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104 Commits
n4.3 ... n3.1.4

Author SHA1 Message Date
Michael Niedermayer
c2ea706282 Changelog: update
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-10-01 02:51:42 +02:00
Michael Niedermayer
622ccbd8ab avformat/avidec: Check nb_streams in read_gab2_sub()
Fixes null pointer dereference
Fixes: 1/null_point.avi

Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2679ad4773)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-10-01 02:50:54 +02:00
Michael Niedermayer
c8c5f66b42 avformat/avidec: Remove ancient assert
This assert can with crafted files fail, a warning is already printed
for this case.

Fixes assertion failure
Fixes:1/assert.avi

Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 14bac7e00d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-10-01 02:50:54 +02:00
James Almer
bc6174d4af Changelog: update after the last few commits
Signed-off-by: James Almer <jamrial@gmail.com>
2016-09-28 17:42:41 -03:00
James Almer
2303cea5be avfilter/vf_colorspace: fix range for output colorspace option
Rreviewed-by: BBB
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit e4bfc9ecf7)
2016-09-28 17:40:10 -03:00
Matthieu Bouron
d0590d9349 lavc/mediacodecdec_h264: fix SODB escaping
Fixes escaping of consecutive 0x00, 0x00, 0x0{0-3} sequences.

(cherry picked from commit f574012d5f)
2016-09-28 16:22:24 +02:00
Timo Rothenpieler
e60a00e0cc avcodec/nvenc: fix const options for hevc gpu setting 2016-09-28 16:10:49 +02:00
Michael Niedermayer
e6351504dc Update for 3.1.4
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:25 +02:00
Michael Niedermayer
8834e080c2 avformat/avidec: Fix memleak with dv in avi
Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b98dafe045)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:07 +02:00
Sasi Inguva
39dc26f0c1 lavc/movtextdec.c: Avoid infinite loop on invalid data.
Signed-off-by: Sasi Inguva <isasi@google.com>
(cherry picked from commit 7e9e1b7070)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:07 +02:00
Michael Niedermayer
496267f8e9 avcodec/ansi: Check dimensions
Fixes: 1.avi

Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 69449da436)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:07 +02:00
Michael Niedermayer
9d738e6968 avcodec/cavsdsp: use av_clip_uint8() for idct
Fixes out of array read
Fixes: 1.swf

Found-by: 连一汉 <lianyihan@360.cn>
Tested-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0e318f110b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:07 +02:00
Michael Niedermayer
77c9c35093 avformat/movenc: Check packet in mov_write_single_packet() too
Fixes assertion failure

Found-by: durandal117
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2834313933)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Michael Niedermayer
03f996d183 avformat/movenc: Factor check_pkt() out
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit deabcd2c05)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Xinzheng Zhang
c68ce48260 avformat/utils: fix timebase error in avformat_seek_file()
When there is only one stream and stream_index has not specified,
The ts has been transferd by the timebase of stream0 without modifying the stream_index
In this condation it cause seek failure.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ecc04b4f2f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Michael Niedermayer
ac8ac46641 avcodec/g726: Add missing ADDB output mask
Fixes: 1.poc
Fixes out of array read

Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a5af1240fc)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Michael Niedermayer
c2087fc48b avcodec/avpacket: clear side_data_elems
Fixes null pointer dereference

Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 5e1bf9d8c0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Michael Niedermayer
21a9797737 avformat/movenc: Check first DTS similar to dts difference
Fixes assertion failure
Fixes: b84b53855a0b74560e64c6f45f505a13/signal_sigabrt_7ffff6ae7c37_3837_ef4e243ea5b4fa8d0becf4afe9166604.avi

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 68f4c2163e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Michael Niedermayer
65c10f0f5c avcodec/ccaption_dec: Use simple array instead of AVBuffer
This is simpler and fixes an out of array read, fixing it with AVBuffers
would be more complex

Fixes: e00d9e6e50e5495cc93fea41147b97bb/asan_heap-oob_12dcdbb_8798_b32a97ea722dd37bb5066812cc674552.mov

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 752e6dfa3e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Michael Niedermayer
ed1c6f701a avcodec/svq3: Reintroduce slice_type
Fixes out of array read
Fixes: 1642cd3962249d6aaf0eec2836023fb6/signal_sigsegv_2557a72_2995_04efaf2ff57a052f609a3b4a2ea4e622.mov

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2d3099ad8e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-28 14:23:06 +02:00
Sergey Volk
7a3dc2f7b6 avformat/mov: Fix potential integer overflow in mov_read_keys
Actual allocation size is computed as (count + 1)*sizeof(meta_keys), so
we need to check that (count + 1) won't cause overflow.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 347cb14b7c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-27 13:42:11 +02:00
Michael Niedermayer
e91b7852df swscale/swscale_unscaled: Try to fix Rgb16ToPlanarRgb16Wrapper() with slices
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e57d99dd4e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-27 13:42:11 +02:00
Michael Niedermayer
5aaf7e3182 swscale/swscale_unscaled: Fix packed_16bpc_bswap() with slices
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 47bc1bdafb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-27 13:42:11 +02:00
Michael Niedermayer
ed38046c5c avformat/avidec: Fix infinite loop in avi_read_nikon()
Fixes: 360/test.poc

Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e4e4a9cad7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-27 13:42:11 +02:00
Michael Niedermayer
ba642f0319 avformat/utils: End probing if the expected codec surpasses AVPROBE_SCORE_STREAM_RETRY
Fixes Ticket5800

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c75273310c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-27 13:42:11 +02:00
Carl Eugen Hoyos
8b21b44e7e lavf/utils: Avoid an overflow for huge negative durations.
Fixes ticket #5135.
(cherry picked from commit 267da70ea8)
2016-09-24 21:07:19 +02:00
Anssi Hannula
748a4747da avformat/hls: Fix handling of EXT-X-BYTERANGE streams over 2GB
Replace uses of atoi() with strtoll() when trying to read values into
int64_t variables.

Fixes Kodi trac #16926:
http://trac.kodi.tv/ticket/16926

(cherry picked from commit a6f5e25ad9)
2016-09-24 09:49:26 +03:00
Carl Eugen Hoyos
6fc29572fb lavc/avpacket: Fix undefined behaviour, do not pass a null pointer to memcpy().
Fixes ticket #5857.
(cherry picked from commit c54eef46f9)
2016-09-22 08:39:40 +02:00
Carl Eugen Hoyos
677ea4a49b lavc/mjpegdec: Do not skip reading quantization tables.
They may contain 0xFFs, confusing the start code finding algorithm.

Fixes ticket #5819.
(cherry picked from commit cef5bc0e6e)
2016-09-03 15:39:33 +02:00
Tobias Rapp
12320c0822 cmdutils: fix implicit declaration of SetDllDirectory function
Pre-processor check changed by commiter.

Signed-off-by: James Almer <jamrial@gmail.com>
2016-08-29 20:00:30 -03:00
James Almer
c46d22a4a5 Changelog: update after last commit
Signed-off-by: James Almer <jamrial@gmail.com>
2016-08-24 20:43:33 -03:00
James Almer
40ab55746e examples/demuxing_decoding: convert to codecpar
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit bba6a03b28)
2016-08-24 20:42:03 -03:00
Michael Niedermayer
949094a4cd Update for 3.1.3 2016-08-25 03:35:17 +02:00
Michael Niedermayer
79f52a0dbd avcodec/exr: Check tile positions
This also disabled the case of mixed x/ymin with tiles, the code
handles these cases inconsistent for the 2 coordinate axis and is
unlikely working correctly.

Fixes crash
Fixes: poc1.exr, poc2.exr

Found-by: Yaoguang Chen of Aliapy unLimit Security Team
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 01aee8148d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:34:55 +02:00
Michael Niedermayer
ae89381962 avcodec/aacenc: Tighter input checks
Fixes occurance of NaN/Inf leading to assertion failures and out of array access
Fixes: d1c38a09acc34845c6be3a127a5aacaf/signal_sigsegv_3982225_6121_d18bd5451d4245ee09408f04badd1b83.wmv

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 77bf96b047)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
596513ca2c avformat/wtvdec: Check pointer before use
Fixes out of array read
Fixes: 049fdf78565f1ce5665df236d90f8657/asan_heap-oob_10a5a97_1026_42f9d4855547329560f385768de2f3fb.wtv

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit cc5e5548df)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
2f07937926 libavcodec/wmalosslessdec: Check the remaining bits
Fixes assertion failure
Fixes: 24ebfda03228b5cc1ef792608cfba458/signal_sigabrt_7ffff6ae7c37_6473_3fa8a111dbc752b1a7c411c5ab79aaa4.wma

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 67318187fb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
4943abe051 avcodec/adpcm: Fix adpcm_ima_wav padding
Fixes out of array read
Fixes: f29f134ea5f5590df554a7733294a587/asan_stack-oob_309d14e_9188_ea01743d6355aff20530f3d4cdaa841a.wav

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f2a9a30fd6)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
8c4a67183b avcodec/svq3: fix slice size check
Fixes out of array read
Fixes: 09f46aa2175cade93e3e3932646a56a9/asan_heap-oob_4a5385_2995_498f6abfdc0248288cefe5f4b7ad316c.mov

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2624695484)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
049d767715 avcodec/diracdec: Check numx/y
Fixes division by 0
Fixes: 60261c4469ba3e11059890fb2832a515/asan_generic_135e694_2790_beb94eaa0aeb7d11c0437375a8964a99.drc

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a31e08fa1a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
8003a5d237 avcodec/h2645_parse: fix nal size
Found-by: <durandal_1707>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 15dd56c093)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
ec30a498e6 avcodec/h2645_parse: Use get_nalsize() in ff_h2645_packet_split()
This fixes several regressions in h.264

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 528171ba84)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Hendrik Leppkes
fabc1c9e56 h2645_parse: only read avc length code at the correct position
Reading it from any other position would result in a wrong size being
read, instead fallback to the re-sync mechanic in the else clause.

(cherry picked from commit c3e9b098e1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Hendrik Leppkes
0ad4d4198a h2645_parse: don't overread AnnexB NALs within an avc stream
We know the maximum size of an AnnexB NAL, signaling it as the maximum
NAL size allows ff_h2645_extract_rbsp to determine the correct size.

