Update for FFmpeg 4.2.2

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

.gitignore

@@ -19,12 +19,8 @@
 *.swp
 *.ver
 *.version
-*.metal.air
-*.metallib
-*.metallib.c
 *.ptx
 *.ptx.c
-*.ptx.gz
 *_g
 \#*
 .\#*
@@ -35,8 +31,8 @@
 /ffprobe
 /config.asm
 /config.h
-/config_components.h
 /coverage.info
+/avversion.h
 /lcov/
 /src
 /mapfile

.mailmap

@@ -1,24 +0,0 @@
-<jeebjp@gmail.com> <jan.ekstrom@aminocom.com>
-<sw@jkqxz.net> <mrt@jkqxz.net>
-<u@pkh.me> <cboesch@gopro.com>
-<zhilizhao@tencent.com> <quinkblack@foxmail.com>
-<zhilizhao@tencent.com> <wantlamy@gmail.com>
-<modmaker@google.com> <modmaker-at-google.com@ffmpeg.org>
-<stebbins@jetheaddev.com> <jstebbins@jetheaddev.com>
-<barryjzhao@tencent.com> <mypopydev@gmail.com>
-<barryjzhao@tencent.com> <jun.zhao@intel.com>
-<josh@itanimul.li> <joshdk@obe.tv>
-<michael@niedermayer.cc> <michaelni@gmx.at>
-<linjie.justin.fu@gmail.com> <linjie.fu@intel.com>
-<linjie.justin.fu@gmail.com> <fulinjie@zju.edu.cn>
-<ceffmpeg@gmail.com> <cehoyos@ag.or.at>
-<ceffmpeg@gmail.com> <cehoyos@rainbow.studorg.tuwien.ac.at>
-<ffmpeg@gyani.pro> <gyandoshi@gmail.com>
-<atomnuker@gmail.com> <rpehlivanov@obe.tv>
-<lizhong1008@gmail.com> <zhong.li@intel.com>
-<lizhong1008@gmail.com> <zhongli_dev@126.com>
-<andreas.rheinhardt@gmail.com> <andreas.rheinhardt@googlemail.com>
-rcombs <rcombs@rcombs.me> <rodger.combs@gmail.com>
-<thilo.borgmann@mail.de> <thilo.borgmann@googlemail.com>
-<liuqi05@kuaishou.com> <lq@chinaffmpeg.org>
-<ruiling.song83@gmail.com> <ruiling.song@intel.com>

CREDITS

@@ -1,6 +1,6 @@
-See the Git history of the project (https://git.ffmpeg.org/ffmpeg) to
+See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
 get the names of people who have contributed to FFmpeg.
 
 To check the log, you can type the command "git log" in the FFmpeg
 source directory, or browse the online repository at
-https://git.ffmpeg.org/ffmpeg
+http://source.ffmpeg.org.

Changelog

@@ -1,465 +1,303 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 6.0.1:
+version 4.2.2
- avcodec/4xm: Check for cfrm exhaustion
+- cbs_mpeg2: Fix parsing the last unit
- avformat/mov: Disallow FTYP after streams
+- cbs_mpeg2: Rearrange start code search
- doc/html: fix styling issue with Texinfo 7.0
+- cbs_mpeg2: Decompose Sequence End
- doc/html: support texinfo 7.0
+- cbs_mpeg2: Fix parsing of picture and slice headers
- Changelog: update
+- cbs: Remove useless initializations
- avformat/lafdec: Check for 0 parameters
+- mpeg2_metadata, cbs_mpeg2: Fix handling of colour_description
- avformat/lafdec: Check for 0 parameters
+- lavc/cbs_h2645_syntax_template: Fix memleak
- avfilter/buffersink: fix order of operation with = and <0
+- avcodec/cbs: Fix potential overflow
- avfilter/framesync: fix order of operation with = and <0
+- avcodec/cbs: Factor out common code for writing units
- tools/target_dec_fuzzer: Adjust threshold for CSCD
+- avcodec/ffwavesynth: Fix undefined overflow in wavesynth_synth_sample()
- avcodec/dovi_rpu: Use 64 bit in get_us/se_coeff()
+- avcodec/ffwavesynth: Fix undefined overflow in wavesynth_synth_sample()
- avformat/mov: Check that is_still_picture_avif has no trak based streams
+- avcodec/cook: Use 3 stage VLC decoding for channel_coupling
- avformat/matroskadec: Fix declaration-after-statement warnings
+- avcodec/wmalosslessdec: Fixes undefined overflow in dequantization in decode_subframe()
- Update for FFmpeg 6.0.1
+- avcodec/sonic: Check e in get_symbol()
- fftools/ffmpeg_mux_init: Restrict disabling automatic copying of metadata
+- avcodec/twinvqdec: Correct overflow in block align check
- avformat/rtsp: Use rtsp_st->stream_index
+- avcodec/vc1dec: Fix "return -1" cases
- avformat/rtsp: Use rtsp_st->stream_index
+- avcodec/vc1dec: Free sprite_output_frame on error
- avutil/tx_template: fix integer ovberflwo in fft3()
+- avcodec/atrac9dec: Clamp band_ext_data to max that can be read if skipped.
- avcodec/jpeg2000dec: Check image offset
+- avcodec/agm: Include block size in the MV check for flags == 3
- avformat/mxfdec: Check klv offset
+- avcodec/wmadec: Keep track of exponent initialization per channel
- libavutil/ppc/cpu.c: check that AT_HWCAP2 is defined
+- avcodec/iff: Check that video_size is large enough for the read parameters
- avcodec/h2645_parse: Avoid EAGAIN
+- avcodec/cbs_vp9: Check data_size
- avcodec/xvididct: Make c* unsigned to avoid undefined overflows
+- avcodec/cbs_vp9: Check index_size
- avcodec/bonk: Fix undefined overflow in predictor_calc_error()
+- avcodec/adpcm: Clip predictor for APC
- avformat/tmv: Check video chunk size
+- avcodec/targa: Check colors vs. available space
- avcodec/h264_parser: saturate dts a bit
+- avcodec/dstdec: Use get_ur_golomb_jpegls()
- avformat/asfdec_f: Saturate presentation time in marker
+- avcodec/wmavoice: Check remaining input in parse_packet_header()
- avformat/xwma: sanity check bits_per_coded_sample
+- avcodec/wmalosslessdec: Fix 2 overflows in mclms
- avformat/matroskadec: Check prebuffered_ns for overflow
+- avcodec/wmaprodec: Fixes integer overflow with 32bit samples
- avformat/wavdec: Check left avio_tell for overflow
+- avcodec/adpcm: Fix invalid shift in xa_decode()
- avformat/tta: Better totalframes check
+- avcodec/wmalosslessdec: Fix several integer issues
- avformat/rpl: Check for number_of_chunks overflow
+- avcodec/wmalosslessdec: Check that padding bits is not more than sample bits
- avformat/mov: compute absolute dts difference without overflow in mov_find_next_sample()
+- avcodec/iff: Skip overflowing runs in decode_delta_d()
- avformat/jacosubdec: Check timeres
+- avcodec/pnm: Check that the header is not truncated
- avformat/jacosubdec: avoid signed integer overflows in get_shift()
+- avcodec/mp3_header_decompress_bsf: Check sample_rate_index
- avformat/jacosubdec: Factorize code in get_shift() a bit
+- avcodec/cbs_av1_syntax_template: Check num_y_points
- avformat/sbgdec: Check for negative duration or un-representable end pts
+- avformat/rmdec: Initialize and sanity check offset in ivr_read_header()
- avcodec/escape124: Do not return random numbers
+- avcodec/agm: Do not allow MVs out of the picture area as no edge is allocated
- avcodec/apedec: Fix an integer overflow in predictor_update_filter()
+- avcodec/apedec: Fix 2 integer overflows
- tools/target_dec_fuzzer: Adjust wmapro threshold
+- avformat/id3v2: Fix double-free on error
- avcodec/wavarc: Allocate AV_INPUT_BUFFER_PADDING_SIZE
+- avcodec/wmaprodec: Set packet_loss when we error out on a sanity check
- avcodec/wavarc: Fix integer overflwo in do_stereo()
+- avcodec/wmaprodec: Check offset
- avutil/tx_template: Fix some signed integer overflows in DECL_FFT5()
+- avcodec/truemotion2: Fix 2 integer overflows in tm2_low_res_block()
- avcodec/aacdec_template: Better avoidance of signed integer overflow in imdct_and_windowing_eld()
+- avcodec/wmaprodec: Check if the channel sum of all internal contexts match the external
- tools/target_dec_fuzzer: Adjust threshold for MVHA
+- avcodec/atrac9dec: Check q_unit_cnt more completely before using it to access at9_tab_band_ext_group
- avformat/avs: Check if return code is representable
+- avcodec/fitsdec: Use lrint()
- avcodec/flacdec: Fix integer overflow in "33bit" DECODER_SUBFRAME_FIXED_WIDE()
+- avcodec/g729dec: Avoid using buf_size
- avcodec/flacdec: Fix overflow in "33bit" decorrelate
+- avcodec/g729dec: Factor block_size out
- avcodec/lcldec: Make PNG filter addressing match the code afterwards
+- avcodec/g729dec: require buf_size to be non 0
- avformat/westwood_vqa: Check chunk size
+- avcodec/alac: Fix integer overflow in lpc_prediction() with sign
- avformat/sbgdec: Check for period overflow
+- avcodec/wmaprodec: Fix buflen computation in save_bits()
- avformat/concatdec: Check in/outpoint for overflow
+- avcodec/vc1_block: Fix integer overflow in AC rescaling in vc1_decode_i_block_adv()
- avformat/mov: Check avif_info
+- avcodec/vmdaudio: Check chunk counts to avoid integer overflow
- avformat/mxfdec: Remove this_partition
+- avformat/mxfdec: Clear metadata_sets_count in mxf_read_close()
- avcodec/xvididct: Fix integer overflow in idct_row()
+- avcodec/nuv: Use ff_set_dimensions()
- avcodec/celp_math: avoid overflow in shift
+- avformat/vividas: Error out on audio packets in the absence of audio streams
- tools/target_dec_fuzzer: Adjust threshold for rtv1
+- avformat/vividas: Check and require 1 video stream
- avformat/hls: reduce default max reload to 3
+- avcodec/ffwavesynth: Fix integer overflow with pink_ts_cur/next
- avformat/format: Stop reading data at EOF during probing
+- avcodec/ralf: Fix integer overflows with the filter coefficient in decode_channel()
- avcodec/bonk: Fix integer overflow in predictor_calc_error()
+- avcodec/g729dec: Use 64bit and clip in scalar product
- avcodec/jpeg2000dec: jpeg2000 has its own lowres option
+- avcodec/mxpegdec: Check for multiple SOF
- avcodec/huffyuvdec: avoid undefined behavior with get_vlc2() failure
+- avcodec/nuv: Move comptype check up
- avcodec/cscd: Fix "CamStudio Lossless Codec 1.0" gzip files
+- avcodec/wmavoice: Fix integer overflow in synth_frame()
- avcodec/cscd: Check for CamStudio Lossless Codec 1.0 behavior in end check of LZO files
+- avcodec/rawdec: Check bits_per_coded_sample more pedantically for 16bit cases
- avcodec/mpeg4videodec: consider lowres in dest_pcm[]
+- avutil/lfg: Correct index increment type to avoid undefined behavior
- avcodec/hevcdec: Fix undefined memcpy()
+- avcodec/cngdec: Remove AV_CODEC_CAP_DELAY
- avcodec/mpeg4videodec: more unsigned in amv computation
+- avcodec/iff: Move index use after check in decodeplane8()
- avcodec/tta: fix signed overflow in decorrelate
+- avcodec/atrac3: Check for huge block aligns
- avcodec/apedec: remove unused variable
+- avcodec/ralf: use multiply instead of shift to avoid undefined behavior in decode_block()
- avcodec/apedec: Fix 48khz 24bit below insane level
+- avcodec/wmadec: Require previous exponents for reuse
- avcodec/apedec: Fix CRC for 24bps and bigendian
+- avcodec/vc1_block: Fix undefined behavior in ac prediction rescaling
- avcodec/wavarc: Check that nb_samples is not negative
+- avcodec/qdm2: The smallest header seems to have 2 bytes so treat 1 as invalid
- avcodec/wavarc: Check shift
+- avcodec/apedec: Fixes integer overflow of res+*data in do_apply_filter()
- avcodec/xvididct: Fix integer overflow in idct_row()
+- avcodec/sonic: Fix integer overflow in predictor_calc_error()
- avformat/avr: Check sample rate
+- avformat/vividas: Add EOF check in val_1 loop in track_header()
- avformat/imf_cpl: Replace NULL content_title_utf8 by ""
+- avcodec/atrac9dec: Check precision_fine/coarse
- avformat/imf_cpl: xmlNodeListGetString() can return NULL
+- avformat/mp3dec: Check that the frame fits within the probe buffer
- avcodec/aacdec_template: Fix undefined signed interger operations
+- vcodec/agm: Alloc based on coded dimensions
- avcodec/wavarc: Fix k limit
+- avcodec/wmaprodec: get frame during frame decode
- avcodec/rka: Fix integer overflow in decode_filter()
+- avcodec/interplayacm: Fix overflow of last unused value
- avformat/rka: bps < 8 is invalid
+- avcodec/adpcm: Fix undefined behavior with negative predictions in IMA OKI
- avcodec/pcm: allow Changing parameters
+- avcodec/cook: Move up and extend block_align check
- avutil/tx_template: extend to 2M
+- avcodec/sbcdec: Fix integer overflows in sbc_synthesize_four()
- avcodec/jpeg2000dec: Check for reduction factor and image offset
+- avcodec/twinvq: Check block_align
- avutil/softfloat: Basic documentation for av_sincos_sf()
+- avcodec/cook: Enlarge gain table
- avutil/softfloat: fix av_sincos_sf()
+- avcodec/cook: Check samples_per_channel earlier
- tools/target_dec_fuzzer: Adjust threshold for speex
+- avcodec/atrac3plus: Check split point in fill mode 3
- avcodec/utils: fix 2 integer overflows in get_audio_frame_duration()
+- avcodec/wmavoice: Check sample_rate
- avcodec/hevcdec: Avoid null pointer dereferences in MC
+- avcodec/xsubdec: fix overflow in alpha handling
- avcodec/takdsp: Fix integer overflows
+- avcodec/iff: Check available space before entering loop in decode_long_vertical_delta2() / decode_long_vertical_delta()
- avcodec/mpegvideo_dec: consider interlaced lowres 4:2:0 chroma in edge emulation check better
+- avcodec/apedec: Fix integer overflow in filter_3800()
- avcodec/rka: use unsigned for buf0 additions
+- avutil/lfg: Document the AVLFG struct
- avcodec/rka: Avoid undefined left shift
+- avcodec/ffv1dec: Use a different error message for the slice level CRC
- avcodec: Ignoring errors is only possible before the input end
+- avcodec/apedec: Fix undefined integer overflow in long_filter_ehigh_3830()
- avformat/jpegxl_probe: Forward error codes
+- avcodec/dstdec: Check that AC probabilities are within range
- avformat/jpegxl_probe: check length instead of blindly reading
+- avcodec/dstdec: Check read_table() for failure
- avformat/jpegxl_probe: Remove intermediate macro obfuscation around get_bits*()
+- avformat/vividas: Fix n_sb_blocks Check
- avcodec/noise_bsf: Check for wrapped frames
+- avcodec/snowenc: Set mb_num to avoid ratecontrol floating point divisions by 0.0
- avformat/oggparsetheora: clip duration within 64bit
+- avcodec/snowenc: Fix 2 undefined shifts
- avcodec/rka: avoid undefined multiply in  cmode==0
+- avformat/nutenc: Do not pass NULL to memcmp() in get_needed_flags()
- avcodec/rka: use 64bit for srate_pad computation
+- avcodec/aptx: Check the number of channels
- avcodec/bonk: Avoid undefined integer overflow in predictor_calc_error()
+- avcodec/aacdec_template: Check samplerate
- avformat/wavdec: Check that smv block fits in available space
+- avcodec/truemotion2: Fix several integer overflows in tm2_low_res_block()
- avcodec/adpcm: Fix integer overflow in intermediate in ADPCM_XMD
+- avcodec/utils: Check block_align
- avcodec/dpcm: fix undefined interger overflow in wady
+- avcodec/wmalosslessdec: Fix some integer anomalies
- avcodec/tiff: add a zero DNG_LINEARIZATION_TABLE check
+- avcodec/adpcm: Fix invalid shifts in ADPCM DTK
- avcodec/tak: Check remaining bits in ff_tak_decode_frame_header()
+- avcodec/apedec: Only clear the needed buffer space, instead of all
- avcodec/sonic: Fix two undefined integer overflows
+- avcodec/libvorbisdec: Fix insufficient input checks leading to out of array reads
- avcodec/utils: the IFF_ILBM implementation assumes that there are a multiple of 16 allocated
+- avcodec/g723_1dec: fix invalid shift with negative sid_gain
- avcodec/flacdec: Fix signed integre overflow
+- avcodec/vp5: Check render_x/y
- avcodec/exr: Cleanup befor return
+- avcodec/hcom: Check the root entry and the right entries
- avcodec/pngdec: Do not pass AVFrame into global header decode
+- avcodec/qdrw: Check input for header/skiped space before get_buffer()
- avcodec/pngdec: remove AVFrame argument from decode_iccp_chunk()
+- avcodec/ralf: Skip initializing unused filter variables
- avcodec/wavarc: Check order before using it to write the list
+- avcodec/takdec: Fix overflow with large sample rates
- avcodec/bonk: decode multiple passes in intlist_read() at once
+- avcodec/atrac9dec: Set channels
- avcodec/vorbisdec: Check codebook float values to be finite
+- avcodec/alsdec: Check that input space for header exists in read_diff_float_data()
- avcodec/g2meet: Replace fake allocation avoidance for framebuf
+- avformat/pjsdec: Check duration for overflow
- avutil/tx_priv: Use unsigned in BF() to avoid signed overflows
+- avcodec/agm: Check for reference frame earlier
- avcodec/lcldec: More space for rgb24
+- avcodec/ptx: Check that the input contains at least one line
- avcodec/lcldec: Support 4:1:1 and 4:2:2 with odd width
+- avcodec/alac: Fix integer overflow in LPC
- libavcodec/lcldec: width and height should not be unsigned
+- avcodec/smacker: Fix integer overflows in pred[] in smka_decode_frame()
- avformat/imf: fix invalid resource handling
+- avcodec/aliaspixdec: Check input size against minimal picture size
- avcodec/escape124: Check that blocks are allocated before use
+- avcodec/ffwavesynth: Fix integer overflows in pink noise addition
- avcodec/rka: Fix signed integer overflow in decode_filter()
+- avcodec/vc1_block: Fixes integer overflow in vc1_decode_i_block_adv()
- avcodec/huffyuvdec: Fix undefined behavior with shift
+- avcodec/wmalosslessdec: Check block_align
- avcodec/j2kenc: Replace RGB24 special case by generic test
+- avcodec/g729dec: Avoid computing invalid temporary pointers for ff_acelp_weighted_vector_sum()
- avcodec/j2kenc: Replace BGR48 / GRAY16 test by test for number of bits
+- avcodec/g729postfilter: Fix left shift of negative value
- avcodec/j2kenc: simplify pixel format setup
+- avcodec/binkaudio: Check sample rate
- avcodec/j2kenc: Fix funky bpno errors on decoding
+- avcodec/sbcdec: Fix integer overflows in sbc_synthesize_eight()
- avcodec/j2kenc: remove misleading pred value
+- avcodec/adpcm: Check initial predictor for ADPCM_IMA_EA_EACS
- avcodec/j2kenc: fix 5/3 DWT identifer
+- avcodec/g723_1dec: Fix overflow in shift
- avcodec/vp3: Check width to avoid assertion failure
+- avcodec/apedec: Fix integer overflow in predictor_update_3930()
- avcodec/g729postfilter: Limit shift in long term filter
+- avcodec/g729postfilter: Fix undefined intermediate pointers
- avcodec/wavarc: Fix several integer overflows
+- avcodec/g729postfilter: Fix undefined shifts
- avcodec/tests/snowenc: Fix 2nd test
+- avcodec/lsp: Fix undefined shifts in lsp2poly()
- avcodec/tests/snowenc: return a failure if DWT/IDWT mismatches
+- avcodec/adpcm: Fix left shifts in AV_CODEC_ID_ADPCM_EA
- avcodec/snowenc: Fix visual weight calculation
+- avformat/shortendec: Check k in probe
- avcodec/tests/snowenc: unbreak DWT tests
+- avfilter/vf_geq: Use av_clipd() instead of av_clipf()
- avcodec/mpeg12dec: Check input size
+- avcodec/wmaprodec: Check that the streams channels do not exceed the overall channels
- avcodec/escape124: Fix some return codes
+- avcodec/qdmc: Check input space in qdmc_get_vlc()
- avcodec/escape124: fix signdness of end of input check
+- avcodec/wmaprodec: Fix cleanup on error
- Use https for repository links
+- avcodec/pcm: Check bits_per_coded_sample
- avcodec/nvdec_hevc: fail to initialize on unsupported profiles
+- avcodec/exr: Allow duplicate use of channel indexes
- fftools/ffmpeg_enc: apply -top to individual encoded frames
+- avcodec/fitsdec: Fail on 0 naxisn
- avcodec/on2avc: use correct fft sizes
+- avcodec/dxv: Subtract 12 earlier in dxv_decompress_cocg()
- avcodec/on2avc: use the matching AVTX context for the 512 sized iMDCT
+- libavcodec/dxv: Remove redundant seek
- examples: fix build of mux and resample_audio
+- avcodec/ituh263dec: Check input for minimal frame size
- avcodec/nvenc: stop using deprecated rc modes with SDK 12.1
+- avcodec/truemotion1: Check that the input has enough space for a minimal index_stream
- configure: use non-deprecated nvenc GUID for conftest
+- avformat/mpsubdec: Clear queue on error
- avcodec/x86/mathops: clip constants used with shift instructions within inline assembly
+- avcodec/sunrast: Check that the input is large enough for the maximally compressed image
- avfilter/vsrc_ddagrab: calculate pointer position on rotated screens
+- avcodec/sunrast: Check for availability of maplength before allocating image
- avfilter/vsrc_ddagrab: account for mouse-only frames during probing
+- avformat/subtitles: Check nb_subs in ff_subtitles_queue_finalize()
- avcodec/aac_ac3_parser: add preprocessor checks for codec specific code
+- avcodec/vc1_block: Fix invalid left shift in vc1_decode_p_mb()
- avcodec/nvenc: handle frame durations and AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE
+- avcodec/wmaprodec: Check if there is a stream
- Revert "lavc/nvenc: handle frame durations and AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE"
+- avcodec/g2meet: Check for end of input in jpg_decode_block()
- Revert "avcodec/nvenc: fix b-frame DTS behavior with fractional framerates"
+- avcodec/g2meet: Check if adjusted pixel was on the stack
- avcodec/vdpau_mpeg4: fix order of quant matrix coefficients
+- avformat/electronicarts: If no packet has been read at the end do not treat it as if theres a packet
- avcodec/vdpau_mpeg12: fix order of quant matrix coefficients
+- avcodec/dxv: Check op_offset in dxv_decompress_yo()
- avcodec/nvdec_mpeg4: fix order of quant matrix coefficients
+- avcodec/utils: Check sample_rate before opening the decoder
- avcodec/nvdec_mpeg2: fix order of quant matrix coefficients
+- avcodec/aptx: Fix multiple shift anomalies
- fftools/ffmpeg_filter: fix leak of AVIOContext in read_binary()
+- avcodec/fitsdec: fix use of uninitialised values
- fftools/ffmpeg: avoid possible invalid reads with short -tag values
+- avcodec/motionpixels: Mark 2 functions as always_inline
- avcodec/mp_cmp: reject invalid comparison function values
+- avcodec/ituh263dec: Make the condition for the studio slice start code match between ff_h263_resync() and ff_mpeg4_decode_studio_slice_header()
- avcodec/aacpsy: clip global_quality within the psy_vbr_map array boundaries
+- avcodec/ralf: Fix integer overflow in decode_channel()
- avutil/wchar_filename: propagate MultiByteToWideChar() and WideCharToMultiByte() failures
+- vcodec/vc1: compute rangex/y only for P/B frames
- avformat/concatf: check if any nodes were allocated
+- avcodec/vc1_pred: Fix invalid shifts in scaleforopp()
- avcodec/nvenc: fix b-frame DTS behavior with fractional framerates
+- avcodec/vc1_block: Fix invalid shift with rangeredfrm
- avcodec/vorbisdec: export skip_samples instead of dropping frames
+- avcodec/vc1: Check for excessive resolution
- fftools/ffmpeg_mux_init: avoid invalid reads in forced keyframe parsing
+- avcodec/vc1: check REFDIST
- lavfi/vf_vpp_qsv: set the right timestamp for AVERROR_EOF
+- avcodec/apedec: Fix several integer overflows in predictor_update_filter() and do_apply_filter()
- avfilter/vf_untile: swap the chroma shift values used for plane offsets
+- avcodec/hevc_cabac: Tighten the limit on k in ff_hevc_cu_qp_delta_abs()
- lavc/decode: stop mangling last_pkt_props->opaque
+- avcodec/4xm: Check index in decode_i_block() also in the path where its not used.
- avcodec/nvenc: avoid failing b_ref_mode check when unset
+- avcodec/loco: Check for end of input in the first line
- lavu/vulkan: fix handle type for 32-bit targets
+- avcodec/atrac3: Check block_align
- vulkan: Fix win/i386 calling convention
+- avcodec/alsdec: Avoid dereferencing context pointer in inner interleave loop
- avfilter/graphparser: fix filter instance name when an id is provided
+- avcodec/hcom: Check that there are dictionary entries
- avcodec/aacps_tablegen: fix build error after avutil bump
+- avcodec/fitsdec: Prevent division by 0 with huge data_max
- avcodec/nvenc: fix potential NULL pointer dereference
+- avcodec/dstdec: Fix integer overflow in samples_per_frame computation
-
+- avcodec/g729_parser: Check block_size
-
+- avcodec/sbcdec: Initialize number of channels
-version 6.0:
+- avcodec/utils: Optimize ff_color_frame() using memcpy()
-- Radiance HDR image support
+- avcodec/aacdec: Check if we run out of input in read_stream_mux_config()
-- ddagrab (Desktop Duplication) video capture filter
+- avcodec/utils: Use av_memcpy_backptr() in ff_color_frame()
-- ffmpeg -shortest_buf_duration option
+- avcodec/smacker: Fix integer overflow in signed int multiply in SMK_BLK_FILL
-- ffmpeg now requires threading to be built
+- avcodec/alac: Fix invalid shifts in 20/24 bps
-- ffmpeg now runs every muxer in a separate thread
+- avcodec/alac: fix undefined behavior with INT_MIN in lpc_prediction()
-- Add new mode to cropdetect filter to detect crop-area based on motion vectors and edges
+- avcodec/ffwavesynth: Fix integer overflow in timestamps
-- VAAPI decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
+- avformat/vividas: Test size and packet numbers a bit more
-- WBMP (Wireless Application Protocol Bitmap) image format
+- avformat/vividas: Check n_sb_blocks against input space
-- a3dscope filter
+- avcodec/dxv: Check op_offset in both directions
-- bonk decoder and demuxer
+- avcodec/adpcm: Check number of channels for MTAF
-- Micronas SC-4 audio decoder
+- avcodec/sunrast: Fix indention
-- LAF demuxer
+- avcodec/sunrast: Fix return type for "unsupported (compression) type"
-- APAC decoder and demuxer
+- avcodec/utils: Check channels fully earlier
-- Media 100i decoders
+- avformat/mov: Check for EOF in mov_read_meta()
-- DTS to PTS reorder bsf
+- avcodec/hevcdec: Fix memleak of a53_caption
-- ViewQuest VQC decoder
+- avformat/vividas: Remove align offset which is always masked off
-- backgroundkey filter
+- avformat/vividas: remove dead assignment
-- nvenc AV1 encoding support
+- avformat/cdxl: Fix integer overflow in intermediate
-- MediaCodec decoder via NDKMediaCodec
+- avcodec/hevcdec: repeat character in skiped
-- MediaCodec encoder
+- repeat an even number of characters in occured
-- oneVPL support for QSV
+- avcodec/gdv: Replace assert() checking bitstream by if()
-- QSV AV1 encoder
+- libavcodec/utils: Free threads on init failure
-- QSV decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
+- avcodec/htmlsubtitles: Avoid locale dependant isdigit()
-- showcwt multimedia filter
+- avcodec/alsdec: Check k from being outside what our implementation can handle
-- corr video filter
+- avcodec/takdec: Fix integer overflow in decorrelate()
-- adrc audio filter
+- avcodec/aacps: Fix integer overflows in hybrid_synthesis()
-- afdelaysrc audio filter
+- avcodec/mpeg4videodec: Fix integer overflow in mpeg4_decode_studio_block()
-- WADY DPCM decoder and demuxer
+- avcodec/vp56rac: delay signaling an error on truncated input
-- CBD2 DPCM decoder
+- avcodec/pnm_parser: Use memchr() in pnm_parse()
-- ssim360 video filter
+- tests: Fix bash errors in lavf_container tests.
-- ffmpeg CLI new options: -stats_enc_pre[_fmt], -stats_enc_post[_fmt],
+- avformat/matroskadec: Fix use-after-free when demuxing ProRes
-  -stats_mux_pre[_fmt]
+- avformat/matroskadec: Fix demuxing ProRes
-- hstack_vaapi, vstack_vaapi and xstack_vaapi filters
+- avcodec/cbs_av1: fix array size for ar_coeffs_cb_plus_128 and ar_coeffs_cr_plus_128
-- XMD ADPCM decoder and demuxer
+- avcodec/cbs_av1: avoid reading trailing bits when obu type is OBU_TILE_LIST
-- media100 to mjpegb bsf
+- lavc/cbs_h2645: Fix incorrect max size of nalu unit
-- ffmpeg CLI new option: -fix_sub_duration_heartbeat
+- avcodec/extract_extradata_bsf: Don't unref uninitialized buffers
-- WavArc decoder and demuxer
+- avformat/av1: Fix leak of dynamic buffer in case of parsing failure
-- CrystalHD decoders deprecated
+- libavformat/rtsp: return error if rtsp_hd_out is null instead of crash
-- SDNS demuxer
+- cbs_h264: Fix missing inferred colour description fields
-- RKA decoder and demuxer
+- avcodec/cbs_av1: keep separate reference frame state for reading and writing
-- filtergraph syntax in ffmpeg CLI now supports passing file contents
+- avcodec/cbs_av1: fix reading reference order hint in skip_mode_params()
-  as option values, by prefixing option name with '/'
+- avcodec/amfnec: allocate packets using av_new_packet()
-- hstack_qsv, vstack_qsv and xstack_qsv filters
+- avcodec/nvenc: make sure newly allocated packets are refcounted
-
+- lavc/mpeg4audio: add chan_config check to avoid indeterminate channels
-
+- aformat/movenc: add missing padding to output track extradata
-version 5.1:
+- avcodec/nvenc: add driver version info for SDK 9.1
-- add ipfs/ipns gateway support
+- avcodec/bsf: check that AVBSFInternal was allocated before dereferencing it
-- dialogue enhance audio filter
-- dropped obsolete XvMC hwaccel
-- pcm-bluray encoder
-- DFPWM audio encoder/decoder and raw muxer/demuxer
-- SITI filter
-- Vizrt Binary Image encoder/decoder
-- avsynctest source filter
-- feedback video filter
-- pixelize video filter
-- colormap video filter
-- colorchart video source filter
-- multiply video filter
-- PGS subtitle frame merge bitstream filter
-- blurdetect filter
-- tiltshelf audio filter
-- QOI image format support
-- ffprobe -o option
-- virtualbass audio filter
-- VDPAU AV1 hwaccel
-- PHM image format support
-- remap_opencl filter
-- added chromakey_cuda filter
-- added bilateral_cuda filter
-
-
-version 5.0:
-- ADPCM IMA Westwood encoder
-- Westwood AUD muxer
-- ADPCM IMA Acorn Replay decoder
-- Argonaut Games CVG demuxer
-- Argonaut Games CVG muxer
-- Concatf protocol
-- afwtdn audio filter
-- audio and video segment filters
-- Apple Graphics (SMC) encoder
-- hsvkey and hsvhold video filters
-- adecorrelate audio filter
-- atilt audio filter
-- grayworld video filter
-- AV1 Low overhead bitstream format muxer
-- swscale slice threading
-- MSN Siren decoder
-- scharr video filter
-- apsyclip audio filter
-- morpho video filter
-- amr parser
-- (a)latency filters
-- GEM Raster image decoder
-- asdr audio filter
-- speex decoder
-- limitdiff video filter
-- xcorrelate video filter
-- varblur video filter
-- huesaturation video filter
-- colorspectrum source video filter
-- RTP packetizer for uncompressed video (RFC 4175)
-- bitpacked encoder
-- VideoToolbox VP9 hwaccel
-- VideoToolbox ProRes hwaccel
-- support loongarch.
-- aspectralstats audio filter
-- adynamicsmooth audio filter
-- libplacebo filter
-- vflip_vulkan, hflip_vulkan and flip_vulkan filters
-- adynamicequalizer audio filter
-- yadif_videotoolbox filter
-- VideoToolbox ProRes encoder
-- anlmf audio filter
-- IMF demuxer (experimental)
-
-
-version 4.4:
-- AudioToolbox output device
-- MacCaption demuxer
-- PGX decoder
-- chromanr video filter
-- VDPAU accelerated HEVC 10/12bit decoding
-- ADPCM IMA Ubisoft APM encoder
-- Rayman 2 APM muxer
-- AV1 encoding support SVT-AV1
-- Cineform HD encoder
-- ADPCM Argonaut Games encoder
-- Argonaut Games ASF muxer
-- AV1 Low overhead bitstream format demuxer
-- RPZA video encoder
-- ADPCM IMA MOFLEX decoder
-- MobiClip FastAudio decoder
-- MobiClip video decoder
-- MOFLEX demuxer
-- MODS demuxer
-- PhotoCD decoder
-- MCA demuxer
-- AV1 decoder (Hardware acceleration used only)
-- SVS demuxer
-- Argonaut Games BRP demuxer
-- DAT demuxer
-- aax demuxer
-- IPU decoder, parser and demuxer
-- Intel QSV-accelerated AV1 decoding
-- Argonaut Games Video decoder
-- libwavpack encoder removed
-- ACE demuxer
-- AVS3 demuxer
-- AVS3 video decoder via libuavs3d
-- Cintel RAW decoder
-- VDPAU accelerated VP9 10/12bit decoding
-- afreqshift and aphaseshift filters
-- High Voltage Software ADPCM encoder
-- LEGO Racers ALP (.tun & .pcm) muxer
-- AV1 VAAPI decoder
-- adenorm filter
-- ADPCM IMA AMV encoder
-- AMV muxer
-- NVDEC AV1 hwaccel
-- DXVA2/D3D11VA hardware accelerated AV1 decoding
-- speechnorm filter
-- SpeedHQ encoder
-- asupercut filter
-- asubcut filter
-- Microsoft Paint (MSP) version 2 decoder
-- Microsoft Paint (MSP) demuxer
-- AV1 monochrome encoding support via libaom >= 2.0.1
-- asuperpass and asuperstop filter
-- shufflepixels filter
-- tmidequalizer filter
-- estdif filter
-- epx filter
-- Dolby E parser
-- shear filter
-- kirsch filter
-- colortemperature filter
-- colorcontrast filter
-- PFM encoder
-- colorcorrect filter
-- binka demuxer
-- XBM parser
-- xbm_pipe demuxer
-- colorize filter
-- CRI parser
-- aexciter audio filter
-- exposure video filter
-- monochrome video filter
-- setts bitstream filter
-- vif video filter
-- OpenEXR image encoder
-- Simbiosis IMX decoder
-- Simbiosis IMX demuxer
-- Digital Pictures SGA demuxer and decoders
-- TTML subtitle encoder and muxer
-- identity video filter
-- msad video filter
-- gophers protocol
-- RIST protocol via librist
-
-
-version 4.3:
-- v360 filter
-- Intel QSV-accelerated MJPEG decoding
-- Intel QSV-accelerated VP9 decoding
-- Support for TrueHD in mp4
-- Support AMD AMF encoder on Linux (via Vulkan)
-- IMM5 video decoder
-- ZeroMQ protocol
-- support Sipro ACELP.KELVIN decoding
-- streamhash muxer
-- sierpinski video source
-- scroll video filter
-- photosensitivity filter
-- anlms filter
-- arnndn filter
-- bilateral filter
-- maskedmin and maskedmax filters
-- VDPAU VP9 hwaccel
-- median filter
-- QSV-accelerated VP9 encoding
-- AV1 encoding support via librav1e
-- AV1 frame merge bitstream filter
-- AV1 Annex B demuxer
-- axcorrelate filter
-- mvdv decoder
-- mvha decoder
-- MPEG-H 3D Audio support in mp4
-- thistogram filter
-- freezeframes filter
-- Argonaut Games ADPCM decoder
-- Argonaut Games ASF demuxer
-- xfade video filter
-- xfade_opencl filter
-- afirsrc audio filter source
-- pad_opencl filter
-- Simon & Schuster Interactive ADPCM decoder
-- Real War KVAG demuxer
-- CDToons video decoder
-- siren audio decoder
-- Rayman 2 ADPCM decoder
-- Rayman 2 APM demuxer
-- cas video filter
-- High Voltage Software ADPCM decoder
-- LEGO Racers ALP (.tun & .pcm) demuxer
-- AMQP 0-9-1 protocol (RabbitMQ)
-- Vulkan support
-- avgblur_vulkan, overlay_vulkan, scale_vulkan and chromaber_vulkan filters
-- ADPCM IMA MTF decoder
-- FWSE demuxer
-- DERF DPCM decoder
-- DERF demuxer
-- CRI HCA decoder
-- CRI HCA demuxer
-- overlay_cuda filter
-- switch from AvxSynth to AviSynth+ on Linux
-- mv30 decoder
-- Expanded styling support for 3GPP Timed Text Subtitles (movtext)
-- WebP parser
-- tmedian filter
-- maskedthreshold filter
-- Support for muxing pcm and pgs in m2ts
-- Cunning Developments ADPCM decoder
-- asubboost filter
-- Pro Pinball Series Soundbank demuxer
-- pcm_rechunk bitstream filter
-- scdet filter
-- NotchLC decoder
-- gradients source video filter
-- MediaFoundation encoder wrapper
-- untile filter
-- Simon & Schuster Interactive ADPCM encoder
-- PFM decoder
-- dblur video filter
-- Real War KVAG muxer
 
+version 4.2.1:
+- avformat/vividas: check for tiny blocks using alignment
+- avcodec/vc1_pred: Fix refdist in scaleforopp()
+- avcodec/vorbisdec: fix FASTDIV usage for vr_type == 2
+- avcodec/iff: Check for overlap in cmap_read_palette()
+- avcodec/apedec: Fix 32bit int overflow in do_apply_filter()
+- lavf/rawenc: Only accept the appropriate stream type for raw muxers.
+- avformat/matroskadec: use av_fast_realloc to reallocate ebml list arrays
+- avformat/matroskadec: use proper types for some EbmlSyntax fields
+- avcodec/ralf: fix undefined shift in extend_code()
+- avcodec/ralf: fix undefined shift
+- avcodec/bgmc: Check input space in ff_bgmc_decode_init()
+- avcodec/vp3: Check for end of input in 2 places of vp4_unpack_macroblocks()
+- avcodec/truemotion2: Fix multiple integer overflows in tm2_null_res_block()
+- avcodec/vc1_block: Check the return code from vc1_decode_p_block()
+- avcodec/vc1dec: Require res_sprite for wmv3images
+- avcodec/vc1_block: Check for double escapes
+- avcodec/vorbisdec: Check get_vlc2() failure
+- avcodec/tta: Fix integer overflow in prediction
+- avcodec/vb: Check input packet size to be large enough to contain flags
+- avcodec/cavsdec: Limit the number of access units per packet to 2
+- avcodec/atrac9dec: Check block_align
+- avcodec/alac: Check for bps of 0
+- avcodec/alac: Fix multiple integer overflows in lpc_prediction()
+- avcodec/rl2: set dimensions
+- avcodec/aacdec: Add FF_CODEC_CAP_INIT_CLEANUP
+- avcodec/idcinvideo: Add 320x240 default maximum resolution
+- avformat/realtextdec: free queue on error
+- avcodec/vp5/6/8: use vpX_rac_is_end()
+- avformat/vividas: Check av_xiphlacing() return value before use
+- avcodec/alsdec: Fix integer overflow in decode_var_block_data()
+- avcodec/alsdec: Limit maximum channels to 512
+- avcodec/anm: Check input size for a frame with just a stop code
+- avcodec/flicvideo: Optimize and Simplify FLI_COPY in flic_decode_frame_24BPP() by using bytestream2_get_buffer()
+- avcodec/loco: Check left column value
+- avcodec/ffwavesynth: Fixes invalid shift with pink noise seeking
+- avcodec/ffwavesynth: Fix integer overflow for some corner case values
+- avcodec/indeo2: Check remaining input more often
+- avcodec/diracdec: Check that slices are fewer than pixels
+- avcodec/vp56: Consider the alpha start as end of the prior header
+- avcodec/4xm: Check for end of input in decode_p_block()
+- avcodec/hevcdec: Check delta_luma_weight_l0/1
+- avcodec/hnm4video: Optimize postprocess_current_frame()
+- avcodec/hevc_refs: Optimize 16bit generate_missing_ref()
+- avcodec/scpr: Use av_memcpy_backptr() in type 17 and 33
+- avcodec/tiff: Enforce increasing offsets
+- avcodec/dds: Use ff_set_dimensions()
+- avformat/vividas: Fix another infinite loop
+- avformat/vividas: Fix infinite loop in header parser
+- avcodec/mpc8: Fix 32bit mask/enum
+- avcodec/alsdec: Fix integer overflows of raw_samples in decode_var_block_data()
+- avcodec/alsdec: Fix integer overflow of raw_samples in decode_blocks()
+- avcodec/alsdec: fix mantisse shift
+- avcodec/pngdec: consider chunk size in minimal size check
+- avcodec/vc1_block: Fix invalid shifts in vc1_decode_i_blocks()
+- avcodec/vc1_block: fix invalid shift in vc1_decode_p_mb()
+- avcodec/aacdec_template: fix integer overflow in imdct_and_windowing()
+- avformat/mpegts: Check if ready on SCTE reception
+- avcodec/omx: fix xFramerate calculation
+- avformat/avidec: add support for recognizing HEVC fourcc when demuxing
+- avformat/mpegts: fix teletext PTS when selecting teletext streams only
+- avcodec/h2645_parse: zero initialize the rbsp buffer
+- avcodec/omx: Fix handling of fragmented buffers
+- avcodec/omx: ensure zerocopy mode can be disabled on rpi builds
+- avformat/mxfdec: do not ignore bad size errors
+- avformat/matroskadec: Fix seeking
+- ffplay: properly detect all window size changes
 
 version 4.2:
 - tpad filter

LICENSE.md

@@ -21,11 +21,10 @@ Specifically, the GPL parts of FFmpeg are:
     - `compat/solaris/make_sunver.pl`
     - `doc/t2h.pm`
     - `doc/texi2pod.pl`
-    - `libswresample/tests/swresample.c`
+    - `libswresample/swresample-test.c`
     - `tests/checkasm/*`
     - `tests/tiny_ssim.c`
 - the following filters in libavfilter:
-    - `signature_lookup.c`
     - `vf_blackframe.c`
     - `vf_boxblur.c`
     - `vf_colormatrix.c`
@@ -35,13 +34,13 @@ Specifically, the GPL parts of FFmpeg are:
     - `vf_eq.c`
     - `vf_find_rect.c`
     - `vf_fspp.c`
+    - `vf_geq.c`
     - `vf_histeq.c`
     - `vf_hqdn3d.c`
+    - `vf_interlace.c`
     - `vf_kerndeint.c`
-    - `vf_lensfun.c` (GPL version 3 or later)
     - `vf_mcdeint.c`
     - `vf_mpdecimate.c`
-    - `vf_nnedi.c`
     - `vf_owdenoise.c`
     - `vf_perspective.c`
     - `vf_phase.c`
@@ -50,14 +49,12 @@ Specifically, the GPL parts of FFmpeg are:
     - `vf_pullup.c`
     - `vf_repeatfields.c`
     - `vf_sab.c`
-    - `vf_signature.c`
     - `vf_smartblur.c`
     - `vf_spp.c`
     - `vf_stereo3d.c`
     - `vf_super2xsai.c`
     - `vf_tinterlace.c`
     - `vf_uspp.c`
-    - `vf_vaguedenoiser.c`
     - `vsrc_mptestsrc.c`
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
@@ -83,39 +80,24 @@ affect the licensing of binaries resulting from the combination.
 
 ### Compatible libraries
 
-The following libraries are under GPL version 2:
+The following libraries are under GPL:
-- avisynth
 - frei0r
 - libcdio
-- libdavs2
 - librubberband
 - libvidstab
 - libx264
 - libx265
 - libxavs
-- libxavs2
 - libxvid
 
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
 passing `--enable-gpl` to configure.
 
-The following libraries are under LGPL version 3:
+The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
-- gmp
+license is incompatible with the LGPL v2.1 and the GPL v2, but not with
-- libaribb24
-- liblensfun
-
-When combining them with FFmpeg, use the configure option `--enable-version3` to
-upgrade FFmpeg to the LGPL v3.
-
-The VMAF, mbedTLS, RK MPI, OpenCORE and VisualOn libraries are under the Apache License
-2.0. That license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 version 3 of those licenses. So to combine these libraries with FFmpeg, the
 license version needs to be upgraded by passing `--enable-version3` to configure.
 
-The smbclient library is under the GPL v3, to combine it with FFmpeg,
-the options `--enable-gpl` and `--enable-version3` have to be passed to
-configure to upgrade FFmpeg to the GPL v3.
-
 ### Incompatible libraries
 
 There are certain libraries you can combine with FFmpeg whose licenses are not

MAINTAINERS

@@ -11,11 +11,17 @@ A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
 patches and related discussions.
 
