*j] = (int)(sin(t) * 10000);
+
+            for (k = 1; k < c->channels; k++)
+                samples[2*j + k] = samples[2*j];
+            t += tincr;
+        }
+        /* encode the samples */
+        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding audio frame\n");
+            exit(1);
+        }
+        if (got_output) {
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* get the delayed frames */
+    for (got_output = 1; got_output; i++) {
+        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+    fclose(f);
+
+    av_freep(&samples);
+    av_frame_free(&frame);
+    avcodec_close(c);
+    av_free(c);
+}
+
+/*
+ * Audio decoding.
+ */
+static void audio_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int len;
+    FILE *f, *outfile;
+    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+    AVFrame *decoded_frame = NULL;
+
+    av_init_packet(&avpkt);
+
+    printf("Decode audio file %s to %s\n", filename, outfilename);
+
+    /* find the MPEG audio decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate audio codec context\n");
+        exit(1);
+    }
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+    outfile = fopen(outfilename, "wb");
+    if (!outfile) {
+        av_free(c);
+        exit(1);
+    }
+
+    /* decode until eof */
+    avpkt.data = inbuf;
+    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+    while (avpkt.size > 0) {
+        int i, ch;
+        int got_frame = 0;
+
+        if (!decoded_frame) {
+            if (!(decoded_frame = av_frame_alloc())) {
+                fprintf(stderr, "Could not allocate audio frame\n");
+                exit(1);
+            }
+        }
+
+        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
+        if (len < 0) {
+            fprintf(stderr, "Error while decoding\n");
+            exit(1);
+        }
+        if (got_frame) {
+            /* if a frame has been decoded, output it */
+            int data_size = av_get_bytes_per_sample(c->sample_fmt);
+            if (data_size < 0) {
+                /* This should not occur, checking just for paranoia */
+                fprintf(stderr, "Failed to calculate data size\n");
+                exit(1);
+            }
+            for (i=0; i<decoded_frame->nb_samples; i++)
+                for (ch=0; ch<c->channels; ch++)
+                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
+        }
+        avpkt.size -= len;
+        avpkt.data += len;
+        avpkt.dts =
+        avpkt.pts = AV_NOPTS_VALUE;
+        if (avpkt.size < AUDIO_REFILL_THRESH) {
+            /* Refill the input buffer, to avoid trying to decode
+             * incomplete frames. Instead of this, one could also use
+             * a parser, or use a proper container format through
+             * libavformat. */
+            memmove(inbuf, avpkt.data, avpkt.size);
+            avpkt.data = inbuf;
+            len = fread(avpkt.data + avpkt.size, 1,
+                        AUDIO_INBUF_SIZE - avpkt.size, f);
+            if (len > 0)
+                avpkt.size += len;
+        }
+    }
+
+    fclose(outfile);
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_frame_free(&decoded_frame);
+}
+
+/*
+ * Video encoding example
+ */
+static void video_encode_example(const char *filename, int codec_id)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int i, ret, x, y, got_output;
+    FILE *f;
+    AVFrame *frame;
+    AVPacket pkt;
+    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
+
+    printf("Encode video file %s\n", filename);
+
+    /* find the video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    /* put sample parameters */
+    c->bit_rate = 400000;
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* frames per second */
+    c->time_base = (AVRational){1,25};
+    /* emit one intra frame every ten frames
+     * check frame pict_type before passing frame
+     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
+     * then gop_size is ignored and the output of encoder
+     * will always be I frame irrespective to gop_size
+     */
+    c->gop_size = 10;
+    c->max_b_frames = 1;
+    c->pix_fmt = AV_PIX_FMT_YUV420P;
+
+    if (codec_id == AV_CODEC_ID_H264)
+        av_opt_set(c->priv_data, "preset", "slow", 0);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+    frame->format = c->pix_fmt;
+    frame->width  = c->width;
+    frame->height = c->height;
+
+    /* the image can be allocated by any means and av_image_alloc() is
+     * just the most convenient way if av_malloc() is to be used */
+    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
+                         c->pix_fmt, 32);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate raw picture buffer\n");
+        exit(1);
+    }
+
+    /* encode 1 second of video */
+    for (i = 0; i < 25; i++) {
+        av_init_packet(&pkt);
+        pkt.data = NULL;    // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        fflush(stdout);
+        /* prepare a dummy image */
+        /* Y */
+        for (y = 0; y < c->height; y++) {
+            for (x = 0; x < c->width; x++) {
+                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
+            }
+        }
+
+        /* Cb and Cr */
+        for (y = 0; y < c->height/2; y++) {
+            for (x = 0; x < c->width/2; x++) {
+                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
+                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
+            }
+        }
+
+        frame->pts = i;
+
+        /* encode the image */
+        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* get the delayed frames */
+    for (got_output = 1; got_output; i++) {
+        fflush(stdout);
+
+        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* add sequence end code to have a real MPEG file */
+    fwrite(endcode, 1, sizeof(endcode), f);
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_freep(&frame->data[0]);
+    av_frame_free(&frame);
+    printf("\n");
+}
+
+/*
+ * Video decoding example
+ */
+
+static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
+                     char *filename)
+{
+    FILE *f;
+    int i;
+
+    f = fopen(filename,"w");
+    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
+    for (i = 0; i < ysize; i++)
+        fwrite(buf + i * wrap, 1, xsize, f);
+    fclose(f);
+}
+
+static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
+                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
+{
+    int len, got_frame;
+    char buf[1024];
+
+    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
+    if (len < 0) {
+        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
+        return len;
+    }
+    if (got_frame) {
+        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder, no need to free it */
+        snprintf(buf, sizeof(buf), outfilename, *frame_count);
+        pgm_save(frame->data[0], frame->linesize[0],
+                 frame->width, frame->height, buf);
+        (*frame_count)++;
+    }
+    if (pkt->data) {
+        pkt->size -= len;
+        pkt->data += len;
+    }
+    return 0;
+}
+
+static void video_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int frame_count;
+    FILE *f;
+    AVFrame *frame;
+    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+
+    av_init_packet(&avpkt);
+
+    /* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
+    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+
+    printf("Decode video file %s to %s\n", filename, outfilename);
+
+    /* find the MPEG-1 video decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
+        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
+
+    /* For some codecs, such as msmpeg4 and mpeg4, width and height
+       MUST be initialized there because this information is not
+       available in the bitstream. */
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+
+    frame_count = 0;
+    for (;;) {
+        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
+        if (avpkt.size == 0)
+            break;
+
+        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
+           and this is the only method to use them because you cannot
+           know the compressed data size before analysing it.
+
+           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
+           based, so you must call them with all the data for one
+           frame exactly. You must also initialize 'width' and
+           'height' before initializing them. */
+
+        /* NOTE2: some codecs allow the raw parameters (frame size,
+           sample rate) to be changed at any frame. We handle this, so
+           you should also take care of it */
+
+        /* here, we use a stream based decoder (mpeg1video), so we
+           feed decoder and see if it could decode a frame */
+        avpkt.data = inbuf;
+        while (avpkt.size > 0)
+            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
+                exit(1);
+    }
+
+    /* Some codecs, such as MPEG, transmit the I- and P-frame with a
+       latency of one frame. You must do the following to have a
+       chance to get the last frame of the video. */
+    avpkt.data = NULL;
+    avpkt.size = 0;
+    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
+
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_frame_free(&frame);
+    printf("\n");
+}
+
+int main(int argc, char **argv)
+{
+    const char *output_type;
+
+    /* register all the codecs */
+    avcodec_register_all();
+
+    if (argc < 2) {
+        printf("usage: %s output_type\n"
+               "API example program to decode/encode a media stream with libavcodec.\n"
+               "This program generates a synthetic stream and encodes it to a file\n"
+               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
+               "The encoded stream is then decoded and written to a raw data output.\n"
+               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
+               argv[0]);
+        return 1;
+    }
+    output_type = argv[1];
+
+    if (!strcmp(output_type, "h264")) {
+        video_encode_example("test.h264", AV_CODEC_ID_H264);
+    } else if (!strcmp(output_type, "mp2")) {
+        audio_encode_example("test.mp2");
+        audio_decode_example("test.pcm", "test.mp2");
+    } else if (!strcmp(output_type, "mpg")) {
+        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
+        video_decode_example("test%02d.pgm", "test.mpg");
+    } else {
+        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
+                output_type);
+        return 1;
+    }
+
+    return 0;
+}

doc/examples/demuxing_decoding.c

@@ -252,6 +252,9 @@ int main (int argc, char **argv)
     video_dst_filename = argv[2];
     audio_dst_filename = argv[3];
 
+    /* register all formats and codecs */
+    av_register_all();
+
     /* open input file, and allocate format context */
     if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
         fprintf(stderr, "Could not open source file %s\n", src_filename);

doc/examples/encode_audio.c

@@ -1,238 +0,0 @@
-/*
- * Copyright (c) 2001 Fabrice Bellard
- *
- * 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
- * audio encoding with libavcodec API example.
- *
- * @example encode_audio.c
- */
-
-#include <stdint.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <libavcodec/avcodec.h>
-
-#include <libavutil/channel_layout.h>
-#include <libavutil/common.h>
-#include <libavutil/frame.h>
-#include <libavutil/samplefmt.h>
-
-/* check that a given sample format is supported by the encoder */
-static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
-{
-    const enum AVSampleFormat *p = codec->sample_fmts;
-
-    while (*p != AV_SAMPLE_FMT_NONE) {
-        if (*p == sample_fmt)
-            return 1;
-        p++;
-    }
-    return 0;
-}
-
-/* just pick the highest supported samplerate */
-static int select_sample_rate(const AVCodec *codec)
-{
-    const int *p;
-    int best_samplerate = 0;
-
-    if (!codec->supported_samplerates)
-        return 44100;
-
-    p = codec->supported_samplerates;
-    while (*p) {
-        if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
-            best_samplerate = *p;
-        p++;
-    }
-    return best_samplerate;
-}
-
-/* select layout with the highest channel count */
-static int select_channel_layout(const AVCodec *codec)
-{
-    const uint64_t *p;
-    uint64_t best_ch_layout = 0;
-    int best_nb_channels   = 0;
-
-    if (!codec->channel_layouts)
-        return AV_CH_LAYOUT_STEREO;
-
-    p = codec->channel_layouts;
-    while (*p) {
-        int nb_channels = av_get_channel_layout_nb_channels(*p);
-
-        if (nb_channels > best_nb_channels) {
-            best_ch_layout    = *p;
-            best_nb_channels = nb_channels;
-        }
-        p++;
-    }
-    return best_ch_layout;
-}
-
-static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
-                   FILE *output)
-{
-    int ret;
-
-    /* send the frame for encoding */
-    ret = avcodec_send_frame(ctx, frame);
-    if (ret < 0) {
-        fprintf(stderr, "Error sending the frame to the encoder\n");
-        exit(1);
-    }
-
-    /* read all the available output packets (in general there may be any
-     * number of them */
-    while (ret >= 0) {
-        ret = avcodec_receive_packet(ctx, pkt);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            return;
-        else if (ret < 0) {
-            fprintf(stderr, "Error encoding audio frame\n");
-            exit(1);
-        }
-
-        fwrite(pkt->data, 1, pkt->size, output);
-        av_packet_unref(pkt);
-    }
-}
-
-int main(int argc, char **argv)
-{
-    const char *filename;
-    const AVCodec *codec;
-    AVCodecContext *c= NULL;
-    AVFrame *frame;
-    AVPacket *pkt;
-    int i, j, k, ret;
-    FILE *f;
-    uint16_t *samples;
-    float t, tincr;
-
-    if (argc <= 1) {
-        fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
-        return 0;
-    }
-    filename = argv[1];
-
-    /* find the MP2 encoder */
-    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
-    if (!codec) {
-        fprintf(stderr, "Codec not found\n");
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
-
-    /* put sample parameters */
-    c->bit_rate = 64000;
-
-    /* check that the encoder supports s16 pcm input */
-    c->sample_fmt = AV_SAMPLE_FMT_S16;
-    if (!check_sample_fmt(codec, c->sample_fmt)) {
-        fprintf(stderr, "Encoder does not support sample format %s",
-                av_get_sample_fmt_name(c->sample_fmt));
-        exit(1);
-    }
-
-    /* select other audio parameters supported by the encoder */
-    c->sample_rate    = select_sample_rate(codec);
-    c->channel_layout = select_channel_layout(codec);
-    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
-
-    /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
-        fprintf(stderr, "Could not open codec\n");
-        exit(1);
-    }
-
-    f = fopen(filename, "wb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    /* packet for holding encoded output */
-    pkt = av_packet_alloc();
-    if (!pkt) {
-        fprintf(stderr, "could not allocate the packet\n");
-        exit(1);
-    }
-
-    /* frame containing input raw audio */
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate audio frame\n");
-        exit(1);
-    }
-
-    frame->nb_samples     = c->frame_size;
-    frame->format         = c->sample_fmt;
-    frame->channel_layout = c->channel_layout;
-
-    /* allocate the data buffers */
-    ret = av_frame_get_buffer(frame, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate audio data buffers\n");
-        exit(1);
-    }
-
-    /* encode a single tone sound */
-    t = 0;
-    tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for (i = 0; i < 200; i++) {
-        /* make sure the frame is writable -- makes a copy if the encoder
-         * kept a reference internally */
-        ret = av_frame_make_writable(frame);
-        if (ret < 0)
-            exit(1);
-        samples = (uint16_t*)frame->data[0];
-
-        for (j = 0; j < c->frame_size; j++) {
-            samples[2*j] = (int)(sin(t) * 10000);
-
-            for (k = 1; k < c->channels; k++)
-                samples[2*j + k] = samples[2*j];
-            t += tincr;
-        }
-        encode(c, frame, pkt, f);
-    }
-
-    /* flush the encoder */
-    encode(c, NULL, pkt, f);
-
-    fclose(f);
-
-    av_frame_free(&frame);
-    av_packet_free(&pkt);
-    avcodec_free_context(&c);
-
-    return 0;
-}

doc/examples/encode_video.c

@@ -1,197 +0,0 @@
-/*
- * Copyright (c) 2001 Fabrice Bellard
- *
- * 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
- * video encoding with libavcodec API example
- *
- * @example encode_video.c
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <libavcodec/avcodec.h>
-
-#include <libavutil/opt.h>
-#include <libavutil/imgutils.h>
-
-static void encode(AVCodecContext *enc_ctx, AVFrame *frame, AVPacket *pkt,
-                   FILE *outfile)
-{
-    int ret;
-
-    /* send the frame to the encoder */
-    if (frame)
-        printf("Send frame %3"PRId64"\n", frame->pts);
-
-    ret = avcodec_send_frame(enc_ctx, frame);
-    if (ret < 0) {
-        fprintf(stderr, "Error sending a frame for encoding\n");
-        exit(1);
-    }
-
-    while (ret >= 0) {
-        ret = avcodec_receive_packet(enc_ctx, pkt);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            return;
-        else if (ret < 0) {
-            fprintf(stderr, "Error during encoding\n");
-            exit(1);
-        }
-
-        printf("Write packet %3"PRId64" (size=%5d)\n", pkt->pts, pkt->size);
-        fwrite(pkt->data, 1, pkt->size, outfile);
-        av_packet_unref(pkt);
-    }
-}
-
-int main(int argc, char **argv)
-{
-    const char *filename, *codec_name;
-    const AVCodec *codec;
-    AVCodecContext *c= NULL;
-    int i, ret, x, y;
-    FILE *f;
-    AVFrame *frame;
-    AVPacket *pkt;
-    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
-
-    if (argc <= 2) {
-        fprintf(stderr, "Usage: %s <output file> <codec name>\n", argv[0]);
-        exit(0);
-    }
-    filename = argv[1];
-    codec_name = argv[2];
-
-    /* find the mpeg1video encoder */
-    codec = avcodec_find_encoder_by_name(codec_name);
-    if (!codec) {
-        fprintf(stderr, "Codec '%s' not found\n", codec_name);
-        exit(1);
-    }
-
-    c = avcodec_alloc_context3(codec);
-    if (!c) {
-        fprintf(stderr, "Could not allocate video codec context\n");
-        exit(1);
-    }
-
-    pkt = av_packet_alloc();
-    if (!pkt)
-        exit(1);
-
-    /* put sample parameters */
-    c->bit_rate = 400000;
-    /* resolution must be a multiple of two */
-    c->width = 352;
-    c->height = 288;
-    /* frames per second */
-    c->time_base = (AVRational){1, 25};
-    c->framerate = (AVRational){25, 1};
-
-    /* emit one intra frame every ten frames
-     * check frame pict_type before passing frame
-     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
-     * then gop_size is ignored and the output of encoder
-     * will always be I frame irrespective to gop_size
-     */
-    c->gop_size = 10;
-    c->max_b_frames = 1;
-    c->pix_fmt = AV_PIX_FMT_YUV420P;
-
-    if (codec->id == AV_CODEC_ID_H264)
-        av_opt_set(c->priv_data, "preset", "slow", 0);
-
-    /* open it */
-    ret = avcodec_open2(c, codec, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
-        exit(1);
-    }
-
-    f = fopen(filename, "wb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        exit(1);
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate video frame\n");
-        exit(1);
-    }
-    frame->format = c->pix_fmt;
-    frame->width  = c->width;
-    frame->height = c->height;
-
-    ret = av_frame_get_buffer(frame, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate the video frame data\n");
-        exit(1);
-    }
-
-    /* encode 1 second of video */
-    for (i = 0; i < 25; i++) {
-        fflush(stdout);
-
-        /* make sure the frame data is writable */
-        ret = av_frame_make_writable(frame);
-        if (ret < 0)
-            exit(1);
-
-        /* prepare a dummy image */
-        /* Y */
-        for (y = 0; y < c->height; y++) {
-            for (x = 0; x < c->width; x++) {
-                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
-            }
-        }
-
-        /* Cb and Cr */
-        for (y = 0; y < c->height/2; y++) {
-            for (x = 0; x < c->width/2; x++) {
-                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
-                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
-            }
-        }
-
-        frame->pts = i;
-
-        /* encode the image */
-        encode(c, frame, pkt, f);
-    }
-
-    /* flush the encoder */
-    encode(c, NULL, pkt, f);
-
-    /* add sequence end code to have a real MPEG file */
-    fwrite(endcode, 1, sizeof(endcode), f);
-    fclose(f);
-
-    avcodec_free_context(&c);
-    av_frame_free(&frame);
-    av_packet_free(&pkt);
-
-    return 0;
-}

doc/examples/extract_mvs.c

@@ -31,26 +31,23 @@ static const char *src_filename = NULL;
 
 static int video_stream_idx = -1;
 static AVFrame *frame = NULL;
+static AVPacket pkt;
 static int video_frame_count = 0;
 
-static int decode_packet(const AVPacket *pkt)
+static int decode_packet(int *got_frame, int cached)
 {
-    int ret = avcodec_send_packet(video_dec_ctx, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error while sending a packet to the decoder: %s\n", av_err2str(ret));
-        return ret;
-    }
+    int decoded = pkt.size;
 
-    while (ret >= 0)  {
-        ret = avcodec_receive_frame(video_dec_ctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
-            break;
-        } else if (ret < 0) {
-            fprintf(stderr, "Error while receiving a frame from the decoder: %s\n", av_err2str(ret));
+    *got_frame = 0;
+
+    if (pkt.stream_index == video_stream_idx) {
+        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
             return ret;
         }
 
-        if (ret >= 0) {
+        if (*got_frame) {
             int i;
             AVFrameSideData *sd;
 
@@ -61,19 +58,19 @@ static int decode_packet(const AVPacket *pkt)
                 for (i = 0; i < sd->size / sizeof(*mvs); i++) {
                     const AVMotionVector *mv = &mvs[i];
                     printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
-                        video_frame_count, mv->source,
-                        mv->w, mv->h, mv->src_x, mv->src_y,
-                        mv->dst_x, mv->dst_y, mv->flags);
+                           video_frame_count, mv->source,
+                           mv->w, mv->h, mv->src_x, mv->src_y,
+                           mv->dst_x, mv->dst_y, mv->flags);
                 }
             }
-            av_frame_unref(frame);
         }
     }
 
-    return 0;
+    return decoded;
 }
 
-static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
+static int open_codec_context(int *stream_idx,
+                              AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
     int ret;
     AVStream *st;
@@ -81,27 +78,24 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
     AVCodec *dec = NULL;
     AVDictionary *opts = NULL;
 
-    ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
+    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
     if (ret < 0) {
         fprintf(stderr, "Could not find %s stream in input file '%s'\n",
                 av_get_media_type_string(type), src_filename);
         return ret;
     } else {
-        int stream_idx = ret;
-        st = fmt_ctx->streams[stream_idx];
+        *stream_idx = ret;
+        st = fmt_ctx->streams[*stream_idx];
 
-        dec_ctx = avcodec_alloc_context3(dec);
-        if (!dec_ctx) {
-            fprintf(stderr, "Failed to allocate codec\n");
+        /* find decoder for the stream */
+        dec_ctx = st->codec;
+        dec = avcodec_find_decoder(dec_ctx->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
             return AVERROR(EINVAL);
         }
 
-        ret = avcodec_parameters_to_context(dec_ctx, st->codecpar);
-        if (ret < 0) {
-            fprintf(stderr, "Failed to copy codec parameters to codec context\n");
-            return ret;
-        }
-
         /* Init the video decoder */
         av_dict_set(&opts, "flags2", "+export_mvs", 0);
         if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
@@ -109,10 +103,6 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
                     av_get_media_type_string(type));
             return ret;
         }
-
-        video_stream_idx = stream_idx;
-        video_stream = fmt_ctx->streams[video_stream_idx];
-        video_dec_ctx = dec_ctx;
     }
 
     return 0;
@@ -120,8 +110,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
 
 int main(int argc, char **argv)
 {
-    int ret = 0;
-    AVPacket pkt = { 0 };
+    int ret = 0, got_frame;
 
     if (argc != 2) {
         fprintf(stderr, "Usage: %s <video>\n", argv[0]);
@@ -129,6 +118,8 @@ int main(int argc, char **argv)
     }
     src_filename = argv[1];
 
+    av_register_all();
+
     if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
         fprintf(stderr, "Could not open source file %s\n", src_filename);
         exit(1);
@@ -139,7 +130,10 @@ int main(int argc, char **argv)
         exit(1);
     }
 
-    open_codec_context(fmt_ctx, AVMEDIA_TYPE_VIDEO);
+    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+        video_dec_ctx = video_stream->codec;
+    }
 
     av_dump_format(fmt_ctx, 0, src_filename, 0);
 
@@ -158,20 +152,33 @@ int main(int argc, char **argv)
 
     printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
 
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
     /* read frames from the file */
     while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        if (pkt.stream_index == video_stream_idx)
-            ret = decode_packet(&pkt);
-        av_packet_unref(&pkt);
-        if (ret < 0)
-            break;
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_packet_unref(&orig_pkt);
     }
 
     /* flush cached frames */
-    decode_packet(NULL);
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
 
 end:
-    avcodec_free_context(&video_dec_ctx);
+    avcodec_close(video_dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     return ret < 0;

doc/examples/filter_audio.c

@@ -64,13 +64,13 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
 {
     AVFilterGraph *filter_graph;
     AVFilterContext *abuffer_ctx;
-    const AVFilter  *abuffer;
+    AVFilter        *abuffer;
     AVFilterContext *volume_ctx;
-    const AVFilter  *volume;
+    AVFilter        *volume;
     AVFilterContext *aformat_ctx;
-    const AVFilter  *aformat;
+    AVFilter        *aformat;
     AVFilterContext *abuffersink_ctx;
-    const AVFilter  *abuffersink;
+    AVFilter        *abuffersink;
 
     AVDictionary *options_dict = NULL;
     uint8_t options_str[1024];
@@ -289,6 +289,8 @@ int main(int argc, char *argv[])
         return 1;
     }
 
+    avfilter_register_all();
+
     /* Allocate the frame we will be using to store the data. */
     frame  = av_frame_alloc();
     if (!frame) {

doc/examples/filtering_audio.c

@@ -32,6 +32,7 @@
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@@ -68,12 +69,7 @@ static int open_input_file(const char *filename)
         return ret;
     }
     audio_stream_index = ret;
-
-    /* create decoding context */
-    dec_ctx = avcodec_alloc_context3(dec);
-    if (!dec_ctx)
-        return AVERROR(ENOMEM);
-    avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
+    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
     av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the audio decoder */
@@ -89,8 +85,8 @@ static int init_filters(const char *filters_descr)
 {
     char args[512];
     int ret = 0;
-    const AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
+    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
@@ -200,7 +196,7 @@ end:
 
 static void print_frame(const AVFrame *frame)
 {
-    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
+    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
     const uint16_t *p     = (uint16_t*)frame->data[0];
     const uint16_t *p_end = p + n;
 
@@ -215,9 +211,10 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket packet;
+    AVPacket packet0, packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
+    int got_frame;
 
     if (!frame || !filt_frame) {
         perror("Could not allocate frame");
@@ -228,58 +225,63 @@ int main(int argc, char **argv)
         exit(1);
     }
 
+    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 */
+    packet0.data = NULL;
+    packet.data = NULL;
     while (1) {
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
+        if (!packet0.data) {
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+                break;
+            packet0 = packet;
+        }
 
         if (packet.stream_index == audio_stream_index) {
-            ret = avcodec_send_packet(dec_ctx, &packet);
+            got_frame = 0;
+            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
-                break;
+                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
+                continue;
             }
+            packet.size -= ret;
+            packet.data += ret;
 
-            while (ret >= 0) {
-                ret = avcodec_receive_frame(dec_ctx, frame);
-                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+            if (got_frame) {
+                /* push the audio data from decoded frame into the filtergraph */
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                     break;
-                } else if (ret < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
-                    goto end;
                 }
 
-                if (ret >= 0) {
-                    /* push the audio data from decoded frame into the filtergraph */
-                    if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
-                        av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
+                /* pull filtered audio from the filtergraph */
+                while (1) {
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                         break;
-                    }
-
-                    /* pull filtered audio from the filtergraph */
-                    while (1) {
-                        ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                            break;
-                        if (ret < 0)
-                            goto end;
-                        print_frame(filt_frame);
-                        av_frame_unref(filt_frame);
-                    }
-                    av_frame_unref(frame);
+                    if (ret < 0)
+                        goto end;
+                    print_frame(filt_frame);
+                    av_frame_unref(filt_frame);
                 }
             }
+
+            if (packet.size <= 0)
+                av_packet_unref(&packet0);
+        } else {
+            /* discard non-wanted packets */
+            av_packet_unref(&packet0);
         }
-        av_packet_unref(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    avcodec_free_context(&dec_ctx);
+    avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);

doc/examples/filtering_video.c

@@ -32,6 +32,7 @@
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@@ -71,12 +72,7 @@ static int open_input_file(const char *filename)
         return ret;
     }
     video_stream_index = ret;
-
-    /* create decoding context */
-    dec_ctx = avcodec_alloc_context3(dec);
-    if (!dec_ctx)
-        return AVERROR(ENOMEM);
-    avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
+    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
     av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the video decoder */
@@ -92,8 +88,8 @@ static int init_filters(const char *filters_descr)
 {
     char args[512];
     int ret = 0;
-    const AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    const AVFilter *buffersink = avfilter_get_by_name("buffersink");
+    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
@@ -212,6 +208,7 @@ int main(int argc, char **argv)
     AVPacket packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
+    int got_frame;
 
     if (!frame || !filt_frame) {
         perror("Could not allocate frame");
@@ -222,6 +219,9 @@ int main(int argc, char **argv)
         exit(1);
     }
 
+    av_register_all();
+    avfilter_register_all();
+
     if ((ret = open_input_file(argv[1])) < 0)
         goto end;
     if ((ret = init_filters(filter_descr)) < 0)
@@ -233,49 +233,40 @@ int main(int argc, char **argv)
             break;
 
         if (packet.stream_index == video_stream_index) {
-            ret = avcodec_send_packet(dec_ctx, &packet);
+            got_frame = 0;
+            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
+                av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                 break;
             }
 
-            while (ret >= 0) {
-                ret = avcodec_receive_frame(dec_ctx, frame);
-                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+            if (got_frame) {
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
+
+                /* push the decoded frame into the filtergraph */
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
                     break;
-                } else if (ret < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
-                    goto end;
                 }
 