(cherry picked from commit 83a940e7fb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
93422bc92e avcodec/h264_parser: Factor get_avc_nalsize() out
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f10ea03df3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:36 +02:00
Michael Niedermayer
22a0c0e764 avcodec/cfhd: Increase minimum band dimension to 3
The implementation does not currently support len=2

Fixes out of array accesses
Fixes: 29d1b3db5ba2205e82b0b3a533e057a3/asan_heap-oob_12b650c_9254_3b8c4e4d931eb2c32841c18ebb297f1d.avi

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit b8b3671721)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:35 +02:00
Michael Niedermayer
77f978996b avcodec/indeo2: check ctab
Fixes out of array access
Fixes: 6b73fa392ac808f02e95a4e0a5770026/asan_static-oob_1b15f9a_1969_e7778535e5f27225fe0d6ded14721430.AVI

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9ffe44c5c7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:35 +02:00
Michael Niedermayer
4770eac663 avformat/swfdec: Fix inflate() error code check
Fixes infinite loop
Fixes endless.poc

Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a453bbb68f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:35 +02:00
Michael Niedermayer
afd57722e1 avcodec/rawdec: Fix bits_per_coded_sample checks
Fixes assertion failure
Fixes: 9eb9cf5b8c26dd0fa7107ed0348dcc1f/signal_sigabrt_7ffff6ae7c37_8926_4609a5c3f071d555d2d557625f9687b1.swf

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 237207645b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:35 +02:00
Michael Niedermayer
7d42daeea2 vcodec/h2645_parse: Clear buffer padding
Fixes use of uninitialized memory
Fixes: 044100cb22845944988a4bd821ff8074/asan_heap-oob_329927a_1366_c3de34ce9217dac820fbb46171031bbb.jsv

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 382a68b008)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:35 +02:00
Michael Niedermayer
055e5c80ee avcodec/h2645: Fix NAL unit padding
The parser changes have lost the support for the needed padding, this adds it back
Fixes out of array reads
Fixes: 03ea21d271abc8acf428d42ace51d8b4/asan_heap-oob_3358eef_5692_16f0cc01ab5225e9ce591659e5c20e35.mkv

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit cc13bc8c4f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:35 +02:00
Michael Niedermayer
905372be8f avfilter/drawutils: Fix single plane with alpha
Fixes Ticket5720

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 369ed11e3c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-25 03:29:35 +02:00
James Almer
f4b8892ccb cmdutils: check for SetDllDirectory() availability
It's only available on Windows XP or newer.

Should fix compilation with mingw32 using the default OS target.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
2016-08-22 19:25:50 -03:00
Michael Niedermayer
4275b27a23 Update for 3.1.2
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-08 21:42:18 +02:00
Hendrik Leppkes
9745c5ebf8 cmdutils: remove the current working directory from the DLL search path on win32
Reviewed-by: Matt Oliver <protogonoi@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3bf142c773)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-08 18:41:13 +02:00
Michael Niedermayer
19d2921bbf avcodec/rawdec: Fix palette handling with changing palettes
Fixes out of array access

Fixes: poc.swf
Found-by: 连一汉 <lianyihan@360.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 6aa39080cc)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-08 18:40:56 +02:00
Michael Niedermayer
e160064d39 avcodec/raw: Fix decoding of ilacetest.mov
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit bbec14de31)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-07 17:33:59 +02:00
Michael Niedermayer
a75a7feebd avformat/mov: Enable mp3 parsing if a packet needs it
Fixes Ticket5689

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 803c058a6f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 22:44:47 +02:00
Anssi Hannula
309fa24f36 avformat/hls: Use an array instead of stream offset for stream mapping
This will be useful when the amount of streams per subdemuxer is not
known at hls_read_header time in a following commit.

(cherry picked from commit 9884f17e34)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 03:43:50 +02:00
Anssi Hannula
3586c68687 avformat/hls: Sync starting segment across variants on live streams
This will avoid a large time difference between variants in the most
common case.

(cherry picked from commit 4d85069e5d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 03:43:47 +02:00
Anssi Hannula
456cf87de9 avformat/hls: Fix regression with ranged media segments
Commit 81306fd4bdf ("hls: eliminate ffurl_* usage", merged in d0fc5de3a6)
changed the hls demuxer to use AVIOContext instead of URLContext for its
HTTP requests.

HLS demuxer uses the "offset" option of the http demuxer, requesting
the initial file offset for the I/O (http URLProtocol uses the "Range:"
HTTP header to try to accommodate that).

However, the code in libavformat/aviobuf.c seems to be doing its own
accounting for the current file offset (AVIOContext.pos), with the
assumption that the initial offset is always zero.

HLS demuxer does an explicit seek after open_url to account for cases
where the "offset" was not effective (due to the URL being a local file
or the HTTP server not obeying it), which should be a no-op in case the
file offset is already at that position.

However, since aviobuf.c code thinks the starting offset is 0, this
doesn't work properly.

This breaks retrieval of ranged media segments.

To fix the regression, just drop the seek call from the HLS demuxer when
the HTTP(S) protocol is used.

(cherry picked from commit 9cb30f7a88)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 03:43:42 +02:00
Michael Niedermayer
54d48c8e90 avcodec/ffv1enc: Fix assertion failure with non zero bits per sample
Fixes Ticket5736
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

(cherry picked from commit c1bfeda5a3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 02:24:06 +02:00
Burt P
43407bde3e avfilter/af_hdcd: small fix in af_hdcd.c where gain was not being adjusted for "attenuate slowly"
Signed-off-by: Burt P <pburt0@gmail.com>
Taken from ba69a81019
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 00:37:41 +02:00
Michael Niedermayer
7c9ee83d2f avformat/oggdec: Fix integer overflow with invalid pts
If negative pts are possible for some codecs in ogg then the code needs to be
changed to use signed values.

Found-by: Thomas Guilbert <tguilbert@google.com>
Fixes: clusterfuzz_usan-2016-08-02
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c5cc3b08e5)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 00:34:54 +02:00
Michael Niedermayer
67f421fd77 ffplay: Fix invalid array index
Found-by: Thomas Guilbert <tguilbert@google.com>
Fixes: clusterfuzz_usan-2016-08-02
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 6cd9a8b67a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-06 00:34:54 +02:00
Paul B Mahol
46732e6a55 avcodec/alacenc: allocate bigger packets
(cherry picked from commit 82b84c71b0)
2016-08-05 23:02:27 +02:00
Steven Robertson
5222f660d7 libavcodec/dnxhd: Enable 12-bit DNxHR support.
10- and 12-bit DNxHR use the same DC coefficient decoding process and
VLC table, just with a different shift value. From SMPTE 2019-1:2016,
8.2.4 DC Coefficient Decoding:

"For 8-bit video sampling, the maximum value of η=11 and for
10-/12-bit video sampling, the maximum value of η=13."

A sample file will be uploaded to show that with this patch, things
decode correctly:
dnxhr_hqx_12bit_1080p_smpte_colorbars_davinci_resolve.mov

Signed-off-by: Steven Robertson <steven@strobe.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e1be80aa11)
2016-08-05 23:00:58 +02:00
Carl Eugen Hoyos
c70b1ae930 lavc/vaapi_encode_h26x: Fix a crash if "." is not the decimal separator.
Fixes Debian bugs #831529, #831909, #832964.

Signed-off-by: Mark Thompson <sw@jkqxz.net>
(cherry picked from commit 82e53b3cef)
2016-08-05 23:00:01 +02:00
Timothy Gu
327033d913 jni: Return ENOSYS on unsupported platforms 2016-08-02 22:33:03 -07:00
Carl Eugen Hoyos
9a345b235f lavu/hwcontext_vaapi: Fix compilation if VA_FOURCC_ABGR is not defined.
Fixes ticket #5484.
(cherry picked from commit 5aede05120)
2016-08-02 23:25:07 +02:00
Michael Niedermayer
8f6a95a103 avcodec/vp9_parser: Check the input frame sizes for being consistent
Suggested-by: BBB
Fixed-by: BBB
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 77b0f3f26d)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-01 17:29:14 +02:00
Xinzheng Zhang
b4922daead avformat/flvdec: parse keyframe before a\v stream was created add_keyframes_index() when stream created or keyframe parsed
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit ad14aab3b4)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-01 17:29:14 +02:00
Xinzheng Zhang
88e3e6b943 avformat/flvdec: splitting add_keyframes_index() out from parse_keyframes_index()
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit cd141e71bd)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-01 17:29:14 +02:00
Kacper Michajłow
87d5146fb7 libavformat/rtpdec_asf: zero initialize the AVIOContext struct
This fixes crash in avformat_open_input() when accessing
protocol_whitelist field.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e947b75b1c)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-01 17:29:14 +02:00
Kacper Michajłow
caf32880fd libavutil/opt: Small bugfix in example.
Fix const corectness and zero init the struct. This example code would actually crash when initializing string.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 69630f4d30)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-01 17:29:14 +02:00
Sasi Inguva
7c01fa962e libx264: Increase x264 opts character limit to 4096
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 282477bf45)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-01 17:29:14 +02:00
Michael Niedermayer
e4eab67a0a avcodec/h264_parser: Set sps/pps_ref
Fixes use of freed memory
Should fix valgrind failures of fate-h264-skip-nointra

Found-by: logan
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit febc862b53)

Conflicts:

	libavcodec/h264_parser.c
2016-08-01 17:29:14 +02:00
Luca Barbato
86f9228740 librtmp: Avoid an infiniloop setting connection arguments
The exit condition was missing.

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
(cherry picked from commit e85d38c20a)
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2016-07-26 12:07:40 -07:00
James Almer
7cab4142c5 avformat/oggparsevp8: fix pts calculation on pages ending with an invisible frame
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 5adfbd3918)
2016-07-25 10:01:11 -03:00
Carl Eugen Hoyos
2e1be22715 lavc/Makefile: Fix standalone compilation of the svq3 decoder.
Regression since 0bf5fd2e
(cherry picked from commit 71167f7f84)
2016-07-24 23:56:39 +02:00
Clément Bœsch
7da59005be lavf/vplayerdec: Improve auto-detection.
Fixes the incorrect detection of 16_selma_OneFrame_QP39.yuv (gray16le
rawvideo) as vplayer format.
(cherry picked from commit 77726d32a8)
2016-07-15 10:36:59 +02:00
Matthieu Bouron
1410732621 lavc/mediacodecdec_h264: properly convert extradata to annex-b
H264ParamSets has its SPS/PPS stored raw (SODB) and needs to be
converted to NAL units before sending them to MediaCodec.

This patch adds the missing convertion of the SPS/PPS from SOBP to RBSP
which makes the resulting NAL units correct.

Fixes codec initialization on Nexus 4 and Nexus 7.

(cherry picked from commit 88d9c30cf5)
2016-07-11 15:32:30 +02:00
James Almer
f9a150fc31 Revert "configure: Enable GCC vectorization on ≥4.9 on x86"
This reverts commit cb8646af24.

This change has brough more issues than benefits, between compilation
time failures depending on flags used and code miscompilation causing
runtime crashes.

See the "[PATCH 2/2] configure: Enable GCC vectorization on ≥4.9"
thread in the ffmpeg-devel mailing list for the relevant discussion.

(cherry picked from commit fd6dbc5385)
2016-07-09 17:38:48 -03:00
Michael Niedermayer
ce36e74e75 doc/APIchanges: fill in missing git hash
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2a8dadb38f)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-01 02:43:01 +02:00
Michael Niedermayer
fc25481d17 Update for 3.1.1
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-01 02:13:51 +02:00
Michael Niedermayer
5c695ce903 doc/APIchanges: document the lavu/lavf field moves
Based-on: patch by James Almer
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 86fec7a7e8)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-01 02:12:28 +02:00
Michael Niedermayer
f617b94c23 avformat/avformat: Move new field to the end of AVStream
This fixes part of Ticket5676
This fixes kodi, mpv, chromium and ffplay build against 3.0 and linked to 3.1

This is a similar ABI fix to 1eb43af1a0

Approved-by: BBB
Approved-by: jamrial
Approved-by: BtbN
Approved-by: nevcairiel
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c1c7e0abb0)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-30 17:58:11 +02:00
Hendrik Leppkes
79af094b93 avformat/utils: update deprecated AVStream->codec when the context is updated
This ensures the AVStream->codec entry is kept in sync when new streams are
discovered mid-playback or changes to the context occur from other sources.