 
+Project Leader
+==============
+
+  final design decisions
+
+
 Applications
 ============
 
 ffmpeg:
-  ffmpeg.c                              Michael Niedermayer, Anton Khirnov
+  ffmpeg.c                              Michael Niedermayer
 
 ffplay:
   ffplay.c                              Marton Balint
@@ -34,8 +40,7 @@ Miscellaneous Areas
 ===================
 
 documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
-project server day to day operations    Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
-project server emergencies              Árpád Gereöffy, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@@ -48,11 +53,11 @@ Communication
 website                                 Deby Barbara Lepage
 fate.ffmpeg.org                         Timothy Gu
 Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
-Patchwork                               Andriy Gelman
 mailing lists                           Baptiste Coudurier
-Twitter                                 Reynaldo H. Verdejo Pinochet
+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, rcombs, wm4
+ffmpeg-security                         Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
 
 
 libavutil
@@ -73,7 +78,6 @@ Other:
   float_dsp                             Loren Merritt
   hash                                  Reimar Doeffinger
   hwcontext_cuda*                       Timo Rothenpieler
-  hwcontext_vulkan*                     Lynne
   intfloat*                             Michael Niedermayer
   integer.c, integer.h                  Michael Niedermayer
   lzo                                   Reimar Doeffinger
@@ -84,7 +88,6 @@ Other:
   rational.c, rational.h                Michael Niedermayer
   rc4                                   Reimar Doeffinger
   ripemd.c, ripemd.h                    James Almer
-  tx*                                   Lynne
 
 
 libavcodec
@@ -110,6 +113,8 @@ 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:
@@ -131,15 +136,13 @@ Codecs:
   8bps.c                                Roberto Togni
   8svx.c                                Jaikrishnan Menon
   aacenc*, aaccoder.c                   Rostislav Pehlivanov
-  adpcm.c                               Zane van Iperen
   alacenc.c                             Jaikrishnan Menon
   alsdec.c                              Thilo Borgmann, Umair Khan
-  amfenc*                               Dmitrii Ovchinnikov
   aptx.c                                Aurelien Jacobs
   ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3plus*                           Maxim Poliakovski
-  audiotoolbox*                         rcombs
+  audiotoolbox*                         Rodger Combs
   avs2*                                 Huiwen Ren
   bgmc.c, bgmc.h                        Thilo Borgmann
   binkaudio.c                           Peter Ross
@@ -151,10 +154,10 @@ Codecs:
   ccaption_dec.c                        Anshul Maheshwari, Aman Gupta
   cljr                                  Alex Beregszaszi
   cpia.c                                Stephan Hilb
+  crystalhd.c                           Philip Langdale
   cscd.c                                Reimar Doeffinger
   cuviddec.c                            Timo Rothenpieler
   dca*                                  foo86
-  dfpwm*                                Jack Bruienne
   dirac*                                Rostislav Pehlivanov
   dnxhd*                                Baptiste Coudurier
   dolby_e*                              foo86
@@ -187,17 +190,14 @@ Codecs:
   libcodec2.c                           Tomas Härdin
   libdirac*                             David Conrad
   libdavs2.c                            Huiwen Ren
-  libjxl*.c, libjxl.h                   Leo Izen
   libgsm.c                              Michel Bardiaux
   libkvazaar.c                          Arttu Ylä-Outinen
-  libopenh264enc.c                      Martin Storsjo, Linjie Fu
   libopenjpeg.c                         Jaikrishnan Menon
   libopenjpegenc.c                      Michael Bradshaw
   libtheoraenc.c                        David Conrad
   libvorbis.c                           David Conrad
   libvpx*                               James Zern
   libxavs.c                             Stefan Gehrer
-  libxavs2.c                            Huiwen Ren
   libzvbi-teletextdec.c                 Marton Balint
   lzo.h, lzo.c                          Reimar Doeffinger
   mdec.c                                Michael Niedermayer
@@ -213,7 +213,6 @@ Codecs:
   msvideo1.c                            Mike Melanson
   nuv.c                                 Reimar Doeffinger
   nvdec*, nvenc*                        Timo Rothenpieler
-  omx.c                                 Martin Storsjo, Aman Gupta
   opus*                                 Rostislav Pehlivanov
   paf.*                                 Paul B Mahol
   pcx.c                                 Ivo van Poorten
@@ -221,7 +220,7 @@ Codecs:
   ptx.c                                 Ivo van Poorten
   qcelp*                                Reynaldo H. Verdejo Pinochet
   qdm2.c, qdm2data.h                    Roberto Togni
-  qsv*                                  Mark Thompson, Zhong Li, Haihao Xiang
+  qsv*                                  Mark Thompson, Zhong Li
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
@@ -231,6 +230,7 @@ Codecs:
   rv10.c                                Michael Niedermayer
   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
@@ -261,14 +261,16 @@ Codecs:
   xan.c                                 Mike Melanson
   xbm*                                  Paul B Mahol
   xface                                 Stefano Sabatini
+  xvmc.c                                Ivan Kalvachev
   xwd*                                  Paul B Mahol
 
 Hardware acceleration:
+  crystalhd.c                           Philip Langdale
   dxva2*                                Hendrik Leppkes, Laurent Aimar, Steve Lhomme
   d3d11va*                              Steve Lhomme
   mediacodec*                           Matthieu Bouron, Aman Gupta
-  vaapi*                                Haihao Xiang
+  vaapi*                                Gwenole Beauchesne
-  vaapi_encode*                         Mark Thompson, Haihao Xiang
+  vaapi_encode*                         Mark Thompson
   vdpau*                                Philip Langdale, Carl Eugen Hoyos
   videotoolbox*                         Rick Kern, Aman Gupta
 
@@ -347,7 +349,6 @@ Filters:
   vf_il.c                               Paul B Mahol
   vf_(t)interlace                       Thomas Mundt (CC <thomas.mundt@hr.de>)
   vf_lenscorrection.c                   Daniel Oberhoff
-  vf_libplacebo.c                       Niklas Haas
   vf_mergeplanes.c                      Paul B Mahol
   vf_mestimate.c                        Davinder Singh
   vf_minterpolate.c                     Davinder Singh
@@ -367,8 +368,6 @@ Filters:
 Sources:
   vsrc_mandelbrot.c                     Michael Niedermayer
 
-dnn                                     Yejun Guo
-
 libavformat
 ===========
 
@@ -387,13 +386,7 @@ Muxers/Demuxers:
   afc.c                                 Paul B Mahol
   aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
   aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
-  alp.c                                 Zane van Iperen
-  amvenc.c                              Zane van Iperen
-  apm.c                                 Zane van Iperen
   apngdec.c                             Benoit Fouet
-  argo_asf.c                            Zane van Iperen
-  argo_brp.c                            Zane van Iperen
-  argo_cvg.c                            Zane van Iperen
   ass*                                  Aurelien Jacobs
   astdec.c                              Paul B Mahol
   astenc.c                              James Almer
@@ -410,7 +403,6 @@ Muxers/Demuxers:
   dashdec.c                             Steven Liu
   dashenc.c                             Karthick Jeyapal
   daud.c                                Reimar Doeffinger
-  dfpwmdec.c                            Jack Bruienne
   dss.c                                 Oleksij Rempel
   dtsdec.c                              foo86
   dtshddec.c                            Paul B Mahol
@@ -427,22 +419,19 @@ Muxers/Demuxers:
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
-  imf*                                  Pierre-Anthony Lemieux
   img2*.c                               Michael Niedermayer
   ipmovie.c                             Mike Melanson
   ircam*                                Paul B Mahol
   iss.c                                 Stefan Gehrer
-  jpegxl_probe.*                        Leo Izen
   jvdec.c                               Peter Ross
-  kvag.c                                Zane van Iperen
   libmodplug.c                          Clément Bœsch
   libopenmpt.c                          Josh de Kock
   lmlm4.c                               Ivo van Poorten
   lvfdec.c                              Paul B Mahol
   lxfdec.c                              Tomas Härdin
-  matroska.c                            Aurelien Jacobs, Andreas Rheinhardt
+  matroska.c                            Aurelien Jacobs
-  matroskadec.c                         Aurelien Jacobs, Andreas Rheinhardt
+  matroskadec.c                         Aurelien Jacobs
-  matroskaenc.c                         David Conrad, Andreas Rheinhardt
+  matroskaenc.c                         David Conrad
   matroska subtitles (matroskaenc.c)    John Peebles
   metadata*                             Aurelien Jacobs
   mgsts.c                               Paul B Mahol
@@ -457,7 +446,7 @@ Muxers/Demuxers:
   mpegtsenc.c                           Baptiste Coudurier
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
-  mxf*                                  Baptiste Coudurier, Tomas Härdin
+  mxf*                                  Baptiste Coudurier
   nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
   nut*                                  Michael Niedermayer
@@ -465,9 +454,9 @@ Muxers/Demuxers:
   oggdec.c, oggdec.h                    David Conrad
   oggenc.c                              Baptiste Coudurier
   oggparse*.c                           David Conrad
+  oggparsedaala*                        Rostislav Pehlivanov
   oma.c                                 Maxim Poliakovski
   paf.c                                 Paul B Mahol
-  pp_bnk.c                              Zane van Iperen
   psxstr.c                              Mike Melanson
   pva.c                                 Ivo van Poorten
   pvfdec.c                              Paul B Mahol
@@ -512,9 +501,7 @@ Protocols:
   bluray.c                              Petri Hintukainen
   ftp.c                                 Lukasz Marek
   http.c                                Ronald S. Bultje
-  libsrt.c                              Zhao Zhili
   libssh.c                              Lukasz Marek
-  libzmq.c                              Andriy Gelman
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
   icecast.c                             Marvin Scholz
@@ -539,7 +526,6 @@ Operating systems / CPU architectures
 
 Alpha                                   Falk Hueffner
 MIPS                                    Manojkumar Bhosale, Shiyou Yin
-LoongArch                               Shiyou Yin
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Lauri Kasanen
@@ -571,7 +557,6 @@ Joakim Plate
 Jun Zhao
 Kieran Kunhya
 Kirill Gavrilov
-Limin Wang
 Martin Storsjö
 Panagiotis Issaris
 Pedro Arthur
@@ -610,23 +595,17 @@ Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 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
-Haihao Xiang (haihao)         1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
 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
-Leo Izen (thebombzen)         B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
-Lynne                         FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
+Lou Logan (llogan)            7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-                              DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
-Niklas Haas (haasn)           1DDB 8076 B14D 5B48 32FC 99D9 EB52 DA9C 02BA 6FB4
 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
-Pierre-Anthony Lemieux (pal)  F4B3 9492 E6F2 E4AF AEC8 46CB 698F A1F0 F8D4 EED4
-Ramiro Polla                  7859 C65B 751B 1179 792E DAE8 8E95 8B2F 9B6C 5700
 Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
@@ -635,9 +614,7 @@ 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
-Thilo Borgmann (thilo)        CE1D B7F4 4D20 FC3A DD9F FE5A 257C 5B8F 1D20 B92F
 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
 Tomas Härdin (thardin)        A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
-Zane van Iperen (zane)        61AE D40F 368B 6F26 9DAE 3892 6861 6B2D 8AC4 DCC5

Makefile

@@ -13,19 +13,17 @@ vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %.cu   $(SRC_PATH)
 vpath %.ptx  $(SRC_PATH)
-vpath %.metal $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
 
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
-
 # $(FFLIBS-yes) needs to be in linking order
 FFLIBS-$(CONFIG_AVDEVICE)   += avdevice
 FFLIBS-$(CONFIG_AVFILTER)   += avfilter
 FFLIBS-$(CONFIG_AVFORMAT)   += avformat
 FFLIBS-$(CONFIG_AVCODEC)    += avcodec
+FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_POSTPROC)   += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
 FFLIBS-$(CONFIG_SWSCALE)    += swscale
@@ -52,49 +50,32 @@ $(TOOLS): %$(EXESUF): %.o
 target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
 
-tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_fuzzer.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
-
-target_dem_%_fuzzer$(EXESUF): target_dem_%_fuzzer.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
-
 tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
 
-tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
-
-
-tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
-tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
-tools/scale_slice_test$(EXESUF): $(FF_DEP_LIBS)
-tools/scale_slice_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
 tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
-tools/target_dem_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
 
 CONFIGURABLE_COMPONENTS =                                           \
     $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
     $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
-    $(SRC_PATH)/libavcodec/hwaccels.h                               \
     $(SRC_PATH)/libavcodec/parsers.c                                \
     $(SRC_PATH)/libavformat/protocols.c                             \
 
-config_components.h: ffbuild/.config
+config.h: ffbuild/.config
 ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
 	@-tput bold 2>/dev/null
-	@-printf '\nWARNING: $(?) newer than config_components.h, rerun configure\n\n'
+	@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
 
 SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
                HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
                ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
-               ALTIVEC-OBJS VSX-OBJS RVV-OBJS MMX-OBJS X86ASM-OBJS       \
+               ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS                \
                MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS         \
-               MMI-OBJS LSX-OBJS LASX-OBJS OBJS SLIBOBJS SHLIBOBJS       \
+               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
-               STLIBOBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -116,13 +97,12 @@ include $(SRC_PATH)/fftools/Makefile
 include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/doc/examples/Makefile
 
-$(ALLFFLIBS:%=lib%/version.o): libavutil/ffversion.h
+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
-	$(RM) $@
 	$(CP) $< $@
 	$(STRIP) $@
 endif
@@ -163,12 +143,11 @@ clean::
 	$(RM) -rf coverage.info coverage.info.in lcov
 
 distclean:: clean
-	$(RM) .version config.asm config.h config_components.h mapfile  \
+	$(RM) .version avversion.h config.asm config.h mapfile  \
 		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
 		version.h libavutil/ffversion.h libavcodec/codec_names.h \
 		libavcodec/bsf_list.c libavformat/protocol_list.c \
 		libavcodec/codec_list.c libavcodec/parser_list.c \
-		libavfilter/filter_list.c libavdevice/indev_list.c libavdevice/outdev_list.c \
 		libavformat/muxer_list.c libavformat/demuxer_list.c
 ifeq ($(SRC_LINK),src)
 	$(RM) src

README.md

@@ -9,7 +9,7 @@ such as audio, video, subtitles and related metadata.
 * `libavcodec` provides implementation of a wider range of codecs.
 * `libavformat` implements streaming protocols, container formats and basic I/O access.
 * `libavutil` includes hashers, decompressors and miscellaneous utility functions.
-* `libavfilter` provides means to alter decoded audio and video through a directed graph of connected filters.
+* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
 * `libavdevice` provides an abstraction to access capture and playback devices.
 * `libswresample` implements audio mixing and resampling routines.
 * `libswscale` implements color conversion and scaling routines.

RELEASE

@@ -1 +1 @@
-6.0.1
+4.2.2

RELEASE_NOTES

@@ -1,15 +1,15 @@
 
-            ┌────────────────────────────────────────────┐
+              ┌────────────────────────────────────┐
-            │ RELEASE NOTES for FFmpeg 6.0 "Von Neumann" │
+              │ RELEASE NOTES for FFmpeg 4.2 "Ada" │
-            └────────────────────────────────────────────┘
+              └────────────────────────────────────┘
 
-   The FFmpeg Project proudly presents FFmpeg 6.0 "Von Neumann", about 6
+   The FFmpeg Project proudly presents FFmpeg 4.2 "Ada", about 8
-   months after the release of FFmpeg 5.1.
+   months after the release of FFmpeg 4.1.
 
    A complete Changelog is available at the root of the project, and the
    complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
 
    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,
-   feel free to join us on the #ffmpeg IRC channel (on irc.libera.chat) or ask
+   feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
    on the mailing-lists.

compat/atomics/win32/stdatomic.h

@@ -96,7 +96,7 @@ do {                                    \
     atomic_load(object)
 
 #define atomic_exchange(object, desired) \
-    InterlockedExchangePointer((PVOID volatile *)object, (PVOID)desired)
+    InterlockedExchangePointer(object, desired);
 
 #define atomic_exchange_explicit(object, desired, order) \
     atomic_exchange(object, desired)

compat/avisynth/avs/capi.h

@@ -0,0 +1,94 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CAPI_H
+#define AVS_CAPI_H
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#ifdef BUILDING_AVSCORE
+#  if defined(GCC) && defined(X86_32)
+#    define AVSC_CC
+#  else // MSVC builds and 64-bit GCC
+#    ifndef AVSC_USE_STDCALL
+#      define AVSC_CC __cdecl
+#    else
+#      define AVSC_CC __stdcall
+#    endif
+#  endif
+#else // needed for programs that talk to AviSynth+
+#  ifndef AVSC_WIN32_GCC32 // see comment below
+#    ifndef AVSC_USE_STDCALL
+#      define AVSC_CC __cdecl
+#    else
+#      define AVSC_CC __stdcall
+#    endif
+#  else
+#    define AVSC_CC
+#  endif
+#endif
+
+// On 64-bit Windows, there's only one calling convention,
+// so there is no difference between MSVC and GCC. On 32-bit,
+// this isn't true. The convention that GCC needs to use to
+// even build AviSynth+ as 32-bit makes anything that uses
+// it incompatible with 32-bit MSVC builds of AviSynth+.
+// The AVSC_WIN32_GCC32 define is meant to provide a user
+// switchable way to make builds of FFmpeg to test 32-bit
+// GCC builds of AviSynth+ without having to screw around
+// with alternate headers, while still default to the usual
+// situation of using 32-bit MSVC builds of AviSynth+.
+
+// Hopefully, this situation will eventually be resolved
+// and a broadly compatible solution will arise so the
+// same 32-bit FFmpeg build can handle either MSVC or GCC
+// builds of AviSynth+.
+
+#define AVSC_INLINE static __inline
+
+#ifdef BUILDING_AVSCORE
+#  define AVSC_EXPORT __declspec(dllexport)
+#  define AVSC_API(ret, name) EXTERN_C AVSC_EXPORT ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+#endif //AVS_CAPI_H

compat/avisynth/avs/config.h

@@ -0,0 +1,70 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CONFIG_H
+#define AVS_CONFIG_H
+
+// Undefine this to get cdecl calling convention
+#define AVSC_USE_STDCALL 1
+
+// NOTE TO PLUGIN AUTHORS:
+// Because FRAME_ALIGN can be substantially higher than the alignment
+// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
+// alignment. They should always request the exact alignment value they need.
+// This is to make sure that plugins work over the widest range of AviSynth
+// builds possible.
+#define FRAME_ALIGN 64
+
+#if   defined(_M_AMD64) || defined(__x86_64)
+#   define X86_64
+#elif defined(_M_IX86) || defined(__i386__)
+#   define X86_32
+#else
+#   error Unsupported CPU architecture.
+#endif
+
+#if   defined(_MSC_VER)
+#   define MSVC
+#elif defined(__GNUC__)
+#   define GCC
+#elif defined(__clang__)
+#   define CLANG
+#else
+#   error Unsupported compiler.
+#endif
+
+#if   defined(GCC)
+#   undef __forceinline
+#   define __forceinline inline
+#endif
+
+#endif //AVS_CONFIG_H

compat/avisynth/avs/types.h

@@ -0,0 +1,57 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_TYPES_H
+#define AVS_TYPES_H
+
+// Define all types necessary for interfacing with avisynth.dll
+
+#ifdef __cplusplus
+  #include <cstddef>
+#else
+  #include <stddef.h>
+#endif
+
+// Raster types used by VirtualDub & Avisynth
+typedef unsigned int    Pixel32;
+typedef unsigned char   BYTE;
+
+// Audio Sample information
+typedef float SFLOAT;
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+#endif //AVS_TYPES_H

compat/avisynth/avxsynth_c.h

@@ -0,0 +1,728 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef __AVXSYNTH_C__
+#define __AVXSYNTH_C__
+
+#include "windowsPorts/windows2linux.h"
+#include <stdarg.h>
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#define AVSC_USE_STDCALL 1
+
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
+#endif
+
+#define AVSC_INLINE static __inline
+
+#ifdef AVISYNTH_C_EXPORTS
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// Constants
+//
+
+#ifndef __AVXSYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 3 };
+#endif
+
+enum {AVS_SAMPLE_INT8  = 1<<0,
+      AVS_SAMPLE_INT16 = 1<<1,
+      AVS_SAMPLE_INT24 = 1<<2,
+      AVS_SAMPLE_INT32 = 1<<3,
+      AVS_SAMPLE_FLOAT = 1<<4};
+
+enum {AVS_PLANAR_Y=1<<0,
+      AVS_PLANAR_U=1<<1,
+      AVS_PLANAR_V=1<<2,
+      AVS_PLANAR_ALIGNED=1<<3,
+      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
+
+  // Colorspace properties.
+enum {AVS_CS_BGR = 1<<28,
+      AVS_CS_YUV = 1<<29,
+      AVS_CS_INTERLEAVED = 1<<30,
+      AVS_CS_PLANAR = 1<<31};
+
+  // Specific colorformats
+enum {
+  AVS_CS_UNKNOWN = 0,
+  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
+  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
+  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
+  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
+};
+
+enum {
+  AVS_IT_BFF = 1<<0,
+  AVS_IT_TFF = 1<<1,
+  AVS_IT_FIELDBASED = 1<<2};
+
+enum {
+  AVS_FILTER_TYPE=1,
+  AVS_FILTER_INPUT_COLORSPACE=2,
+  AVS_FILTER_OUTPUT_TYPE=9,
+  AVS_FILTER_NAME=4,
+  AVS_FILTER_AUTHOR=5,
+  AVS_FILTER_VERSION=6,
+  AVS_FILTER_ARGS=7,
+  AVS_FILTER_ARGS_INFO=8,
+  AVS_FILTER_ARGS_DESCRIPTION=10,
+  AVS_FILTER_DESCRIPTION=11};
+
+enum {  //SUBTYPES
+  AVS_FILTER_TYPE_AUDIO=1,
+  AVS_FILTER_TYPE_VIDEO=2,
+  AVS_FILTER_OUTPUT_TYPE_SAME=3,
+  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
+
+enum {
+  AVS_CACHE_NOTHING=0,
+  AVS_CACHE_RANGE=1,
+  AVS_CACHE_ALL=2,
+  AVS_CACHE_AUDIO=3,
+  AVS_CACHE_AUDIO_NONE=4,
+  AVS_CACHE_AUDIO_AUTO=5
+};
+
+#define AVS_FRAME_ALIGN 16
+
+typedef struct AVS_Clip AVS_Clip;
+typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoInfo
+//
+
+// AVS_VideoInfo is layed out identicly to VideoInfo
+typedef struct AVS_VideoInfo {
+  int width, height;    // width=0 means no video
+  unsigned fps_numerator, fps_denominator;
+  int num_frames;
+
+  int pixel_type;
+
+  int audio_samples_per_second;   // 0 means no audio
+  int sample_type;
+  INT64 num_audio_samples;
+  int nchannels;
+
+  // Imagetype properties
+
+  int image_type;
+} AVS_VideoInfo;
+
+// useful functions of the above
+AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
+        { return (p->width!=0); }
+
+AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
+        { return (p->audio_samples_per_second!=0); }
+
+AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_BGR); }
+
+AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
+        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
+
+AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
+
+AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_YUV ); }
+
+AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
+
+AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
+        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
+
+AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
+        { return ((p->pixel_type & c_space) == c_space); }
+
+AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
+        { return ((p->pixel_type & property)==property ); }
+
+AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type & AVS_CS_PLANAR); }
+
+AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_FIELDBASED); }
+
+AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
+        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
+
+AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_BFF); }
+
+AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_TFF); }
+
+AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
+{
+  switch (p->pixel_type) {
+      case AVS_CS_BGR24: return 24;
+      case AVS_CS_BGR32: return 32;
+      case AVS_CS_YUY2:  return 16;
+      case AVS_CS_YV12:
+      case AVS_CS_I420:  return 12;
+      default:           return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
+        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
+
+AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
+        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
+
+AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
+        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
+
+AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
+        { return p->audio_samples_per_second; }
+
+
+AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
+{
+    switch (p->sample_type) {
+      case AVS_SAMPLE_INT8:  return sizeof(signed char);
+      case AVS_SAMPLE_INT16: return sizeof(signed short);
+      case AVS_SAMPLE_INT24: return 3;
+      case AVS_SAMPLE_INT32: return sizeof(signed int);
+      case AVS_SAMPLE_FLOAT: return sizeof(float);
+      default: return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
+        { return p->nchannels*avs_bytes_per_channel_sample(p);}
+
+AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
+        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
+
+AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
+
+AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
+        { return bytes / avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return samples * avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
+        { return p->nchannels; }
+
+AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
+        { return p->sample_type;}
+
+// useful mutator
+AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
+        { p->image_type|=property; }
+
+AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
+        { p->image_type&=~property; }
+
+AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
+        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
+
+AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
+{
+    unsigned x=numerator, y=denominator;
+    while (y) {   // find gcd
+      unsigned t = x%y; x = y; y = t;
+    }
+    p->fps_numerator = numerator/x;
+    p->fps_denominator = denominator/x;
+}
+
+AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
+{
+        return (x->pixel_type == y->pixel_type)
+                || (avs_is_yv12(x) && avs_is_yv12(y));
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoFrame
+//
+
+// VideoFrameBuffer holds information about a memory block which is used
+// for video data.  For efficiency, instances of this class are not deleted
+// when the refcount reaches zero; instead they're stored in a linked list
+// to be reused.  The instances are deleted when the corresponding AVS
+// file is closed.
+
+// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrameBuffer {
+  unsigned char * data;
+  int data_size;
+  // sequence_number is incremented every time the buffer is changed, so
+  // that stale views can tell they're no longer valid.
+  long sequence_number;
+
+  long refcount;
+} AVS_VideoFrameBuffer;
+
+// VideoFrame holds a "window" into a VideoFrameBuffer.
+
+// AVS_VideoFrame is layed out identicly to IVideoFrame
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrame {
+  int refcount;
+  AVS_VideoFrameBuffer * vfb;
+  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
+} AVS_VideoFrame;
+
+// Access functions for AVS_VideoFrame
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return p->pitch;}
+
+AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
+  switch (plane) {
+  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
+  return p->pitch;}
+
+AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
+        return p->row_size; }
+
+AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
+        int r;
+    switch (plane) {
+    case AVS_PLANAR_U: case AVS_PLANAR_V:
+                if (p->pitchUV) return p->row_size>>1;
+                else            return 0;
+    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
+                if (p->pitchUV) {
+                        r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
+                        if (r < p->pitchUV)
+                                return r;
+                        return p->row_size>>1;
+                } else return 0;
+    case AVS_PLANAR_Y_ALIGNED:
+                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
+                if (r <= p->pitch)
+                        return r;
+                return p->row_size;
+    }
+    return p->row_size;
+}
+
+AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
+        return p->height;}
+
+AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
+        switch (plane) {
+                case AVS_PLANAR_U: case AVS_PLANAR_V:
+                        if (p->pitchUV) return p->height>>1;
+                        return 0;
+        }
+        return p->height;}
+
+AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
+        return p->vfb->data + p->offset;}
+
+AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        switch (plane) {
+                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                default:           return p->vfb->data + p->offset;}
+}
+
+AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
+        return (p->refcount == 1 && p->vfb->refcount == 1);}
+
+AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
+{
+        if (avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else
+                return 0;
+}
+
+AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
+{
+        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
+                ++p->vfb->sequence_number;
+                return p->vfb->data + p->offset;
+        } else if (plane==AVS_PLANAR_Y) {
+                return 0;
+        } else {
+                switch (plane) {
+                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
+                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
+                        default:       return p->vfb->data + p->offset;
+                }
+        }
+}
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
+// makes a shallow copy of a video frame
+AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
+  {avs_release_video_frame(f);}
+AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
+  {return avs_copy_video_frame(f);}
+#endif
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Value
+//
+
+// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
+// and avs_release_value appropiaty as you would if AVS_Value was
+// a pointer.
+
+// To maintain source code compatibility with future versions of the
+// avisynth_c API don't use the AVS_Value directly.  Use the helper
+// functions below.
+
+// AVS_Value is layed out identicly to AVSValue
+typedef struct AVS_Value AVS_Value;
+struct AVS_Value {
+  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
+               // for some function e'rror
+  short array_size;
+  union {
+    void * clip; // do not use directly, use avs_take_clip
+    char boolean;
+    int integer;
+    INT64 integer64; // match addition of __int64 to avxplugin.h
+    float floating_pt;
+    const char * string;
+    const AVS_Value * array;
+  } d;
+};
+
+// AVS_Value should be initilized with avs_void.
+// Should also set to avs_void after the value is released
+// with avs_copy_value.  Consider it the equalvent of setting
+// a pointer to NULL
+static const AVS_Value avs_void = {'v'};
+
+AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
+AVSC_API(void, avs_release_value)(AVS_Value);
+
+AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
+AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
+AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
+AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
+AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
+AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
+AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
+AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
+AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+AVSC_INLINE int avs_as_bool(AVS_Value v)
+        { return v.d.boolean; }
+AVSC_INLINE int avs_as_int(AVS_Value v)
+        { return v.d.integer; }
+AVSC_INLINE const char * avs_as_string(AVS_Value v)
+        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
+AVSC_INLINE double avs_as_float(AVS_Value v)
+        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
+AVSC_INLINE const char * avs_as_error(AVS_Value v)
+        { return avs_is_error(v) ? v.d.string : 0; }
+AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
+        { return v.d.array; }
+AVSC_INLINE int avs_array_size(AVS_Value v)
+        { return avs_is_array(v) ? v.array_size : 1; }
+AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
+        { return avs_is_array(v) ? v.d.array[index] : v; }
+
+// only use these functions on am AVS_Value that does not already have
+// an active value.  Remember, treat AVS_Value as a fat pointer.
+AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
+        { AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+AVSC_INLINE AVS_Value avs_new_value_int(int v0)
+        { AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
+        { AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
+AVSC_INLINE AVS_Value avs_new_value_float(float v0)
+        { AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
+AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
+        { AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
+        { AVS_Value v = {0}; 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 = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Clip
+//
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_clip)(AVS_Clip *);
+AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
+
+AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
+
+AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
+
+AVSC_API(int, avs_get_version)(AVS_Clip *);
+
+AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
+// The returned video frame must be released with avs_release_video_frame
+
+AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
+// return field parity if field_based, else parity of first field in frame
+
+AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
+                                  INT64 start, INT64 count);
+// start and count are in samples
+
+AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
+                                        int cachehints, size_t frame_range);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// This is the callback type used by avs_add_function
+typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
+                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
+
+typedef struct AVS_FilterInfo AVS_FilterInfo;
+struct AVS_FilterInfo
+{
+  // these members should not be modified outside of the AVS_ApplyFunc callback
+  AVS_Clip * child;
+  AVS_VideoInfo vi;
+  AVS_ScriptEnvironment * env;
+  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
+                                  INT64 start, INT64 count);
+  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
+                                        int frame_range);
+  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
+
+  // Should be set when ever there is an error to report.
+  // It is cleared before any of the above methods are called
+  const char * error;
+  // this is to store whatever and may be modified at will
+  void * user_data;
+};
+
+// Create a new filter
+// fi is set to point to the AVS_FilterInfo so that you can
+//   modify it once it is initilized.
+// store_child should generally be set to true.  If it is not
+//    set than ALL methods (the function pointers) must be defined
+// If it is set than you do not need to worry about freeing the child
+//    clip.
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
+                                      AVS_FilterInfo * * fi,
+                                      AVS_Value child, int store_child);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_ScriptEnvironment
+//
+
+// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
+enum {
+                                /* slowest CPU to support extension */
+  AVS_CPU_FORCE        = 0x01,   // N/A
+  AVS_CPU_FPU          = 0x02,   // 386/486DX
+  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
+  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
+  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
+  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
+  AVS_CPU_3DNOW        = 0x40,   // K6-2
+  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
+  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
+                                 // which only Hammer will have anyway)
+};
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
+
+AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
+
+AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
+AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
+
+AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
+ // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
+
+AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
+                                     const char * name, const char * params,
+                                     AVS_ApplyFunc apply, void * user_data);
+
+AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
+
+AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
+                               AVS_Value args, const char** arg_names);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
+
+AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
+
+//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
+//void avs_pop_context(AVS_ScriptEnvironment *);
+
+AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
+                                          const AVS_VideoInfo * vi, int align);
+// align should be at least 16
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE
+AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
+                                     const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+
+AVSC_INLINE
+AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
+                               const AVS_VideoInfo * vi)
+  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+#endif
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
+
+AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
+
+typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
+AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
+
+AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
+// The returned video frame must be be released
+
+AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
+
+AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
+
+// avisynth.dll exports this; it's a way to use it as a library, without
+// writing an AVS script or without going through AVIFile.
+AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// this symbol is the entry point for the plugin and must
+// be defined
+AVSC_EXPORT
+const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
+
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
+
+
+AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
+// The returned video frame must be be released
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#endif //__AVXSYNTH_C__

compat/avisynth/windowsPorts/basicDataTypeConversions.h

@@ -0,0 +1,85 @@
+#ifndef __DATA_TYPE_CONVERSIONS_H__
+#define __DATA_TYPE_CONVERSIONS_H__
+
+#include <stdint.h>
+#include <wchar.h>
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+
+typedef int64_t __int64;
+typedef int32_t __int32;
+#ifdef __cplusplus
+typedef bool BOOL;
+#else
+typedef uint32_t BOOL;
+#endif // __cplusplus
+typedef void* HMODULE;
+typedef void* LPVOID;
+typedef void* PVOID;
+typedef PVOID HANDLE;
+typedef HANDLE HWND;
+typedef HANDLE HINSTANCE;
+typedef void* HDC;
+typedef void* HBITMAP;
+typedef void* HICON;
+typedef void* HFONT;
+typedef void* HGDIOBJ;
+typedef void* HBRUSH;
+typedef void* HMMIO;
+typedef void* HACMSTREAM;
+typedef void* HACMDRIVER;
+typedef void* HIC;
+typedef void* HACMOBJ;
+typedef HACMSTREAM* LPHACMSTREAM;
+typedef void* HACMDRIVERID;
+typedef void* LPHACMDRIVER;
+typedef unsigned char BYTE;
+typedef BYTE* LPBYTE;
+typedef char TCHAR;
+typedef TCHAR* LPTSTR;
+typedef const TCHAR* LPCTSTR;
+typedef char* LPSTR;
+typedef LPSTR LPOLESTR;
+typedef const char* LPCSTR;
+typedef LPCSTR LPCOLESTR;
+typedef wchar_t WCHAR;
+typedef unsigned short WORD;
+typedef unsigned int UINT;
+typedef UINT MMRESULT;
+typedef uint32_t DWORD;
+typedef DWORD COLORREF;
+typedef DWORD FOURCC;
+typedef DWORD HRESULT;
+typedef DWORD* LPDWORD;
+typedef DWORD* DWORD_PTR;
+typedef int32_t LONG;
+typedef int32_t* LONG_PTR;
+typedef LONG_PTR LRESULT;
+typedef uint32_t ULONG;
+typedef uint32_t* ULONG_PTR;
+//typedef __int64_t intptr_t;
+typedef uint64_t _fsize_t;
+
+
+//
+// Structures
+//
+
+typedef struct _GUID {
+  DWORD Data1;
+  WORD  Data2;
+  WORD  Data3;
+  BYTE  Data4[8];
+} GUID;
+
+typedef GUID REFIID;
+typedef GUID CLSID;
+typedef CLSID* LPCLSID;
+typedef GUID IID;
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+#endif //  __DATA_TYPE_CONVERSIONS_H__

compat/avisynth/windowsPorts/windows2linux.h

@@ -0,0 +1,77 @@
+#ifndef __WINDOWS2LINUX_H__
+#define __WINDOWS2LINUX_H__
+
+/*
+ * LINUX SPECIFIC DEFINITIONS
+*/
+//
+// Data types conversions
+//
+#include <stdlib.h>
+#include <string.h>
+#include "basicDataTypeConversions.h"
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+//
+// purposefully define the following MSFT definitions
+// to mean nothing (as they do not mean anything on Linux)
+//
+#define __stdcall
+#define __cdecl
+#define noreturn
+#define __declspec(x)
+#define STDAPI       extern "C" HRESULT
+#define STDMETHODIMP HRESULT __stdcall
+#define STDMETHODIMP_(x) x __stdcall
+
+#define STDMETHOD(x)    virtual HRESULT x
+#define STDMETHOD_(a, x) virtual a x
+
+#ifndef TRUE
+#define TRUE  true
+#endif
+
+#ifndef FALSE
+#define FALSE false
+#endif
+
+#define S_OK                (0x00000000)
+#define S_FALSE             (0x00000001)
+#define E_NOINTERFACE       (0X80004002)
+#define E_POINTER           (0x80004003)
+#define E_FAIL              (0x80004005)
+#define E_OUTOFMEMORY       (0x8007000E)
+
+#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
+#define FAILED(hr)              ((hr) & 0x80000000)
+#define SUCCEEDED(hr)           (!FAILED(hr))
+
+
+//
+// Functions
+//
+#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
+#define MAKEWORD(a,b) (((a) << 8) | (b))
+
+#define lstrlen                             strlen
+#define lstrcpy                             strcpy
+#define lstrcmpi                            strcasecmp
+#define _stricmp                            strcasecmp
+#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
+#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
+// Windows uses (new, old) ordering but GCC has (old, new)
+#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
+
+#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
+#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
+#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
+
+#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+
+#endif //  __WINDOWS2LINUX_H__

compat/cuda/cuda_runtime.h

@@ -1,7 +1,7 @@
 /*
  * Minimum CUDA compatibility definitions header
  *
- * Copyright (c) 2019 rcombs
+ * Copyright (c) 2019 Rodger Combs
  *
  * This file is part of FFmpeg.
  *
@@ -49,16 +49,6 @@ typedef struct __device_builtin__ __align__(4) ushort2
     unsigned short x, y;
 } ushort2;
 
-typedef struct __device_builtin__ __align__(8) float2
-{
-    float x, y;
-} float2;
-
-typedef struct __device_builtin__ __align__(8) int2
-{
-    int x, y;
-} int2;
-
 typedef struct __device_builtin__ uint3
 {
     unsigned int x, y, z;
@@ -66,6 +56,11 @@ typedef struct __device_builtin__ uint3
 
 typedef struct uint3 dim3;
 
+typedef struct __device_builtin__ __align__(8) int2
+{
+    int x, y;
+} int2;
+
 typedef struct __device_builtin__ __align__(4) uchar4
 {
     unsigned char x, y, z, w;
@@ -73,7 +68,7 @@ typedef struct __device_builtin__ __align__(4) uchar4
 
 typedef struct __device_builtin__ __align__(8) ushort4
 {
-    unsigned short x, y, z, w;
+    unsigned char x, y, z, w;
 } ushort4;
 
 typedef struct __device_builtin__ __align__(16) int4
@@ -81,11 +76,6 @@ typedef struct __device_builtin__ __align__(16) int4
     int x, y, z, w;
 } int4;
 
-typedef struct __device_builtin__ __align__(16) float4
-{
-    float x, y, z, w;
-} float4;
-
 // Accessors for special registers
 #define GETCOMP(reg, comp) \
     asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
@@ -110,31 +100,24 @@ GET(getThreadIdx, tid)
 #define threadIdx (getThreadIdx())
 
 // Basic initializers (simple macros rather than inline functions)
-#define make_int2(a, b) ((int2){.x = a, .y = b})
 #define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
 #define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
-#define make_float2(a, b) ((float2){.x = a, .y = b})
-#define make_int4(a, b, c, d) ((int4){.x = a, .y = b, .z = c, .w = d})
 #define make_uchar4(a, b, c, d) ((uchar4){.x = a, .y = b, .z = c, .w = d})
 #define make_ushort4(a, b, c, d) ((ushort4){.x = a, .y = b, .z = c, .w = d})
-#define make_float4(a, b, c, d) ((float4){.x = a, .y = b, .z = c, .w = d})
 
 // Conversions from the tex instruction's 4-register output to various types
 #define TEX2D(type, ret) static inline __device__ void conv(type* out, unsigned a, unsigned b, unsigned c, unsigned d) {*out = (ret);}
 
 TEX2D(unsigned char, a & 0xFF)
 TEX2D(unsigned short, a & 0xFFFF)
-TEX2D(float, a)
 TEX2D(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
 TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
-TEX2D(float2, make_float2(a, b))
 TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
 TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
-TEX2D(float4, make_float4(a, b, c, d))
 
 // Template calling tex instruction and converting the output to the selected type
-template<typename T>
+template <class T>
-inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
+static inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
 {
   T ret;
   unsigned ret1, ret2, ret3, ret4;
@@ -145,48 +128,4 @@ inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
   return ret;
 }
 
-template<>
-inline __device__ float4 tex2D<float4>(cudaTextureObject_t texObject, float x, float y)
-{
-    float4 ret;
-    asm("tex.2d.v4.f32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
-        "=r"(ret.x), "=r"(ret.y), "=r"(ret.z), "=r"(ret.w) :
-        "l"(texObject), "f"(x), "f"(y));
-    return ret;
-}
-
-template<>
-inline __device__ float tex2D<float>(cudaTextureObject_t texObject, float x, float y)
-{
-    return tex2D<float4>(texObject, x, y).x;
-}
-
-template<>
-inline __device__ float2 tex2D<float2>(cudaTextureObject_t texObject, float x, float y)
-{
-    float4 ret = tex2D<float4>(texObject, x, y);
-    return make_float2(ret.x, ret.y);
-}
-
-// Math helper functions
-static inline __device__ float floorf(float a) { return __builtin_floorf(a); }
-static inline __device__ float floor(float a) { return __builtin_floorf(a); }
-static inline __device__ double floor(double a) { return __builtin_floor(a); }
-static inline __device__ float ceilf(float a) { return __builtin_ceilf(a); }
-static inline __device__ float ceil(float a) { return __builtin_ceilf(a); }
-static inline __device__ double ceil(double a) { return __builtin_ceil(a); }
-static inline __device__ float truncf(float a) { return __builtin_truncf(a); }
-static inline __device__ float trunc(float a) { return __builtin_truncf(a); }
-static inline __device__ double trunc(double a) { return __builtin_trunc(a); }
-static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
-static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
-static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
-static inline __device__ float sqrtf(float a) { return __builtin_sqrtf(a); }
-
-static inline __device__ float __saturatef(float a) { return __nvvm_saturate_f(a); }
-static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }
-static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a); }
-static inline __device__ float __expf(float a) { return __nvvm_ex2_approx_f(a * (float)__builtin_log2(__builtin_exp(1))); }
-static inline __device__ float __powf(float a, float b) { return __nvvm_ex2_approx_f(__nvvm_lg2_approx_f(a) * b); }
-
 #endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

compat/cuda/ptx2c.sh

@@ -0,0 +1,36 @@
+#!/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" > "$OUT"
+while IFS= read -r LINE
+do
+    printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
+done < "$IN"
+printf ";\n" >> "$OUT"
+
+exit 0

compat/msvcrt/snprintf.c

@@ -59,7 +59,7 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt,
      * recommends to provide _snprintf/_vsnprintf() a buffer size that
      * is one less than the actual buffer, and zero it before calling
      * _snprintf/_vsnprintf() to workaround this problem.
-     * See https://web.archive.org/web/20151214111935/http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
+     * See http://msdn.microsoft.com/en-us/library/1kt27hek(v=vs.80).aspx */
     memset(s, 0, n);
     va_copy(ap_copy, ap);
     ret = _vsnprintf(s, n - 1, fmt, ap_copy);

compat/os2threads.h

@@ -27,19 +27,15 @@
 #define COMPAT_OS2THREADS_H
 
 #define INCL_DOS
-#define INCL_DOSERRORS
 #include <os2.h>
 
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>
-#include <time.h>
 
 #include <sys/builtin.h>
 #include <sys/fmutex.h>
 
 #include "libavutil/attributes.h"
-#include "libavutil/common.h"
-#include "libavutil/time.h"
 
 typedef struct {
     TID tid;
@@ -167,28 +163,6 @@ static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
     return 0;
 }
 
-static av_always_inline int pthread_cond_timedwait(pthread_cond_t *cond,
-                                                   pthread_mutex_t *mutex,
-                                                   const struct timespec *abstime)
-{
-    int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
-    ULONG t = av_clip64(abs_milli - av_gettime() / 1000, 0, ULONG_MAX);
-
-    __atomic_increment(&cond->wait_count);
-
-    pthread_mutex_unlock(mutex);
-
-    APIRET ret = DosWaitEventSem(cond->event_sem, t);
-
-    __atomic_decrement(&cond->wait_count);
-
-    DosPostEventSem(cond->ack_sem);
-
-    pthread_mutex_lock(mutex);
-
-    return (ret == ERROR_TIMEOUT) ? ETIMEDOUT : 0;
-}
-
 static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
                                               pthread_mutex_t *mutex)
 {

compat/w32dlfcn.h

@@ -20,40 +20,11 @@
 #define COMPAT_W32DLFCN_H
 
 #ifdef _WIN32
-#include <stdint.h>
-
 #include <windows.h>
-
 #include "config.h"
-#include "libavutil/macros.h"
+#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
 #include "libavutil/wchar_filename.h"
-
+#endif
-static inline wchar_t *get_module_filename(HMODULE module)
-{
-    wchar_t *path = NULL, *new_path;
-    DWORD path_size = 0, path_len;
-
-    do {
-        path_size = path_size ? FFMIN(2 * path_size, INT16_MAX + 1) : MAX_PATH;
-        new_path = av_realloc_array(path, path_size, sizeof *path);
-        if (!new_path) {
-            av_free(path);
-            return NULL;
-        }
-        path = new_path;
-        // Returns path_size in case of insufficient buffer.
-        // Whether the error is set or not and whether the output
-        // is null-terminated or not depends on the version of Windows.
-        path_len = GetModuleFileNameW(module, path, path_size);
-    } while (path_len && path_size <= INT16_MAX && path_size <= path_len);
-
-    if (!path_len) {
-        av_free(path);
-        return NULL;
-    }
-    return path;
-}
-
 /**
  * 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
@@ -63,53 +34,29 @@ static inline wchar_t *get_module_filename(HMODULE module)
  */
 static inline HMODULE win32_dlopen(const char *name)
 {
-    wchar_t *name_w;
-    HMODULE module = NULL;
-    if (utf8towchar(name, &name_w))
-        name_w = NULL;
 #if _WIN32_WINNT < 0x0602
-    // On Win7 and earlier we check if KB2533623 is available
+    // Need to check if KB2533623 is available
     if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
-        wchar_t *path = NULL, *new_path;
+        HMODULE module = NULL;
-        DWORD pathlen, pathsize, namelen;
+        wchar_t *path = NULL, *name_w = NULL;
-        if (!name_w)
+        DWORD pathlen;
+        if (utf8towchar(name, &name_w))
             goto exit;
-        namelen = wcslen(name_w);
+        path = (wchar_t *)av_mallocz_array(MAX_PATH, sizeof(wchar_t));
         // Try local directory first
-        path = get_module_filename(NULL);
+        pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
-        if (!path)
+        pathlen = wcsrchr(path, '\\') - path;
+        if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
             goto exit;
-        new_path = wcsrchr(path, '\\');
+        path[pathlen] = '\\';
-        if (!new_path)
-            goto exit;
-        pathlen = new_path - path;
-        pathsize = pathlen + namelen + 2;
-        new_path = av_realloc_array(path, pathsize, sizeof *path);
-        if (!new_path)
-            goto exit;
-        path = new_path;
         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, pathsize);
+            pathlen = GetSystemDirectoryW(path, MAX_PATH);
-            if (!pathlen)
+            if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
                 goto exit;
-            // Buffer is not enough in two cases:
+            path[pathlen] = '\\';
-            // 1. system directory + \ + module name
-            // 2. system directory even without the module name.
-            if (pathlen + namelen + 2 > pathsize) {
-                pathsize = pathlen + namelen + 2;
-                new_path = av_realloc_array(path, pathsize, sizeof *path);
-                if (!new_path)
-                    goto exit;
-                path = new_path;
-                // Query again to handle the case #2.
-                pathlen = GetSystemDirectoryW(path, pathsize);
-                if (!pathlen)
-                    goto exit;
-            }
-            path[pathlen] = L'\\';
             wcscpy(path + pathlen + 1, name_w);
             module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
         }
@@ -126,19 +73,16 @@ exit:
 #   define LOAD_LIBRARY_SEARCH_SYSTEM32        0x00000800
 #endif
 #if HAVE_WINRT
-    if (!name_w)
+    wchar_t *name_w = NULL;
+    int ret;
+    if (utf8towchar(name, &name_w))
         return NULL;
-    module = LoadPackagedLibrary(name_w, 0);
+    ret = LoadPackagedLibrary(name_w, 0);
-#else
-#define LOAD_FLAGS (LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32)
-    /* filename may be be in CP_ACP */
-    if (!name_w)
-        return LoadLibraryExA(name, NULL, LOAD_FLAGS);
-    module = LoadLibraryExW(name_w, NULL, LOAD_FLAGS);
-#undef LOAD_FLAGS
-#endif
     av_free(name_w);
-    return module;
+    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

compat/w32pthreads.h

@@ -38,13 +38,11 @@
 #define WIN32_LEAN_AND_MEAN
 #include <windows.h>
 #include <process.h>
-#include <time.h>
 
 #include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"
-#include "libavutil/time.h"
 
 typedef struct pthread_t {
     void *handle;
@@ -63,9 +61,6 @@ typedef CONDITION_VARIABLE pthread_cond_t;
 #define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
 #define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
 
-#define PTHREAD_CANCEL_ENABLE 1
-#define PTHREAD_CANCEL_DISABLE 0
-
 static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
     pthread_t *h = (pthread_t*)arg;
@@ -161,31 +156,10 @@ static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex
     return 0;
 }
 
-static inline int pthread_cond_timedwait(pthread_cond_t *cond, pthread_mutex_t *mutex,
-                                         const struct timespec *abstime)
-{
-    int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
-    DWORD t = av_clip64(abs_milli - av_gettime() / 1000, 0, UINT32_MAX);
-
-    if (!SleepConditionVariableSRW(cond, mutex, t, 0)) {
-        DWORD err = GetLastError();
-        if (err == ERROR_TIMEOUT)
-            return ETIMEDOUT;
-        else
-            return EINVAL;
-    }
-    return 0;
-}
-
 static inline int pthread_cond_signal(pthread_cond_t *cond)
 {
     WakeConditionVariable(cond);
     return 0;
 }
 
-static inline int pthread_setcancelstate(int state, int *oldstate)
-{
-    return 0;
-}
-
 #endif /* COMPAT_W32PTHREADS_H */

compat/windows/mswindres

@@ -1,32 +0,0 @@
-#!/bin/sh
-
-if [ "$1" = "--version" ]; then
-    rc.exe -?
-    exit $?
-fi
-
-if [ $# -lt 2 ]; then
-    echo "Usage: mswindres [-I/include/path ...] [-DSOME_DEFINE ...] [-o output.o] input.rc [output.o]" >&2
-    exit 0
-fi
-
-EXTRA_OPTS="-nologo"
-
-while [ $# -gt 2 ]; do
-    case $1 in
-    -D*) EXTRA_OPTS="$EXTRA_OPTS -d$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-    -I*) EXTRA_OPTS="$EXTRA_OPTS -i$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-    -o)  OPT_OUT="$2"; shift ;;
-    esac
-    shift
-done
-
-IN="$1"
-if [ -z "$OPT_OUT" ]; then
-    OUT="$2"
-else
-    OUT="$OPT_OUT"
-fi
-
-eval set -- $EXTRA_OPTS
-rc.exe "$@" -fo "$OUT" "$IN"

doc/APIchanges

@@ -1,665 +1,20 @@
-The last version increases of all libraries were on 2023-02-09
+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
+
 
 API changes, most recent first:
 