-                if (ret >= 0) {
-                    frame->pts = frame->best_effort_timestamp;
-
-                    /* push the decoded frame into the filtergraph */
-                    if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
-                        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
+                /* pull filtered frames from the filtergraph */
+                while (1) {
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                         break;
-                    }
-
-                    /* pull filtered frames from the filtergraph */
-                    while (1) {
-                        ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                            break;
-                        if (ret < 0)
-                            goto end;
-                        display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
-                        av_frame_unref(filt_frame);
-                    }
-                    av_frame_unref(frame);
+                    if (ret < 0)
+                        goto end;
+                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
+                    av_frame_unref(filt_frame);
                 }
+                av_frame_unref(frame);
             }
         }
         av_packet_unref(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    avcodec_free_context(&dec_ctx);
+    avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);

doc/examples/http_multiclient.c

@@ -33,19 +33,18 @@
 #include <libavutil/opt.h>
 #include <unistd.h>
 
-static void process_client(AVIOContext *client, const char *in_uri)
+void process_client(AVIOContext *client, const char *in_uri)
 {
     AVIOContext *input = NULL;
     uint8_t buf[1024];
     int ret, n, reply_code;
-    uint8_t *resource = NULL;
+    char *resource = NULL;
     while ((ret = avio_handshake(client)) > 0) {
         av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
         // check for strlen(resource) is necessary, because av_opt_get()
         // may return empty string.
         if (resource && strlen(resource))
             break;
-        av_freep(&resource);
     }
     if (ret < 0)
         goto end;
@@ -94,16 +93,15 @@ end:
     avio_close(client);
     fprintf(stderr, "Closing input\n");
     avio_close(input);
-    av_freep(&resource);
 }
 
 int main(int argc, char **argv)
 {
+    av_log_set_level(AV_LOG_TRACE);
     AVDictionary *options = NULL;
     AVIOContext *client = NULL, *server = NULL;
     const char *in_uri, *out_uri;
     int ret, pid;
-    av_log_set_level(AV_LOG_TRACE);
     if (argc < 3) {
         printf("usage: %s input http://hostname[:port]\n"
                "API example program to serve http to multiple clients.\n"
@@ -114,6 +112,7 @@ int main(int argc, char **argv)
     in_uri = argv[1];
     out_uri = argv[2];
 
+    av_register_all();
     avformat_network_init();
 
     if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {

doc/examples/hw_decode.c

@@ -1,251 +0,0 @@
-/*
- * Copyright (c) 2017 Jun Zhao
- * Copyright (c) 2017 Kaixuan Liu
- *
- * HW Acceleration API (video decoding) decode sample
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * HW-Accelerated decoding example.
- *
- * @example hw_decode.c
- * This example shows how to do HW-accelerated decoding with output
- * frames from the HW video surfaces.
- */
-
-#include <stdio.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavutil/pixdesc.h>
-#include <libavutil/hwcontext.h>
-#include <libavutil/opt.h>
-#include <libavutil/avassert.h>
-#include <libavutil/imgutils.h>
-
-static AVBufferRef *hw_device_ctx = NULL;
-static enum AVPixelFormat hw_pix_fmt;
-static FILE *output_file = NULL;
-
-static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
-{
-    int err = 0;
-
-    if ((err = av_hwdevice_ctx_create(&hw_device_ctx, type,
-                                      NULL, NULL, 0)) < 0) {
-        fprintf(stderr, "Failed to create specified HW device.\n");
-        return err;
-    }
-    ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
-
-    return err;
-}
-
-static enum AVPixelFormat get_hw_format(AVCodecContext *ctx,
-                                        const enum AVPixelFormat *pix_fmts)
-{
-    const enum AVPixelFormat *p;
-
-    for (p = pix_fmts; *p != -1; p++) {
-        if (*p == hw_pix_fmt)
-            return *p;
-    }
-
-    fprintf(stderr, "Failed to get HW surface format.\n");
-    return AV_PIX_FMT_NONE;
-}
-
-static int decode_write(AVCodecContext *avctx, AVPacket *packet)
-{
-    AVFrame *frame = NULL, *sw_frame = NULL;
-    AVFrame *tmp_frame = NULL;
-    uint8_t *buffer = NULL;
-    int size;
-    int ret = 0;
-
-    ret = avcodec_send_packet(avctx, packet);
-    if (ret < 0) {
-        fprintf(stderr, "Error during decoding\n");
-        return ret;
-    }
-
-    while (1) {
-        if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
-            fprintf(stderr, "Can not alloc frame\n");
-            ret = AVERROR(ENOMEM);
-            goto fail;
-        }
-
-        ret = avcodec_receive_frame(avctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
-            av_frame_free(&frame);
-            av_frame_free(&sw_frame);
-            return 0;
-        } else if (ret < 0) {
-            fprintf(stderr, "Error while decoding\n");
-            goto fail;
-        }
-
-        if (frame->format == hw_pix_fmt) {
-            /* retrieve data from GPU to CPU */
-            if ((ret = av_hwframe_transfer_data(sw_frame, frame, 0)) < 0) {
-                fprintf(stderr, "Error transferring the data to system memory\n");
-                goto fail;
-            }
-            tmp_frame = sw_frame;
-        } else
-            tmp_frame = frame;
-
-        size = av_image_get_buffer_size(tmp_frame->format, tmp_frame->width,
-                                        tmp_frame->height, 1);
-        buffer = av_malloc(size);
-        if (!buffer) {
-            fprintf(stderr, "Can not alloc buffer\n");
-            ret = AVERROR(ENOMEM);
-            goto fail;
-        }
-        ret = av_image_copy_to_buffer(buffer, size,
-                                      (const uint8_t * const *)tmp_frame->data,
-                                      (const int *)tmp_frame->linesize, tmp_frame->format,
-                                      tmp_frame->width, tmp_frame->height, 1);
-        if (ret < 0) {
-            fprintf(stderr, "Can not copy image to buffer\n");
-            goto fail;
-        }
-
-        if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
-            fprintf(stderr, "Failed to dump raw data.\n");
-            goto fail;
-        }
-
-    fail:
-        av_frame_free(&frame);
-        av_frame_free(&sw_frame);
-        av_freep(&buffer);
-        if (ret < 0)
-            return ret;
-    }
-}
-
-int main(int argc, char *argv[])
-{
-    AVFormatContext *input_ctx = NULL;
-    int video_stream, ret;
-    AVStream *video = NULL;
-    AVCodecContext *decoder_ctx = NULL;
-    AVCodec *decoder = NULL;
-    AVPacket packet;
-    enum AVHWDeviceType type;
-    int i;
-
-    if (argc < 4) {
-        fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
-        return -1;
-    }
-
-    type = av_hwdevice_find_type_by_name(argv[1]);
-    if (type == AV_HWDEVICE_TYPE_NONE) {
-        fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
-        fprintf(stderr, "Available device types:");
-        while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
-            fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
-        fprintf(stderr, "\n");
-        return -1;
-    }
-
-    /* open the input file */
-    if (avformat_open_input(&input_ctx, argv[2], NULL, NULL) != 0) {
-        fprintf(stderr, "Cannot open input file '%s'\n", argv[2]);
-        return -1;
-    }
-
-    if (avformat_find_stream_info(input_ctx, NULL) < 0) {
-        fprintf(stderr, "Cannot find input stream information.\n");
-        return -1;
-    }
-
-    /* find the video stream information */
-    ret = av_find_best_stream(input_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot find a video stream in the input file\n");
-        return -1;
-    }
-    video_stream = ret;
-
-    for (i = 0;; i++) {
-        const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
-        if (!config) {
-            fprintf(stderr, "Decoder %s does not support device type %s.\n",
-                    decoder->name, av_hwdevice_get_type_name(type));
-            return -1;
-        }
-        if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
-            config->device_type == type) {
-            hw_pix_fmt = config->pix_fmt;
-            break;
-        }
-    }
-
-    if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
-        return AVERROR(ENOMEM);
-
-    video = input_ctx->streams[video_stream];
-    if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
-        return -1;
-
-    decoder_ctx->get_format  = get_hw_format;
-    av_opt_set_int(decoder_ctx, "refcounted_frames", 1, 0);
-
-    if (hw_decoder_init(decoder_ctx, type) < 0)
-        return -1;
-
-    if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0) {
-        fprintf(stderr, "Failed to open codec for stream #%u\n", video_stream);
-        return -1;
-    }
-
-    /* open the file to dump raw data */
-    output_file = fopen(argv[3], "w+");
-
-    /* actual decoding and dump the raw data */
-    while (ret >= 0) {
-        if ((ret = av_read_frame(input_ctx, &packet)) < 0)
-            break;
-
-        if (video_stream == packet.stream_index)
-            ret = decode_write(decoder_ctx, &packet);
-
-        av_packet_unref(&packet);
-    }
-
-    /* flush the decoder */
-    packet.data = NULL;
-    packet.size = 0;
-    ret = decode_write(decoder_ctx, &packet);
-    av_packet_unref(&packet);
-
-    if (output_file)
-        fclose(output_file);
-    avcodec_free_context(&decoder_ctx);
-    avformat_close_input(&input_ctx);
-    av_buffer_unref(&hw_device_ctx);
-
-    return 0;
-}

doc/examples/metadata.c

@@ -44,6 +44,7 @@ int main (int argc, char **argv)
         return 1;
     }
 
+    av_register_all();
     if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
         return ret;
 

doc/examples/muxing.c

@@ -335,15 +335,15 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
         if (ret < 0)
             exit(1);
 
-        /* convert to destination format */
-        ret = swr_convert(ost->swr_ctx,
-                          ost->frame->data, dst_nb_samples,
-                          (const uint8_t **)frame->data, frame->nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            exit(1);
-        }
-        frame = ost->frame;
+            /* convert to destination format */
+            ret = swr_convert(ost->swr_ctx,
+                              ost->frame->data, dst_nb_samples,
+                              (const uint8_t **)frame->data, frame->nb_samples);
+            if (ret < 0) {
+                fprintf(stderr, "Error while converting\n");
+                exit(1);
+            }
+            frame = ost->frame;
 
         frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
         ost->samples_count += dst_nb_samples;
@@ -440,7 +440,15 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
 static void fill_yuv_image(AVFrame *pict, int frame_index,
                            int width, int height)
 {
-    int x, y, i;
+    int x, y, i, ret;
+
+    /* when we pass a frame to the encoder, it may keep a reference to it
+     * internally;
+     * make sure we do not overwrite it here
+     */
+    ret = av_frame_make_writable(pict);
+    if (ret < 0)
+        exit(1);
 
     i = frame_index;
 
@@ -467,11 +475,6 @@ static AVFrame *get_video_frame(OutputStream *ost)
                       STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
         return NULL;
 
-    /* when we pass a frame to the encoder, it may keep a reference to it
-     * internally; make sure we do not overwrite it here */
-    if (av_frame_make_writable(ost->frame) < 0)
-        exit(1);
-
     if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
         /* as we only generate a YUV420P picture, we must convert it
          * to the codec pixel format if needed */
@@ -488,9 +491,9 @@ static AVFrame *get_video_frame(OutputStream *ost)
             }
         }
         fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
-        sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
-                  ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
-                  ost->frame->linesize);
+        sws_scale(ost->sws_ctx,
+                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
+                  0, c->height, ost->frame->data, ost->frame->linesize);
     } else {
         fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
     }
@@ -564,6 +567,9 @@ int main(int argc, char **argv)
     AVDictionary *opt = NULL;
     int i;
 
+    /* Initialize libavcodec, and register all codecs and formats. */
+    av_register_all();
+
     if (argc < 2) {
         printf("usage: %s output_file\n"
                "API example program to output a media file with libavformat.\n"

doc/examples/qsvdec.c

@@ -26,55 +26,185 @@
  *
  * @example qsvdec.c
  * This example shows how to do QSV-accelerated H.264 decoding with output
- * frames in the GPU video surfaces.
+ * frames in the VA-API video surfaces.
  */
 
 #include "config.h"
 
 #include <stdio.h>
 
+#include <mfx/mfxvideo.h>
+
+#include <va/va.h>
+#include <va/va_x11.h>
+#include <X11/Xlib.h>
+
 #include "libavformat/avformat.h"
 #include "libavformat/avio.h"
 
 #include "libavcodec/avcodec.h"
+#include "libavcodec/qsv.h"
 
-#include "libavutil/buffer.h"
 #include "libavutil/error.h"
-#include "libavutil/hwcontext.h"
-#include "libavutil/hwcontext_qsv.h"
 #include "libavutil/mem.h"
 
 typedef struct DecodeContext {
-    AVBufferRef *hw_device_ref;
+    mfxSession mfx_session;
+    VADisplay va_dpy;
+
+    VASurfaceID *surfaces;
+    mfxMemId    *surface_ids;
+    int         *surface_used;
+    int       nb_surfaces;
+
+    mfxFrameInfo frame_info;
 } DecodeContext;
 
+static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
+                             mfxFrameAllocResponse *resp)
+{
+    DecodeContext *decode = pthis;
+    int err, i;
+
+    if (decode->surfaces) {
+        fprintf(stderr, "Multiple allocation requests.\n");
+        return MFX_ERR_MEMORY_ALLOC;
+    }
+    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
+        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
+        return MFX_ERR_UNSUPPORTED;
+    }
+    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
+        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
+        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
+        fprintf(stderr, "Unsupported surface properties.\n");
+        return MFX_ERR_UNSUPPORTED;
+    }
+
+    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
+    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
+    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
+    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
+        goto fail;
+
+    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
+                           req->Info.Width, req->Info.Height,
+                           decode->surfaces, req->NumFrameSuggested,
+                           NULL, 0);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Error allocating VA surfaces\n");
+        goto fail;
+    }
+    decode->nb_surfaces = req->NumFrameSuggested;
+
+    for (i = 0; i < decode->nb_surfaces; i++)
+        decode->surface_ids[i] = &decode->surfaces[i];
+
+    resp->mids           = decode->surface_ids;
+    resp->NumFrameActual = decode->nb_surfaces;
+
+    decode->frame_info = req->Info;
+
+    return MFX_ERR_NONE;
+fail:
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+
+    return MFX_ERR_MEMORY_ALLOC;
+}
+
+static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
+{
+    return MFX_ERR_NONE;
+}
+
+static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
+{
+    *hdl = mid;
+    return MFX_ERR_NONE;
+}
+
+static void free_surfaces(DecodeContext *decode)
+{
+    if (decode->surfaces)
+        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+    decode->nb_surfaces = 0;
+}
+
+static void free_buffer(void *opaque, uint8_t *data)
+{
+    int *used = opaque;
+    *used = 0;
+    av_freep(&data);
+}
+
+static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
+{
+    DecodeContext *decode = avctx->opaque;
+
+    mfxFrameSurface1 *surf;
+    AVBufferRef *surf_buf;
+    int idx;
+
+    for (idx = 0; idx < decode->nb_surfaces; idx++) {
+        if (!decode->surface_used[idx])
+            break;
+    }
+    if (idx == decode->nb_surfaces) {
+        fprintf(stderr, "No free surfaces\n");
+        return AVERROR(ENOMEM);
+    }
+
+    surf = av_mallocz(sizeof(*surf));
+    if (!surf)
+        return AVERROR(ENOMEM);
+    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
+                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
+    if (!surf_buf) {
+        av_freep(&surf);
+        return AVERROR(ENOMEM);
+    }
+
+    surf->Info       = decode->frame_info;
+    surf->Data.MemId = &decode->surfaces[idx];
+
+    frame->buf[0]  = surf_buf;
+    frame->data[3] = (uint8_t*)surf;
+
+    decode->surface_used[idx] = 1;
+
+    return 0;
+}
+
 static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
 {
     while (*pix_fmts != AV_PIX_FMT_NONE) {
         if (*pix_fmts == AV_PIX_FMT_QSV) {
-            DecodeContext *decode = avctx->opaque;
-            AVHWFramesContext  *frames_ctx;
-            AVQSVFramesContext *frames_hwctx;
-            int ret;
+            if (!avctx->hwaccel_context) {
+                DecodeContext *decode = avctx->opaque;
+                AVQSVContext *qsv = av_qsv_alloc_context();
+                if (!qsv)
+                    return AV_PIX_FMT_NONE;
 
-            /* create a pool of surfaces to be used by the decoder */
-            avctx->hw_frames_ctx = av_hwframe_ctx_alloc(decode->hw_device_ref);
-            if (!avctx->hw_frames_ctx)
-                return AV_PIX_FMT_NONE;
-            frames_ctx   = (AVHWFramesContext*)avctx->hw_frames_ctx->data;
-            frames_hwctx = frames_ctx->hwctx;
+                qsv->session   = decode->mfx_session;
+                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
 
-            frames_ctx->format            = AV_PIX_FMT_QSV;
-            frames_ctx->sw_format         = avctx->sw_pix_fmt;
-            frames_ctx->width             = FFALIGN(avctx->coded_width,  32);
-            frames_ctx->height            = FFALIGN(avctx->coded_height, 32);
-            frames_ctx->initial_pool_size = 32;
-
-            frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
-
-            ret = av_hwframe_ctx_init(avctx->hw_frames_ctx);
-            if (ret < 0)
-                return AV_PIX_FMT_NONE;
+                avctx->hwaccel_context = qsv;
+            }
 
             return AV_PIX_FMT_QSV;
         }
@@ -88,47 +218,86 @@ static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
 }
 
 static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
-                         AVFrame *frame, AVFrame *sw_frame,
-                         AVPacket *pkt, AVIOContext *output_ctx)
+                         AVFrame *frame, AVPacket *pkt,
+                         AVIOContext *output_ctx)
 {
     int ret = 0;
+    int got_frame = 1;
 
-    ret = avcodec_send_packet(decoder_ctx, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error during decoding\n");
-        return ret;
-    }
-
-    while (ret >= 0) {
-        int i, j;
-
-        ret = avcodec_receive_frame(decoder_ctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            break;
-        else if (ret < 0) {
+    while (pkt->size > 0 || (!pkt->data && got_frame)) {
+        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
+        if (ret < 0) {
             fprintf(stderr, "Error during decoding\n");
             return ret;
         }
 
+        pkt->data += ret;
+        pkt->size -= ret;
+
         /* A real program would do something useful with the decoded frame here.
          * We just retrieve the raw data and write it to a file, which is rather
          * useless but pedagogic. */
-        ret = av_hwframe_transfer_data(sw_frame, frame, 0);
-        if (ret < 0) {
-            fprintf(stderr, "Error transferring the data to system memory\n");
-            goto fail;
-        }
+        if (got_frame) {
+            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
+            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
 
-        for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++)
-            for (j = 0; j < (sw_frame->height >> (i > 0)); j++)
-                avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], sw_frame->width);
+            VAImageFormat img_fmt = {
+                .fourcc         = VA_FOURCC_NV12,
+                .byte_order     = VA_LSB_FIRST,
+                .bits_per_pixel = 8,
+                .depth          = 8,
+            };
+
+            VAImage img;
+
+            VAStatus err;
+            uint8_t *data;
+            int i, j;
+
+            img.buf      = VA_INVALID_ID;
+            img.image_id = VA_INVALID_ID;
+
+            err = vaCreateImage(decode->va_dpy, &img_fmt,
+                                frame->width, frame->height, &img);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error creating an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaGetImage(decode->va_dpy, surface, 0, 0,
+                             frame->width, frame->height,
+                             img.image_id);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error getting an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error mapping the image buffer: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            for (i = 0; i < img.num_planes; i++)
+                for (j = 0; j < (img.height >> (i > 0)); j++)
+                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
 
 fail:
-        av_frame_unref(sw_frame);
-        av_frame_unref(frame);
+            if (img.buf != VA_INVALID_ID)
+                vaUnmapBuffer(decode->va_dpy, img.buf);
+            if (img.image_id != VA_INVALID_ID)
+                vaDestroyImage(decode->va_dpy, img.image_id);
+            av_frame_unref(frame);
 
-        if (ret < 0)
-            return ret;
+            if (ret < 0)
+                return ret;
+        }
     }
 
     return 0;
@@ -142,13 +311,30 @@ int main(int argc, char **argv)
     const AVCodec *decoder;
 
     AVPacket pkt = { 0 };
-    AVFrame *frame = NULL, *sw_frame = NULL;
+    AVFrame *frame = NULL;
 
     DecodeContext decode = { NULL };
 
+    Display *dpy = NULL;
+    int va_ver_major, va_ver_minor;
+
+    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
+    mfxVersion mfx_ver = { { 1, 1 } };
+
+    mfxFrameAllocator frame_allocator = {
+        .pthis = &decode,
+        .Alloc = frame_alloc,
+        .Lock  = frame_lock,
+        .Unlock = frame_unlock,
+        .GetHDL = frame_get_hdl,
+        .Free   = frame_free,
+    };
+
     AVIOContext *output_ctx = NULL;
 
-    int ret, i;
+    int ret, i, err;
+
+    av_register_all();
 
     if (argc < 3) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -176,13 +362,34 @@ int main(int argc, char **argv)
         goto finish;
     }
 
-    /* open the hardware device */
-    ret = av_hwdevice_ctx_create(&decode.hw_device_ref, AV_HWDEVICE_TYPE_QSV,
-                                 "auto", NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot open the hardware device\n");
+    /* initialize VA-API */
+    dpy = XOpenDisplay(NULL);
+    if (!dpy) {
+        fprintf(stderr, "Cannot open the X display\n");
         goto finish;
     }
+    decode.va_dpy = vaGetDisplay(dpy);
+    if (!decode.va_dpy) {
+        fprintf(stderr, "Cannot open the VA display\n");
+        goto finish;
+    }
+
+    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
+        goto finish;
+    }
+    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
+
+    /* initialize an MFX session */
+    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
+    if (err != MFX_ERR_NONE) {
+        fprintf(stderr, "Error initializing an MFX session\n");
+        goto finish;
+    }
+
+    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
+    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
 
     /* initialize the decoder */
     decoder = avcodec_find_decoder_by_name("h264_qsv");
@@ -208,8 +415,10 @@ int main(int argc, char **argv)
                video_st->codecpar->extradata_size);
         decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
     }
+    decoder_ctx->refcounted_frames = 1;
 
     decoder_ctx->opaque      = &decode;
+    decoder_ctx->get_buffer2 = get_buffer;
     decoder_ctx->get_format  = get_format;
 
     ret = avcodec_open2(decoder_ctx, NULL, NULL);
@@ -225,9 +434,8 @@ int main(int argc, char **argv)
         goto finish;
     }
 
-    frame    = av_frame_alloc();
-    sw_frame = av_frame_alloc();
-    if (!frame || !sw_frame) {
+    frame = av_frame_alloc();
+    if (!frame) {
         ret = AVERROR(ENOMEM);
         goto finish;
     }
@@ -239,7 +447,7 @@ int main(int argc, char **argv)
             break;
 
         if (pkt.stream_index == video_st->index)
-            ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
+            ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
 
         av_packet_unref(&pkt);
     }
@@ -247,7 +455,7 @@ int main(int argc, char **argv)
     /* flush the decoder */
     pkt.data = NULL;
     pkt.size = 0;
-    ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
+    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
 
 finish:
     if (ret < 0) {
@@ -259,11 +467,19 @@ finish:
     avformat_close_input(&input_ctx);
 
     av_frame_free(&frame);
-    av_frame_free(&sw_frame);
 
+    if (decoder_ctx)
+        av_freep(&decoder_ctx->hwaccel_context);
     avcodec_free_context(&decoder_ctx);
 
-    av_buffer_unref(&decode.hw_device_ref);
+    free_surfaces(&decode);
+
+    if (decode.mfx_session)
+        MFXClose(decode.mfx_session);
+    if (decode.va_dpy)
+        vaTerminate(decode.va_dpy);
+    if (dpy)
+        XCloseDisplay(dpy);
 
     avio_close(output_ctx);
 

doc/examples/remuxing.c

@@ -50,9 +50,6 @@ int main(int argc, char **argv)
     AVPacket pkt;
     const char *in_filename, *out_filename;
     int ret, i;
-    int stream_index = 0;
-    int *stream_mapping = NULL;
-    int stream_mapping_size = 0;
 
     if (argc < 3) {
         printf("usage: %s input output\n"
@@ -65,6 +62,8 @@ int main(int argc, char **argv)
     in_filename  = argv[1];
     out_filename = argv[2];
 
+    av_register_all();
+
     if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
         fprintf(stderr, "Could not open input file '%s'", in_filename);
         goto end;
@@ -84,42 +83,25 @@ int main(int argc, char **argv)
         goto end;
     }
 
-    stream_mapping_size = ifmt_ctx->nb_streams;
-    stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
-    if (!stream_mapping) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
     ofmt = ofmt_ctx->oformat;
 
     for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *out_stream;
         AVStream *in_stream = ifmt_ctx->streams[i];
-        AVCodecParameters *in_codecpar = in_stream->codecpar;
-
-        if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
-            in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO &&
-            in_codecpar->codec_type != AVMEDIA_TYPE_SUBTITLE) {
-            stream_mapping[i] = -1;
-            continue;
-        }
-
-        stream_mapping[i] = stream_index++;
-
-        out_stream = avformat_new_stream(ofmt_ctx, NULL);
+        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
         if (!out_stream) {
             fprintf(stderr, "Failed allocating output stream\n");
             ret = AVERROR_UNKNOWN;
             goto end;
         }
 
-        ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
+        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
         if (ret < 0) {
-            fprintf(stderr, "Failed to copy codec parameters\n");
+            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
             goto end;
         }
-        out_stream->codecpar->codec_tag = 0;
+        out_stream->codec->codec_tag = 0;
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
     }
     av_dump_format(ofmt_ctx, 0, out_filename, 1);
 
@@ -145,14 +127,8 @@ int main(int argc, char **argv)
             break;
 
         in_stream  = ifmt_ctx->streams[pkt.stream_index];
-        if (pkt.stream_index >= stream_mapping_size ||
-            stream_mapping[pkt.stream_index] < 0) {
-            av_packet_unref(&pkt);
-            continue;
-        }
-
-        pkt.stream_index = stream_mapping[pkt.stream_index];
         out_stream = ofmt_ctx->streams[pkt.stream_index];
+
         log_packet(ifmt_ctx, &pkt, "in");
 
         /* copy packet */
@@ -180,8 +156,6 @@ end:
         avio_closep(&ofmt_ctx->pb);
     avformat_free_context(ofmt_ctx);
 
-    av_freep(&stream_mapping);
-
     if (ret < 0 && ret != AVERROR_EOF) {
         fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
         return 1;

doc/examples/transcode_aac.c

@@ -1,6 +1,4 @@
 /*
- * Copyright (c) 2013-2018 Andreas Unterweger
- *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
@@ -10,7 +8,7 @@
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * 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
@@ -20,11 +18,10 @@
 
 /**
  * @file
- * Simple audio converter
+ * simple audio converter
  *
  * @example transcode_aac.c
  * Convert an input audio file to AAC in an MP4 container using FFmpeg.
- * Formats other than MP4 are supported based on the output file extension.
  * @author Andreas Unterweger (dustsigns@gmail.com)
  */
 
@@ -43,18 +40,24 @@
 
 #include "libswresample/swresample.h"
 
-/* The output bit rate in bit/s */
+/** The output bit rate in kbit/s */
 #define OUTPUT_BIT_RATE 96000
-/* The number of output channels */
+/** The number of output channels */
 #define OUTPUT_CHANNELS 2
 
 /**
- * Open an input file and the required decoder.
- * @param      filename             File to be opened
- * @param[out] input_format_context Format context of opened file
- * @param[out] input_codec_context  Codec context of opened file
- * @return Error code (0 if successful)
+ * Convert an error code into a text message.
+ * @param error Error code to be converted
+ * @return Corresponding error text (not thread-safe)
  */
+static const char *get_error_text(const int error)
+{
+    static char error_buffer[255];
+    av_strerror(error, error_buffer, sizeof(error_buffer));
+    return error_buffer;
+}
+
+/** Open an input file and the required decoder. */
 static int open_input_file(const char *filename,
                            AVFormatContext **input_format_context,
                            AVCodecContext **input_codec_context)
@@ -63,24 +66,24 @@ static int open_input_file(const char *filename,
     AVCodec *input_codec;
     int error;
 