Fixes trac 5678.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c2e13d2ecd)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-30 17:58:11 +02:00
Michael Niedermayer
7747300289 avutil/frame: Move new field to the end of AVFrame
This fixes part of Ticket5676
This fixes kodi, mpv, chromium and ffplay build against 3.0 and linked to 3.1

This is a similar ABI fix to 1eb43af1a0

Approved-by: BBB
Approved-by: jamrial
Approved-by: BtbN
Approved-by: nevcairiel
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 042fb69deb)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-30 17:58:10 +02:00
Martin Vignali
37c83b5373 libavcodec/exr : fix decoding piz float file.
fix ticket #5674

the size of data to process in piz_uncompress, is now calc
using the pixel type of each channel.

the data reorganization, alos take care about the size of
each channel

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit d9e1e08133)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-30 17:57:59 +02:00
Michael Niedermayer
3e730278f5 avformat/mov: Check sample size
Fixes integer overflow
Fixes: poc.mp4

Found-by: ajax secure <ajax4sec@hotmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 8a3221cc67)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-29 20:00:49 +02:00
Timo Rothenpieler
1fdf549462 lavfi: Move new field to the end of AVFilterContext
This fixes an accidental ABI break introduced at 8688d3a.
2016-06-29 18:24:06 +02:00
Timo Rothenpieler
0a6d760230 lavfi: Move new field to the end of AVFilterLink
Even though this is not part of the public API, some external
applications access fields after it, thus breaking after updating from
ffmpeg 3.0 or earlier.
Since it is not public, it can be freely moved to the end to avoid
that problem in the future.

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-29 12:59:21 +02:00
Timo Rothenpieler
cd427a9d07 ffplay: Fix usage of private lavfi API
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-29 12:59:19 +02:00
Matthieu Bouron
8fd5669077 lavc/mediacodecdec_h264: add missing NAL headers to SPS/PPS buffers
Fixes a regression introduced by 0cd5e281df.

(cherry picked from commit db0af7250a)
2016-06-29 11:00:42 +02:00
Clément Bœsch
25f0ea9ece lavc/pnm_parser: disable parsing for text based PNMs
P1, P2, and P3 are respectively the text versions of PBM, PGM and PPM
files.

We can not obtain the buffer size using av_imgage_get_buffer_size() as
every pixel in the picture will occupy a random size between 16 and 32
bits ("4 " and "231 " are such example).

Ideally, we could look for the next header (or EOF) in the bytestream,
but this commit is meant to fix a decoding regression introduced by
48ac4532d4.

Fix Ticket #5670

(cherry picked from commit c5566f0a94)
2016-06-29 11:00:34 +02:00
Rick Kern
36fcb8cc55 Changelog: Add VideoToolbox encoder entry for 3.1
Signed-off-by: Rick Kern <kernrj@gmail.com>
(cherry picked from commit d956171813)
2016-06-27 11:45:11 -04:00
Rick Kern
18ce5a4d1b configure: use c++98 for c++ files
Use c++98 standard instead of c++11.

Signed-off-by: Rick Kern <kernrj@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 729d82abae)

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-26 23:27:22 +02:00
James Almer
cf09348b9e changelog: fix entry order
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit c6f2d1a21f)
2016-06-26 15:28:16 -03:00
James Almer
970f2ad966 Update FFmpeg 3.1 cut marker
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 069fd69662)
2016-06-26 15:17:48 -03:00
James Almer
104c357b6a Merge branch 'master' into release/3.1
Merged-by: James Almer <jamrial@gmail.com>
2016-06-26 15:14:17 -03:00
Michael Niedermayer
b2a74dd629 Set version to 3.1
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-26 11:35:22 +02:00
Michael Niedermayer
182cfe4832 release notes (based on release/3.0)
Better release notes are welcome
write better ones or do not complain later!

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-26 03:57:55 +02:00
Michael Niedermayer
e5d434b840 tests/checkasm/checkasm: Disable checkasm_check_pixblockdsp for ppc64be
See: Ticket5508

Suggested-by: Carl
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-26 03:56:11 +02:00
4485 changed files with 150205 additions and 487911 deletions

1
.gitattributes vendored
View File

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

8
.gitignore vendored
View File

@@ -18,9 +18,6 @@
*.so.*
*.swp
*.ver
*.version
*.ptx
*.ptx.c
*_g
\#*
.\#*
@@ -29,11 +26,10 @@
/ffmpeg
/ffplay
/ffprobe
/config.asm
/config.h
/ffserver
/config.*
/coverage.info
/avversion.h
/lcov/
/src
/mapfile
/tools/python/__pycache__/

View File

@@ -1,21 +0,0 @@
<james.darnley@gmail.com> <jdarnley@obe.tv>
<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.fu@intel.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>
<zhong.li@intel.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>