--------- 8< --------- FFmpeg 6.0 was cut here -------- 8< ---------
-
-2023-02-16 - 927042b409 - lavf 60.2.100 - avformat.h
-  Deprecate AVFormatContext io_close callback.
-  The superior io_close2 callback should be used instead.
-
-2023-02-13 - 2296078397 - lavu 58.1.100 - frame.h
-  Deprecate AVFrame.coded_picture_number and display_picture_number.
-  Their usefulness is questionable and very few decoders set them.
-
-2023-02-13 - 6b6f7db819 - lavc 60.2.100 - avcodec.h
-  Add AVCodecContext.frame_num as a 64bit version of frame_number.
-  Deprecate AVCodecContext.frame_number.
-
-2023-02-12 - d1b9a3ddb4 - lavfi 9.1.100 - avfilter.h
-  Add filtergraph segment parsing API.
-  New structs:
-    - AVFilterGraphSegment
-    - AVFilterChain
-    - AVFilterParams
-    - AVFilterPadParams
-  New functions:
-    - avfilter_graph_segment_parse()
-    - avfilter_graph_segment_create_filters()
-    - avfilter_graph_segment_apply_opts()
-    - avfilter_graph_segment_init()
-    - avfilter_graph_segment_link()
-    - avfilter_graph_segment_apply()
-
-2023-02-09 - 719a93f4e4 - lavu 58.0.100 - csp.h
-  Add av_csp_approximate_trc_gamma() and av_csp_trc_func_from_id().
-  Add av_csp_trc_function.
-
-2023-02-09 - 868a31b42d - lavc 60.0.100 - avcodec.h
-  avcodec_decode_subtitle2() now accepts const AVPacket*.
-
-2023-02-04 - d02340b9e3 - lavc 59.63.100
-  Allow AV_CODEC_FLAG_COPY_OPAQUE to be used with decoders.
-
-2023-01-29 - a1a80f2e64 - lavc 59.59.100 - avcodec.h
-  Add AV_CODEC_FLAG_COPY_OPAQUE and AV_CODEC_FLAG_FRAME_DURATION.
-
-2023-01-13 - 002d0ec740 - lavu 57.44.100 - ambient_viewing_environment.h frame.h
-  Adds a new structure for holding H.274 Ambient Viewing Environment metadata,
-  AVAmbientViewingEnvironment.
-  Adds a new AVFrameSideDataType entry AV_FRAME_DATA_AMBIENT_VIEWING_ENVIRONMENT
-  for it.
-
-2022-12-10 - 7a8d78f7e3 - lavc 59.55.100 - avcodec.h
-  Add AV_HWACCEL_FLAG_UNSAFE_OUTPUT.
-
-2022-11-24 - e97368eba5 - lavu 57.43.100 - tx.h
-  Add AV_TX_FLOAT_DCT, AV_TX_DOUBLE_DCT and AV_TX_INT32_DCT.
-
-2022-11-06 - 9dad237928 - lavu 57.42.100 - dict.h
-  Add av_dict_iterate().
-
-2022-11-03 - 6228ba141d - lavu 57.41.100 - channel_layout.h
-  Add AV_CH_LAYOUT_7POINT1_TOP_BACK and AV_CHANNEL_LAYOUT_7POINT1_TOP_BACK.
-
-2022-10-30 - 83e918de71 - lavu 57.40.100 - channel_layout.h
-  Add AV_CH_LAYOUT_CUBE and AV_CHANNEL_LAYOUT_CUBE.
-
-2022-10-11 - 479747645f - lavu 57.39.101 - pixfmt.h
-  Add AV_PIX_FMT_RGBF32 and AV_PIX_FMT_RGBAF32.
-
-2022-10-05 - 37d5ddc317 - lavu 57.39.100 - cpu.h
-  Add AV_CPU_FLAG_RVB_BASIC.
-
-2022-10-03 - d09776d486 - lavf 59.34.100 - avio.h
-  Make AVIODirContext an opaque type in a future major version bump.
-
-2022-09-27 - 0c0a3deb18 - lavu 57.38.100 - cpu.h
-  Add CPU flags for RISC-V vector extensions:
-  AV_CPU_FLAG_RVV_I32, AV_CPU_FLAG_RVV_F32, AV_CPU_FLAG_RVV_I64,
-  AV_CPU_FLAG_RVV_F64
-
-2022-09-26 - a02a0e8db4 - lavc 59.48.100 - avcodec.h
-  Deprecate avcodec_enum_to_chroma_pos() and avcodec_chroma_pos_to_enum().
-  Use av_chroma_location_enum_to_pos() or av_chroma_location_pos_to_enum()
-  instead.
-
-2022-09-26 - xxxxxxxxxx - lavu 57.37.100 - pixdesc.h pixfmt.h
-  Add av_chroma_location_enum_to_pos() and av_chroma_location_pos_to_enum().
-  Add AV_PIX_FMT_RGBF32BE, AV_PIX_FMT_RGBF32LE, AV_PIX_FMT_RGBAF32BE,
-  AV_PIX_FMT_RGBAF32LE.
-
-2022-09-26 - cf856d8957 - lavc 59.47.100 - avcodec.h defs.h
-  Move the AV_EF_* and FF_COMPLIANCE_* defines from avcodec.h to defs.h.
-
-2022-09-03 - d75c4693fe - lavu 57.36.100 - pixfmt.h
-  Add AV_PIX_FMT_P012, AV_PIX_FMT_Y212, AV_PIX_FMT_XV30, AV_PIX_FMT_XV36
-
-2022-09-03 - dea9744560 - lavu 57.35.100 - file.h
-  Deprecate av_tempfile() without replacement.
-
-2022-08-03 - cc5a5c9860 - lavu 57.34.100 - pixfmt.h
-  Add AV_PIX_FMT_VUYX.
-
-2022-08-22 - 14726571dd - lavf 59 - avformat.h
-  Deprecate av_stream_get_end_pts() without replacement.
-
-2022-08-19 - 352799dca8 - lavc 59.42.102 - codec_id.h
-  Deprecate AV_CODEC_ID_AYUV and ayuv decoder/encoder. The rawvideo codec
-  and vuya pixel format combination will be used instead from now on.
-
-2022-08-07 - e95b08a7dd - lavu 57.33.101 - pixfmt.h
-  Add AV_PIX_FMT_RGBAF16{BE,LE} pixel formats.
-
-2022-08-12 - e0bbdbe0a6 - lavu 57.33.100 - hwcontext_qsv.h
-  Add loader field to AVQSVDeviceContext
-
-2022-08-03 - 6ab8a9d375 - lavu 57.32.100 - pixfmt.h
-  Add AV_PIX_FMT_VUYA.
-
-2022-08-02 - e3838b856f - lavc 59.41.100 - avcodec.h codec.h
-  Add AV_CODEC_FLAG_RECON_FRAME and AV_CODEC_CAP_ENCODER_RECON_FRAME.
-  avcodec_receive_frame() may now be used on encoders when
-  AV_CODEC_FLAG_RECON_FRAME is active.
-
-2022-08-02 - eede1d2927 - lavu 57.31.100 - frame.h
-  av_frame_make_writable() may now be called on non-refcounted
-  frames and will make a refcounted copy out of them.
-  Previously an error was returned in such cases.
-
-2022-07-30 - e1a0f2df3d - lavc 59.40.100 - avcodec.h
-  Add the AV_CODEC_FLAG2_ICC_PROFILES flag to AVCodecContext, to enable
-  automatic reading and writing of embedded ICC profiles in image files.
-  The "flags2" option now supports the corresponding flag "icc_profiles".
-
-2022-07-19 - 4397f9a5a0 - lavu 57.30.100 - frame.h
-  Add AVFrame.duration, deprecate AVFrame.pkt_duration.
-
--------- 8< --------- FFmpeg 5.1 was cut here -------- 8< ---------
-
-2022-06-12 - 7cae3d8b76 - lavf 59.25.100 - avio.h
-  Add avio_vprintf(), similar to avio_printf() but allow to use it
-  from within a function taking a variable argument list as input.
-
-2022-06-12 - ff59ecc4de - lavu 57.27.100 - uuid.h
-  Add UUID handling functions.
-  Add av_uuid_parse(), av_uuid_urn_parse(), av_uuid_parse_range(),
-  av_uuid_parse_range(), av_uuid_equal(), av_uuid_copy(), and av_uuid_nil().
-
-2022-06-01 - d42b410e05 - lavu 57.26.100 - csp.h
-  Add public API for colorspace structs.
-  Add av_csp_luma_coeffs_from_avcsp(), av_csp_primaries_desc_from_id(),
-  and av_csp_primaries_id_from_desc().
-
-2022-05-23 - 4cdc14aa95 - lavu 57.25.100 - avutil.h
-  Deprecate av_fopen_utf8() without replacement.
-
-2022-03-16 - f3a0e2ee2b - all libraries - version_major.h
-  Add lib<name>/version_major.h as new installed headers, which only
-  contain the major version number (and corresponding API deprecation
-  defines).
-
-2022-03-15 - cdba98bb80 - swr 4.5.100 - swresample.h
-  Add swr_alloc_set_opts2() and swr_build_matrix2().
-  Deprecate swr_alloc_set_opts() and swr_build_matrix().
-
-2022-03-15 - cdba98bb80 - lavfi 8.28.100 - avfilter.h buffersink.h buffersrc.h
-  Update AVFilterLink for the new channel layout API: add ch_layout,
-  deprecate channel_layout.
-
-  Update the buffersink filter sink for the new channel layout API:
-  add av_buffersink_get_ch_layout() and the ch_layouts option,
-  deprecate av_buffersink_get_channel_layout() and the channel_layouts option.
-
-  Update AVBufferSrcParameters for the new channel layout API:
-  add ch_layout, deprecate channel_layout.
-
-2022-03-15 - cdba98bb80 - lavf 59.19.100 - avformat.h
-  Add AV_DISPOSITION_NON_DIEGETIC.
-
-2022-03-15 - cdba98bb80 - lavc 59.24.100 - avcodec.h codec_par.h
-  Update AVCodecParameters for the new channel layout API: add ch_layout,
-  deprecate channels/channel_layout.
-
-  Update AVCodecContext for the new channel layout API: add ch_layout,
-  deprecate channels/channel_layout.
-
-  Update AVCodec for the new channel layout API: add ch_layouts,
-  deprecate channel_layouts.
-
-2022-03-15 - cdba98bb80 - lavu 57.24.100 - channel_layout.h frame.h opt.h
-  Add new channel layout API based on the AVChannelLayout struct.
-  Add support for Ambisonic audio.
-  Deprecate previous channel layout API based on uint64 bitmasks.
-
-  Add AV_OPT_TYPE_CHLAYOUT option type, deprecate AV_OPT_TYPE_CHANNEL_LAYOUT.
-  Update AVFrame for the new channel layout API: add ch_layout, deprecate
-  channels/channel_layout.
-
-2022-03-10 - f629ea2e18 - lavu 57.23.100 - cpu.h
-  Add AV_CPU_FLAG_AVX512ICL.
-
-2022-02-07 - a10f1aec1f - lavu 57.21.100 - fifo.h
-  Deprecate AVFifoBuffer and the API around it, namely av_fifo_alloc(),
-  av_fifo_alloc_array(), av_fifo_free(), av_fifo_freep(), av_fifo_reset(),
-  av_fifo_size(), av_fifo_space(), av_fifo_generic_peek_at(),
-  av_fifo_generic_peek(), av_fifo_generic_read(), av_fifo_generic_write(),
-  av_fifo_realloc2(), av_fifo_grow(), av_fifo_drain() and av_fifo_peek2().
-  Users should switch to the AVFifo-API.
-
-2022-02-07 - 7329b22c05 - lavu 57.20.100 - fifo.h
-  Add a new FIFO API, which allows setting a FIFO element size.
-  This API operates on these elements rather than on bytes.
-  Add av_fifo_alloc2(), av_fifo_elem_size(), av_fifo_can_read(),
-  av_fifo_can_write(), av_fifo_grow2(), av_fifo_drain2(), av_fifo_write(),
-  av_fifo_write_from_cb(), av_fifo_read(), av_fifo_read_to_cb(),
-  av_fifo_peek(), av_fifo_peek_to_cb(), av_fifo_drain2(), av_fifo_reset2(),
-  av_fifo_freep2(), av_fifo_auto_grow_limit().
-
-2022-01-26 - af94ab7c7c0 - lavu 57.19.100 - tx.h
-  Add AV_TX_FLOAT_RDFT, AV_TX_DOUBLE_RDFT and AV_TX_INT32_RDFT.
-
--------- 8< --------- FFmpeg 5.0 was cut here -------- 8< ---------
-
-2022-01-04 - 78dc21b123e - lavu 57.16.100 - frame.h
-  Add AV_FRAME_DATA_DOVI_METADATA.
-
-2022-01-03 - 70f318e6b6c - lavf 59.13.100 - avformat.h
-  Add AVFMT_EXPERIMENTAL flag.
-
-2021-12-22 - b7e1ec7bda9 - lavu 57.13.100 - hwcontext_videotoolbox.h
-  Add av_vt_pixbuf_set_attachments
-
-2021-12-22 - 69bd95dcd8d - lavu 57.13.100 - hwcontext_videotoolbox.h
-  Add av_map_videotoolbox_chroma_loc_from_av
-  Add av_map_videotoolbox_color_matrix_from_av
-  Add av_map_videotoolbox_color_primaries_from_av
-  Add av_map_videotoolbox_color_trc_from_av
-
-2021-12-21 - ffbab99f2c2 - lavu 57.12.100 - cpu.h
-  Add AV_CPU_FLAG_SLOW_GATHER.
-
-2021-12-20 - 278068dc60d - lavu 57.11.101 - display.h
-  Modified the documentation of av_display_rotation_set()
-  to match its longstanding actual behaviour of treating
-  the angle as directed clockwise.
-
-2021-12-12 - 64834bb86a1 - lavf 59.10.100 - avformat.h
-  Add AVFormatContext io_close2 which returns an int
-
-2021-12-10 - f45cbb775e4 - lavu 57.11.100 - hwcontext_vulkan.h
-  Add AVVkFrame.offset and AVVulkanFramesContext.flags.
-
-2021-12-04 - b9c928a486f - lavfi 8.19.100 - avfilter.h
-  Add AVFILTER_FLAG_METADATA_ONLY.
-
-2021-12-03 - b236ef0a594 - lavu 57.10.100 - frame.h
-  Add AVFrame.time_base
-
-2021-11-22 - b2cd1fb2ec6 - lavu 57.9.100 - pixfmt.h
-  Add AV_PIX_FMT_P210, AV_PIX_FMT_P410, AV_PIX_FMT_P216, and AV_PIX_FMT_P416.
-
-2021-11-17 - 54e65aa38ab - lavf 57.9.100 - frame.h
-  Add AV_FRAME_DATA_DOVI_RPU_BUFFER.
-
-2021-11-16 - ed75a08d36c - lavf 59.9.100 - avformat.h
-  Add av_stream_get_class(). Schedule adding AVStream.av_class at libavformat
-  major version 60.
-  Add av_disposition_to_string() and av_disposition_from_string().
-  Add "disposition" AVOption to AVStream's class.
-
-2021-11-12 - 8478d60d5b5 - lavu 57.8.100 - hwcontext_vulkan.h
-  Added AVVkFrame.sem_value, AVVulkanDeviceContext.queue_family_encode_index,
-  nb_encode_queues, queue_family_decode_index, and nb_decode_queues.
-
-2021-10-18 - 682bafdb125 - lavf 59.8.100 - avio.h
-  Introduce public bytes_{read,written} statistic fields to AVIOContext.
-
-2021-10-13 - a5622ed16f8 - lavf 59.7.100 - avio.h
-  Deprecate AVIOContext.written. Originally added as a private entry in
-  commit 3f75e5116b900f1428aa13041fc7d6301bf1988a, its grouping with
-  the comment noting its private state was missed during merging of the field
-  from Libav (most likely due to an already existing field in between).
-
-2021-09-21 - 0760d9153c3 - lavu 57.7.100 - pixfmt.h
-  Add AV_PIX_FMT_X2BGR10.
-
-2021-09-20 - 8d5de914d31 - lavu 57.6.100 - mem.h
-  Deprecate av_mallocz_array() as it is identical to av_calloc().
-
-2021-09-20 - 176b8d785bf - lavc 59.9.100 - avcodec.h
-  Deprecate AVCodecContext.sub_text_format and the corresponding
-  AVOptions. It is unused since the last major bump.
-
-2021-09-20 - dd846bc4a91 - lavc 59.8.100 - avcodec.h codec.h
-  Deprecate AV_CODEC_FLAG_TRUNCATED and AV_CODEC_CAP_TRUNCATED,
-  as they are redundant with parsers.
-
-2021-09-17 - ccfdef79b13 - lavu 57.5.101 - buffer.h
-  Constified the input parameters in av_buffer_replace(), av_buffer_ref(),
-  and av_buffer_pool_buffer_get_opaque().
-
-2021-09-08 - 4f78711f9c2 - lavu 57.5.100 - hwcontext_d3d11va.h
-  Add AVD3D11VAFramesContext.texture_infos
-
-2021-09-06 - 42cd64c1826 - lsws 6.1.100 - swscale.h
-  Add AVFrame-based scaling API:
-    - sws_scale_frame()
-    - sws_frame_start()
-    - sws_frame_end()
-    - sws_send_slice()
-    - sws_receive_slice()
-    - sws_receive_slice_alignment()
-
-2021-09-02 - cbf111059d2 - lavc 59.7.100 - avcodec.h
-  Incremented the number of elements of AVCodecParser.codec_ids to seven.
-
-2021-08-24 - 590a7e02f04 - lavc 59.6.100 - avcodec.h
-  Add FF_CODEC_PROPERTY_FILM_GRAIN
-
-2021-08-20 - 7c5f998196d - lavfi 8.3.100 - avfilter.H
-  Add avfilter_filter_pad_count() as a replacement for avfilter_pad_count().
-  Deprecate avfilter_pad_count().
-
-2021-08-17 - 8c53b145993 - lavu 57.4.101 - opt.h
-  av_opt_copy() now guarantees that allocated src and dst options
-  don't alias each other even on error.
-
-2021-08-14 - d5de9965ef6 - lavu 57.4.100 - imgutils.h
-  Add av_image_copy_plane_uc_from()
-
-2021-08-02 - a1a0fddfd05 - lavc 59.4.100 - packet.h
-  Add AVPacket.opaque, AVPacket.opaque_ref, AVPacket.time_base.
-
-2021-07-23 - 2dd8acbe800 - lavu 57.3.100 - common.h macros.h
-  Move several macros (AV_NE, FFDIFFSIGN, FFMAX, FFMAX3, FFMIN, FFMIN3,
-  FFSWAP, FF_ARRAY_ELEMS, MKTAG, MKBETAG) from common.h to macros.h.
-
-2021-07-22 - e3b5ff17c2e - lavu 57.2.100 - film_grain_params.h
-  Add AV_FILM_GRAIN_PARAMS_H274, AVFilmGrainH274Params
-
-2021-07-19 - c1bf56a526f - lavu 57.1.100 - cpu.h
-  Add av_cpu_force_count()
-
-2021-06-17 - aca923b3653 - lavc 59.2.100 - packet.h
-  Add AV_PKT_DATA_DYNAMIC_HDR10_PLUS
-
-2021-06-09 - 2cccab96f6f - lavf 59.3.100 - avformat.h
-  Add pts_wrap_bits to AVStream
-
-2021-06-10 - 7c9763070d9 - lavc 59.1.100 - avcodec.h codec.h
-  Move av_get_profile_name() from avcodec.h to codec.h.
-
-2021-06-10 - bb3648e6766 - lavc 59.1.100 - avcodec.h codec_par.h
-  Move av_get_audio_frame_duration2() from avcodec.h to codec_par.h.
-
-2021-06-10 - 881db34f6a0 - lavc 59.1.100 - avcodec.h codec_id.h
-  Move av_get_bits_per_sample(), av_get_exact_bits_per_sample(),
-  avcodec_profile_name(), and av_get_pcm_codec() from avcodec.h
-  to codec_id.h.
-
-2021-06-10 - ff0a96046d8 - lavc 59.1.100 - avcodec.h defs.h
-  Add new installed header defs.h. The following definitions are moved
-  into it from avcodec.h:
-    - AVDiscard
-    - AVAudioServiceType
-    - AVPanScan
-    - AVCPBProperties and av_cpb_properties_alloc()
-    - AVProducerReferenceTime
-    - av_xiphlacing()
-
-2021-04-27 - cb3ac722f4 - lavc 59.0.100 - avcodec.h
-  Constified AVCodecParserContext.parser.
-
-2021-04-27 - 8b3e6ce5f4 - lavd 59.0.100 - avdevice.h
-  The av_*_device_next API functions now accept and return
-  pointers to const AVInputFormat resp. AVOutputFormat.
-
-2021-04-27 - d7e0d428fa - lavd 59.0.100 - avdevice.h
-  avdevice_list_input_sources and avdevice_list_output_sinks now accept
-  pointers to const AVInputFormat resp. const AVOutputFormat.
-
-2021-04-27 - 46dac8cf3d - lavf 59.0.100 - avformat.h
-  av_find_best_stream now uses a const AVCodec ** parameter
-  for the returned decoder.
-
-2021-04-27 - 626535f6a1 - lavc 59.0.100 - codec.h
-  avcodec_find_encoder_by_name(), avcodec_find_encoder(),
-  avcodec_find_decoder_by_name() and avcodec_find_decoder()
-  now return a pointer to const AVCodec.
-
-2021-04-27 - 14fa0a4efb - lavf 59.0.100 - avformat.h
-  Constified AVFormatContext.*_codec.
-
-2021-04-27 - 56450a0ee4 - lavf 59.0.100 - avformat.h
-  Constified the pointers to AVInputFormats and AVOutputFormats
-  in AVFormatContext, avformat_alloc_output_context2(),
-  av_find_input_format(), av_probe_input_format(),
-  av_probe_input_format2(), av_probe_input_format3(),
-  av_probe_input_buffer2(), av_probe_input_buffer(),
-  avformat_open_input(), av_guess_format() and av_guess_codec().
-  Furthermore, constified the AVProbeData in av_probe_input_format(),
-  av_probe_input_format2() and av_probe_input_format3().
-
-2021-04-19 - 18af1ea8d1 - lavu 56.74.100 - tx.h
-  Add AV_TX_FULL_IMDCT and AV_TX_UNALIGNED.
-
-2021-04-17 - f1bf465aa0 - lavu 56.73.100 - frame.h detection_bbox.h
-  Add AV_FRAME_DATA_DETECTION_BBOXES
-
-2021-04-06 - 557953a397 - lavf 58.78.100 - avformat.h
-  Add avformat_index_get_entries_count(), avformat_index_get_entry(),
-  and avformat_index_get_entry_from_timestamp().
-
-2021-03-21 - a77beea6c8 - lavu 56.72.100 - frame.h
-  Deprecated av_get_colorspace_name().
-  Use av_color_space_name() instead.
-
--------- 8< --------- FFmpeg 4.4 was cut here -------- 8< ---------
-
-2021-03-19 - e8c0bca6bd - lavu 56.69.100 - adler32.h
-  Added a typedef for the type of the Adler-32 checksums
-  used by av_adler32_update(). It will be changed to uint32_t
-  at the next major bump.
-  The type of the parameter for the length of the input buffer
-  will also be changed to size_t at the next major bump.
-
-2021-03-19 - e318438f2f - lavf 58.75.100  - avformat.h
-  AVChapter.id will be changed from int to int64_t
-  on the next major version bump.
-
-2021-03-17 - f7db77bd87 - lavc 58.133.100 - codec.h
-  Deprecated av_init_packet(). Once removed, sizeof(AVPacket) will
-  no longer be a part of the public ABI.
-  Deprecated AVPacketList.
-
-2021-03-16 - 7d09579190 - lavc 58.132.100 - codec.h
-  Add AV_CODEC_CAP_OTHER_THREADS as a new name for
-  AV_CODEC_CAP_AUTO_THREADS. AV_CODEC_CAP_AUTO_THREADS
-  is now deprecated.
-
-2021-03-12 - 6e7e3a3820 - lavc 58.131.100 - avcodec.h codec.h
-  Add a get_encode_buffer callback to AVCodecContext, similar to
-  get_buffer2 but for encoders.
-  Add avcodec_default_get_encode_buffer().
-  Add AV_GET_ENCODE_BUFFER_FLAG_REF.
-  Encoders may now be flagged as AV_CODEC_CAP_DR1 capable.
-
-2021-03-10 - 42e68fe015 - lavf 58.72.100 - avformat.h
-  Change AVBufferRef related AVStream function and struct size
-  parameter and fields type to size_t at next major bump.
-
-2021-03-10 - d79e0fe65c - lavc 58.130.100 - packet.h
-  Change AVBufferRef related AVPacket function and struct size
-  parameter and fields type to size_t at next major bump.
-
-2021-03-10 - 14040a1d91 - lavu 56.68.100 - buffer.h frame.h
-  Change AVBufferRef and relevant AVFrame function and struct size
-  parameter and fields type to size_t at next major bump.
-
-2021-03-04 - a0eec776b6 - lavc 58.128.101 - avcodec.h
-  Enable err_recognition to be set for encoders.
-
-2021-03-03 - 2ff40b98ec - lavf 58.70.100 - avformat.h
-  Deprecate AVFMT_FLAG_PRIV_OPT. It will do nothing
-  as soon as av_demuxer_open() is removed.
-
-2021-02-27 - dd9227e48f - lavc 58.126.100 - avcodec.h
-  Deprecated avcodec_get_frame_class().
-
-2021-02-21 - 5ca40d6d94 - lavu 56.66.100 - tx.h
-  Add enum AVTXFlags and AVTXFlags.AV_TX_INPLACE
-
-2021-02-14 - 4f49ca7bbc - lavd 58.12.100 - avdevice.h
-  Deprecated avdevice_capabilities_create() and
-  avdevice_capabilities_free().
-
-2021-02-10 - 1bda9bb68a - lavu 56.65.100 - common.h
-  Add FFABS64U()
-
-2021-01-26 - 5dd9567080 - lavu 56.64.100 - common.h
-  Add FFABSU()
-
-2021-01-25 - 56709ca8aa - lavc 58.119.100 - avcodec.h
-  Deprecate AVCodecContext.debug_mv, FF_DEBUG_VIS_MV_P_FOR, FF_DEBUG_VIS_MV_B_FOR,
-  FF_DEBUG_VIS_MV_B_BACK
-
-2021-01-11 - ebdd33086a - lavc 58.116.100 - avcodec.h
-  Add FF_PROFILE_VVC_MAIN_10 and FF_PROFILE_VVC_MAIN_10_444.
-
-2020-01-01 - baecaa16c1 - lavu 56.63.100 - video_enc_params.h
-  Add AV_VIDEO_ENC_PARAMS_MPEG2
-
-2020-12-03 - eca12f4d5a - lavu 56.62.100 - timecode.h
-  Add av_timecode_init_from_components.
-
-2020-11-27 - a83098ab03 - lavc 58.114.100 - avcodec.h
-  Deprecate AVCodecContext.thread_safe_callbacks. Starting with
-  LIBAVCODEC_VERSION_MAJOR=60, user callbacks must always be
-  thread-safe when frame threading is used.
-
-2020-11-25 - d243dd540a - lavc 58.113.100 - avcodec.h
-  Adds a new flag AV_CODEC_EXPORT_DATA_FILM_GRAIN for export_side_data.
-
-2020-11-25 - 4f9ee87253 - lavu 56.61.100 - film_grain_params.h
-  Adds a new API for extracting codec film grain parameters as side data.
-  Adds a new AVFrameSideDataType entry AV_FRAME_DATA_FILM_GRAIN_PARAMS for it.
-
-2020-10-28 - f95d9510ff - lavf 58.64.100 - avformat.h
-  Add AVSTREAM_EVENT_FLAG_NEW_PACKETS.
-
-2020-09-28 - 68918d3b7f - lavu 56.60.100 - buffer.h
-  Add a av_buffer_replace() convenience function.
-
-2020-09-13 - 837b6eb90e - lavu 56.59.100 - timecode.h
-  Add av_timecode_make_smpte_tc_string2.
-
-2020-08-21 - 06f2651204 - lavu 56.58.100 - avstring.h
-  Deprecate av_d2str(). Use av_asprintf() instead.
-
-2020-08-04 - 34de0abbe7 - lavu 56.58.100 - channel_layout.h
-  Add AV_CH_LAYOUT_22POINT2 together with its newly required pieces:
-  AV_CH_TOP_SIDE_LEFT, AV_CH_TOP_SIDE_RIGHT, AV_CH_BOTTOM_FRONT_CENTER,
-  AV_CH_BOTTOM_FRONT_LEFT, AV_CH_BOTTOM_FRONT_RIGHT.
-
-2020-07-23 - 84655b7101 - lavu 56.57.100 - cpu.h
-  Add AV_CPU_FLAG_MMI and AV_CPU_FLAG_MSA.
-
-2020-07-22 - 3a8e927176 - lavu 56.56.100 - imgutils.h
-  Add av_image_fill_plane_sizes().
-
-2020-07-15 - 448a9aaa78 - lavc 58.96.100 - packet.h
-  Add AV_PKT_DATA_S12M_TIMECODE.
-
-2020-06-12 - b09fb030c1 - lavu 56.55.100 - pixdesc.h
-  Add AV_PIX_FMT_X2RGB10.
-
-2020-06-11 - bc8ab084fb - lavu 56.54.100 - frame.h
-  Add AV_FRAME_DATA_SEI_UNREGISTERED.
-
-2020-06-10 - 1b4a98b029 - lavu 56.53.100 - log.h opt.h
-  Add av_opt_child_class_iterate() and AVClass.child_class_iterate().
-  Deprecate av_opt_child_class_next() and AVClass.child_class_next().
-
--------- 8< --------- FFmpeg 4.3 was cut here -------- 8< ---------
-
-2020-06-05 - ec39c2276a - lavu 56.50.100 - buffer.h
-  Passing NULL as alloc argument to av_buffer_pool_init2() is now allowed.
-
-2020-05-27 - ba6cada92e - lavc 58.88.100 - avcodec.h codec.h
-  Move AVCodec-related public API to new header codec.h.
-
-2020-05-23 - 064b875e89 - lavu 56.49.100 - video_enc_params.h
-  Add AV_VIDEO_ENC_PARAMS_H264.
-
-2020-05-23 - 2e08b39444 - lavu 56.48.100 - hwcontext.h
-  Add av_hwdevice_ctx_create_derived_opts.
-
-2020-05-23 - 6b65c4ec54 - lavu 56.47.100 - rational.h
-  Add av_gcd_q().
-
-2020-05-22 - af9e622776 - lavu 56.46.101 - opt.h
-  Add AV_OPT_FLAG_CHILD_CONSTS.
-
-2020-05-22 - 9d443c3e68 - lavc 58.87.100 - avcodec.h codec_par.h
-  Move AVBitstreamFilter-related public API to new header bsf.h.
-  Move AVCodecParameters-related public API to new header codec_par.h.
-
-2020-05-21 - 13b1bbff0b - lavc 58.86.101 - avcodec.h
-  Deprecated AV_CODEC_CAP_INTRA_ONLY and AV_CODEC_CAP_LOSSLESS.
-
-2020-05-17 - 84af196c65 - lavu 56.46.100 - common.h
-  Add av_sat_add64() and av_sat_sub64()
-
-2020-05-12 - 991d417692 - lavu 56.45.100 - video_enc_params.h
-                          lavc 58.84.100 - avcodec.h
-  Add a new API for exporting video encoding information.
-  Replaces the deprecated API for exporting QP tables from decoders.
-  Add AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS to request this information from
-  decoders.
-
-2020-05-10 - dccd07f66d - lavu 56.44.100 - hwcontext_vulkan.h
-  Add enabled_inst_extensions, num_enabled_inst_extensions, enabled_dev_extensions
-  and num_enabled_dev_extensions fields to AVVulkanDeviceContext
-
-2020-04-22 - 0e1db79e37 - lavc 58.81.100 - packet.h
-                        - lavu 56.43.100 - dovi_meta.h
-  Add AV_PKT_DATA_DOVI_CONF and AVDOVIDecoderConfigurationRecord.
-
-2020-04-15 - 22b25b3ea5 - lavc 58.79.100 - avcodec.h
-  Add formal support for calling avcodec_flush_buffers() on encoders.
-  Encoders that set the cap AV_CODEC_CAP_ENCODER_FLUSH will be flushed.
-  For all other encoders, the call is now a no-op rather than undefined
-  behaviour.
-
-2020-04-10 - 672946c7fe - lavc 58.78.100 - avcodec.h codec_desc.h codec_id.h packet.h
-  Move AVCodecDesc-related public API to new header codec_desc.h.
-  Move AVCodecID enum to new header codec_id.h.
-  Move AVPacket-related public API to new header packet.h.
-
-2020-03-29 - 4cb0dda555 - lavf 58.42.100 - avformat.h
-  av_read_frame() now guarantees to handle uninitialized input packets
-  and to return refcounted packets on success.
-
-2020-03-27 - c52ec0367d - lavc 58.77.100 - avcodec.h
-  av_packet_ref() now guarantees to return the destination packet
-  in a blank state on error.
-
-2020-03-10 - 05d27f342b - lavc 58.75.100 - avcodec.h
-  Add AV_PKT_DATA_ICC_PROFILE.
-
-2020-02-21 - d005a7cdfd - lavc 58.73.101 - avcodec.h
-  Add AV_CODEC_EXPORT_DATA_PRFT.
-
-2020-02-21 - c666689491 - lavc 58.73.100 - avcodec.h
-  Add AVCodecContext.export_side_data and AV_CODEC_EXPORT_DATA_MVS.
-
-2020-02-13 - e8f054b095 - lavu 56.41.100 - tx.h
-  Add AV_TX_INT32_FFT and AV_TX_INT32_MDCT
-
-2020-02-12 - 3182114f88 - lavu 56.40.100 - log.h
-  Add av_log_once().
-
-2020-02-04 - a88449ffb2 - lavu 56.39.100 - hwcontext.h
-  Add AV_PIX_FMT_VULKAN
-  Add AV_HWDEVICE_TYPE_VULKAN and implementation.
-
-2020-01-30 - 27529eeb27 - lavf 58.37.100 - avio.h
-  Add avio_protocol_get_class().
-
-2020-01-15 - 717b2074ec - lavc 58.66.100 - avcodec.h
-  Add AV_PKT_DATA_PRFT and AVProducerReferenceTime.
-
-2019-12-27 - 45259a0ee4 - lavu 56.38.100 - eval.h
-  Add av_expr_count_func().
-
-2019-12-26 - 16685114d5 - lavu 56.37.100 - buffer.h
-  Add av_buffer_pool_buffer_get_opaque().
-
-2019-11-17 - 1c23abc88f - lavu 56.36.100 - eval API
-  Add av_expr_count_vars().
-
-2019-10-14 - f3746d31f9 - lavu 56.35.101 - opt.h
-  Add AV_OPT_FLAG_RUNTIME_PARAM.
-
-2019-09-25 - f8406ab4b9 - lavc 58.59.100 - avcodec.h
-  Add max_samples
-
-2019-09-04 - 2a9d461abc - lavu 56.35.100 - hwcontext_videotoolbox.h
-  Add av_map_videotoolbox_format_from_pixfmt2() for full range pixfmt
-
-2019-09-01 - 8821d1f56e - lavu 56.34.100 - pixfmt.h
-  Add EBU Tech. 3213-E AVColorPrimaries value
-
-2019-08-17 - 95fa73a2b4 - lavf 58.31.101 - avio.h
-  4K limit removed from avio_printf.
-
-2019-08-17 - a82f8f2f10 - lavf 58.31.100 - avio.h
-  Add avio_print_string_array and avio_print.
-
-2019-07-27 - 42e2319ba9 - lavu 56.33.100 - tx.h
-  Add AV_TX_DOUBLE_FFT and AV_TX_DOUBLE_MDCT
-
 -------- 8< --------- FFmpeg 4.2 was cut here -------- 8< ---------
 
 2019-06-21 - a30e44098a - lavu 56.30.100 - frame.h
@@ -1810,7 +1165,7 @@ API changes, most recent first:
 2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
   Add av_format_inject_global_side_data()
 
-2014-04-12 - 4f698be8f - lavu 52.76.100 - log.h
+2014-04-12 - 4f698be - lavu 52.76.100 - log.h
   Add av_log_get_flags()
 
 2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h

doc/Doxyfile

@@ -38,7 +38,7 @@ PROJECT_NAME           = FFmpeg
 # could be handy for archiving the generated documentation or if some version
 # control system is used.
 
-PROJECT_NUMBER         = 6.0.1
+PROJECT_NUMBER         = 4.2.2
 
 # Using the PROJECT_BRIEF tag one can provide an optional one line description
 # for a project that appears at the top of each page and should give viewer a
@@ -1980,7 +1980,6 @@ PREDEFINED             = __attribute__(x)= \
                          av_alloc_size(...)= \
                          AV_GCC_VERSION_AT_LEAST(x,y)=1 \
                          AV_GCC_VERSION_AT_MOST(x,y)=0 \
-                         "FF_PAD_STRUCTURE(name,size,...)=typedef struct name { __VA_ARGS__ } name;" \
                          __GNUC__
 
 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then this

doc/Makefile

@@ -27,9 +27,6 @@ HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
               doc/mailing-list-faq.html                                 \
               doc/nut.html                                              \
               doc/platform.html                                         \
-              $(SRC_PATH)/doc/bootstrap.min.css                         \
-              $(SRC_PATH)/doc/style.min.css                             \
-              $(SRC_PATH)/doc/default.css                               \
 
 TXTPAGES    = doc/fate.txt                                              \
 
@@ -105,7 +102,7 @@ DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak
 
 doc/doxy/html: TAG = DOXY
 doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $$PWD/doc/doxy $(SRC_PATH) doc/Doxyfile $(DOXYGEN) $(DOXY_INPUT);
+	$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
 
 install-doc: install-html install-man
 

doc/authors.texi

@@ -3,9 +3,9 @@
 The FFmpeg developers.
 
 For details about the authorship, see the Git history of the project
-(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
 @command{git log} in the FFmpeg source directory, or browsing the
-online repository at @url{https://git.ffmpeg.org/ffmpeg}.
+online repository at @url{http://source.ffmpeg.org}.
 
 Maintainers for the specific components are listed in the file
 @file{MAINTAINERS} in the source code tree.

doc/bitstream_filters.texi

@@ -81,7 +81,7 @@ Top-left position.
 @end table
 
 @item tick_rate
-Set the tick rate (@emph{time_scale / num_units_in_display_tick}) in
+Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
 the timing info in the sequence header.
 @item num_ticks_per_picture
 Set the number of ticks in each picture, to indicate that the stream
@@ -132,36 +132,6 @@ 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 dv_error_marker
-
-Blocks in DV which are marked as damaged are replaced by blocks of the specified color.
-
-@table @option
-@item color
-The color to replace damaged blocks by
-@item sta
-A 16 bit mask which specifies which of the 16 possible error status values are
-to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0
-error status values.
-@table @samp
-@item ok
-No error, no concealment
-@item err
-Error, No concealment
-@item res
-Reserved
-@item notok
-Error or concealment
-@item notres
-Not reserved
-@item Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
-The specific error status code
-@end table
-see page 44-46 or section 5.5 of
-@url{http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf}
-
-@end table
-
 @section eac3_core
 
 Extract the core from a E-AC-3 stream, dropping extra channels.
@@ -247,20 +217,12 @@ Modify metadata embedded in an H.264 stream.
 Insert or remove AUD NAL units in all access units of the stream.
 
 @table @samp
-@item pass
 @item insert
 @item remove
 @end table
 
-Default is pass.
-
 @item sample_aspect_ratio
 Set the sample aspect ratio of the stream in the VUI parameters.
-See H.264 table E-1.
-
-@item overscan_appropriate_flag
-Set whether the stream is suitable for display using overscan
-or not (see H.264 section E.2.1).
 
 @item video_format
 @item video_full_range_flag
@@ -278,7 +240,7 @@ 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 (time_scale / num_units_in_tick) in the VUI
+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).
@@ -287,11 +249,6 @@ 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 zero_new_constraint_set_flags
-Zero constraint_set4_flag and constraint_set5_flag in the SPS. These
-bits were reserved in a previous version of the H.264 spec, and thus
-some hardware decoders require these to be zero. The result of zeroing
-this is still a valid bitstream.
 
 @item crop_left
 @item crop_right
@@ -315,37 +272,6 @@ insert the string ``hello'' associated with the given UUID.
 @item delete_filler
 Deletes both filler NAL units and filler SEI messages.
 
-@item display_orientation
-Insert, extract or remove Display orientation SEI messages.
-See H.264 section D.1.27 and D.2.27 for syntax and semantics.
-
-@table @samp
-@item pass
-@item insert
-@item remove
-@item extract
-@end table
-
-Default is pass.
-
-Insert mode works in conjunction with @code{rotate} and @code{flip} options.
-Any pre-existing Display orientation messages will be removed in insert or remove mode.
-Extract mode attaches the display matrix to the packet as side data.
-
-@item rotate
-Set rotation in display orientation SEI (anticlockwise angle in degrees).
-Range is -360 to +360. Default is NaN.
-
-@item flip
-Set flip in display orientation SEI.
-
-@table @samp
-@item horizontal
-@item vertical
-@end table
-
-Default is unset.
-
 @item level
 Set the level in the SPS.  Refer to H.264 section A.3 and tables A-1
 to A-5.
@@ -382,6 +308,9 @@ 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.
@@ -414,8 +343,8 @@ 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 (time_scale /
+Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
-num_units_in_tick). Combined with @option{num_ticks_poc_diff_one}, this can
+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.
 
@@ -596,110 +525,25 @@ container. Can be used for fuzzing or testing error resilience/concealment.
 Parameters:
 @table @option
 @item amount
-Accepts an expression whose evaluation per-packet determines how often bytes in that
+A numeral string, whose value is related to how often output bytes will
-packet will be modified. A value below 0 will result in a variable frequency.
+be modified. Therefore, values below or equal to 0 are forbidden, and
-Default is 0 which results in no modification. However, if neither amount nor drop is specified,
+the lower the more frequent bytes will be modified, with 1 meaning
-amount will be set to @var{-1}. See below for accepted variables.
+every byte is modified.
-@item drop
-Accepts an expression evaluated per-packet whose value determines whether that packet is dropped.
-Evaluation to a positive value results in the packet being dropped. Evaluation to a negative
-value results in a variable chance of it being dropped, roughly inverse in proportion to the magnitude
-of the value. Default is 0 which results in no drops. See below for accepted variables.
 @item dropamount
-Accepts a non-negative integer, which assigns a variable chance of it being dropped, roughly inverse
+A numeral string, whose value is related to how often packets will be dropped.
-in proportion to the value. Default is 0 which results in no drops. This option is kept for backwards
+Therefore, values below or equal to 0 are forbidden, and the lower the more
-compatibility and is equivalent to setting drop to a negative value with the same magnitude
+frequent packets will be dropped, with 1 meaning every packet is dropped.
-i.e. @code{dropamount=4} is the same as @code{drop=-4}. Ignored if drop is also specified.
 @end table
 
-Both @code{amount} and @code{drop} accept expressions containing the following variables:
+The following example applies the modification to every byte but does not drop
-
+any packets.
-@table @samp
-@item n
-The index of the packet, starting from zero.
-@item tb
-The timebase for packet timestamps.
-@item pts
-Packet presentation timestamp.
-@item dts
-Packet decoding timestamp.
-@item nopts
-Constant representing AV_NOPTS_VALUE.
-@item startpts
-First non-AV_NOPTS_VALUE PTS seen in the stream.
-@item startdts
-First non-AV_NOPTS_VALUE DTS seen in the stream.
-@item duration
-@itemx d
-Packet duration, in timebase units.
-@item pos
-Packet position in input; may be -1 when unknown or not set.
-@item size
-Packet size, in bytes.
-@item key
-Whether packet is marked as a keyframe.
-@item state
-A pseudo random integer, primarily derived from the content of packet payload.
-@end table
-
-@subsection Examples
-Apply modification to every byte but don't drop any packets.
 @example
-ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv
+ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
-@end example
-
-Drop every video packet not marked as a keyframe after timestamp 30s but do not
-modify any of the remaining packets.
-@example
-ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
-@end example
-
-Drop one second of audio every 10 seconds and add some random noise to the rest.
-@example
-ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
 @end example
 
 @section null
 This bitstream filter passes the packets through unchanged.
 
-@section pcm_rechunk
-
-Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
-rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
-filter,ffmpeg-filters} but works on audio packets instead of audio frames.
-
-@table @option
-@item nb_out_samples, n
-Set the number of samples per each output audio packet. The number is intended
-as the number of samples @emph{per each channel}. Default value is 1024.
-
-@item pad, p
-If set to 1, the filter will pad the last audio packet with silence, so that it
-will contain the same number of samples (or roughly the same number of samples,
-see @option{frame_rate}) as the previous ones. Default value is 1.
-
-@item frame_rate, r
-This option makes the filter output a fixed number of packets per second instead
-of a fixed number of samples per packet. If the audio sample rate is not
-divisible by the frame rate then the number of samples will not be constant but
-will vary slightly so that each packet will start as close to the frame
-boundary as possible. Using this option has precedence over @option{nb_out_samples}.
-@end table
-
-You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
-for NTSC frame rate using the @option{frame_rate} option.
-@example
-ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
-@end example
-
-@section pgs_frame_merge
-
-Merge a sequence of PGS Subtitle segments ending with an "end of display set"
-segment into a single packet.
-
-This is required by some containers that support PGS subtitles
-(muxer @code{matroska}).
-
 @section prores_metadata
 
 Modify color property metadata embedded in prores stream.
@@ -741,10 +585,6 @@ Keep the same transfer characteristics property (default).
 @item unknown
 @item bt709
 BT 601, BT 709, BT 2020
-@item smpte2084
-SMPTE ST 2084
-@item arib-std-b67
-ARIB STD-B67
 @end table
 
 
@@ -754,7 +594,7 @@ Available values are:
 
 @table @samp
 @item auto
-Keep the same colorspace property (default).
+Keep the same transfer characteristics property (default).
 
 @item unknown
 @item bt709
@@ -770,11 +610,6 @@ Set Rec709 colorspace for each frame of the file
 ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
 @end example
 
-Set Hybrid Log-Gamma parameters for each frame of the file
-@example
-ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
-@end example
-
 @section remove_extra
 
 Remove extradata from packets.
@@ -797,91 +632,6 @@ Remove extradata from all frames.
 @end table
 @end table
 
-@section setts
-Set PTS and DTS in packets.
-
-It accepts the following parameters:
-@table @option
-@item ts
-@item pts
-@item dts
-Set expressions for PTS, DTS or both.
-@item duration
-Set expression for duration.
-@item time_base
-Set output time base.
-@end table
-
-The expressions are evaluated through the eval API and can contain the following
-constants:
-
-@table @option
-@item N
-The count of the input packet. Starting from 0.
-
-@item TS
-The demux timestamp in input in case of @code{ts} or @code{dts} option or presentation
-timestamp in case of @code{pts} option.
-
-@item POS
-The original position in the file of the packet, or undefined if undefined
-for the current packet
-
-@item DTS
-The demux timestamp in input.
-
-@item PTS
-The presentation timestamp in input.
-
-@item DURATION
-The duration in input.
-
-@item STARTDTS
-The DTS of the first packet.
-
-@item STARTPTS
-The PTS of the first packet.
-
-@item PREV_INDTS
-The previous input DTS.
-
-@item PREV_INPTS
-The previous input PTS.
-
-@item PREV_INDURATION
-The previous input duration.
-
-@item PREV_OUTDTS
-The previous output DTS.
-
-@item PREV_OUTPTS
-The previous output PTS.
-
-@item PREV_OUTDURATION
-The previous output duration.
-
-@item NEXT_DTS
-The next input DTS.
-
-@item NEXT_PTS
-The next input PTS.
-
-@item NEXT_DURATION
-The next input duration.
-
-@item TB
-The timebase of stream packet belongs.
-
-@item TB_OUT
-The output timebase.
-
-@item SR
-The sample rate of stream packet belongs.
-
-@item NOPTS
-The AV_NOPTS_VALUE constant.
-@end table
-
 @anchor{text2movsub}
 @section text2movsub
 
@@ -909,9 +659,7 @@ Modify metadata embedded in a VP9 stream.
 
 @table @option
 @item color_space
-Set the color space value in the frame header.  Note that any frame
+Set the color space value in the frame header.
-set to RGB will be implicitly set to PC range and that RGB is
-incompatible with profiles 0 and 2.
 @table @samp
 @item unknown
 @item bt601
@@ -923,8 +671,8 @@ incompatible with profiles 0 and 2.
 @end table
 
 @item color_range
-Set the color range value in the frame header.  Note that any value
+Set the color range value in the frame header.  Note that this cannot
-imposed by the color space will take precedence over this value.
+be set in RGB streams.
 @table @samp
 @item tv
 @item pc

doc/build_system.txt

@@ -48,8 +48,6 @@ config
 tools/target_dec_<decoder>_fuzzer
     Build fuzzer to fuzz the specified decoder.
 
-tools/target_bsf_<filter>_fuzzer
-    Build fuzzer to fuzz the specified bitstream filter.
 
 Useful standard make commands:
 make -t <target>

doc/codecs.texi

@@ -50,10 +50,11 @@ Use internal 2pass ratecontrol in first pass mode.
 Use internal 2pass ratecontrol in second pass mode.
 @item gray
 Only decode/encode grayscale.
+@item emu_edge
+Do not draw edges.
 @item psnr
 Set error[?] variables during encoding.
 @item truncated
-Input bitstream might be randomly truncated.
 @item drop_changed
 Don't output frames whose parameters differ from first decoded frame in stream.
 Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
@@ -70,14 +71,50 @@ This ensures that file and data checksums are reproducible and match between
 platforms. Its primary use is for regression testing.
 @item aic
 Apply H263 advanced intra coding / mpeg4 ac prediction.
+@item cbp
+Deprecated, use mpegvideo private options instead.
+@item qprd
+Deprecated, use mpegvideo private options instead.
 @item ilme
 Apply interlaced motion estimation.
 @item cgop
 Use closed gop.
-@item output_corrupt
-Output even potentially corrupted frames.
 @end table
 
+@item me_method @var{integer} (@emph{encoding,video})
+Set motion estimation method.
+
+Possible values:
+@table @samp
+@item zero
+zero motion estimation (fastest)
+@item full
+full motion estimation (slowest)
+@item epzs
+EPZS motion estimation (default)
+@item esa
+esa motion estimation (alias for full)
+@item tesa
+tesa motion estimation
+@item dia
+dia motion estimation (alias for epzs)
+@item log
+log motion estimation
+@item phods
+phods motion estimation
+@item x1
+X1 motion estimation
+@item hex
+hex motion estimation
+@item umh
+umh motion estimation
+@item iter
+iter motion estimation
+@end table
+
+@item extradata_size @var{integer}
+Set extradata size.
+
 @item time_base @var{rational number}
 Set codec time base.
 