-    /* Open the input file to read from it. */
+    /** Open the input file to read from it. */
     if ((error = avformat_open_input(input_format_context, filename, NULL,
                                      NULL)) < 0) {
         fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
-                filename, av_err2str(error));
+                filename, get_error_text(error));
         *input_format_context = NULL;
         return error;
     }
 
-    /* Get information on the input file (number of streams etc.). */
+    /** Get information on the input file (number of streams etc.). */
     if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
         fprintf(stderr, "Could not open find stream info (error '%s')\n",
-                av_err2str(error));
+                get_error_text(error));
         avformat_close_input(input_format_context);
         return error;
     }
 
-    /* Make sure that there is only one stream in the input file. */
+    /** Make sure that there is only one stream in the input file. */
     if ((*input_format_context)->nb_streams != 1) {
         fprintf(stderr, "Expected one audio input stream, but found %d\n",
                 (*input_format_context)->nb_streams);
@@ -88,14 +91,14 @@ static int open_input_file(const char *filename,
         return AVERROR_EXIT;
     }
 
-    /* Find a decoder for the audio stream. */
+    /** Find a decoder for the audio stream. */
     if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
         fprintf(stderr, "Could not find input codec\n");
         avformat_close_input(input_format_context);
         return AVERROR_EXIT;
     }
 
-    /* Allocate a new decoding context. */
+    /** allocate a new decoding context */
     avctx = avcodec_alloc_context3(input_codec);
     if (!avctx) {
         fprintf(stderr, "Could not allocate a decoding context\n");
@@ -103,7 +106,7 @@ static int open_input_file(const char *filename,
         return AVERROR(ENOMEM);
     }
 
-    /* Initialize the stream parameters with demuxer information. */
+    /** initialize the stream parameters with demuxer information */
     error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
     if (error < 0) {
         avformat_close_input(input_format_context);
@@ -111,16 +114,16 @@ static int open_input_file(const char *filename,
         return error;
     }
 
-    /* Open the decoder for the audio stream to use it later. */
+    /** Open the decoder for the audio stream to use it later. */
     if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
         fprintf(stderr, "Could not open input codec (error '%s')\n",
-                av_err2str(error));
+                get_error_text(error));
         avcodec_free_context(&avctx);
         avformat_close_input(input_format_context);
         return error;
     }
 
-    /* Save the decoder context for easier access later. */
+    /** Save the decoder context for easier access later. */
     *input_codec_context = avctx;
 
     return 0;
@@ -130,11 +133,6 @@ static int open_input_file(const char *filename,
  * Open an output file and the required encoder.
  * Also set some basic encoder parameters.
  * Some of these parameters are based on the input file's parameters.
- * @param      filename              File to be opened
- * @param      input_codec_context   Codec context of input file
- * @param[out] output_format_context Format context of output file
- * @param[out] output_codec_context  Codec context of output file
- * @return Error code (0 if successful)
  */
 static int open_output_file(const char *filename,
                             AVCodecContext *input_codec_context,
@@ -147,43 +145,40 @@ static int open_output_file(const char *filename,
     AVCodec *output_codec          = NULL;
     int error;
 
-    /* Open the output file to write to it. */
+    /** Open the output file to write to it. */
     if ((error = avio_open(&output_io_context, filename,
                            AVIO_FLAG_WRITE)) < 0) {
         fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
-                filename, av_err2str(error));
+                filename, get_error_text(error));
         return error;
     }
 
-    /* Create a new format context for the output container format. */
+    /** Create a new format context for the output container format. */
     if (!(*output_format_context = avformat_alloc_context())) {
         fprintf(stderr, "Could not allocate output format context\n");
         return AVERROR(ENOMEM);
     }
 
-    /* Associate the output file (pointer) with the container format context. */
+    /** Associate the output file (pointer) with the container format context. */
     (*output_format_context)->pb = output_io_context;
 
-    /* Guess the desired container format based on the file extension. */
+    /** Guess the desired container format based on the file extension. */
     if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
                                                               NULL))) {
         fprintf(stderr, "Could not find output file format\n");
         goto cleanup;
     }
 
-    if (!((*output_format_context)->url = av_strdup(filename))) {
-        fprintf(stderr, "Could not allocate url.\n");
-        error = AVERROR(ENOMEM);
-        goto cleanup;
-    }
+    av_strlcpy((*output_format_context)->filename, filename,
+               sizeof((*output_format_context)->filename));
 
-    /* Find the encoder to be used by its name. */
+    /** Find the encoder to be used by its name. */
     if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
         fprintf(stderr, "Could not find an AAC encoder.\n");
         goto cleanup;
     }
 
-    /* Create a new audio stream in the output file container. */
+    /** Create a new audio stream in the output file container. */
     if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
         fprintf(stderr, "Could not create new stream\n");
         error = AVERROR(ENOMEM);
@@ -197,30 +192,34 @@ static int open_output_file(const char *filename,
         goto cleanup;
     }
 
-    /* Set the basic encoder parameters.
-     * The input file's sample rate is used to avoid a sample rate conversion. */
+    /**
+     * Set the basic encoder parameters.
+     * The input file's sample rate is used to avoid a sample rate conversion.
+     */
     avctx->channels       = OUTPUT_CHANNELS;
     avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
     avctx->sample_rate    = input_codec_context->sample_rate;
     avctx->sample_fmt     = output_codec->sample_fmts[0];
     avctx->bit_rate       = OUTPUT_BIT_RATE;
 
-    /* Allow the use of the experimental AAC encoder. */
+    /** Allow the use of the experimental AAC encoder */
     avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
 
-    /* Set the sample rate for the container. */
+    /** Set the sample rate for the container. */
     stream->time_base.den = input_codec_context->sample_rate;
     stream->time_base.num = 1;
 
-    /* Some container formats (like MP4) require global headers to be present.
-     * Mark the encoder so that it behaves accordingly. */
+    /**
+     * Some container formats (like MP4) require global headers to be present
+     * Mark the encoder so that it behaves accordingly.
+     */
     if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
         avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
-    /* Open the encoder for the audio stream to use it later. */
+    /** Open the encoder for the audio stream to use it later. */
     if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
         fprintf(stderr, "Could not open output codec (error '%s')\n",
-                av_err2str(error));
+                get_error_text(error));
         goto cleanup;
     }
 
@@ -230,7 +229,7 @@ static int open_output_file(const char *filename,
         goto cleanup;
     }
 
-    /* Save the encoder context for easier access later. */
+    /** Save the encoder context for easier access later. */
     *output_codec_context = avctx;
 
     return 0;
@@ -243,23 +242,16 @@ cleanup:
     return error < 0 ? error : AVERROR_EXIT;
 }
 
-/**
- * Initialize one data packet for reading or writing.
- * @param packet Packet to be initialized
- */
+/** Initialize one data packet for reading or writing. */
 static void init_packet(AVPacket *packet)
 {
     av_init_packet(packet);
-    /* Set the packet data and size so that it is recognized as being empty. */
+    /** Set the packet data and size so that it is recognized as being empty. */
     packet->data = NULL;
     packet->size = 0;
 }
 
-/**
- * Initialize one audio frame for reading from the input file.
- * @param[out] frame Frame to be initialized
- * @return Error code (0 if successful)
- */
+/** Initialize one audio frame for reading from the input file */
 static int init_input_frame(AVFrame **frame)
 {
     if (!(*frame = av_frame_alloc())) {
@@ -273,10 +265,6 @@ static int init_input_frame(AVFrame **frame)
  * Initialize the audio resampler based on the input and output codec settings.
  * If the input and output sample formats differ, a conversion is required
  * libswresample takes care of this, but requires initialization.
- * @param      input_codec_context  Codec context of the input file
- * @param      output_codec_context Codec context of the output file
- * @param[out] resample_context     Resample context for the required conversion
- * @return Error code (0 if successful)
  */
 static int init_resampler(AVCodecContext *input_codec_context,
                           AVCodecContext *output_codec_context,
@@ -284,7 +272,7 @@ static int init_resampler(AVCodecContext *input_codec_context,
 {
         int error;
 
-        /*
+        /**
          * Create a resampler context for the conversion.
          * Set the conversion parameters.
          * Default channel layouts based on the number of channels
@@ -303,14 +291,14 @@ static int init_resampler(AVCodecContext *input_codec_context,
             fprintf(stderr, "Could not allocate resample context\n");
             return AVERROR(ENOMEM);
         }
-        /*
+        /**
         * Perform a sanity check so that the number of converted samples is
         * not greater than the number of samples to be converted.
         * If the sample rates differ, this case has to be handled differently
         */
         av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
 
-        /* Open the resampler with the specified parameters. */
+        /** Open the resampler with the specified parameters. */
         if ((error = swr_init(*resample_context)) < 0) {
             fprintf(stderr, "Could not open resample context\n");
             swr_free(resample_context);
@@ -319,15 +307,10 @@ static int init_resampler(AVCodecContext *input_codec_context,
     return 0;
 }
 
-/**
- * Initialize a FIFO buffer for the audio samples to be encoded.
- * @param[out] fifo                 Sample buffer
- * @param      output_codec_context Codec context of the output file
- * @return Error code (0 if successful)
- */
+/** Initialize a FIFO buffer for the audio samples to be encoded. */
 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
-    /* Create the FIFO buffer based on the specified output sample format. */
+    /** Create the FIFO buffer based on the specified output sample format. */
     if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
                                       output_codec_context->channels, 1))) {
         fprintf(stderr, "Could not allocate FIFO\n");
@@ -336,103 +319,69 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
     return 0;
 }
 
-/**
- * Write the header of the output file container.
- * @param output_format_context Format context of the output file
- * @return Error code (0 if successful)
- */
+/** Write the header of the output file container. */
 static int write_output_file_header(AVFormatContext *output_format_context)
 {
     int error;
     if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
         fprintf(stderr, "Could not write output file header (error '%s')\n",
-                av_err2str(error));
+                get_error_text(error));
         return error;
     }
     return 0;
 }
 
-/**
- * Decode one audio frame from the input file.
- * @param      frame                Audio frame to be decoded
- * @param      input_format_context Format context of the input file
- * @param      input_codec_context  Codec context of the input file
- * @param[out] data_present         Indicates whether data has been decoded
- * @param[out] finished             Indicates whether the end of file has
- *                                  been reached and all data has been
- *                                  decoded. If this flag is false, there
- *                                  is more data to be decoded, i.e., this
- *                                  function has to be called again.
- * @return Error code (0 if successful)
- */
+/** Decode one audio frame from the input file. */
 static int decode_audio_frame(AVFrame *frame,
                               AVFormatContext *input_format_context,
                               AVCodecContext *input_codec_context,
                               int *data_present, int *finished)
 {
-    /* Packet used for temporary storage. */
+    /** Packet used for temporary storage. */
     AVPacket input_packet;
     int error;
     init_packet(&input_packet);
 
-    /* Read one audio frame from the input file into a temporary packet. */
+    /** Read one audio frame from the input file into a temporary packet. */
     if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
-        /* If we are at the end of the file, flush the decoder below. */
+        /** If we are at the end of the file, flush the decoder below. */
         if (error == AVERROR_EOF)
             *finished = 1;
         else {
             fprintf(stderr, "Could not read frame (error '%s')\n",
-                    av_err2str(error));
+                    get_error_text(error));
             return error;
         }
     }
 
-    /* Send the audio frame stored in the temporary packet to the decoder.
-     * The input audio stream decoder is used to do this. */
-    if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
-        fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
-                av_err2str(error));
+    /**
+     * Decode the audio frame stored in the temporary packet.
+     * The input audio stream decoder is used to do this.
+     * If we are at the end of the file, pass an empty packet to the decoder
+     * to flush it.
+     */
+    if ((error = avcodec_decode_audio4(input_codec_context, frame,
+                                       data_present, &input_packet)) < 0) {
+        fprintf(stderr, "Could not decode frame (error '%s')\n",
+                get_error_text(error));
+        av_packet_unref(&input_packet);
         return error;
     }
 
-    /* Receive one frame from the decoder. */
-    error = avcodec_receive_frame(input_codec_context, frame);
-    /* If the decoder asks for more data to be able to decode a frame,
-     * return indicating that no data is present. */
-    if (error == AVERROR(EAGAIN)) {
-        error = 0;
-        goto cleanup;
-    /* If the end of the input file is reached, stop decoding. */
-    } else if (error == AVERROR_EOF) {
-        *finished = 1;
-        error = 0;
-        goto cleanup;
-    } else if (error < 0) {
-        fprintf(stderr, "Could not decode frame (error '%s')\n",
-                av_err2str(error));
-        goto cleanup;
-    /* Default case: Return decoded data. */
-    } else {
-        *data_present = 1;
-        goto cleanup;
-    }
-
-cleanup:
+    /**
+     * If the decoder has not been flushed completely, we are not finished,
+     * so that this function has to be called again.
+     */
+    if (*finished && *data_present)
+        *finished = 0;
     av_packet_unref(&input_packet);
-    return error;
+    return 0;
 }
 
 /**
  * Initialize a temporary storage for the specified number of audio samples.
  * The conversion requires temporary storage due to the different format.
  * The number of audio samples to be allocated is specified in frame_size.
- * @param[out] converted_input_samples Array of converted samples. The
- *                                     dimensions are reference, channel
- *                                     (for multi-channel audio), sample.
- * @param      output_codec_context    Codec context of the output file
- * @param      frame_size              Number of samples to be converted in
- *                                     each round
- * @return Error code (0 if successful)
  */
 static int init_converted_samples(uint8_t ***converted_input_samples,
                                   AVCodecContext *output_codec_context,
@@ -440,7 +389,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
 {
     int error;
 
-    /* Allocate as many pointers as there are audio channels.
+    /**
+     * Allocate as many pointers as there are audio channels.
      * Each pointer will later point to the audio samples of the corresponding
      * channels (although it may be NULL for interleaved formats).
      */
@@ -450,15 +400,17 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
         return AVERROR(ENOMEM);
     }
 
-    /* Allocate memory for the samples of all channels in one consecutive
-     * block for convenience. */
+    /**
+     * Allocate memory for the samples of all channels in one consecutive
+     * block for convenience.
+     */
     if ((error = av_samples_alloc(*converted_input_samples, NULL,
                                   output_codec_context->channels,
                                   frame_size,
                                   output_codec_context->sample_fmt, 0)) < 0) {
         fprintf(stderr,
                 "Could not allocate converted input samples (error '%s')\n",
-                av_err2str(error));
+                get_error_text(error));
         av_freep(&(*converted_input_samples)[0]);
         free(*converted_input_samples);
         return error;
@@ -468,15 +420,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
 
 /**
  * Convert the input audio samples into the output sample format.
- * The conversion happens on a per-frame basis, the size of which is
- * specified by frame_size.
- * @param      input_data       Samples to be decoded. The dimensions are
- *                              channel (for multi-channel audio), sample.
- * @param[out] converted_data   Converted samples. The dimensions are channel
- *                              (for multi-channel audio), sample.
- * @param      frame_size       Number of samples to be converted
- * @param      resample_context Resample context for the conversion
- * @return Error code (0 if successful)
+ * The conversion happens on a per-frame basis, the size of which is specified
+ * by frame_size.
  */
 static int convert_samples(const uint8_t **input_data,
                            uint8_t **converted_data, const int frame_size,
@@ -484,40 +429,35 @@ static int convert_samples(const uint8_t **input_data,
 {
     int error;
 
-    /* Convert the samples using the resampler. */
+    /** Convert the samples using the resampler. */
     if ((error = swr_convert(resample_context,
                              converted_data, frame_size,
                              input_data    , frame_size)) < 0) {
         fprintf(stderr, "Could not convert input samples (error '%s')\n",
-                av_err2str(error));
+                get_error_text(error));
         return error;
     }
 
     return 0;
 }
 
-/**
- * Add converted input audio samples to the FIFO buffer for later processing.
- * @param fifo                    Buffer to add the samples to
- * @param converted_input_samples Samples to be added. The dimensions are channel
- *                                (for multi-channel audio), sample.
- * @param frame_size              Number of samples to be converted
- * @return Error code (0 if successful)
- */
+/** Add converted input audio samples to the FIFO buffer for later processing. */
 static int add_samples_to_fifo(AVAudioFifo *fifo,
                                uint8_t **converted_input_samples,
                                const int frame_size)
 {
     int error;
 
-    /* Make the FIFO as large as it needs to be to hold both,
-     * the old and the new samples. */
+    /**
+     * Make the FIFO as large as it needs to be to hold both,
+     * the old and the new samples.
+     */
     if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
         fprintf(stderr, "Could not reallocate FIFO\n");
         return error;
     }
 
-    /* Store the new samples in the FIFO buffer. */
+    /** Store the new samples in the FIFO buffer. */
     if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
                             frame_size) < frame_size) {
         fprintf(stderr, "Could not write data to FIFO\n");
@@ -527,20 +467,8 @@ static int add_samples_to_fifo(AVAudioFifo *fifo,
 }
 
 /**
- * Read one audio frame from the input file, decode, convert and store
+ * Read one audio frame from the input file, decodes, converts and stores
  * it in the FIFO buffer.
- * @param      fifo                 Buffer used for temporary storage
- * @param      input_format_context Format context of the input file
- * @param      input_codec_context  Codec context of the input file
- * @param      output_codec_context Codec context of the output file
- * @param      resampler_context    Resample context for the conversion
- * @param[out] finished             Indicates whether the end of file has
- *                                  been reached and all data has been
- *                                  decoded. If this flag is false,
- *                                  there is more data to be decoded,
- *                                  i.e., this function has to be called
- *                                  again.
- * @return Error code (0 if successful)
  */
 static int read_decode_convert_and_store(AVAudioFifo *fifo,
                                          AVFormatContext *input_format_context,
@@ -549,41 +477,45 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
                                          SwrContext *resampler_context,
                                          int *finished)
 {
-    /* Temporary storage of the input samples of the frame read from the file. */
+    /** Temporary storage of the input samples of the frame read from the file. */
     AVFrame *input_frame = NULL;
-    /* Temporary storage for the converted input samples. */
+    /** Temporary storage for the converted input samples. */
     uint8_t **converted_input_samples = NULL;
-    int data_present = 0;
+    int data_present;
     int ret = AVERROR_EXIT;
 
-    /* Initialize temporary storage for one input frame. */
+    /** Initialize temporary storage for one input frame. */
     if (init_input_frame(&input_frame))
         goto cleanup;
-    /* Decode one frame worth of audio samples. */
+    /** Decode one frame worth of audio samples. */
     if (decode_audio_frame(input_frame, input_format_context,
                            input_codec_context, &data_present, finished))
         goto cleanup;
-    /* If we are at the end of the file and there are no more samples
+    /**
+     * If we are at the end of the file and there are no more samples
      * in the decoder which are delayed, we are actually finished.
-     * This must not be treated as an error. */
-    if (*finished) {
+     * This must not be treated as an error.
+     */
+    if (*finished && !data_present) {
         ret = 0;
         goto cleanup;
     }
-    /* If there is decoded data, convert and store it. */
+    /** If there is decoded data, convert and store it */
     if (data_present) {
-        /* Initialize the temporary storage for the converted input samples. */
+        /** Initialize the temporary storage for the converted input samples. */
         if (init_converted_samples(&converted_input_samples, output_codec_context,
                                    input_frame->nb_samples))
             goto cleanup;
 
-        /* Convert the input samples to the desired output sample format.
-         * This requires a temporary storage provided by converted_input_samples. */
+        /**
+         * Convert the input samples to the desired output sample format.
+         * This requires a temporary storage provided by converted_input_samples.
+         */
         if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
                             input_frame->nb_samples, resampler_context))
             goto cleanup;
 
-        /* Add the converted input samples to the FIFO buffer for later processing. */
+        /** Add the converted input samples to the FIFO buffer for later processing. */
         if (add_samples_to_fifo(fifo, converted_input_samples,
                                 input_frame->nb_samples))
             goto cleanup;
@@ -604,10 +536,6 @@ cleanup:
 /**
  * Initialize one input frame for writing to the output file.
  * The frame will be exactly frame_size samples large.
- * @param[out] frame                Frame to be initialized
- * @param      output_codec_context Codec context of the output file
- * @param      frame_size           Size of the frame
- * @return Error code (0 if successful)
  */
 static int init_output_frame(AVFrame **frame,
                              AVCodecContext *output_codec_context,
@@ -615,27 +543,31 @@ static int init_output_frame(AVFrame **frame,
 {
     int error;
 
-    /* Create a new frame to store the audio samples. */
+    /** Create a new frame to store the audio samples. */
     if (!(*frame = av_frame_alloc())) {
         fprintf(stderr, "Could not allocate output frame\n");
         return AVERROR_EXIT;
     }
 
-    /* Set the frame's parameters, especially its size and format.
+    /**
+     * Set the frame's parameters, especially its size and format.
      * av_frame_get_buffer needs this to allocate memory for the
      * audio samples of the frame.
      * Default channel layouts based on the number of channels
-     * are assumed for simplicity. */
+     * are assumed for simplicity.
+     */
     (*frame)->nb_samples     = frame_size;
     (*frame)->channel_layout = output_codec_context->channel_layout;
     (*frame)->format         = output_codec_context->sample_fmt;
     (*frame)->sample_rate    = output_codec_context->sample_rate;
 
-    /* Allocate the samples of the created frame. This call will make
-     * sure that the audio frame can hold as many samples as specified. */
+    /**
+     * Allocate the samples of the created frame. This call will make
+     * sure that the audio frame can hold as many samples as specified.
+     */
     if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
-        fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
-                av_err2str(error));
+        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
+                get_error_text(error));
         av_frame_free(frame);
         return error;
     }
@@ -643,114 +575,87 @@ static int init_output_frame(AVFrame **frame,
     return 0;
 }
 
-/* Global timestamp for the audio frames. */
+/** Global timestamp for the audio frames */
 static int64_t pts = 0;
 
-/**
- * Encode one frame worth of audio to the output file.
- * @param      frame                 Samples to be encoded
- * @param      output_format_context Format context of the output file
- * @param      output_codec_context  Codec context of the output file
- * @param[out] data_present          Indicates whether data has been
- *                                   encoded
- * @return Error code (0 if successful)
- */
+/** Encode one frame worth of audio to the output file. */
 static int encode_audio_frame(AVFrame *frame,
                               AVFormatContext *output_format_context,
                               AVCodecContext *output_codec_context,
                               int *data_present)
 {
-    /* Packet used for temporary storage. */
+    /** Packet used for temporary storage. */
     AVPacket output_packet;
     int error;
     init_packet(&output_packet);
 
-    /* Set a timestamp based on the sample rate for the container. */
+    /** Set a timestamp based on the sample rate for the container. */
     if (frame) {
         frame->pts = pts;
         pts += frame->nb_samples;
     }
 
-    /* Send the audio frame stored in the temporary packet to the encoder.
-     * The output audio stream encoder is used to do this. */
-    error = avcodec_send_frame(output_codec_context, frame);
-    /* The encoder signals that it has nothing more to encode. */
-    if (error == AVERROR_EOF) {
-        error = 0;
-        goto cleanup;
-    } else if (error < 0) {
-        fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
-                av_err2str(error));
+    /**
+     * Encode the audio frame and store it in the temporary packet.
+     * The output audio stream encoder is used to do this.
+     */
+    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
+                                       frame, data_present)) < 0) {
+        fprintf(stderr, "Could not encode frame (error '%s')\n",
+                get_error_text(error));
+        av_packet_unref(&output_packet);
         return error;
     }
 
-    /* Receive one encoded frame from the encoder. */
-    error = avcodec_receive_packet(output_codec_context, &output_packet);
-    /* If the encoder asks for more data to be able to provide an
-     * encoded frame, return indicating that no data is present. */
-    if (error == AVERROR(EAGAIN)) {
-        error = 0;
-        goto cleanup;
-    /* If the last frame has been encoded, stop encoding. */
-    } else if (error == AVERROR_EOF) {
-        error = 0;
-        goto cleanup;
-    } else if (error < 0) {
-        fprintf(stderr, "Could not encode frame (error '%s')\n",
-                av_err2str(error));
-        goto cleanup;
-    /* Default case: Return encoded data. */
-    } else {
-        *data_present = 1;
+    /** Write one audio frame from the temporary packet to the output file. */
+    if (*data_present) {
+        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
+            fprintf(stderr, "Could not write frame (error '%s')\n",
+                    get_error_text(error));
+            av_packet_unref(&output_packet);
+            return error;
+        }
+
+        av_packet_unref(&output_packet);
     }
 
-    /* Write one audio frame from the temporary packet to the output file. */
-    if (*data_present &&
-        (error = av_write_frame(output_format_context, &output_packet)) < 0) {
-        fprintf(stderr, "Could not write frame (error '%s')\n",
-                av_err2str(error));
-        goto cleanup;
-    }
-
-cleanup:
-    av_packet_unref(&output_packet);
-    return error;
+    return 0;
 }
 
 /**
  * Load one audio frame from the FIFO buffer, encode and write it to the
  * output file.
- * @param fifo                  Buffer used for temporary storage
- * @param output_format_context Format context of the output file
- * @param output_codec_context  Codec context of the output file
- * @return Error code (0 if successful)
  */
 static int load_encode_and_write(AVAudioFifo *fifo,
                                  AVFormatContext *output_format_context,
                                  AVCodecContext *output_codec_context)
 {
-    /* Temporary storage of the output samples of the frame written to the file. */
+    /** Temporary storage of the output samples of the frame written to the file. */
     AVFrame *output_frame;
-    /* Use the maximum number of possible samples per frame.
+    /**
+     * Use the maximum number of possible samples per frame.
      * If there is less than the maximum possible frame size in the FIFO
-     * buffer use this number. Otherwise, use the maximum possible frame size. */
+     * buffer use this number. Otherwise, use the maximum possible frame size
+     */
     const int frame_size = FFMIN(av_audio_fifo_size(fifo),
                                  output_codec_context->frame_size);
     int data_written;
 
-    /* Initialize temporary storage for one output frame. */
+    /** Initialize temporary storage for one output frame. */
     if (init_output_frame(&output_frame, output_codec_context, frame_size))
         return AVERROR_EXIT;
 
-    /* Read as many samples from the FIFO buffer as required to fill the frame.
-     * The samples are stored in the frame temporarily. */
+    /**
+     * Read as many samples from the FIFO buffer as required to fill the frame.
+     * The samples are stored in the frame temporarily.
+     */
     if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
         fprintf(stderr, "Could not read data from FIFO\n");
         av_frame_free(&output_frame);
         return AVERROR_EXIT;
     }
 
-    /* Encode one frame worth of audio samples. */
+    /** Encode one frame worth of audio samples. */
     if (encode_audio_frame(output_frame, output_format_context,
                            output_codec_context, &data_written)) {
         av_frame_free(&output_frame);
@@ -760,22 +665,19 @@ static int load_encode_and_write(AVAudioFifo *fifo,
     return 0;
 }
 
-/**
- * Write the trailer of the output file container.
- * @param output_format_context Format context of the output file
- * @return Error code (0 if successful)
- */
+/** Write the trailer of the output file container. */
 static int write_output_file_trailer(AVFormatContext *output_format_context)
 {
     int error;
     if ((error = av_write_trailer(output_format_context)) < 0) {
         fprintf(stderr, "Could not write output file trailer (error '%s')\n",
-                av_err2str(error));
+                get_error_text(error));
         return error;
     }
     return 0;
 }
 
+/** Convert an audio file to an AAC file in an MP4 container. */
 int main(int argc, char **argv)
 {
     AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
@@ -784,75 +686,90 @@ int main(int argc, char **argv)
     AVAudioFifo *fifo = NULL;
     int ret = AVERROR_EXIT;
 
-    if (argc != 3) {
+    if (argc < 3) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
         exit(1);
     }
 
-    /* Open the input file for reading. */
+    /** Register all codecs and formats so that they can be used. */
+    av_register_all();
+    /** Open the input file for reading. */
     if (open_input_file(argv[1], &input_format_context,
                         &input_codec_context))
         goto cleanup;
-    /* Open the output file for writing. */
+    /** Open the output file for writing. */
     if (open_output_file(argv[2], input_codec_context,
                          &output_format_context, &output_codec_context))
         goto cleanup;
-    /* Initialize the resampler to be able to convert audio sample formats. */
+    /** Initialize the resampler to be able to convert audio sample formats. */
     if (init_resampler(input_codec_context, output_codec_context,
                        &resample_context))
         goto cleanup;
-    /* Initialize the FIFO buffer to store audio samples to be encoded. */
+    /** Initialize the FIFO buffer to store audio samples to be encoded. */
     if (init_fifo(&fifo, output_codec_context))
         goto cleanup;
-    /* Write the header of the output file container. */
+    /** Write the header of the output file container. */
     if (write_output_file_header(output_format_context))
         goto cleanup;
 