View File

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

View File

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

430
Changelog
View File

@@ -1,348 +1,102 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
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 <next>:
version 3.1.4:
- avformat/avidec: Check nb_streams in read_gab2_sub()
- avformat/avidec: Remove ancient assert
- avfilter/vf_colorspace: fix range for output colorspace option
- lavc/mediacodecdec_h264: fix SODB escaping
- avcodec/nvenc: fix const options for hevc gpu setting
- avformat/avidec: Fix memleak with dv in avi
- lavc/movtextdec.c: Avoid infinite loop on invalid data.
- avcodec/ansi: Check dimensions
- avcodec/cavsdsp: use av_clip_uint8() for idct
- avformat/movenc: Check packet in mov_write_single_packet() too
- avformat/movenc: Factor check_pkt() out
- avformat/utils: fix timebase error in avformat_seek_file()
- avcodec/g726: Add missing ADDB output mask
- avcodec/avpacket: clear side_data_elems
- avformat/movenc: Check first DTS similar to dts difference
- avcodec/ccaption_dec: Use simple array instead of AVBuffer
- avcodec/svq3: Reintroduce slice_type
- avformat/mov: Fix potential integer overflow in mov_read_keys
- swscale/swscale_unscaled: Try to fix Rgb16ToPlanarRgb16Wrapper() with slices
- swscale/swscale_unscaled: Fix packed_16bpc_bswap() with slices
- avformat/avidec: Fix infinite loop in avi_read_nikon()
- lavf/utils: Avoid an overflow for huge negative durations.
- avformat/hls: Fix handling of EXT-X-BYTERANGE streams over 2GB
- lavc/avpacket: Fix undefined behaviour, do not pass a null pointer to memcpy().
- lavc/mjpegdec: Do not skip reading quantization tables.
- cmdutils: fix implicit declaration of SetDllDirectory function
version 4.2:
- tpad filter
- AV1 decoding support through libdav1d
- dedot filter
- chromashift and rgbashift filters
- freezedetect filter
- truehd_core bitstream filter
- dhav demuxer
- PCM-DVD encoder
- GIF parser
- vividas demuxer
- hymt decoder
- anlmdn filter
- maskfun filter
- hcom demuxer and decoder
- ARBC decoder
- libaribb24 based ARIB STD-B24 caption support (profiles A and C)
- Support decoding of HEVC 4:4:4 content in nvdec and cuviddec
- removed libndi-newtek
- agm decoder
- KUX demuxer
- AV1 frame split bitstream filter
- lscr decoder
- lagfun filter
- asoftclip filter
- Support decoding of HEVC 4:4:4 content in vdpau
- colorhold filter
- xmedian filter
- asr filter
- showspatial multimedia filter
- VP4 video decoder
- IFV demuxer
- derain filter
- deesser filter
- mov muxer writes tracks with unspecified language instead of English by default
- add support for using clang to compile CUDA kernels
version 3.1.3:
- examples/demuxing_decoding: convert to codecpar
- avcodec/exr: Check tile positions
- avcodec/aacenc: Tighter input checks
- avformat/wtvdec: Check pointer before use
- libavcodec/wmalosslessdec: Check the remaining bits
- avcodec/adpcm: Fix adpcm_ima_wav padding
- avcodec/svq3: fix slice size check
- avcodec/diracdec: Check numx/y
- avcodec/h2645_parse: fix nal size
- avcodec/h2645_parse: Use get_nalsize() in ff_h2645_packet_split()
- h2645_parse: only read avc length code at the correct position
- h2645_parse: don't overread AnnexB NALs within an avc stream
- avcodec/h264_parser: Factor get_avc_nalsize() out
- avcodec/cfhd: Increase minimum band dimension to 3
- avcodec/indeo2: check ctab
- avformat/swfdec: Fix inflate() error code check
- avcodec/rawdec: Fix bits_per_coded_sample checks
- vcodec/h2645_parse: Clear buffer padding
- avcodec/h2645: Fix NAL unit padding
- avfilter/drawutils: Fix single plane with alpha
- cmdutils: check for SetDllDirectory() availability
version 3.1.2:
- cmdutils: remove the current working directory from the DLL search path on win32
- avcodec/rawdec: Fix palette handling with changing palettes
- avcodec/raw: Fix decoding of ilacetest.mov
- avformat/mov: Enable mp3 parsing if a packet needs it
- avformat/hls: Use an array instead of stream offset for stream mapping
- avformat/hls: Sync starting segment across variants on live streams
- avformat/hls: Fix regression with ranged media segments
- avcodec/ffv1enc: Fix assertion failure with non zero bits per sample
- avfilter/af_hdcd: small fix in af_hdcd.c where gain was not being adjusted for "attenuate slowly"
- avformat/oggdec: Fix integer overflow with invalid pts
- ffplay: Fix invalid array index
- avcodec/alacenc: allocate bigger packets (cherry picked from commit 82b84c71b009884c8d041361027718b19922c76d)
- libavcodec/dnxhd: Enable 12-bit DNxHR support.
- lavc/vaapi_encode_h26x: Fix a crash if "." is not the decimal separator.
- jni: Return ENOSYS on unsupported platforms
- lavu/hwcontext_vaapi: Fix compilation if VA_FOURCC_ABGR is not defined.
- avcodec/vp9_parser: Check the input frame sizes for being consistent
- avformat/flvdec: parse keyframe before a\v stream was created add_keyframes_index() when stream created or keyframe parsed
- avformat/flvdec: splitting add_keyframes_index() out from parse_keyframes_index()
- libavformat/rtpdec_asf: zero initialize the AVIOContext struct
- libavutil/opt: Small bugfix in example.
- libx264: Increase x264 opts character limit to 4096
- avcodec/h264_parser: Set sps/pps_ref
- librtmp: Avoid an infiniloop setting connection arguments
- avformat/oggparsevp8: fix pts calculation on pages ending with an invisible frame
- lavc/Makefile: Fix standalone compilation of the svq3 decoder.
- lavf/vplayerdec: Improve auto-detection.
- lavc/mediacodecdec_h264: properly convert extradata to annex-b
- Revert "configure: Enable GCC vectorization on ≥4.9 on x86"
version 4.1:
- deblock filter
- tmix filter
- amplify filter
- fftdnoiz filter
- aderivative and aintegral audio filters
- pal75bars and pal100bars video filter sources
- support mbedTLS based TLS
- adeclick filter
- adeclip filter
- libtensorflow backend for DNN based filters like srcnn
- vc1 decoder is now bit-exact
- ATRAC9 decoder
- lensfun wrapper filter
- colorconstancy filter
- AVS2 video decoder via libdavs2
- IMM4 video decoder
- Brooktree ProSumer video decoder
- MatchWare Screen Capture Codec decoder
- WinCam Motion Video decoder
- 1D LUT filter (lut1d)
- RemotelyAnywhere Screen Capture decoder
- cue and acue filters
- support for AV1 in MP4
- transpose_npp filter
- AVS2 video encoder via libxavs2
- amultiply filter
- Block-Matching 3d (bm3d) denoising filter
- acrossover filter
- ilbc decoder
- audio denoiser as afftdn filter
- AV1 parser
- SER demuxer
- sinc audio filter source
- chromahold filter
- setparams filter
- vibrance filter
- decoding S12M timecode in h264
- xstack filter
- pcm vidc decoder and encoder
- (a)graphmonitor filter
- yadif_cuda filter
version 4.0:
- Bitstream filters for editing metadata in H.264, HEVC and MPEG-2 streams
- Dropped support for OpenJPEG versions 2.0 and below. Using OpenJPEG now
requires 2.1 (or later) and pkg-config.
- VDA dropped (use VideoToolbox instead)
- MagicYUV encoder
- Raw AMR-NB and AMR-WB demuxers
- TiVo ty/ty+ demuxer
- Intel QSV-accelerated MJPEG encoding
- PCE support for extended channel layouts in the AAC encoder
- native aptX and aptX HD encoder and decoder
- Raw aptX and aptX HD muxer and demuxer
- NVIDIA NVDEC-accelerated H.264, HEVC, MJPEG, MPEG-1/2/4, VC1, VP8/9 hwaccel decoding
- Intel QSV-accelerated overlay filter
- mcompand audio filter
- acontrast audio filter
- OpenCL overlay filter
- video mix filter
- video normalize filter
- audio lv2 wrapper filter
- VAAPI MJPEG and VP8 decoding
- AMD AMF H.264 and HEVC encoders
- video fillborders filter
- video setrange filter
- nsp demuxer
- support LibreSSL (via libtls)
- AVX-512/ZMM support added
- Dropped support for building for Windows XP. The minimum supported Windows
version is Windows Vista.
- deconvolve video filter
- entropy video filter
- hilbert audio filter source
- aiir audio filter
- aiff: add support for CD-ROM XA ADPCM
- Removed the ffserver program
- Removed the ffmenc and ffmdec muxer and demuxer
- VideoToolbox HEVC encoder and hwaccel
- VAAPI-accelerated ProcAmp (color balance), denoise and sharpness filters
- Add android_camera indev
- codec2 en/decoding via libcodec2
- muxer/demuxer for raw codec2 files and .c2 files
- Moved nvidia codec headers into an external repository.
They can be found at http://git.videolan.org/?p=ffmpeg/nv-codec-headers.git
- native SBC encoder and decoder
- drmeter audio filter
- hapqa_extract bitstream filter
- filter_units bitstream filter
- AV1 Support through libaom
- E-AC-3 dependent frames support
- bitstream filter for extracting E-AC-3 core
- Haivision SRT protocol via libsrt
- segafilm muxer
- vfrdet filter
- SRCNN filter
version 3.4:
- deflicker video filter
- doubleweave video filter
- lumakey video filter
- pixscope video filter
- oscilloscope video filter
- config.log and other configuration files moved into ffbuild/ directory
- update cuvid/nvenc headers to Video Codec SDK 8.0.14
- afir audio filter
- scale_cuda CUDA based video scale filter
- librsvg support for svg rasterization
- crossfeed audio filter
- spec compliant VP9 muxing support in MP4
- remove the libnut muxer/demuxer wrappers
- remove the libschroedinger encoder/decoder wrappers
- surround audio filter
- sofalizer filter switched to libmysofa
- Gremlin Digital Video demuxer and decoder
- headphone audio filter
- superequalizer audio filter
- roberts video filter
- The x86 assembler default switched from yasm to nasm, pass
--x86asmexe=yasm to configure to restore the old behavior.
- additional frame format support for Interplay MVE movies
- support for decoding through D3D11VA in ffmpeg
- limiter video filter
- libvmaf video filter
- Dolby E decoder and SMPTE 337M demuxer
- unpremultiply video filter
- tlut2 video filter
- floodfill video filter
- pseudocolor video filter
- raw G.726 muxer and demuxer, left- and right-justified
- NewTek NDI input/output device
- Some video filters with several inputs now use a common set of options:
blend, libvmaf, lut3d, overlay, psnr, ssim.
They must always be used by name.
- FITS demuxer and decoder
- FITS muxer and encoder
- add --disable-autodetect build switch
- drop deprecated qtkit input device (use avfoundation instead)
- despill video filter
- haas audio filter
- SUP/PGS subtitle muxer
- convolve video filter
- VP9 tile threading support
- KMS screen grabber
- CUDA thumbnail filter
- V4L2 mem2mem HW assisted codecs
- Rockchip MPP hardware decoding
- vmafmotion video filter
- use MIME type "G726" for little-endian G.726, "AAL2-G726" for big-endian G.726
version 3.3:
- CrystalHD decoder moved to new decode API
- add internal ebur128 library, remove external libebur128 dependency
- Pro-MPEG CoP #3-R2 FEC protocol
- premultiply video filter
- Support for spherical videos
- configure now fails if autodetect-libraries are requested but not found
- PSD Decoder
- 16.8 floating point pcm decoder
- 24.0 floating point pcm decoder
- Apple Pixlet decoder
- QDMC audio decoder
- NewTek SpeedHQ decoder
- MIDI Sample Dump Standard demuxer
- readeia608 filter
- Sample Dump eXchange demuxer
- abitscope multimedia filter
- Scenarist Closed Captions demuxer and muxer
- threshold filter
- midequalizer filter
- Optimal Huffman tables for (M)JPEG encoding
- VAAPI-accelerated MPEG-2 and VP8 encoding
- FM Screen Capture Codec decoder
- native Opus encoder
- ScreenPressor decoder
- incomplete ClearVideo decoder
- Intel QSV video scaling and deinterlacing filters
- Support MOV with multiple sample description tables
- XPM decoder
- Removed the legacy X11 screen grabber, use XCB instead
- MPEG-7 Video Signature filter
- Removed asyncts filter (use af_aresample instead)
- Intel QSV-accelerated VP8 video decoding
- VAAPI-accelerated deinterlacing
version 3.2:
- libopenmpt demuxer
- tee protocol
- Changed metadata print option to accept general urls
- Alias muxer for Ogg Video (.ogv)
- VP8 in Ogg muxing
- curves filter doesn't automatically insert points at x=0 and x=1 anymore
- 16-bit support in curves filter and selectivecolor filter
- OpenH264 decoder wrapper
- MediaCodec H.264/HEVC/MPEG-4/VP8/VP9 hwaccel
- True Audio (TTA) muxer
- crystalizer audio filter
- acrusher audio filter
- bitplanenoise video filter
- floating point support in als decoder
- fifo muxer
- maskedclamp filter
- hysteresis filter
- lut2 filter
- yuvtestsrc filter
- CUDA CUVID H.263/VP8/VP9/10 bit HEVC (Dithered) Decoding
- vaguedenoiser filter
- added threads option per filter instance
- weave filter
- gblur filter
- avgblur filter
- sobel and prewitt filter
- MediaCodec HEVC/MPEG-4/VP8/VP9 decoding
- Meridian Lossless Packing (MLP) / TrueHD encoder
- Non-Local Means (nlmeans) denoising filter
- sdl2 output device and ffplay support
- sdl1 output device and sdl1 support removed
- extended mov edit list support
- libfaac encoder removed
- Matroska muxer now writes CRC32 elements by default in all Level 1 elements
- sidedata video and asidedata audio filter
- Changed mapping of rtp MIME type G726 to codec g726le.
- spec compliant VAAPI/DXVA2 VC-1 decoding of slices in frame-coded images
version 3.1.1:
- doc/APIchanges: document the lavu/lavf field moves
- avformat/avformat: Move new field to the end of AVStream
- avformat/utils: update deprecated AVStream->codec when the context is updated
- avutil/frame: Move new field to the end of AVFrame
- libavcodec/exr : fix decoding piz float file.
- avformat/mov: Check sample size
- lavfi: Move new field to the end of AVFilterContext
- lavfi: Move new field to the end of AVFilterLink
- ffplay: Fix usage of private lavfi API
- lavc/mediacodecdec_h264: add missing NAL headers to SPS/PPS buffers
- lavc/pnm_parser: disable parsing for text based PNMs
version 3.1:

View File

@@ -1,4 +1,4 @@
## Installing FFmpeg
#Installing FFmpeg:
1. Type `./configure` to create the configuration. A list of configure
options is printed by running `configure --help`.

View File

@@ -17,15 +17,16 @@ Specifically, the GPL parts of FFmpeg are:
- `libavcodec/x86/flac_dsp_gpl.asm`
- `libavcodec/x86/idct_mmx.c`
- `libavfilter/x86/vf_removegrain.asm`
- the X11 grabber in `libavdevice/x11grab.c`
- the following building and testing tools
- `compat/solaris/make_sunver.pl`
- `doc/t2h.pm`
- `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`
- `f_ebur128.c`
- `vf_blackframe.c`
- `vf_boxblur.c`
- `vf_colormatrix.c`
@@ -35,13 +36,13 @@ 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 +51,12 @@ 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
@@ -83,47 +82,43 @@ affect the licensing of binaries resulting from the combination.
### Compatible libraries
The following libraries are under GPL version 2:
- avisynth
The following libraries are under GPL:
- 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:
- gmp
- 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
The 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
compatible with the GPL and/or the LGPL. If you wish to enable these
libraries, even in circumstances that their license may be incompatible, pass
`--enable-nonfree` to configure. This will cause the resulting binary to be
`--enable-nonfree` to configure. But note that if you enable any of these
libraries the resulting binary will be under a complex license mix that is
more restrictive than the LGPL and that may result in additional obligations.
It is possible that these restrictions cause the resulting binary to be
unredistributable.
The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
incompatible with the GPLv2 and v3. To the best of our knowledge, they are
compatible with the LGPL.
The FAAC library is incompatible with all versions of GPL and LGPL.
The NVENC library, while its header file is licensed under the compatible MIT
license, requires a proprietary binary blob at run time, and is deemed to be
incompatible with the GPL. We are not certain if it is compatible with the
LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
it is not.