@@ -144,6 +181,24 @@ Default value is 0.
 @item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
 
+@item rc_strategy @var{integer} (@emph{encoding,video})
+Set ratecontrol method.
+
+@item b_strategy @var{integer} (@emph{encoding,video})
+Set strategy to choose between I/P/B-frames.
+
+@item ps @var{integer} (@emph{encoding,video})
+Set RTP payload size in bytes.
+
+@item mv_bits @var{integer}
+@item header_bits @var{integer}
+@item i_tex_bits @var{integer}
+@item p_tex_bits @var{integer}
+@item i_count @var{integer}
+@item p_count @var{integer}
+@item skip_count @var{integer}
+@item misc_bits @var{integer}
+@item frame_bits @var{integer}
 @item codec_tag @var{integer}
 @item bug @var{flags} (@emph{decoding,video})
 Workaround not auto detected encoder bugs.
@@ -152,6 +207,8 @@ Possible values:
 @table @samp
 @item autodetect
 
+@item old_msmpeg4
+some old lavc generated msmpeg4v3 files (no autodetection)
 @item xvid_ilace
 Xvid interlacing bug (autodetected if fourcc==XVIX)
 @item ump4
@@ -160,6 +217,8 @@ Xvid interlacing bug (autodetected if fourcc==XVIX)
 padding bug (autodetected)
 @item amv
 
+@item ac_vlc
+illegal vlc bug (autodetected per fourcc)
 @item qpel_chroma
 
 @item std_qpel
@@ -180,6 +239,14 @@ Workaround various bugs in microsoft broken decoders.
 trancated frames
 @end table
 
+@item lelim @var{integer} (@emph{encoding,video})
+Set single coefficient elimination threshold for luminance (negative
+values also consider DC coefficient).
+
+@item celim @var{integer} (@emph{encoding,video})
+Set single coefficient elimination threshold for chrominance (negative
+values also consider dc coefficient)
+
 @item strict @var{integer} (@emph{decoding/encoding,audio,video})
 Specify how strictly to follow the standards.
 
@@ -233,8 +300,29 @@ consider things that a sane encoder should not do as an error
 
 @item block_align @var{integer}
 
+@item mpeg_quant @var{integer} (@emph{encoding,video})
+Use MPEG quantizers instead of H.263.
+
+@item qsquish @var{float} (@emph{encoding,video})
+How to keep quantizer between qmin and qmax (0 = clip, 1 = use
+differentiable function).
+
+@item rc_qmod_amp @var{float} (@emph{encoding,video})
+Set experimental quantizer modulation.
+
+@item rc_qmod_freq @var{integer} (@emph{encoding,video})
+Set experimental quantizer modulation.
+
 @item rc_override_count @var{integer}
 
+@item rc_eq @var{string} (@emph{encoding,video})
+Set rate control equation. When computing the expression, besides the
+standard functions defined in the section 'Expression Evaluation', the
+following functions are available: bits2qp(bits), qp2bits(qp). Also
+the following constants are available: iTex pTex tex mv fCode iCount
+mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex
+avgTex.
+
 @item maxrate @var{integer} (@emph{encoding,audio,video})
 Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
 
@@ -245,12 +333,18 @@ encode. It is of little use elsewise.
 @item bufsize @var{integer} (@emph{encoding,audio,video})
 Set ratecontrol buffer size (in bits).
 
+@item rc_buf_aggressivity @var{float} (@emph{encoding,video})
+Currently useless.
+
 @item i_qfactor @var{float} (@emph{encoding,video})
 Set QP factor between P and I frames.
 
 @item i_qoffset @var{float} (@emph{encoding,video})
 Set QP offset between P and I frames.
 
+@item rc_init_cplx @var{float} (@emph{encoding,video})
+Set initial complexity for 1-pass encoding.
+
 @item dct @var{integer} (@emph{encoding,video})
 Set DCT algorithm.
 
@@ -315,7 +409,11 @@ Automatically pick a IDCT compatible with the simple one
 
 @item simpleneon
 
-@item xvid
+@item simplealpha
+
+@item ipp
+
+@item xvidmmx
 
 @item faani
 floating point AAN IDCT
@@ -338,6 +436,19 @@ favor predicting from the previous frame instead of the current
 
 @item bits_per_coded_sample @var{integer}
 
+@item pred @var{integer} (@emph{encoding,video})
+Set prediction method.
+
+Possible values:
+@table @samp
+@item left
+
+@item plane
+
+@item median
+
+@end table
+
 @item aspect @var{rational number} (@emph{encoding,video})
 Set sample aspect ratio.
 
@@ -532,28 +643,13 @@ noise preserving sum of squared differences
 
 @item dia_size @var{integer} (@emph{encoding,video})
 Set diamond type & size for motion estimation.
-@table @samp
-@item (1024, INT_MAX)
-full motion estimation(slowest)
-@item (768, 1024]
-umh motion estimation
-@item (512, 768]
-hex motion estimation
-@item (256, 512]
-l2s diamond motion estimation
-@item [2,256]
-var diamond motion estimation
-@item (-1,  2)
-small diamond motion estimation
-@item -1
-funny diamond motion estimation
-@item (INT_MIN, -1)
-sab diamond motion estimation
-@end table
 
 @item last_pred @var{integer} (@emph{encoding,video})
 Set amount of motion predictors from the previous frame.
 
+@item preme @var{integer} (@emph{encoding,video})
+Set pre motion estimation.
+
 @item precmp @var{integer} (@emph{encoding,video})
 Set pre motion estimation compare function.
 
@@ -597,11 +693,40 @@ Set diamond type & size for motion estimation pre-pass.
 @item subq @var{integer} (@emph{encoding,video})
 Set sub pel motion estimation quality.
 
+@item dtg_active_format @var{integer}
+
 @item me_range @var{integer} (@emph{encoding,video})
 Set limit motion vectors range (1023 for DivX player).
 
+@item ibias @var{integer} (@emph{encoding,video})
+Set intra quant bias.
+
+@item pbias @var{integer} (@emph{encoding,video})
+Set inter quant bias.
+
+@item color_table_id @var{integer}
+
 @item global_quality @var{integer} (@emph{encoding,audio,video})
 
+@item coder @var{integer} (@emph{encoding,video})
+
+Possible values:
+@table @samp
+@item vlc
+variable length coder / huffman coder
+@item ac
+arithmetic coder
+@item raw
+raw (no encoding)
+@item rle
+run-length coder
+@item deflate
+deflate-based coder
+@end table
+
+@item context @var{integer} (@emph{encoding,video})
+Set context model.
+
 @item slice_flags @var{integer}
 
 @item mbd @var{integer} (@emph{encoding,video})
@@ -617,16 +742,32 @@ use fewest bits
 use best rate distortion
 @end table
 
+@item stream_codec_tag @var{integer}
+
+@item sc_threshold @var{integer} (@emph{encoding,video})
+Set scene change threshold.
+
+@item lmin @var{integer} (@emph{encoding,video})
+Set min lagrange factor (VBR).
+
+@item lmax @var{integer} (@emph{encoding,video})
+Set max lagrange factor (VBR).
+
+@item nr @var{integer} (@emph{encoding,video})
+Set noise reduction.
+
 @item rc_init_occupancy @var{integer} (@emph{encoding,video})
 Set number of bits which should be loaded into the rc buffer before
 decoding starts.
 
-@item flags2 @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
+@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
 
 Possible values:
 @table @samp
 @item fast
 Allow non spec compliant speedup tricks.
+@item sgop
+Deprecated, use mpegvideo private options instead.
 @item noout
 Skip bitstream encoding.
 @item ignorecrop
@@ -640,31 +781,12 @@ Show all frames before the first keyframe.
 @item export_mvs
 Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
 for codecs that support it. See also @file{doc/examples/export_mvs.c}.
-@item skip_manual
-Do not skip samples and export skip information as frame side data.
-@item ass_ro_flush_noop
-Do not reset ASS ReadOrder field on flush.
-@item icc_profiles
-Generate/parse embedded ICC profiles from/to colorimetry tags.
 @end table
 
-@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
+@item error @var{integer} (@emph{encoding,video})
 
-Possible values:
+@item qns @var{integer} (@emph{encoding,video})
-@table @samp
+Deprecated, use mpegvideo private options instead.
-@item mvs
-Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
-for codecs that support it. See also @file{doc/examples/export_mvs.c}.
-@item prft
-Export encoder Producer Reference Time into packet side-data (see @code{AV_PKT_DATA_PRFT})
-for codecs that support it.
-@item venc_params
-Export video encoding parameters through frame side data (see @code{AV_FRAME_DATA_VIDEO_ENC_PARAMS})
-for codecs that support it. At present, those are H.264 and VP9.
-@item film_grain
-Export film grain parameters through frame side data (see @code{AV_FRAME_DATA_FILM_GRAIN_PARAMS}).
-Supported at present by AV1 decoders.
-@end table
 
 @item threads @var{integer} (@emph{decoding/encoding,video})
 Set the number of threads to be used, in case the selected codec
@@ -678,6 +800,12 @@ automatically select the number of threads to set
 
 Default value is @samp{auto}.
 
+@item me_threshold @var{integer} (@emph{encoding,video})
+Set motion estimation threshold.
+
+@item mb_threshold @var{integer} (@emph{encoding,video})
+Set macroblock threshold.
+
 @item dc @var{integer} (@emph{encoding,video})
 Set intra_dc_precision.
 
@@ -692,8 +820,49 @@ Set number of macroblock rows at the bottom which are skipped.
 
 @item profile @var{integer} (@emph{encoding,audio,video})
 
-Set encoder codec profile. Default value is @samp{unknown}. Encoder specific
+Possible values:
-profiles are documented in the relevant encoder documentation.
+@table @samp
+@item unknown
+
+@item aac_main
+
+@item aac_low
+
+@item aac_ssr
+
+@item aac_ltp
+
+@item aac_he
+
+@item aac_he_v2
+
+@item aac_ld
+
+@item aac_eld
+
+@item mpeg2_aac_low
+
+@item mpeg2_aac_he
+
+@item mpeg4_sp
+
+@item mpeg4_core
+
+@item mpeg4_main
+
+@item mpeg4_asp
+
+@item dts
+
+@item dts_es
+
+@item dts_96_24
+
+@item dts_hd_hra
+
+@item dts_hd_ma
+
+@end table
 
 @item level @var{integer} (@emph{encoding,audio,video})
 
@@ -706,12 +875,67 @@ Possible values:
 @item lowres @var{integer} (@emph{decoding,audio,video})
 Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
 
+@item skip_threshold @var{integer} (@emph{encoding,video})
+Set frame skip threshold.
+
+@item skip_factor @var{integer} (@emph{encoding,video})
+Set frame skip factor.
+
+@item skip_exp @var{integer} (@emph{encoding,video})
+Set frame skip exponent.
+Negative values behave identical to the corresponding positive ones, except
+that the score is normalized.
+Positive values exist primarily for compatibility reasons and are not so useful.
+
+@item skipcmp @var{integer} (@emph{encoding,video})
+Set frame skip compare function.
+
+Possible values:
+@table @samp
+@item sad
+sum of absolute differences, fast (default)
+@item sse
+sum of squared errors
+@item satd
+sum of absolute Hadamard transformed differences
+@item dct
+sum of absolute DCT transformed differences
+@item psnr
+sum of squared quantization errors (avoid, low quality)
+@item bit
+number of bits needed for the block
+@item rd
+rate distortion optimal, slow
+@item zero
+0
+@item vsad
+sum of absolute vertical differences
+@item vsse
+sum of squared vertical differences
+@item nsse
+noise preserving sum of squared differences
+@item w53
+5/3 wavelet, only used in snow
+@item w97
+9/7 wavelet, only used in snow
+@item dctmax
+
+@item chroma
+
+@end table
+
+@item border_mask @var{float} (@emph{encoding,video})
+Increase the quantizer for macroblocks close to borders.
+
 @item mblmin @var{integer} (@emph{encoding,video})
 Set min macroblock lagrange factor (VBR).
 
 @item mblmax @var{integer} (@emph{encoding,video})
 Set max macroblock lagrange factor (VBR).
 
+@item mepc @var{integer} (@emph{encoding,video})
+Set motion estimation bitrate penalty compensation (1.0 = 256).
+
 @item skip_loop_filter @var{integer} (@emph{decoding,video})
 @item skip_idct        @var{integer} (@emph{decoding,video})
 @item skip_frame       @var{integer} (@emph{decoding,video})
@@ -751,17 +975,34 @@ Default value is @samp{default}.
 @item bidir_refine @var{integer} (@emph{encoding,video})
 Refine the two motion vectors used in bidirectional macroblocks.
 
+@item brd_scale @var{integer} (@emph{encoding,video})
+Downscale frames for dynamic B-frame decision.
+
 @item keyint_min @var{integer} (@emph{encoding,video})
 Set minimum interval between IDR-frames.
 
 @item refs @var{integer} (@emph{encoding,video})
 Set reference frames to consider for motion compensation.
 
+@item chromaoffset @var{integer} (@emph{encoding,video})
+Set chroma qp offset from luma.
+
 @item trellis @var{integer} (@emph{encoding,audio,video})
 Set rate-distortion optimal quantization.
 
 @item mv0_threshold @var{integer} (@emph{encoding,video})
+@item b_sensitivity @var{integer} (@emph{encoding,video})
+Adjust sensitivity of b_frame_strategy 1.
+
 @item compression_level @var{integer} (@emph{encoding,audio,video})
+@item min_prediction_order @var{integer} (@emph{encoding,audio})
+@item max_prediction_order @var{integer} (@emph{encoding,audio})
+@item timecode_frame_start @var{integer} (@emph{encoding,video})
+Set GOP timecode frame start number, in non drop frame format.
+
+@item request_channels @var{integer} (@emph{decoding,audio})
+Set desired number of audio channels.
+
 @item bits_per_raw_sample @var{integer}
 @item channel_layout @var{integer} (@emph{decoding/encoding,audio})
 
@@ -875,12 +1116,6 @@ BT.2020 NCL
 BT.2020 CL
 @item smpte2085
 SMPTE 2085
-@item chroma-derived-nc
-Chroma-derived NCL
-@item chroma-derived-c
-Chroma-derived CL
-@item ictcp
-ICtCp
 @end table
 
 @item color_range @var{integer} (@emph{decoding/encoding,video})

doc/decoders.texi

@@ -25,19 +25,6 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.
 
-@section av1
-
-AOMedia Video 1 (AV1) decoder.
-
-@subsection Options
-
-@table @option
-
-@item operating_point
-Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.
-
-@end table
-
 @section rawvideo
 
 Raw video decoder.
@@ -70,36 +57,19 @@ You need to explicitly configure the build with @code{--enable-libdav1d}.
 
 @subsection Options
 
-The following options are supported by the libdav1d wrapper.
+The following option is supported by the libdav1d wrapper.
 
 @table @option
 
 @item framethreads
 Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
-This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
-option @code{max_frame_delay} and the global option @code{threads} instead.
 
 @item tilethreads
 Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
-This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
-global option @code{threads} instead.
-
-@item max_frame_delay
-Set max amount of frames the decoder may buffer internally. The default value is 0
-(autodetect).
 
 @item filmgrain
-Apply film grain to the decoded video if present in the bitstream. Defaults to the
+Apply film grain to the decoded video if present in the bitstream. The default value
-internal default of the library.
+is true.
-This option is deprecated and will be removed in the future. See the global option
-@code{export_side_data} to export Film Grain parameters instead of applying it.
-
-@item oppoint
-Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
-internal default of the library.
-
-@item alllayers
-Output all spatial layers of a scalable AV1 bitstream. The default value is false.
 
 @end table
 
@@ -111,84 +81,6 @@ This decoder allows libavcodec to decode AVS2 streams with davs2 library.
 
 @c man end VIDEO DECODERS
 
-@section libuavs3d
-
-AVS3-P2/IEEE1857.10 video decoder.
-
-libuavs3d allows libavcodec to decode AVS3 streams.
-Requires the presence of the libuavs3d headers and library during configuration.
-You need to explicitly configure the build with @code{--enable-libuavs3d}.
-
-@subsection Options
-
-The following option is supported by the libuavs3d wrapper.
-
-@table @option
-
-@item frame_threads
-Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
-
-@end table
-
-@section QSV Decoders
-
-The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC,
-JPEG/MJPEG, VP8, VP9, AV1).
-
-@subsection Common Options
-
-The following options are supported by all qsv decoders.
-
-@table @option
-
-@item @var{async_depth}
-Internal parallelization depth, the higher the value the higher the latency.
-
-@item @var{gpu_copy}
-A GPU-accelerated copy between video and system memory
-@table @samp
-@item default
-@item on
-@item off
-@end table
-
-@end table
-
-@subsection HEVC Options
-Extra options for hevc_qsv.
-
-@table @option
-
-@item @var{load_plugin}
-A user plugin to load in an internal session
-@table @samp
-@item none
-@item hevc_sw
-@item hevc_hw
-@end table
-
-@item @var{load_plugins}
-A :-separate list of hexadecimal plugin UIDs to load in an internal session
-
-@end table
-
-@section v210
-
-Uncompressed 4:2:2 10-bit decoder.
-
-@subsection Options
-
-@table @option
-
-@item custom_stride
-Set the line size of the v210 data in bytes. The default value is 0
-(autodetect). You can use the special -1 value for a strideless v210 as seen in
-BOXX files.
-
-@end table
-
-@c man end VIDEO DECODERS
-
 @chapter Audio Decoders
 @c man begin AUDIO DECODERS
 
@@ -208,7 +100,7 @@ the undocumented RealAudio 3 (a.k.a. dnet).
 
 @item -drc_scale @var{value}
 Dynamic Range Scale Factor. The factor to apply to dynamic range values
-from the AC-3 stream. This factor is applied exponentially. The default value is 1.
+from the AC-3 stream. This factor is applied exponentially.
 There are 3 notable scale factor ranges:
 @table @option
 @item drc_scale == 0
@@ -360,8 +252,6 @@ Enabled by default.
 @table @option
 @item compute_clut
 @table @option
-@item -2
-Compute clut once if no matching CLUT is in the stream.
 @item -1
 Compute clut if no matching CLUT is in the stream.
 @item 0
@@ -390,7 +280,7 @@ palette is stored in the IFO file, and therefore not available when reading
 from dumped VOB files.
 
 The format for this option is a string containing 16 24-bits hexadecimal
-numbers (without 0x prefix) separated by commas, for example @code{0d00ee,
+numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
 ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
 
@@ -419,11 +309,6 @@ List of teletext page numbers to decode. Pages that do not match the specified
 list are dropped. You may use the special @code{*} string to match all pages,
 or @code{subtitle} to match all subtitle pages.
 Default value is *.
-@item txt_default_region
-Set default character set used for decoding, a value between 0 and 87 (see
-ETS 300 706, Section 15, Table 32). Default value is -1, which does not
-override the libzvbi default. This option is needed for some legacy level 1.0
-transmissions which cannot signal the proper charset.
 @item txt_chop_top
 Discards the top teletext line. Default value is 1.
 @item txt_format

doc/demuxers.texi

@@ -25,13 +25,6 @@ Audible Format 2, 3, and 4 demuxer.
 
 This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
 
-@section aac
-
-Raw Audio Data Transport Stream AAC demuxer.
-
-This demuxer is used to demux an ADTS input containing a single AAC stream
-alongwith any ID3v1/2 or APE tags in it.
-
 @section apng
 
 Animated Portable Network Graphics demuxer.
@@ -44,15 +37,12 @@ between the last fcTL and IEND chunks.
 
 @table @option
 @item -ignore_loop @var{bool}
-Ignore the loop variable in the file if set. Default is enabled.
+Ignore the loop variable in the file if set.
-
 @item -max_fps @var{int}
-Maximum framerate in frames per second. Default of 0 imposes no limit.
+Maximum framerate in frames per second (0 for no limit).
-
 @item -default_fps @var{int}
 Default framerate in frames per second when none is specified in the file
-(0 meaning as fast as possible). Default is 15.
+(0 meaning as fast as possible).
-
 @end table
 
 @section asf
@@ -103,7 +93,8 @@ backslash or single quotes.
 All subsequent file-related directives apply to that file.
 
 @item @code{ffconcat version 1.0}
-Identify the script type and version.
+Identify the script type and version. It also sets the @option{safe} option
+to 1 if it was -1.
 
 To make FFmpeg recognize the format automatically, this directive must
 appear exactly as is (no extra space or byte-order-mark) on the very first
@@ -157,16 +148,6 @@ directive) will be reduced based on their specified Out point.
 Metadata of the packets of the file. The specified metadata will be set for
 each file packet. You can specify this directive multiple times to add multiple
 metadata entries.
-This directive is deprecated, use @code{file_packet_meta} instead.
-
-@item @code{file_packet_meta @var{key} @var{value}}
-Metadata of the packets of the file. The specified metadata will be set for
-each file packet. You can specify this directive multiple times to add multiple
-metadata entries.
-
-@item @code{option @var{key} @var{value}}
-Option to access, open and probe the file.
-Can be present multiple times.
 
 @item @code{stream}
 Introduce a stream in the virtual file.
@@ -184,20 +165,6 @@ subfiles will be used.
 This is especially useful for MPEG-PS (VOB) files, where the order of the
 streams is not reliable.
 
-@item @code{stream_meta @var{key} @var{value}}
-Metadata for the stream.
-Can be present multiple times.
-
-@item @code{stream_codec @var{value}}
-Codec for the stream.
-
-@item @code{stream_extradata @var{hex_string}}
-Extradata for the string, encoded in hexadecimal.
-
-@item @code{chapter @var{id} @var{start} @var{end}}
-Add a chapter. @var{id} is an unique identifier, possibly small and
-consecutive.
-
 @end table
 
 @subsection Options
@@ -207,8 +174,7 @@ This demuxer accepts the following option:
 @table @option
 
 @item safe
-If set to 1, reject unsafe file paths and directives.
+If set to 1, reject unsafe file paths. A file path is considered safe if it
-A file path is considered safe if it
 does not contain a protocol specification and is relative and all components
 only contain characters from the portable character set (letters, digits,
 period, underscore and hyphen) and have no period at the beginning of a
@@ -218,6 +184,9 @@ If set to 0, any file name is accepted.
 
 The default is 1.
 
+-1 is equivalent to 1 if the format was automatically
+probed and 0 otherwise.
+
 @item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
@@ -274,47 +243,11 @@ 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.
 
-@subsection Options
+@section flv, live_flv
-
-This demuxer accepts the following option:
-
-@table @option
-
-@item cenc_decryption_key
-16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
-
-@end table
-
-@section ea
-
-Electronic Arts Multimedia format demuxer.
-
-This format is used by various Electronic Arts games.
-
-@subsection Options
-
-@table @option
-
-@item merge_alpha @var{bool}
-
-Normally the VP6 alpha channel (if exists) is returned as a secondary video
-stream, by setting this option you can make the demuxer return a single video
-stream which contains the alpha channel in addition to the ordinary video.
-
-@end table
-
-@section imf
-
-Interoperable Master Format demuxer.
-
-This demuxer presents audio and video streams found in an IMF Composition.
-
-@section flv, live_flv, kux
 
 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.
-KUX is a flv variant used on the Youku platform.
 
 @example
 ffmpeg -f flv -i myfile.flv ...
@@ -391,9 +324,6 @@ It accepts the following options:
 @item live_start_index
 segment index to start live streams at (negative values are from the end).
 
-@item prefer_x_start
-prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
-
 @item allowed_extensions
 ',' separated list of file extensions that hls is allowed to access.
 
@@ -401,10 +331,6 @@ prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
 Maximum number of times a insufficient list is attempted to be reloaded.
 Default value is 1000.
 
-@item m3u8_hold_counters
-The maximum number of times to load m3u8 when it refreshes without new segments.
-Default value is 1000.
-
 @item http_persistent
 Use persistent HTTP connections. Applicable only for HTTP streams.
 Enabled by default.
@@ -412,17 +338,6 @@ Enabled by default.
 @item http_multiple
 Use multiple HTTP connections for downloading HTTP segments.
 Enabled by default for HTTP/1.1 servers.
-
-@item http_seekable
-Use HTTP partial requests for downloading HTTP segments.
-0 = disable, 1 = enable, -1 = auto, Default is auto.
-
-@item seg_format_options
-Set options for the demuxer of media segments using a list of key=value pairs separated by @code{:}.
-
-@item seg_max_retry
-Maximum number of times to reload a segment on error, useful when segment skip on network error is not desired.
-Default value is 0.
 @end table
 
 @section image2
@@ -533,17 +448,6 @@ nanosecond precision.
 @item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
-@item export_path_metadata
-If set to 1, will add two extra fields to the metadata found in input, making them
-also available for other filters (see @var{drawtext} filter for examples). Default
-value is 0. The extra fields are described below:
-@table @option
-@item lavf.image2dec.source_path
-Corresponds to the full path to the input file being read.
-@item lavf.image2dec.source_basename
-Corresponds to the name of the file being read.
-@end table
-
 @end table
 
 @subsection Examples
@@ -681,13 +585,9 @@ 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
+@section mov/mp4/3gp/QuickTime
 
-Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
+QuickTime / MP4 demuxer.
-
-Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v
-
-@subsection Options
 
 This demuxer accepts the following options:
 @table @option
@@ -698,88 +598,10 @@ Enabling this can theoretically leak information in some use cases.
 @item use_absolute_path
 Allows loading of external tracks via absolute paths, disabled by default.
 Enabling this poses a security risk. It should only be enabled if the source
-is known to be non-malicious.
+is known to be non malicious.
 
-@item seek_streams_individually
-When seeking, identify the closest point in each stream individually and demux packets in
-that stream from identified point. This can lead to a different sequence of packets compared
-to demuxing linearly from the beginning. Default is true.
-
-@item ignore_editlist
-Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the
-timeline described by the edit list. Default is false.
-
-@item advanced_editlist
-Modify the stream index to reflect the timeline described by the edit list. @code{ignore_editlist}
-must be set to false for this option to be effective.
-If both @code{ignore_editlist} and this option are set to false, then only the
-start of the stream index is modified to reflect initial dwell time or starting timestamp
-described by the edit list. Default is true.
-
-@item ignore_chapters
-Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are
-only parsed when input is seekable. Default is false.
-
-@item use_mfra_for
-For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.
-
-Following options are available:
-@table @samp
-@item auto
-Auto-detect whether to set mfra timestamps as PTS or DTS @emph{(default)}
-
-@item dts
-Set mfra timestamps as DTS
-
-@item pts
-Set mfra timestamps as PTS
-
-@item 0
-Don't use mfra box to set timestamps
 @end table
 
-@item use_tfdt
-For fragmented input, set fragment's starting timestamp to @code{baseMediaDecodeTime} from the @code{tfdt} box.
-Default is enabled, which will prefer to use the @code{tfdt} box to set DTS. Disable to use the @code{earliest_presentation_time} from the @code{sidx} box.
-In either case, the timestamp from the @code{mfra} box will be used if it's available and @code{use_mfra_for} is
-set to pts or dts.
-
-@item export_all
-Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
-characters of the box type are set as the key. Default is false.
-
-@item export_xmp
-Export entire contents of @var{XMP_} box and @var{uuid} box as a string with key @code{xmp}. Note that
-if @code{export_all} is set and this option isn't, the contents of @var{XMP_} box are still exported
-but with key @code{XMP_}. Default is false.
-
-@item activation_bytes
-4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
-
-@item audible_fixed_key
-Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
-specify.
-
-@item decryption_key
-16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
-
-@item max_stts_delta
-Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in
-error or used to store a negative value for dts correction when treated as signed 32-bit integers. This option lets
-the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when
-cast to int32 are used to adjust onward dts.
-
-Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows upto
-a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
-@end table
-
-@subsection Audible AAX
-
-Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
-@example
-ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
-@end example
-
 @section mpegts
 
 MPEG-2 transport stream demuxer.
@@ -811,10 +633,6 @@ disabled). Default value is -1.
 @item merge_pmt_versions
 Re-use existing streams when a PMT's version is updated and elementary
 streams move to different PIDs. Default value is 0.
-
-@item max_packet_size
-Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size
-are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.
 @end table
 
 @section mpjpeg

doc/dev_community/community.md

@@ -1,82 +0,0 @@
-# FFmpeg project
-
-## Organisation
-
-The FFmpeg project is organized through a community working on global consensus.
-
-Decisions are taken by the ensemble of active members, through voting and
-are aided by two committees.
-
-## General Assembly
-
-The ensemble of active members is called the General Assembly (GA).
-
-The General Assembly is sovereign and legitimate for all its decisions
-regarding the FFmpeg project.
-
-The General Assembly is made up of active contributors.
-
-Contributors are considered "active contributors" if they have pushed more
-than 20 patches in the last 36 months in the main FFmpeg repository, or
-if they have been voted in by the GA.
-
-Additional members are added to the General Assembly through a vote after
-proposal by a member of the General Assembly.
-They are part of the GA for two years, after which they need a confirmation by
-the GA.
-
-A script to generate the current members of the general assembly (minus members
-voted in) can be found in `tools/general_assembly.pl`.
-
-## Voting
-
-Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
-
-Majority vote means more than 50% of the expressed ballots.
-
-## Technical Committee
-
-The Technical Committee (TC) is here to arbitrate and make decisions when
-technical conflicts occur in the project.
-They will consider the merits of all the positions, judge them and make a
-decision.
-
-The TC resolves technical conflicts but is not a technical steering committee.
-
-Decisions by the TC are binding for all the contributors.
-
-Decisions made by the TC can be re-opened after 1 year or by a majority vote
-of the General Assembly, requested by one of the member of the GA.
-
-The TC is elected by the General Assembly for a duration of 1 year, and
-is composed of 5 members.
-Members can be re-elected if they wish. A majority vote in the General Assembly
-can trigger a new election of the TC.
-
-The members of the TC can be elected from outside of the GA.
-Candidates for election can either be suggested or self-nominated.
-
-The conflict resolution process is detailed in the [resolution process](resolution_process.md) document.
-
-## Community committee
-
-The Community Committee (CC) is here to arbitrage and make decisions when
-inter-personal conflicts occur in the project. It will decide quickly and
-take actions, for the sake of the project.
-
-The CC can remove privileges of offending members, including removal of
-commit access and temporary ban from the community.
-
-Decisions made by the CC can be re-opened after 1 year or by a majority vote
-of the General Assembly. Indefinite bans from the community must be confirmed
-by the General Assembly, in a majority vote.
-
-The CC is elected by the General Assembly for a duration of 1 year, and is
-composed of 5 members.
-Members can be re-elected if they wish. A majority vote in the General Assembly
-can trigger a new election of the CC.
-
-The members of the CC can be elected from outside of the GA.
-Candidates for election can either be suggested or self-nominated.
-
-The CC is governed by and responsible for enforcing the Code of Conduct.

doc/dev_community/resolution_process.md

@@ -1,91 +0,0 @@
-# Technical Committee
-
-_This document only makes sense with the rules from [the community document](community)_.
-
-The Technical Committee (**TC**) is here to arbitrate and make decisions when
-technical conflicts occur in the project.
-
-The TC main role is to resolve technical conflicts.
-It is therefore not a technical steering committee, but it is understood that
-some decisions might impact the future of the project.
-
-# Process
-
-## Seizing
-
-The TC can take possession of any technical matter that it sees fit.
-
-To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
-
-As members of TC are developers, they also can email tc@ to raise an issue.
-
-## Announcement
-
-The TC, once seized, must announce itself on the main mailing list, with a _[TC]_ tag.
-
-The TC has 2 modes of operation: a RFC one and an internal one.
-
-If the TC thinks it needs the input from the larger community, the TC can call
-for a RFC. Else, it can decide by itself.
-
-If the disagreement involves a member of the TC, that member should recuse
-themselves from the decision.
-
-The decision to use a RFC process or an internal discussion is a discretionary
-decision of the TC.
-
-The TC can also reject a seizure for a few reasons such as:
-the matter was not discussed enough previously; it lacks expertise to reach a
-beneficial decision on the matter; or the matter is too trivial.
-
-### RFC call
-
-In the RFC mode, one person from the TC posts on the mailing list the
-technical question and will request input from the community.
-
-The mail will have the following specification:
-* a precise title
-* a specific tag [TC RFC]
-* a top-level email
-* contain a precise question that does not exceed 100 words and that is answerable by developers
-* may have an extra description, or a link to a previous discussion, if deemed necessary,
-* contain a precise end date for the answers.
-
-The answers from the community must be on the main mailing list and must have
-the following specification:
-* keep the tag and the title unchanged
-* limited to 400 words
-* a first-level, answering directly to the main email
-* answering to the question.
-
-Further replies to answers are permitted, as long as they conform to the
-community standards of politeness, they are limited to 100 words, and are not
-nested more than once. (max-depth=2)
-
-After the end-date, mails on the thread will be ignored.
-
-Violations of those rules will be escalated through the Community Committee.
-
-After all the emails are in, the TC has 96 hours to give its final decision.
-Exceptionally, the TC can request an extra delay, that will be notified on the
-mailing list.
-
-### Within TC
-
-In the internal case, the TC has 96 hours to give its final decision.
-Exceptionally, the TC can request an extra delay.
-
-
-## Decisions
-
-The decisions from the TC will be sent on the mailing list, with the _[TC]_ tag.
-
-Internally, the TC should take decisions with a majority, or using
-ranked-choice voting.
-
-The decision from the TC should be published with a summary of the reasons that
-lead to this decision.
-
-The decisions from the TC are final, until the matters are reopened after
-no less than one year.
-

doc/developer.texi

@@ -10,79 +10,41 @@
 
 @contents
 
-@chapter Introduction
+@chapter Notes for external developers
 
-This text is concerned with the development @emph{of} FFmpeg itself. Information
+This document is mostly useful for internal FFmpeg developers.
-on using the FFmpeg libraries in other programs can be found elsewhere, e.g. in:
+External developers who need to use the API in their application should
-@itemize @bullet
+refer to the API doxygen documentation in the public headers, and
-@item
+check the examples in @file{doc/examples} and in the source code to
-the installed header files
+see how the public API is employed.
-@item
-@url{http://ffmpeg.org/doxygen/trunk/index.html, the Doxygen documentation}
-generated from the headers
-@item
-the examples under @file{doc/examples}
-@end itemize
 
-If you modify FFmpeg code for your own use case, you are highly encouraged to
+You can use the FFmpeg libraries in your commercial program, but you
-@emph{submit your changes back to us}, using this document as a guide. There are
+are encouraged to @emph{publish any patch you make}. In this case the
-both pragmatic and ideological reasons to do so:
+best way to proceed is to send your patches to the ffmpeg-devel
-@itemize @bullet
+mailing list following the guidelines illustrated in the remainder of
-@item
+this document.
-Maintaining external changes to keep up with upstream development is
-time-consuming and error-prone. With your code in the main tree, it will be
-maintained by FFmpeg developers.
-@item
-FFmpeg developers include leading experts in the field who can find bugs or
-design flaws in your code.
-@item
-By supporting the project you find useful you ensure it continues to be
-maintained and developed.
-@end itemize
 
 For more detailed legal information about the use of FFmpeg in
 external programs read the @file{LICENSE} file in the source tree and
 consult @url{https://ffmpeg.org/legal.html}.
 
-@section Contributing code
+@chapter Contributing
 
-All proposed code changes should be submitted for review to
+There are 2 ways by which code gets into FFmpeg:
-@url{mailto:ffmpeg-devel@@ffmpeg.org, the development mailing list}, as
+@itemize @bullet
-described in more detail in the @ref{Submitting patches} chapter. The code
+@item Submitting patches to the ffmpeg-devel mailing list.
-should comply with the @ref{Development Policy} and follow the @ref{Coding Rules}.
+      See @ref{Submitting patches} for details.
+@item Directly committing changes to the main tree.
+@end itemize
+
+Whichever way, changes should be reviewed by the maintainer of the code
+before they are committed. And they should follow the @ref{Coding Rules}.
 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 C language features
-
-FFmpeg is programmed in the ISO C99 language, extended with:
-@itemize @bullet
-@item
-Atomic operations from C11 @file{stdatomic.h}. They are emulated on
-architectures/compilers that do not support them, so all FFmpeg-internal code
-may use atomics without any extra checks. However, @file{stdatomic.h} must not
-be included in public headers, so they stay C99-compatible.
-@end itemize
-
-Compiler-specific extensions may be used with good reason, but must not be
-depended on, i.e. the code must still compile and work with compilers lacking
-the extension.
-
-The following C99 features must not be used anywhere in the codebase:
-@itemize @bullet
-@item
-variable-length arrays;
-
-@item
-complex numbers;
-
-@item
-mixed statements and declarations.
-@end itemize
-
 @section Code formatting conventions
 
 There are the following guidelines regarding the indentation in files:
@@ -105,39 +67,8 @@ K&R coding style is used.
 @end itemize
 The presentation is one inspired by 'indent -i4 -kr -nut'.
 
-@subsection Vim configuration
+The main priority in FFmpeg is simplicity and small code size in order to
-In order to configure Vim to follow FFmpeg formatting conventions, paste
+minimize the bug count.
-the following snippet into your @file{.vimrc}:
-@example
-" indentation rules for FFmpeg: 4 spaces, no tabs
-set expandtab
-set shiftwidth=4
-set softtabstop=4
-set cindent
-set cinoptions=(0
-" Allow tabs in Makefiles.
-autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
-" Trailing whitespace and tabs are forbidden, so highlight them.
-highlight ForbiddenWhitespace ctermbg=red guibg=red
-match ForbiddenWhitespace /\s\+$\|\t/
-" Do not highlight spaces at the end of line while typing on that line.
-autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
-@end example
-
-@subsection Emacs configuration
-For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
-@lisp
-(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode . nil)
-               (show-trailing-whitespace . t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
-@end lisp
 
 @section Comments
 Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
@@ -179,51 +110,89 @@ int myfunc(int my_parameter)
 ...
 @end example
 
-@section Naming conventions
+@section C language features
 
-Names of functions, variables, and struct members must be lowercase, using
+FFmpeg is programmed in the ISO C90 language with a few additional
-underscores (_) to separate words. For example, @samp{avfilter_get_video_buffer}
+features from ISO C99, namely:
-is an acceptable function name and @samp{AVFilterGetVideo} is not.
 
-Struct, union, enum, and typedeffed type names must use CamelCase. All structs
-and unions should be typedeffed to the same name as the struct/union tag, e.g.
-@code{typedef struct AVFoo @{ ... @} AVFoo;}. Enums are typically not
-typedeffed.
-
-Enumeration constants and macros must be UPPERCASE, except for macros
-masquerading as functions, which should use the function naming convention.
-
-All identifiers in the libraries should be namespaced as follows:
 @itemize @bullet
 @item
-No namespacing for identifiers with file and lower scope (e.g. local variables,
+the @samp{inline} keyword;
-static functions), and struct and union members,
 
 @item
-The @code{ff_} prefix must be used for variables and functions visible outside
+@samp{//} comments;
-of file scope, but only used internally within a single library, e.g.
+
-@samp{ff_w64_demuxer}. This prevents name collisions when FFmpeg is statically
+@item
-linked.
+designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
+
+@item
+compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
+
+@item
+for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
+
+@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
+accept patches to remove their use unless they absolutely do not impair
+clarity and performance.
+
+All code must compile with recent versions of GCC and a number of other
+currently supported compilers. To ensure compatibility, please do not use
+additional C99 features or GCC extensions. Especially watch out for:
+
+@itemize @bullet
+@item
+mixing statements and declarations;
+
+@item
+@samp{long long} (use @samp{int64_t} instead);
+
+@item
+@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
+
+@item
+GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
+@end itemize
+
+@section 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
+for example structs and enums; they should always be in CamelCase.
+
+There are the following conventions for naming variables and functions:
+
+@itemize @bullet
+@item
+For local variables no prefix is required.
+
+@item
+For file-scope variables and functions declared as @code{static}, no prefix
+is required.
+
+@item
+For variables and functions visible outside of file scope, but only used
+internally by a library, an @code{ff_} prefix should be used,
+e.g. @samp{ff_w64_demuxer}.
 
 @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}.
 
-@item
-All other internal identifiers, like private type or macro names, should be
-namespaced only to avoid possible internal conflicts. E.g. @code{H264_NAL_SPS}
-vs. @code{HEVC_NAL_SPS}.
-
 @item
 Each library has its own prefix for public symbols, in addition to the
 commonly used @code{av_} (@code{avformat_} for libavformat,
 @code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
 Check the existing code and choose names accordingly.
-
+Note that some symbols without these prefixes are also exported for
-@item
+retro-compatibility reasons. These exceptions are declared in the
-Other public identifiers (struct, union, enum, macro, type names) must use their
+@code{lib<name>/lib<name>.v} files.
-library's public prefix (@code{AV}, @code{Sws}, or @code{Swr}).
 @end itemize
 
 Furthermore, name space reserved for the system should not be invaded.
@@ -246,7 +215,39 @@ 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
 
-@anchor{Development Policy}
+@section Editor configuration
+In order to configure Vim to follow FFmpeg formatting conventions, paste
+the following snippet into your @file{.vimrc}:
+@example
+" indentation rules for FFmpeg: 4 spaces, no tabs
+set expandtab
+set shiftwidth=4
+set softtabstop=4
+set cindent
+set cinoptions=(0
+" Allow tabs in Makefiles.
+autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+" Trailing whitespace and tabs are forbidden, so highlight them.
+highlight ForbiddenWhitespace ctermbg=red guibg=red
+match ForbiddenWhitespace /\s\+$\|\t/
+" Do not highlight spaces at the end of line while typing on that line.
+autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
+@end example
+
+For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
+@lisp
+(c-add-style "ffmpeg"
+             '("k&r"
+               (c-basic-offset . 4)
+               (indent-tabs-mode . nil)
+               (show-trailing-whitespace . t)
+               (c-offsets-alist
+                (statement-cont . (c-lineup-assignments +)))
+               )
+             )
+(setq c-default-style "ffmpeg")
+@end lisp
+
 @chapter Development Policy
 
 @section Patches/Committing
@@ -490,22 +491,6 @@ 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.
 
-@subheading Sending patches from email clients
-Using @code{git send-email} might not be desirable for everyone. The
-following trick allows to send patches via email clients in a safe
-way. It has been tested with Outlook and Thunderbird (with X-Unsent
-extension) and might work with other applications.
-
-Create your patch like this:
-
-@verbatim
-git format-patch -s -o "outputfolder" --add-header "X-Unsent: 1" --suffix .eml --to ffmpeg-devel@ffmpeg.org -1 1a2b3c4d
-@end verbatim
-
-Now you'll just need to open the eml file with the email application
-and execute 'Send'.
-
-@subheading Reviews
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -637,7 +622,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
 If the patch fixes a bug, did you provide enough information, including
 a sample, so the bug can be reproduced and the fix can be verified?
 Note please do not attach samples >100k to mails but rather provide a
-URL, you can upload to @url{https://streams.videolan.org/upload/}.
+URL, you can upload to ftp://upload.ffmpeg.org.
 
 @item
 Did you provide a verbose summary about what the patch does change?

doc/doxy-wrapper.sh

@@ -1,13 +1,10 @@
 #!/bin/sh
 
 OUT_DIR="${1}"
-SRC_DIR="${2}"
+DOXYFILE="${2}"
-DOXYFILE="${3}"
+DOXYGEN="${3}"
-DOXYGEN="${4}"
 
-shift 4
+shift 3
-
-cd ${SRC_DIR}
 
 if [ -e "VERSION" ]; then
     VERSION=`cat "VERSION"`

doc/examples/.gitignore

@@ -1,4 +1,4 @@
-/avio_list_dir
+/avio_dir_cmd
 /avio_reading
 /decode_audio
 /decode_video
@@ -22,4 +22,3 @@
 /transcoding
 /vaapi_encode
 /vaapi_transcode
-/qsv_transcode

doc/examples/Makefile

@@ -1,27 +1,26 @@
-EXAMPLES-$(CONFIG_AVIO_HTTP_SERVE_FILES)     += avio_http_serve_files
+EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
-EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
+EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-EXAMPLES-$(CONFIG_AVIO_READ_CALLBACK_EXAMPLE) += avio_read_callback
 EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE)      += decode_audio
-EXAMPLES-$(CONFIG_DECODE_FILTER_AUDIO_EXAMPLE) += decode_filter_audio
-EXAMPLES-$(CONFIG_DECODE_FILTER_VIDEO_EXAMPLE) += decode_filter_video
 EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE)      += decode_video
-EXAMPLES-$(CONFIG_DEMUX_DECODE_EXAMPLE)      += demux_decode
+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_MUX_EXAMPLE)               += mux
+EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-EXAMPLES-$(CONFIG_QSV_DECODE_EXAMPLE)        += qsv_decode
+EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-EXAMPLES-$(CONFIG_REMUX_EXAMPLE)             += remux
+EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
-EXAMPLES-$(CONFIG_RESAMPLE_AUDIO_EXAMPLE)    += resample_audio
+EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-EXAMPLES-$(CONFIG_SCALE_VIDEO_EXAMPLE)       += scale_video
+EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-EXAMPLES-$(CONFIG_SHOW_METADATA_EXAMPLE)     += show_metadata
+EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
 EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-EXAMPLES-$(CONFIG_TRANSCODE_EXAMPLE)         += transcode
+EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
 EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE)      += vaapi_encode
 EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE)   += vaapi_transcode
-EXAMPLES-$(CONFIG_QSV_TRANSCODE_EXAMPLE)     += qsv_transcode
 
 EXAMPLES       := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
 EXAMPLES_G     := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))

doc/examples/Makefile.example

@@ -11,40 +11,33 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-# missing the following targets, since they need special options in the FFmpeg build:
+EXAMPLES=       avio_dir_cmd                       \
-# qsv_decode
+                avio_reading                       \
-# qsv_transcode
-# vaapi_encode
-# vaapi_transcode
-
-EXAMPLES=\
-                avio_http_serve_files              \
-                avio_list_dir                      \
-                avio_read_callback                 \
                 decode_audio                       \
-                decode_filter_audio                \
-                decode_filter_video                \
                 decode_video                       \
-                demux_decode                       \
+                demuxing_decoding                  \
                 encode_audio                       \
                 encode_video                       \
                 extract_mvs                        \
+                filtering_video                    \
+                filtering_audio                    \
+                http_multiclient                   \
                 hw_decode                          \
-                mux                                \
+                metadata                           \
-                remux                              \
+                muxing                             \
-                resample_audio                     \
+                remuxing                           \
-                scale_video                        \
+                resampling_audio                   \
-                show_metadata                      \
+                scaling_video                      \
                 transcode_aac                      \
-                transcode
+                transcoding                        \
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
 avcodec:           LDLIBS += -lm
 encode_audio:      LDLIBS += -lm
-mux:               LDLIBS += -lm
+muxing:            LDLIBS += -lm
-resample_audio:    LDLIBS += -lm
+resampling_audio:  LDLIBS += -lm
 
 .phony: all clean-test clean
 

doc/examples/README

@@ -7,10 +7,8 @@ that you have them installed and working on your system.
 
 Method 1: build the installed examples in a generic read/write user directory
 
-Copy to a read/write user directory and run:
+Copy to a read/write user directory and just use "make", it will link
-make -f Makefile.example
+to the libraries on your system, assuming the PKG_CONFIG_PATH is
-
-It will link to the libraries on your system, assuming the PKG_CONFIG_PATH is
 correctly configured.
 