-    /* Loop as long as we have input samples to read or output samples
-     * to write; abort as soon as we have neither. */
+    /**
+     * Loop as long as we have input samples to read or output samples
+     * to write; abort as soon as we have neither.
+     */
     while (1) {
-        /* Use the encoder's desired frame size for processing. */
+        /** Use the encoder's desired frame size for processing. */
         const int output_frame_size = output_codec_context->frame_size;
         int finished                = 0;
 
-        /* Make sure that there is one frame worth of samples in the FIFO
+        /**
+         * Make sure that there is one frame worth of samples in the FIFO
          * buffer so that the encoder can do its work.
          * Since the decoder's and the encoder's frame size may differ, we
          * need to FIFO buffer to store as many frames worth of input samples
-         * that they make up at least one frame worth of output samples. */
+         * that they make up at least one frame worth of output samples.
+         */
         while (av_audio_fifo_size(fifo) < output_frame_size) {
-            /* Decode one frame worth of audio samples, convert it to the
-             * output sample format and put it into the FIFO buffer. */
+            /**
+             * Decode one frame worth of audio samples, convert it to the
+             * output sample format and put it into the FIFO buffer.
+             */
             if (read_decode_convert_and_store(fifo, input_format_context,
                                               input_codec_context,
                                               output_codec_context,
                                               resample_context, &finished))
                 goto cleanup;
 
-            /* If we are at the end of the input file, we continue
-             * encoding the remaining audio samples to the output file. */
+            /**
+             * If we are at the end of the input file, we continue
+             * encoding the remaining audio samples to the output file.
+             */
             if (finished)
                 break;
         }
 
-        /* If we have enough samples for the encoder, we encode them.
+        /**
+         * If we have enough samples for the encoder, we encode them.
          * At the end of the file, we pass the remaining samples to
-         * the encoder. */
+         * the encoder.
+         */
         while (av_audio_fifo_size(fifo) >= output_frame_size ||
                (finished && av_audio_fifo_size(fifo) > 0))
-            /* Take one frame worth of audio samples from the FIFO buffer,
-             * encode it and write it to the output file. */
+            /**
+             * Take one frame worth of audio samples from the FIFO buffer,
+             * encode it and write it to the output file.
+             */
             if (load_encode_and_write(fifo, output_format_context,
                                       output_codec_context))
                 goto cleanup;
 
-        /* If we are at the end of the input file and have encoded
-         * all remaining samples, we can exit this loop and finish. */
+        /**
+         * If we are at the end of the input file and have encoded
+         * all remaining samples, we can exit this loop and finish.
+         */
         if (finished) {
             int data_written;
-            /* Flush the encoder as it may have delayed frames. */
+            /** Flush the encoder as it may have delayed frames. */
             do {
-                data_written = 0;
                 if (encode_audio_frame(NULL, output_format_context,
                                        output_codec_context, &data_written))
                     goto cleanup;
@@ -861,7 +778,7 @@ int main(int argc, char **argv)
         }
     }
 
-    /* Write the trailer of the output file container. */
+    /** Write the trailer of the output file container. */
     if (write_output_file_trailer(output_format_context))
         goto cleanup;
     ret = 0;

doc/examples/transcoding.c

@@ -30,6 +30,7 @@
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
@@ -44,12 +45,6 @@ typedef struct FilteringContext {
 } FilteringContext;
 static FilteringContext *filter_ctx;
 
-typedef struct StreamContext {
-    AVCodecContext *dec_ctx;
-    AVCodecContext *enc_ctx;
-} StreamContext;
-static StreamContext *stream_ctx;
-
 static int open_input_file(const char *filename)
 {
     int ret;
@@ -66,42 +61,22 @@ static int open_input_file(const char *filename)
         return ret;
     }
 
-    stream_ctx = av_mallocz_array(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
-    if (!stream_ctx)
-        return AVERROR(ENOMEM);
-
     for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *stream = ifmt_ctx->streams[i];
-        AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
+        AVStream *stream;
         AVCodecContext *codec_ctx;
-        if (!dec) {
-            av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
-            return AVERROR_DECODER_NOT_FOUND;
-        }
-        codec_ctx = avcodec_alloc_context3(dec);
-        if (!codec_ctx) {
-            av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
-            return AVERROR(ENOMEM);
-        }
-        ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
-                   "for stream #%u\n", i);
-            return ret;
-        }
+        stream = ifmt_ctx->streams[i];
+        codec_ctx = stream->codec;
         /* Reencode video & audio and remux subtitles etc. */
         if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
                 || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
-                codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
             /* Open decoder */
-            ret = avcodec_open2(codec_ctx, dec, NULL);
+            ret = avcodec_open2(codec_ctx,
+                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
                 return ret;
             }
         }
-        stream_ctx[i].dec_ctx = codec_ctx;
     }
 
     av_dump_format(ifmt_ctx, 0, filename, 0);
@@ -133,7 +108,8 @@ static int open_output_file(const char *filename)
         }
 
         in_stream = ifmt_ctx->streams[i];
-        dec_ctx = stream_ctx[i].dec_ctx;
+        dec_ctx = in_stream->codec;
+        enc_ctx = out_stream->codec;
 
         if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
                 || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
@@ -143,11 +119,6 @@ static int open_output_file(const char *filename)
                 av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
                 return AVERROR_INVALIDDATA;
             }
-            enc_ctx = avcodec_alloc_context3(encoder);
-            if (!enc_ctx) {
-                av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
-                return AVERROR(ENOMEM);
-            }
 
             /* In this example, we transcode to same properties (picture size,
              * sample rate etc.). These properties can be changed for output
@@ -162,7 +133,7 @@ static int open_output_file(const char *filename)
                 else
                     enc_ctx->pix_fmt = dec_ctx->pix_fmt;
                 /* video time_base can be set to whatever is handy and supported by encoder */
-                enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
+                enc_ctx->time_base = dec_ctx->time_base;
             } else {
                 enc_ctx->sample_rate = dec_ctx->sample_rate;
                 enc_ctx->channel_layout = dec_ctx->channel_layout;
@@ -178,29 +149,22 @@ static int open_output_file(const char *filename)
                 av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
                 return ret;
             }
-            ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
-                return ret;
-            }
-            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-                enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
-
-            out_stream->time_base = enc_ctx->time_base;
-            stream_ctx[i].enc_ctx = enc_ctx;
         } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
             av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
             return AVERROR_INVALIDDATA;
         } else {
             /* if this stream must be remuxed */
-            ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
+            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
+                    ifmt_ctx->streams[i]->codec);
             if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
+                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
                 return ret;
             }
-            out_stream->time_base = in_stream->time_base;
         }
 
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
     }
     av_dump_format(ofmt_ctx, 0, filename, 1);
 
@@ -227,8 +191,8 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
 {
     char args[512];
     int ret = 0;
-    const AVFilter *buffersrc = NULL;
-    const AVFilter *buffersink = NULL;
+    AVFilter *buffersrc = NULL;
+    AVFilter *buffersink = NULL;
     AVFilterContext *buffersrc_ctx = NULL;
     AVFilterContext *buffersink_ctx = NULL;
     AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -384,17 +348,17 @@ static int init_filters(void)
         filter_ctx[i].buffersrc_ctx  = NULL;
         filter_ctx[i].buffersink_ctx = NULL;
         filter_ctx[i].filter_graph   = NULL;
-        if (!(ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
-                || ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
+        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
+                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
             continue;
 
 
-        if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
+        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
             filter_spec = "null"; /* passthrough (dummy) filter for video */
         else
             filter_spec = "anull"; /* passthrough (dummy) filter for audio */
-        ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
-                stream_ctx[i].enc_ctx, filter_spec);
+        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
+                ofmt_ctx->streams[i]->codec, filter_spec);
         if (ret)
             return ret;
     }
@@ -406,7 +370,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
     int got_frame_local;
     AVPacket enc_pkt;
     int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
-        (ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
+        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
          AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
 
     if (!got_frame)
@@ -417,7 +381,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
     enc_pkt.data = NULL;
     enc_pkt.size = 0;
     av_init_packet(&enc_pkt);
-    ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
+    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
             filt_frame, got_frame);
     av_frame_free(&filt_frame);
     if (ret < 0)
@@ -428,7 +392,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
     /* prepare packet for muxing */
     enc_pkt.stream_index = stream_index;
     av_packet_rescale_ts(&enc_pkt,
-                         stream_ctx[stream_index].enc_ctx->time_base,
+                         ofmt_ctx->streams[stream_index]->codec->time_base,
                          ofmt_ctx->streams[stream_index]->time_base);
 
     av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
@@ -486,7 +450,7 @@ static int flush_encoder(unsigned int stream_index)
     int ret;
     int got_frame;
 
-    if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
+    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
                 AV_CODEC_CAP_DELAY))
         return 0;
 
@@ -517,6 +481,9 @@ int main(int argc, char **argv)
         return 1;
     }
 
+    av_register_all();
+    avfilter_register_all();
+
     if ((ret = open_input_file(argv[1])) < 0)
         goto end;
     if ((ret = open_output_file(argv[2])) < 0)
@@ -529,7 +496,7 @@ int main(int argc, char **argv)
         if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
             break;
         stream_index = packet.stream_index;
-        type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
+        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
         av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
                 stream_index);
 
@@ -542,10 +509,10 @@ int main(int argc, char **argv)
             }
             av_packet_rescale_ts(&packet,
                                  ifmt_ctx->streams[stream_index]->time_base,
-                                 stream_ctx[stream_index].dec_ctx->time_base);
+                                 ifmt_ctx->streams[stream_index]->codec->time_base);
             dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
                 avcodec_decode_audio4;
-            ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
+            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
                     &got_frame, &packet);
             if (ret < 0) {
                 av_frame_free(&frame);
@@ -554,7 +521,7 @@ int main(int argc, char **argv)
             }
 
             if (got_frame) {
-                frame->pts = frame->best_effort_timestamp;
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
                 ret = filter_encode_write_frame(frame, stream_index);
                 av_frame_free(&frame);
                 if (ret < 0)
@@ -599,14 +566,13 @@ end:
     av_packet_unref(&packet);
     av_frame_free(&frame);
     for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        avcodec_free_context(&stream_ctx[i].dec_ctx);
-        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
-            avcodec_free_context(&stream_ctx[i].enc_ctx);
+        avcodec_close(ifmt_ctx->streams[i]->codec);
+        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
+            avcodec_close(ofmt_ctx->streams[i]->codec);
         if (filter_ctx && filter_ctx[i].filter_graph)
             avfilter_graph_free(&filter_ctx[i].filter_graph);
     }
     av_free(filter_ctx);
-    av_free(stream_ctx);
     avformat_close_input(&ifmt_ctx);
     if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
         avio_closep(&ofmt_ctx->pb);

doc/examples/vaapi_encode.c

@@ -1,222 +0,0 @@
-/*
- * Video Acceleration API (video encoding) encode sample
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * Intel VAAPI-accelerated encoding example.
- *
- * @example vaapi_encode.c
- * This example shows how to do VAAPI-accelerated encoding. now only support NV12
- * raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
- *
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <errno.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavutil/pixdesc.h>
-#include <libavutil/hwcontext.h>
-
-static int width, height;
-static AVBufferRef *hw_device_ctx = NULL;
-
-static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
-{
-    AVBufferRef *hw_frames_ref;
-    AVHWFramesContext *frames_ctx = NULL;
-    int err = 0;
-
-    if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
-        fprintf(stderr, "Failed to create VAAPI frame context.\n");
-        return -1;
-    }
-    frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
-    frames_ctx->format    = AV_PIX_FMT_VAAPI;
-    frames_ctx->sw_format = AV_PIX_FMT_NV12;
-    frames_ctx->width     = width;
-    frames_ctx->height    = height;
-    frames_ctx->initial_pool_size = 20;
-    if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
-        fprintf(stderr, "Failed to initialize VAAPI frame context."
-                "Error code: %s\n",av_err2str(err));
-        av_buffer_unref(&hw_frames_ref);
-        return err;
-    }
-    ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
-    if (!ctx->hw_frames_ctx)
-        err = AVERROR(ENOMEM);
-
-    av_buffer_unref(&hw_frames_ref);
-    return err;
-}
-
-static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
-{
-    int ret = 0;
-    AVPacket enc_pkt;
-
-    av_init_packet(&enc_pkt);
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-
-    if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
-        fprintf(stderr, "Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-    while (1) {
-        ret = avcodec_receive_packet(avctx, &enc_pkt);
-        if (ret)
-            break;
-
-        enc_pkt.stream_index = 0;
-        ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
-        av_packet_unref(&enc_pkt);
-    }
-
-end:
-    ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
-    return ret;
-}
-
-int main(int argc, char *argv[])
-{
-    int size, err;
-    FILE *fin = NULL, *fout = NULL;
-    AVFrame *sw_frame = NULL, *hw_frame = NULL;
-    AVCodecContext *avctx = NULL;
-    AVCodec *codec = NULL;
-    const char *enc_name = "h264_vaapi";
-
-    if (argc < 5) {
-        fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
-        return -1;
-    }
-
-    width  = atoi(argv[1]);
-    height = atoi(argv[2]);
-    size   = width * height;
-
-    if (!(fin = fopen(argv[3], "r"))) {
-        fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
-        return -1;
-    }
-    if (!(fout = fopen(argv[4], "w+b"))) {
-        fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
-        err = -1;
-        goto close;
-    }
-
-    err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
-                                 NULL, NULL, 0);
-    if (err < 0) {
-        fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
-        goto close;
-    }
-
-    if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
-        fprintf(stderr, "Could not find encoder.\n");
-        err = -1;
-        goto close;
-    }
-
-    if (!(avctx = avcodec_alloc_context3(codec))) {
-        err = AVERROR(ENOMEM);
-        goto close;
-    }
-
-    avctx->width     = width;
-    avctx->height    = height;
-    avctx->time_base = (AVRational){1, 25};
-    avctx->framerate = (AVRational){25, 1};
-    avctx->sample_aspect_ratio = (AVRational){1, 1};
-    avctx->pix_fmt   = AV_PIX_FMT_VAAPI;
-
-    /* set hw_frames_ctx for encoder's AVCodecContext */
-    if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
-        fprintf(stderr, "Failed to set hwframe context.\n");
-        goto close;
-    }
-
-    if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
-        fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
-        goto close;
-    }
-
-    while (1) {
-        if (!(sw_frame = av_frame_alloc())) {
-            err = AVERROR(ENOMEM);
-            goto close;
-        }
-        /* read data into software frame, and transfer them into hw frame */
-        sw_frame->width  = width;
-        sw_frame->height = height;
-        sw_frame->format = AV_PIX_FMT_NV12;
-        if ((err = av_frame_get_buffer(sw_frame, 32)) < 0)
-            goto close;
-        if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
-            break;
-        if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
-            break;
-
-        if (!(hw_frame = av_frame_alloc())) {
-            err = AVERROR(ENOMEM);
-            goto close;
-        }
-        if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
-            fprintf(stderr, "Error code: %s.\n", av_err2str(err));
-            goto close;
-        }
-        if (!hw_frame->hw_frames_ctx) {
-            err = AVERROR(ENOMEM);
-            goto close;
-        }
-        if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
-            fprintf(stderr, "Error while transferring frame data to surface."
-                    "Error code: %s.\n", av_err2str(err));
-            goto close;
-        }
-
-        if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
-            fprintf(stderr, "Failed to encode.\n");
-            goto close;
-        }
-        av_frame_free(&hw_frame);
-        av_frame_free(&sw_frame);
-    }
-
-    /* flush encoder */
-    err = encode_write(avctx, NULL, fout);
-    if (err == AVERROR_EOF)
-        err = 0;
-
-close:
-    if (fin)
-        fclose(fin);
-    if (fout)
-        fclose(fout);
-    av_frame_free(&sw_frame);
-    av_frame_free(&hw_frame);
-    avcodec_free_context(&avctx);
-    av_buffer_unref(&hw_device_ctx);
-
-    return err;
-}

doc/examples/vaapi_transcode.c

@@ -1,304 +0,0 @@
-/*
- * Video Acceleration API (video transcoding) transcode sample
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * Intel VAAPI-accelerated transcoding example.
- *
- * @example vaapi_transcode.c
- * This example shows how to do VAAPI-accelerated transcoding.
- * Usage: vaapi_transcode input_stream codec output_stream
- * e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
- *      - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
- */
-
-#include <stdio.h>
-#include <errno.h>
-
-#include <libavutil/hwcontext.h>
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-static AVBufferRef *hw_device_ctx = NULL;
-static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
-static int video_stream = -1;
-static AVStream *ost;
-static int initialized = 0;
-
-static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
-                                           const enum AVPixelFormat *pix_fmts)
-{
-    const enum AVPixelFormat *p;
-
-    for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
-        if (*p == AV_PIX_FMT_VAAPI)
-            return *p;
-    }
-
-    fprintf(stderr, "Unable to decode this file using VA-API.\n");
-    return AV_PIX_FMT_NONE;
-}
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *decoder = NULL;
-    AVStream *video = NULL;
-
-    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
-        fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
-                filename, av_err2str(ret));
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
-        fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
-                av_err2str(ret));
-        return ret;
-    }
-
-    ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot find a video stream in the input file. "
-                "Error code: %s\n", av_err2str(ret));
-        return ret;
-    }
-    video_stream = ret;
-
-    if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
-        return AVERROR(ENOMEM);
-
-    video = ifmt_ctx->streams[video_stream];
-    if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
-        fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
-                av_err2str(ret));
-        return ret;
-    }
-
-    decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
-    if (!decoder_ctx->hw_device_ctx) {
-        fprintf(stderr, "A hardware device reference create failed.\n");
-        return AVERROR(ENOMEM);
-    }
-    decoder_ctx->get_format    = get_vaapi_format;
-
-    if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
-        fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
-                av_err2str(ret));
-
-    return ret;
-}
-
-static int encode_write(AVFrame *frame)
-{
-    int ret = 0;
-    AVPacket enc_pkt;
-
-    av_init_packet(&enc_pkt);
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-
-    if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
-        fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-    while (1) {
-        ret = avcodec_receive_packet(encoder_ctx, &enc_pkt);
-        if (ret)
-            break;
-
-        enc_pkt.stream_index = 0;
-        av_packet_rescale_ts(&enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
-                             ofmt_ctx->streams[0]->time_base);
-        ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error during writing data to output file. "
-                    "Error code: %s\n", av_err2str(ret));
-            return -1;
-        }
-    }
-
-end:
-    if (ret == AVERROR_EOF)
-        return 0;
-    ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
-    return ret;
-}
-
-static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
-{
-    AVFrame *frame;
-    int ret = 0;
-
-    ret = avcodec_send_packet(decoder_ctx, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
-        return ret;
-    }
-
-    while (ret >= 0) {
-        if (!(frame = av_frame_alloc()))
-            return AVERROR(ENOMEM);
-
-        ret = avcodec_receive_frame(decoder_ctx, frame);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
-            av_frame_free(&frame);
-            return 0;
-        } else if (ret < 0) {
-            fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
-            goto fail;
-        }
-
-        if (!initialized) {
-            /* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
-               Only after we get a decoded frame, can we obtain its hw_frames_ctx */
-            encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
-            if (!encoder_ctx->hw_frames_ctx) {
-                ret = AVERROR(ENOMEM);
-                goto fail;
-            }
-            /* set AVCodecContext Parameters for encoder, here we keep them stay
-             * the same as decoder.
-             * xxx: now the the sample can't handle resolution change case.
-             */
-            encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
-            encoder_ctx->pix_fmt   = AV_PIX_FMT_VAAPI;
-            encoder_ctx->width     = decoder_ctx->width;
-            encoder_ctx->height    = decoder_ctx->height;
-
-            if ((ret = avcodec_open2(encoder_ctx, enc_codec, NULL)) < 0) {
-                fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
-                        av_err2str(ret));
-                goto fail;
-            }
-
-            if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
-                fprintf(stderr, "Failed to allocate stream for output format.\n");
-                ret = AVERROR(ENOMEM);
-                goto fail;
-            }
-
-            ost->time_base = encoder_ctx->time_base;
-            ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
-            if (ret < 0) {
-                fprintf(stderr, "Failed to copy the stream parameters. "
-                        "Error code: %s\n", av_err2str(ret));
-                goto fail;
-            }
-
-            /* write the stream header */
-            if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
-                fprintf(stderr, "Error while writing stream header. "
-                        "Error code: %s\n", av_err2str(ret));
-                goto fail;
-            }
-
-            initialized = 1;
-        }
-
-        if ((ret = encode_write(frame)) < 0)
-            fprintf(stderr, "Error during encoding and writing.\n");
-
-fail:
-        av_frame_free(&frame);
-        if (ret < 0)
-            return ret;
-    }
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    int ret = 0;
-    AVPacket dec_pkt;
-    AVCodec *enc_codec;
-
-    if (argc != 4) {
-        fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
-                "The output format is guessed according to the file extension.\n"
-                "\n", argv[0]);
-        return -1;
-    }
-
-    ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI, NULL, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(ret));
-        return -1;
-    }
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-
-    if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
-        fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
-        ret = -1;
-        goto end;
-    }
-
-    if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
-        fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
-                "%s\n", av_err2str(ret));
-        goto end;
-    }
-
-    if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot open output file. "
-                "Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    /* read all packets and only transcoding video */
-    while (ret >= 0) {
-        if ((ret = av_read_frame(ifmt_ctx, &dec_pkt)) < 0)
-            break;
-
-        if (video_stream == dec_pkt.stream_index)
-            ret = dec_enc(&dec_pkt, enc_codec);
-
-        av_packet_unref(&dec_pkt);
-    }
-
-    /* flush decoder */
-    dec_pkt.data = NULL;
-    dec_pkt.size = 0;
-    ret = dec_enc(&dec_pkt, enc_codec);
-    av_packet_unref(&dec_pkt);
-
-    /* flush encoder */
-    ret = encode_write(NULL);
-
-    /* write the trailer for output stream */
-    av_write_trailer(ofmt_ctx);
-
-end:
-    avformat_close_input(&ifmt_ctx);
-    avformat_close_input(&ofmt_ctx);
-    avcodec_free_context(&decoder_ctx);
-    avcodec_free_context(&encoder_ctx);
-    av_buffer_unref(&hw_device_ctx);
-    return ret;
-}

doc/faq.texi

@@ -311,18 +311,18 @@ invoking ffmpeg with several @option{-i} options.
 For audio, to put all channels together in a single stream (example: two
 mono streams into one stereo stream): this is sometimes called to
 @emph{merge} them, and can be done using the
-@url{ffmpeg-filters.html#amerge, @code{amerge}} filter.
+@url{https://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
 
 @item
 For audio, to play one on top of the other: this is called to @emph{mix}
 them, and can be done by first merging them into a single stream and then
-using the @url{ffmpeg-filters.html#pan, @code{pan}} filter to mix
+using the @url{https://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
 the channels at will.
 
 @item
 For video, to display both together, side by side or one on top of a part of
 the other; it can be done using the
-@url{ffmpeg-filters.html#overlay, @code{overlay}} video filter.
+@url{https://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
 
 @end itemize
 
@@ -333,19 +333,19 @@ There are several solutions, depending on the exact circumstances.
 
 @subsection Concatenating using the concat @emph{filter}
 
-FFmpeg has a @url{ffmpeg-filters.html#concat,
+FFmpeg has a @url{https://ffmpeg.org/ffmpeg-filters.html#concat,
 @code{concat}} filter designed specifically for that, with examples in the
 documentation. This operation is recommended if you need to re-encode.
 
 @subsection Concatenating using the concat @emph{demuxer}
 
-FFmpeg has a @url{ffmpeg-formats.html#concat,
+FFmpeg has a @url{https://www.ffmpeg.org/ffmpeg-formats.html#concat,
 @code{concat}} demuxer which you can use when you want to avoid a re-encode and
 your format doesn't support file level concatenation.
 
 @subsection Concatenating using the concat @emph{protocol} (file level)
 
-FFmpeg has a @url{ffmpeg-protocols.html#concat,
+FFmpeg has a @url{https://ffmpeg.org/ffmpeg-protocols.html#concat,
 @code{concat}} protocol designed specifically for that, with examples in the
 documentation.
 
@@ -385,7 +385,7 @@ mkfifo intermediate2.mpg
 ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
 ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
 cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -c:v mpeg4 -c:a libmp3lame output.avi
+ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
 @end example
 
 @subsection Concatenating using raw audio and video
@@ -407,13 +407,13 @@ mkfifo temp2.a
 mkfifo temp2.v
 mkfifo all.a
 mkfifo all.v
-ffmpeg -i input1.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
-ffmpeg -i input2.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
+ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
+ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
 ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
 @{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
 cat temp1.a temp2.a > all.a &
 cat temp1.v temp2.v > all.v &
-ffmpeg -f u16le -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
+ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
        -f yuv4mpegpipe -i all.v \
        -y output.flv
 rm temp[12].[av] all.[av]
@@ -485,7 +485,7 @@ scaling adjusts the SAR to keep the DAR constant.
 
 If you want to stretch, or “unstretch”, the image, you need to override the
 information with the
-@url{ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
+@url{https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
 
 Do not forget to examine carefully the original video to check whether the
 stretching comes from the image or from the aspect ratio information.
@@ -501,71 +501,6 @@ ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
 ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
 @end example
 
-@anchor{background task}
-@section How do I run ffmpeg as a background task?
-
-ffmpeg normally checks the console input, for entries like "q" to stop
-and "?" to give help, while performing operations. ffmpeg does not have a way of
-detecting when it is running as a background task.
-When it checks the console input, that can cause the process running ffmpeg
-in the background to suspend.
-
-To prevent those input checks, allowing ffmpeg to run as a background task,
-use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
-in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
-or invoke ffmpeg in its own process via an operating system API.
-
-As an alternative, when you are running ffmpeg in a shell, you can redirect
-standard input to @code{/dev/null} (on Linux and Mac OS)
-or @code{NUL} (on Windows). You can do this redirect either
-on the ffmpeg invocation, or from a shell script which calls ffmpeg.
-
-For example:
-
-@example
-ffmpeg -nostdin -i INPUT OUTPUT
-@end example
-
-or (on Linux, Mac OS, and other UNIX-like shells):
-
-@example
-ffmpeg -i INPUT OUTPUT </dev/null
-@end example
-
-or (on Windows):
-
-@example
-ffmpeg -i INPUT OUTPUT <NUL
-@end example
-
-@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
-
-If you run ffmpeg in the background, you may find that its process suspends.
-There may be a message like @emph{suspended (tty output)}. The question is how
-to prevent the process from being suspended.
-
-For example:
-
-@example
-% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
-[1] 93352
-%
-[1]  + suspended (tty output)  ffmpeg -i INPUT OUTPUT &>
-@end example
-
-The message "tty output" notwithstanding, the problem here is that
-ffmpeg normally checks the console input when it runs. The operating system
-detects this, and suspends the process until you can bring it to the
-foreground and attend to it.
-
-The solution is to use the right techniques to tell ffmpeg not to consult
-console input. You can use the
-@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
-or redirect standard input with @code{< /dev/null}.
-See FAQ
-@ref{background task, @emph{How do I run ffmpeg as a background task?}}
-for details.
-
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?

doc/fate.texi

@@ -147,26 +147,6 @@ process.
 The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.
 
-@chapter Uploading new samples to the fate suite
-
-This is for developers who have an account on the fate suite server.
-If you upload new samples, please make sure they are as small as possible,
-space on each client, network bandwidth and so on benefit from smaller test cases.
-Also keep in mind older checkouts use existing sample files, that means in
-practice generally do not replace, remove or overwrite files as it likely would
-break older checkouts or releases.
-
-@example
-#First update your local samples copy:
-rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
-
-#Then do a dry run checking what would be uploaded:
-rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
-
-#Upload the files:
-rsync -vaL  --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
-@end example
-
 
 @chapter FATE makefile targets and variables
 
@@ -217,11 +197,6 @@ through @command{ssh}.
 