View File

@@ -29,6 +29,9 @@ ffplay:
ffprobe:
ffprobe.c Stefano Sabatini
ffserver:
ffserver.c Reynaldo H. Verdejo Pinochet
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
@@ -39,12 +42,11 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
API tests Ludmila Glinskih
Communication
@@ -52,12 +54,11 @@ 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
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists Baptiste Coudurier, Lou Logan
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan, Reynaldo H. Verdejo Pinochet
Launchpad Timothy Gu
ffmpeg-security Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
libavutil
@@ -77,8 +78,6 @@ Other:
eval.c, eval.h Michael Niedermayer
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
@@ -89,7 +88,6 @@ Other:
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
tx* Lynne
libavcodec
@@ -115,14 +113,13 @@ 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:
motion* Michael Niedermayer
rate control:
ratecontrol.c Michael Niedermayer
libxvid_rc.c Michael Niedermayer
simple IDCT:
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
@@ -139,13 +136,10 @@ Codecs:
8svx.c Jaikrishnan Menon
aacenc*, aaccoder.c Rostislav Pehlivanov
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann, Umair Khan
aptx.c Aurelien Jacobs
alsdec.c Thilo Borgmann
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3plus* Maxim Poliakovski
audiotoolbox* Rodger Combs
avs2* Huiwen Ren
bgmc.c, bgmc.h Thilo Borgmann
binkaudio.c Peter Ross
cavs* Stefan Gehrer
@@ -153,16 +147,14 @@ Codecs:
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cinepakenc.c Rl / Aetey G.T. AB
ccaption_dec.c Anshul Maheshwari, Aman Gupta
ccaption_dec.c Anshul Maheshwari
cljr Alex Beregszaszi
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
cuviddec.c Timo Rothenpieler
dca* foo86
cuvid.c Timo Rothenpieler
dirac* Rostislav Pehlivanov
dnxhd* Baptiste Coudurier
dolby_e* foo86
dpcm.c Mike Melanson
dss_sp.c Oleksij Rempel
dv.c Roman Shaposhnik
@@ -172,7 +164,6 @@ Codecs:
exif.c, exif.h Thilo Borgmann
ffv1* Michael Niedermayer
ffwavesynth.c Nicolas George
fifo.c Jan Sebechlebsky
flicvideo.c Mike Melanson
g722.c Martin Storsjo
g726.c Roman Shaposhnik
@@ -181,7 +172,7 @@ Codecs:
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
hap* Tom Butterworth
huffyuv* Michael Niedermayer
huffyuv* Michael Niedermayer, Christophe Gisquet
idcinvideo.c Mike Melanson
interplayvideo.c Mike Melanson
jni*, ffjni* Matthieu Bouron
@@ -189,25 +180,22 @@ Codecs:
jvdec.c Peter Ross
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libcodec2.c Tomas Härdin
libdirac* David Conrad
libdavs2.c Huiwen Ren
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenh264enc.c Martin Storsjo, Linjie Fu
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libschroedinger* David Conrad
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
mimic.c Ramiro Polla
mjpeg*.c Michael Niedermayer
mlp* Ramiro Polla, Jai Luthra
mlp* Ramiro Polla
mmvideo.c Peter Ross
mpeg12.c, mpeg12data.h Michael Niedermayer
mpegvideo.c, mpegvideo.h Michael Niedermayer
@@ -216,16 +204,14 @@ Codecs:
msrle.c Mike Melanson
msvideo1.c Mike Melanson
nuv.c Reimar Doeffinger
nvdec*, nvenc* Timo Rothenpieler
omx.c Martin Storsjo, Aman Gupta
opus* Rostislav Pehlivanov
nvenc* Timo Rothenpieler
paf.* Paul B Mahol
pcx.c Ivo van Poorten
pgssubdec.c Reimar Doeffinger
ptx.c Ivo van Poorten
qcelp* Reynaldo H. Verdejo Pinochet
qdm2.c, qdm2data.h Roberto Togni
qsv* Mark Thompson, Zhong Li
qsv* Ivan Uskov
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
@@ -233,12 +219,12 @@ Codecs:
rpza.c Roberto Togni
rtjpeg.c, rtjpeg.h Reimar Doeffinger
rv10.c Michael Niedermayer
rv4* Christophe Gisquet
s3tc* Ivo van Poorten
smc.c Mike Melanson
smvjpegdec.c Ash Hughes
snow* Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
speedhq.c Steinar H. Gunderson
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
@@ -247,10 +233,11 @@ Codecs:
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
v4l2_* Jorge Ramirez-Ortiz
vc1* Christophe Gisquet
vc2* Rostislav Pehlivanov
vcr1.c Michael Niedermayer
videotoolboxenc.c Rick Kern, Aman Gupta
vda_h264_dec.c Xidorn Quan
videotoolboxenc.c Rick Kern
vima.c Paul B Mahol
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
@@ -271,13 +258,12 @@ Codecs:
Hardware acceleration:
crystalhd.c Philip Langdale
dxva2* Hendrik Leppkes, Laurent Aimar, Steve Lhomme
d3d11va* Steve Lhomme
mediacodec* Matthieu Bouron, Aman Gupta
dxva2* Hendrik Leppkes, Laurent Aimar
mediacodec* Matthieu Bouron
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern, Aman Gupta
videotoolbox* Rick Kern
libavdevice
@@ -287,8 +273,7 @@ libavdevice
avfoundation.m Thilo Borgmann
android_camera.c Felix Matouschek
decklink* Marton Balint
decklink* Deti Fliegl
dshow.c Roger Pack (CC rogerdpack@gmail.com)
fbdev_enc.c Lukasz Marek
gdigrab.c Roger Pack (CC rogerdpack@gmail.com)
@@ -297,8 +282,8 @@ libavdevice
libdc1394.c Roman Shaposhnik
opengl_enc.c Lukasz Marek
pulse_audio_enc.c Lukasz Marek
qtkit.m Thilo Borgmann
sdl Stefano Sabatini
sdl2.c Josh de Kock
v4l2.c Giorgio Vazzana
vfwcap.c Ramiro Polla
xv.c Lukasz Marek
@@ -309,8 +294,6 @@ libavfilter
Generic parts:
graphdump.c Nicolas George
motion_estimation.c Davinder Singh
Filters:
f_drawgraph.c Paul B Mahol
af_adelay.c Paul B Mahol
@@ -325,7 +308,6 @@ Filters:
af_chorus.c Paul B Mahol
af_compand.c Paul B Mahol
af_firequalizer.c Muhammad Faiz
af_hdcd.c Burt P.
af_ladspa.c Paul B Mahol
af_loudnorm.c Kyle Swanson
af_pan.c Nicolas George
@@ -335,10 +317,8 @@ Filters:
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_bwdif Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_chromakey.c Timo Rothenpieler
vf_colorchannelmixer.c Paul B Mahol
vf_colorconstancy.c Mina Sami (CC <minas.gorgy@gmail.com>)
vf_colorbalance.c Paul B Mahol
vf_colorkey.c Timo Rothenpieler
vf_colorlevels.c Paul B Mahol
@@ -352,11 +332,8 @@ Filters:
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_(t)interlace Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_lenscorrection.c Daniel Oberhoff
vf_mergeplanes.c Paul B Mahol
vf_mestimate.c Davinder Singh
vf_minterpolate.c Davinder Singh
vf_neighbor.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_random.c Paul B Mahol
@@ -366,15 +343,12 @@ Filters:
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_tonemap_opencl.c Ruiling Song
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c Michael Niedermayer
dnn Yejun Guo
libavformat
===========
@@ -398,31 +372,26 @@ Muxers/Demuxers:
astdec.c Paul B Mahol
astenc.c James Almer
avi* Michael Niedermayer
avisynth.c Stephen Hutchinson
avisynth.c AvxSynth Team (avxsynth.testing at gmail dot com)
avr.c Paul B Mahol
bink.c Peter Ross
boadec.c Michael Niedermayer
brstm.c Paul B Mahol
caf* Peter Ross
cdxl.c Paul B Mahol
codec2.c Tomas Härdin
crc.c Michael Niedermayer
dashdec.c Steven Liu
dashenc.c Karthick Jeyapal
daud.c Reimar Doeffinger
dss.c Oleksij Rempel
dtsdec.c foo86
dtshddec.c Paul B Mahol
dv.c Roman Shaposhnik
electronicarts.c Peter Ross
epafdec.c Paul B Mahol
ffm* Baptiste Coudurier
flic.c Mike Melanson
flvdec.c Michael Niedermayer
flvenc.c Michael Niedermayer, Steven Liu
flvdec.c, flvenc.c Michael Niedermayer
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hlsenc.c Christian Suloway, Steven Liu
hls.c Anssi Hannula
hls encryption (hlsenc.c) Christian Suloway
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
@@ -432,13 +401,13 @@ Muxers/Demuxers:
iss.c Stefan Gehrer
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
libopenmpt.c Josh de Kock
libnut.c Oded Shimon
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs, Andreas Rheinhardt
matroskadec.c Aurelien Jacobs, Andreas Rheinhardt
matroskaenc.c David Conrad, Andreas Rheinhardt
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
matroska subtitles (matroskaenc.c) John Peebles
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
@@ -453,7 +422,8 @@ 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
mxfdec.c Tomas Härdin
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
@@ -461,6 +431,7 @@ 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
psxstr.c Mike Melanson
@@ -481,7 +452,6 @@ Muxers/Demuxers:
rtpdec_vc2hq.*, rtpenc_vc2hq.* Thomas Volkert
rtpdec_vp9.c Thomas Volkert
rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo
s337m.c foo86
sbgdec.c Nicolas George
sdp.c Martin Storsjo
segafilm.c Mike Melanson
@@ -508,7 +478,6 @@ Protocols:
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libssh.c Lukasz Marek
libzmq.c Andriy Gelman
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c Marvin Scholz
@@ -532,10 +501,9 @@ Operating systems / CPU architectures
=====================================
Alpha Falk Hueffner
MIPS Manojkumar Bhosale, Shiyou Yin
MIPS Nedeljko Babic
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Lauri Kasanen
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver, Hendrik Leppkes
@@ -545,35 +513,6 @@ Sparc Roman Shaposhnik
OS/2 KO Myung-Hun
Developers with git write access who are currently not maintaining any specific part
====================================================================================
Alex Converse
Andreas Cadhalpun
Anuradha Suraparaju
Ben Littler
Benjamin Larsson
Bobby Bingham
Daniel Verkamp
Derek Buitenhuis
Ganesh Ajjanagadde
Henrik Gramner
Ivan Uskov
James Darnley
Jan Ekström
Joakim Plate
Jun Zhao
Kieran Kunhya
Kirill Gavrilov
Limin Wang
Martin Storsjö
Panagiotis Issaris
Pedro Arthur
Sebastien Zwickert
Vittorio Giovara
wm4
(this list is incomplete)
Releases
========
@@ -581,15 +520,13 @@ Releases
2.7 Michael Niedermayer
2.6 Michael Niedermayer
2.5 Michael Niedermayer
2.4 Michael Niedermayer
If you want to maintain an older release, please contact us
GnuPG Fingerprints and IRC nicknames of maintainers and contributors
====================================================================
IRC nicknames are in parentheses. These apply
to the IRC channels listed on the website.
GnuPG Fingerprints of maintainers and contributors
==================================================
Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
@@ -604,28 +541,22 @@ FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Ganesh Ajjanagadde C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
James Almer 7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan (llogan) 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Lynne FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Nikolay Aleksandrov 8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
Philip Langdale 5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
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
Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Steinar H. Gunderson C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
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
Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