 Method 2: build the examples in-tree
@@ -22,4 +20,4 @@ examples using "make examplesclean"
 
 If you want to try the dedicated Makefile examples (to emulate the first
 method), go into doc/examples and run a command such as
-PKG_CONFIG_PATH=pc-uninstalled make -f Makefile.example
+PKG_CONFIG_PATH=pc-uninstalled make.

doc/examples/avio_dir_cmd.c

@@ -20,13 +20,6 @@
  * THE SOFTWARE.
  */
 
-/**
- * @file libavformat AVIOContext list directory API usage example
- * @example avio_list_dir.c
- *
- * Show how to list directories through the libavformat AVIOContext API.
- */
-
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavformat/avio.h>
@@ -109,15 +102,38 @@ static int list_op(const char *input_dir)
     return ret;
 }
 
+static int del_op(const char *url)
+{
+    int ret = avpriv_io_delete(url);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
+    return ret;
+}
+
+static int move_op(const char *src, const char *dst)
+{
+    int ret = avpriv_io_move(src, dst);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
+    return ret;
+}
+
+
 static void usage(const char *program_name)
 {
-    fprintf(stderr, "usage: %s input_dir\n"
+    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
-            "API example program to show how to list files in directory "
+            "API example program to show how to manipulate resources "
-            "accessed through AVIOContext.\n", program_name);
+            "accessed through AVIOContext.\n"
+            "OPERATIONS:\n"
+            "list      list content of the directory\n"
+            "move      rename content in directory\n"
+            "del       delete content in directory\n",
+            program_name);
 }
 
 int main(int argc, char *argv[])
 {
+    const char *op = NULL;
     int ret;
 
     av_log_set_level(AV_LOG_DEBUG);
@@ -129,7 +145,32 @@ int main(int argc, char *argv[])
 
     avformat_network_init();
 
-    ret = list_op(argv[1]);
+    op = argv[1];
+    if (strcmp(op, "list") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = list_op(argv[2]);
+        }
+    } else if (strcmp(op, "del") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = del_op(argv[2]);
+        }
+    } else if (strcmp(op, "move") == 0) {
+        if (argc < 4) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = move_op(argv[2], argv[3]);
+        }
+    } else {
+        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
+        ret = AVERROR(EINVAL);
+    }
 
     avformat_network_deinit();
 

doc/examples/avio_reading.c

@@ -21,11 +21,12 @@
  */
 
 /**
- * @file libavformat AVIOContext read callback API usage example
+ * @file
- * @example avio_read_callback.c
+ * libavformat AVIOContext API example.
  *
  * Make libavformat demuxer access media content through a custom
  * AVIOContext read callback.
+ * @example avio_reading.c
  */
 
 #include <libavcodec/avcodec.h>

doc/examples/decode_audio.c

@@ -21,11 +21,10 @@
  */
 
 /**
- * @file libavcodec audio decoding API usage example
+ * @file
- * @example decode_audio.c
+ * audio decoding with libavcodec API example
  *
- * Decode data from an MP2 input file and generate a raw audio file to
+ * @example decode_audio.c
- * be played with ffplay.
  */
 
 #include <stdio.h>
@@ -40,35 +39,6 @@
 #define AUDIO_INBUF_SIZE 20480
 #define AUDIO_REFILL_THRESH 4096
 
-static int get_format_from_sample_fmt(const char **fmt,
-                                      enum AVSampleFormat sample_fmt)
-{
-    int i;
-    struct sample_fmt_entry {
-        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
-    } sample_fmt_entries[] = {
-        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
-        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
-        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
-        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
-        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
-    };
-    *fmt = NULL;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
-        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
-        if (sample_fmt == entry->sample_fmt) {
-            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
-            return 0;
-        }
-    }
-
-    fprintf(stderr,
-            "sample format %s is not supported as output format\n",
-            av_get_sample_fmt_name(sample_fmt));
-    return -1;
-}
-
 static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
                    FILE *outfile)
 {
@@ -98,7 +68,7 @@ static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
             exit(1);
         }
         for (i = 0; i < frame->nb_samples; i++)
-            for (ch = 0; ch < dec_ctx->ch_layout.nb_channels; ch++)
+            for (ch = 0; ch < dec_ctx->channels; ch++)
                 fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
     }
 }
@@ -116,9 +86,6 @@ int main(int argc, char **argv)
     size_t   data_size;
     AVPacket *pkt;
     AVFrame *decoded_frame = NULL;
-    enum AVSampleFormat sfmt;
-    int n_channels = 0;
-    const char *fmt;
 
     if (argc <= 2) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -205,26 +172,6 @@ int main(int argc, char **argv)
     pkt->size = 0;
     decode(c, pkt, decoded_frame, outfile);
 
-    /* print output pcm infomations, because there have no metadata of pcm */
-    sfmt = c->sample_fmt;
-
-    if (av_sample_fmt_is_planar(sfmt)) {
-        const char *packed = av_get_sample_fmt_name(sfmt);
-        printf("Warning: the sample format the decoder produced is planar "
-               "(%s). This example will output the first channel only.\n",
-               packed ? packed : "?");
-        sfmt = av_get_packed_sample_fmt(sfmt);
-    }
-
-    n_channels = c->ch_layout.nb_channels;
-    if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
-        goto end;
-
-    printf("Play the output audio file with the command:\n"
-           "ffplay -f %s -ac %d -ar %d %s\n",
-           fmt, n_channels, c->sample_rate,
-           outfilename);
-end:
     fclose(outfile);
     fclose(f);
 

doc/examples/decode_video.c

@@ -21,11 +21,10 @@
  */
 
 /**
- * @file libavcodec video decoding API usage example
+ * @file
- * @example decode_video.c *
+ * video decoding with libavcodec API example
  *
- * Read from an MPEG1 video file, decode frames, and generate PGM images as
+ * @example decode_video.c
- * output.
  */
 
 #include <stdio.h>
@@ -42,7 +41,7 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
     FILE *f;
     int i;
 
-    f = fopen(filename,"wb");
+    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);
@@ -70,12 +69,12 @@ static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
             exit(1);
         }
 
-        printf("saving frame %3"PRId64"\n", dec_ctx->frame_num);
+        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-%"PRId64, filename, dec_ctx->frame_num);
+        snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
         pgm_save(frame->data[0], frame->linesize[0],
                  frame->width, frame->height, buf);
     }
@@ -93,12 +92,10 @@ int main(int argc, char **argv)
     uint8_t *data;
     size_t   data_size;
     int ret;
-    int eof;
     AVPacket *pkt;
 
     if (argc <= 2) {
-        fprintf(stderr, "Usage: %s <input file> <output file>\n"
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
-                "And check your input file is encoded by mpeg1video please.\n", argv[0]);
         exit(0);
     }
     filename    = argv[1];
@@ -152,16 +149,15 @@ int main(int argc, char **argv)
         exit(1);
     }
 
-    do {
+    while (!feof(f)) {
         /* read raw data from the input file */
         data_size = fread(inbuf, 1, INBUF_SIZE, f);
-        if (ferror(f))
+        if (!data_size)
             break;
-        eof = !data_size;
 
         /* use the parser to split the data into frames */
         data = inbuf;
-        while (data_size > 0 || eof) {
+        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) {
@@ -173,10 +169,8 @@ int main(int argc, char **argv)
 
             if (pkt->size)
                 decode(c, frame, pkt, outfilename);
-            else if (eof)
-                break;
         }
-    } while (!eof);
+    }
 
     /* flush the decoder */
     decode(c, frame, NULL, outfilename);

doc/examples/demuxing_decoding.c

@@ -21,18 +21,17 @@
  */
 
 /**
- * @file libavformat and libavcodec demuxing and decoding API usage example
+ * @file
- * @example demux_decode.c
+ * Demuxing and decoding example.
  *
- * Show how to use the libavformat and libavcodec API to demux and decode audio
+ * Show how to use the libavformat and libavcodec API to demux and
- * and video data. Write the output as raw audio and input files to be played by
+ * decode audio and video data.
- * ffplay.
+ * @example demuxing_decoding.c
  */
 
 #include <libavutil/imgutils.h>
 #include <libavutil/samplefmt.h>
 #include <libavutil/timestamp.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 
 static AVFormatContext *fmt_ctx = NULL;
@@ -52,97 +51,99 @@ static int video_dst_bufsize;
 
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
-static AVPacket *pkt = NULL;
+static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;
 
-static int output_video_frame(AVFrame *frame)
+/* Enable or disable frame reference counting. You are not supposed to support
-{
+ * both paths in your application but pick the one most appropriate to your
-    if (frame->width != width || frame->height != height ||
+ * needs. Look for the use of refcount in this example to see what are the
-        frame->format != pix_fmt) {
+ * differences of API usage between them. */
-        /* To handle this change, one could call av_image_alloc again and
+static int refcount = 0;
-         * decode the following frames into another rawvideo file. */
-        fprintf(stderr, "Error: Width, height and pixel format have to be "
-                "constant in a rawvideo file, but the width, height or "
-                "pixel format of the input video changed:\n"
-                "old: width = %d, height = %d, format = %s\n"
-                "new: width = %d, height = %d, format = %s\n",
-                width, height, av_get_pix_fmt_name(pix_fmt),
-                frame->width, frame->height,
-                av_get_pix_fmt_name(frame->format));
-        return -1;
-    }
 
-    printf("video_frame n:%d\n",
+static int decode_packet(int *got_frame, int cached)
-           video_frame_count++);
-
-    /* copy decoded frame to destination buffer:
-     * this is required since rawvideo expects non aligned data */
-    av_image_copy(video_dst_data, video_dst_linesize,
-                  (const uint8_t **)(frame->data), frame->linesize,
-                  pix_fmt, width, height);
-
-    /* write to rawvideo file */
-    fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
-    return 0;
-}
-
-static int output_audio_frame(AVFrame *frame)
-{
-    size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
-    printf("audio_frame n:%d nb_samples:%d pts:%s\n",
-           audio_frame_count++, frame->nb_samples,
-           av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
-
-    /* Write the raw audio data samples of the first plane. This works
-     * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
-     * most audio decoders output planar audio, which uses a separate
-     * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
-     * In other words, this code will write only the first audio channel
-     * in these cases.
-     * You should use libswresample or libavfilter to convert the frame
-     * to packed data. */
-    fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
-
-    return 0;
-}
-
-static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
 {
     int ret = 0;
+    int decoded = pkt.size;
 
-    // submit the packet to the decoder
+    *got_frame = 0;
-    ret = avcodec_send_packet(dec, pkt);
-    if (ret < 0) {
-        fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
-        return ret;
-    }
 
-    // get all the available frames from the decoder
+    if (pkt.stream_index == video_stream_idx) {
-    while (ret >= 0) {
+        /* decode video frame */
-        ret = avcodec_receive_frame(dec, frame);
+        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
         if (ret < 0) {
-            // those two return values are special and mean there is no output
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
-            // frame available, but there were no errors during decoding
-            if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
-                return 0;
-
-            fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
             return ret;
         }
 
-        // write the frame data to output file
+        if (*got_frame) {
-        if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
-            ret = output_video_frame(frame);
-        else
-            ret = output_audio_frame(frame);
 
-        av_frame_unref(frame);
+            if (frame->width != width || frame->height != height ||
-        if (ret < 0)
+                frame->format != pix_fmt) {
+                /* To handle this change, one could call av_image_alloc again and
+                 * decode the following frames into another rawvideo file. */
+                fprintf(stderr, "Error: Width, height and pixel format have to be "
+                        "constant in a rawvideo file, but the width, height or "
+                        "pixel format of the input video changed:\n"
+                        "old: width = %d, height = %d, format = %s\n"
+                        "new: width = %d, height = %d, format = %s\n",
+                        width, height, av_get_pix_fmt_name(pix_fmt),
+                        frame->width, frame->height,
+                        av_get_pix_fmt_name(frame->format));
+                return -1;
+            }
+
+            printf("video_frame%s n:%d coded_n:%d\n",
+                   cached ? "(cached)" : "",
+                   video_frame_count++, frame->coded_picture_number);
+
+            /* copy decoded frame to destination buffer:
+             * this is required since rawvideo expects non aligned data */
+            av_image_copy(video_dst_data, video_dst_linesize,
+                          (const uint8_t **)(frame->data), frame->linesize,
+                          pix_fmt, width, height);
+
+            /* write to rawvideo file */
+            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
+        }
+    } else if (pkt.stream_index == audio_stream_idx) {
+        /* decode audio frame */
+        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
             return ret;
+        }
+        /* Some audio decoders decode only part of the packet, and have to be
+         * called again with the remainder of the packet data.
+         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
+         * Also, some decoders might over-read the packet. */
+        decoded = FFMIN(ret, pkt.size);
+
+        if (*got_frame) {
+            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
+            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   audio_frame_count++, frame->nb_samples,
+                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
+
+            /* Write the raw audio data samples of the first plane. This works
+             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
+             * most audio decoders output planar audio, which uses a separate
+             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
+             * In other words, this code will write only the first audio channel
+             * in these cases.
+             * You should use libswresample or libavfilter to convert the frame
+             * to packed data. */
+            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
+        }
     }
 
-    return 0;
+    /* If we use frame reference counting, we own the data and need
+     * to de-reference it when we don't use it anymore */
+    if (*got_frame && refcount)
+        av_frame_unref(frame);
+
+    return decoded;
 }
 
 static int open_codec_context(int *stream_idx,
@@ -150,7 +151,8 @@ static int open_codec_context(int *stream_idx,
 {
     int ret, stream_index;
     AVStream *st;
-    const AVCodec *dec = NULL;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
 
     ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
     if (ret < 0) {
@@ -184,8 +186,9 @@ static int open_codec_context(int *stream_idx,
             return ret;
         }
 
-        /* Init the decoders */
+        /* Init the decoders, with or without reference counting */
-        if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 0) {
+        av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
+        if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) {
             fprintf(stderr, "Failed to open %s codec\n",
                     av_get_media_type_string(type));
             return ret;
@@ -227,17 +230,24 @@ static int get_format_from_sample_fmt(const char **fmt,
 
 int main (int argc, char **argv)
 {
-    int ret = 0;
+    int ret = 0, got_frame;
 
-    if (argc != 4) {
+    if (argc != 4 && argc != 5) {
-        fprintf(stderr, "usage: %s  input_file video_output_file audio_output_file\n"
+        fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
                 "API example program to show how to read frames from an input file.\n"
                 "This program reads frames from a file, decodes them, and writes decoded\n"
                 "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n",
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
-                argv[0]);
+                "If the -refcount option is specified, the program use the\n"
+                "reference counting frame system which allows keeping a copy of\n"
+                "the data for longer than one decode call.\n"
+                "\n", argv[0]);
         exit(1);
     }
+    if (argc == 5 && !strcmp(argv[1], "-refcount")) {
+        refcount = 1;
+        argv++;
+    }
     src_filename = argv[1];
     video_dst_filename = argv[2];
     audio_dst_filename = argv[3];
@@ -303,12 +313,10 @@ int main (int argc, char **argv)
         goto end;
     }
 
-    pkt = av_packet_alloc();
+    /* initialize packet, set data to NULL, let the demuxer fill it */
-    if (!pkt) {
+    av_init_packet(&pkt);
-        fprintf(stderr, "Could not allocate packet\n");
+    pkt.data = NULL;
-        ret = AVERROR(ENOMEM);
+    pkt.size = 0;
-        goto end;
-    }
 
     if (video_stream)
         printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
@@ -316,23 +324,24 @@ int main (int argc, char **argv)
         printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
 
     /* read frames from the file */
-    while (av_read_frame(fmt_ctx, pkt) >= 0) {
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        // check if the packet belongs to a stream we are interested in, otherwise
+        AVPacket orig_pkt = pkt;
-        // skip it
+        do {
-        if (pkt->stream_index == video_stream_idx)
+            ret = decode_packet(&got_frame, 0);
-            ret = decode_packet(video_dec_ctx, pkt);
+            if (ret < 0)
-        else if (pkt->stream_index == audio_stream_idx)
+                break;
-            ret = decode_packet(audio_dec_ctx, pkt);
+            pkt.data += ret;
-        av_packet_unref(pkt);
+            pkt.size -= ret;
-        if (ret < 0)
+        } while (pkt.size > 0);
-            break;
+        av_packet_unref(&orig_pkt);
     }
 
-    /* flush the decoders */
+    /* flush cached frames */
-    if (video_dec_ctx)
+    pkt.data = NULL;
-        decode_packet(video_dec_ctx, NULL);
+    pkt.size = 0;
-    if (audio_dec_ctx)
+    do {
-        decode_packet(audio_dec_ctx, NULL);
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
 
     printf("Demuxing succeeded.\n");
 
@@ -345,7 +354,7 @@ int main (int argc, char **argv)
 
     if (audio_stream) {
         enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
-        int n_channels = audio_dec_ctx->ch_layout.nb_channels;
+        int n_channels = audio_dec_ctx->channels;
         const char *fmt;
 
         if (av_sample_fmt_is_planar(sfmt)) {
@@ -374,7 +383,6 @@ end:
         fclose(video_dst_file);
     if (audio_dst_file)
         fclose(audio_dst_file);
-    av_packet_free(&pkt);
     av_frame_free(&frame);
     av_free(video_dst_data[0]);
 

doc/examples/encode_audio.c

@@ -21,10 +21,10 @@
  */
 
 /**
- * @file libavcodec encoding audio API usage examples
+ * @file
- * @example encode_audio.c
+ * audio encoding with libavcodec API example.
  *
- * Generate a synthetic audio signal and encode it to an output MP2 file.
+ * @example encode_audio.c
  */
 
 #include <stdint.h>
@@ -70,25 +70,26 @@ static int select_sample_rate(const AVCodec *codec)
 }
 
 /* select layout with the highest channel count */
-static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
+static int select_channel_layout(const AVCodec *codec)
 {
-    const AVChannelLayout *p, *best_ch_layout;
+    const uint64_t *p;
+    uint64_t best_ch_layout = 0;
     int best_nb_channels   = 0;
 
-    if (!codec->ch_layouts)
+    if (!codec->channel_layouts)
-        return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+        return AV_CH_LAYOUT_STEREO;
 
-    p = codec->ch_layouts;
+    p = codec->channel_layouts;
-    while (p->nb_channels) {
+    while (*p) {
-        int nb_channels = p->nb_channels;
+        int nb_channels = av_get_channel_layout_nb_channels(*p);
 
         if (nb_channels > best_nb_channels) {
-            best_ch_layout   = p;
+            best_ch_layout    = *p;
             best_nb_channels = nb_channels;
         }
         p++;
     }
-    return av_channel_layout_copy(dst, best_ch_layout);
+    return best_ch_layout;
 }
 
 static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
@@ -163,9 +164,8 @@ int main(int argc, char **argv)
 
     /* select other audio parameters supported by the encoder */
     c->sample_rate    = select_sample_rate(codec);
-    ret = select_channel_layout(codec, &c->ch_layout);
+    c->channel_layout = select_channel_layout(codec);
-    if (ret < 0)
+    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
-        exit(1);
 
     /* open it */
     if (avcodec_open2(c, codec, NULL) < 0) {
@@ -195,9 +195,7 @@ int main(int argc, char **argv)
 
     frame->nb_samples     = c->frame_size;
     frame->format         = c->sample_fmt;
-    ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
+    frame->channel_layout = c->channel_layout;
-    if (ret < 0)
-        exit(1);
 
     /* allocate the data buffers */
     ret = av_frame_get_buffer(frame, 0);
@@ -220,7 +218,7 @@ int main(int argc, char **argv)
         for (j = 0; j < c->frame_size; j++) {
             samples[2*j] = (int)(sin(t) * 10000);
 
-            for (k = 1; k < c->ch_layout.nb_channels; k++)
+            for (k = 1; k < c->channels; k++)
                 samples[2*j + k] = samples[2*j];
             t += tincr;
         }

doc/examples/encode_video.c

@@ -21,10 +21,10 @@
  */
 
 /**
- * @file libavcodec encoding video API usage example
+ * @file
- * @example encode_video.c
+ * video encoding with libavcodec API example
  *
- * Generate synthetic video data and encode it to an output file.
+ * @example encode_video.c
  */
 
 #include <stdio.h>
@@ -145,7 +145,7 @@ int main(int argc, char **argv)
     frame->width  = c->width;
     frame->height = c->height;
 
-    ret = av_frame_get_buffer(frame, 0);
+    ret = av_frame_get_buffer(frame, 32);
     if (ret < 0) {
         fprintf(stderr, "Could not allocate the video frame data\n");
         exit(1);
@@ -155,25 +155,12 @@ int main(int argc, char **argv)
     for (i = 0; i < 25; i++) {
         fflush(stdout);
 
-        /* Make sure the frame data is writable.
+        /* make sure the frame data is writable */
-           On the first round, the frame is fresh from av_frame_get_buffer()
-           and therefore we know it is writable.
-           But on the next rounds, encode() will have called
-           avcodec_send_frame(), and the codec may have kept a reference to
-           the frame in its internal structures, that makes the frame
-           unwritable.
-           av_frame_make_writable() checks that and allocates a new buffer
-           for the frame only if necessary.
-         */
         ret = av_frame_make_writable(frame);
         if (ret < 0)
             exit(1);
 
-        /* Prepare a dummy image.
+        /* prepare a dummy image */
-           In real code, this is where you would have your own logic for
-           filling the frame. FFmpeg does not care what you put in the
-           frame.
-         */
         /* Y */
         for (y = 0; y < c->height; y++) {
             for (x = 0; x < c->width; x++) {
@@ -198,14 +185,8 @@ int main(int argc, char **argv)
     /* flush the encoder */
     encode(c, NULL, pkt, f);
 
-    /* Add sequence end code to have a real MPEG file.
+    /* add sequence end code to have a real MPEG file */
-       It makes only sense because this tiny examples writes packets
+    fwrite(endcode, 1, sizeof(endcode), f);
-       directly. This is called "elementary stream" and only works for some
-       codecs. To create a valid file, you usually need to write packets
-       into a proper file format or protocol; see mux.c.
-     */
-    if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
-        fwrite(endcode, 1, sizeof(endcode), f);
     fclose(f);
 
     avcodec_free_context(&c);

doc/examples/extract_mvs.c

@@ -21,16 +21,7 @@
  * THE SOFTWARE.
  */
 
-/**
- * @file libavcodec motion vectors extraction API usage example
- * @example extract_mvs.c
- *
- * Read from input file, decode video stream and print a motion vectors
- * representation to stdout.
- */
-
 #include <libavutil/motion_vector.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 
 static AVFormatContext *fmt_ctx = NULL;
@@ -69,11 +60,10 @@ static int decode_packet(const AVPacket *pkt)
                 const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
                 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",%4d,%4d,%4d\n",
+                    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,
+                        mv->dst_x, mv->dst_y, mv->flags);
-                        mv->motion_x, mv->motion_y, mv->motion_scale);
                 }
             }
             av_frame_unref(frame);
@@ -88,7 +78,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
     int ret;
     AVStream *st;
     AVCodecContext *dec_ctx = NULL;
-    const AVCodec *dec = NULL;
+    AVCodec *dec = NULL;
     AVDictionary *opts = NULL;
 
     ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
@@ -114,9 +104,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
 
         /* Init the video decoder */
         av_dict_set(&opts, "flags2", "+export_mvs", 0);
-        ret = avcodec_open2(dec_ctx, dec, &opts);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
-        av_dict_free(&opts);
-        if (ret < 0) {
             fprintf(stderr, "Failed to open %s codec\n",
                     av_get_media_type_string(type));
             return ret;
@@ -133,7 +121,7 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
 int main(int argc, char **argv)
 {
     int ret = 0;
-    AVPacket *pkt = NULL;
+    AVPacket pkt = { 0 };
 
     if (argc != 2) {
         fprintf(stderr, "Usage: %s <video>\n", argv[0]);
@@ -168,20 +156,13 @@ int main(int argc, char **argv)
         goto end;
     }
 
-    pkt = av_packet_alloc();
+    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
-    if (!pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags,motion_x,motion_y,motion_scale\n");
 
     /* read frames from the file */
-    while (av_read_frame(fmt_ctx, pkt) >= 0) {
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        if (pkt->stream_index == video_stream_idx)
+        if (pkt.stream_index == video_stream_idx)
-            ret = decode_packet(pkt);
+            ret = decode_packet(&pkt);
-        av_packet_unref(pkt);
+        av_packet_unref(&pkt);
         if (ret < 0)
             break;
     }
@@ -193,6 +174,5 @@ end:
     avcodec_free_context(&video_dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
-    av_packet_free(&pkt);
     return ret < 0;
 }

doc/examples/filter_audio.c

@@ -19,11 +19,13 @@
  */
 
 /**
- * @file libavfilter audio filtering API usage example
+ * @file
- * @example filter_audio.c
+ * libavfilter API usage example.
  *
- * This example will generate a sine wave audio, pass it through a simple filter
+ * @example filter_audio.c
- * chain, and then compute the MD5 checksum of the output data.
+ * This example will generate a sine wave audio,
+ * pass it through a simple filter chain, and then compute the MD5 checksum of
+ * the output data.
  *
  * The filter chain it uses is:
  * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
@@ -53,7 +55,7 @@
 
 #define INPUT_SAMPLERATE     48000
 #define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
 
 #define VOLUME_VAL 0.90
 
@@ -98,7 +100,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
     }
 
     /* Set the filter options through the AVOptions API. */
-    av_channel_layout_describe(&INPUT_CHANNEL_LAYOUT, ch_layout, sizeof(ch_layout));
+    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
     av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
     av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
     av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
@@ -152,8 +154,9 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
     /* A third way of passing the options is in a string of the form
      * key1=value1:key2=value2.... */
     snprintf(options_str, sizeof(options_str),
-             "sample_fmts=%s:sample_rates=%d:channel_layouts=stereo",
+             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100);
+             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
+             (uint64_t)AV_CH_LAYOUT_STEREO);
     err = avfilter_init_str(aformat_ctx, options_str);
     if (err < 0) {
         av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
@@ -212,7 +215,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
 static int process_output(struct AVMD5 *md5, AVFrame *frame)
 {
     int planar     = av_sample_fmt_is_planar(frame->format);
-    int channels   = frame->ch_layout.nb_channels;
+    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
     int planes     = planar ? channels : 1;
     int bps        = av_get_bytes_per_sample(frame->format);
     int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
@@ -245,7 +248,7 @@ static int get_input(AVFrame *frame, int frame_num)
     /* Set up the frame properties and allocate the buffer for the data. */
     frame->sample_rate    = INPUT_SAMPLERATE;
     frame->format         = INPUT_FORMAT;
-    av_channel_layout_copy(&frame->ch_layout, &INPUT_CHANNEL_LAYOUT);
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
     frame->nb_samples     = FRAME_SIZE;
     frame->pts            = frame_num * FRAME_SIZE;
 

doc/examples/filtering_audio.c

@@ -23,11 +23,9 @@
  */
 
 /**
- * @file audio decoding and filtering usage example
+ * @file
- * @example decode_filter_audio.c
+ * API example for audio decoding and filtering
- *
+ * @example filtering_audio.c
- * Demux, decode and filter audio input file, generate a raw audio
- * file to be played with ffplay.
  */
 
 #include <unistd.h>
@@ -36,7 +34,6 @@
 #include <libavformat/avformat.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 
 static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
@@ -51,8 +48,8 @@ static int audio_stream_index = -1;
 
 static int open_input_file(const char *filename)
 {
-    const AVCodec *dec;
     int ret;
+    AVCodec *dec;
 
     if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
@@ -96,6 +93,7 @@ static int init_filters(const char *filters_descr)
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
     static const int out_sample_rates[] = { 8000, -1 };
     const AVFilterLink *outlink;
     AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
@@ -107,13 +105,12 @@ static int init_filters(const char *filters_descr)
     }
 
     /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
+    if (!dec_ctx->channel_layout)
-        av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
+        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    ret = snprintf(args, sizeof(args),
+    snprintf(args, sizeof(args),
-            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
+            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
              time_base.num, time_base.den, dec_ctx->sample_rate,
-             av_get_sample_fmt_name(dec_ctx->sample_fmt));
+             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
-    av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
     ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
                                        args, NULL, filter_graph);
     if (ret < 0) {
@@ -136,7 +133,7 @@ static int init_filters(const char *filters_descr)
         goto end;
     }
 
-    ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
+    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
@@ -187,7 +184,7 @@ static int init_filters(const char *filters_descr)
     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
     outlink = buffersink_ctx->inputs[0];
-    av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
+    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
     av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
            (int)outlink->sample_rate,
            (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
@@ -202,7 +199,7 @@ end:
 
 static void print_frame(const AVFrame *frame)
 {
-    const int n = frame->nb_samples * frame->ch_layout.nb_channels;
+    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
     const uint16_t *p     = (uint16_t*)frame->data[0];
     const uint16_t *p_end = p + n;
 
@@ -217,12 +214,12 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket *packet = av_packet_alloc();
+    AVPacket packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
 
-    if (!packet || !frame || !filt_frame) {
+    if (!frame || !filt_frame) {
-        fprintf(stderr, "Could not allocate frame or packet\n");
+        perror("Could not allocate frame");
         exit(1);
     }
     if (argc != 2) {
@@ -237,11 +234,11 @@ int main(int argc, char **argv)
 
     /* read all packets */
     while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
             break;
 
-        if (packet->stream_index == audio_stream_index) {
+        if (packet.stream_index == audio_stream_index) {
-            ret = avcodec_send_packet(dec_ctx, packet);
+            ret = avcodec_send_packet(dec_ctx, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                 break;
@@ -277,13 +274,12 @@ int main(int argc, char **argv)
                 }
             }
         }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
     avcodec_free_context(&dec_ctx);
     avformat_close_input(&fmt_ctx);
-    av_packet_free(&packet);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 

doc/examples/filtering_video.c

@@ -24,7 +24,7 @@
 /**
  * @file
  * API example for decoding and filtering
- * @example decode_filter_video.c
+ * @example filtering_video.c
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -53,8 +53,8 @@ static int64_t last_pts = AV_NOPTS_VALUE;
 
 static int open_input_file(const char *filename)
 {
-    const AVCodec *dec;
     int ret;
+    AVCodec *dec;
 
     if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
@@ -210,7 +210,7 @@ static void display_frame(const AVFrame *frame, AVRational time_base)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket *packet;
+    AVPacket packet;
     AVFrame *frame;
     AVFrame *filt_frame;
 
@@ -221,9 +221,8 @@ int main(int argc, char **argv)
 
     frame = av_frame_alloc();
     filt_frame = av_frame_alloc();
-    packet = av_packet_alloc();
+    if (!frame || !filt_frame) {
-    if (!frame || !filt_frame || !packet) {
+        perror("Could not allocate frame");
-        fprintf(stderr, "Could not allocate frame or packet\n");
         exit(1);
     }
 
@@ -234,11 +233,11 @@ int main(int argc, char **argv)
 
     /* read all packets */
     while (1) {
-        if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
             break;
 
-        if (packet->stream_index == video_stream_index) {
+        if (packet.stream_index == video_stream_index) {
-            ret = avcodec_send_packet(dec_ctx, packet);
+            ret = avcodec_send_packet(dec_ctx, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                 break;
@@ -274,7 +273,7 @@ int main(int argc, char **argv)
                 av_frame_unref(frame);
             }
         }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
@@ -282,7 +281,6 @@ end:
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
-    av_packet_free(&packet);
 
     if (ret < 0 && ret != AVERROR_EOF) {
         fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));

doc/examples/http_multiclient.c

@@ -21,11 +21,12 @@
  */
 
 /**
- * @file libavformat multi-client network API usage example
+ * @file
- * @example avio_http_serve_files.c
+ * libavformat multi-client network API usage example.
  *
- * Serve a file without decoding or demuxing it over the HTTP protocol. Multiple
+ * @example http_multiclient.c
- * clients can connect and will receive the same file.
+ * This example will serve a file without decoding or demuxing it over http.
+ * Multiple clients can connect and will receive the same file.
  */
 
 #include <libavformat/avformat.h>

doc/examples/hw_decode.c

@@ -24,11 +24,12 @@
  */
 
 /**
- * @file HW-accelerated decoding API usage.example
+ * @file
- * @example hw_decode.c
+ * HW-Accelerated decoding example.
  *
- * Perform HW-accelerated decoding with output frames from HW video
+ * @example hw_decode.c
- * surfaces.
+ * This example shows how to do HW-accelerated decoding with output
+ * frames from the HW video surfaces.
  */
 
 #include <stdio.h>
@@ -151,8 +152,8 @@ int main(int argc, char *argv[])
     int video_stream, ret;
     AVStream *video = NULL;
     AVCodecContext *decoder_ctx = NULL;
-    const AVCodec *decoder = NULL;
+    AVCodec *decoder = NULL;
-    AVPacket *packet = NULL;
+    AVPacket packet;
     enum AVHWDeviceType type;
     int i;
 
@@ -171,12 +172,6 @@ int main(int argc, char *argv[])
         return -1;
     }
 
-    packet = av_packet_alloc();
-    if (!packet) {
-        fprintf(stderr, "Failed to allocate AVPacket\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]);
@@ -228,25 +223,27 @@ int main(int argc, char *argv[])
     }
 
     /* open the file to dump raw data */
-    output_file = fopen(argv[3], "w+b");
+    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)
+        if ((ret = av_read_frame(input_ctx, &packet)) < 0)
             break;
 
-        if (video_stream == packet->stream_index)
+        if (video_stream == packet.stream_index)
-            ret = decode_write(decoder_ctx, packet);
+            ret = decode_write(decoder_ctx, &packet);
 
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 
     /* flush the decoder */
-    ret = decode_write(decoder_ctx, NULL);
+    packet.data = NULL;
+    packet.size = 0;
+    ret = decode_write(decoder_ctx, &packet);
+    av_packet_unref(&packet);
 
     if (output_file)
         fclose(output_file);
-    av_packet_free(&packet);
     avcodec_free_context(&decoder_ctx);
     avformat_close_input(&input_ctx);
     av_buffer_unref(&hw_device_ctx);

doc/examples/metadata.c

@@ -21,10 +21,9 @@
  */
 
 /**
- * @file libavformat metadata extraction API usage example
+ * @file
- * @example show_metadata.c
+ * Shows how the metadata API can be used in application programs.
- *
+ * @example metadata.c
- * Show metadata from an input file.
  */
 
 #include <stdio.h>
@@ -35,7 +34,7 @@
 int main (int argc, char **argv)
 {
     AVFormatContext *fmt_ctx = NULL;
-    const AVDictionaryEntry *tag = NULL;
+    AVDictionaryEntry *tag = NULL;
     int ret;
 
     if (argc != 2) {
@@ -53,7 +52,7 @@ int main (int argc, char **argv)
         return ret;
     }
 
-    while ((tag = av_dict_iterate(fmt_ctx->metadata, tag)))
+    while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
         printf("%s=%s\n", tag->key, tag->value);
 
     avformat_close_input(&fmt_ctx);

doc/examples/muxing.c

@@ -21,11 +21,12 @@
  */
 
 /**
- * @file libavformat muxing API usage example
+ * @file
- * @example mux.c
+ * libavformat API example.
  *
- * Generate a synthetic audio and video signal and mux them to a media file in
+ * Output a media file in any supported libavformat format. The default
- * any supported libavformat format. The default codecs are used.
+ * codecs are used.
+ * @example muxing.c
  */
 
 #include <stdlib.h>
@@ -38,7 +39,6 @@
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/timestamp.h>
-#include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
@@ -61,8 +61,6 @@ typedef struct OutputStream {
     AVFrame *frame;
     AVFrame *tmp_frame;
 
-    AVPacket *tmp_pkt;
-
     float t, tincr, tincr2;
 
     struct SwsContext *sws_ctx;
@@ -80,50 +78,20 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
            pkt->stream_index);
 }
 
-static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
-                       AVStream *st, AVFrame *frame, AVPacket *pkt)
 {
-    int ret;
+    /* rescale output packet timestamp values from codec to stream timebase */
+    av_packet_rescale_ts(pkt, *time_base, st->time_base);
+    pkt->stream_index = st->index;
 
-    // send the frame to the encoder
+    /* Write the compressed frame to the media file. */
-    ret = avcodec_send_frame(c, frame);
+    log_packet(fmt_ctx, pkt);
-    if (ret < 0) {
+    return av_interleaved_write_frame(fmt_ctx, pkt);
-        fprintf(stderr, "Error sending a frame to the encoder: %s\n",
-                av_err2str(ret));
-        exit(1);
-    }
-
-    while (ret >= 0) {
-        ret = avcodec_receive_packet(c, pkt);
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-            break;
-        else if (ret < 0) {
-            fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
-            exit(1);
-        }
-
-        /* rescale output packet timestamp values from codec to stream timebase */
-        av_packet_rescale_ts(pkt, c->time_base, st->time_base);
-        pkt->stream_index = st->index;
-
-        /* Write the compressed frame to the media file. */
-        log_packet(fmt_ctx, pkt);
-        ret = av_interleaved_write_frame(fmt_ctx, pkt);
-        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
-         * its contents and resets pkt), so that no unreferencing is necessary.
-         * This would be different if one used av_write_frame(). */
-        if (ret < 0) {
-            fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
-            exit(1);
-        }
-    }
-
-    return ret == AVERROR_EOF ? 1 : 0;
 }
 
 /* Add an output stream. */
 static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       const AVCodec **codec,
+                       AVCodec **codec,
                        enum AVCodecID codec_id)
 {
     AVCodecContext *c;
@@ -137,12 +105,6 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
         exit(1);
     }
 
-    ost->tmp_pkt = av_packet_alloc();
-    if (!ost->tmp_pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        exit(1);
-    }
-
     ost->st = avformat_new_stream(oc, NULL);
     if (!ost->st) {
         fprintf(stderr, "Could not allocate stream\n");
@@ -169,7 +131,16 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
                     c->sample_rate = 44100;
             }
         }
-        av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
+        c->channel_layout = AV_CH_LAYOUT_STEREO;
+        if ((*codec)->channel_layouts) {
+            c->channel_layout = (*codec)->channel_layouts[0];
+            for (i = 0; (*codec)->channel_layouts[i]; i++) {
+                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
+                    c->channel_layout = AV_CH_LAYOUT_STEREO;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
         ost->st->time_base = (AVRational){ 1, c->sample_rate };
         break;
 
@@ -199,7 +170,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
              * the motion of the chroma plane does not match the luma plane. */
             c->mb_decision = 2;
         }
-        break;
+    break;
 
     default:
         break;
@@ -214,22 +185,25 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
 /* audio output */
 
 static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
-                                  const AVChannelLayout *channel_layout,
+                                  uint64_t channel_layout,
                                   int sample_rate, int nb_samples)
 {
     AVFrame *frame = av_frame_alloc();
+    int ret;
+
     if (!frame) {
         fprintf(stderr, "Error allocating an audio frame\n");
         exit(1);
     }
 
     frame->format = sample_fmt;
-    av_channel_layout_copy(&frame->ch_layout, channel_layout);
+    frame->channel_layout = channel_layout;
     frame->sample_rate = sample_rate;
     frame->nb_samples = nb_samples;
 
     if (nb_samples) {
-        if (av_frame_get_buffer(frame, 0) < 0) {
+        ret = av_frame_get_buffer(frame, 0);
+        if (ret < 0) {
             fprintf(stderr, "Error allocating an audio buffer\n");
             exit(1);
         }
@@ -238,8 +212,7 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
     return frame;
 }
 
-static void open_audio(AVFormatContext *oc, const AVCodec *codec,
+static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
-                       OutputStream *ost, AVDictionary *opt_arg)
 {
     AVCodecContext *c;
     int nb_samples;
@@ -268,9 +241,9 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
     else
         nb_samples = c->frame_size;
 
-    ost->frame     = alloc_audio_frame(c->sample_fmt, &c->ch_layout,
+    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
                                        c->sample_rate, nb_samples);
-    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, &c->ch_layout,
+    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
                                        c->sample_rate, nb_samples);
 
     /* copy the stream parameters to the muxer */
@@ -281,25 +254,25 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
     }
 
     /* create resampler context */
-    ost->swr_ctx = swr_alloc();
+        ost->swr_ctx = swr_alloc();
-    if (!ost->swr_ctx) {
+        if (!ost->swr_ctx) {
-        fprintf(stderr, "Could not allocate resampler context\n");
+            fprintf(stderr, "Could not allocate resampler context\n");
-        exit(1);
+            exit(1);
-    }
+        }
 
-    /* set options */
+        /* set options */
-    av_opt_set_chlayout  (ost->swr_ctx, "in_chlayout",       &c->ch_layout,      0);
+        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
-    av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-    av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-    av_opt_set_chlayout  (ost->swr_ctx, "out_chlayout",      &c->ch_layout,      0);
+        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
-    av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-    av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
 
-    /* initialize the resampling context */
+        /* initialize the resampling context */
-    if ((ret = swr_init(ost->swr_ctx)) < 0) {
+        if ((ret = swr_init(ost->swr_ctx)) < 0) {
-        fprintf(stderr, "Failed to initialize the resampling context\n");
+            fprintf(stderr, "Failed to initialize the resampling context\n");
-        exit(1);
+            exit(1);
-    }
+        }
 }
 
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -312,12 +285,12 @@ static AVFrame *get_audio_frame(OutputStream *ost)
 
     /* check if we want to generate more frames */
     if (av_compare_ts(ost->next_pts, ost->enc->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
         return NULL;
 
     for (j = 0; j <frame->nb_samples; j++) {
         v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->enc->ch_layout.nb_channels; i++)
+        for (i = 0; i < ost->enc->channels; i++)
             *q++ = v;
         ost->t     += ost->tincr;
         ost->tincr += ost->tincr2;
@@ -336,20 +309,23 @@ static AVFrame *get_audio_frame(OutputStream *ost)
 static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
 {
     AVCodecContext *c;
+    AVPacket pkt = { 0 }; // data and size must be 0;
     AVFrame *frame;
     int ret;
+    int got_packet;
     int dst_nb_samples;
 
+    av_init_packet(&pkt);
     c = ost->enc;
 
     frame = get_audio_frame(ost);
 
     if (frame) {
         /* convert samples from native format to destination codec format, using the resampler */
-        /* compute destination number of samples */
+            /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
-                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
-        av_assert0(dst_nb_samples == frame->nb_samples);
+            av_assert0(dst_nb_samples == frame->nb_samples);
 
         /* when we pass a frame to the encoder, it may keep a reference to it
          * internally;
@@ -373,7 +349,22 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
         ost->samples_count += dst_nb_samples;
     }
 
-    return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
+    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error while writing audio frame: %s\n",
+                    av_err2str(ret));
+            exit(1);
+        }
+    }
+
+    return (frame || got_packet) ? 0 : 1;
 }
 
 /**************************************************************/
@@ -393,7 +384,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
     picture->height = height;
 
     /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(picture, 0);
+    ret = av_frame_get_buffer(picture, 32);
     if (ret < 0) {
         fprintf(stderr, "Could not allocate frame data.\n");
         exit(1);
@@ -402,8 +393,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
     return picture;
 }
 
-static void open_video(AVFormatContext *oc, const AVCodec *codec,
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
-                       OutputStream *ost, AVDictionary *opt_arg)
 {
     int ret;
     AVCodecContext *c = ost->enc;
@@ -474,7 +464,7 @@ static AVFrame *get_video_frame(OutputStream *ost)
 
     /* check if we want to generate more frames */
     if (av_compare_ts(ost->next_pts, c->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
         return NULL;
 
     /* when we pass a frame to the encoder, it may keep a reference to it
@@ -516,7 +506,37 @@ static AVFrame *get_video_frame(OutputStream *ost)
  */
 static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
 {
-    return write_frame(oc, ost->enc, ost->st, get_video_frame(ost), ost->tmp_pkt);
+    int ret;
+    AVCodecContext *c;
+    AVFrame *frame;
+    int got_packet = 0;
+    AVPacket pkt = { 0 };
+
+    c = ost->enc;
+
+    frame = get_video_frame(ost);
+
+    av_init_packet(&pkt);
+
+    /* encode the image */
+    ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+    } else {
+        ret = 0;
+    }
+
+    if (ret < 0) {
+        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    return (frame || got_packet) ? 0 : 1;
 }
 
 static void close_stream(AVFormatContext *oc, OutputStream *ost)
@@ -524,7 +544,6 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
     avcodec_free_context(&ost->enc);
     av_frame_free(&ost->frame);
     av_frame_free(&ost->tmp_frame);
-    av_packet_free(&ost->tmp_pkt);
     sws_freeContext(ost->sws_ctx);
     swr_free(&ost->swr_ctx);
 }
@@ -535,10 +554,10 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
 int main(int argc, char **argv)
 {
     OutputStream video_st = { 0 }, audio_st = { 0 };
-    const AVOutputFormat *fmt;
     const char *filename;
+    AVOutputFormat *fmt;
     AVFormatContext *oc;
-    const AVCodec *audio_codec, *video_codec;
+    AVCodec *audio_codec, *video_codec;
     int ret;
     int have_video = 0, have_audio = 0;
     int encode_video = 0, encode_audio = 0;
@@ -625,6 +644,10 @@ int main(int argc, char **argv)
         }
     }
 
+    /* Write the trailer, if any. The trailer must be written before you
+     * close the CodecContexts open when you wrote the header; otherwise
+     * av_write_trailer() may try to use memory that was freed on
+     * av_codec_close(). */
     av_write_trailer(oc);
 
     /* Close each codec. */

doc/examples/qsv_transcode.c

@@ -1,438 +0,0 @@
-/*
- * 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 Intel QSV-accelerated video transcoding API usage example
- * @example qsv_transcode.c
- *
- * Perform QSV-accelerated transcoding and show to dynamically change
- * encoder's options.
- *
- * Usage: qsv_transcode input_stream codec output_stream initial option
- *                      { frame_number new_option }
- * e.g: - qsv_transcode input.mp4 h264_qsv output_h264.mp4 "g 60"
- *      - qsv_transcode input.mp4 hevc_qsv output_hevc.mp4 "g 60 async_depth 1"
- *                      100 "g 120"
- *         (initialize codec with gop_size 60 and change it to 120 after 100
- *          frames)
- */
-
-#include <stdio.h>
-#include <errno.h>
-
-#include <libavutil/hwcontext.h>
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavutil/opt.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;
-
-typedef struct DynamicSetting {
-    int frame_number;
-    char* optstr;
-} DynamicSetting;
-static DynamicSetting *dynamic_setting;
-static int setting_number;
-static int current_setting_number;
-
-static int str_to_dict(char* optstr, AVDictionary **opt)
-{
-    char *key, *value;
-    if (strlen(optstr) == 0)
-        return 0;
-    key = strtok(optstr, " ");
-    if (key == NULL)
-        return AVERROR(ENAVAIL);
-    value = strtok(NULL, " ");
-    if (value == NULL)
-        return AVERROR(ENAVAIL);
-    av_dict_set(opt, key, value, 0);
-    do {
-        key = strtok(NULL, " ");
-        if (key == NULL)
-            return 0;
-        value = strtok(NULL, " ");
-        if (value == NULL)
-            return AVERROR(ENAVAIL);
-        av_dict_set(opt, key, value, 0);
-    } while(key != NULL);
-    return 0;
-}
-
-static int dynamic_set_parameter(AVCodecContext *avctx)
-{
-    AVDictionary *opts = NULL;
-    int ret = 0;
-    static int frame_number = 0;
-    frame_number++;
-    if (current_setting_number < setting_number &&
-        frame_number == dynamic_setting[current_setting_number].frame_number) {
-        AVDictionaryEntry *e = NULL;
-        ret = str_to_dict(dynamic_setting[current_setting_number].optstr, &opts);
-        if (ret < 0) {
-            fprintf(stderr, "The dynamic parameter is wrong\n");
-            goto fail;
-        }
-        /* Set common option. The dictionary will be freed and replaced
-         * by a new one containing all options not found in common option list.
-         * Then this new dictionary is used to set private option. */
-        if ((ret = av_opt_set_dict(avctx, &opts)) < 0)
-            goto fail;
-        /* Set codec specific option */
-        if ((ret = av_opt_set_dict(avctx->priv_data, &opts)) < 0)
-            goto fail;
-        /* There is no "framerate" option in commom option list. Use "-r" to set
-         * framerate, which is compatible with ffmpeg commandline. The video is
-         * assumed to be average frame rate, so set time_base to 1/framerate. */
-        e = av_dict_get(opts, "r", NULL, 0);
-        if (e) {
-            avctx->framerate = av_d2q(atof(e->value), INT_MAX);
-            encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
-        }
-    }
-fail:
-    av_dict_free(&opts);
-    return ret;
-}
-
-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) {
-            return AV_PIX_FMT_QSV;
-        }
-
-        pix_fmts++;
-    }
-
-    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
-
-    return AV_PIX_FMT_NONE;
-}
-
-static int open_input_file(char *filename)
-{
-    int ret;
-    const 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, NULL, 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;
-    video = ifmt_ctx->streams[video_stream];
-
-    switch(video->codecpar->codec_id) {
-    case AV_CODEC_ID_H264:
-        decoder = avcodec_find_decoder_by_name("h264_qsv");
-        break;
-    case AV_CODEC_ID_HEVC:
-        decoder = avcodec_find_decoder_by_name("hevc_qsv");
-        break;
-    case AV_CODEC_ID_VP9:
-        decoder = avcodec_find_decoder_by_name("vp9_qsv");
-        break;
-    case AV_CODEC_ID_VP8:
-        decoder = avcodec_find_decoder_by_name("vp8_qsv");
-        break;
-    case AV_CODEC_ID_AV1:
-        decoder = avcodec_find_decoder_by_name("av1_qsv");
-        break;
-    case AV_CODEC_ID_MPEG2VIDEO:
-        decoder = avcodec_find_decoder_by_name("mpeg2_qsv");
-        break;
-    case AV_CODEC_ID_MJPEG:
-        decoder = avcodec_find_decoder_by_name("mjpeg_qsv");
-        break;
-    default:
-        fprintf(stderr, "Codec is not supportted by qsv\n");
-        return AVERROR(ENAVAIL);
-    }
-
-    if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
-        return AVERROR(ENOMEM);
-
-    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->framerate = av_guess_frame_rate(ifmt_ctx, video, NULL);
-
-    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_format;
-    decoder_ctx->pkt_timebase = video->time_base;
-    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(AVPacket *enc_pkt, AVFrame *frame)
-{
-    int ret = 0;
-
-    av_packet_unref(enc_pkt);
-
-    if((ret = dynamic_set_parameter(encoder_ctx)) < 0) {
-        fprintf(stderr, "Failed to set dynamic parameter. Error code: %s\n",
-                av_err2str(ret));
-        goto end;
-    }
-
-    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) {
-        if (ret = avcodec_receive_packet(encoder_ctx, enc_pkt))
-            break;
-        enc_pkt->stream_index = 0;
-        av_packet_rescale_ts(enc_pkt, encoder_ctx->time_base,
-                             ofmt_ctx->streams[0]->time_base);
-        if ((ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt)) < 0) {
-            fprintf(stderr, "Error during writing data to output file. "
-                    "Error code: %s\n", av_err2str(ret));
-            return ret;
-        }
-    }
-
-end:
-    if (ret == AVERROR_EOF)
-        return 0;
-    ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
-    return ret;
-}
-
-static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec, char *optstr)
-{
-    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 (!encoder_ctx->hw_frames_ctx) {
-            AVDictionaryEntry *e = NULL;
-            AVDictionary *opts = NULL;
-            AVStream *ost;
-            /* 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.
-             */
-            encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
-            encoder_ctx->pix_fmt   = AV_PIX_FMT_QSV;
-            encoder_ctx->width     = decoder_ctx->width;
-            encoder_ctx->height    = decoder_ctx->height;
-            if ((ret = str_to_dict(optstr, &opts)) < 0) {
-                fprintf(stderr, "Failed to set encoding parameter.\n");
-                goto fail;
-            }
-            /* There is no "framerate" option in commom option list. Use "-r" to
-            * set framerate, which is compatible with ffmpeg commandline. The
-            * video is assumed to be average frame rate, so set time_base to
-            * 1/framerate. */
-            e = av_dict_get(opts, "r", NULL, 0);
-            if (e) {
-                encoder_ctx->framerate = av_d2q(atof(e->value), INT_MAX);
-                encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
-            }
-            if ((ret = avcodec_open2(encoder_ctx, enc_codec, &opts)) < 0) {
-                fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
-                        av_err2str(ret));
-                av_dict_free(&opts);
-                goto fail;
-            }
-            av_dict_free(&opts);
-
-            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;
-            }
-        }
-        frame->pts = av_rescale_q(frame->pts, decoder_ctx->pkt_timebase,
-                                  encoder_ctx->time_base);
-        if ((ret = encode_write(pkt, 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)
-{
-    const AVCodec *enc_codec;
-    int ret = 0;
-    AVPacket *dec_pkt;
-
-    if (argc < 5 || (argc - 5) % 2) {
-        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <encoder> <output file>"
-               " <\"encoding option set 0\"> [<frame_number> <\"encoding options set 1\">]...\n", argv[0]);
-        return 1;
-    }
-    setting_number = (argc - 5) / 2;
-    dynamic_setting = av_malloc(setting_number * sizeof(*dynamic_setting));
-    current_setting_number = 0;
-    for (int i = 0; i < setting_number; i++) {
-        dynamic_setting[i].frame_number = atoi(argv[i*2 + 5]);
-        dynamic_setting[i].optstr = argv[i*2 + 6];
-    }
-
-    ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_QSV, NULL, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Failed to create a QSV device. Error code: %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    dec_pkt = av_packet_alloc();
-    if (!dec_pkt) {
-        fprintf(stderr, "Failed to allocate decode packet\n");
-        goto end;
-    }
-
-    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, argv[4]);
-
-        av_packet_unref(dec_pkt);
-    }
-
-    /* flush decoder */
-    av_packet_unref(dec_pkt);
-    if ((ret = dec_enc(dec_pkt, enc_codec, argv[4])) < 0) {
-        fprintf(stderr, "Failed to flush decoder %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    /* flush encoder */
-    if ((ret = encode_write(dec_pkt, NULL)) < 0) {
-        fprintf(stderr, "Failed to flush encoder %s\n", av_err2str(ret));
-        goto end;
-    }
-
-    /* write the trailer for output stream */
-    if ((ret = av_write_trailer(ofmt_ctx)) < 0)
-        fprintf(stderr, "Failed to write trailer %s\n", av_err2str(ret));
-
-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);
-    av_packet_free(&dec_pkt);
-    av_freep(&dynamic_setting);
-    return ret;
-}

doc/examples/qsvdec.c

@@ -21,11 +21,12 @@
  */
 
 /**
- * @file Intel QSV-accelerated H.264 decoding API usage example
+ * @file
- * @example qsv_decode.c
+ * Intel QSV-accelerated H.264 decoding example.
  *
- * Perform QSV-accelerated H.264 decoding with output frames in the
+ * @example qsvdec.c
- * GPU video surfaces, write the decoded frames to an output file.
+ * This example shows how to do QSV-accelerated H.264 decoding with output
+ * frames in the GPU video surfaces.
  */
 