 @item GEN
 Set to @samp{1} to generate the missing or mismatched references.
-
-@item HWACCEL
-Specify which hardware acceleration to use while running regression tests,
-by default @samp{none} is used.
-
 @end table
 
 @section Examples

doc/fate_config.sh.template

@@ -6,7 +6,6 @@ workdir=                                 # directory in which to do all the work
 #fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
 comment=                                 # optional description
 build_only=     # set to "yes" for a compile-only instance that skips tests
-ignore_tests=
 
 # the following are optional and map to configure options
 arch=
@@ -27,7 +26,5 @@ extra_conf=     # extra configure options not covered above
 
 #make=          # name of GNU make if not 'make'
 makeopts=       # extra options passed to 'make'
-#makeopts_fate= # extra options passed to 'make' when running tests,
-                # defaulting to makeopts above if this is not set
 #tar=           # command to create a tar archive from its arguments on stdout,
                 # defaults to 'tar c'

doc/ffmpeg-bitstream-filters.texi

@@ -26,12 +26,12 @@ bitstream level modifications without performing decoding.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavcodec.html,libavcodec}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-codecs.texi

@@ -23,12 +23,12 @@ the libavcodec library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavcodec.html,libavcodec}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-devices.texi

@@ -23,12 +23,12 @@ libavdevice library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavdevice.html,libavdevice}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-filters.texi

@@ -23,12 +23,12 @@ libavfilter library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavfilter.html,libavfilter}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavfilter(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-formats.texi

@@ -23,12 +23,12 @@ provided by the libavformat library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavformat.html,libavformat}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-protocols.texi

@@ -23,12 +23,12 @@ libavformat library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavformat.html,libavformat}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-resampler.texi

@@ -25,12 +25,12 @@ and convert audio format and packing layout.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libswresample.html,libswresample}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-scaler.texi

@@ -24,12 +24,12 @@ image rescaling and pixel format conversion.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libswscale.html,libswscale}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libswscale(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg-utils.texi

@@ -23,12 +23,12 @@ by the libavutil library.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{libavutil.html,libavutil}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1), libavutil(3)
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
 @end ifnothtml
 
 @include authors.texi

doc/ffmpeg.texi

@@ -223,7 +223,7 @@ with the highest resolution, for audio, it is the stream with the most channels,
 subtitles, it is the first subtitle stream. In the case where several streams of
 the same type rate equally, the stream with the lowest index is chosen.
 
-You can disable some of those defaults by using the @code{-vn/-an/-sn/-dn} options. For
+You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For
 full manual control, use the @code{-map} option, which disables the defaults just
 described.
 
@@ -289,8 +289,8 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
 
 -to and -t are mutually exclusive and -t has priority.
 
-@item -to @var{position} (@emph{input/output})
-Stop writing the output or reading the input at @var{position}.
+@item -to @var{position} (@emph{output})
+Stop writing the output at @var{position}.
 @var{position} must be a time duration specification,
 see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
@@ -357,40 +357,6 @@ To set the language of the first audio stream:
 ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
 @end example
 
-@item -disposition[:stream_specifier] @var{value} (@emph{output,per-stream})
-Sets the disposition for a stream.
-
-This option overrides the disposition copied from the input stream. It is also
-possible to delete the disposition by setting it to 0.
-
-The following dispositions are recognized:
-@table @option
-@item default
-@item dub
-@item original
-@item comment
-@item lyrics
-@item karaoke
-@item forced
-@item hearing_impaired
-@item visual_impaired
-@item clean_effects
-@item captions
-@item descriptions
-@item metadata
-@end table
-
-For example, to make the second audio stream the default stream:
-@example
-ffmpeg -i in.mkv -disposition:a:1 default out.mkv
-@end example
-
-To make the second subtitle stream the default stream and remove the default
-disposition from the first subtitle stream:
-@example
-ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT
-@end example
-
 @item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
 
 Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@@ -413,13 +379,8 @@ they do not conflict with the standard, as in:
 ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
 @end example
 
-@item -dn (@emph{output})
-Disable data recording. For full manual control see the @code{-map}
-option.
-
 @item -dframes @var{number} (@emph{output})
-Set the number of data frames to output. This is an obsolete alias for
-@code{-frames:d}, which you should use instead.
+Set the number of data frames to output. This is an alias for @code{-frames:d}.
 
 @item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
 Stop writing to the stream after @var{framecount} frames.
@@ -454,11 +415,6 @@ This option is similar to @option{-filter}, the only difference is that its
 argument is the name of the file from which a filtergraph description is to be
 read.
 
-@item -filter_threads @var{nb_threads} (@emph{global})
-Defines how many threads are used to process a filter pipeline. Each pipeline
-will produce a thread pool with this many threads available for parallel processing.
-The default is the number of available CPUs.
-
 @item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
 Specify the preset for matching stream(s).
 
@@ -474,7 +430,6 @@ the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
 consists of only alphanumeric characters. The last key of a sequence of
 progress information is always "progress".
 
-@anchor{stdin option}
 @item -stdin
 Enable interaction on standard input. On by default unless standard input is
 used as an input. To explicitly disable interaction you need to specify
@@ -535,8 +490,7 @@ Disable automatically rotating video based on file metadata.
 
 @table @option
 @item -vframes @var{number} (@emph{output})
-Set the number of video frames to output. This is an obsolete alias for
-@code{-frames:v}, which you should use instead.
+Set the number of video frames to output. 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).
 
@@ -575,8 +529,7 @@ stored at container level, but not the aspect ratio stored in encoded
 frames, if it exists.
 
 @item -vn (@emph{output})
-Disable video recording. For full manual control see the @code{-map}
-option.
+Disable video recording.
 
 @item -vcodec @var{codec} (@emph{output})
 Set the video codec. This is an alias for @code{-codec:v}.
@@ -644,16 +597,6 @@ Calculate PSNR of compressed frames.
 Dump video coding statistics to @file{vstats_HHMMSS.log}.
 @item -vstats_file @var{file}
 Dump video coding statistics to @var{file}.
-@item -vstats_version @var{file}
-Specifies which version of the vstats format to use. Default is 2.
-
-version = 1 :
-
-@code{frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
-
-version > 1:
-
-@code{out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
 @item -top[:@var{stream_specifier}] @var{n} (@emph{output,per-stream})
 top=1/bottom=0/auto=-1 field first
 @item -dc @var{precision}
@@ -721,104 +664,6 @@ would be more efficient.
 When doing stream copy, copy also non-key frames found at the
 beginning.
 
-@item -init_hw_device @var{type}[=@var{name}][:@var{device}[,@var{key=value}...]]
-Initialise a new hardware device of type @var{type} called @var{name}, using the
-given device parameters.
-If no name is specified it will receive a default name of the form "@var{type}%d".
-
-The meaning of @var{device} and the following arguments depends on the
-device type:
-@table @option
-
-@item cuda
-@var{device} is the number of the CUDA device.
-
-@item dxva2
-@var{device} is the number of the Direct3D 9 display adapter.
-
-@item vaapi
-@var{device} is either an X11 display name or a DRM render node.
-If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY})
-and then the first DRM render node (@emph{/dev/dri/renderD128}).
-
-@item vdpau
-@var{device} is an X11 display name.
-If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY}).
-
-@item qsv
-@var{device} selects a value in @samp{MFX_IMPL_*}. Allowed values are:
-@table @option
-@item auto
-@item sw
-@item hw
-@item auto_any
-@item hw_any
-@item hw2
-@item hw3
-@item hw4
-@end table
-If not specified, @samp{auto_any} is used.
-(Note that it may be easier to achieve the desired result for QSV by creating the
-platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
-QSV device from that.)
-
-@item opencl
-@var{device} selects the platform and device as @emph{platform_index.device_index}.
-
-The set of devices can also be filtered using the key-value pairs to find only
-devices matching particular platform or device strings.
-
-The strings usable as filters are:
-@table @option
-@item platform_profile
-@item platform_version
-@item platform_name
-@item platform_vendor
-@item platform_extensions
-@item device_name
-@item device_vendor
-@item driver_version
-@item device_version
-@item device_profile
-@item device_extensions
-@item device_type
-@end table
-
-The indices and filters must together uniquely select a device.
-
-Examples:
-@table @emph
-@item -init_hw_device opencl:0.1
-Choose the second device on the first platform.
-
-@item -init_hw_device opencl:,device_name=Foo9000
-Choose the device with a name containing the string @emph{Foo9000}.
-
-@item -init_hw_device opencl:1,device_type=gpu,device_extensions=cl_khr_fp16
-Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
-extension.
-@end table
-
-@end table
-
-@item -init_hw_device @var{type}[=@var{name}]@@@var{source}
-Initialise a new hardware device of type @var{type} called @var{name},
-deriving it from the existing device with the name @var{source}.
-
-@item -init_hw_device list
-List all hardware device types supported in this build of ffmpeg.
-
-@item -filter_hw_device @var{name}
-Pass the hardware device called @var{name} to all filters in any filter graph.
-This can be used to set the device to upload to with the @code{hwupload} filter,
-or the device to map to with the @code{hwmap} filter.  Other filters may also
-make use of this parameter when they require a hardware device.  Note that this
-is typically only required when the input is not already in hardware frames -
-when it is, filters will derive the device they require from the context of the
-frames they receive as input.
-
-This is a global setting, so all filters will receive the same device.
-
 @item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
 Use hardware acceleration to decode the matching stream(s). The allowed values
 of @var{hwaccel} are:
@@ -829,15 +674,15 @@ Do not use any hardware acceleration (the default).
 @item auto
 Automatically select the hardware acceleration method.
 
+@item vda
+Use Apple VDA hardware acceleration.
+
 @item vdpau
 Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
 
 @item dxva2
 Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
 
-@item vaapi
-Use VAAPI (Video Acceleration API) hardware acceleration.
-
 @item qsv
 Use the Intel QuickSync Video acceleration for video transcoding.
 
@@ -861,11 +706,33 @@ useful for testing.
 @item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
 Select a device to use for hardware acceleration.
 
-This option only makes sense when the @option{-hwaccel} option is also specified.
-It can either refer to an existing device created with @option{-init_hw_device}
-by name, or it can create a new device as if
-@samp{-init_hw_device} @var{type}:@var{hwaccel_device}
-were called immediately before.
+This option only makes sense when the @option{-hwaccel} option is also
+specified. Its exact meaning depends on the specific hardware acceleration
+method chosen.
+
+@table @option
+@item vdpau
+For VDPAU, this option specifies the X11 display/screen to use. If this option
+is not specified, the value of the @var{DISPLAY} environment variable is used
+
+@item dxva2
+For DXVA2, this option should contain the number of the display adapter to use.
+If this option is not specified, the default adapter is used.
+
+@item qsv
+For QSV, this option corresponds to the values of MFX_IMPL_* . Allowed values
+are:
+@table @option
+@item auto
+@item sw
+@item hw
+@item auto_any
+@item hw_any
+@item hw2
+@item hw3
+@item hw4
+@end table
+@end table
 
 @item -hwaccels
 List all hardware acceleration methods supported in this build of ffmpeg.
@@ -876,8 +743,7 @@ List all hardware acceleration methods supported in this build of ffmpeg.
 
 @table @option
 @item -aframes @var{number} (@emph{output})
-Set the number of audio frames to output. This is an obsolete alias for
-@code{-frames:a}, which you should use instead.
+Set the number of audio frames to output. This is an alias for @code{-frames:a}.
 @item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
 Set the audio sampling frequency. For output streams it is set by
 default to the frequency of the corresponding input stream. For input
@@ -891,8 +757,7 @@ default to the number of input audio channels. For input streams
 this option only makes sense for audio grabbing devices and raw demuxers
 and is mapped to the corresponding demuxer options.
 @item -an (@emph{output})
-Disable audio recording. For full manual control see the @code{-map}
-option.
+Disable audio recording.
 @item -acodec @var{codec} (@emph{input/output})
 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})
@@ -927,8 +792,7 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
 @item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
 @item -sn (@emph{output})
-Disable subtitle recording. For full manual control see the @code{-map}
-option.
+Disable subtitle recording.
 @item -sbsf @var{bitstream_filter}
 Deprecated, see -bsf
 @end table
@@ -958,7 +822,7 @@ Set the size of the canvas used to render subtitles.
 @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}]] | @var{[linklabel]} (@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
@@ -974,11 +838,6 @@ 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.
 
-A trailing @code{?} after the stream index will allow the map to be
-optional: if the map matches no streams the map will be ignored instead
-of failing. Note the map will still fail if an invalid input file index
-is used; such as if the map refers to a non-existent input.
-
 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.
@@ -1016,13 +875,6 @@ To map all the streams except the second audio, use negative mappings
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
 @end example
 
-To map the video and audio streams from the first input, and using the
-trailing @code{?}, ignore the audio mapping if no audio streams exist in
-the first input:
-@example
-ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
-@end example
-
 To pick the English audio stream:
 @example
 ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
@@ -1038,7 +890,7 @@ such streams is attempted.
 Allow input streams with unknown type to be copied instead of failing if copying
 such streams is attempted.
 
-@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][?][:@var{output_file_id}.@var{stream_specifier}]
+@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
 be mapped on all the audio streams.
@@ -1047,10 +899,6 @@ Using "-1" instead of
 @var{input_file_id}.@var{stream_specifier}.@var{channel_id} will map a muted
 channel.
 
-A trailing @code{?} will allow the map_channel to be
-optional: if the map_channel matches no channel the map_channel will be ignored instead
-of failing.
-
 For example, assuming @var{INPUT} is a stereo audio file, you can switch the
 two audio channels with the following command:
 @example
@@ -1098,13 +946,6 @@ video stream), you can use the following command:
 ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv
 @end example
 
-To map the first two audio channels from the first input, and using the
-trailing @code{?}, ignore the audio channel mapping if the first input is
-mono instead of stereo:
-@example
-ffmpeg -i INPUT -map_channel 0.0.0 -map_channel 0.0.1? OUTPUT
-@end example
-
 @item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@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.
@@ -1174,6 +1015,10 @@ loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
 of the input(s). It is useful for real-time output (e.g. live streaming).
+@item -loop_input
+Loop over the input stream. Currently it works only for image
+streams. This option is used for automatic FFserver testing.
+This option is deprecated, use -loop 1.
 @item -loop_output @var{number_of_times}
 Repeatedly loop output for formats that support looping such as animated GIF
 (0 will loop the output infinitely).
@@ -1267,32 +1112,6 @@ Try to make the choice automatically, in order to generate a sane output.
 
 Default value is -1.
 
-@item -enc_time_base[:@var{stream_specifier}] @var{timebase} (@emph{output,per-stream})
-Set the encoder timebase. @var{timebase} is a floating point number,
-and can assume one of the following values:
-
-@table @option
-@item 0
-Assign a default value according to the media type.
-
-For video - use 1/framerate, for audio - use 1/samplerate.
-
-@item -1
-Use the input stream timebase when possible.
-
-If an input stream is not available, the default timebase will be used.
-
-@item >0
-Use the provided number as the timebase.
-
-This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000)
-or as a floating point number (e.g. 0.04166, 2.0833e-5)
-@end table
-
-Default value is 0.
-
-@item -bitexact (@emph{input/output})
-Enable bitexact mode for (de)muxer and (de/en)coder
 @item -shortest (@emph{output})
 Finish encoding when the shortest input stream ends.
 @item -dts_delta_threshold
@@ -1382,11 +1201,6 @@ To generate 5 seconds of pure red video using lavfi @code{color} source:
 ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
 @end example
 
-@item -filter_complex_threads @var{nb_threads} (@emph{global})
-Defines how many threads are used to process a filter_complex graph.
-Similar to filter_threads but used for @code{-filter_complex} graphs only.
-The default is the number of available CPUs.
-
 @item -lavfi @var{filtergraph} (@emph{global})
 Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
 outputs. Equivalent to @option{-filter_complex}.
@@ -1415,6 +1229,16 @@ file or device. With low latency / high rate live streams, packets may be
 discarded if they are not read in a timely manner; raising this value can
 avoid it.
 
+@item -override_ffserver (@emph{global})
+Overrides the input specifications from @command{ffserver}. Using this
+option you can map any input stream to @command{ffserver} and control
+many aspects of the encoding from @command{ffmpeg}. Without this
+option @command{ffmpeg} will transmit to @command{ffserver} what is
+requested by @command{ffserver}.
+
+The option is intended for cases where features are needed that cannot be
+specified to @command{ffserver} but can be to @command{ffmpeg}.
+
 @item -sdp_file @var{file} (@emph{global})
 Print sdp information for an output stream to @var{file}.
 This allows dumping sdp information when at least one output isn't an
@@ -1455,15 +1279,6 @@ No packets were passed to the muxer, the output is empty.
 @item -xerror (@emph{global})
 Stop and exit on error
 
-@item -max_muxing_queue_size @var{packets} (@emph{output,per-stream})
-When transcoding audio and/or video streams, ffmpeg will not begin writing into
-the output until it has one packet for each such stream. While waiting for that
-to happen, packets for other streams are buffered. This option sets the size of
-this buffer, in packets, for the matching output stream.
-
-The default value of this option should be high enough for most uses, so only
-touch this option if you are sure that you need it.
-
 @end table
 
 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1669,7 +1484,7 @@ to enable LAME support by passing @code{--enable-libmp3lame} to configure.
 The mapping is particularly useful for DVD transcoding
 to get the desired audio language.
 
-NOTE: To see the supported input formats, use @code{ffmpeg -demuxers}.
+NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
 
 @item
 You can extract images from a video, or create a video from many images:
@@ -1684,8 +1499,8 @@ output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
 etc. Images will be rescaled to fit the new WxH values.
 
 If you want to extract just a limited number of frames, you can use the
-above command in combination with the @code{-frames:v} or @code{-t} option,
-or in combination with -ss to start extracting from a certain point in time.
+above command in combination with the -vframes or -t option, or in
+combination with -ss to start extracting from a certain point in time.
 
 For creating a video from many images:
 @example
@@ -1769,7 +1584,7 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
 @ifset config-not-all
 @url{ffmpeg-all.html,ffmpeg-all},
 @end ifset
-@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-utils.html,ffmpeg-utils},
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -1788,7 +1603,7 @@ ffmpeg(1),
 @ifset config-not-all
 ffmpeg-all(1),
 @end ifset
-ffplay(1), ffprobe(1),
+ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

doc/ffplay.texi

@@ -62,12 +62,6 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
 Seek by bytes.
 @item -nodisp
 Disable graphical display.
-@item -noborder
-Borderless window.
-@item -volume
-Set the startup volume. 0 means silence, 100 means no volume reduction or
-amplification. Negative values are treated as 0, values above 100 are treated
-as 100.
 @item -f @var{fmt}
 Force format.
 @item -window_title @var{title}
@@ -291,7 +285,7 @@ Toggle full screen.
 @ifset config-not-all
 @url{ffplay-all.html,ffmpeg-all},
 @end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-utils.html,ffmpeg-utils},
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -310,7 +304,7 @@ ffplay(1),
 @ifset config-not-all
 ffplay-all(1),
 @end ifset
-ffmpeg(1), ffprobe(1),
+ffmpeg(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

doc/ffprobe.texi

@@ -208,13 +208,6 @@ multimedia stream.
 The information for each single frame is printed within a dedicated
 section with name "FRAME" or "SUBTITLE".
 
-@item -show_log @var{loglevel}
-Show logging information from the decoder about each frame according to
-the value set in @var{loglevel}, (see @code{-loglevel}). This option requires @code{-show_frames}.
-
-The information for each log message is printed within a dedicated
-section with name "LOG".
-
 @item -show_streams
 Show information about each media stream contained in the input
 multimedia stream.
@@ -252,7 +245,7 @@ continue reading from that.
 Each interval is specified by two optional parts, separated by "%".
 
 The first part specifies the interval start position. It is
-interpreted as an absolute position, or as a relative offset from the
+interpreted as an abolute position, or as a relative offset from the
 current position if it is preceded by the "+" character. If this first
 part is not specified, no seeking will be performed when reading this
 interval.
@@ -471,7 +464,7 @@ Perform no escaping.
 @end table
 
 @item print_section, p
-Print the section name at the beginning of each line if the value is
+Print the section name at the begin of each line if the value is
 @code{1}, disable it with value set to @code{0}. Default value is
 @code{1}.
 
@@ -653,7 +646,7 @@ DV, GXF and AVI timecodes are available in format metadata
 @ifset config-not-all
 @url{ffprobe-all.html,ffprobe-all},
 @end ifset
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
 @url{ffmpeg-utils.html,ffmpeg-utils},
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -672,7 +665,7 @@ ffprobe(1),
 @ifset config-not-all
 ffprobe-all(1),
 @end ifset
-ffmpeg(1), ffplay(1),
+ffmpeg(1), ffplay(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

doc/ffprobe.xsd

@@ -83,7 +83,6 @@
     <xsd:complexType name="frameType">
       <xsd:sequence>
             <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-            <xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
             <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
       </xsd:sequence>
 
@@ -120,25 +119,6 @@
       <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="color_range"            type="xsd:string"/>
-      <xsd:attribute name="color_space"            type="xsd:string"/>
-      <xsd:attribute name="color_primaries"        type="xsd:string"/>
-      <xsd:attribute name="color_transfer"         type="xsd:string"/>
-      <xsd:attribute name="chroma_location"        type="xsd:string"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="logsType">
-        <xsd:sequence>
-            <xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-    <xsd:complexType name="logType">
-        <xsd:attribute name="context"                     type="xsd:string"/>
-        <xsd:attribute name="level"                       type="xsd:int"   />
-        <xsd:attribute name="category"                    type="xsd:int"   />
-        <xsd:attribute name="parent_context"              type="xsd:string"/>
-        <xsd:attribute name="parent_category"             type="xsd:int"   />
-        <xsd:attribute name="message"                     type="xsd:string"/>
     </xsd:complexType>
 
     <xsd:complexType name="frameSideDataListType">
@@ -149,7 +129,6 @@
     <xsd:complexType name="frameSideDataType">
         <xsd:attribute name="side_data_type"              type="xsd:string"/>
         <xsd:attribute name="side_data_size"              type="xsd:int"   />
-        <xsd:attribute name="timecode"                    type="xsd:string"/>
     </xsd:complexType>
 
     <xsd:complexType name="subtitleType">
@@ -186,7 +165,6 @@
       <xsd:attribute name="visual_impaired"  type="xsd:int" use="required" />
       <xsd:attribute name="clean_effects"    type="xsd:int" use="required" />
       <xsd:attribute name="attached_pic"     type="xsd:int" use="required" />
-      <xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
     </xsd:complexType>
 
     <xsd:complexType name="streamType">
@@ -222,7 +200,6 @@
       <xsd:attribute name="color_transfer"       type="xsd:string"/>
       <xsd:attribute name="color_primaries"      type="xsd:string"/>
       <xsd:attribute name="chroma_location"      type="xsd:string"/>
-      <xsd:attribute name="field_order"          type="xsd:string"/>
       <xsd:attribute name="timecode"             type="xsd:string"/>
       <xsd:attribute name="refs"                 type="xsd:int"/>
 

doc/ffserver.conf

@@ -0,0 +1,372 @@
+# Port on which the server is listening. You must select a different
+# port from your standard HTTP web server if it is running on the same
+# computer.
+HTTPPort 8090
+
+# Address on which the server is bound. Only useful if you have
+# several network interfaces.
+HTTPBindAddress 0.0.0.0
+
+# Number of simultaneous HTTP connections that can be handled. It has
+# to be defined *before* the MaxClients parameter, since it defines the
+# MaxClients maximum limit.
+MaxHTTPConnections 2000
+
+# Number of simultaneous requests that can be handled. Since FFServer
+# is very fast, it is more likely that you will want to leave this high
+# and use MaxBandwidth, below.
+MaxClients 1000
+
+# This the maximum amount of kbit/sec that you are prepared to
+# consume when streaming to clients.
+MaxBandwidth 1000
+
+# Access log file (uses standard Apache log file format)
+# '-' is the standard output.
+CustomLog -
+
+##################################################################
+# Definition of the live feeds. Each live feed contains one video
+# and/or audio sequence coming from an ffmpeg encoder or another
+# ffserver. This sequence may be encoded simultaneously with several
+# codecs at several resolutions.
+
+<Feed feed1.ffm>
+
+# You must use 'ffmpeg' to send a live feed to ffserver. In this
+# example, you can type:
+#
+# ffmpeg http://localhost:8090/feed1.ffm
+
+# ffserver can also do time shifting. It means that it can stream any
+# previously recorded live stream. The request should contain:
+# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
+# a path where the feed is stored on disk. You also specify the
+# maximum size of the feed, where zero means unlimited. Default:
+# File=/tmp/feed_name.ffm FileMaxSize=5M
+File /tmp/feed1.ffm
+FileMaxSize 200K
+
+# You could specify
+# ReadOnlyFile /saved/specialvideo.ffm
+# This marks the file as readonly and it will not be deleted or updated.
+
+# Specify launch in order to start ffmpeg automatically.
+# First ffmpeg must be defined with an appropriate path if needed,
+# after that options can follow, but avoid adding the http:// field
+#Launch ffmpeg
+
+# Only allow connections from localhost to the feed.
+ACL allow 127.0.0.1
+
+</Feed>
+
+
+##################################################################
+# Now you can define each stream which will be generated from the
+# original audio and video stream. Each format has a filename (here
+# 'test1.mpg'). FFServer will send this stream when answering a
+# request containing this filename.
+
+<Stream test1.mpg>
+
+# coming from live feed 'feed1'
+Feed feed1.ffm
+
+# Format of the stream : you can choose among:
+# mpeg       : MPEG-1 multiplexed video and audio
+# mpegvideo  : only MPEG-1 video
+# mp2        : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
+# ogg        : Ogg format (Vorbis audio codec)
+# rm         : RealNetworks-compatible stream. Multiplexed audio and video.
+# ra         : RealNetworks-compatible stream. Audio only.
+# mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
+# jpeg       : Generate a single JPEG image.
+# mjpeg      : Generate a M-JPEG stream.
+# asf        : ASF compatible streaming (Windows Media Player format).
+# swf        : Macromedia Flash compatible stream
+# avi        : AVI format (MPEG-4 video, MPEG audio sound)
+Format mpeg
+
+# Bitrate for the audio stream. Codecs usually support only a few
+# different bitrates.
+AudioBitRate 32
+
+# Number of audio channels: 1 = mono, 2 = stereo
+AudioChannels 1
+
+# Sampling frequency for audio. When using low bitrates, you should
+# lower this frequency to 22050 or 11025. The supported frequencies
+# depend on the selected audio codec.
+AudioSampleRate 44100
+
+# Bitrate for the video stream
+VideoBitRate 64
+
+# Ratecontrol buffer size
+VideoBufferSize 40
+
+# Number of frames per second
+VideoFrameRate 3
+
+# Size of the video frame: WxH (default: 160x128)
+# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
+# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
+# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
+# hd1080
+VideoSize 160x128
+
+# Transmit only intra frames (useful for low bitrates, but kills frame rate).
+#VideoIntraOnly
+
+# If non-intra only, an intra frame is transmitted every VideoGopSize
+# frames. Video synchronization can only begin at an intra frame.
+VideoGopSize 12
+
+# More MPEG-4 parameters
+# VideoHighQuality
+# Video4MotionVector
+
+# Choose your codecs:
+#AudioCodec mp2
+#VideoCodec mpeg1video
+
+# Suppress audio
+#NoAudio
+
+# Suppress video
+#NoVideo
+
+#VideoQMin 3
+#VideoQMax 31
+
+# Set this to the number of seconds backwards in time to start. Note that
+# most players will buffer 5-10 seconds of video, and also you need to allow
+# for a keyframe to appear in the data stream.
+#Preroll 15
+
+# ACL:
+
+# You can allow ranges of addresses (or single addresses)
+#ACL ALLOW <first address> <last address>
+
+# You can deny ranges of addresses (or single addresses)
+#ACL DENY <first address> <last address>
+
+# You can repeat the ACL allow/deny as often as you like. It is on a per
+# stream basis. The first match defines the action. If there are no matches,
+# then the default is the inverse of the last ACL statement.
+#
+# Thus 'ACL allow localhost' only allows access from localhost.
+# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
+# allow everybody else.
+
+</Stream>
+
+
+##################################################################
+# Example streams
+
+
+# Multipart JPEG
+
+#<Stream test.mjpg>
+#Feed feed1.ffm
+#Format mpjpeg
+#VideoFrameRate 2
+#VideoIntraOnly
+#NoAudio
+#Strict -1
+#</Stream>
+
+
+# Single JPEG
+
+#<Stream test.jpg>
+#Feed feed1.ffm
+#Format jpeg
+#VideoFrameRate 2
+#VideoIntraOnly
+##VideoSize 352x240
+#NoAudio
+#Strict -1
+#</Stream>
+
+
+# Flash
+
+#<Stream test.swf>
+#Feed feed1.ffm
+#Format swf
+#VideoFrameRate 2
+#VideoIntraOnly
+#NoAudio
+#</Stream>
+
+
+# ASF compatible
+
+<Stream test.asf>
+Feed feed1.ffm
+Format asf
+VideoFrameRate 15
+VideoSize 352x240
+VideoBitRate 256
+VideoBufferSize 40
+VideoGopSize 30
+AudioBitRate 64
+StartSendOnKey
+</Stream>
+
+
+# MP3 audio
+
+#<Stream test.mp3>
+#Feed feed1.ffm
+#Format mp2
+#AudioCodec mp3
+#AudioBitRate 64
+#AudioChannels 1
+#AudioSampleRate 44100
+#NoVideo
+#</Stream>
+
+
+# Ogg Vorbis audio
+
+#<Stream test.ogg>
+#Feed feed1.ffm
+#Metadata title "Stream title"
+#AudioBitRate 64
+#AudioChannels 2
+#AudioSampleRate 44100
+#NoVideo
+#</Stream>
+
+
+# Real with audio only at 32 kbits
+
+#<Stream test.ra>
+#Feed feed1.ffm
+#Format rm
+#AudioBitRate 32
+#NoVideo
+#NoAudio
+#</Stream>
+
+
+# Real with audio and video at 64 kbits
+
+#<Stream test.rm>
+#Feed feed1.ffm
+#Format rm
+#AudioBitRate 32
+#VideoBitRate 128
+#VideoFrameRate 25
+#VideoGopSize 25
+#NoAudio
+#</Stream>
+
+
+##################################################################
+# A stream coming from a file: you only need to set the input
+# filename and optionally a new format. Supported conversions:
+#    AVI -> ASF
+
+#<Stream file.rm>
+#File "/usr/local/httpd/htdocs/tlive.rm"
+#NoAudio
+#</Stream>
+
+#<Stream file.asf>
+#File "/usr/local/httpd/htdocs/test.asf"
+#NoAudio
+#Metadata author "Me"
+#Metadata copyright "Super MegaCorp"
+#Metadata title "Test stream from disk"
+#Metadata comment "Test comment"
+#</Stream>
+
+
+##################################################################
+# RTSP examples
+#
+# You can access this stream with the RTSP URL:
+#   rtsp://localhost:5454/test1-rtsp.mpg
+#
+# A non-standard RTSP redirector is also created. Its URL is:
+#   http://localhost:8090/test1-rtsp.rtsp
+
+#<Stream test1-rtsp.mpg>
+#Format rtp
+#File "/usr/local/httpd/htdocs/test1.mpg"
+#</Stream>
+
+
+# Transcode an incoming live feed to another live feed,
+# using libx264 and video presets
+
+#<Stream live.h264>
+#Format rtp
+#Feed feed1.ffm
+#VideoCodec libx264
+#VideoFrameRate 24
+#VideoBitRate 100
+#VideoSize 480x272
+#AVPresetVideo default
+#AVPresetVideo baseline
+#AVOptionVideo flags +global_header
+#
+#AudioCodec libfaac
+#AudioBitRate 32
+#AudioChannels 2
+#AudioSampleRate 22050
+#AVOptionAudio flags +global_header
+#</Stream>
+
+##################################################################
+# SDP/multicast examples
+#
+# If you want to send your stream in multicast, you must set the
+# multicast address with MulticastAddress. The port and the TTL can
+# also be set.
+#
+# An SDP file is automatically generated by ffserver by adding the
+# 'sdp' extension to the stream name (here
+# http://localhost:8090/test1-sdp.sdp). You should usually give this
+# file to your player to play the stream.
+#
+# The 'NoLoop' option can be used to avoid looping when the stream is
+# terminated.
+
+#<Stream test1-sdp.mpg>
+#Format rtp
+#File "/usr/local/httpd/htdocs/test1.mpg"
+#MulticastAddress 224.124.0.1
+#MulticastPort 5000
+#MulticastTTL 16
+#NoLoop
+#</Stream>
+
+
+##################################################################
+# Special streams
+
+# Server status
+
+<Stream stat.html>
+Format status
+
+# Only allow local people to get the status
+ACL allow localhost
+ACL allow 192.168.0.0 192.168.255.255
+
+#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
+</Stream>
+
+
+# Redirect index.html to the appropriate site
+
+<Redirect index.html>
+URL http://www.ffmpeg.org/
+</Redirect>