151
Makefile
View File

@@ -1,5 +1,5 @@
MAIN_MAKEFILE=1
include ffbuild/config.mak
include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
@@ -11,12 +11,40 @@ vpath %.asm $(SRC_PATH)
vpath %.rc $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %.cu $(SRC_PATH)
vpath %.ptx $(SRC_PATH)
vpath %/fate_config.sh.template $(SRC_PATH)
AVPROGS-$(CONFIG_FFMPEG) += ffmpeg
AVPROGS-$(CONFIG_FFPLAY) += ffplay
AVPROGS-$(CONFIG_FFPROBE) += ffprobe
AVPROGS-$(CONFIG_FFSERVER) += ffserver
AVPROGS := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
INSTPROGS = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
PROGS += $(AVPROGS)
AVBASENAMES = ffmpeg ffplay ffprobe ffserver
ALLAVPROGS = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
OBJS-ffmpeg += ffmpeg_opt.o ffmpeg_filter.o
OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
OBJS-ffmpeg-$(CONFIG_LIBMFX) += ffmpeg_qsv.o
OBJS-ffmpeg-$(CONFIG_VAAPI) += ffmpeg_vaapi.o
ifndef CONFIG_VIDEOTOOLBOX
OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_videotoolbox.o
endif
OBJS-ffmpeg-$(CONFIG_CUVID) += ffmpeg_cuvid.o
OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
OBJS-ffserver += ffserver_config.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
TOOLS = qt-faststart trasher uncoded_frame
TOOLS-$(CONFIG_ZLIB) += cws2fws
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
@@ -31,52 +59,36 @@ FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README
SKIPHEADERS = compat/w32pthreads.h
SKIPHEADERS = cmdutils_common_opts.h \
compat/w32pthreads.h
# first so "all" becomes default target
all: all-yes
include $(SRC_PATH)/tools/Makefile
include $(SRC_PATH)/ffbuild/common.mak
include $(SRC_PATH)/common.mak
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
all: $(AVPROGS)
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_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) $(LIBFUZZER_PATH)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
CONFIGURABLE_COMPONENTS = \
$(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c)) \
$(SRC_PATH)/libavcodec/bitstream_filters.c \
$(SRC_PATH)/libavcodec/parsers.c \
$(SRC_PATH)/libavformat/protocols.c \
config.h: ffbuild/.config
ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
config.h: .config
.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@-tput bold 2>/dev/null
@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS \
SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS \
HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS \
ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS \
ALTIVEC-OBJS MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS \
MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
@@ -91,32 +103,41 @@ SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
include $(SRC_PATH)/ffbuild/library.mak
include $(SRC_PATH)/library.mak
endef
$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
include $(SRC_PATH)/fftools/Makefile
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/doc/examples/Makefile
libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
define DOPROG
OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
$$(OBJS-$(1)): CFLAGS += $(CFLAGS-$(1))
$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
-include $$(OBJS-$(1):.o=.d)
endef
$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
ifeq ($(STRIPTYPE),direct)
$(STRIP) -o $@ $<
else
$(CP) $< $@
$(STRIP) $@
endif
%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
VERSION_SH = $(SRC_PATH)/ffbuild/version.sh
OBJDIRS += tools
-include $(wildcard tools/*.d)
VERSION_SH = $(SRC_PATH)/version.sh
GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
.version: M=@
libavutil/ffversion.h .version:
@@ -126,33 +147,44 @@ libavutil/ffversion.h .version:
# force version.sh to run whenever version might have changed
-include .version
ifdef AVPROGS
install: install-progs install-data
endif
install: install-libs install-headers
install-libs: install-libs-yes
install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
install-progs-yes:
install-progs-$(CONFIG_SHARED): install-libs
uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
install-progs: install-progs-yes $(AVPROGS)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
install-data: $(DATA_FILES) $(EXAMPLES_FILES)
$(Q)mkdir -p "$(DATADIR)/examples"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES))
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
distclean:: clean
$(RM) .version avversion.h config.asm config.h mapfile \
ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
version.h libavutil/ffversion.h libavcodec/codec_names.h \
libavcodec/bsf_list.c libavformat/protocol_list.c \
libavcodec/codec_list.c libavcodec/parser_list.c \
libavfilter/filter_list.c libavdevice/indev_list.c libavdevice/outdev_list.c \
libavformat/muxer_list.c libavformat/demuxer_list.c
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .config libavutil/avconfig.h .version mapfile avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h libavcodec/bsf_list.c libavformat/protocol_list.c
ifeq ($(SRC_LINK),src)
$(RM) src
endif
@@ -161,12 +193,11 @@ endif
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
build: all alltools examples testprogs
check: all alltools examples testprogs fate
include $(SRC_PATH)/tests/Makefile
$(sort $(OUTDIRS)):
$(sort $(OBJDIRS)):
$(Q)mkdir -p $@
# Dummy rule to stop make trying to rebuild removed or renamed headers
@@ -177,5 +208,5 @@ $(sort $(OUTDIRS)):
# so this saves some time on slow systems.
.SUFFIXES:
.PHONY: all all-yes alltools build check config testprogs
.PHONY: *clean install* uninstall*
.PHONY: all all-yes alltools check *clean config install*
.PHONY: testprogs uninstall*

View File

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

View File

@@ -1 +1 @@
4.3
3.1.4

View File

@@ -1,13 +1,13 @@
┌────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 4.3 "4:3" │
└────────────────────────────────────┘
┌────────────────────────────────────────
│ RELEASE NOTES for FFmpeg 3.1 "Laplace" │
└────────────────────────────────────────
The FFmpeg Project proudly presents FFmpeg 4.3 "4:3", about 10
months after the release of FFmpeg 4.2.
The FFmpeg Project proudly presents FFmpeg 3.1 "Laplace", about 4
months after the release of FFmpeg 3.0.
A complete Changelog is available at the root of the project, and the
complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
complete Git history on http://source.ffmpeg.org.
We hope you will like this release as much as we enjoyed working on it, and
as usual, if you have any questions about it, or any FFmpeg related topic,