 #include "config.h"
@@ -43,10 +44,38 @@
 #include "libavutil/hwcontext_qsv.h"
 #include "libavutil/mem.h"
 
+typedef struct DecodeContext {
+    AVBufferRef *hw_device_ref;
+} DecodeContext;
+
 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;
+
+            /* 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;
+
+            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;
+
             return AV_PIX_FMT_QSV;
         }
 
@@ -58,7 +87,7 @@ static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
     return AV_PIX_FMT_NONE;
 }
 
-static int decode_packet(AVCodecContext *decoder_ctx,
+static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
                          AVFrame *frame, AVFrame *sw_frame,
                          AVPacket *pkt, AVIOContext *output_ctx)
 {
@@ -112,15 +141,15 @@ int main(int argc, char **argv)
     AVCodecContext *decoder_ctx = NULL;
     const AVCodec *decoder;
 
-    AVPacket *pkt = NULL;
+    AVPacket pkt = { 0 };
     AVFrame *frame = NULL, *sw_frame = NULL;
 
+    DecodeContext decode = { NULL };
+
     AVIOContext *output_ctx = NULL;
 
     int ret, i;
 
-    AVBufferRef *device_ref = NULL;
-
     if (argc < 3) {
         fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
         return 1;
@@ -148,7 +177,7 @@ int main(int argc, char **argv)
     }
 
     /* open the hardware device */
-    ret = av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV,
+    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");
@@ -180,8 +209,7 @@ int main(int argc, char **argv)
         decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
     }
 
-
+    decoder_ctx->opaque      = &decode;
-    decoder_ctx->hw_device_ctx = av_buffer_ref(device_ref);
     decoder_ctx->get_format  = get_format;
 
     ret = avcodec_open2(decoder_ctx, NULL, NULL);
@@ -199,26 +227,27 @@ int main(int argc, char **argv)
 
     frame    = av_frame_alloc();
     sw_frame = av_frame_alloc();
-    pkt      = av_packet_alloc();
+    if (!frame || !sw_frame) {
-    if (!frame || !sw_frame || !pkt) {
         ret = AVERROR(ENOMEM);
         goto finish;
     }
 
     /* actual decoding */
     while (ret >= 0) {
-        ret = av_read_frame(input_ctx, pkt);
+        ret = av_read_frame(input_ctx, &pkt);
         if (ret < 0)
             break;
 
-        if (pkt->stream_index == video_st->index)
+        if (pkt.stream_index == video_st->index)
-            ret = decode_packet(decoder_ctx, frame, sw_frame, pkt, output_ctx);
+            ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
 
-        av_packet_unref(pkt);
+        av_packet_unref(&pkt);
     }
 
     /* flush the decoder */
-    ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
+    pkt.data = NULL;
+    pkt.size = 0;
+    ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
 
 finish:
     if (ret < 0) {
@@ -231,11 +260,10 @@ finish:
 
     av_frame_free(&frame);
     av_frame_free(&sw_frame);
-    av_packet_free(&pkt);
 
     avcodec_free_context(&decoder_ctx);
 
-    av_buffer_unref(&device_ref);
+    av_buffer_unref(&decode.hw_device_ref);
 
     avio_close(output_ctx);
 

doc/examples/remuxing.c

@@ -21,11 +21,11 @@
  */
 
 /**
- * @file libavformat/libavcodec demuxing and muxing API usage example
+ * @file
- * @example remux.c
+ * libavformat/libavcodec demuxing and muxing API example.
  *
- * Remux streams from one container format to another. Data is copied from the
+ * Remux streams from one container format to another.
- * input to the output without transcoding.
+ * @example remuxing.c
  */
 
 #include <libavutil/timestamp.h>
@@ -45,9 +45,9 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, cons
 
 int main(int argc, char **argv)
 {
-    const AVOutputFormat *ofmt = NULL;
+    AVOutputFormat *ofmt = NULL;
     AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket *pkt = NULL;
+    AVPacket pkt;
     const char *in_filename, *out_filename;
     int ret, i;
     int stream_index = 0;
@@ -65,12 +65,6 @@ int main(int argc, char **argv)
     in_filename  = argv[1];
     out_filename = argv[2];
 
-    pkt = av_packet_alloc();
-    if (!pkt) {
-        fprintf(stderr, "Could not allocate AVPacket\n");
-        return 1;
-    }
-
     if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
         fprintf(stderr, "Could not open input file '%s'", in_filename);
         goto end;
@@ -91,7 +85,7 @@ int main(int argc, char **argv)
     }
 
     stream_mapping_size = ifmt_ctx->nb_streams;
-    stream_mapping = av_calloc(stream_mapping_size, sizeof(*stream_mapping));
+    stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
     if (!stream_mapping) {
         ret = AVERROR(ENOMEM);
         goto end;
@@ -146,39 +140,38 @@ int main(int argc, char **argv)
     while (1) {
         AVStream *in_stream, *out_stream;
 
-        ret = av_read_frame(ifmt_ctx, pkt);
+        ret = av_read_frame(ifmt_ctx, &pkt);
         if (ret < 0)
             break;
 
-        in_stream  = ifmt_ctx->streams[pkt->stream_index];
+        in_stream  = ifmt_ctx->streams[pkt.stream_index];
-        if (pkt->stream_index >= stream_mapping_size ||
+        if (pkt.stream_index >= stream_mapping_size ||
-            stream_mapping[pkt->stream_index] < 0) {
+            stream_mapping[pkt.stream_index] < 0) {
-            av_packet_unref(pkt);
+            av_packet_unref(&pkt);
             continue;
         }
 
-        pkt->stream_index = stream_mapping[pkt->stream_index];
+        pkt.stream_index = stream_mapping[pkt.stream_index];
-        out_stream = ofmt_ctx->streams[pkt->stream_index];
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
-        log_packet(ifmt_ctx, pkt, "in");
+        log_packet(ifmt_ctx, &pkt, "in");
 
         /* copy packet */
-        av_packet_rescale_ts(pkt, in_stream->time_base, out_stream->time_base);
+        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt->pos = -1;
+        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        log_packet(ofmt_ctx, pkt, "out");
+        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
+        pkt.pos = -1;
+        log_packet(ofmt_ctx, &pkt, "out");
 
-        ret = av_interleaved_write_frame(ofmt_ctx, pkt);
+        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
-        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
-         * its contents and resets pkt), so that no unreferencing is necessary.
-         * This would be different if one used av_write_frame(). */
         if (ret < 0) {
             fprintf(stderr, "Error muxing packet\n");
             break;
         }
+        av_packet_unref(&pkt);
     }
 
     av_write_trailer(ofmt_ctx);
 end:
-    av_packet_free(&pkt);
 
     avformat_close_input(&ifmt_ctx);
 

doc/examples/resampling_audio.c

@@ -21,12 +21,8 @@
  */
 
 /**
- * @file audio resampling API usage example
+ * @example resampling_audio.c
- * @example resample_audio.c
+ * libswresample API use example.
- *
- * Generate a synthetic audio signal, and Use libswresample API to perform audio
- * resampling. The output is written to a raw audio file to be played with
- * ffplay.
  */
 
 #include <libavutil/opt.h>
@@ -84,7 +80,7 @@ static void fill_samples(double *dst, int nb_samples, int nb_channels, int sampl
 
 int main(int argc, char **argv)
 {
-    AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
+    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
     int src_rate = 48000, dst_rate = 44100;
     uint8_t **src_data = NULL, **dst_data = NULL;
     int src_nb_channels = 0, dst_nb_channels = 0;
@@ -96,7 +92,6 @@ int main(int argc, char **argv)
     int dst_bufsize;
     const char *fmt;
     struct SwrContext *swr_ctx;
-    char buf[64];
     double t;
     int ret;
 
@@ -125,11 +120,11 @@ int main(int argc, char **argv)
     }
 
     /* set options */
-    av_opt_set_chlayout(swr_ctx, "in_chlayout",    &src_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
     av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
     av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
 
-    av_opt_set_chlayout(swr_ctx, "out_chlayout",    &dst_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
     av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
     av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
 
@@ -141,7 +136,7 @@ int main(int argc, char **argv)
 
     /* allocate source and destination samples buffers */
 
-    src_nb_channels = src_ch_layout.nb_channels;
+    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
     ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
                                              src_nb_samples, src_sample_fmt, 0);
     if (ret < 0) {
@@ -156,7 +151,7 @@ int main(int argc, char **argv)
         av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
 
     /* buffer is going to be directly written to a rawaudio file, no alignment */
-    dst_nb_channels = dst_ch_layout.nb_channels;
+    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
     ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
                                              dst_nb_samples, dst_sample_fmt, 0);
     if (ret < 0) {
@@ -199,10 +194,9 @@ int main(int argc, char **argv)
 
     if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
         goto end;
-    av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
     fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
-            "ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
+            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
-            fmt, buf, dst_nb_channels, dst_rate, dst_filename);
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
 
 end:
     fclose(dst_file);

doc/examples/scaling_video.c

@@ -21,10 +21,9 @@
  */
 
 /**
- * @file libswscale API usage example
+ * @file
- * @example scale_video.c
+ * libswscale API use example.
- *
+ * @example scaling_video.c
- * Generate a synthetic video signal and use libswscale to perform rescaling.
  */
 
 #include <libavutil/imgutils.h>

doc/examples/transcode_aac.c

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2013-2022 Andreas Unterweger
+ * Copyright (c) 2013-2018 Andreas Unterweger
  *
  * This file is part of FFmpeg.
  *
@@ -19,11 +19,12 @@
  */
 
 /**
- * @file audio transcoding to MPEG/AAC API usage example
+ * @file
- * @example transcode_aac.c
+ * Simple audio converter
  *
- * Convert an input audio file to AAC in an MP4 container. Formats other than
+ * @example transcode_aac.c
- * MP4 are supported based on the output file extension.
+ * 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)
  */
 
@@ -37,7 +38,6 @@
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
-#include "libavutil/channel_layout.h"
 #include "libavutil/frame.h"
 #include "libavutil/opt.h"
 
@@ -60,8 +60,7 @@ static int open_input_file(const char *filename,
                            AVCodecContext **input_codec_context)
 {
     AVCodecContext *avctx;
-    const AVCodec *input_codec;
+    AVCodec *input_codec;
-    const AVStream *stream;
     int error;
 
     /* Open the input file to read from it. */
@@ -89,10 +88,8 @@ static int open_input_file(const char *filename,
         return AVERROR_EXIT;
     }
 
-    stream = (*input_format_context)->streams[0];
-
     /* Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
+    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;
@@ -107,7 +104,7 @@ static int open_input_file(const char *filename,
     }
 
     /* Initialize the stream parameters with demuxer information. */
-    error = avcodec_parameters_to_context(avctx, stream->codecpar);
+    error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
     if (error < 0) {
         avformat_close_input(input_format_context);
         avcodec_free_context(&avctx);
@@ -123,9 +120,6 @@ static int open_input_file(const char *filename,
         return error;
     }
 
-    /* Set the packet timebase for the decoder. */
-    avctx->pkt_timebase = stream->time_base;
-
     /* Save the decoder context for easier access later. */
     *input_codec_context = avctx;
 
@@ -150,7 +144,7 @@ static int open_output_file(const char *filename,
     AVCodecContext *avctx          = NULL;
     AVIOContext *output_io_context = NULL;
     AVStream *stream               = NULL;
-    const AVCodec *output_codec    = NULL;
+    AVCodec *output_codec          = NULL;
     int error;
 
     /* Open the output file to write to it. */
@@ -205,11 +199,15 @@ static int open_output_file(const char *filename,
 
     /* Set the basic encoder parameters.
      * The input file's sample rate is used to avoid a sample rate conversion. */
-    av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
+    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. */
+    avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+
     /* Set the sample rate for the container. */
     stream->time_base.den = input_codec_context->sample_rate;
     stream->time_base.num = 1;
@@ -247,16 +245,14 @@ cleanup:
 
 /**
  * Initialize one data packet for reading or writing.
- * @param[out] packet Packet to be initialized
+ * @param packet Packet to be initialized
- * @return Error code (0 if successful)
  */
-static int init_packet(AVPacket **packet)
+static void init_packet(AVPacket *packet)
 {
-    if (!(*packet = av_packet_alloc())) {
+    av_init_packet(packet);
-        fprintf(stderr, "Could not allocate packet\n");
+    /* Set the packet data and size so that it is recognized as being empty. */
-        return AVERROR(ENOMEM);
+    packet->data = NULL;
-    }
+    packet->size = 0;
-    return 0;
 }
 
 /**
@@ -291,18 +287,21 @@ static int init_resampler(AVCodecContext *input_codec_context,
         /*
          * Create a resampler context for the conversion.
          * Set the conversion parameters.
+         * Default channel layouts based on the number of channels
+         * are assumed for simplicity (they are sometimes not detected
+         * properly by the demuxer and/or decoder).
          */
-        error = swr_alloc_set_opts2(resample_context,
+        *resample_context = swr_alloc_set_opts(NULL,
-                                             &output_codec_context->ch_layout,
+                                              av_get_default_channel_layout(output_codec_context->channels),
                                               output_codec_context->sample_fmt,
                                               output_codec_context->sample_rate,
-                                             &input_codec_context->ch_layout,
+                                              av_get_default_channel_layout(input_codec_context->channels),
                                               input_codec_context->sample_fmt,
                                               input_codec_context->sample_rate,
                                               0, NULL);
-        if (error < 0) {
+        if (!*resample_context) {
             fprintf(stderr, "Could not allocate resample context\n");
-            return error;
+            return AVERROR(ENOMEM);
         }
         /*
         * Perform a sanity check so that the number of converted samples is
@@ -330,7 +329,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
     /* 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->ch_layout.nb_channels, 1))) {
+                                      output_codec_context->channels, 1))) {
         fprintf(stderr, "Could not allocate FIFO\n");
         return AVERROR(ENOMEM);
     }
@@ -372,33 +371,28 @@ static int decode_audio_frame(AVFrame *frame,
                               int *data_present, int *finished)
 {
     /* Packet used for temporary storage. */
-    AVPacket *input_packet;
+    AVPacket input_packet;
     int error;
+    init_packet(&input_packet);
 
-    error = init_packet(&input_packet);
-    if (error < 0)
-        return error;
-
-    *data_present = 0;
-    *finished = 0;
     /* Read one audio frame from the input file into a temporary packet. */
-    if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
+    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 (error == AVERROR_EOF)
             *finished = 1;
         else {
             fprintf(stderr, "Could not read frame (error '%s')\n",
                     av_err2str(error));
-            goto cleanup;
+            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) {
+    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));
-        goto cleanup;
+        return error;
     }
 
     /* Receive one frame from the decoder. */
@@ -424,7 +418,7 @@ static int decode_audio_frame(AVFrame *frame,
     }
 
 cleanup:
-    av_packet_free(&input_packet);
+    av_packet_unref(&input_packet);
     return error;
 }
 
@@ -450,7 +444,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
      * Each pointer will later point to the audio samples of the corresponding
      * channels (although it may be NULL for interleaved formats).
      */
-    if (!(*converted_input_samples = calloc(output_codec_context->ch_layout.nb_channels,
+    if (!(*converted_input_samples = calloc(output_codec_context->channels,
                                             sizeof(**converted_input_samples)))) {
         fprintf(stderr, "Could not allocate converted input sample pointers\n");
         return AVERROR(ENOMEM);
@@ -459,7 +453,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
     /* 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->ch_layout.nb_channels,
+                                  output_codec_context->channels,
                                   frame_size,
                                   output_codec_context->sample_fmt, 0)) < 0) {
         fprintf(stderr,
@@ -559,7 +553,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
     AVFrame *input_frame = NULL;
     /* Temporary storage for the converted input samples. */
     uint8_t **converted_input_samples = NULL;
-    int data_present;
+    int data_present = 0;
     int ret = AVERROR_EXIT;
 
     /* Initialize temporary storage for one input frame. */
@@ -633,7 +627,7 @@ static int init_output_frame(AVFrame **frame,
      * Default channel layouts based on the number of channels
      * are assumed for simplicity. */
     (*frame)->nb_samples     = frame_size;
-    av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
+    (*frame)->channel_layout = output_codec_context->channel_layout;
     (*frame)->format         = output_codec_context->sample_fmt;
     (*frame)->sample_rate    = output_codec_context->sample_rate;
 
@@ -667,12 +661,9 @@ static int encode_audio_frame(AVFrame *frame,
                               int *data_present)
 {
     /* Packet used for temporary storage. */
-    AVPacket *output_packet;
+    AVPacket output_packet;
     int error;
-
+    init_packet(&output_packet);
-    error = init_packet(&output_packet);
-    if (error < 0)
-        return error;
 
     /* Set a timestamp based on the sample rate for the container. */
     if (frame) {
@@ -680,20 +671,21 @@ static int encode_audio_frame(AVFrame *frame,
         pts += frame->nb_samples;
     }
 
-    *data_present = 0;
     /* 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);
-    /* Check for errors, but proceed with fetching encoded samples if the
+    /* The encoder signals that it has nothing more to encode. */
-     *  encoder signals that it has nothing more to encode. */
+    if (error == AVERROR_EOF) {
-    if (error < 0 && error != AVERROR_EOF) {
+        error = 0;
-      fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
+        goto cleanup;
-              av_err2str(error));
+    } else if (error < 0) {
-      goto cleanup;
+        fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
+                av_err2str(error));
+        return error;
     }
 
     /* Receive one encoded frame from the encoder. */
-    error = avcodec_receive_packet(output_codec_context, output_packet);
+    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)) {
@@ -714,14 +706,14 @@ static int encode_audio_frame(AVFrame *frame,
 
     /* 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) {
+        (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_free(&output_packet);
+    av_packet_unref(&output_packet);
     return error;
 }
 
@@ -860,6 +852,7 @@ int main(int argc, char **argv)
             int data_written;
             /* 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;

doc/examples/transcoding.c

@@ -23,18 +23,15 @@
  */
 
 /**
- * @file demuxing, decoding, filtering, encoding and muxing API usage example
+ * @file
- * @example transcode.c
+ * API example for demuxing, decoding, filtering, encoding and muxing
- *
+ * @example transcoding.c
- * Convert input to output file, applying some hard-coded filter-graph on both
- * audio and video streams.
  */
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 #include <libavutil/pixdesc.h>
 
@@ -44,17 +41,12 @@ typedef struct FilteringContext {
     AVFilterContext *buffersink_ctx;
     AVFilterContext *buffersrc_ctx;
     AVFilterGraph *filter_graph;
-
-    AVPacket *enc_pkt;
-    AVFrame *filtered_frame;
 } FilteringContext;
 static FilteringContext *filter_ctx;
 
 typedef struct StreamContext {
     AVCodecContext *dec_ctx;
     AVCodecContext *enc_ctx;
-
-    AVFrame *dec_frame;
 } StreamContext;
 static StreamContext *stream_ctx;
 
@@ -74,13 +66,13 @@ static int open_input_file(const char *filename)
         return ret;
     }
 
-    stream_ctx = av_calloc(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
+    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];
-        const AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
+        AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
         AVCodecContext *codec_ctx;
         if (!dec) {
             av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
@@ -110,10 +102,6 @@ static int open_input_file(const char *filename)
             }
         }
         stream_ctx[i].dec_ctx = codec_ctx;
-
-        stream_ctx[i].dec_frame = av_frame_alloc();
-        if (!stream_ctx[i].dec_frame)
-            return AVERROR(ENOMEM);
     }
 
     av_dump_format(ifmt_ctx, 0, filename, 0);
@@ -125,7 +113,7 @@ static int open_output_file(const char *filename)
     AVStream *out_stream;
     AVStream *in_stream;
     AVCodecContext *dec_ctx, *enc_ctx;
-    const AVCodec *encoder;
+    AVCodec *encoder;
     int ret;
     unsigned int i;
 
@@ -177,9 +165,8 @@ static int open_output_file(const char *filename)
                 enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
             } else {
                 enc_ctx->sample_rate = dec_ctx->sample_rate;
-                ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
+                enc_ctx->channel_layout = dec_ctx->channel_layout;
-                if (ret < 0)
+                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
-                    return ret;
                 /* take first format from list of supported formats */
                 enc_ctx->sample_fmt = encoder->sample_fmts[0];
                 enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
@@ -292,7 +279,6 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
             goto end;
         }
     } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        char buf[64];
         buffersrc = avfilter_get_by_name("abuffer");
         buffersink = avfilter_get_by_name("abuffersink");
         if (!buffersrc || !buffersink) {
@@ -301,14 +287,14 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
             goto end;
         }
 
-        if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
+        if (!dec_ctx->channel_layout)
-            av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
+            dec_ctx->channel_layout =
-        av_channel_layout_describe(&dec_ctx->ch_layout, buf, sizeof(buf));
+                av_get_default_channel_layout(dec_ctx->channels);
         snprintf(args, sizeof(args),
-                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
+                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
                 dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
                 av_get_sample_fmt_name(dec_ctx->sample_fmt),
-                buf);
+                dec_ctx->channel_layout);
         ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                 args, NULL, filter_graph);
         if (ret < 0) {
@@ -331,9 +317,9 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
             goto end;
         }
 
-        av_channel_layout_describe(&enc_ctx->ch_layout, buf, sizeof(buf));
+        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
-        ret = av_opt_set(buffersink_ctx, "ch_layouts",
+                (uint8_t*)&enc_ctx->channel_layout,
-                         buf, AV_OPT_SEARCH_CHILDREN);
+                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
         if (ret < 0) {
             av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
             goto end;
@@ -412,63 +398,54 @@ static int init_filters(void)
                 stream_ctx[i].enc_ctx, filter_spec);
         if (ret)
             return ret;
-
-        filter_ctx[i].enc_pkt = av_packet_alloc();
-        if (!filter_ctx[i].enc_pkt)
-            return AVERROR(ENOMEM);
-
-        filter_ctx[i].filtered_frame = av_frame_alloc();
-        if (!filter_ctx[i].filtered_frame)
-            return AVERROR(ENOMEM);
     }
     return 0;
 }
 
-static int encode_write_frame(unsigned int stream_index, int flush)
+static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
-{
-    StreamContext *stream = &stream_ctx[stream_index];
-    FilteringContext *filter = &filter_ctx[stream_index];
-    AVFrame *filt_frame = flush ? NULL : filter->filtered_frame;
-    AVPacket *enc_pkt = filter->enc_pkt;
     int ret;
+    int got_frame_local;
+    AVPacket enc_pkt;
+    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
+        (ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
+         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
+
+    if (!got_frame)
+        got_frame = &got_frame_local;
 
     av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
     /* encode filtered frame */
-    av_packet_unref(enc_pkt);
+    enc_pkt.data = NULL;
-
+    enc_pkt.size = 0;
-    ret = avcodec_send_frame(stream->enc_ctx, filt_frame);
+    av_init_packet(&enc_pkt);
-
+    ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
+            filt_frame, got_frame);
+    av_frame_free(&filt_frame);
     if (ret < 0)
         return ret;
+    if (!(*got_frame))
+        return 0;
 
-    while (ret >= 0) {
+    /* prepare packet for muxing */
-        ret = avcodec_receive_packet(stream->enc_ctx, enc_pkt);
+    enc_pkt.stream_index = stream_index;
-
+    av_packet_rescale_ts(&enc_pkt,
-        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                         stream_ctx[stream_index].enc_ctx->time_base,
-            return 0;
+                         ofmt_ctx->streams[stream_index]->time_base);
-
-        /* prepare packet for muxing */
-        enc_pkt->stream_index = stream_index;
-        av_packet_rescale_ts(enc_pkt,
-                             stream->enc_ctx->time_base,
-                             ofmt_ctx->streams[stream_index]->time_base);
-
-        av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
-        /* mux encoded frame */
-        ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt);
-    }
 
+    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
+    /* mux encoded frame */
+    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
     return ret;
 }
 
 static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 {
-    FilteringContext *filter = &filter_ctx[stream_index];
     int ret;
+    AVFrame *filt_frame;
 
     av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
     /* push the decoded frame into the filtergraph */
-    ret = av_buffersrc_add_frame_flags(filter->buffersrc_ctx,
+    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
             frame, 0);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
@@ -477,9 +454,14 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 
     /* pull filtered frames from the filtergraph */
     while (1) {
+        filt_frame = av_frame_alloc();
+        if (!filt_frame) {
+            ret = AVERROR(ENOMEM);
+            break;
+        }
         av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
-        ret = av_buffersink_get_frame(filter->buffersink_ctx,
+        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
-                                      filter->filtered_frame);
+                filt_frame);
         if (ret < 0) {
             /* if no more frames for output - returns AVERROR(EAGAIN)
              * if flushed and no more frames for output - returns AVERROR_EOF
@@ -487,12 +469,12 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
              */
             if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                 ret = 0;
+            av_frame_free(&filt_frame);
             break;
         }
 
-        filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
+        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
-        ret = encode_write_frame(stream_index, 0);
+        ret = encode_write_frame(filt_frame, stream_index, NULL);
-        av_frame_unref(filter->filtered_frame);
         if (ret < 0)
             break;
     }
@@ -502,20 +484,34 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
 
 static int flush_encoder(unsigned int stream_index)
 {
+    int ret;
+    int got_frame;
+
     if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
                 AV_CODEC_CAP_DELAY))
         return 0;
 
-    av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
+    while (1) {
-    return encode_write_frame(stream_index, 1);
+        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
+        ret = encode_write_frame(NULL, stream_index, &got_frame);
+        if (ret < 0)
+            break;
+        if (!got_frame)
+            return 0;
+    }
+    return ret;
 }
 
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket *packet = NULL;
+    AVPacket packet = { .data = NULL, .size = 0 };
+    AVFrame *frame = NULL;
+    enum AVMediaType type;
     unsigned int stream_index;
     unsigned int i;
+    int got_frame;
+    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
 
     if (argc != 3) {
         av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -528,54 +524,56 @@ int main(int argc, char **argv)
         goto end;
     if ((ret = init_filters()) < 0)
         goto end;
-    if (!(packet = av_packet_alloc()))
-        goto end;
 
     /* read all packets */
     while (1) {
-        if ((ret = av_read_frame(ifmt_ctx, packet)) < 0)
+        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
             break;
-        stream_index = packet->stream_index;
+        stream_index = packet.stream_index;
+        type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
         av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
                 stream_index);
 
         if (filter_ctx[stream_index].filter_graph) {
-            StreamContext *stream = &stream_ctx[stream_index];
-
             av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
-
+            frame = av_frame_alloc();
-            av_packet_rescale_ts(packet,
+            if (!frame) {
+                ret = AVERROR(ENOMEM);
+                break;
+            }
+            av_packet_rescale_ts(&packet,
                                  ifmt_ctx->streams[stream_index]->time_base,
-                                 stream->dec_ctx->time_base);
+                                 stream_ctx[stream_index].dec_ctx->time_base);
-            ret = avcodec_send_packet(stream->dec_ctx, packet);
+            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
+                avcodec_decode_audio4;
+            ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
+                    &got_frame, &packet);
             if (ret < 0) {
+                av_frame_free(&frame);
                 av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
                 break;
             }
 
-            while (ret >= 0) {
+            if (got_frame) {
-                ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
+                frame->pts = frame->best_effort_timestamp;
-                if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
+                ret = filter_encode_write_frame(frame, stream_index);
-                    break;
+                av_frame_free(&frame);
-                else if (ret < 0)
-                    goto end;
-
-                stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
-                ret = filter_encode_write_frame(stream->dec_frame, stream_index);
                 if (ret < 0)
                     goto end;
+            } else {
+                av_frame_free(&frame);
             }
         } else {
             /* remux this frame without reencoding */
-            av_packet_rescale_ts(packet,
+            av_packet_rescale_ts(&packet,
                                  ifmt_ctx->streams[stream_index]->time_base,
                                  ofmt_ctx->streams[stream_index]->time_base);
 
-            ret = av_interleaved_write_frame(ofmt_ctx, packet);
+            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
             if (ret < 0)
                 goto end;
         }
-        av_packet_unref(packet);
+        av_packet_unref(&packet);
     }
 
     /* flush filters and encoders */
@@ -599,18 +597,14 @@ int main(int argc, char **argv)
 
     av_write_trailer(ofmt_ctx);
 end:
-    av_packet_free(&packet);
+    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);
-        if (filter_ctx && filter_ctx[i].filter_graph) {
+        if (filter_ctx && filter_ctx[i].filter_graph)
             avfilter_graph_free(&filter_ctx[i].filter_graph);
-            av_packet_free(&filter_ctx[i].enc_pkt);
-            av_frame_free(&filter_ctx[i].filtered_frame);
-        }
-
-        av_frame_free(&stream_ctx[i].dec_frame);
     }
     av_free(filter_ctx);
     av_free(stream_ctx);

doc/examples/vaapi_encode.c

@@ -1,4 +1,6 @@
 /*
+ * Video Acceleration API (video encoding) encode sample
+ *
  * 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
@@ -19,12 +21,13 @@
  */
 
 /**
- * @file Intel VAAPI-accelerated encoding API usage example
+ * @file
- * @example vaapi_encode.c
+ * 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
  *
- * Perform VAAPI-accelerated encoding. Read input from an NV12 raw
- * file, and write the H.264 encoded data to an output raw file.
- * Usage: vaapi_encode 1920 1080 input.yuv output.h264
  */
 
 #include <stdio.h>
@@ -71,27 +74,27 @@ static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
 static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
 {
     int ret = 0;
-    AVPacket *enc_pkt;
+    AVPacket enc_pkt;
 
-    if (!(enc_pkt = av_packet_alloc()))
+    av_init_packet(&enc_pkt);
-        return AVERROR(ENOMEM);
+    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);
+        ret = avcodec_receive_packet(avctx, &enc_pkt);
         if (ret)
             break;
 
-        enc_pkt->stream_index = 0;
+        enc_pkt.stream_index = 0;
-        ret = fwrite(enc_pkt->data, enc_pkt->size, 1, fout);
+        ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
-        av_packet_unref(enc_pkt);
+        av_packet_unref(&enc_pkt);
     }
 
 end:
-    av_packet_free(&enc_pkt);
     ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
     return ret;
 }
@@ -102,7 +105,7 @@ int main(int argc, char *argv[])
     FILE *fin = NULL, *fout = NULL;
     AVFrame *sw_frame = NULL, *hw_frame = NULL;
     AVCodecContext *avctx = NULL;
-    const AVCodec *codec = NULL;
+    AVCodec *codec = NULL;
     const char *enc_name = "h264_vaapi";
 
     if (argc < 5) {
@@ -169,7 +172,7 @@ int main(int argc, char *argv[])
         sw_frame->width  = width;
         sw_frame->height = height;
         sw_frame->format = AV_PIX_FMT_NV12;
-        if ((err = av_frame_get_buffer(sw_frame, 0)) < 0)
+        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;

doc/examples/vaapi_transcode.c

@@ -1,4 +1,6 @@
 /*
+ * Video Acceleration API (video transcoding) transcode sample
+ *
  * 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
@@ -19,10 +21,11 @@
  */
 
 /**
- * @file Intel VAAPI-accelerated transcoding API usage example
+ * @file
- * @example vaapi_transcode.c
+ * Intel VAAPI-accelerated transcoding example.
  *
- * Perform VAAPI-accelerated transcoding.
+ * @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
@@ -59,7 +62,7 @@ static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
 static int open_input_file(const char *filename)
 {
     int ret;
-    const AVCodec *decoder = NULL;
+    AVCodec *decoder = NULL;
     AVStream *video = NULL;
 
     if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
@@ -106,25 +109,28 @@ static int open_input_file(const char *filename)
     return ret;
 }
 
-static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
+static int encode_write(AVFrame *frame)
 {
     int ret = 0;
+    AVPacket enc_pkt;
 
-    av_packet_unref(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);
+        ret = avcodec_receive_packet(encoder_ctx, &enc_pkt);
         if (ret)
             break;
 
-        enc_pkt->stream_index = 0;
+        enc_pkt.stream_index = 0;
-        av_packet_rescale_ts(enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
+        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);
+        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));
@@ -139,7 +145,7 @@ end:
     return ret;
 }
 
-static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)
+static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
 {
     AVFrame *frame;
     int ret = 0;
@@ -210,7 +216,7 @@ static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec)
             initialized = 1;
         }
 
-        if ((ret = encode_write(pkt, frame)) < 0)
+        if ((ret = encode_write(frame)) < 0)
             fprintf(stderr, "Error during encoding and writing.\n");
 
 fail:
@@ -223,9 +229,9 @@ fail:
 
 int main(int argc, char **argv)
 {
-    const AVCodec *enc_codec;
     int ret = 0;
-    AVPacket *dec_pkt;
+    AVPacket dec_pkt;
+    AVCodec *enc_codec;
 
     if (argc != 4) {
         fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
@@ -240,12 +246,6 @@ int main(int argc, char **argv)
         return -1;
     }
 
-    dec_pkt = av_packet_alloc();
-    if (!dec_pkt) {
-        fprintf(stderr, "Failed to allocate decode packet\n");
-        goto end;
-    }
-
     if ((ret = open_input_file(argv[1])) < 0)
         goto end;
 
@@ -275,21 +275,23 @@ int main(int argc, char **argv)
 
     /* read all packets and only transcoding video */
     while (ret >= 0) {
-        if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 0)
+        if ((ret = av_read_frame(ifmt_ctx, &dec_pkt)) < 0)
             break;
 
-        if (video_stream == dec_pkt->stream_index)
+        if (video_stream == dec_pkt.stream_index)
-            ret = dec_enc(dec_pkt, enc_codec);
+            ret = dec_enc(&dec_pkt, enc_codec);
 
-        av_packet_unref(dec_pkt);
+        av_packet_unref(&dec_pkt);
     }
 
     /* flush decoder */
-    av_packet_unref(dec_pkt);
+    dec_pkt.data = NULL;
-    ret = dec_enc(dec_pkt, enc_codec);
+    dec_pkt.size = 0;
+    ret = dec_enc(&dec_pkt, enc_codec);
+    av_packet_unref(&dec_pkt);
 
     /* flush encoder */
-    ret = encode_write(dec_pkt, NULL);
+    ret = encode_write(NULL);
 
     /* write the trailer for output stream */
     av_write_trailer(ofmt_ctx);
@@ -300,6 +302,5 @@ end:
     avcodec_free_context(&decoder_ctx);
     avcodec_free_context(&encoder_ctx);
     av_buffer_unref(&hw_device_ctx);
-    av_packet_free(&dec_pkt);
     return ret;
 }

doc/fate.texi

@@ -79,21 +79,6 @@ Do not put a '~' character in the samples path to indicate a home
 directory. Because of shell nuances, this will cause FATE to fail.
 @end float
 
-To get the complete list of tests, run the command:
-@example
-make fate-list
-@end example
-
-You can specify a subset of tests to run by specifying the
-corresponding elements from the list with the @code{fate-} prefix,
-e.g. as in:
-@example
-make fate-ffprobe_compact fate-ffprobe_xml
-@end example
-
-This makes it easier to run a few tests in case of failure without
-running the complete test suite.
-
 To use a custom wrapper to run the test, pass @option{--target-exec} to
 @command{configure} or set the @var{TARGET_EXEC} Make variable.
 
@@ -164,8 +149,6 @@ the synchronisation of the samples directory.
 
 @chapter Uploading new samples to the fate suite
 
-If you need a sample uploaded send a mail to samples-request.
-
 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.
@@ -174,8 +157,6 @@ practice generally do not replace, remove or overwrite files as it likely would
 break older checkouts or releases.
 Also all needed samples for a commit should be uploaded, ideally 24
 hours, before the push.
-If you need an account for frequently uploading samples or you wish to help
-others by doing that send a mail to ffmpeg-devel.
 
 @example
 #First update your local samples copy:

doc/ffmpeg.texi

@@ -449,11 +449,6 @@ output file already exists.
 Set number of times input stream shall be looped. Loop 0 means no loop,
 loop -1 means infinite loop.
 
-@item -recast_media (@emph{global})
-Allow forcing a decoder of a different media type than the one
-detected or designated by the demuxer. Useful for decoding media
-data muxed as data streams.
-
 @item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 @itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
 Select an encoder (when used before an output file) or a decoder (when used
@@ -518,21 +513,6 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
 Like the @code{-ss} option but relative to the "end of file". That is negative
 values are earlier in the file, 0 is at EOF.
 
-@item -isync @var{input_index} (@emph{input})
-Assign an input as a sync source.
-
-This will take the difference between the start times of the target and reference inputs and
-offset the timestamps of the target file by that difference. The source timestamps of the two
-inputs should derive from the same clock source for expected results. If @code{copyts} is set
-then @code{start_at_zero} must also be set. If either of the inputs has no starting timestamp
-then no sync adjustment is made.
-
-Acceptable values are those that refer to a valid ffmpeg input index. If the sync reference is
-the target index itself or @var{-1}, then no adjustment is made to target timestamps. A sync
-reference may not itself be synced to any other input.
-
-Default value is @var{-1}.
-
 @item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.
 
@@ -575,22 +555,27 @@ ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
 @item -disposition[:stream_specifier] @var{value} (@emph{output,per-stream})
 Sets the disposition for a stream.
 
-By default, the disposition is copied from the input stream, unless the output
+This option overrides the disposition copied from the input stream. It is also
-stream this option applies to is fed by a complex filtergraph - in that case the
+possible to delete the disposition by setting it to 0.
-disposition is unset by default.
 
-@var{value} is a sequence of items separated by '+' or '-'. The first item may
+The following dispositions are recognized:
-also be prefixed with '+' or '-', in which case this option modifies the default
+@table @option
-value. Otherwise (the first item is not prefixed) this options overrides the
+@item default
-default value. A '+' prefix adds the given disposition, '-' removes it. It is
+@item dub
-also possible to clear the disposition by setting it to 0.
+@item original
-
+@item comment
-If no @code{-disposition} options were specified for an output file, ffmpeg will
+@item lyrics
-automatically set the 'default' disposition on the first stream of each type,
+@item karaoke
-when there are multiple streams of this type in the output file and no stream of
+@item forced
-that type is already marked as default.
+@item hearing_impaired
-
+@item visual_impaired
-The @code{-dispositions} option lists the known dispositions.
+@item clean_effects
+@item attached_pic
+@item captions
+@item descriptions
+@item dependent
+@item metadata
+@end table
 
 For example, to make the second audio stream the default stream:
 @example
@@ -632,102 +617,6 @@ they do not conflict with the standard, as in:
 ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
 @end example
 
-The parameters set for each target are as follows.
-
-@strong{VCD}
-@example
-@var{pal}:
--f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
--s 352x288 -r 25
--codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
--ar 44100 -ac 2
--codec:a mp2 -b:a 224k
-
-@var{ntsc}:
--f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
--s 352x240 -r 30000/1001
--codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
--ar 44100 -ac 2
--codec:a mp2 -b:a 224k
-
-@var{film}:
--f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
--s 352x240 -r 24000/1001
--codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
--ar 44100 -ac 2
--codec:a mp2 -b:a 224k
-@end example
-
-@strong{SVCD}
-@example
-@var{pal}:
--f svcd -packetsize 2324
--s 480x576 -pix_fmt yuv420p -r 25
--codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
--ar 44100
--codec:a mp2 -b:a 224k
-
-@var{ntsc}:
--f svcd -packetsize 2324
--s 480x480 -pix_fmt yuv420p -r 30000/1001
--codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
--ar 44100
--codec:a mp2 -b:a 224k
-
-@var{film}:
--f svcd -packetsize 2324
--s 480x480 -pix_fmt yuv420p -r 24000/1001
--codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
--ar 44100
--codec:a mp2 -b:a 224k
-@end example
-
-@strong{DVD}
-@example
-@var{pal}:
--f dvd -muxrate 10080k -packetsize 2048
--s 720x576 -pix_fmt yuv420p -r 25
--codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
--ar 48000
--codec:a ac3 -b:a 448k
-
-@var{ntsc}:
--f dvd -muxrate 10080k -packetsize 2048
--s 720x480 -pix_fmt yuv420p -r 30000/1001
--codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
--ar 48000
--codec:a ac3 -b:a 448k
-
-@var{film}:
--f dvd -muxrate 10080k -packetsize 2048
--s 720x480 -pix_fmt yuv420p -r 24000/1001
--codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
--ar 48000
--codec:a ac3 -b:a 448k
-@end example
-
-@strong{DV}
-@example
-@var{pal}:
--f dv
--s 720x576 -pix_fmt yuv420p -r 25
--ar 48000 -ac 2
-
-@var{ntsc}:
--f dv
--s 720x480 -pix_fmt yuv411p -r 30000/1001
--ar 48000 -ac 2
-
-@var{film}:
--f dv
--s 720x480 -pix_fmt yuv411p -r 24000/1001
--ar 48000 -ac 2
-@end example
-The @code{dv50} target is identical to the @code{dv} target except that the pixel format set is @code{yuv422p} for all three standards.
-
-Any user-set value for a parameter above will override the target preset value. In that case, the output may
-not comply with the target standard.
-
 @item -dn (@emph{input/output})
 As an input option, blocks all data streams of a file from being filtered or
 being automatically selected or mapped for any output. See @code{-discard}
@@ -774,16 +663,6 @@ 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 -reinit_filter[:@var{stream_specifier}] @var{integer} (@emph{input,per-stream})
-This boolean option determines if the filtergraph(s) to which this stream is fed gets
-reinitialized when input frame parameters change mid-stream. This option is enabled by
-default as most video and all audio filters cannot handle deviation in input frame properties.
-Upon reinitialization, existing filter state is lost, like e.g. the frame count @code{n}
-reference available in some filters. Any frames buffered at time of reinitialization are lost.
-The properties where a change triggers reinitialization are,
-for video, frame resolution or pixel format;
-for audio, sample format, sample rate, channel count or channel layout.
-
 @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.
@@ -796,19 +675,14 @@ Specify the preset for matching stream(s).
 Print encoding progress/statistics. It is on by default, to explicitly
 disable it you need to specify @code{-nostats}.
 
-@item -stats_period @var{time} (@emph{global})
-Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.
-
 @item -progress @var{url} (@emph{global})
 Send program-friendly progress information to @var{url}.
 
-Progress information is written periodically and at the end of
+Progress information is written approximately every second and at the end of
 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".
 
-The update period is set using @code{-stats_period}.
-
 @anchor{stdin option}
 @item -stdin
 Enable interaction on standard input. On by default unless standard input is
@@ -860,6 +734,10 @@ ffmpeg -dump_attachment:t "" -i INPUT
 Technical note -- attachments are implemented as codec extradata, so this
 option can actually be used to extract extradata from any stream, not just
 attachments.
+
+@item -noautorotate
+Disable automatically rotating video based on file metadata.
+
 @end table
 
 @section Video Options
@@ -877,27 +755,9 @@ This is not the same as the @option{-framerate} option used for some input forma
 like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
 If in doubt use @option{-framerate} instead of the input option @option{-r}.
 
-As an output option:
+As an output option, duplicate or drop input frames to achieve constant output
-@table @option
-@item video encoding
-Duplicate or drop frames right before encoding them to achieve constant output
 frame rate @var{fps}.
 
-@item video streamcopy
-Indicate to the muxer that @var{fps} is the stream frame rate. No data is
-dropped or duplicated in this case. This may produce invalid files if @var{fps}
-does not match the actual stream frame rate as determined by packet timestamps.
-See also the @code{setts} bitstream filter.
-
-@end table
-
-@item -fpsmax[:@var{stream_specifier}] @var{fps} (@emph{output,per-stream})
-Set maximum frame rate (Hz value, fraction or abbreviation).
-
-Clamps output frame rate when output framerate is auto-set and is higher than this value.
-Useful in batch processing or when input framerate is wrongly detected as very high.
-It cannot be set together with @code{-r}. It is ignored during streamcopy.
-
 @item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
 Set frame size.
 
@@ -923,32 +783,6 @@ If used together with @option{-vcodec copy}, it will affect the aspect ratio
 stored at container level, but not the aspect ratio stored in encoded
 frames, if it exists.
 
-@item -display_rotation[:@var{stream_specifier}] @var{rotation} (@emph{input,per-stream})
-Set video rotation metadata.
-
-@var{rotation} is a decimal number specifying the amount in degree by
-which the video should be rotated counter-clockwise before being
-displayed.
-
-This option overrides the rotation/display transform metadata stored in
-the file, if any. When the video is being transcoded (rather than
-copied) and @code{-autorotate} is enabled, the video will be rotated at
-the filtering stage. Otherwise, the metadata will be written into the
-output file if the muxer supports it.
-
-If the @code{-display_hflip} and/or @code{-display_vflip} options are
-given, they are applied after the rotation specified by this option.
-
-@item -display_hflip[:@var{stream_specifier}] (@emph{input,per-stream})
-Set whether on display the image should be horizontally flipped.
-
-See the @code{-display_rotation} option for more details.
-
-@item -display_vflip[:@var{stream_specifier}] (@emph{input,per-stream})
-Set whether on display the image should be vertically flipped.
-
-See the @code{-display_rotation} option for more details.
-
 @item -vn (@emph{input/output})
 As an input option, blocks all video streams of a file from being filtered or
 being automatically selected or mapped for any output. See @code{-discard}
@@ -985,18 +819,6 @@ Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.
 
 This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
-
-@item -autorotate
-Automatically rotate the video according to file metadata. Enabled by
-default, use @option{-noautorotate} to disable it.
-
-@item -autoscale
-Automatically scale the video according to the resolution of first frame.
-Enabled by default, use @option{-noautoscale} to disable it. When autoscale is
-disabled, all output frames of filter graph might not be in the same resolution
-and may be inadequate for some encoder/muxer. Therefore, it is not recommended
-to disable it unless you really know what you are doing.
-Disable autoscale at your own risk.
 @end table
 
 @section Advanced Video options
@@ -1022,9 +844,14 @@ list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.
 
+@item -ilme
+Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
+Use this option if your input file is interlaced and you want
+to keep the interlaced format for minimum losses.
+The alternative is to deinterlace the input stream with
+@option{-deinterlace}, but deinterlacing introduces losses.
 @item -psnr
-Calculate PSNR of compressed frames. This option is deprecated, pass the
+Calculate PSNR of compressed frames.
-PSNR flag to the encoder instead, using @code{-flags +psnr}.
 @item -vstats
 Dump video coding statistics to @file{vstats_HHMMSS.log}.
 @item -vstats_file @var{file}
@@ -1041,6 +868,8 @@ 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}
+Intra_dc_precision.
 @item -vtag @var{fourcc/tag} (@emph{output})
 Force video tag/fourcc. This is an alias for @code{-tag:v}.
 @item -qphist (@emph{global})
@@ -1050,20 +879,12 @@ Deprecated see -bsf
 
 @item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
 @item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
-@item -force_key_frames[:@var{stream_specifier}] source (@emph{output,per-stream})
+Force key frames at the specified timestamps, more precisely at the first
-@item -force_key_frames[:@var{stream_specifier}] source_no_drop (@emph{output,per-stream})
+frames after each specified time.
 