doc/ffserver.texi

@@ -0,0 +1,923 @@
+\input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
+
+@settitle ffserver Documentation
+@titlepage
+@center @titlefont{ffserver Documentation}
+@end titlepage
+
+@top
+
+@contents
+
+@chapter Synopsis
+
+ffserver [@var{options}]
+
+@chapter Description
+@c man begin DESCRIPTION
+
+@command{ffserver} is a streaming server for both audio and video.
+It supports several live feeds, streaming from files and time shifting
+on live feeds. You can seek to positions in the past on each live
+feed, provided you specify a big enough feed storage.
+
+@command{ffserver} is configured through a configuration file, which
+is read at startup. If not explicitly specified, it will read from
+@file{/etc/ffserver.conf}.
+
+@command{ffserver} receives prerecorded files or FFM streams from some
+@command{ffmpeg} instance as input, then streams them over
+RTP/RTSP/HTTP.
+
+An @command{ffserver} instance will listen on some port as specified
+in the configuration file. You can launch one or more instances of
+@command{ffmpeg} and send one or more FFM streams to the port where
+ffserver is expecting to receive them. Alternately, you can make
+@command{ffserver} launch such @command{ffmpeg} instances at startup.
+
+Input streams are called feeds, and each one is specified by a
+@code{<Feed>} section in the configuration file.
+
+For each feed you can have different output streams in various
+formats, each one specified by a @code{<Stream>} section in the
+configuration file.
+
+@chapter Detailed description
+
+@command{ffserver} works by forwarding streams encoded by
+@command{ffmpeg}, or pre-recorded streams which are read from disk.
+
+Precisely, @command{ffserver} acts as an HTTP server, accepting POST
+requests from @command{ffmpeg} to acquire the stream to publish, and
+serving RTSP clients or HTTP clients GET requests with the stream
+media content.
+
+A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
+a port where @command{ffserver} is listening.
+
+Each feed is identified by a unique name, corresponding to the name
+of the resource published on @command{ffserver}, and is configured by
+a dedicated @code{Feed} section in the configuration file.
+
+The feed publish URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
+@end example
+
+where @var{ffserver_ip_address} is the IP address of the machine where
+@command{ffserver} is installed, @var{http_port} is the port number of
+the HTTP server (configured through the @option{HTTPPort} option), and
+@var{feed_name} is the name of the corresponding feed defined in the
+configuration file.
+
+Each feed is associated to a file which is stored on disk. This stored
+file is used to send pre-recorded data to a player as fast as
+possible when new content is added in real-time to the stream.
+
+A "live-stream" or "stream" is a resource published by
+@command{ffserver}, and made accessible through the HTTP protocol to
+clients.
+
+A stream can be connected to a feed, or to a file. In the first case,
+the published stream is forwarded from the corresponding feed
+generated by a running instance of @command{ffmpeg}, in the second
+case the stream is read from a pre-recorded file.
+
+Each stream is identified by a unique name, corresponding to the name
+of the resource served by @command{ffserver}, and is configured by
+a dedicated @code{Stream} section in the configuration file.
+
+The stream access HTTP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
+@end example
+
+The stream access RTSP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
+@end example
+
+@var{stream_name} is the name of the corresponding stream defined in
+the configuration file. @var{options} is a list of options specified
+after the URL which affects how the stream is served by
+@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
+and RTSP ports configured with the options @var{HTTPPort} and
+@var{RTSPPort} respectively.
+
+In case the stream is associated to a feed, the encoding parameters
+must be configured in the stream configuration. They are sent to
+@command{ffmpeg} when setting up the encoding. This allows
+@command{ffserver} to define the encoding parameters used by
+the @command{ffmpeg} encoders.
+
+The @command{ffmpeg} @option{override_ffserver} commandline option
+allows one to override the encoding parameters set by the server.
+
+Multiple streams can be connected to the same feed.
+
+For example, you can have a situation described by the following
+graph:
+
+@verbatim
+               _________       __________
+              |         |     |          |
+ffmpeg 1 -----| feed 1  |-----| stream 1 |
+    \         |_________|\    |__________|
+     \                    \
+      \                    \   __________
+       \                    \ |          |
+        \                    \| stream 2 |
+         \                    |__________|
+          \
+           \   _________       __________
+            \ |         |     |          |
+             \| feed 2  |-----| stream 3 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+ffmpeg 2 -----| feed 3  |-----| stream 4 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+              | file 1  |-----| stream 5 |
+              |_________|     |__________|
+
+@end verbatim
+
+@anchor{FFM}
+@section FFM, FFM2 formats
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
+@section Status stream
+
+@command{ffserver} supports an HTTP interface which exposes the
+current status of the server.
+
+Simply point your browser to the address of the special status stream
+specified in the configuration file.
+
+For example if you have:
+@example
+<Stream status.html>
+Format status
+
+# Only allow local people to get the status
+ACL allow localhost
+ACL allow 192.168.0.0 192.168.255.255
+</Stream>
+@end example
+
+then the server will post a page with the status information when
+the special stream @file{status.html} is requested.
+
+@section How do I make it work?
+
+As a simple test, just run the following two command lines where INPUTFILE
+is some file which you can decode with ffmpeg:
+
+@example
+ffserver -f doc/ffserver.conf &
+ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
+@end example
+
+At this point you should be able to go to your Windows machine and fire up
+Windows Media Player (WMP). Go to Open URL and enter
+
+@example
+    http://<linuxbox>:8090/test.asf
+@end example
+
+You should (after a short delay) see video and hear audio.
+
+WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
+transfer the entire file before starting to play.
+The same is true of AVI files.
+
+You should edit the @file{ffserver.conf} file to suit your needs (in
+terms of frame rates etc). Then install @command{ffserver} and
+@command{ffmpeg}, write a script to start them up, and off you go.
+
+@section What else can it do?
+
+You can replay video from .ffm files that was recorded earlier.
+However, there are a number of caveats, including the fact that the
+ffserver parameters must match the original parameters used to record the
+file. If they do not, then ffserver deletes the file before recording into it.
+(Now that I write this, it seems broken).
+
+You can fiddle with many of the codec choices and encoding parameters, and
+there are a bunch more parameters that you cannot control. Post a message
+to the mailing list if there are some 'must have' parameters. Look in
+ffserver.conf for a list of the currently available controls.
+
+It will automatically generate the ASX or RAM files that are often used
+in browsers. These files are actually redirections to the underlying ASF
+or RM file. The reason for this is that the browser often fetches the
+entire file before starting up the external viewer. The redirection files
+are very small and can be transferred quickly. [The stream itself is
+often 'infinite' and thus the browser tries to download it and never
+finishes.]
+
+@section Tips
+
+* When you connect to a live stream, most players (WMP, RA, etc) want to
+buffer a certain number of seconds of material so that they can display the
+signal continuously. However, ffserver (by default) starts sending data
+in realtime. This means that there is a pause of a few seconds while the
+buffering is being done by the player. The good news is that this can be
+cured by adding a '?buffer=5' to the end of the URL. This means that the
+stream should start 5 seconds in the past -- and so the first 5 seconds
+of the stream are sent as fast as the network will allow. It will then
+slow down to real time. This noticeably improves the startup experience.
+
+You can also add a 'Preroll 15' statement into the ffserver.conf that will
+add the 15 second prebuffering on all requests that do not otherwise
+specify a time. In addition, ffserver will skip frames until a key_frame
+is found. This further reduces the startup delay by not transferring data
+that will be discarded.
+
+@section Why does the ?buffer / Preroll stop working after a time?
+
+It turns out that (on my machine at least) the number of frames successfully
+grabbed is marginally less than the number that ought to be grabbed. This
+means that the timestamp in the encoded data stream gets behind realtime.
+This means that if you say 'Preroll 10', then when the stream gets 10
+or more seconds behind, there is no Preroll left.
+
+Fixing this requires a change in the internals of how timestamps are
+handled.
+
+@section Does the @code{?date=} stuff work.
+
+Yes (subject to the limitation outlined above). Also note that whenever you
+start ffserver, it deletes the ffm file (if any parameters have changed),
+thus wiping out what you had recorded before.
+
+The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
+of the following formats (the 'T' is literal):
+
+@example
+* YYYY-MM-DDTHH:MM:SS     (localtime)
+* YYYY-MM-DDTHH:MM:SSZ    (UTC)
+@end example
+
+You can omit the YYYY-MM-DD, and then it refers to the current day. However
+note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
+may be in the future and so is unlikely to be useful.
+
+You use this by adding the ?date= to the end of the URL for the stream.
+For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
+@c man end
+
+@chapter Options
+@c man begin OPTIONS
+
+@include fftools-common-opts.texi
+
+@section Main options
+
+@table @option
+@item -f @var{configfile}
+Read configuration file @file{configfile}. If not specified it will
+read by default from @file{/etc/ffserver.conf}.
+
+@item -n
+Enable no-launch mode. This option disables all the @code{Launch}
+directives within the various @code{<Feed>} sections. Since
+@command{ffserver} will not launch any @command{ffmpeg} instances, you
+will have to launch them manually.
+
+@item -d
+Enable debug mode. This option increases log verbosity, and directs
+log messages to stdout. When specified, the @option{CustomLog} option
+is ignored.
+@end table
+
+@chapter Configuration file syntax
+
+@command{ffserver} reads a configuration file containing global
+options and settings for each stream and feed.
+
+The configuration file consists of global options and dedicated
+sections, which must be introduced by "<@var{SECTION_NAME}
+@var{ARGS}>" on a separate line and must be terminated by a line in
+the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
+
+Currently the following sections are recognized: @samp{Feed},
+@samp{Stream}, @samp{Redirect}.
+
+A line starting with @code{#} is ignored and treated as a comment.
+
+Name of options and sections are case-insensitive.
+
+@section ACL syntax
+An ACL (Access Control List) specifies the address which are allowed
+to access a given stream, or to write a given feed.
+
+It accepts the folling forms
+@itemize
+@item
+Allow/deny access to @var{address}.
+@example
+ACL ALLOW <address>
+ACL DENY <address>
+@end example
+
+@item
+Allow/deny access to ranges of addresses from @var{first_address} to
+@var{last_address}.
+@example
+ACL ALLOW <first_address> <last_address>
+ACL DENY <first_address> <last_address>
+@end example
+@end itemize
+
+You can repeat the ACL allow/deny as often as you like. It is on a per
+stream basis. The first match defines the action. If there are no matches,
+then the default is the inverse of the last ACL statement.
+
+Thus 'ACL allow localhost' only allows access from localhost.
+'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
+allow everybody else.
+
+@section Global options
+@table @option
+@item HTTPPort @var{port_number}
+@item Port @var{port_number}
+@item RTSPPort @var{port_number}
+
+@var{HTTPPort} sets the HTTP server listening TCP port number,
+@var{RTSPPort} sets the RTSP server listening TCP port number.
+
+@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
+
+You must select a different port from your standard HTTP web server if
+it is running on the same computer.
+
+If not specified, no corresponding server will be created.
+
+@item HTTPBindAddress @var{ip_address}
+@item BindAddress @var{ip_address}
+@item RTSPBindAddress @var{ip_address}
+Set address on which the HTTP/RTSP server is bound. Only useful if you
+have several network interfaces.
+
+@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
+deprecated.
+
+@item MaxHTTPConnections @var{n}
+Set number of simultaneous HTTP connections that can be handled. It
+has to be defined @emph{before} the @option{MaxClients} parameter,
+since it defines the @option{MaxClients} maximum limit.
+
+Default value is 2000.
+
+@item MaxClients @var{n}
+Set number of simultaneous requests that can be handled. Since
+@command{ffserver} is very fast, it is more likely that you will want
+to leave this high and use @option{MaxBandwidth}.
+
+Default value is 5.
+
+@item MaxBandwidth @var{kbps}
+Set the maximum amount of kbit/sec that you are prepared to consume
+when streaming to clients.
+
+Default value is 1000.
+
+@item CustomLog @var{filename}
+Set access log file (uses standard Apache log file format). '-' is the
+standard output.
+
+If not specified @command{ffserver} will produce no log.
+
+In case the commandline option @option{-d} is specified this option is
+ignored, and the log is written to standard output.
+
+@item NoDaemon
+Set no-daemon mode. This option is currently ignored since now
+@command{ffserver} will always work in no-daemon mode, and is
+deprecated.
+
+@item UseDefaults
+@item NoDefaults
+Control whether default codec options are used for the all streams or not.
+Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
+The lastest occurrence overrides previous if multiple definitions.
+@end table
+
+@section Feed section
+
+A Feed section defines a feed provided to @command{ffserver}.
+
+Each live feed contains one video and/or audio sequence coming from an
+@command{ffmpeg} encoder or another @command{ffserver}. This sequence
+may be encoded simultaneously with several codecs at several
+resolutions.
+
+A feed instance specification is introduced by a line in the form:
+@example
+<Feed FEED_FILENAME>
+@end example
+
+where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
+
+The following options are recognized within a Feed section.
+
+@table @option
+@item File @var{filename}
+@item ReadOnlyFile @var{filename}
+Set the path where the feed file is stored on disk.
+
+If not specified, the @file{/tmp/FEED.ffm} is assumed, where
+@var{FEED} is the feed name.
+
+If @option{ReadOnlyFile} is used the file is marked as read-only and
+it will not be deleted or updated.
+
+@item Truncate
+Truncate the feed file, rather than appending to it. By default
+@command{ffserver} will append data to the file, until the maximum
+file size value is reached (see @option{FileMaxSize} option).
+
+@item FileMaxSize @var{size}
+Set maximum size of the feed file in bytes. 0 means unlimited. The
+postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
+recognized.
+
+Default value is 5M.
+
+@item Launch @var{args}
+Launch an @command{ffmpeg} command when creating @command{ffserver}.
+
+@var{args} must be a sequence of arguments to be provided to an
+@command{ffmpeg} instance. The first provided argument is ignored, and
+it is replaced by a path with the same dirname of the @command{ffserver}
+instance, followed by the remaining argument and terminated with a
+path corresponding to the feed.
+
+When the launched process exits, @command{ffserver} will launch
+another program instance.
+
+In case you need a more complex @command{ffmpeg} configuration,
+e.g. if you need to generate multiple FFM feeds with a single
+@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
+
+This option is ignored in case the commandline option @option{-n} is
+specified.
+
+@item ACL @var{spec}
+Specify the list of IP address which are allowed or denied to write
+the feed. Multiple ACL options can be specified.
+@end table
+
+@section Stream section
+
+A Stream section defines a stream provided by @command{ffserver}, and
+identified by a single name.
+
+The stream is sent when answering a request containing the stream
+name.
+
+A stream section must be introduced by the line:
+@example
+<Stream STREAM_NAME>
+@end example
+
+where @var{STREAM_NAME} specifies the unique name of the stream.
+
+The following options are recognized within a Stream section.
+
+Encoding options are marked with the @emph{encoding} tag, and they are
+used to set the encoding parameters, and are mapped to libavcodec
+encoding options. Not all encoding options are supported, in
+particular it is not possible to set encoder private options. In order
+to override the encoding options specified by @command{ffserver}, you
+can use the @command{ffmpeg} @option{override_ffserver} commandline
+option.
+
+Only one of the @option{Feed} and @option{File} options should be set.
+
+@table @option
+@item Feed @var{feed_name}
+Set the input feed. @var{feed_name} must correspond to an existing
+feed defined in a @code{Feed} section.
+
+When this option is set, encoding options are used to setup the
+encoding operated by the remote @command{ffmpeg} process.
+
+@item File @var{filename}
+Set the filename of the pre-recorded input file to stream.
+
+When this option is set, encoding options are ignored and the input
+file content is re-streamed as is.
+
+@item Format @var{format_name}
+Set the format of the output stream.
+
+Must be the name of a format recognized by FFmpeg. If set to
+@samp{status}, it is treated as a status stream.
+
+@item InputFormat @var{format_name}
+Set input format. If not specified, it is automatically guessed.
+
+@item Preroll @var{n}
+Set this to the number of seconds backwards in time to start. Note that
+most players will buffer 5-10 seconds of video, and also you need to allow
+for a keyframe to appear in the data stream.
+
+Default value is 0.
+
+@item StartSendOnKey
+Do not send stream until it gets the first key frame. By default
+@command{ffserver} will send data immediately.
+
+@item MaxTime @var{n}
+Set the number of seconds to run. This value set the maximum duration
+of the stream a client will be able to receive.
+
+A value of 0 means that no limit is set on the stream duration.
+
+@item ACL @var{spec}
+Set ACL for the stream.
+
+@item DynamicACL @var{spec}
+
+@item RTSPOption @var{option}
+
+@item MulticastAddress @var{address}
+
+@item MulticastPort @var{port}
+
+@item MulticastTTL @var{integer}
+
+@item NoLoop
+
+@item FaviconURL @var{url}
+Set favicon (favourite icon) for the server status page. It is ignored
+for regular streams.
+
+@item Author @var{value}
+@item Comment @var{value}
+@item Copyright @var{value}
+@item Title @var{value}
+Set metadata corresponding to the option. All these options are
+deprecated in favor of @option{Metadata}.
+
+@item Metadata @var{key} @var{value}
+Set metadata value on the output stream.
+
+@item UseDefaults
+@item NoDefaults
+Control whether default codec options are used for the stream or not.
+Default is @var{UseDefaults} unless disabled globally.
+
+@item NoAudio
+@item NoVideo
+Suppress audio/video.
+
+@item AudioCodec @var{codec_name} (@emph{encoding,audio})
+Set audio codec.
+
+@item AudioBitRate @var{rate} (@emph{encoding,audio})
+Set bitrate for the audio stream in kbits per second.
+
+@item AudioChannels @var{n} (@emph{encoding,audio})
+Set number of audio channels.
+
+@item AudioSampleRate @var{n} (@emph{encoding,audio})
+Set sampling frequency for audio. When using low bitrates, you should
+lower this frequency to 22050 or 11025. The supported frequencies
+depend on the selected audio codec.
+
+@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
+Set generic or private option for audio stream.
+Private option must be prefixed with codec name or codec must be defined before.
+
+@item AVPresetAudio @var{preset} (@emph{encoding,audio})
+Set preset for audio stream.
+
+@item VideoCodec @var{codec_name} (@emph{encoding,video})
+Set video codec.
+
+@item VideoBitRate @var{n} (@emph{encoding,video})
+Set bitrate for the video stream in kbits per second.
+
+@item VideoBitRateRange @var{range} (@emph{encoding,video})
+Set video bitrate range.
+
+A range must be specified in the form @var{minrate}-@var{maxrate}, and
+specifies the @option{minrate} and @option{maxrate} encoding options
+expressed in kbits per second.
+
+@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
+Set video bitrate tolerance in kbits per second.
+
+@item PixelFormat @var{pixel_format} (@emph{encoding,video})
+Set video pixel format.
+
+@item Debug @var{integer} (@emph{encoding,video})
+Set video @option{debug} encoding option.
+
+@item Strict @var{integer} (@emph{encoding,video})
+Set video @option{strict} encoding option.
+
+@item VideoBufferSize @var{n} (@emph{encoding,video})
+Set ratecontrol buffer size, expressed in KB.
+
+@item VideoFrameRate @var{n} (@emph{encoding,video})
+Set number of video frames per second.
+
+@item VideoSize (@emph{encoding,video})
+Set size of the video frame, must be an abbreviation or in the form
+@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
+in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+Default value is @code{160x128}.
+
+@item VideoIntraOnly (@emph{encoding,video})
+Transmit only intra frames (useful for low bitrates, but kills frame rate).
+
+@item VideoGopSize @var{n} (@emph{encoding,video})
+If non-intra only, an intra frame is transmitted every VideoGopSize
+frames. Video synchronization can only begin at an intra frame.
+
+@item VideoTag @var{tag} (@emph{encoding,video})
+Set video tag.
+
+@item VideoHighQuality (@emph{encoding,video})
+@item Video4MotionVector (@emph{encoding,video})
+
+@item BitExact (@emph{encoding,video})
+Set bitexact encoding flag.
+
+@item IdctSimple (@emph{encoding,video})
+Set simple IDCT algorithm.
+
+@item Qscale @var{n} (@emph{encoding,video})
+Enable constant quality encoding, and set video qscale (quantization
+scale) value, expressed in @var{n} QP units.
+
+@item VideoQMin @var{n} (@emph{encoding,video})
+@item VideoQMax @var{n} (@emph{encoding,video})
+Set video qmin/qmax.
+
+@item VideoQDiff @var{integer} (@emph{encoding,video})
+Set video @option{qdiff} encoding option.
+
+@item LumiMask @var{float} (@emph{encoding,video})
+@item DarkMask @var{float} (@emph{encoding,video})
+Set @option{lumi_mask}/@option{dark_mask} encoding options.
+
+@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
+Set generic or private option for video stream.
+Private option must be prefixed with codec name or codec must be defined before.
+
+@item AVPresetVideo @var{preset} (@emph{encoding,video})
+Set preset for video stream.
+
+@var{preset} must be the path of a preset file.
+@end table
+
+@subsection Server status stream
+
+A server status stream is a special stream which is used to show
+statistics about the @command{ffserver} operations.
+
+It must be specified setting the option @option{Format} to
+@samp{status}.
+
+@section Redirect section
+
+A redirect section specifies where to redirect the requested URL to
+another page.
+
+A redirect section must be introduced by the line:
+@example
+<Redirect NAME>
+@end example
+
+where @var{NAME} is the name of the page which should be redirected.
+
+It only accepts the option @option{URL}, which specify the redirection
+URL.
+
+@chapter Stream examples
+
+@itemize
+@item
+Multipart JPEG
+@example
+<Stream test.mjpg>
+Feed feed1.ffm
+Format mpjpeg
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Single JPEG
+@example
+<Stream test.jpg>
+Feed feed1.ffm
+Format jpeg
+VideoFrameRate 2
+VideoIntraOnly
+VideoSize 352x240
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Flash
+@example
+<Stream test.swf>
+Feed feed1.ffm
+Format swf
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+</Stream>
+@end example
+
+@item
+ASF compatible
+@example
+<Stream test.asf>
+Feed feed1.ffm
+Format asf
+VideoFrameRate 15
+VideoSize 352x240
+VideoBitRate 256
+VideoBufferSize 40
+VideoGopSize 30
+AudioBitRate 64
+StartSendOnKey
+</Stream>
+@end example
+
+@item
+MP3 audio
+@example
+<Stream test.mp3>
+Feed feed1.ffm
+Format mp2
+AudioCodec mp3
+AudioBitRate 64
+AudioChannels 1
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Ogg Vorbis audio
+@example
+<Stream test.ogg>
+Feed feed1.ffm
+Metadata title "Stream title"
+AudioBitRate 64
+AudioChannels 2
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio only at 32 kbits
+@example
+<Stream test.ra>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio and video at 64 kbits
+@example
+<Stream test.rm>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+VideoBitRate 128
+VideoFrameRate 25
+VideoGopSize 25
+</Stream>
+@end example
+
+@item
+For stream coming from a file: you only need to set the input filename
+and optionally a new format.
+
+@example
+<Stream file.rm>
+File "/usr/local/httpd/htdocs/tlive.rm"
+NoAudio
+</Stream>
+@end example
+
+@example
+<Stream file.asf>
+File "/usr/local/httpd/htdocs/test.asf"
+NoAudio
+Metadata author "Me"
+Metadata copyright "Super MegaCorp"
+Metadata title "Test stream from disk"
+Metadata comment "Test comment"
+</Stream>
+@end example
+@end itemize
+
+@c man end
+
+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
+@chapter See Also
+
+@ifhtml
+@ifset config-all
+@url{ffserver.html,ffserver},
+@end ifset
+@ifset config-not-all
+@url{ffserver-all.html,ffserver-all},
+@end ifset
+the @file{doc/ffserver.conf} example,
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg-utils.html,ffmpeg-utils},
+@url{ffmpeg-scaler.html,ffmpeg-scaler},
+@url{ffmpeg-resampler.html,ffmpeg-resampler},
+@url{ffmpeg-codecs.html,ffmpeg-codecs},
+@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
+@url{ffmpeg-formats.html,ffmpeg-formats},
+@url{ffmpeg-devices.html,ffmpeg-devices},
+@url{ffmpeg-protocols.html,ffmpeg-protocols},
+@url{ffmpeg-filters.html,ffmpeg-filters}
+@end ifhtml
+
+@ifnothtml
+@ifset config-all
+ffserver(1),
+@end ifset
+@ifset config-not-all
+ffserver-all(1),
+@end ifset
+the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
+ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
+ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
+@end ifnothtml
+
+@include authors.texi
+
+@ignore
+
+@setfilename ffserver
+@settitle ffserver video server
+
+@end ignore
+
+@bye

doc/fftools-common-opts.texi

@@ -42,20 +42,10 @@ streams, 'V' only matches video streams which are not attached pictures, video
 thumbnails or cover arts.  If @var{stream_index} is given, then it matches
 stream number @var{stream_index} of this type. Otherwise, it matches all
 streams of this type.
-@item p:@var{program_id}[:@var{stream_index}] or p:@var{program_id}[:@var{stream_type}[:@var{stream_index}]] or
-p:@var{program_id}:m:@var{key}[:@var{value}]
-In first version, if @var{stream_index} is given, then it matches the stream with number @var{stream_index}
+@item p:@var{program_id}[:@var{stream_index}]
+If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
 in the program with the id @var{program_id}. Otherwise, it matches all streams in the
-program. In the second version, @var{stream_type} is one of following: 'v' for video, 'a' for audio, 's'
-for subtitle, 'd' for data. If @var{stream_index} is also given, then it matches
-stream number @var{stream_index} of this type in the program with the id @var{program_id}.
-Otherwise, if only @var{stream_type} is given, it matches all
-streams of this type in the program with the id @var{program_id}.
-In the third version matches streams in the program with the id @var{program_id} with the metadata
-tag @var{key} having the specified value. If
-@var{value} is not given, matches streams that contain the given tag with any
-value.
-
+program.
 @item #@var{stream_id} or i:@var{stream_id}
 Match the stream by stream id (e.g. PID in MPEG-TS container).
 @item m:@var{key}[:@var{value}]
@@ -120,12 +110,6 @@ Show version.
 @item -formats
 Show available formats (including devices).
 