View File

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

View File

@@ -38,7 +38,6 @@
#include "libswscale/swscale.h"
#include "libswresample/swresample.h"
#include "libpostproc/postprocess.h"
#include "libavutil/attributes.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
@@ -55,11 +54,14 @@
#include "libavutil/ffversion.h"
#include "libavutil/version.h"
#include "cmdutils.h"
#if CONFIG_NETWORK
#include "libavformat/network.h"
#endif
#if HAVE_SYS_RESOURCE_H
#include <sys/time.h>
#include <sys/resource.h>
#endif
#ifdef _WIN32
#if HAVE_SETDLLDIRECTORY
#include <windows.h>
#endif
@@ -73,12 +75,6 @@ static FILE *report_file;
static int report_file_level = AV_LOG_DEBUG;
int hide_banner = 0;
enum show_muxdemuxers {
SHOW_DEFAULT,
SHOW_DEMUXERS,
SHOW_MUXERS,
};
void init_opts(void)
{
av_dict_set(&sws_dict, "flags", "bicubic", 0);
@@ -116,7 +112,7 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
void init_dynload(void)
{
#if HAVE_SETDLLDIRECTORY && defined(_WIN32)
#if HAVE_SETDLLDIRECTORY
/* Calling SetDllDirectory with the empty string (but not NULL) removes the
* current working directory from the DLL search path as a security pre-caution. */
SetDllDirectory("");
@@ -179,7 +175,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
first = 1;
for (po = options; po->name; po++) {
char buf[128];
char buf[64];
if (((po->flags & req_flags) != req_flags) ||
(alt_flags && !(po->flags & alt_flags)) ||
@@ -229,6 +225,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
* by default. HAVE_COMMANDLINETOARGVW is true on cygwin, while
* it doesn't provide the actual command line via GetCommandLineW(). */
#if HAVE_COMMANDLINETOARGVW && defined(_WIN32)
#include <windows.h>
#include <shellapi.h>
/* Will be leaked on exit */
static char** win32_argv_utf8 = NULL;
@@ -845,8 +842,8 @@ do { \
}
if (octx->cur_group.nb_opts || codec_opts || format_opts || resample_opts)
av_log(NULL, AV_LOG_WARNING, "Trailing option(s) found in the "
"command: may be ignored.\n");
av_log(NULL, AV_LOG_WARNING, "Trailing options were found on the "
"commandline.\n");
av_log(NULL, AV_LOG_DEBUG, "Finished splitting the commandline.\n");
@@ -878,54 +875,28 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
{ "debug" , AV_LOG_DEBUG },
{ "trace" , AV_LOG_TRACE },
};
const char *token;
char *tail;
int flags = av_log_get_flags();
int level = av_log_get_level();
int cmd, i = 0;
int level;
int flags;
int i;
av_assert0(arg);
while (*arg) {
token = arg;
if (*token == '+' || *token == '-') {
cmd = *token++;
} else {
cmd = 0;
}
if (!i && !cmd) {
flags = 0; /* missing relative prefix, build absolute value */
}
if (!strncmp(token, "repeat", 6)) {
if (cmd == '-') {
flags |= AV_LOG_SKIP_REPEATED;
} else {
flags &= ~AV_LOG_SKIP_REPEATED;
}
arg = token + 6;
} else if (!strncmp(token, "level", 5)) {
if (cmd == '-') {
flags &= ~AV_LOG_PRINT_LEVEL;
} else {
flags |= AV_LOG_PRINT_LEVEL;
}
arg = token + 5;
} else {
break;
}
i++;
}
if (!*arg) {
goto end;
} else if (*arg == '+') {
arg++;
} else if (!i) {
flags = av_log_get_flags(); /* level value without prefix, reset flags */
}
flags = av_log_get_flags();
tail = strstr(arg, "repeat");
if (tail)
flags &= ~AV_LOG_SKIP_REPEATED;
else
flags |= AV_LOG_SKIP_REPEATED;
av_log_set_flags(flags);
if (tail == arg)
arg += 6 + (arg[6]=='+');
if(tail && !*arg)
return 0;
for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) {
if (!strcmp(log_levels[i].name, arg)) {
level = log_levels[i].level;
goto end;
av_log_set_level(log_levels[i].level);
return 0;
}
}
@@ -937,9 +908,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
av_log(NULL, AV_LOG_FATAL, "\"%s\"\n", log_levels[i].name);
exit_program(1);
}
end:
av_log_set_flags(flags);
av_log_set_level(level);
return 0;
}
@@ -977,7 +945,6 @@ static int init_report(const char *env)
char *filename_template = NULL;
char *key, *val;
int ret, count = 0;
int prog_loglevel, envlevel = 0;
time_t now;
struct tm *tm;
AVBPrint filename;
@@ -1009,7 +976,6 @@ static int init_report(const char *env)
av_log(NULL, AV_LOG_FATAL, "Invalid report file level\n");
exit_program(1);
}
envlevel = 1;
} else {
av_log(NULL, AV_LOG_ERROR, "Unknown key '%s' in FFREPORT\n", key);
}
@@ -1017,7 +983,7 @@ static int init_report(const char *env)
av_free(key);
}
av_bprint_init(&filename, 0, AV_BPRINT_SIZE_AUTOMATIC);
av_bprint_init(&filename, 0, 1);
expand_filename_template(&filename,
av_x_if_null(filename_template, "%p-%t.log"), tm);
av_free(filename_template);
@@ -1026,10 +992,6 @@ static int init_report(const char *env)
return AVERROR(ENOMEM);
}
prog_loglevel = av_log_get_level();
if (!envlevel)
report_file_level = FFMAX(report_file_level, prog_loglevel);
report_file = fopen(filename.str, "w");
if (!report_file) {
int ret = AVERROR(errno);
@@ -1040,17 +1002,16 @@ static int init_report(const char *env)
av_log_set_callback(log_callback_report);
av_log(NULL, AV_LOG_INFO,
"%s started on %04d-%02d-%02d at %02d:%02d:%02d\n"
"Report written to \"%s\"\n"
"Log level: %d\n",
"Report written to \"%s\"\n",
program_name,
tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday,
tm->tm_hour, tm->tm_min, tm->tm_sec,
filename.str, report_file_level);
filename.str);
av_bprint_finalize(&filename, NULL);
return 0;
}
int opt_report(void *optctx, const char *opt, const char *arg)
int opt_report(const char *opt)
{
return init_report(NULL);
}
@@ -1290,12 +1251,10 @@ static int is_device(const AVClass *avclass)
return AV_IS_INPUT_DEVICE(avclass->category) || AV_IS_OUTPUT_DEVICE(avclass->category);
}
static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only, int muxdemuxers)
static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
{
void *ifmt_opaque = NULL;
const AVInputFormat *ifmt = NULL;
void *ofmt_opaque = NULL;
const AVOutputFormat *ofmt = NULL;
AVInputFormat *ifmt = NULL;
AVOutputFormat *ofmt = NULL;
const char *last_name;
int is_dev;
@@ -1310,35 +1269,29 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
const char *name = NULL;
const char *long_name = NULL;
if (muxdemuxers !=SHOW_DEMUXERS) {
ofmt_opaque = NULL;
while ((ofmt = av_muxer_iterate(&ofmt_opaque))) {
is_dev = is_device(ofmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ofmt->name, name) < 0) &&
strcmp(ofmt->name, last_name) > 0) {
name = ofmt->name;
long_name = ofmt->long_name;
encode = 1;
}
while ((ofmt = av_oformat_next(ofmt))) {
is_dev = is_device(ofmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ofmt->name, name) < 0) &&
strcmp(ofmt->name, last_name) > 0) {
name = ofmt->name;
long_name = ofmt->long_name;
encode = 1;
}
}
if (muxdemuxers != SHOW_MUXERS) {
ifmt_opaque = NULL;
while ((ifmt = av_demuxer_iterate(&ifmt_opaque))) {
is_dev = is_device(ifmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ifmt->name, name) < 0) &&
strcmp(ifmt->name, last_name) > 0) {
name = ifmt->name;
long_name = ifmt->long_name;
encode = 0;
}
if (name && strcmp(ifmt->name, name) == 0)
decode = 1;
while ((ifmt = av_iformat_next(ifmt))) {
is_dev = is_device(ifmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ifmt->name, name) < 0) &&
strcmp(ifmt->name, last_name) > 0) {
name = ifmt->name;
long_name = ifmt->long_name;
encode = 0;
}
if (name && strcmp(ifmt->name, name) == 0)
decode = 1;
}
if (!name)
break;
@@ -1355,22 +1308,12 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
int show_formats(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 0, SHOW_DEFAULT);
}
int show_muxers(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 0, SHOW_MUXERS);
}
int show_demuxers(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 0, SHOW_DEMUXERS);
return show_formats_devices(optctx, opt, arg, 0);
}
int show_devices(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 1, SHOW_DEFAULT);
return show_formats_devices(optctx, opt, arg, 1);
}
#define PRINT_CODEC_SUPPORTED(codec, field, type, list_name, term, get_name) \
@@ -1418,12 +1361,6 @@ static void print_codec(const AVCodec *c)
AV_CODEC_CAP_SLICE_THREADS |
AV_CODEC_CAP_AUTO_THREADS))
printf("threads ");
if (c->capabilities & AV_CODEC_CAP_AVOID_PROBING)
printf("avoidprobe ");
if (c->capabilities & AV_CODEC_CAP_HARDWARE)
printf("hardware ");
if (c->capabilities & AV_CODEC_CAP_HYBRID)
printf("hybrid ");
if (!c->capabilities)
printf("none");
printf("\n");
@@ -1444,17 +1381,6 @@ static void print_codec(const AVCodec *c)
printf("\n");
}
if (avcodec_get_hw_config(c, 0)) {
printf(" Supported hardware devices: ");
for (int i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(c, i);
if (!config)
break;
printf("%s ", av_hwdevice_get_type_name(config->device_type));
}
printf("\n");
}
if (c->supported_framerates) {
const AVRational *fps = c->supported_framerates;
@@ -1493,14 +1419,13 @@ static char get_media_type_char(enum AVMediaType type)
}
}
static const AVCodec *next_codec_for_id(enum AVCodecID id, void **iter,
static const AVCodec *next_codec_for_id(enum AVCodecID id, const AVCodec *prev,
int encoder)
{
const AVCodec *c;
while ((c = av_codec_iterate(iter))) {
if (c->id == id &&
(encoder ? av_codec_is_encoder(c) : av_codec_is_decoder(c)))
return c;
while ((prev = av_codec_next(prev))) {
if (prev->id == id &&
(encoder ? av_codec_is_encoder(prev) : av_codec_is_decoder(prev)))
return prev;
}
return NULL;
}
@@ -1537,12 +1462,11 @@ static unsigned get_codecs_sorted(const AVCodecDescriptor ***rcodecs)
static void print_codecs_for_id(enum AVCodecID id, int encoder)
{
void *iter = NULL;
const AVCodec *codec;
const AVCodec *codec = NULL;
printf(" (%s: ", encoder ? "encoders" : "decoders");
while ((codec = next_codec_for_id(id, &iter, encoder)))
while ((codec = next_codec_for_id(id, codec, encoder)))
printf("%s ", codec->name);
printf(")");
@@ -1565,8 +1489,7 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
" -------\n");
for (i = 0; i < nb_codecs; i++) {
const AVCodecDescriptor *desc = codecs[i];
const AVCodec *codec;
void *iter = NULL;
const AVCodec *codec = NULL;
if (strstr(desc->name, "_deprecated"))
continue;
@@ -1584,14 +1507,14 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
/* print decoders/encoders when there's more than one or their
* names are different from codec name */
while ((codec = next_codec_for_id(desc->id, &iter, 0))) {
while ((codec = next_codec_for_id(desc->id, codec, 0))) {
if (strcmp(codec->name, desc->name)) {
print_codecs_for_id(desc->id, 0);
break;
}
}
iter = NULL;
while ((codec = next_codec_for_id(desc->id, &iter, 1))) {
codec = NULL;
while ((codec = next_codec_for_id(desc->id, codec, 1))) {
if (strcmp(codec->name, desc->name)) {
print_codecs_for_id(desc->id, 1);
break;
@@ -1622,10 +1545,9 @@ static void print_codecs(int encoder)
encoder ? "Encoders" : "Decoders");
for (i = 0; i < nb_codecs; i++) {
const AVCodecDescriptor *desc = codecs[i];
const AVCodec *codec;
void *iter = NULL;
const AVCodec *codec = NULL;
while ((codec = next_codec_for_id(desc->id, &iter, encoder))) {
while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
printf(" %c", get_media_type_char(desc->type));
printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
@@ -1657,11 +1579,10 @@ int show_encoders(void *optctx, const char *opt, const char *arg)
int show_bsfs(void *optctx, const char *opt, const char *arg)
{
const AVBitStreamFilter *bsf = NULL;
void *opaque = NULL;
AVBitStreamFilter *bsf = NULL;
printf("Bitstream filters:\n");
while ((bsf = av_bsf_iterate(&opaque)))
while ((bsf = av_bitstream_filter_next(bsf)))
printf("%s\n", bsf->name);
printf("\n");
return 0;
@@ -1687,7 +1608,6 @@ int show_filters(void *optctx, const char *opt, const char *arg)
#if CONFIG_AVFILTER
const AVFilter *filter = NULL;
char descr[64], *descr_cur;
void *opaque = NULL;
int i, j;
const AVFilterPad *pad;
@@ -1699,7 +1619,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
" V = Video input/output\n"
" N = Dynamic number and/or type of input/output\n"
" | = Source or sink filter\n");
while ((filter = av_filter_iterate(&opaque))) {
while ((filter = avfilter_next(filter))) {
descr_cur = descr;
for (i = 0; i < 2; i++) {
if (i) {
@@ -1762,7 +1682,7 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg)
#endif
while ((pix_desc = av_pix_fmt_desc_next(pix_desc))) {
enum AVPixelFormat av_unused pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
enum AVPixelFormat pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
printf("%c%c%c%c%c %-16s %d %2d\n",
sws_isSupportedInput (pix_fmt) ? 'I' : '.',
sws_isSupportedOutput(pix_fmt) ? 'O' : '.',
@@ -1830,10 +1750,9 @@ static void show_help_codec(const char *name, int encoder)
if (codec)
print_codec(codec);
else if ((desc = avcodec_descriptor_get_by_name(name))) {
void *iter = NULL;
int printed = 0;
while ((codec = next_codec_for_id(desc->id, &iter, encoder))) {
while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
printed = 1;
print_codec(codec);
}
@@ -1868,24 +1787,6 @@ static void show_help_demuxer(const char *name)
show_help_children(fmt->priv_class, AV_OPT_FLAG_DECODING_PARAM);
}
static void show_help_protocol(const char *name)
{
const AVClass *proto_class;
if (!name) {
av_log(NULL, AV_LOG_ERROR, "No protocol name specified.\n");
return;
}
proto_class = avio_protocol_get_class(name);
if (!proto_class) {
av_log(NULL, AV_LOG_ERROR, "Unknown protocol '%s'.\n", name);
return;
}
show_help_children(proto_class, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM);
}
static void show_help_muxer(const char *name)
{
const AVCodecDescriptor *desc;
@@ -1975,25 +1876,6 @@ static void show_help_filter(const char *name)
}
#endif
static void show_help_bsf(const char *name)
{
const AVBitStreamFilter *bsf = av_bsf_get_by_name(name);
if (!name) {
av_log(NULL, AV_LOG_ERROR, "No bitstream filter name specified.\n");
return;
} else if (!bsf) {
av_log(NULL, AV_LOG_ERROR, "Unknown bit stream filter '%s'.\n", name);
return;
}
printf("Bit stream filter %s\n", bsf->name);
PRINT_CODEC_SUPPORTED(bsf, codec_ids, enum AVCodecID, "codecs",
AV_CODEC_ID_NONE, GET_CODEC_NAME);
if (bsf->priv_class)
show_help_children(bsf->priv_class, AV_OPT_FLAG_BSF_PARAM);
}
int show_help(void *optctx, const char *opt, const char *arg)
{
char *topic, *par;
@@ -2016,14 +1898,10 @@ int show_help(void *optctx, const char *opt, const char *arg)
show_help_demuxer(par);
} else if (!strcmp(topic, "muxer")) {
show_help_muxer(par);
} else if (!strcmp(topic, "protocol")) {
show_help_protocol(par);
#if CONFIG_AVFILTER
} else if (!strcmp(topic, "filter")) {
show_help_filter(par);
#endif
} else if (!strcmp(topic, "bsf")) {
show_help_bsf(par);
} else {
show_help_default(topic, par);
}
@@ -2057,7 +1935,7 @@ FILE *get_preset_file(char *filename, size_t filename_size,
av_strlcpy(filename, preset_name, filename_size);
f = fopen(filename, "r");
} else {
#if HAVE_GETMODULEHANDLE && defined(_WIN32)
#ifdef _WIN32
char datadir[MAX_PATH], *ls;
base[2] = NULL;
@@ -2115,7 +1993,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
codec = s->oformat ? avcodec_find_encoder(codec_id)
: avcodec_find_decoder(codec_id);
switch (st->codecpar->codec_type) {
switch (st->codec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
prefix = 'v';
flags |= AV_OPT_FLAG_VIDEO_PARAM;
@@ -2173,7 +2051,7 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
return NULL;
}
for (i = 0; i < s->nb_streams; i++)
opts[i] = filter_codec_opts(codec_opts, s->streams[i]->codecpar->codec_id,
opts[i] = filter_codec_opts(codec_opts, s->streams[i]->codec->codec_id,
s, s->streams[i], NULL);
return opts;
}
@@ -2199,10 +2077,18 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
double get_rotation(AVStream *st)
{
AVDictionaryEntry *rotate_tag = av_dict_get(st->metadata, "rotate", NULL, 0);
uint8_t* displaymatrix = av_stream_get_side_data(st,
AV_PKT_DATA_DISPLAYMATRIX, NULL);
double theta = 0;
if (displaymatrix)
if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
char *tail;
theta = av_strtod(rotate_tag->value, &tail);
if (*tail)
theta = 0;
}
if (displaymatrix && !theta)
theta = -av_display_rotation_get((int32_t*) displaymatrix);
theta -= 360*floor(theta/360 + 0.9/360);
@@ -2210,7 +2096,7 @@ double get_rotation(AVStream *st)
if (fabs(theta - 90*round(theta/90)) > 2)
av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
"If you want to help, upload a sample "
"of this file to https://streams.videolan.org/upload/ "
"of this file to ftp://upload.ffmpeg.org/incoming/ "
"and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");
return theta;
@@ -2225,7 +2111,7 @@ static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
if (!fmt || !fmt->priv_class || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
return AVERROR(EINVAL);
printf("Auto-detected sources for %s:\n", fmt->name);
printf("Audo-detected sources for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sources. Not implemented.\n");
@@ -2255,7 +2141,7 @@ static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
if (!fmt || !fmt->priv_class || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
return AVERROR(EINVAL);
printf("Auto-detected sinks for %s:\n", fmt->name);
printf("Audo-detected sinks for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sinks. Not implemented.\n");