-@var{force_key_frames} can take arguments of the following form:
+If the argument is prefixed with @code{expr:}, the string @var{expr}
-
+is interpreted like an expression and is evaluated for each frame. A
-@table @option
+key frame is forced in case the evaluation is non-zero.
-
-@item @var{time}[,@var{time}...]
-If the argument consists of timestamps, ffmpeg will round the specified times to the nearest
-output timestamp as per the encoder time base and force a keyframe at the first frame having
-timestamp equal or greater than the computed timestamp. Note that if the encoder time base is too
-coarse, then the keyframes may be forced on frames with timestamps lower than the specified time.
-The default encoder time base is the inverse of the output framerate but may be set otherwise
-via @code{-enc_time_base}.
 
 If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
 the time of the beginning of all chapters in the file, shifted by
@@ -1077,11 +898,6 @@ before the beginning of every chapter:
 -force_key_frames 0:05:00,chapters-0.1
 @end example
 
-@item expr:@var{expr}
-If the argument is prefixed with @code{expr:}, the string @var{expr}
-is interpreted like an expression and is evaluated for each frame. A
-key frame is forced in case the evaluation is non-zero.
-
 The expression in @var{expr} can contain the following constants:
 @table @option
 @item n
@@ -1109,18 +925,6 @@ starting from second 13:
 -force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
 @end example
 
-@item source
-If the argument is @code{source}, ffmpeg will force a key frame if
-the current frame being encoded is marked as a key frame in its source.
-
-@item source_no_drop
-If the argument is @code{source_no_drop}, ffmpeg will force a key frame if
-the current frame being encoded is marked as a key frame in its source.
-In cases where this particular source frame has to be dropped,
-enforce the next available frame to become a key frame instead.
-
-@end table
-
 Note that forcing too many keyframes is very harmful for the lookahead
 algorithms of certain encoders: using fixed-GOP options or similar
 would be more efficient.
@@ -1141,27 +945,9 @@ device type:
 @item cuda
 @var{device} is the number of the CUDA device.
 
-The following options are recognized:
-@table @option
-@item primary_ctx
-If set to 1, uses the primary device context instead of creating a new one.
-@end table
-
-Examples:
-@table @emph
-@item -init_hw_device cuda:1
-Choose the second device on the system.
-
-@item -init_hw_device cuda:0,primary_ctx=1
-Choose the first device and use the primary device context.
-@end table
-
 @item dxva2
 @var{device} is the number of the Direct3D 9 display adapter.
 
-@item d3d11va
-@var{device} is the number of the Direct3D 11 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{$DISPLAY})
@@ -1185,21 +971,9 @@ If not specified, it will attempt to open the default X11 display (@emph{$DISPLA
 @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{d3d11va} or @samp{vaapi}) and then deriving a
+platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
 QSV device from that.)
 
-Alternatively, @samp{child_device_type} helps to choose platform-appropriate subdevice type.
-On Windows @samp{d3d11va} is used as default subdevice type.
-
-Examples:
-@table @emph
-@item -init_hw_device qsv:hw,child_device_type=d3d11va
-Choose the GPU subdevice with type @samp{d3d11va} and create QSV device with @samp{MFX_IMPL_HARDWARE}.
-
-@item -init_hw_device qsv:hw,child_device_type=dxva2
-Choose the GPU subdevice with type @samp{dxva2} and create QSV device with @samp{MFX_IMPL_HARDWARE}.
-@end table
-
 @item opencl
 @var{device} selects the platform and device as @emph{platform_index.device_index}.
 
@@ -1237,35 +1011,6 @@ Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
 extension.
 @end table
 
-@item vulkan
-If @var{device} is an integer, it selects the device by its index in a
-system-dependent list of devices.  If @var{device} is any other string, it
-selects the first device with a name containing that string as a substring.
-
-The following options are recognized:
-@table @option
-@item debug
-If set to 1, enables the validation layer, if installed.
-@item linear_images
-If set to 1, images allocated by the hwcontext will be linear and locally mappable.
-@item instance_extensions
-A plus separated list of additional instance extensions to enable.
-@item device_extensions
-A plus separated list of additional device extensions to enable.
-@end table
-
-Examples:
-@table @emph
-@item -init_hw_device vulkan:1
-Choose the second device on the system.
-
-@item -init_hw_device vulkan:RADV
-Choose the first device with a name containing the string @emph{RADV}.
-
-@item -init_hw_device vulkan:0,instance_extensions=VK_KHR_wayland_surface+VK_KHR_xcb_surface
-Choose the first device and enable the Wayland and XCB instance extensions.
-@end table
-
 @end table
 
 @item -init_hw_device @var{type}[=@var{name}]@@@var{source}
@@ -1302,9 +1047,6 @@ Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
 @item dxva2
 Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
 
-@item d3d11va
-Use D3D11VA (DirectX Video Acceleration) hardware acceleration.
-
 @item vaapi
 Use VAAPI (Video Acceleration API) hardware acceleration.
 
@@ -1338,25 +1080,7 @@ by name, or it can create a new device as if
 were called immediately before.
 
 @item -hwaccels
-List all hardware acceleration components enabled in this build of ffmpeg.
+List all hardware acceleration methods supported in this build of ffmpeg.
-Actual runtime availability depends on the hardware and its suitable driver
-being installed.
-
-@item -fix_sub_duration_heartbeat[:@var{stream_specifier}]
-Set a specific output video stream as the heartbeat stream according to which
-to split and push through currently in-progress subtitle upon receipt of a
-random access packet.
-
-This lowers the latency of subtitles for which the end packet or the following
-subtitle has not yet been received. As a drawback, this will most likely lead
-to duplication of subtitle events in order to cover the full duration, so
-when dealing with use cases where latency of when the subtitle event is passed
-on to output is not relevant this option should not be utilized.
-
-Requires @option{-fix_sub_duration} to be set for the relevant input subtitle
-stream for this to have any effect, as well as for the input subtitle stream
-having to be directly mapped to the same output in which the heartbeat stream
-resides.
 
 @end table
 
@@ -1456,18 +1180,18 @@ 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{[linklabel]} (@emph{output})
+@item -map [-]@var{input_file_id}[:@var{stream_specifier}][?][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
 
-Create one or more streams in the output file. This option has two forms for
+Designate one or more input streams as a source for the output file. Each input
-specifying the data source(s): the first selects one or more streams from some
+stream is identified by the input file index @var{input_file_id} and
-input file (specified with @code{-i}), the second takes an output from some
+the input stream index @var{input_stream_id} within the input
-complex filtergraph (specified with @code{-filter_complex} or
+file. Both indices start at 0. If specified,
-@code{-filter_complex_script}).
+@var{sync_file_id}:@var{stream_specifier} sets which input stream
+is used as a presentation sync reference.
 
-In the first form, an output stream is created for every stream from the input
+The first @code{-map} option on the command line specifies the
-file with the index @var{input_file_id}. If @var{stream_specifier} is given,
+source for output stream 0, the second @code{-map} option specifies
-only those streams that match the specifier are used (see the
+the source for output stream 1, etc.
-@ref{Stream specifiers} section for the @var{stream_specifier} syntax).
 
 A @code{-} character before the stream identifier creates a "negative" mapping.
 It disables matching streams from already created mappings.
@@ -1481,56 +1205,39 @@ 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.
 
-This option may be specified multiple times, each adding more streams to the
+For example, to map ALL streams from the first input file to output
-output file. Any given input stream may also be mapped any number of times as a
-source for different output streams, e.g. in order to use different encoding
-options and/or filters. The streams are created in the output in the same order
-in which the @code{-map} options are given on the commandline.
-
-Using this option disables the default mappings for this output file.
-
-Examples:
-
-@table @emph
-
-@item map everything
-To map ALL streams from the first input file to output
 @example
 ffmpeg -i INPUT -map 0 output
 @end example
 
-@item select specific stream
+For example, if you have two audio streams in the first input file,
-If you have two audio streams in the first input file, these streams are
+these streams are identified by "0:0" and "0:1". You can use
-identified by @var{0:0} and @var{0:1}. You can use @code{-map} to select which
+@code{-map} to select which streams to place in an output file. For
-streams to place in an output file. For example:
+example:
 @example
 ffmpeg -i INPUT -map 0:1 out.wav
 @end example
-will map the second input stream in @file{INPUT} to the (single) output stream
+will map the input stream in @file{INPUT} identified by "0:1" to
-in @file{out.wav}.
+the (single) output stream in @file{out.wav}.
 
-@item create multiple streams
+For example, to select the stream with index 2 from input file
-To select the stream with index 2 from input file @file{a.mov} (specified by the
+@file{a.mov} (specified by the identifier "0:2"), and stream with
-identifier @var{0:2}), and stream with index 6 from input @file{b.mov}
+index 6 from input @file{b.mov} (specified by the identifier "1:6"),
-(specified by the identifier @var{1:6}), and copy them to the output file
+and copy them to the output file @file{out.mov}:
-@file{out.mov}:
 @example
 ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov
 @end example
 
-@item create multiple streams 2
 To select all video and the third audio stream from an input file:
 @example
 ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT
 @end example
 
-@item negative map
 To map all the streams except the second audio, use negative mappings
 @example
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
 @end example
 
-@item optional map
 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:
@@ -1538,13 +1245,12 @@ the first input:
 ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
 @end example
 
-@item map by language
 To pick the English audio stream:
 @example
 ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
 @end example
 
-@end table
+Note that using this option disables the default mappings for this output file.
 
 @item -ignore_unknown
 Ignore input streams with unknown type instead of failing if copying
@@ -1555,10 +1261,6 @@ 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}]
-This option is deprecated and will be removed. It can be replaced by the
-@var{pan} filter. In some cases it may be easier to use some combination of the
-@var{channelsplit}, @var{channelmap}, or @var{amerge} filters.
-
 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.
@@ -1681,49 +1383,38 @@ it will usually display as 0 if not supported.
 Show benchmarking information during the encode.
 Shows real, system and user time used in various steps (audio/video encode/decode).
 @item -timelimit @var{duration} (@emph{global})
-Exit after ffmpeg has been running for @var{duration} seconds in CPU user time.
+Exit after ffmpeg has been running for @var{duration} seconds.
 @item -dump (@emph{global})
 Dump each input packet to stderr.
 @item -hex (@emph{global})
 When dumping packets, also dump the payload.
-@item -readrate @var{speed} (@emph{input})
-Limit input read speed.
-
-Its value is a floating-point positive number which represents the maximum duration of
-media, in seconds, that should be ingested in one second of wallclock time.
-Default value is zero and represents no imposed limitation on speed of ingestion.
-Value @code{1} represents real-time speed and is equivalent to @code{-re}.
-
-Mainly used to simulate a capture device or live input stream (e.g. when reading from a file).
-Should not be used with a low value when input is an actual capture device or live stream as
-it may cause packet loss.
-
-It is useful for when flow speed of output packets is important, such as live streaming.
 @item -re (@emph{input})
-Read input at native frame rate. This is equivalent to setting @code{-readrate 1}.
+Read input at native frame rate. Mainly used to simulate a grab device,
-@item -vsync @var{parameter} (@emph{global})
+or live input stream (e.g. when reading from a file). Should not be used
-@itemx -fps_mode[:@var{stream_specifier}] @var{parameter} (@emph{output,per-stream})
+with actual grab devices or live input streams (where it can cause packet
-Set video sync method / framerate mode. vsync is applied to all output video streams
+loss).
-but can be overridden for a stream by setting fps_mode. vsync is deprecated and will be
+By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
-removed in the future.
+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).
-For compatibility reasons some of the values for vsync can be specified as numbers (shown
+@item -vsync @var{parameter}
-in parentheses in the following table).
+Video sync method.
+For compatibility reasons old values can be specified as numbers.
+Newly added values will have to be specified as strings always.
 
 @table @option
-@item passthrough (0)
+@item 0, passthrough
 Each frame is passed with its timestamp from the demuxer to the muxer.
-@item cfr (1)
+@item 1, cfr
 Frames will be duplicated and dropped to achieve exactly the requested
 constant frame rate.
-@item vfr (2)
+@item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
 @item drop
 As passthrough but destroys all timestamps, making the muxer generate
 fresh timestamps based on frame-rate.
-@item auto (-1)
+@item -1, auto
-Chooses between cfr and vfr depending on muxer capabilities. This is the
+Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
 @end table
 
@@ -1742,16 +1433,17 @@ The default is -1.1. One possible usecase is to avoid framedrops in case
 of noisy timestamps or to increase frame drop precision in case of exact
 timestamps.
 
-@item -adrift_threshold @var{time}
+@item -async @var{samples_per_second}
-Set the minimum difference between timestamps and audio data (in seconds) to trigger
+Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
-adding/dropping samples to make it match the timestamps. This option effectively is
+the parameter is the maximum samples per second by which the audio is changed.
-a threshold to select between hard (add/drop) and soft (squeeze/stretch) compensation.
+-async 1 is a special case where only the start of the audio stream is corrected
-@code{-async} must be set to a positive value.
+without any later correction.
 
-@item -apad @var{parameters} (@emph{output,per-stream})
+Note that the timestamps may be further modified by the muxer, after this.
-Pad the output audio stream(s). This is the same as applying @code{-af apad}.
+For example, in the case that the format option @option{avoid_negative_ts}
-Argument is a string of filter parameters composed the same as with the @code{apad} filter.
+is enabled.
-@code{-shortest} must be set for this output for the option to take effect.
+
+This option has been deprecated. Use the @code{aresample} audio filter instead.
 
 @item -copyts
 Do not process input timestamps, but keep their values without trying
@@ -1820,29 +1512,9 @@ 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 output stream ends.
+Finish encoding when the shortest input stream ends.
-
-Note that this option may require buffering frames, which introduces extra
-latency. The maximum amount of this latency may be controlled with the
-@code{-shortest_buf_duration} option.
-
-@item -shortest_buf_duration @var{duration} (@emph{output})
-The @code{-shortest} option may require buffering potentially large amounts
-of data when at least one of the streams is "sparse" (i.e. has large gaps
-between frames – this is typically the case for subtitles).
-
-This option controls the maximum duration of buffered frames in seconds.
-Larger values may allow the @code{-shortest} option to produce more accurate
-results, but increase memory use and latency.
-
-The default value is 10 seconds.
-
 @item -dts_delta_threshold
 Timestamp discontinuity delta threshold.
-@item -dts_error_threshold @var{seconds}
-Timestamp error delta threshold. This threshold use to discard crazy/damaged
-timestamps and the default is 30 hours which is arbitrarily picked and quite
-conservative.
 @item -muxdelay @var{seconds} (@emph{output})
 Set the maximum demux-decode delay.
 @item -muxpreload @var{seconds} (@emph{output})
@@ -1923,22 +1595,6 @@ graph will be added to the output file automatically, so we can simply write
 ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
 @end example
 
-As a special exception, you can use a bitmap subtitle stream as input: it
-will be converted into a video with the same size as the largest video in
-the file, or 720x576 if no video is present. Note that this is an
-experimental and temporary solution. It will be removed once libavfilter has
-proper support for subtitles.
-
-For example, to hardcode subtitles on top of a DVB-T recording stored in
-MPEG-TS format, delaying the subtitles by 1 second:
-@example
-ffmpeg -i input.ts -filter_complex \
-  '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
-  -sn -map '#0x2dc' output.mkv
-@end example
-(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
-audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
-
 To generate 5 seconds of pure red video using lavfi @code{color} source:
 @example
 ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
@@ -1971,15 +1627,11 @@ to the @option{-ss} option is considered an actual timestamp, and is not
 offset by the start time of the file. This matters only for files which do
 not start from timestamp 0, such as transport streams.
 
-@item -thread_queue_size @var{size} (@emph{input/output})
+@item -thread_queue_size @var{size} (@emph{input})
-For input, this option sets the maximum number of queued packets when reading
+This option sets the maximum number of queued packets when reading from the
-from the file or device. With low latency / high rate live streams, packets may
+file or device. With low latency / high rate live streams, packets may be
-be discarded if they are not read in a timely manner; setting this value can
+discarded if they are not read in a timely manner; raising this value can
-force ffmpeg to use a separate input thread and read packets as soon as they
+avoid it.
-arrive. By default ffmpeg only does this if multiple inputs are specified.
-
-For output, this option specified the maximum number of packets that may be
-queued to each muxing thread.
 
 @item -sdp_file @var{file} (@emph{global})
 Print sdp information for an output stream to @var{file}.
@@ -2018,15 +1670,8 @@ Stop and abort on various conditions. The following flags are available:
 @table @option
 @item empty_output
 No packets were passed to the muxer, the output is empty.
-@item empty_output_stream
-No packets were passed to the muxer in some of the output streams.
 @end table
 
-@item -max_error_rate (@emph{global})
-Set fraction of decoding frame failures across all inputs which when crossed
-ffmpeg will return exit code 69. Crossing this threshold does not terminate
-processing. Range is a floating-point number between 0 to 1. Default is 2/3.
-
 @item -xerror (@emph{global})
 Stop and exit on error
 
@@ -2039,139 +1684,23 @@ 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.
 
-@item -muxing_queue_data_threshold @var{bytes} (@emph{output,per-stream})
-This is a minimum threshold until which the muxing queue size is not taken into
-account. Defaults to 50 megabytes per stream, and is based on the overall size
-of packets passed to the muxer.
-
-@item -auto_conversion_filters (@emph{global})
-Enable automatically inserting format conversion filters in all filter
-graphs, including those defined by @option{-vf}, @option{-af},
-@option{-filter_complex} and @option{-lavfi}. If filter format negotiation
-requires a conversion, the initialization of the filters will fail.
-Conversions can still be performed by inserting the relevant conversion
-filter (scale, aresample) in the graph.
-On by default, to explicitly disable it you need to specify
-@code{-noauto_conversion_filters}.
-
-@item -bits_per_raw_sample[:@var{stream_specifier}] @var{value} (@emph{output,per-stream})
-Declare the number of bits per raw sample in the given output stream to be
-@var{value}. Note that this option sets the information provided to the
-encoder/muxer, it does not change the stream to conform to this value. Setting
-values that do not match the stream properties may result in encoding failures
-or invalid output files.
-
-@item -stats_enc_pre[:@var{stream_specifier}] @var{path} (@emph{output,per-stream})
-@item -stats_enc_post[:@var{stream_specifier}] @var{path} (@emph{output,per-stream})
-@item -stats_mux_pre[:@var{stream_specifier}] @var{path} (@emph{output,per-stream})
-Write per-frame encoding information about the matching streams into the file
-given by @var{path}.
-
-@option{-stats_enc_pre} writes information about raw video or audio frames right
-before they are sent for encoding, while @option{-stats_enc_post} writes
-information about encoded packets as they are received from the encoder.
-@option{-stats_mux_pre} writes information about packets just as they are about to
-be sent to the muxer. Every frame or packet produces one line in the specified
-file. The format of this line is controlled by @option{-stats_enc_pre_fmt} /
-@option{-stats_enc_post_fmt} / @option{-stats_mux_pre_fmt}.
-
-When stats for multiple streams are written into a single file, the lines
-corresponding to different streams will be interleaved. The precise order of
-this interleaving is not specified and not guaranteed to remain stable between
-different invocations of the program, even with the same options.
-
-@item -stats_enc_pre_fmt[:@var{stream_specifier}] @var{format_spec} (@emph{output,per-stream})
-@item -stats_enc_post_fmt[:@var{stream_specifier}] @var{format_spec} (@emph{output,per-stream})
-@item -stats_mux_pre_fmt[:@var{stream_specifier}] @var{format_spec} (@emph{output,per-stream})
-Specify the format for the lines written with @option{-stats_enc_pre} /
-@option{-stats_enc_post} / @option{-stats_mux_pre}.
-
-@var{format_spec} is a string that may contain directives of the form
-@var{@{fmt@}}. @var{format_spec} is backslash-escaped --- use \@{, \@}, and \\
-to write a literal @{, @}, or \, respectively, into the output.
-
-The directives given with @var{fmt} may be one of the following:
-@table @option
-@item fidx
-Index of the output file.
-
-@item sidx
-Index of the output stream in the file.
-
-@item n
-Frame number. Pre-encoding: number of frames sent to the encoder so far.
-Post-encoding: number of packets received from the encoder so far.
-Muxing: number of packets submitted to the muxer for this stream so far.
-
-@item ni
-Input frame number. Index of the input frame (i.e. output by a decoder) that
-corresponds to this output frame or packet. -1 if unavailable.
-
-@item tb
-Encoder timebase, as a rational number @var{num/den}. Note that this may be
-different from the timebase used by the muxer.
-
-@item tbi
-Timebase for @var{ptsi}, as a rational number @var{num/den}. Available when
-@var{ptsi} is available, @var{0/1} otherwise.
-
-@item pts
-Presentation timestamp of the frame or packet, as an integer. Should be
-multiplied by the timebase to compute presentation time.
-
-@item ptsi
-Presentation timestamp of the input frame (see @var{ni}), as an integer. Should
-be multiplied by @var{tbi} to compute presentation time. Printed as
-(2^63 - 1 = 9223372036854775807) when not available.
-
-@item t
-Presentation time of the frame or packet, as a decimal number. Equal to
-@var{pts} multiplied by @var{tb}.
-
-@item ti
-Presentation time of the input frame (see @var{ni}), as a decimal number. Equal
-to @var{ptsi} multiplied by @var{tbi}. Printed as inf when not available.
-
-@item dts
-Decoding timestamp of the packet, as an integer. Should be multiplied by the
-timebase to compute presentation time. Post-encoding only.
-
-@item dt
-Decoding time of the frame or packet, as a decimal number. Equal to
-@var{dts} multiplied by @var{tb}.
-
-@item sn
-Number of audio samples sent to the encoder so far. Audio and pre-encoding only.
-
-@item samp
-Number of audio samples in the frame. Audio and pre-encoding only.
-
-@item size
-Size of the encoded packet in bytes. Post-encoding only.
-
-@item br
-Current bitrate in bits per second. Post-encoding only.
-
-@item abr
-Average bitrate for the whole stream so far, in bits per second, -1 if it cannot
-be determined at this point. Post-encoding only.
 @end table
 
-The default format strings are:
+As a special exception, you can use a bitmap subtitle stream as input: it
-@table @option
+will be converted into a video with the same size as the largest video in
-@item pre-encoding
+the file, or 720x576 if no video is present. Note that this is an
-@{fidx@} @{sidx@} @{n@} @{t@}
+experimental and temporary solution. It will be removed once libavfilter has
-@item post-encoding
+proper support for subtitles.
-@{fidx@} @{sidx@} @{n@} @{t@}
-@end table
-In the future, new items may be added to the end of the default formatting
-strings. Users who depend on the format staying exactly the same, should
-prescribe it manually.
 
-Note that stats for different streams written into the same file may have
+For example, to hardcode subtitles on top of a DVB-T recording stored in
-different formats.
+MPEG-TS format, delaying the subtitles by 1 second:
-
+@example
-@end table
+ffmpeg -i input.ts -filter_complex \
+  '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
+  -sn -map '#0x2dc' output.mkv
+@end example
+(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
+audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
 
 @section Preset files
 A preset file contains a sequence of @var{option}=@var{value} pairs,
@@ -2449,7 +1978,6 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
 @ifset config-avfilter
 @include filters.texi
 @end ifset
-@include general_contents.texi
 @end ifset
 
 @chapter See Also

doc/ffplay.texi

@@ -34,6 +34,10 @@ various FFmpeg APIs.
 Force displayed width.
 @item -y @var{height}
 Force displayed height.
+@item -s @var{size}
+Set frame size (WxH or abbreviation), needed for videos which do
+not contain a header with the frame size like raw YUV.  This option
+has been deprecated in favor of private options, try -video_size.
 @item -fs
 Start in fullscreen mode.
 @item -an
@@ -122,12 +126,15 @@ Read @var{input_url}.
 
 @section Advanced options
 @table @option
+@item -pix_fmt @var{format}
+Set pixel format.
+This option has been deprecated in favor of private options, try -pixel_format.
+
 @item -stats
 Print several playback statistics, in particular show the stream
 duration, the codec parameters, the current position in the stream and
-the audio/video synchronisation drift. It is shown by default, unless the
+the audio/video synchronisation drift. It is on by default, to
-log level is lower than @code{info}. Its display can be forced by manually
+explicitly disable it you need to specify @code{-nostats}.
-specifying this option. To disable it, you need to specify @code{-nostats}.
 
 @item -fast
 Non-spec-compliant optimizations.
@@ -214,6 +221,8 @@ Pause.
 Toggle mute.
 
 @item 9, 0
+Decrease and increase volume respectively.
+
 @item /, *
 Decrease and increase volume respectively.
 
@@ -285,7 +294,6 @@ Toggle full screen.
 @ifset config-avfilter
 @include filters.texi
 @end ifset
-@include general_contents.texi
 @end ifset
 
 @chapter See Also

doc/ffprobe.texi

@@ -12,7 +12,7 @@
 
 @chapter Synopsis
 
-ffprobe [@var{options}] @file{input_url}
+ffprobe [@var{options}] [@file{input_url}]
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -28,9 +28,6 @@ If a url is specified in input, ffprobe will try to open and
 probe the url content. If the url cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.
 
-If no output is specified as output with @option{o} ffprobe will write
-to stdout.
-
 ffprobe may be employed both as a standalone application or in
 combination with a textual filter, which may perform more
 sophisticated processing, e.g. statistical processing or plotting.
@@ -338,12 +335,6 @@ Show information about all pixel formats supported by FFmpeg.
 Pixel format information for each format is printed within a section
 with name "PIXEL_FORMAT".
 
-@item -show_optional_fields @var{value}
-Some writers viz. JSON and XML, omit the printing of fields with invalid or non-applicable values,
-while other writers always print them. This option enables one to control this behaviour.
-Valid values are @code{always}/@code{1}, @code{never}/@code{0} and @code{auto}/@code{-1}.
-Default is @var{auto}.
-
 @item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
@@ -351,10 +342,6 @@ on the specific build.
 @item -i @var{input_url}
 Read @var{input_url}.
 
-@item -o @var{output_url}
-Write output to @var{output_url}. If not specified, the output is sent
-to stdout.
-
 @end table
 @c man end
 
@@ -655,7 +642,6 @@ DV, GXF and AVI timecodes are available in format metadata
 @ifset config-avfilter
 @include filters.texi
 @end ifset
-@include general_contents.texi
 @end ifset
 
 @chapter See Also

doc/ffprobe.xsd

@@ -29,18 +29,22 @@
     </xsd:complexType>
 
     <xsd:complexType name="framesType">
-        <xsd:choice minOccurs="0" maxOccurs="unbounded">
+        <xsd:sequence>
-            <xsd:element name="frame"    type="ffprobe:frameType"/>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
-            <xsd:element name="subtitle" type="ffprobe:subtitleType"/>
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:choice>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
+        </xsd:sequence>
     </xsd:complexType>
 
     <xsd:complexType name="packetsAndFramesType">
-        <xsd:choice minOccurs="0" maxOccurs="unbounded">
+        <xsd:sequence>
-            <xsd:element name="packet"   type="ffprobe:packetType"/>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
-            <xsd:element name="frame"    type="ffprobe:frameType"/>
+                <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-            <xsd:element name="subtitle" type="ffprobe:subtitleType"/>
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:choice>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
+        </xsd:sequence>
     </xsd:complexType>
 
     <xsd:complexType name="packetType">
@@ -57,6 +61,8 @@
       <xsd:attribute name="dts_time"      type="xsd:float" />
       <xsd:attribute name="duration"      type="xsd:long"  />
       <xsd:attribute name="duration_time" type="xsd:float" />
+      <xsd:attribute name="convergence_duration"      type="xsd:long"  />
+      <xsd:attribute name="convergence_duration_time" type="xsd:float" />
       <xsd:attribute name="size"          type="xsd:long" use="required" />
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
@@ -86,14 +92,14 @@
       <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
       <xsd:attribute name="pts"           type="xsd:long" />
       <xsd:attribute name="pts_time"      type="xsd:float"/>
+      <xsd:attribute name="pkt_pts"       type="xsd:long" />
+      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
       <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
       <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
       <xsd:attribute name="pkt_duration"  type="xsd:long" />
       <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
-      <xsd:attribute name="duration"      type="xsd:long" />
-      <xsd:attribute name="duration_time" type="xsd:float"/>
       <xsd:attribute name="pkt_pos"       type="xsd:long" />
       <xsd:attribute name="pkt_size"      type="xsd:int" />
 
@@ -195,11 +201,6 @@
       <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:attribute name="captions"         type="xsd:int" use="required" />
-      <xsd:attribute name="descriptions"     type="xsd:int" use="required" />
-      <xsd:attribute name="metadata"         type="xsd:int" use="required" />
-      <xsd:attribute name="dependent"        type="xsd:int" use="required" />
-      <xsd:attribute name="still_image"      type="xsd:int" use="required" />
     </xsd:complexType>
 
     <xsd:complexType name="streamType">
@@ -214,10 +215,10 @@
       <xsd:attribute name="codec_long_name"  type="xsd:string" />
       <xsd:attribute name="profile"          type="xsd:string" />
       <xsd:attribute name="codec_type"       type="xsd:string" />
+      <xsd:attribute name="codec_time_base"  type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
       <xsd:attribute name="extradata"        type="xsd:string" />
-      <xsd:attribute name="extradata_size"   type="xsd:int"    />
       <xsd:attribute name="extradata_hash"   type="xsd:string" />
 
       <!-- video attributes -->
@@ -225,8 +226,6 @@
       <xsd:attribute name="height"               type="xsd:int"/>
       <xsd:attribute name="coded_width"          type="xsd:int"/>
       <xsd:attribute name="coded_height"         type="xsd:int"/>
-      <xsd:attribute name="closed_captions"      type="xsd:boolean"/>
-      <xsd:attribute name="film_grain"           type="xsd:boolean"/>
       <xsd:attribute name="has_b_frames"         type="xsd:int"/>
       <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
       <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -238,6 +237,7 @@
       <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"/>
 
       <!-- audio attributes -->
@@ -246,7 +246,6 @@
       <xsd:attribute name="channels"         type="xsd:int"/>
       <xsd:attribute name="channel_layout"   type="xsd:string"/>
       <xsd:attribute name="bits_per_sample"  type="xsd:int"/>
-      <xsd:attribute name="initial_padding"  type="xsd:int"/>
 
       <xsd:attribute name="id"               type="xsd:string"/>
       <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
@@ -273,6 +272,10 @@
       <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
       <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
       <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="start_time"           type="xsd:float"/>
+      <xsd:attribute name="start_pts"            type="xsd:long"/>
+      <xsd:attribute name="end_time"             type="xsd:float"/>
+      <xsd:attribute name="end_pts"              type="xsd:long"/>
       <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
       <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
     </xsd:complexType>
@@ -354,6 +357,7 @@
       <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
       <xsd:attribute name="planar"     type="xsd:int" use="required"/>
       <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
+      <xsd:attribute name="pseudopal"  type="xsd:int" use="required"/>
       <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
     </xsd:complexType>
 

doc/fftools-common-opts.texi

@@ -107,24 +107,17 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
 @option{-formats} option to get a list of all muxers and demuxers.
 
 @item filter=@var{filter_name}
-Print detailed information about the filter named @var{filter_name}. Use the
+Print detailed information about the filter name @var{filter_name}. Use the
 @option{-filters} option to get a list of all filters.
 
 @item bsf=@var{bitstream_filter_name}
-Print detailed information about the bitstream filter named @var{bitstream_filter_name}.
+Print detailed information about the bitstream filter name @var{bitstream_filter_name}.
 Use the @option{-bsfs} option to get a list of all bitstream filters.
-
-@item protocol=@var{protocol_name}
-Print detailed information about the protocol named @var{protocol_name}.
-Use the @option{-protocols} option to get a list of all protocols.
 @end table
 
 @item -version
 Show version.
 
-@item -buildconf
-Show the build configuration, one option per line.
-
 @item -formats
 Show available formats (including devices).
 
@@ -167,9 +160,6 @@ Show available sample formats.
 @item -layouts
 Show channel names and standard channel layouts.
 
-@item -dispositions
-Show stream dispositions.
-
 @item -colors
 Show recognized color names.
 
@@ -246,11 +236,13 @@ ffmpeg [...] -loglevel +repeat
 By default the program logs to stderr. If coloring is supported by the
 terminal, colors are used to mark errors and warnings. Log coloring
 can be disabled setting the environment variable
-@env{AV_LOG_FORCE_NOCOLOR}, or can be forced setting
+@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
 the environment variable @env{AV_LOG_FORCE_COLOR}.
+The use of the environment variable @env{NO_COLOR} is deprecated and
+will be dropped in a future FFmpeg version.
 
 @item -report
-Dump full command line and log output to a file named
+Dump full command line and console output to a file named
 @code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
 directory.
 This file can be useful for bug reports.
@@ -355,19 +347,6 @@ Possible flags for this option are:
 @item k8
 @end table
 @end table
-
-@item -cpucount @var{count} (@emph{global})
-Override detection of CPU count. This option is intended
-for testing. Do not use it unless you know what you're doing.
-@example
-ffmpeg -cpucount 2
-@end example
-
-@item -max_alloc @var{bytes}
-Set the maximum size limit for allocating a block on the heap by ffmpeg's
-family of malloc functions. Exercise @strong{extreme caution} when using
-this option. Don't use if you do not understand the full consequence of doing so.
-Default is INT_MAX.
 @end table
 
 @section AVOptions

doc/formats.texi

@@ -27,10 +27,6 @@ stream information. A higher value will enable detecting more
 information in case it is dispersed into the stream, but will increase
 latency. Must be an integer not lesser than 32. It is 5000000 by default.
 
-@item max_probe_packets @var{integer} (@emph{input})
-Set the maximum number of buffered packets when probing a codec.
-Default is 2500 packets.
-
 @item packetsize @var{integer} (@emph{output})
 Set packet size.
 
@@ -49,6 +45,7 @@ Generate missing PTS if DTS is present.
 Ignore DTS if PTS is set. Inert when nofillin is set.
 @item ignidx
 Ignore index.
+@item keepside (@emph{deprecated},@emph{inert})
 @item nobuffer
 Reduce the latency introduced by buffering during initial input streams analysis.
 @item nofillin
@@ -69,6 +66,7 @@ This ensures that file and data checksums are reproducible and match between
 platforms. Its primary use is for regression testing.
 @item flush_packets
 Write out packets immediately.
+@item latm (@emph{deprecated},@emph{inert})
 @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
@@ -141,7 +139,7 @@ Consider things that a sane encoder should not do as an error.
 
 @item max_interleave_delta @var{integer} (@emph{output})
 Set maximum buffering duration for interleaving. The duration is
-expressed in microseconds, and defaults to 10000000 (10 seconds).
+expressed in microseconds, and defaults to 1000000 (1 second).
 
 To ensure all the streams are interleaved correctly, libavformat will
 wait until it has at least one packet for each stream before actually

doc/git-howto.texi

@@ -53,7 +53,7 @@ Most distribution and operating system provide a package for it.
 @section Cloning the source tree
 
 @example
-git clone https://git.ffmpeg.org/ffmpeg.git <target>
+git clone git://source.ffmpeg.org/ffmpeg <target>
 @end example
 
 This will put the FFmpeg sources into the directory @var{<target>}.
@@ -187,18 +187,11 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
 @end example
 
-Make sure you have told Git your name, email address and GPG key
+Make sure you have told Git your name and email address
 
 @example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
-git config --global user.signingkey ABCDEF0123245
-@end example
-
-Enable signing all commits or use -S
-
-@example
-git config --global commit.gpgsign true
 @end example
 
 Use @option{--global} to set the global configuration for all your Git checkouts.
@@ -224,46 +217,16 @@ git config --global core.editor
 or set by one of the following environment variables:
 @var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
 
-@section Writing a commit message
+Log messages should be concise but descriptive. Explain why you made a change,
+what you did will be obvious from the changes themselves most of the time.
+Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
+levels look at and educate themselves while reading through your code. Don't
+include filenames in log messages, Git provides that information.
 
-Log messages should be concise but descriptive.
+Possibly make the commit message have a terse, descriptive first line, an
-
+empty line and then a full description. The first line will be used to name
-The first line must contain the context, a colon and a very short
-summary of what the commit does. Details can be added, if necessary,
-separated by an empty line. These details should not exceed 60-72 characters
-per line, except when containing code.
-
-Example of a good commit message:
-
-@example
-avcodec/cbs: add a helper to read extradata within packet side data
-
-Using ff_cbs_read() on the raw buffer will not parse it as extradata,
-resulting in parsing errors for example when handling ISOBMFF avcC.
-This helper works around that.
-@end example
-
-@example
-ptr might be NULL
-@end example
-
-If the summary on the first line is not enough, in the body of the message,
-explain why you made a change, what you did will be obvious from the changes
-themselves most of the time. Saying just "bug fix" or "10l" is bad. Remember
-that people of varying skill levels look at and educate themselves while
-reading through your code. Don't include filenames in log messages except in
-the context, Git provides that information.
-
-If the commit fixes a registered issue, state it in a separate line of the
-body: @code{Fix Trac ticket #42.}
-
-The first line will be used to name
 the patch by @command{git format-patch}.
 
-Common mistakes for the first line, as seen in @command{git log --oneline}
-include: missing context at the beginning; description of what the code did
-before the patch; line too long or wrapped to the second line.
-
 @section Preparing a patchset
 
 @example
@@ -430,19 +393,6 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
 
-@chapter gpg key generation
-
-If you have no gpg key yet, we recommend that you create a ed25519 based key as it
-is small, fast and secure. Especially it results in small signatures in git.
-
-@example
-gpg --default-new-key-algo "ed25519/cert,sign+cv25519/encr" --quick-generate-key "human@@server.com"
-@end example
-
-When generating a key, make sure the email specified matches the email used in git as some sites like
-github consider mismatches a reason to declare such commits unverified. After generating a key you
-can add it to the MAINTAINER file and upload it to a keyserver.
-
 @chapter Pre-push checklist
 
 Once you have a set of commits that you feel are ready for pushing,

doc/indevs.texi

@@ -277,8 +277,8 @@ audio track.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}. This option is deprecated, please use the
+Defaults to @option{false}. Alternatively you can use the @code{-sources}
-@code{-sources} option of ffmpeg to list the available input devices.
+option of ffmpeg to list the available input devices.
 
 @item list_formats
 If set to @option{true}, print a list of supported formats and exit.
@@ -292,35 +292,25 @@ 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 auto
-
-This is the default which means 8-bit YUV 422 or 8-bit ARGB if format
-autodetection is used, 8-bit YUV 422 otherwise.
-
 @item uyvy422
 
-8-bit YUV 422.
-
 @item yuv422p10
 
-10-bit YUV 422.
-
 @item argb
 
-8-bit RGB.
-
 @item bgra
 
-8-bit RGB.
-
 @item rgb10
 
-10-bit RGB.
-
 @end table
 
 @item teletext_lines
@@ -344,33 +334,14 @@ Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp
 Defaults to @samp{2}.
 
 @item duplex_mode
-Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
+Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
 Defaults to @samp{unset}.
 
-Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
-For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
-sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
-is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
-
-Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
-
-Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
-@samp{half}, @samp{full}
-
 @item timecode_format
 Timecode type to include in the frame and video stream metadata. Must be
 @samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
-@samp{rp188hfr}, @samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}.
+@samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}. Defaults to
-Defaults to @samp{none} (not included).
+@samp{none} (not included).
-
-In order to properly support 50/60 fps timecodes, the ordering of the queried
-timecode types for @samp{rp188any} is HFR, VITC1, VITC2 and LTC for >30 fps
-content. Note that this is slightly different to the ordering used by the
-DeckLink API, which is HFR, VITC1, LTC, VITC2.
 
 @item video_input
 Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
@@ -424,20 +395,6 @@ Either sync could go wrong by 1 frame or in a rarer case
 @option{timestamp_align} seconds.
 Defaults to @samp{0}.
 
-@item wait_for_tc (@emph{bool})
-Drop frames till a frame with timecode is received. Sometimes serial timecode
-isn't received with the first input frame. If that happens, the stored stream
-timecode will be inaccurate. If this option is set to @option{true}, input frames
-are dropped till a frame with timecode is received.
-Option @var{timecode_format} must be specified.
-Defaults to @option{false}.
-
-@item enable_klv(@emph{bool})
-If set to @option{true}, extracts KLV data from VANC and outputs KLV packets.
-KLV VANC packets are joined based on MID and PSC fields and aggregated into
-one KLV packet.
-Defaults to @option{false}.
-
 @end table
 
 @subsection Examples
@@ -447,7 +404,7 @@ Defaults to @option{false}.
 @item
 List input devices:
 @example
-ffmpeg -sources decklink
+ffmpeg -f decklink -list_devices 1 -i dummy
 @end example
 
 @item
@@ -465,7 +422,7 @@ ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy outp
 @item
 Capture video clip at 1080i50 10 bit:
 @example
-ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
+ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
 @end example
 
 @item
@@ -625,12 +582,6 @@ Save the currently used video capture filter device and its
 parameters (if the filter supports it) to a file.
 If a file with the same name exists it will be overwritten.
 
-@item use_video_device_timestamps
-If set to @option{false}, the timestamp for video frames will be
-derived from the wallclock instead of the timestamp provided by
-the capture device. This allows working around devices that
-provide unreliable timestamps.
-
 @end table
 
 @subsection Examples
@@ -929,15 +880,11 @@ If you don't understand what all of that means, you probably don't want this.  L
 DRM device to capture on.  Defaults to @option{/dev/dri/card0}.
 
 @item format
-Pixel format of the framebuffer.  This can be autodetected if you are running Linux 5.7
+Pixel format of the framebuffer.  Defaults to @option{bgr0}.
-or later, but needs to be provided for earlier versions.  Defaults to @option{bgr0},
-which is the most common format used by the Linux console and Xorg X server.
 
 @item format_modifier
 Format modifier to signal on output frames.  This is necessary to import correctly into
-some APIs.  It can be autodetected if you are running Linux 5.7 or later, but will need
+some APIs, but can't be autodetected.  See the libdrm documentation for possible values.
-to be provided explicitly when needed in earlier versions.  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
@@ -1289,11 +1236,11 @@ Specify the samplerate in Hz, by default 48kHz is used.
 Specify the channels in use, by default 2 (stereo) is set.
 
 @item frame_size
-This option does nothing and is deprecated.
+Specify the number of bytes per frame, by default it is set to 1024.
 
 @item fragment_size
-Specify the size in bytes of the minimal buffering fragment in PulseAudio, it
+Specify the minimal buffering fragment in PulseAudio, it will affect the
-will affect the audio latency. By default it is set to 50 ms amount of data.
+audio latency. By default it is unset.
 
 @item wallclock
 Set the initial PTS using the current time. Default is 1.
@@ -1528,14 +1475,6 @@ ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
 @subsection Options
 
 @table @option
-@item select_region
-Specify whether to select the grabbing area graphically using the pointer.
-A value of @code{1} prompts the user to select the grabbing area graphically
-by clicking and dragging. A single click with no dragging will select the
-whole screen. A region with zero width or height will also select the whole
-screen. This option overwrites the @var{video_size}, @var{grab_x}, and
-@var{grab_y} options. Default value is @code{0}.
-
 @item draw_mouse
 Specify whether to draw the mouse pointer. A value of @code{0} specifies
 not to draw the pointer. Default value is @code{1}.
@@ -1584,21 +1523,8 @@ With @var{follow_mouse}:
 ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
 @end example
 
-@item window_id
-Grab this window, instead of the whole screen. Default value is 0, which maps to
-the whole screen (root window).
-
-The id of a window can be found using the @command{xwininfo} program, possibly with options -tree and
--root.
-
-If the window is later enlarged, the new area is not recorded. Video ends when
-the window is closed, unmapped (i.e., iconified) or shrunk beyond the video
-size (which defaults to the initial window size).
-
-This option disables options @option{follow_mouse} and @option{select_region}.
-
 @item video_size
-Set the video frame size. Default is the full desktop or window.
+Set the video frame size. Default value is @code{vga}.
 
 @item grab_x
 @item grab_y

doc/mailing-list-faq.texi

@@ -116,7 +116,7 @@ or is abusive towards others).
 @section How long does it take for my message in the moderation queue to be approved?
 
 The queue is not checked on a regular basis. You can ask on the
-@t{#ffmpeg-devel} IRC channel on Libera Chat for someone to approve your message.
+@t{#ffmpeg-devel} IRC channel on Freenode for someone to approve your message.
 
 @anchor{How do I delete my message in the moderation queue?}
 @section How do I delete my message in the moderation queue?
@@ -155,7 +155,7 @@ Perform a site search using your favorite search engine. Example:
 
 @section Is there an alternative to the mailing list?
 
-You can ask for help in the official @t{#ffmpeg} IRC channel on Libera Chat.
+You can ask for help in the official @t{#ffmpeg} IRC channel on Freenode.
 
 Some users prefer the third-party @url{http://www.ffmpeg-archive.org/, Nabble}
 interface which presents the mailing lists in a typical forum layout.

doc/multithreading.txt

@@ -20,7 +20,8 @@ Slice threading -
 
 Frame threading -
 * Restrictions with slice threading also apply.
-* Custom get_buffer2() and get_format() callbacks must be thread-safe.
+* For best performance, the client should set thread_safe_callbacks if it
+  provides a thread-safe get_buffer() callback.
 * There is one frame of delay added for every thread beyond the first one.
   Clients must be able to handle this; the pkt_dts and pkt_pts fields in
   AVFrame will work as usual.
@@ -50,14 +51,16 @@ the decode process starts. Call ff_thread_finish_setup() afterwards. If
 some code can't be moved, have update_thread_context() run it in the next
 thread.
 
+If the codec allocates writable tables in its init(), add an init_thread_copy()
+which re-allocates them for other threads.
+
 Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.
 
 If there are inter-frame dependencies, so the codec calls
-ff_thread_report/await_progress(), set FF_CODEC_CAP_ALLOCATE_PROGRESS in
+ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
-AVCodec.caps_internal and use ff_thread_get_buffer() to allocate frames. The
 frames must then be freed with ff_thread_release_buffer().
-Otherwise decode directly into the user-supplied frames.
+Otherwise leave it at zero and decode directly into the user-supplied frames.
 
 Call ff_thread_report_progress() after some part of the current picture has decoded.
 A good place to put this is where draw_horiz_band() is called - add this if it isn't

doc/optimization.txt

@@ -267,11 +267,6 @@ CELL/SPU:
 http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
 http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf
 
-RISC-V-specific:
-----------------
-The RISC-V Instruction Set Manual, Volume 1, Unprivileged ISA:
-https://riscv.org/technical/specifications/
-
 GCC asm links:
 --------------
 official doc but quite ugly

doc/outdevs.texi

@@ -38,52 +38,6 @@ ffmpeg -i INPUT -f alsa hw:1,7
 @end example
 @end itemize
 
-@section AudioToolbox
-
-AudioToolbox output device.
-
-Allows native output to CoreAudio devices on OSX.
-
-The output filename can be empty (or @code{-}) to refer to the default system output device or a number that refers to the device index as shown using: @code{-list_devices true}.
-
-Alternatively, the audio input device can be chosen by index using the
-@option{
-    -audio_device_index <INDEX>
-}
-, overriding any device name or index given in the input filename.
-
-All available devices can be enumerated by using @option{-list_devices true}, listing
-all device names, UIDs and corresponding indices.
-
-@subsection Options
-
-AudioToolbox supports the following options:
-
-@table @option
-
-@item -audio_device_index <INDEX>
-Specify the audio device by its index. Overrides anything given in the output filename.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Print the list of supported devices and output a sine wave to the default device:
-@example
-$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -
-@end example
-
-@item
-Output a sine wave to the device with the index 2, overriding any output filename:
-@example
-$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -
-@end example
-
-@end itemize
-
 @section caca
 
 CACA output device.
@@ -186,8 +140,8 @@ device with @command{-list_formats 1}. Audio sample rate is always 48 kHz.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}. This option is deprecated, please use the
+Defaults to @option{false}. Alternatively you can use the @code{-sinks}
-@code{-sinks} option of ffmpeg to list the available output devices.
+option of ffmpeg to list the available output devices.
 
 @item list_formats
 If set to @option{true}, print a list of supported formats and exit.
@@ -198,43 +152,13 @@ Amount of time to preroll video in seconds.
 Defaults to @option{0.5}.
 
 @item duplex_mode
-Sets the decklink device duplex/profile mode. Must be @samp{unset}, @samp{half}, @samp{full},
+Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
 Defaults to @samp{unset}.
 
-Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property.
-For the DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2
-sub-devices that utilize the same connectors. For the DeckLink 8K Pro, a profile
-is shared between all 4 sub-devices. So DeckLink 8K Pro support four profiles.
-
-Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
-@samp{one_sub_device_full}, @samp{one_sub_device_half}, @samp{two_sub_device_full},
-@samp{four_sub_device_half}
-
-Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2:
-@samp{half}, @samp{full}
-
 @item timing_offset
 Sets the genlock timing pixel offset on the used output.
 Defaults to @samp{unset}.
 
-@item link
-Sets the SDI video link configuration on the used output. Must be
-@samp{unset}, @samp{single} link SDI, @samp{dual} link SDI or @samp{quad} link
-SDI.
-Defaults to @samp{unset}.
-
-@item sqd
-Enable Square Division Quad Split mode for Quad-link SDI output.
-Must be @samp{unset}, @samp{true} or @samp{false}.
-Defaults to @option{unset}.
-
-@item level_a
-Enable SMPTE Level A mode on the used output.
-Must be @samp{unset}, @samp{true} or @samp{false}.
-Defaults to @option{unset}.
-
 @end table
 
 @subsection Examples
@@ -244,7 +168,7 @@ Defaults to @option{unset}.
 @item
 List output devices:
 @example
-ffmpeg -sinks decklink
+ffmpeg -i test.avi -f decklink -list_devices 1 dummy
 @end example
 
 @item
@@ -405,8 +329,6 @@ ffmpeg  -i INPUT -f pulse "stream name"
 
 SDL (Simple DirectMedia Layer) output device.
 