-@item -demuxers
-Show available demuxers.
-
-@item -muxers
-Show available muxers.
-
 @item -devices
 Show available devices.
 
@@ -163,7 +147,7 @@ Show channel names and standard channel layouts.
 Show recognized color names.
 
 @item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
-Show autodetected sources of the input device.
+Show autodetected sources of the intput device.
 Some devices may provide system-dependent source names that cannot be autodetected.
 The returned list cannot be assumed to be always complete.
 @example
@@ -178,24 +162,14 @@ The returned list cannot be assumed to be always complete.
 ffmpeg -sinks pulse,server=192.168.0.4
 @end example
 
-@item -loglevel [@var{flags}+]@var{loglevel} | -v [@var{flags}+]@var{loglevel}
-Set logging level and flags used by the library.
-
-The optional @var{flags} prefix can consist of the following values:
-@table @samp
-@item repeat
-Indicates that repeated log output should not be compressed to the first line
-and the "Last message repeated n times" line will be omitted.
-@item level
-Indicates that log output should add a @code{[level]} prefix to each message
-line. This can be used as an alternative to log coloring, e.g. when dumping the
-log to file.
-@end table
-Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single
-flag without affecting other @var{flags} or changing @var{loglevel}. When
-setting both @var{flags} and @var{loglevel}, a '+' separator is expected
-between the last @var{flags} value and before @var{loglevel}.
-
+@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
+Set the logging level used by the library.
+Adding "repeat+" indicates that repeated log output should not be compressed
+to the first line and the "Last message repeated n times" line will be
+omitted. "repeat" can also be used alone.
+If "repeat" is used alone, and with no prior loglevel set, the default
+loglevel will be used. If multiple loglevel parameters are given, using
+'repeat' will not change the loglevel.
 @var{loglevel} is a string or a number containing one of the following values:
 @table @samp
 @item quiet, -8
@@ -221,17 +195,6 @@ Show everything, including debugging information.
 @item trace, 56
 @end table
 
-For example to enable repeated log output, add the @code{level} prefix, and set
-@var{loglevel} to @code{verbose}:
-@example
-ffmpeg -loglevel repeat+level+verbose -i input output
-@end example
-Another example that enables repeated log output without affecting current
-state of @code{level} prefix flag or @var{loglevel}:
-@example
-ffmpeg [...] -loglevel +repeat
-@end example
-
 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
@@ -346,6 +309,51 @@ Possible flags for this option are:
 @item k8
 @end table
 @end table
+
+@item -opencl_bench
+This option is used to benchmark all available OpenCL devices and print the
+results. This option is only available when FFmpeg has been compiled with
+@code{--enable-opencl}.
+
+When FFmpeg is configured with @code{--enable-opencl}, the options for the
+global OpenCL context are set via @option{-opencl_options}. See the
+"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
+supported options. Amongst others, these options include the ability to select
+a specific platform and device to run the OpenCL code on. By default, FFmpeg
+will run on the first device of the first platform. While the options for the
+global OpenCL context provide flexibility to the user in selecting the OpenCL
+device of their choice, most users would probably want to select the fastest
+OpenCL device for their system.
+
+This option assists the selection of the most efficient configuration by
+identifying the appropriate device for the user's system. The built-in
+benchmark is run on all the OpenCL devices and the performance is measured for
+each device. The devices in the results list are sorted based on their
+performance with the fastest device listed first. The user can subsequently
+invoke @command{ffmpeg} using the device deemed most appropriate via
+@option{-opencl_options} to obtain the best performance for the OpenCL
+accelerated code.
+
+Typical usage to use the fastest OpenCL device involve the following steps.
+
+Run the command:
+@example
+ffmpeg -opencl_bench
+@end example
+Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
+i.e. fastest device in the list.
+Select the platform and device using the command:
+@example
+ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
+@end example
+
+@item -opencl_options options (@emph{global})
+Set OpenCL environment options. This option is only available when
+FFmpeg has been compiled with @code{--enable-opencl}.
+
+@var{options} must be a list of @var{key}=@var{value} option pairs
+separated by ':'. See the ``OpenCL Options'' section in the
+ffmpeg-utils manual for the list of supported options.
 @end table
 
 @section AVOptions

doc/filter_design.txt

@@ -5,7 +5,7 @@ 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 AVFrame structure.
+of audio samples, as stored in an AVFilterBuffer structure.
 
 
 Format negotiation
@@ -35,31 +35,32 @@ Format negotiation
   to set the formats supported on another.
 
 
-Frame references ownership and permissions
-==========================================
+Buffer references ownership and permissions
+===========================================
 
   Principle
   ---------
 
-    Audio and video data are voluminous; the frame and frame reference
+    Audio and video data are voluminous; the buffer and buffer reference
     mechanism is intended to avoid, as much as possible, expensive copies of
     that data while still allowing the filters to produce correct results.
 
-    The data is stored in buffers represented by AVFrame structures.
-    Several references can point to the same frame buffer; the buffer is
-    automatically deallocated once all corresponding references have been
-    destroyed.
+    The data is stored in buffers represented by AVFilterBuffer structures.
+    They must not be accessed directly, but through references stored in
+    AVFilterBufferRef structures. Several references can point to the
+    same buffer; the buffer is automatically deallocated once all
+    corresponding references have been destroyed.
 
     The characteristics of the data (resolution, sample rate, etc.) are
     stored in the reference; different references for the same buffer can
     show different characteristics. In particular, a video reference can
     point to only a part of a video buffer.
 
-    A reference is usually obtained as input to the filter_frame method or
-    requested using the ff_get_video_buffer or ff_get_audio_buffer
-    functions. A new reference on an existing buffer can be created with
-    av_frame_ref(). A reference is destroyed using
-    the av_frame_free() function.
+    A reference is usually obtained as input to the start_frame or
+    filter_frame method or requested using the ff_get_video_buffer or
+    ff_get_audio_buffer functions. A new reference on an existing buffer can
+    be created with the avfilter_ref_buffer. A reference is destroyed using
+    the avfilter_unref_bufferp function.
 
   Reference ownership
   -------------------
@@ -72,13 +73,17 @@ Frame references ownership and permissions
 
     Here are the (fairly obvious) rules for reference ownership:
 
-    * A reference received by the filter_frame method belongs to the
-      corresponding filter.
+    * A reference received by the filter_frame method (or its start_frame
+      deprecated version) belongs to the corresponding filter.
 
-    * A reference passed to ff_filter_frame is given away and must no longer
-      be used.
+      Special exception: for video references: the reference may be used
+      internally for automatic copying and must not be destroyed before
+      end_frame; it can be given away to ff_start_frame.
 
-    * A reference created with av_frame_ref() belongs to the code that
+    * A reference passed to ff_filter_frame (or the deprecated
+      ff_start_frame) is given away and must no longer be used.
+
+    * A reference created with avfilter_ref_buffer belongs to the code that
       created it.
 
     * A reference obtained with ff_get_video_buffer or ff_get_audio_buffer
@@ -90,32 +95,89 @@ Frame references ownership and permissions
   Link reference fields
   ---------------------
 
-    The AVFilterLink structure has a few AVFrame fields.
-
-    partial_buf is used by libavfilter internally and must not be accessed
-    by filters.
-
-    fifo contains frames queued in the filter's input. They belong to the
-    framework until they are taken by the filter.
+    The AVFilterLink structure has a few AVFilterBufferRef fields. The
+    cur_buf and out_buf were used with the deprecated
+    start_frame/draw_slice/end_frame API and should no longer be used.
+    src_buf and partial_buf are used by libavfilter internally
+    and must not be accessed by filters.
 
   Reference permissions
   ---------------------
 
-    Since the same frame data can be shared by several frames, modifying may
-    have unintended consequences. A frame is considered writable if only one
-    reference to it exists. The code owning that reference it then allowed
-    to modify the data.
+    The AVFilterBufferRef structure has a perms field that describes what
+    the code that owns the reference is allowed to do to the buffer data.
+    Different references for the same buffer can have different permissions.
 
-    A filter can check if a frame is writable by using the
-    av_frame_is_writable() function.
+    For video filters that implement the deprecated
+    start_frame/draw_slice/end_frame API, the permissions only apply to the
+    parts of the buffer that have already been covered by the draw_slice
+    method.
 
-    A filter can ensure that a frame is writable at some point of the code
-    by using the ff_inlink_make_frame_writable() function. It will duplicate
-    the frame if needed.
+    The value is a binary OR of the following constants:
 
-    A filter can ensure that the frame passed to the filter_frame() callback
-    is writable by setting the needs_writable flag on the corresponding
-    input pad. It does not apply to the activate() callback.
+    * AV_PERM_READ: the owner can read the buffer data; this is essentially
+      always true and is there for self-documentation.
+
+    * AV_PERM_WRITE: the owner can modify the buffer data.
+
+    * AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data
+      will not be modified by previous filters.
+
+    * AV_PERM_REUSE: the owner can output the buffer several times, without
+      modifying the data in between.
+
+    * AV_PERM_REUSE2: the owner can output the buffer several times and
+      modify the data in between (useless without the WRITE permissions).
+
+    * AV_PERM_ALIGN: the owner can access the data using fast operations
+      that require data alignment.
+
+    The READ, WRITE and PRESERVE permissions are about sharing the same
+    buffer between several filters to avoid expensive copies without them
+    doing conflicting changes on the data.
+
+    The REUSE and REUSE2 permissions are about special memory for direct
+    rendering. For example a buffer directly allocated in video memory must
+    not modified once it is displayed on screen, or it will cause tearing;
+    it will therefore not have the REUSE2 permission.
+
+    The ALIGN permission is about extracting part of the buffer, for
+    copy-less padding or cropping for example.
+
+
+    References received on input pads are guaranteed to have all the
+    permissions stated in the min_perms field and none of the permissions
+    stated in the rej_perms.
+
+    References obtained by ff_get_video_buffer and ff_get_audio_buffer are
+    guaranteed to have at least all the permissions requested as argument.
+
+    References created by avfilter_ref_buffer have the same permissions as
+    the original reference minus the ones explicitly masked; the mask is
+    usually ~0 to keep the same permissions.
+
+    Filters should remove permissions on reference they give to output
+    whenever necessary. It can be automatically done by setting the
+    rej_perms field on the output pad.
+
+    Here are a few guidelines corresponding to common situations:
+
+    * Filters that modify and forward their frame (like drawtext) need the
+      WRITE permission.
+
+    * Filters that read their input to produce a new frame on output (like
+      scale) need the READ permission on input and must request a buffer
+      with the WRITE permission.
+
+    * Filters that intend to keep a reference after the filtering process
+      is finished (after filter_frame returns) must have the PRESERVE
+      permission on it and remove the WRITE permission if they create a new
+      reference to give it away.
+
+    * Filters that intend to modify a reference they have kept after the end
+      of the filtering process need the REUSE2 permission and must remove
+      the PRESERVE permission if they create a new reference to give it
+      away.
 
 
 Frame scheduling
@@ -127,100 +189,11 @@ Frame scheduling
   Simple filters that output one frame for each input frame should not have
   to worry about it.
 
-  There are two design for filters: one using the filter_frame() and
-  request_frame() callbacks and the other using the activate() callback.
-
-  The design using filter_frame() and request_frame() is legacy, but it is
-  suitable for filters that have a single input and process one frame at a
-  time. New filters with several inputs, that treat several frames at a time
-  or that require a special treatment at EOF should probably use the design
-  using activate().
-
-  activate
-  --------
-
-    This method is called when something must be done in a filter; the
-    definition of that "something" depends on the semantic of the filter.
-
-    The callback must examine the status of the filter's links and proceed
-    accordingly.
-
-    The status of output links is stored in the frame_wanted_out, status_in
-    and status_out fields and tested by the ff_outlink_frame_wanted()
-    function. If this function returns true, then the processing requires a
-    frame on this link and the filter is expected to make efforts in that
-    direction.
-
-    The status of input links is stored by the status_in, fifo and
-    status_out fields; they must not be accessed directly. The fifo field
-    contains the frames that are queued in the input for processing by the
-    filter. The status_in and status_out fields contains the queued status
-    (EOF or error) of the link; status_in is a status change that must be
-    taken into account after all frames in fifo have been processed;
-    status_out is the status that have been taken into account, it is final
-    when it is not 0.
-
-    The typical task of an activate callback is to first check the backward
-    status of output links, and if relevant forward it to the corresponding
-    input. Then, if relevant, for each input link: test the availability of
-    frames in fifo and process them;  if no frame is available, test and
-    acknowledge a change of status using ff_inlink_acknowledge_status(); and
-    forward the result (frame or status change) to the corresponding input.
-    If nothing is possible, test the status of outputs and forward it to the
-    corresponding input(s). If still not possible, return FFERROR_NOT_READY.
-
-    If the filters stores internally one or a few frame for some input, it
-    can consider them to be part of the FIFO and delay acknowledging a
-    status change accordingly.
-
-    Example code:
-
-    ret = ff_outlink_get_status(outlink);
-    if (ret) {
-        ff_inlink_set_status(inlink, ret);
-        return 0;
-    }
-    if (priv->next_frame) {
-        /* use it */
-        return 0;
-    }
-    ret = ff_inlink_consume_frame(inlink, &frame);
-    if (ret < 0)
-        return ret;
-    if (ret) {
-        /* use it */
-        return 0;
-    }
-    ret = ff_inlink_acknowledge_status(inlink, &status, &pts);
-    if (ret) {
-        /* flush */
-        ff_outlink_set_status(outlink, status, pts);
-        return 0;
-    }
-    if (ff_outlink_frame_wanted(outlink)) {
-        ff_inlink_request_frame(inlink);
-        return 0;
-    }
-    return FFERROR_NOT_READY;
-
-    The exact code depends on how similar the /* use it */ blocks are and
-    how related they are to the /* flush */ block, and needs to apply these
-    operations to the correct inlink or outlink if there are several.
-
-    Macros are available to factor that when no extra processing is needed:
-
-    FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
-    FF_FILTER_FORWARD_STATUS_ALL(outlink, filter);
-    FF_FILTER_FORWARD_STATUS(inlink, outlink);
-    FF_FILTER_FORWARD_STATUS_ALL(inlink, filter);
-    FF_FILTER_FORWARD_WANTED(outlink, inlink);
-
   filter_frame
   ------------
 
-    For filters that do not use the activate() callback, this method is
-    called when a frame is pushed to the filter's input. It can be called at
-    any time except in a reentrant way.
+    This method is called when a frame is pushed to the filter's input. It
+    can be called at any time except in a reentrant way.
 
     If the input frame is enough to produce output, then the filter should
     push the output frames on the output link immediately.
@@ -249,10 +222,9 @@ Frame scheduling
   request_frame
   -------------
 
-    For filters that do not use the activate() callback, this method is
-    called when a frame is wanted on an output.
+    This method is called when a frame is wanted on an output.
 
-    For a source, it should directly call filter_frame on the corresponding
+    For an input, it should directly call filter_frame on the corresponding
     output.
 
     For a filter, if there are queued frames already ready, one of these
@@ -282,7 +254,16 @@ Frame scheduling
         }
         return 0;
 
-    Note that, except for filters that can have queued frames and sources,
-    request_frame does not push frames: it requests them to its input, and
-    as a reaction, the filter_frame method possibly will be called and do
-    the work.
+    Note that, except for filters that can have queued frames, request_frame
+    does not push frames: it requests them to its input, and as a reaction,
+    the filter_frame method possibly will be called and do the work.
+
+Legacy API
+==========
+
+  Until libavfilter 3.23, the filter_frame method was split:
+
+  - for video filters, it was made of start_frame, draw_slice (that could be
+    called several times on distinct parts of the frame) and end_frame;
+
+  - for audio filters, it was called filter_samples.

doc/formats.texi

@@ -61,10 +61,6 @@ Reduce the latency introduced by optional buffering
 Only write platform-, build- and time-independent data.
 This ensures that file and data checksums are reproducible and match between
 platforms. Its primary use is for regression testing.
-@item shortest
-Stop muxing at the end of the shortest stream.
-It may be needed to increase max_interleave_delta to avoid flushing the longer
-streams before EOF.
 @end table
 
 @item seek2any @var{integer} (@emph{input})
@@ -182,10 +178,9 @@ Default is 0.
 Correct single timestamp overflows if set to 1. Default is 1.
 
 @item flush_packets @var{integer} (@emph{output})
-Flush the underlying I/O stream after each packet. Default is -1 (auto), which
-means that the underlying protocol will decide, 1 enables it, and has the
-effect of reducing the latency, 0 disables it and may increase IO throughput in
-some cases.
+Flush the underlying I/O stream after each packet. Default 1 enables it, and
+has the effect of reducing the latency; 0 disables it and may slightly
+increase performance in some cases.
 
 @item output_ts_offset @var{offset} (@emph{output})
 Set the output time offset.

doc/general.texi

@@ -17,14 +17,6 @@ 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}.
 
-@section Alliance for Open Media libaom
-
-FFmpeg can make use of the libaom library for AV1 decoding.
-
-Go to @url{http://aomedia.org/} and follow the instructions for
-installing the library. Then pass @code{--enable-libaom} to configure to
-enable it.
-
 @section OpenJPEG
 
 FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos.  Go to
@@ -93,24 +85,6 @@ Go to @url{http://www.twolame.org/} and follow the
 instructions for installing the library.
 Then pass @code{--enable-libtwolame} to configure to enable it.
 
-@section libcodec2 / codec2 general
-
-FFmpeg can make use of libcodec2 for codec2 encoding and decoding.
-There is currently no native decoder, so libcodec2 must be used for decoding.
-
-Go to @url{http://freedv.org/}, download "Codec 2 source archive".
-Build and install using CMake. Debian users can install the libcodec2-dev package instead.
-Once libcodec2 is installed you can pass @code{--enable-libcodec2} to configure to enable it.
-
-The easiest way to use codec2 is with .c2 files, since they contain the mode information required for decoding.
-To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode option:
-@code{ffmpeg -i input.wav -mode 700C output.c2}.
-Playback is as simple as @code{ffplay output.c2}.
-For a list of supported modes, run @code{ffmpeg -h encoder=libcodec2}.
-Raw codec2 files are also supported.
-To make sense of them the mode in use needs to be specified as a format option:
-@code{ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav}.
-
 @section libvpx
 
 FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
@@ -127,29 +101,14 @@ Go to @url{http://www.wavpack.com/} and follow the instructions for
 installing the library. Then pass @code{--enable-libwavpack} to configure to
 enable it.
 
-@section libxavs
-
-FFmpeg can make use of the libxavs library for Xavs encoding.
-
-Go to @url{http://xavs.sf.net/} and follow the instructions for
-installing the library. Then pass @code{--enable-libxavs} to configure to
-enable it.
-
 @section OpenH264
 
-FFmpeg can make use of the OpenH264 library for H.264 encoding and decoding.
+FFmpeg can make use of the OpenH264 library for H.264 encoding.
 
 Go to @url{http://www.openh264.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libopenh264} to configure to
 enable it.
 
-For decoding, this library is much more limited than the built-in decoder
-in libavcodec; currently, this library lacks support for decoding B-frames
-and some other main/high profile features. (It currently only supports
-constrained baseline profile and CABAC.) Using it is mostly useful for
-testing and for taking advantage of Cisco's patent portfolio license
-(@url{http://www.openh264.org/BINARY_LICENSE.txt}).
-
 @section x264
 
 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -220,19 +179,6 @@ For Windows, supported AviSynth variants are
 For Linux and OS X, the supported AviSynth variant is
 @url{https://github.com/avxsynth/avxsynth, AvxSynth}.
 
-@float NOTE
-There is currently a regression in AviSynth+'s @code{capi.h} header as of
-October 2016, which interferes with the ability for builds of FFmpeg to use
-MSVC-built binaries of AviSynth. Until this is resolved, you can make sure
-a known good version is installed by checking out a version from before
-the regression occurred:
-
-@code{git clone -b MT git://github.com/AviSynth/AviSynthPlus.git @*
-cd AviSynthPlus @*
-git checkout -b oldheader b4f292b4dbfad149697fb65c6a037bb3810813f9 @*
-make install PREFIX=/install/prefix}
-@end float
-
 @float NOTE
 AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
 with @code{--enable-avisynth}, and the binaries will work regardless of the
@@ -251,18 +197,6 @@ The dispatcher is open source and can be downloaded from
 with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
 locate the dispatcher's @code{.pc} files.
 
-@section AMD VCE
-
-FFmpeg can use the AMD Advanced Media Framework library for accelerated H.264
-and HEVC encoding on VCE enabled hardware under Windows.
-
-To enable support you must obtain the AMF framework header files from
-@url{https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git}.
-
-Create an @code{AMF/} directory in the system include path.
-Copy the contents of @code{AMF/amf/public/include/} into that directory.
-Then configure FFmpeg with @code{--enable-amf}.
-
 