-"sdl2" can be used as alias for "sdl".
-
 This output device allows one to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
 have only one instance of this output device in an application.

doc/protocols.texi

@@ -51,82 +51,6 @@ in microseconds.
 
 A description of the currently available protocols follows.
 
-@section amqp
-
-Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based
-publish-subscribe communication protocol.
-
-FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate
-AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.
-
-After starting the broker, an FFmpeg client may stream data to the broker using
-the command:
-
-@example
-ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@@]hostname[:port][/vhost]
-@end example
-
-Where hostname and port (default is 5672) is the address of the broker. The
-client may also set a user/password for authentication. The default for both
-fields is "guest". Name of virtual host on broker can be set with vhost. The
-default value is "/".
-
-Muliple subscribers may stream from the broker using the command:
-@example
-ffplay amqp://[[user]:[password]@@]hostname[:port][/vhost]
-@end example
-
-In RabbitMQ all data published to the broker flows through a specific exchange,
-and each subscribing client has an assigned queue/buffer. When a packet arrives
-at an exchange, it may be copied to a client's queue depending on the exchange
-and routing_key fields.
-
-The following options are supported:
-
-@table @option
-
-@item exchange
-Sets the exchange to use on the broker. RabbitMQ has several predefined
-exchanges: "amq.direct" is the default exchange, where the publisher and
-subscriber must have a matching routing_key; "amq.fanout" is the same as a
-broadcast operation (i.e. the data is forwarded to all queues on the fanout
-exchange independent of the routing_key); and "amq.topic" is similar to
-"amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
-documentation).
-
-@item routing_key
-Sets the routing key. The default value is "amqp". The routing key is used on
-the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
-to the queue of a subscriber.
-
-@item pkt_size
-Maximum size of each packet sent/received to the broker. Default is 131072.
-Minimum is 4096 and max is any large value (representable by an int). When
-receiving packets, this sets an internal buffer size in FFmpeg. It should be
-equal to or greater than the size of the published packets to the broker. Otherwise
-the received message may be truncated causing decoding errors.
-
-@item connection_timeout
-The timeout in seconds during the initial connection to the broker. The
-default value is rw_timeout, or 5 seconds if rw_timeout is not set.
-
-@item delivery_mode @var{mode}
-Sets the delivery mode of each message sent to broker.
-The following values are accepted:
-@table @samp
-@item persistent
-Delivery mode set to "persistent" (2). This is the default value.
-Messages may be written to the broker's disk depending on its setup.
-
-@item non-persistent
-Delivery mode set to "non-persistent" (1).
-Messages will stay in broker's memory unless the broker is under memory
-pressure.
-
-@end table
-
-@end table
-
 @section async
 
 Asynchronous data filling wrapper for input stream.
@@ -175,16 +99,6 @@ Caching wrapper for input stream.
 
 Cache the input stream to temporary file. It brings seeking capability to live streams.
 
-The accepted options are:
-@table @option
-
-@item read_ahead_limit
-Amount in bytes that may be read ahead when seeking isn't supported. Range is -1 to INT_MAX.
--1 for unlimited. Default is 65536.
-
-@end table
-
-URL Syntax is
 @example
 cache:@var{URL}
 @end example
@@ -215,38 +129,6 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 Note that you may need to escape the character "|" which is special for
 many shells.
 
-@section concatf
-
-Physical concatenation protocol using a line break delimited list of
-resources.
-
-Read and seek from many resources in sequence as if they were
-a unique resource.
-
-A URL accepted by this protocol has the syntax:
-@example
-concatf:@var{URL}
-@end example
-
-where @var{URL} is the url containing a line break delimited list of
-resources to be concatenated, each one possibly specifying a distinct
-protocol. Special characters must be escaped with backslash or single
-quotes. See @ref{quoting_and_escaping,,the "Quoting and escaping"
-section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} listed in separate lines within
-a file @file{split.txt} with @command{ffplay} use the command:
-@example
-ffplay concatf:split.txt
-@end example
-Where @file{split.txt} contains the lines:
-@example
-split1.mpeg
-split2.mpeg
-split3.mpeg
-@end example
-
 @section crypto
 
 AES-encrypted stream reading protocol.
@@ -275,33 +157,6 @@ For example, to convert a GIF file given inline with @command{ffmpeg}:
 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
 @end example
 
-@section fd
-
-File descriptor access protocol.
-
-The accepted syntax is:
-@example
-fd: -fd @var{file_descriptor}
-@end example
-
-If @option{fd} is not specified, by default the stdout file descriptor will be
-used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has
-seek support if it corresponding to a regular file. fd protocol doesn't support
-pass file descriptor via URL for security.
-
-This protocol accepts the following options:
-
-@table @option
-@item blocksize
-Set I/O operation maximum block size, in bytes. Default value is
-@code{INT_MAX}, which results in not limiting the requested block size.
-Setting this value reasonably low improves user termination request reaction
-time, which is valuable if data transmission is slow.
-
-@item fd
-Set file descriptor.
-@end table
-
 @section file
 
 File access protocol.
@@ -373,14 +228,6 @@ Set timeout in microseconds of socket I/O operations used by the underlying low
 operation. By default it is set to -1, which means that the timeout is
 not specified.
 
-@item ftp-user
-Set a user to be used for authenticating to the FTP server. This is overridden by the
-user in the FTP URL.
-
-@item ftp-password
-Set a password to be used for authenticating to the FTP server. This is overridden by
-the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
-
 @item ftp-anonymous-password
 Password used when login as anonymous user. Typically an e-mail address
 should be used.
@@ -400,12 +247,6 @@ operation. ff* tools may produce incomplete content due to server limitations.
 
 Gopher protocol.
 
-@section gophers
-
-Gophers protocol.
-
-The Gopher protocol with TLS encapsulation.
-
 @section hls
 
 Read Apple HTTP Live Streaming compliant segmented stream as
@@ -465,6 +306,14 @@ Set the Referer header. Include 'Referer: URL' header in HTTP request.
 Override the User-Agent header. If not specified the protocol will use a
 string describing the libavformat build. ("Lavf/<version>")
 
+@item user-agent
+This is a deprecated option, you can use user_agent instead it.
+
+@item timeout
+Set timeout in microseconds of socket I/O operations used by the underlying low level
+operation. By default it is set to -1, which means that the timeout is
+not specified.
+
 @item reconnect_at_eof
 If set then eof is treated like an error and causes reconnection, this is useful
 for live / endless streams.
@@ -472,13 +321,6 @@ for live / endless streams.
 @item reconnect_streamed
 If set then even streamed/non seekable streams will be reconnected on errors.
 
-@item reconnect_on_network_error
-Reconnect automatically in case of TCP/TLS errors during connect.
-
-@item reconnect_on_http_error
-A comma separated list of HTTP status codes to reconnect on. The list can
-include specific status codes (e.g. '503') or the strings '4xx' / '5xx'.
-
 @item reconnect_delay_max
 Sets the maximum delay in seconds after which to give up reconnecting
 
@@ -556,28 +398,6 @@ Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
 value is -1.
 
-@item auth_type
-
-Set HTTP authentication type. No option for Digest, since this method requires
-getting nonce parameters from the server first and can't be used straight away like
-Basic.
-
-@table @option
-@item none
-Choose the HTTP authentication type automatically. This is the default.
-@item basic
-
-Choose the HTTP basic authentication.
-
-Basic authentication sends a Base64-encoded string that contains a user name and password
-for the client. Base64 is not a form of encryption and should be considered the same as
-sending the user name and password in clear text (Base64 is a reversible encoding).
-If a resource needs to be protected, strongly consider using an authentication scheme
-other than basic authentication. HTTPS/TLS should be used with basic authentication.
-Without these additional security enhancements, basic authentication should not be used
-to protect sensitive or valuable information.
-@end table
-
 @end table
 
 @subsection HTTP Cookies
@@ -632,44 +452,12 @@ audio/mpeg.
 This enables support for Icecast versions < 2.4.0, that do not support the
 HTTP PUT method but the SOURCE method.
 
-@item tls
-Establish a TLS (HTTPS) connection to Icecast.
-
 @end table
 
 @example
 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
 @end example
 
-@section ipfs
-
-InterPlanetary File System (IPFS) protocol support. One can access files stored
-on the IPFS network through so-called gateways. These are http(s) endpoints.
-This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent
-to such a gateway. Users can (and should) host their own node which means this
-protocol will use one's local gateway to access files on the IPFS network.
-
-This protocol accepts the following options:
-
-@table @option
-
-@item gateway
-Defines the gateway to use. When not set, the protocol will first try
-locating the local gateway by looking at @code{$IPFS_GATEWAY}, @code{$IPFS_PATH}
-and @code{$HOME/.ipfs/}, in that order.
-
-@end table
-
-One can use this protocol in 2 ways. Using IPFS:
-@example
-ffplay ipfs://<hash>
-@end example
-
-Or the IPNS protocol (IPNS is mutable IPFS):
-@example
-ffplay ipns://<hash>
-@end example
-
 @section mmst
 
 MMS (Microsoft Media Server) protocol over TCP.
@@ -714,7 +502,7 @@ The accepted syntax is:
 pipe:[@var{number}]
 @end example
 
-If @option{fd} isn't specified, @var{number} is the number corresponding to the file descriptor of the
+@var{number} is the number corresponding to the file descriptor of the
 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{number}
 is not specified, by default the stdout file descriptor will be used
 for writing, stdin for reading.
@@ -741,8 +529,6 @@ Set I/O operation maximum block size, in bytes. Default value is
 @code{INT_MAX}, which results in not limiting the requested block size.
 Setting this value reasonably low improves user termination request reaction
 time, which is valuable if data transmission is slow.
-@item fd
-Set file descriptor.
 @end table
 
 Note that some formats (typically MOV), require the output protocol to
@@ -783,50 +569,6 @@ Example usage:
 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
 @end example
 
-@section rist
-
-Reliable Internet Streaming Transport protocol
-
-The accepted options are:
-@table @option
-@item rist_profile
-Supported values:
-@table @samp
-@item simple
-@item main
-This one is default.
-@item advanced
-@end table
-
-@item buffer_size
-Set internal RIST buffer size in milliseconds for retransmission of data.
-Default value is 0 which means the librist default (1 sec). Maximum value is 30
-seconds.
-
-@item fifo_size
-Size of the librist receiver output fifo in number of packets. This must be a
-power of 2.
-Defaults to 8192 (vs the librist default of 1024).
-
-@item overrun_nonfatal=@var{1|0}
-Survive in case of librist fifo buffer overrun. Default value is 0.
-
-@item pkt_size
-Set maximum packet size for sending data. 1316 by default.
-
-@item log_level
-Set loglevel for RIST logging messages. You only need to set this if you
-explicitly want to enable debug level messages or packet loss simulation,
-otherwise the regular loglevel is respected.
-
-@item secret
-Set override of encryption secret, by default is unset.
-
-@item encryption
-Set encryption type, by default is disabled.
-Acceptable values are 128 and 256.
-@end table
-
 @section rtmp
 
 Real-Time Messaging Protocol.
@@ -941,11 +683,6 @@ URL to player swf file, compute hash/size automatically.
 @item rtmp_tcurl
 URL of the target stream. Defaults to proto://host[:port]/app.
 
-@item tcp_nodelay=@var{1|0}
-Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
-
-@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
-
 @end table
 
 For example to read with @command{ffplay} a multimedia resource named
@@ -1133,9 +870,6 @@ Set the local RTCP port to @var{n}.
 @item pkt_size=@var{n}
 Set max packet size (in bytes) to @var{n}.
 
-@item buffer_size=@var{size}
-Set the maximum UDP socket buffer size in bytes.
-
 @item connect=0|1
 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
 to 0).
@@ -1153,13 +887,6 @@ set to 1) or to a default remote address (if set to 0).
 @item localport=@var{n}
 Set the local RTP port to @var{n}.
 
-@item localaddr=@var{addr}
-Local IP address of a network interface used for sending packets or joining
-multicast groups.
-
-@item timeout=@var{n}
-Set timeout (in microseconds) of socket I/O operations to @var{n}.
-
 This is a deprecated option. Instead, @option{localrtpport} should be
 used.
 
@@ -1204,59 +931,6 @@ Options can be set on the @command{ffmpeg}/@command{ffplay} command
 line, or set in code via @code{AVOption}s or in
 @code{avformat_open_input}.
 
-@subsection Muxer
-The following options are supported.
-
-@table @option
-@item rtsp_transport
-Set RTSP transport protocols.
-
-It accepts the following values:
-@table @samp
-@item udp
-Use UDP as lower transport protocol.
-
-@item tcp
-Use TCP (interleaving within the RTSP control channel) as lower
-transport protocol.
-@end table
-
-Default value is @samp{0}.
-
-@item rtsp_flags
-Set RTSP flags.
-
-The following values are accepted:
-@table @samp
-@item latm
-Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.
-@item rfc2190
-Use RFC 2190 packetization instead of RFC 4629 for H.263.
-@item skip_rtcp
-Don't send RTCP sender reports.
-@item h264_mode0
-Use mode 0 for H.264 in RTP.
-@item send_bye
-Send RTCP BYE packets when finishing.
-@end table
-
-Default value is @samp{0}.
-
-
-@item min_port
-Set minimum local UDP port. Default value is 5000.
-
-@item max_port
-Set maximum local UDP port. Default value is 65000.
-
-@item buffer_size
-Set the maximum socket buffer size in bytes.
-
-@item pkt_size
-Set max send packet size (in bytes). Default value is 1472.
-@end table
-
-@subsection Demuxer
 The following options are supported.
 
 @table @option
@@ -1282,10 +956,6 @@ Use UDP multicast as lower transport protocol.
 @item http
 Use HTTP tunneling as lower transport protocol, which is useful for
 passing proxies.
-
-@item https
-Use HTTPs tunneling as lower transport protocol, which is useful for
-passing proxies and widely used for security consideration.
 @end table
 
 Multiple lower transport protocols may be specified, in that case they are
@@ -1303,9 +973,6 @@ Accept packets only from negotiated peer address and port.
 Act as a server, listening for an incoming connection.
 @item prefer_tcp
 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
-@item satip_raw
-Export raw MPEG-TS stream instead of demuxing. The flag will simply write out
-the raw stream, with the original PAT/PMT/PIDs intact.
 @end table
 
 Default value is @samp{none}.
@@ -1318,7 +985,6 @@ The following flags are accepted:
 @item video
 @item audio
 @item data
-@item subtitle
 @end table
 
 By default it accepts all media types.
@@ -1329,23 +995,21 @@ Set minimum local UDP port. Default value is 5000.
 @item max_port
 Set maximum local UDP port. Default value is 65000.
 
-@item listen_timeout
+@item timeout
-Set maximum timeout (in seconds) to establish an initial connection. Setting
+Set maximum timeout (in seconds) to wait for incoming connections.
-@option{listen_timeout} > 0 sets @option{rtsp_flags} to @samp{listen}. Default is -1
+
-which means an infinite timeout when @samp{listen} mode is set.
+A value of -1 means infinite (default). This option implies the
+@option{rtsp_flags} set to @samp{listen}.
 
 @item reorder_queue_size
 Set number of packets to buffer for handling of reordered packets.
 
-@item timeout
+@item stimeout
 Set socket TCP I/O timeout in microseconds.
 
-@item user_agent
+@item user-agent
 Override User-Agent header. If not specified, it defaults to the
 libavformat identifier string.
-
-@item buffer_size
-Set the maximum socket buffer size in bytes.
 @end table
 
 When receiving data over UDP, the demuxer tries to reorder received packets
@@ -1523,7 +1187,7 @@ options.
 This protocol accepts the following options.
 
 @table @option
-@item connect_timeout=@var{milliseconds}
+@item connect_timeout
 Connection timeout; SRT cannot connect for RTT > 1500 msec
 (2 handshake exchanges) with the default connect timeout of
 3 seconds. This option applies to the caller and rendezvous
@@ -1554,7 +1218,7 @@ IP Type of Service. Applies to sender only. Default value is 0xB8.
 @item ipttl=@var{ttl}
 IP Time To Live. Applies to sender only. Default value is 64.
 
-@item latency=@var{microseconds}
+@item latency
 Timestamp-based Packet Delivery Delay.
 Used to absorb bursts of missed packet retransmissions.
 This flag sets both @option{rcvlatency} and @option{peerlatency}
@@ -1565,7 +1229,7 @@ when side is sender and @option{rcvlatency}
 when side is receiver, and the bidirectional stream
 sending is not supported.
 
-@item listen_timeout=@var{microseconds}
+@item listen_timeout
 Set socket listen timeout.
 
 @item maxbw=@var{bytes/seconds}
@@ -1610,32 +1274,6 @@ only if @option{pbkeylen} is non-zero. It is used on
 the receiver only if the received data is encrypted.
 The configured passphrase cannot be recovered (write-only).
 
-@item enforced_encryption=@var{1|0}
-If true, both connection parties must have the same password
-set (including empty, that is, with no encryption). If the
-password doesn't match or only one side is unencrypted,
-the connection is rejected. Default is true.
-
-@item kmrefreshrate=@var{packets}
-The number of packets to be transmitted after which the
-encryption key is switched to a new key. Default is -1.
--1 means auto (0x1000000 in srt library). The range for
-this option is integers in the 0 - @code{INT_MAX}.
-
-@item kmpreannounce=@var{packets}
-The interval between when a new encryption key is sent and
-when switchover occurs. This value also applies to the
-subsequent interval between when switchover occurs and
-when the old encryption key is decommissioned. Default is -1.
--1 means auto (0x1000 in srt library). The range for
-this option is integers in the 0 - @code{INT_MAX}.
-
-@item snddropdelay=@var{microseconds}
-The sender's extra delay before dropping packets. This delay is
-added to the default drop delay time interval value.
-
-Special value -1: Do not drop packets on the sender at all.
-
 @item payload_size=@var{bytes}
 Sets the maximum declared size of a packet transferred
 during the single call to the sending function in Live
@@ -1651,7 +1289,7 @@ use a bigger maximum frame size, though not greater than
 @item pkt_size=@var{bytes}
 Alias for @samp{payload_size}.
 
-@item peerlatency=@var{microseconds}
+@item peerlatency
 The latency value (as described in @option{rcvlatency}) that is
 set by the sender side as a minimum value for the receiver.
 
@@ -1663,7 +1301,7 @@ Not required on receiver (set to 0),
 key size obtained from sender in HaiCrypt handshake.
 Default value is 0.
 
-@item rcvlatency=@var{microseconds}
+@item rcvlatency
 The time that should elapse since the moment when the
 packet was sent and the moment when it's delivered to
 the receiver application in the receiving function.
@@ -1681,10 +1319,12 @@ Set UDP receive buffer size, expressed in bytes.
 @item send_buffer_size=@var{bytes}
 Set UDP send buffer size, expressed in bytes.
 
-@item timeout=@var{microseconds}
+@item rw_timeout
-Set raise error timeouts for read, write and connect operations. Note that the
+Set raise error timeout for read/write optations.
-SRT library has internal timeouts which can be controlled separately, the
+
-value set here is only a cap on those.
+This option is only relevant in read mode:
+if no data arrived in more than this time
+interval, raise error.
 
 @item tlpktdrop=@var{1|0}
 Too-late Packet Drop. When enabled on receiver, it skips
@@ -1735,9 +1375,6 @@ This option doesn’t make sense in Rendezvous connection; the result
 might be that simply one side will override the value from the other
 side and it’s the matter of luck which one would win
 
-@item srt_streamid=@var{string}
-Alias for @samp{streamid} to avoid conflict with ffmpeg command line option.
-
 @item smoother=@var{live|file}
 The type of Smoother used for the transmission for that socket, which
 is responsible for the transmission and congestion control. The Smoother
@@ -1781,17 +1418,6 @@ the overhead transmission (retransmitted and control packets).
 file: Set options as for non-live transmission. See @option{messageapi}
 for further explanations
 
-@item linger=@var{seconds}
-The number of seconds that the socket waits for unsent data when closing.
-Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
-seconds in file mode). The range for this option is integers in the
-0 - @code{INT_MAX}.
-
-@item tsbpd=@var{1|0}
-When true, use Timestamp-based Packet Delivery mode. The default behavior
-depends on the transmission type: enabled in live mode, disabled in file
-mode.
-
 @end table
 
 For more information see: @url{https://github.com/Haivision/srt}.
@@ -1878,9 +1504,8 @@ tcp://@var{hostname}:@var{port}[?@var{options}]
 The list of supported options follows.
 
 @table @option
-@item listen=@var{2|1|0}
+@item listen=@var{1|0}
-Listen for an incoming connection. 0 disables listen, 1 enables listen in
+Listen for an incoming connection. Default value is 0.
-single client mode, 2 enables listen in multi-client mode. Default value is 0.
 
 @item timeout=@var{microseconds}
 Set raise error timeout, expressed in microseconds.
@@ -1900,8 +1525,6 @@ Set send buffer size, expressed bytes.
 @item tcp_nodelay=@var{1|0}
 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
 
-@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
-
 @item tcp_mss=@var{bytes}
 Set maximum segment size for outgoing TCP packets, expressed in bytes.
 @end table
@@ -1958,10 +1581,6 @@ A file containing the private key for the certificate.
 If enabled, listen for connections on the provided port, and assume
 the server role in the handshake instead of the client role.
 
-@item http_proxy
-The HTTP proxy to tunnel through, e.g. @code{http://example.com:1234}.
-The proxy must support the CONNECT method.
-
 @end table
 
 Example command lines:
@@ -2000,7 +1619,7 @@ The list of supported options follows.
 @item buffer_size=@var{size}
 Set the UDP maximum socket buffer size in bytes. This is used to set either
 the receive or send buffer size, depending on what the socket is used for.
-Default is 32 KB for output, 384 KB for input.  See also @var{fifo_size}.
+Default is 64KB.  See also @var{fifo_size}.
 
 @item bitrate=@var{bitrate}
 If set to nonzero, the output will have the specified constant bitrate if the
@@ -2109,50 +1728,4 @@ Timeout in ms.
 Create the Unix socket in listening mode.
 @end table
 
-@section zmq
-
-ZeroMQ asynchronous messaging using the libzmq library.
-
-This library supports unicast streaming to multiple clients without relying on
-an external server.
-
-The required syntax for streaming or connecting to a stream is:
-@example
-zmq:tcp://ip-address:port
-@end example
-
-Example:
-Create a localhost stream on port 5555:
-@example
-ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
-@end example
-
-Multiple clients may connect to the stream using:
-@example
-ffplay zmq:tcp://127.0.0.1:5555
-@end example
-
-Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
-The server side binds to a port and publishes data. Clients connect to the
-server (via IP address/port) and subscribe to the stream. The order in which
-the server and client start generally does not matter.
-
-ffmpeg must be compiled with the --enable-libzmq option to support
-this protocol.
-
-Options can be set on the @command{ffmpeg}/@command{ffplay} command
-line. The following options are supported:
-
-@table @option
-
-@item pkt_size
-Forces the maximum packet size for sending/receiving data. The default value is
-131,072 bytes. On the server side, this sets the maximum size of sent packets
-via ZeroMQ. On the clients, it sets an internal buffer size for receiving
-packets. Note that pkt_size on the clients should be equal to or greater than
-pkt_size on the server. Otherwise the received message may be truncated causing
-decoding errors.
-
-@end table
-
 @c man end PROTOCOLS

doc/resampler.texi

@@ -11,8 +11,18 @@ programmatic use.
 
 @table @option
 
-@item uchl, used_chlayout
+@item ich, in_channel_count
-Set used input channel layout. Default is unset. This option is
+Set the number of input channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{in_channel_layout} is set.
+
+@item och, out_channel_count
+Set the number of output channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{out_channel_layout} is set.
+
+@item uch, used_channel_count
+Set the number of used input channels. Default value is 0. This option is
 only used for special remapping.
 
 @item isr, in_sample_rate
@@ -31,8 +41,8 @@ Specify the output sample format. It is set by default to @code{none}.
 Set the internal sample format. Default value is @code{none}.
 This will automatically be chosen when it is not explicitly set.
 
-@item ichl, in_chlayout
+@item icl, in_channel_layout
-@item ochl, out_chlayout
+@item ocl, out_channel_layout
 Set the input/output channel layout.
 
 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}

doc/t2h.pm

@@ -20,45 +20,8 @@
 # License along with FFmpeg; if not, write to the Free Software
 # Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 
-# Texinfo 7.0 changed the syntax of various functions.
-# Provide a shim for older versions.
-sub ff_set_from_init_file($$) {
-    my $key = shift;
-    my $value = shift;
-    if (exists &{'texinfo_set_from_init_file'}) {
-        texinfo_set_from_init_file($key, $value);
-    } else {
-        set_from_init_file($key, $value);
-    }
-}
-
-sub ff_get_conf($) {
-    my $key = shift;
-    if (exists &{'texinfo_get_conf'}) {
-        texinfo_get_conf($key);
-    } else {
-        get_conf($key);
-    }
-}
-
-sub get_formatting_function($$) {
-    my $obj = shift;
-    my $func = shift;
-
-    my $sub = $obj->can('formatting_function');
-    if ($sub) {
-        return $obj->formatting_function($func);
-    } else {
-        return $obj->{$func};
-    }
-}
-
-# determine texinfo version
-my $program_version_num = version->declare(ff_get_conf('PACKAGE_VERSION'))->numify;
-my $program_version_6_8 = $program_version_num >= 6.008000;
-
 # no navigation elements
-ff_set_from_init_file('HEADERS', 0);
+set_from_init_file('HEADERS', 0);
 
 sub ffmpeg_heading_command($$$$$)
 {
@@ -92,7 +55,7 @@ sub ffmpeg_heading_command($$$$$)
         $element = $command->{'parent'};
     }
     if ($element) {
-        $result .= &{get_formatting_function($self, 'format_element_header')}($self, $cmdname,
+        $result .= &{$self->{'format_element_header'}}($self, $cmdname,
                                                        $command, $element);
     }
 
@@ -149,11 +112,7 @@ sub ffmpeg_heading_command($$$$$)
                 $cmdname
                     = $Texinfo::Common::level_to_structuring_command{$cmdname}->[$heading_level];
             }
-            # format_heading_text expects an array of headings for texinfo >= 7.0
+            $result .= &{$self->{'format_heading_text'}}(
-            if ($program_version_num >= 7.000000) {
-                $heading = [$heading];
-            }
-            $result .= &{get_formatting_function($self,'format_heading_text')}(
                         $self, $cmdname, $heading,
                         $heading_level +
                         $self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command);
@@ -168,18 +127,14 @@ foreach my $command (keys(%Texinfo::Common::sectioning_commands), 'node') {
 }
 
 # print the TOC where @contents is used
-if ($program_version_6_8) {
+set_from_init_file('INLINE_CONTENTS', 1);
-    ff_set_from_init_file('CONTENTS_OUTPUT_LOCATION', 'inline');
-} else {
-    ff_set_from_init_file('INLINE_CONTENTS', 1);
-}
 
 # make chapters <h2>
-ff_set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
+set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
 
 # Do not add <hr>
-ff_set_from_init_file('DEFAULT_RULE', '');
+set_from_init_file('DEFAULT_RULE', '');
-ff_set_from_init_file('BIG_RULE', '');
+set_from_init_file('BIG_RULE', '');
 
 # Customized file beginning
 sub ffmpeg_begin_file($$$)
@@ -196,18 +151,7 @@ sub ffmpeg_begin_file($$$)
     my ($title, $description, $encoding, $date, $css_lines,
         $doctype, $bodytext, $copying_comment, $after_body_open,
         $extra_head, $program_and_version, $program_homepage,
-        $program, $generator);
+        $program, $generator) = $self->_file_header_informations($command);
-    if ($program_version_num >= 7.000000) {
-        ($title, $description, $encoding, $date, $css_lines,
-         $doctype, $bodytext, $copying_comment, $after_body_open,
-         $extra_head, $program_and_version, $program_homepage,
-         $program, $generator) = $self->_file_header_information($command);
-    } else {
-        ($title, $description, $encoding, $date, $css_lines,
-         $doctype, $bodytext, $copying_comment, $after_body_open,
-         $extra_head, $program_and_version, $program_homepage,
-         $program, $generator) = $self->_file_header_informations($command);
-    }
 
     my $links = $self->_get_links ($filename, $element);
 
@@ -240,11 +184,7 @@ EOT
 
     return $head1 . $head_title . $head2 . $head_title . $head3;
 }
-if ($program_version_6_8) {
+texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
-    texinfo_register_formatting_function('format_begin_file', \&ffmpeg_begin_file);
-} else {
-    texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
-}
 
 sub ffmpeg_program_string($)
 {
@@ -261,17 +201,13 @@ sub ffmpeg_program_string($)
       $self->gdt('This document was generated automatically.'));
   }
 }
-if ($program_version_6_8) {
+texinfo_register_formatting_function('program_string', \&ffmpeg_program_string);
-    texinfo_register_formatting_function('format_program_string', \&ffmpeg_program_string);
-} else {
-    texinfo_register_formatting_function('program_string', \&ffmpeg_program_string);
-}
 
 # Customized file ending
 sub ffmpeg_end_file($)
 {
     my $self = shift;
-    my $program_string = &{get_formatting_function($self,'format_program_string')}($self);
+    my $program_string = &{$self->{'format_program_string'}}($self);
     my $program_text = <<EOT;
       <p style="font-size: small;">
         $program_string
@@ -284,15 +220,11 @@ EOT
 EOT
     return $program_text . $footer;
 }
-if ($program_version_6_8) {
+texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
-    texinfo_register_formatting_function('format_end_file', \&ffmpeg_end_file);
-} else {
-    texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
-}
 
 # Dummy title command
 # Ignore title. Title is handled through ffmpeg_begin_file().
-ff_set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
+set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
 sub ffmpeg_title($$$$)
 {
     return '';
@@ -310,14 +242,8 @@ sub ffmpeg_float($$$$$)
     my $args = shift;
     my $content = shift;
 
-    my ($caption, $prepended);
+    my ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
-    if ($program_version_num >= 7.000000) {
+                                                                $command);
-        ($caption, $prepended) = Texinfo::Convert::Converter::float_name_caption($self,
-                                                                                 $command);
-    } else {
-        ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
-                                                                     $command);
-    }
     my $caption_text = '';
     my $prepended_text;
     my $prepended_save = '';
@@ -389,13 +315,8 @@ sub ffmpeg_float($$$$$)
             $caption->{'args'}->[0], 'float caption');
     }
     if ($prepended_text.$caption_text ne '') {
-        if ($program_version_num >= 7.000000) {
+        $prepended_text = $self->_attribute_class('div','float-caption'). '>'
-            $prepended_text = $self->html_attribute_class('div',['float-caption']). '>'
+                . $prepended_text;
-                    . $prepended_text;
-        } else {
-            $prepended_text = $self->_attribute_class('div','float-caption'). '>'
-                    . $prepended_text;
-        }
         $caption_text .= '</div>';
     }
     my $html_class = '';
@@ -408,13 +329,8 @@ sub ffmpeg_float($$$$$)
         $prepended_text = '';
         $caption_text   = '';
     }
-    if ($program_version_num >= 7.000000) {
+    return $self->_attribute_class('div', $html_class). '>' . "\n" .
-        return $self->html_attribute_class('div', [$html_class]). '>' . "\n" .
+        $prepended_text . $caption_text . $content . '</div>';
-            $prepended_text . $caption_text . $content . '</div>';
-    } else {
-        return $self->_attribute_class('div', $html_class). '>' . "\n" .
-            $prepended_text . $caption_text . $content . '</div>';
-    }
 }
 
 texinfo_register_command_formatting('float',

doc/texi2pod.pl

@@ -172,9 +172,6 @@ INF: while(<$inf>) {
         } elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
             $_ = "\n=back\n";
             $ic = pop @icstack;
-        } elsif ($ended =~ /^float$/) {
-            $_ = "\n=back\n";
-            $ic = pop @icstack;
         } else {
             die "unknown command \@end $ended at line $.\n";
         }
@@ -300,12 +297,6 @@ INF: while(<$inf>) {
         $_ = "";        # need a paragraph break
     };
 
-    /^\@(float)\s+\w+/ and do {
-        push @endwstack, $endw;
-        $endw = $1;
-        $_ = "\n=over 4\n";
-    };
-
     /^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
         my $columns = $1;
         $columns =~ s/\@tab/ : /;

doc/utils.texi

@@ -110,13 +110,11 @@ maximum of 2 digits. The @var{m} at the end expresses decimal value for
 @emph{or}
 
 @example
-[-]@var{S}+[.@var{m}...][s|ms|us]
+[-]@var{S}+[.@var{m}...]
 @end example
 
 @var{S} expresses the number of seconds, with the optional decimal part
-@var{m}.  The optional literal suffixes @samp{s}, @samp{ms} or @samp{us}
+@var{m}.
-indicate to interpret the value as seconds, milliseconds or microseconds,
-respectively.
 
 In both expressions, the optional @samp{-} indicates negative duration.
 
@@ -128,15 +126,6 @@ The following examples are all valid time duration:
 @item 55
 55 seconds
 
-@item 0.2
-0.2 seconds
-
-@item 200ms
-200 milliseconds, that's 0.2s
-
-@item 200000us
-200000 microseconds, that's 0.2s
-
 @item 12:03:45
 12 hours, 03 minutes and 45 seconds
 
@@ -713,42 +702,26 @@ FL+FR+FC+LFE+BL+BR+SL+SR
 FL+FR+FC+LFE+BL+BR+FLC+FRC
 @item 7.1(wide-side)
 FL+FR+FC+LFE+FLC+FRC+SL+SR
-@item 7.1(top)
-FL+FR+FC+LFE+BL+BR+TFL+TFR
 @item octagonal
 FL+FR+FC+BL+BR+BC+SL+SR
-@item cube
-FL+FR+BL+BR+TFL+TFR+TBL+TBR
-@item hexadecagonal
-FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
 @item downmix
 DL+DR
-@item 22.2
-FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR
 @end table
 
-A custom channel layout can be specified as a sequence of terms, separated by '+'.
+A custom channel layout can be specified as a sequence of terms, separated by
-Each term can be:
+'+' or '|'. Each term can be:
 @itemize
-@item
-the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.),
-each optionally containing a custom name after a '@@', (e.g. @samp{FL@@Left},
-@samp{FR@@Right}, @samp{FC@@Center}, @samp{LFE@@Low_Frequency}, etc.)
-@end itemize
-
-A standard channel layout can be specified by the following:
-@itemize
-@item
-the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
-
 @item
 the name of a standard channel layout (e.g. @samp{mono},
 @samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.)
 
+@item
+the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
+
 @item
 a number of channels, in decimal, followed by 'c', yielding the default channel
 layout for that number of channels (see the function
-@code{av_channel_layout_default}). Note that not all channel counts have a
+@code{av_get_default_channel_layout}). Note that not all channel counts have a
 default layout.
 
 @item
@@ -765,7 +738,7 @@ Before libavutil version 53 the trailing character "c" to specify a number of
 channels was optional, but now it is required, while a channel layout mask can
 also be specified as a decimal number (if and only if not followed by "c" or "C").
 
-See also the function @code{av_channel_layout_from_string} defined in
+See also the function @code{av_get_channel_layout} defined in
 @file{libavutil/channel_layout.h}.
 @c man end SYNTAX
 
@@ -947,9 +920,6 @@ corresponding input value will be returned.
 @item round(expr)
 Round the value of expression @var{expr} to the nearest integer. For example, "round(1.5)" is "2.0".
 
-@item sgn(x)
-Compute sign of @var{x}.
-
 @item sin(x)
 Compute sine of @var{x}.
 
@@ -1077,13 +1047,13 @@ indication of the corresponding powers of 10 and of 2.
 @item T
 10^12 / 2^40
 @item P
-10^15 / 2^50
+10^15 / 2^40
 @item E
-10^18 / 2^60
+10^18 / 2^50
 @item Z
-10^21 / 2^70
+10^21 / 2^60
 @item Y
-10^24 / 2^80
+10^24 / 2^70
 @end table
 
 @c man end EXPRESSION EVALUATION

doc/writing_filters.txt

@@ -418,4 +418,4 @@ done:
 
 When all of this is done, you can submit your patch to the ffmpeg-devel
 mailing-list for review.  If you need any help, feel free to come on our IRC
-channel, #ffmpeg-devel on irc.libera.chat.
+channel, #ffmpeg-devel on irc.freenode.net.

ffbuild/.gitignore

@@ -1,6 +1,4 @@
 /.config
-/bin2c
-/bin2c.exe
 /config.fate
 /config.log
 /config.mak

ffbuild/arch.mak

@@ -8,14 +8,10 @@ OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPSDSP)   += $(MIPSDSP-OBJS)    $(MIPSDSP-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
-OBJS-$(HAVE_MMI)       += $(MMI-OBJS)        $(MMI-OBJS-yes)
+OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)
-OBJS-$(HAVE_LSX)       += $(LSX-OBJS)        $(LSX-OBJS-yes)
-OBJS-$(HAVE_LASX)      += $(LASX-OBJS)       $(LASX-OBJS-yes)
 
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
 
-OBJS-$(HAVE_RVV)     += $(RVV-OBJS)     $(RVV-OBJS-yes)
-
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_X86ASM)  += $(X86ASM-OBJS)  $(X86ASM-OBJS-yes)

ffbuild/bin2c.c

@@ -1,76 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * 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.
- */
-
-#include <string.h>
-#include <stdio.h>
-
-int main(int argc, char **argv)
-{
-    const char *name;
-    FILE *input, *output;
-    unsigned int length = 0;
-    unsigned char data;
-
-    if (argc < 3 || argc > 4)
-        return 1;
-
-    input = fopen(argv[1], "rb");
-    if (!input)
-        return -1;
-
-    output = fopen(argv[2], "wb");
-    if (!output)
-        return -1;
-
-    if (argc == 4) {
-        name = argv[3];
-    } else {
-        size_t arglen = strlen(argv[1]);
-        name = argv[1];
-
-        for (int i = 0; i < arglen; i++) {
-            if (argv[1][i] == '.')
-                argv[1][i] = '_';
-            else if (argv[1][i] == '/')
-                name = &argv[1][i+1];
-        }
-    }
-
-    fprintf(output, "const unsigned char ff_%s_data[] = { ", name);
-
-    while (fread(&data, 1, 1, input) > 0) {
-        fprintf(output, "0x%02x, ", data);
-        length++;
-    }
-
-    fprintf(output, "0x00 };\n");
-    fprintf(output, "const unsigned int ff_%s_len = %u;\n", name, length);
-
-    fclose(output);
-
-    if (ferror(input) || !feof(input))
-        return -1;
-
-    fclose(input);
-
-    return 0;
-}

ffbuild/common.mak

@@ -12,13 +12,10 @@ endif
 
 ifndef SUBDIR
 
-BIN2CEXE = ffbuild/bin2c$(HOSTEXESUF)
-BIN2C = $(BIN2CEXE)
-
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC BIN2C
+BRIEF  = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
 SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
 
 MSG    = $@
@@ -29,8 +26,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-# Prepend to a recursively expanded variable without making it simply expanded.
+ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
-PREPEND = $(eval $(1) = $(patsubst %,$$(%), $(2)) $(value $(1)))
 
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_LINK)/
@@ -40,9 +36,7 @@ CCFLAGS     = $(CPPFLAGS) $(CFLAGS)
 OBJCFLAGS  += $(EOBJCFLAGS)
 OBJCCFLAGS  = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
 ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
-# Use PREPEND here so that later (target-dependent) additions to CPPFLAGS
+CXXFLAGS   := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
-# end up in CXXFLAGS.
-$(call PREPEND,CXXFLAGS, CPPFLAGS CFLAGS)
 X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
 
 HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
@@ -50,7 +44,7 @@ LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
 
 define COMPILE
        $(call $(1)DEP,$(1))
-       $($(1)) $($(1)FLAGS) $($(2)) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
+       $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
 endef
 
 COMPILE_C = $(call COMPILE,CC)
@@ -60,22 +54,6 @@ COMPILE_M = $(call COMPILE,OBJCC)
 COMPILE_X86ASM = $(call COMPILE,X86ASM)
 COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 COMPILE_NVCC = $(call COMPILE,NVCC)
-COMPILE_MMI = $(call COMPILE,CC,MMIFLAGS)
-COMPILE_MSA = $(call COMPILE,CC,MSAFLAGS)
-COMPILE_LSX = $(call COMPILE,CC,LSXFLAGS)
-COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
-
-%_mmi.o: %_mmi.c
-	$(COMPILE_MMI)
-
-%_msa.o: %_msa.c
-	$(COMPILE_MSA)
-
-%_lsx.o: %_lsx.c
-	$(COMPILE_LSX)
-
-%_lasx.o: %_lasx.c
-	$(COMPILE_LASX)
 
 %.o: %.c
 	$(COMPILE_C)
@@ -104,7 +82,7 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 	-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
 
 %.o: %.rc
-	$(WINDRES) $(IFLAGS) $(foreach ARG,$(CC_DEPFLAGS),--preprocessor-arg "$(ARG)") -o $@ $<
+	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
 
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
@@ -112,40 +90,16 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 %.h.c:
 	$(Q)echo '#include "$*.h"' >$@
 
-$(BIN2CEXE): ffbuild/bin2c_host.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
-
-%.metal.air: %.metal
-	$(METALCC) $< -o $@
-
-%.metallib: %.metal.air
-	$(METALLIB) --split-module-without-linking $< -o $@
-
-%.metallib.c: %.metallib $(BIN2CEXE)
-	$(BIN2C) $< $@ $(subst .,_,$(basename $(notdir $@)))
-
 %.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
 	$(COMPILE_NVCC)
 
-ifdef CONFIG_PTX_COMPRESSION
+%.ptx.c: %.ptx
-%.ptx.gz: TAG = GZIP
+	$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
-%.ptx.gz: %.ptx
-	$(M)gzip -c9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
-
-%.ptx.c: %.ptx.gz $(BIN2CEXE)
-	$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
-else
-%.ptx.c: %.ptx $(BIN2CEXE)
-	$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
-endif
-
-clean::
-	$(RM) $(BIN2CEXE)
 
 %.c %.h %.pc %.ver %.version: TAG = GEN
 
 # Dummy rule to stop make trying to rebuild removed or renamed headers
-%.h %_template.c:
+%.h:
 	@:
 
 # Disable suffix rules.  Most of the builtin rules are suffix rules,
@@ -160,8 +114,6 @@ include $(SRC_PATH)/ffbuild/arch.mak
 
 OBJS      += $(OBJS-yes)
 SLIBOBJS  += $(SLIBOBJS-yes)
-SHLIBOBJS += $(SHLIBOBJS-yes)
-STLIBOBJS += $(STLIBOBJS-yes)
 FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
@@ -170,8 +122,6 @@ FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=
 
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
 SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
-SHLIBOBJS := $(sort $(SHLIBOBJS:%=$(SUBDIR)%))
-STLIBOBJS := $(sort $(STLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -193,7 +143,7 @@ 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=.gz) $(PTXOBJS:.o=)
+.SECONDARY:   $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)
 
 alltools: $(TOOLS)
 
@@ -207,14 +157,12 @@ $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
-$(SHLIBOBJS): | $(sort $(dir $(SHLIBOBJS)))
-$(STLIBOBJS): | $(sort $(dir $(STLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools
 
-OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(SHLIBOBJS) $(STLIBOBJS) $(TESTOBJS))
+OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.gz *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
+CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
 define RULES
@@ -224,4 +172,4 @@ endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SHLIBOBJS:.o=.d) $(STLIBOBJS:.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_X86ASMD).d)

ffbuild/library.mak

@@ -14,26 +14,10 @@ INSTHEADERS := $(INSTHEADERS) $(HEADERS:%=$(SUBDIR)%)
 all-$(CONFIG_STATIC): $(SUBDIR)$(LIBNAME)  $(SUBDIR)lib$(FULLNAME).pc
 all-$(CONFIG_SHARED): $(SUBDIR)$(SLIBNAME) $(SUBDIR)lib$(FULLNAME).pc
 
-LIBOBJS := $(OBJS) $(SHLIBOBJS) $(STLIBOBJS) $(SUBDIR)%.h.o $(TESTOBJS)
+LIBOBJS := $(OBJS) $(SUBDIR)%.h.o $(TESTOBJS)
 $(LIBOBJS) $(LIBOBJS:.o=.s) $(LIBOBJS:.o=.i):   CPPFLAGS += -DHAVE_AV_CONFIG_H
 
-ifdef CONFIG_SHARED
+$(SUBDIR)$(LIBNAME): $(OBJS)
-# In case both shared libs and static libs are enabled, it can happen
-# that a user might want to link e.g. libavformat statically, but
-# libavcodec and the other libs dynamically. In this case
-# libavformat won't be able to access libavcodec's internal symbols,
-# so that they have to be duplicated into the archive just like
-# for purely shared builds.
-# Test programs are always statically linked against their library
-# to be able to access their library's internals, even with shared builds.
-# Yet linking against dependend libraries still uses dynamic linking.
-# This means that we are in the scenario described above.
-# In case only static libs are used, the linker will only use
-# one of these copies; this depends on the duplicated object files
-# containing exactly the same symbols.
-OBJS += $(SHLIBOBJS)
-endif
-$(SUBDIR)$(LIBNAME): $(OBJS) $(STLIBOBJS)
 	$(RM) $@
 	$(AR) $(ARFLAGS) $(AR_O) $^
 	$(RANLIB) $@
@@ -52,8 +36,8 @@ $(LIBOBJS): CPPFLAGS += -DBUILDING_$(NAME)
 $(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
 	$$(LD) $(LDFLAGS) $(LDEXEFLAGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $(FFEXTRALIBS) $$(EXTRALIBS-$$(*F)) $$(ELIBS)
 
-$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h $(SUBDIR)version_major.h | $(SUBDIR)
+$(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h | $(SUBDIR)
-	$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$^ > $$@
+	$$(M) $$(SRC_PATH)/ffbuild/libversion.sh $(NAME) $$< > $$@
 
 $(SUBDIR)lib$(FULLNAME).pc: $(SUBDIR)version.h ffbuild/config.sh | $(SUBDIR)
 	$$(M) $$(SRC_PATH)/ffbuild/pkgconfig_generate.sh $(NAME) "$(DESC)"
@@ -64,7 +48,7 @@ $(SUBDIR)lib$(NAME).ver: $(SUBDIR)lib$(NAME).v $(OBJS)
 $(SUBDIR)$(SLIBNAME): $(SUBDIR)$(SLIBNAME_WITH_MAJOR)
 	$(Q)cd ./$(SUBDIR) && $(LN_S) $(SLIBNAME_WITH_MAJOR) $(SLIBNAME)
 
-$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SHLIBOBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
+$(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SLIBOBJS) $(SUBDIR)lib$(NAME).ver
 	$(SLIB_CREATE_DEF_CMD)
 	$$(LD) $(SHFLAGS) $(LDFLAGS) $(LDSOFLAGS) $$(LD_O) $$(filter %.o,$$^) $(FFEXTRALIBS)
 	$(SLIB_EXTRA_CMD)

ffbuild/libversion.sh

@@ -5,12 +5,8 @@ toupper(){
 name=lib$1
 ucname=$(toupper ${name})
 file=$2
-file2=$3
 
 eval $(awk "/#define ${ucname}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file")
-if [ -f "$file2" ]; then
-  eval $(awk "/#define ${ucname}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file2")
-fi
 eval ${ucname}_VERSION=\$${ucname}_VERSION_MAJOR.\$${ucname}_VERSION_MINOR.\$${ucname}_VERSION_MICRO
 eval echo "${name}_VERSION=\$${ucname}_VERSION"
 eval echo "${name}_VERSION_MAJOR=\$${ucname}_VERSION_MAJOR"

fftools/Makefile

@@ -9,22 +9,16 @@ AVBASENAMES  = ffmpeg ffplay ffprobe
 ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
 
-OBJS-ffmpeg +=                  \
+OBJS-ffmpeg                        += fftools/ffmpeg_opt.o fftools/ffmpeg_filter.o fftools/ffmpeg_hw.o
-    fftools/ffmpeg_demux.o      \
+OBJS-ffmpeg-$(CONFIG_CUVID)        += fftools/ffmpeg_cuvid.o
-    fftools/ffmpeg_filter.o     \
+OBJS-ffmpeg-$(CONFIG_LIBMFX)       += fftools/ffmpeg_qsv.o
-    fftools/ffmpeg_hw.o         \
+ifndef CONFIG_VIDEOTOOLBOX
-    fftools/ffmpeg_mux.o        \
+OBJS-ffmpeg-$(CONFIG_VDA)          += fftools/ffmpeg_videotoolbox.o
-    fftools/ffmpeg_mux_init.o   \
+endif
-    fftools/ffmpeg_opt.o        \
+OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += fftools/ffmpeg_videotoolbox.o
-    fftools/objpool.o           \
-    fftools/sync_queue.o        \
-    fftools/thread_queue.o      \
 
 define DOFFTOOL
-OBJS-$(1) += fftools/cmdutils.o fftools/opt_common.o fftools/$(1).o $(OBJS-$(1)-yes)
+OBJS-$(1) += fftools/cmdutils.o fftools/$(1).o $(OBJS-$(1)-yes)
-ifdef HAVE_GNU_WINDRES
-OBJS-$(1) += fftools/fftoolsres.o
-endif
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): | fftools
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))