 @chapter Supported File Formats, Codecs or Features
 
@@ -328,10 +262,6 @@ library:
 @item BRSTM                     @tab   @tab X
     @tab Audio format used on the Nintendo Wii.
 @item BWF                       @tab X @tab X
-@item codec2 (raw)              @tab X @tab X
-    @tab Must be given -mode format option to decode correctly.
-@item codec2 (.c2 files)        @tab X @tab X
-    @tab Contains header with version and mode info, simplifying playback.
 @item CRI ADX                   @tab X @tab X
     @tab Audio-only format used in console video games.
 @item Discworld II BMV          @tab   @tab X
@@ -383,7 +313,6 @@ library:
 @item FunCom ISS                @tab   @tab X
     @tab Audio format used in various games from FunCom like The Longest Journey.
 @item G.723.1                   @tab X @tab X
-@item G.726                     @tab   @tab X @tab Both left- and right-justified.
 @item G.729 BIT                 @tab X @tab X
 @item G.729 raw                 @tab   @tab X
 @item GENH                      @tab   @tab X
@@ -406,8 +335,6 @@ library:
 @item iLBC                      @tab X @tab X
 @item Interplay MVE             @tab   @tab X
     @tab Format used in various Interplay computer games.
-@item Iterated Systems ClearVideo @tab     @tab  X
-    @tab I-frames only
 @item IV8                       @tab   @tab X
     @tab A format generated by IndigoVision 8000 video server.
 @item IVF (On2)                 @tab X @tab X
@@ -453,7 +380,6 @@ library:
     @tab Audio format used on the PS3.
 @item Mirillis FIC video        @tab   @tab X
     @tab No cursor rendering.
-@item MIDI Sample Dump Standard @tab   @tab X
 @item MIME multipart JPEG       @tab X @tab
 @item MSN TCP webcam            @tab   @tab X
     @tab Used by MSN Messenger webcam streams.
@@ -467,7 +393,6 @@ library:
 @item NC camera feed            @tab   @tab X
     @tab NC (AVIP NC4600) camera streams
 @item NIST SPeech HEader REsources @tab   @tab X
-@item Computerized Speech Lab NSP @tab   @tab X
 @item NTT TwinVQ (VQF)          @tab   @tab X
     @tab Nippon Telegraph and Telephone Corporation TwinVQ.
 @item Nullsoft Streaming Video  @tab   @tab X
@@ -482,10 +407,6 @@ library:
 @item QCP                       @tab   @tab X
 @item raw ADTS (AAC)            @tab X @tab X
 @item raw AC-3                  @tab X @tab X
-@item raw AMR-NB                @tab   @tab X
-@item raw AMR-WB                @tab   @tab X
-@item raw aptX                  @tab X @tab X
-@item raw aptX HD               @tab X @tab X
 @item raw Chinese AVS video     @tab X @tab X
 @item raw CRI ADX               @tab X @tab X
 @item raw Dirac                 @tab X @tab X
@@ -509,7 +430,6 @@ library:
 @item raw NULL                  @tab X @tab
 @item raw video                 @tab X @tab X
 @item raw id RoQ                @tab X @tab
-@item raw SBC                   @tab X @tab X
 @item raw Shorten               @tab   @tab X
 @item raw TAK                   @tab   @tab X
 @item raw TrueHD                @tab X @tab X
@@ -523,8 +443,6 @@ library:
 @item raw PCM signed 24 bit little-endian  @tab X @tab X
 @item raw PCM signed 32 bit big-endian  @tab X @tab X
 @item raw PCM signed 32 bit little-endian  @tab X @tab X
-@item raw PCM signed 64 bit big-endian  @tab X @tab X
-@item raw PCM signed 64 bit little-endian  @tab X @tab X
 @item raw PCM unsigned 8 bit    @tab X @tab X
 @item raw PCM unsigned 16 bit big-endian  @tab X @tab X
 @item raw PCM unsigned 16 bit little-endian  @tab X @tab X
@@ -532,8 +450,6 @@ library:
 @item raw PCM unsigned 24 bit little-endian  @tab X @tab X
 @item raw PCM unsigned 32 bit big-endian  @tab X @tab X
 @item raw PCM unsigned 32 bit little-endian  @tab X @tab X
-@item raw PCM 16.8 floating point little-endian @tab   @tab X
-@item raw PCM 24.0 floating point little-endian @tab   @tab X
 @item raw PCM floating-point 32 bit big-endian  @tab X @tab X
 @item raw PCM floating-point 32 bit little-endian  @tab X @tab X
 @item raw PCM floating-point 64 bit big-endian  @tab X @tab X
@@ -555,11 +471,10 @@ library:
     @tab Output is performed by publishing stream to RTMP server
 @item RTP                       @tab X @tab X
 @item RTSP                      @tab X @tab X
-@item Sample Dump eXchange      @tab   @tab X
 @item SAP                       @tab X @tab X
 @item SBG                       @tab   @tab X
 @item SDP                       @tab   @tab X
-@item Sega FILM/CPK             @tab X @tab X
+@item Sega FILM/CPK             @tab   @tab X
     @tab Used in many Sega Saturn console games.
 @item Silicon Graphics Movie    @tab   @tab X
 @item Sierra SOL                @tab   @tab X
@@ -570,7 +485,6 @@ library:
     @tab Multimedia format used by many games.
 @item SMJPEG                    @tab X @tab X
     @tab Used in certain Loki game ports.
-@item SMPTE 337M encapsulation  @tab   @tab X
 @item Smush                     @tab   @tab X
     @tab Multimedia format used in some LucasArts games.
 @item Sony OpenMG (OMA)         @tab X @tab X
@@ -579,7 +493,7 @@ library:
 @item Sony Wave64 (W64)         @tab X @tab X
 @item SoX native format         @tab X @tab X
 @item SUN AU format             @tab X @tab X
-@item SUP raw PGS subtitles     @tab X @tab X
+@item SUP raw PGS subtitles     @tab   @tab X
 @item SVAG                      @tab   @tab X
     @tab Audio format used in Konami PS2 games.
 @item TDSC                      @tab   @tab X
@@ -588,7 +502,7 @@ library:
     @tab Used on the Nintendo GameCube.
 @item Tiertex Limited SEQ       @tab   @tab X
     @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
-@item True Audio                @tab X @tab X
+@item True Audio                @tab   @tab X
 @item VAG                       @tab   @tab X
     @tab Audio format used in many Sony PS2 games.
 @item VC-1 test bitstream       @tab X @tab X
@@ -642,8 +556,6 @@ following image formats are supported:
     @tab Digital Picture Exchange
 @item EXR          @tab   @tab X
     @tab OpenEXR
-@item FITS         @tab X @tab X
-    @tab Flexible Image Transport System
 @item JPEG         @tab X @tab X
     @tab Progressive JPEG is not supported.
 @item JPEG 2000    @tab X @tab X
@@ -665,8 +577,6 @@ following image formats are supported:
 @item PNG          @tab X @tab X
 @item PPM          @tab X @tab X
     @tab Portable PixelMap image
-@item PSD          @tab   @tab X
-    @tab Photoshop
 @item PTX          @tab   @tab X
     @tab V.Flash PTX format
 @item SGI          @tab X @tab X
@@ -683,8 +593,6 @@ following image formats are supported:
     @tab X BitMap image format
 @item XFace @tab X @tab X
     @tab X-Face image format
-@item XPM  @tab   @tab X
-    @tab X PixMap image format
 @item XWD  @tab X @tab X
     @tab X Window Dump image format
 @end multitable
@@ -710,7 +618,6 @@ following image formats are supported:
 @item ANSI/ASCII art         @tab     @tab  X
 @item Apple Intermediate Codec @tab     @tab  X
 @item Apple MJPEG-B          @tab     @tab  X
-@item Apple Pixlet           @tab     @tab  X
 @item Apple ProRes           @tab  X  @tab  X
 @item Apple QuickDraw        @tab     @tab  X
     @tab fourcc: qdrw
@@ -727,8 +634,6 @@ following image formats are supported:
 @item Autodesk Animator Flic video  @tab     @tab  X
 @item Autodesk RLE           @tab     @tab  X
     @tab fourcc: AASC
-@item AV1                    @tab     @tab  E
-    @tab Supported through external library libaom
 @item Avid 1:1 10-bit RGB Packer  @tab  X  @tab  X
     @tab fourcc: AVrp
 @item AVS (Audio Video Standard) video  @tab     @tab  X
@@ -766,7 +671,7 @@ following image formats are supported:
 @item DFA                    @tab     @tab  X
     @tab Codec used in Chronomaster game.
 @item Dirac                  @tab  E  @tab  X
-    @tab supported though the native vc2 (Dirac Pro) encoder
+    @tab supported through external library libschroedinger
 @item Deluxe Paint Animation @tab     @tab  X
 @item DNxHD                  @tab   X @tab  X
     @tab aka SMPTE VC3
@@ -774,8 +679,6 @@ following image formats are supported:
     @tab fourcc: DUCK
 @item Duck TrueMotion 2.0    @tab     @tab  X
     @tab fourcc: TM20
-@item Duck TrueMotion 2.0 RT @tab     @tab  X
-    @tab fourcc: TR20
 @item DV (Digital Video)     @tab  X  @tab  X
 @item Dxtory capture format  @tab     @tab  X
 @item Feeble Files/ScummVM DXA  @tab     @tab  X
@@ -795,7 +698,6 @@ following image formats are supported:
 @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 FM Screen Capture Codec  @tab     @tab  X
 @item Forward Uncompressed   @tab     @tab  X
 @item Fraps                  @tab     @tab  X
 @item Go2Meeting             @tab     @tab  X
@@ -843,8 +745,7 @@ following image formats are supported:
 @item LucasArts SANM/Smush   @tab     @tab  X
     @tab Used in LucasArts games / SMUSH animations.
 @item lossless MJPEG         @tab  X  @tab  X
-@item MagicYUV Video         @tab  X  @tab  X
-@item Mandsoft Screen Capture Codec  @tab     @tab  X
+@item MagicYUV Lossless Video @tab     @tab  X
 @item Microsoft ATC Screen   @tab     @tab  X
     @tab Also known as Microsoft Screen 3.
 @item Microsoft Expression Encoder Screen  @tab     @tab  X
@@ -869,7 +770,6 @@ following image formats are supported:
 @item MPEG-4 part 2 Microsoft variant version 1  @tab     @tab  X
 @item MPEG-4 part 2 Microsoft variant version 2  @tab  X  @tab  X
 @item MPEG-4 part 2 Microsoft variant version 3  @tab  X  @tab  X
-@item Newtek SpeedHQ               @tab     @tab  X
 @item Nintendo Gamecube THP video  @tab     @tab  X
 @item NuppelVideo/RTjpeg     @tab     @tab  X
     @tab Video encoding used in NuppelVideo files.
@@ -910,9 +810,7 @@ following image formats are supported:
     @tab Texture dictionaries used by the Renderware Engine.
 @item RL2 video              @tab     @tab  X
     @tab used in some games by Entertainment Software Partners
-@item ScreenPressor          @tab     @tab  X
 @item Screenpresso           @tab     @tab  X
-@item Screen Recorder Gold Codec  @tab     @tab  X
 @item Sierra VMD video       @tab     @tab  X
     @tab Used in Sierra VMD files.
 @item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
@@ -980,7 +878,7 @@ following image formats are supported:
 @item 8SVX exponential       @tab     @tab  X
 @item 8SVX fibonacci         @tab     @tab  X
 @item AAC                    @tab EX  @tab  X
-    @tab encoding supported through internal encoder and external library libfdk-aac
+    @tab encoding supported through internal encoder and external libraries libfaac and libfdk-aac
 @item AAC+                   @tab  E  @tab  IX
     @tab encoding supported through external library libfdk-aac
 @item AC-3                   @tab IX  @tab  IX
@@ -1041,10 +939,6 @@ following image formats are supported:
 @item Amazing Studio PAF Audio @tab     @tab  X
 @item Apple lossless audio   @tab  X  @tab  X
     @tab QuickTime fourcc 'alac'
-@item aptX                   @tab  X  @tab  X
-    @tab Used in Bluetooth A2DP
-@item aptX HD                @tab  X  @tab  X
-    @tab Used in Bluetooth A2DP
 @item ATRAC1                 @tab     @tab  X
 @item ATRAC3                 @tab     @tab  X
 @item ATRAC3+                @tab     @tab  X
@@ -1052,8 +946,6 @@ following image formats are supported:
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
     @tab decoding supported through external library libcelt
-@item codec2                 @tab  E  @tab  E
-    @tab en/decoding supported through external library libcodec2
 @item Delphine Software International CIN audio  @tab     @tab  X
     @tab Codec used in Delphine Software International games.
 @item Digital Speech Standard - Standard Play mode (DSS SP) @tab     @tab  X
@@ -1062,7 +954,6 @@ following image formats are supported:
     @tab All versions except 5.1 are supported.
 @item DCA (DTS Coherent Acoustics)  @tab  X  @tab  X
     @tab supported extensions: XCh, XXCH, X96, XBR, XLL, LBR (partially)
-@item Dolby E  @tab     @tab  X
 @item DPCM id RoQ            @tab  X  @tab  X
     @tab Used in Quake III, Jedi Knight 2 and other computer games.
 @item DPCM Interplay         @tab     @tab  X
@@ -1097,7 +988,7 @@ following image formats are supported:
 @item Interplay ACM            @tab     @tab  X
 @item MACE (Macintosh Audio Compression/Expansion) 3:1  @tab     @tab  X
 @item MACE (Macintosh Audio Compression/Expansion) 6:1  @tab     @tab  X
-@item MLP (Meridian Lossless Packing)  @tab  X  @tab  X
+@item MLP (Meridian Lossless Packing)  @tab     @tab  X
     @tab Used in DVD-Audio discs.
 @item Monkey's Audio         @tab     @tab  X
 @item MP1 (MPEG audio layer 1)  @tab     @tab IX
@@ -1141,7 +1032,6 @@ following image formats are supported:
 @item PCM unsigned 32-bit little-endian  @tab  X  @tab  X
 @item PCM Zork               @tab     @tab  X
 @item QCELP / PureVoice      @tab     @tab  X
-@item QDesign Music Codec 1  @tab     @tab  X
 @item QDesign Music Codec 2  @tab     @tab  X
     @tab There are still some distortions.
 @item RealAudio 1.0 (14.4K)  @tab  X  @tab  X
@@ -1152,8 +1042,6 @@ following image formats are supported:
     @tab Real low bitrate AC-3 codec
 @item RealAudio Lossless     @tab     @tab  X
 @item RealAudio SIPR / ACELP.NET @tab     @tab  X
-@item SBC (low-complexity subband codec) @tab  X  @tab  X
-    @tab Used in Bluetooth A2DP
 @item Shorten                @tab     @tab  X
 @item Sierra VMD audio       @tab     @tab  X
     @tab Used in Sierra VMD files.
@@ -1167,7 +1055,7 @@ following image formats are supported:
     @tab supported through external library libspeex
 @item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
 @item True Audio (TTA)       @tab  X  @tab  X
-@item TrueHD                 @tab  X  @tab  X
+@item TrueHD                 @tab     @tab  X
     @tab Used in HD-DVD and Blu-Ray discs.
 @item TwinVQ (VQF flavor)    @tab     @tab  X
 @item VIMA                   @tab     @tab  X
@@ -1240,7 +1128,6 @@ performance on systems without hardware floating point support).
 @item MMSH         @tab X
 @item MMST         @tab X
 @item pipe         @tab X
-@item Pro-MPEG FEC @tab X
 @item RTMP         @tab X
 @item RTMPE        @tab X
 @item RTMPS        @tab X

doc/indevs.texi

@@ -63,51 +63,12 @@ Set the number of channels. Default is 2.
 
 @end table
 
-@section android_camera
-
-Android camera input device.
-
-This input devices uses the Android Camera2 NDK API which is
-available on devices with API level 24+. The availability of
-android_camera is autodetected during configuration.
-
-This device allows capturing from all cameras on an Android device,
-which are integrated into the Camera2 NDK API.
-
-The available cameras are enumerated internally and can be selected
-with the @var{camera_index} parameter. The input file string is
-discarded.
-
-Generally the back facing camera has index 0 while the front facing
-camera has index 1.
-
-@subsection Options
-
-@table @option
-
-@item video_size
-Set the video size given as a string such as 640x480 or hd720.
-Falls back to the first available configuration reported by
-Android if requested video size is not available or by default.
-
-@item framerate
-Set the video framerate.
-Falls back to the first available configuration reported by
-Android if requested framerate is not available or by default (-1).
-
-@item camera_index
-Set the index of the camera to use. Default is 0.
-
-@item input_queue_size
-Set the maximum number of frames to buffer. Default is 5.
-
-@end table
-
 @section avfoundation
 
 AVFoundation input device.
 
 AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
+The older QTKit framework has been marked deprecated since OSX version 10.7.
 
 The input filename has to be given in the following syntax:
 @example
@@ -254,9 +215,8 @@ need to configure with the appropriate @code{--extra-cflags}
 and @code{--extra-ldflags}.
 On Windows, you need to run the IDL files through @command{widl}.
 
-DeckLink is very picky about the formats it supports. Pixel format of the
-input can be set with @option{raw_format}.
-Framerate and video size must be determined for your device with
+DeckLink is very picky about the formats it supports. Pixel format is
+uyvy422 or v210, framerate and video size must be determined for your device with
 @command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
 of channels can be 2, 8 or 16. Note that all audio channels are bundled in one single
 audio track.
@@ -273,97 +233,24 @@ Defaults to @option{false}.
 If set to @option{true}, print a list of supported formats and exit.
 Defaults to @option{false}.
 
-@item format_code <FourCC>
-This sets the input video format to the format given by the FourCC. To see
-the supported values of your device(s) use @option{list_formats}.
-Note that there is a FourCC @option{'pal '} that can also be used
-as @option{pal} (3 letters).
-Default behavior is autodetection of the input video format, if the hardware
-supports it.
-
 @item bm_v210
-This is a deprecated option, you can use @option{raw_format} instead.
 If set to @samp{1}, video is captured in 10 bit v210 instead
 of uyvy422. Not all Blackmagic devices support this option.
 
-@item raw_format
-Set the pixel format of the captured video.
-Available values are:
-@table @samp
-@item uyvy422
-
-@item yuv422p10
-
-@item argb
-
-@item bgra
-
-@item rgb10
-
-@end table
-
 @item teletext_lines
 If set to nonzero, an additional teletext stream will be captured from the
-vertical ancillary data. Both SD PAL (576i) and HD (1080i or 1080p)
-sources are supported. In case of HD sources, OP47 packets are decoded.
-
-This option is a bitmask of the SD PAL VBI lines captured, specifically lines 6
-to 22, and lines 318 to 335. Line 6 is the LSB in the mask. Selected lines
-which do not contain teletext information will be ignored. You can use the
-special @option{all} constant to select all possible lines, or
-@option{standard} to skip lines 6, 318 and 319, which are not compatible with
-all receivers.
-
-For SD sources, ffmpeg needs to be compiled with @code{--enable-libzvbi}. For
-HD sources, on older (pre-4K) DeckLink card models you have to capture in 10
-bit mode.
+vertical ancillary data. This option is a bitmask of the VBI lines checked,
+specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB in the mask.
+Selected lines which do not contain teletext information will be ignored. You
+can use the special @option{all} constant to select all possible lines, or
+@option{standard} to skip lines 6, 318 and 319, which are not compatible with all
+receivers. Capturing teletext only works for SD PAL sources in 8 bit mode.
+To use this option, ffmpeg needs to be compiled with @code{--enable-libzvbi}.
 
 @item channels
 Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp{16}.
 Defaults to @samp{2}.
 
-@item duplex_mode
-Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
-Defaults to @samp{unset}.
-
-@item video_input
-Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
-@samp{optical_sdi}, @samp{component}, @samp{composite} or @samp{s_video}.
-Defaults to @samp{unset}.
-
-@item audio_input
-Sets the audio input source. Must be @samp{unset}, @samp{embedded},
-@samp{aes_ebu}, @samp{analog}, @samp{analog_xlr}, @samp{analog_rca} or
-@samp{microphone}. Defaults to @samp{unset}.
-
-@item video_pts
-Sets the video packet timestamp source. Must be @samp{video}, @samp{audio},
-@samp{reference}, @samp{wallclock} or @samp{abs_wallclock}.
-Defaults to @samp{video}.
-
-@item audio_pts
-Sets the audio packet timestamp source. Must be @samp{video}, @samp{audio},
-@samp{reference}, @samp{wallclock} or @samp{abs_wallclock}.
-Defaults to @samp{audio}.
-
-@item draw_bars
-If set to @samp{true}, color bars are drawn in the event of a signal loss.
-Defaults to @samp{true}.
-
-@item queue_size
-Sets maximum input buffer size in bytes. If the buffering reaches this value,
-incoming frames will be dropped.
-Defaults to @samp{1073741824}.
-
-@item audio_depth
-Sets the audio sample bit depth. Must be @samp{16} or @samp{32}.
-Defaults to @samp{16}.
-
-@item decklink_copyts
-If set to @option{true}, timestamps are forwarded as they are without removing
-the initial offset.
-Defaults to @option{false}.
-
 @end table
 
 @subsection Examples
@@ -383,131 +270,21 @@ ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
 @end example
 
 @item
-Capture video clip at 1080i50:
+Capture video clip at 1080i50 (format 11):
 @example
-ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
+ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
 @end example
 
 @item
 Capture video clip at 1080i50 10 bit:
 @example
-ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
+ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
 @end example
 
 @item
 Capture video clip at 1080i50 with 16 audio channels:
 @example
-ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
-@end example
-
-@end itemize
-
-@section kmsgrab
-
-KMS video input device.
-
-Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
-DRM object that can be passed to other hardware functions.
-
-Requires either DRM master or CAP_SYS_ADMIN to run.
-
-If you don't understand what all of that means, you probably don't want this.  Look at
-@option{x11grab} instead.
-
-@subsection Options
-
-@table @option
-
-@item device
-DRM device to capture on.  Defaults to @option{/dev/dri/card0}.
-
-@item format
-Pixel format of the framebuffer.  Defaults to @option{bgr0}.
-
-@item format_modifier
-Format modifier to signal on output frames.  This is necessary to import correctly into
-some APIs, but can't be autodetected.  See the libdrm documentation for possible values.
-
-@item crtc_id
-KMS CRTC ID to define the capture source.  The first active plane on the given CRTC
-will be used.
-
-@item plane_id
-KMS plane ID to define the capture source.  Defaults to the first active plane found if
-neither @option{crtc_id} nor @option{plane_id} are specified.
-
-@item framerate
-Framerate to capture at.  This is not synchronised to any page flipping or framebuffer
-changes - it just defines the interval at which the framebuffer is sampled.  Sampling
-faster than the framebuffer update rate will generate independent frames with the same
-content.  Defaults to @code{30}.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Capture from the first active plane, download the result to normal frames and encode.
-This will only work if the framebuffer is both linear and mappable - if not, the result
-may be scrambled or fail to download.
-@example
-ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
-@end example
-
-@item
-Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
-@example
-ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
-@end example
-
-@end itemize
-
-@section libndi_newtek
-
-The libndi_newtek input device provides capture capabilities for using NDI (Network
-Device Interface, standard created by NewTek).
-
-Input filename is a NDI source name that could be found by sending -find_sources 1
-to command line - it has no specific syntax but human-readable formatted.
-
-To enable this input device, you need the NDI SDK and you
-need to configure with the appropriate @code{--extra-cflags}
-and @code{--extra-ldflags}.
-
-@subsection Options
-
-@table @option
-
-@item find_sources
-If set to @option{true}, print a list of found/available NDI sources and exit.
-Defaults to @option{false}.
-
-@item wait_sources
-Override time to wait until the number of online sources have changed.
-Defaults to @option{0.5}.
-
-@item allow_video_fields
-When this flag is @option{false}, all video that you receive will be progressive.
-Defaults to @option{true}.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-List input devices:
-@example
-ffmpeg -f libndi_newtek -find_sources 1 -i dummy
-@end example
-
-@item
-Restream to NDI:
-@example
-ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
+ffmpeg -channels 16 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
 @end example
 
 @end itemize
@@ -712,6 +489,31 @@ $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_numbe
 
 @end itemize
 
+@section dv1394
+
+Linux DV 1394 input device.
+
+@subsection Options
+
+@table @option
+
+@item framerate
+Set the frame rate. Default is 25.
+
+@item standard
+
+Available values are:
+@table @samp
+@item pal
+
+@item ntsc
+
+@end table
+
+Default value is @code{ntsc}.
+
+@end table
+
 @section fbdev
 
 Linux framebuffer input device.
@@ -1248,6 +1050,49 @@ Record a stream from default device:
 ffmpeg -f pulse -i default /tmp/pulse.wav
 @end example
 
+@section qtkit
+
+QTKit input device.
+
+The filename passed as input is parsed to contain either a device name or index.
+The device index can also be given by using -video_device_index.
+A given device index will override any given device name.
+If the desired device consists of numbers only, use -video_device_index to identify it.
+The default device will be chosen if an empty string  or the device name "default" is given.
+The available devices can be enumerated by using -list_devices.
+
+@example
+ffmpeg -f qtkit -i "0" out.mpg
+@end example
+
+@example
+ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg
+@end example
+
+@example
+ffmpeg -f qtkit -i "default" out.mpg
+@end example
+
+@example
+ffmpeg -f qtkit -list_devices true -i ""
+@end example
+
+@subsection Options
+
+@table @option
+
+@item frame_rate
+Set frame rate. Default is 30.
+
+@item list_devices
+If set to @code{true}, print a list of devices and exit. Default is
+@code{false}.
+
+@item video_device_index
+Select the video device by index for devices with the same name (starts at 0).
+
+@end table
+
 @section sndio
 
 sndio input device.
@@ -1434,6 +1279,9 @@ To enable this input device during configuration you need libxcb
 installed on your system. It will be automatically detected during
 configuration.
 
+Alternatively, the configure option @option{--enable-x11grab} exists
+for legacy Xlib users.
+
 This device allows one to capture a region of an X11 display.
 
 The filename passed as input has the syntax:
@@ -1521,6 +1369,11 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
 @item video_size
 Set the video frame size. Default value is @code{vga}.
 
+@item use_shm
+Use the MIT-SHM extension for shared memory. Default value is @code{1}.
+It may be necessary to disable it for remote displays (legacy x11grab
+only).
+
 @item grab_x
 @item grab_y
 Set the grabbing region coordinates. They are expressed as offset from

doc/issue_tracker.txt

@@ -193,6 +193,9 @@ ffplay
 ffprobe
     issues in or related to ffprobe.c
 
+ffserver
+    issues in or related to ffserver.c
+
 postproc
     issues in libpostproc/*
 

doc/lexicon

@@ -1,27 +0,0 @@
-Common abbreviations/shorthands we use that don't need a comment
-================================================================
-
-dsp: digital signal processing
-dst/adst: (asymmetric) discrete sine transform
-ec: entropy coding or error concealment
-er: error resilience
-fdct/idct: forward/inverse discrete cosine transform
-fft: fast Fourier transform
-gop: group of pictures
-hw/sw: hardware/software
-lp: lowpass
-lpf: loop filter
-lut: lookup table
-mb: macroblock
-mc: motion compensation
-me: motion estimation
-mv: motion vector
-nal: network abstraction layer
-pel/qpel/epel/hpel/fpel: pixel / quarter-pixel / eighth-pixel / half-pixel / full-pixel
-pp: post process
-qp: quantization parameter
-rc: rate control
-sei: supplemental enhancement information
-sl: slice
-vlc: variable length coding
-vq: vector quantization

doc/libav-merge.txt

@@ -1,116 +0,0 @@
-CONTEXT
-=======
-
-The FFmpeg project merges all the changes from the Libav project
-(https://libav.org) since the origin of the fork (around 2011).
-
-With the exceptions of some commits due to technical/political disagreements or
-issues, the changes are merged on a more or less regular schedule (daily for
-years thanks to Michael, but more sparse nowadays).
-
-WHY
-===
-
-The majority of the active developers believe the project needs to keep this
-policy for various reasons.
-
-The most important one is that we don't want our users to have to choose
-between two distributors of libraries of the exact same name in order to have a
-different set of features and bugfixes. By taking the responsibility of
-unifying the two codebases, we allow users to benefit from the changes from the
-two teams.
-
-Today, FFmpeg has a much larger user database (we are distributed by every
-major distribution), so we consider this mission a priority.
-
-A different approach to the merge could have been to pick the changes we are
-interested in and drop most of the cosmetics and other less important changes.
-Unfortunately, this makes the following picks much harder, especially since the
-Libav project is involved in various deep API changes. As a result, we decide
-to virtually take everything done there.
-
-Any Libav developer is of course welcome anytime to contribute directly to the
-FFmpeg tree. Of course, we fully understand and are forced to accept that very
-few Libav developers are interested in doing so, but we still want to recognize
-their work. This leads us to create merge commits for every single one from
-Libav. The original commit appears totally unchanged with full authorship in
-our history (and the conflict are solved in the merge one). That way, not a
-single thing from Libav will be lost in the future in case some reunification
-happens, or that project disappears one way or another.
-
-DOWNSIDES
-=========
-
-Of course, there are many downsides to this approach.
-
-- It causes a non negligible merge commits pollution. We make sure there are
-  not several level of merges entangled (we do a 1:1 merge/commit), but it's
-  still a non-linear history.
-
-- Many duplicated work. For instance, we added libavresample in our tree to
-  keep compatibility with Libav when our libswresample was already covering the
-  exact same purpose. The same thing happened for various elements such as the
-  ProRes support (but differences in features, bugs, licenses, ...). There are
-  many work to do to unify them, and any help is very much welcome.
-
-- So much manpower from both FFmpeg and Libav is lost because of this mess. We
-  know it, and we don't know how to fix it. It takes incredible time to do
-  these merges, so we have even less time to work on things we personally care
-  about. The bad vibes also do not help with keeping our developers motivated.
-
-- There is a growing technical risk factor with the merges due to the codebase
-  differing more and more.
-
-MERGE GUIDELINES
-================
-
-The following gives developer guidelines on how to proceed when merging Libav commits.
-
-Before starting, you can reduce the risk of errors on merge conflicts by using
-a different merge conflict style:
-
-    $ git config --global merge.conflictstyle diff3
-
-tools/libav-merge-next-commit is a script to help merging the next commit in
-the queue. It assumes a remote named libav. It has two modes: merge, and noop.
-The noop mode creates a merge with no change to the HEAD. You can pass a hash
-as extra argument to reference a justification (it is common that we already
-have the change done in FFmpeg).
-
-Also see tools/murge, you can copy and paste a 3 way conflict into its stdin
-and it will display colored diffs. Any arguments to murge (like ones to suppress
-whitespace differences) are passed into colordiff.
-
-TODO/FIXME/UNMERGED
-===================
-
-Stuff that didn't reach the codebase:
--------------------------------------
-
-- HEVC DSP and x86 MC SIMD improvements from Libav (see https://ffmpeg.org/pipermail/ffmpeg-devel/2015-December/184777.html)
-  - 1f821750f hevcdsp: split the qpel functions by width instead of by the subpixel fraction
-  - 818bfe7f0 hevcdsp: split the epel functions by width
-  - 688417399 hevcdsp: split the pred functions by width
-  - a853388d2 hevc: change the stride of the MC buffer to be in bytes instead of elements
-  - 0cef06df0 checkasm: add HEVC MC tests
-  - e7078e842 hevcdsp: add x86 SIMD for MC
-  - 7993ec19a hevc: Add hevc_get_pixel_4/8/12/16/24/32/48/64
-- new bitstream reader (see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-April/209609.html)
-- use av_cpu_max_align() instead of hardcoding alignment requirements (see https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/215834.html)
-  - f44ec22e0 lavc: use av_cpu_max_align() instead of hardcoding alignment requirements
-  - 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
-- Support recovery from an already present HLS playlist (see 16cb06bb30)
-- Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
-
-Collateral damage that needs work locally:
-------------------------------------------
-
-- Merge proresdec2.c and proresdec_lgpl.c
-- Merge proresenc_anatoliy.c and proresenc_kostya.c
-- Fix MIPS AC3 downmix
-
-Extra changes needed to be aligned with Libav:
-----------------------------------------------
-
-- Switching our examples to the new encode/decode API (see 67d28f4a0f)
-- HEVC IDCT bit depth 12-bit support (Libav added 8 and 10 but doesn't have 12)

doc/libavcodec.texi

@@ -26,13 +26,13 @@ implementing robust and fast codecs as well as for experimentation.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
 @url{libavutil.html,libavutil}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
 libavutil(3)
 @end ifnothtml

doc/libavdevice.texi

@@ -23,13 +23,13 @@ VfW, DShow, and ALSA.
 @chapter See Also
 
 @ifhtml
-@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
 @url{ffmpeg-devices.html,ffmpeg-devices},
 @url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
 @end ifhtml
 
 @ifnothtml
-ffmpeg(1), ffplay(1), ffprobe(1),
+ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-devices(1),
 libavutil(3), libavcodec(3), libavformat(3)
 @end ifnothtml