View File

@@ -19,8 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef FFTOOLS_CMDUTILS_H
#define FFTOOLS_CMDUTILS_H
#ifndef CMDUTILS_H
#define CMDUTILS_H
#include <stdint.h>
@@ -99,12 +99,18 @@ int opt_default(void *optctx, const char *opt, const char *arg);
*/
int opt_loglevel(void *optctx, const char *opt, const char *arg);
int opt_report(void *optctx, const char *opt, const char *arg);
int opt_report(const char *opt);
int opt_max_alloc(void *optctx, const char *opt, const char *arg);
int opt_codec_debug(void *optctx, const char *opt, const char *arg);
#if CONFIG_OPENCL
int opt_opencl(void *optctx, const char *opt, const char *arg);
int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
#endif
/**
* Limit the execution time.
*/
@@ -149,7 +155,6 @@ typedef struct SpecifierOpt {
uint8_t *str;
int i;
int64_t i64;
uint64_t ui64;
float f;
double dbl;
} u;
@@ -201,47 +206,6 @@ typedef struct OptionDef {
void show_help_options(const OptionDef *options, const char *msg, int req_flags,
int rej_flags, int alt_flags);
#if CONFIG_AVDEVICE
#define CMDUTILS_COMMON_OPTIONS_AVDEVICE \
{ "sources" , OPT_EXIT | HAS_ARG, { .func_arg = show_sources }, \
"list sources of the input device", "device" }, \
{ "sinks" , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks }, \
"list sinks of the output device", "device" }, \
#else
#define CMDUTILS_COMMON_OPTIONS_AVDEVICE
#endif
#define CMDUTILS_COMMON_OPTIONS \
{ "L", OPT_EXIT, { .func_arg = show_license }, "show license" }, \
{ "h", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "?", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "help", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "-help", OPT_EXIT, { .func_arg = show_help }, "show help", "topic" }, \
{ "version", OPT_EXIT, { .func_arg = show_version }, "show version" }, \
{ "buildconf", OPT_EXIT, { .func_arg = show_buildconf }, "show build configuration" }, \
{ "formats", OPT_EXIT, { .func_arg = show_formats }, "show available formats" }, \
{ "muxers", OPT_EXIT, { .func_arg = show_muxers }, "show available muxers" }, \
{ "demuxers", OPT_EXIT, { .func_arg = show_demuxers }, "show available demuxers" }, \
{ "devices", OPT_EXIT, { .func_arg = show_devices }, "show available devices" }, \
{ "codecs", OPT_EXIT, { .func_arg = show_codecs }, "show available codecs" }, \
{ "decoders", OPT_EXIT, { .func_arg = show_decoders }, "show available decoders" }, \
{ "encoders", OPT_EXIT, { .func_arg = show_encoders }, "show available encoders" }, \
{ "bsfs", OPT_EXIT, { .func_arg = show_bsfs }, "show available bit stream filters" }, \
{ "protocols", OPT_EXIT, { .func_arg = show_protocols }, "show available protocols" }, \
{ "filters", OPT_EXIT, { .func_arg = show_filters }, "show available filters" }, \
{ "pix_fmts", OPT_EXIT, { .func_arg = show_pix_fmts }, "show available pixel formats" }, \
{ "layouts", OPT_EXIT, { .func_arg = show_layouts }, "show standard channel layouts" }, \
{ "sample_fmts", OPT_EXIT, { .func_arg = show_sample_fmts }, "show available audio sample formats" }, \
{ "colors", OPT_EXIT, { .func_arg = show_colors }, "show available color names" }, \
{ "loglevel", HAS_ARG, { .func_arg = opt_loglevel }, "set logging level", "loglevel" }, \
{ "v", HAS_ARG, { .func_arg = opt_loglevel }, "set logging level", "loglevel" }, \
{ "report", 0, { .func_arg = opt_report }, "generate a report" }, \
{ "max_alloc", HAS_ARG, { .func_arg = opt_max_alloc }, "set maximum size of a single allocated block", "bytes" }, \
{ "cpuflags", HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" }, \
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" }, \
CMDUTILS_COMMON_OPTIONS_AVDEVICE \
/**
* Show help for all options with given flags in class and all its
* children.
@@ -477,20 +441,6 @@ int show_license(void *optctx, const char *opt, const char *arg);
*/
int show_formats(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the muxers supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_muxers(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the demuxer supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_demuxers(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the devices supported by the
* program.
@@ -500,13 +450,13 @@ int show_devices(void *optctx, const char *opt, const char *arg);
#if CONFIG_AVDEVICE
/**
* Print a listing containing autodetected sinks of the output device.
* Print a listing containing audodetected sinks of the output device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sinks(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing autodetected sources of the input device.
* Print a listing containing audodetected sources of the input device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sources(void *optctx, const char *opt, const char *arg);
@@ -625,9 +575,6 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
#define GET_PIX_FMT_NAME(pix_fmt)\
const char *name = av_get_pix_fmt_name(pix_fmt);
#define GET_CODEC_NAME(id)\
const char *name = avcodec_descriptor_get(id)->name;
#define GET_SAMPLE_FMT_NAME(sample_fmt)\
const char *name = av_get_sample_fmt_name(sample_fmt)
@@ -645,4 +592,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
double get_rotation(AVStream *st);
#endif /* FFTOOLS_CMDUTILS_H */
#endif /* CMDUTILS_H */

35
cmdutils_common_opts.h Normal file
View File

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

278
cmdutils_opencl.c Normal file
View File

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

View File

@@ -2,12 +2,15 @@
# common bits used by all libraries
#
DEFAULT_X86ASMD=.dbg
# first so "all" becomes default target
all: all-yes
DEFAULT_YASMD=.dbg
ifeq ($(DBG),1)
X86ASMD=$(DEFAULT_X86ASMD)
YASMD=$(DEFAULT_YASMD)
else
X86ASMD=
YASMD=
endif
ifndef SUBDIR
@@ -15,8 +18,8 @@ ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC
SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
MSG = $@
M = @$(call ECHO,$(TAG),$@);
@@ -37,7 +40,7 @@ OBJCFLAGS += $(EOBJCFLAGS)
OBJCCFLAGS = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
@@ -51,9 +54,7 @@ COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
COMPILE_M = $(call COMPILE,OBJCC)
COMPILE_X86ASM = $(call COMPILE,X86ASM)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
COMPILE_NVCC = $(call COMPILE,NVCC)
%.o: %.c
$(COMPILE_C)
@@ -73,14 +74,6 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
%_host.o: %.c
$(COMPILE_HOSTC)
%$(DEFAULT_X86ASMD).asm: %.asm
$(DEPX86ASM) $(X86ASMFLAGS) -M -o $@ $< > $(@:.asm=.d)
$(X86ASM) $(X86ASMFLAGS) -e $< | sed '/^%/d;/^$$/d;' > $@
%.o: %.asm
$(COMPILE_X86ASM)
-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
%.o: %.rc
$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
@@ -90,13 +83,12 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
%.h.c:
$(Q)echo '#include "$*.h"' >$@
%.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
$(COMPILE_NVCC)
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ | sed -e 's/:/:\
/' -e 's/; /;\
/g' > $@
%.ptx.c: %.ptx
$(Q)sh $(SRC_PATH)/compat/cuda/ptx2c.sh $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
%.c %.h %.pc %.ver %.version: TAG = GEN
%.c %.h: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@@ -110,7 +102,7 @@ COMPILE_NVCC = $(call COMPILE,NVCC)
$(OBJS):
endif
include $(SRC_PATH)/ffbuild/arch.mak
include $(SRC_PATH)/arch.mak
OBJS += $(OBJS-yes)
SLIBOBJS += $(SLIBOBJS-yes)
@@ -118,7 +110,7 @@ FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
LDLIBS = $(FFLIBS:%=%$(BUILDSUF))
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=EXTRALIBS-%),$($(lib))) $(EXTRALIBS)
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
@@ -140,10 +132,8 @@ ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
HOBJS = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
PTXOBJS = $(filter %.ptx.o,$(OBJS))
$(HOBJS): CCFLAGS += $(CFLAGS_HEADERS)
checkheaders: $(HOBJS)
.SECONDARY: $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=)
.SECONDARY: $(HOBJS:.o=.c)
alltools: $(TOOLS)
@@ -151,7 +141,7 @@ $(HOSTOBJS): %.o: %.c
$(COMPILE_HOSTC)
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
@@ -160,9 +150,10 @@ $(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.c *.ver *.version *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ver-sol2 *.ho *.gcno *.gcda *$(DEFAULT_YASMD).asm
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
define RULES
@@ -172,4 +163,4 @@ endef
$(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)

View File

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

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

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

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

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

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// 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.
// NOTE: this is a partial update of the Avisynth C interface to recognize
// new color spaces added in Avisynth 2.60. By no means is this document
// completely Avisynth 2.60 compliant.
#ifndef __AVISYNTH_C__
#define __AVISYNTH_C__
#include "avs/config.h"
#include "avs/capi.h"
#include "avs/types.h"
/////////////////////////////////////////////////////////////////////
//
// Constants
//
#ifndef __AVISYNTH_6_H__
enum { AVISYNTH_INTERFACE_VERSION = 6 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
AVS_SAMPLE_INT16 = 1<<1,
AVS_SAMPLE_INT24 = 1<<2,
AVS_SAMPLE_INT32 = 1<<3,
AVS_SAMPLE_FLOAT = 1<<4};
enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_U=1<<1,
AVS_PLANAR_V=1<<2,
AVS_PLANAR_ALIGNED=1<<3,
AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
AVS_PLANAR_A=1<<4,
AVS_PLANAR_R=1<<5,
AVS_PLANAR_G=1<<6,
AVS_PLANAR_B=1<<7,
AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31,
AVS_CS_SHIFT_SUB_WIDTH = 0,
AVS_CS_SHIFT_SUB_HEIGHT = 8,
AVS_CS_SHIFT_SAMPLE_BITS = 16,
AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
AVS_CS_SUB_WIDTH_2 = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
AVS_CS_SUB_WIDTH_4 = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
AVS_CS_VPLANEFIRST = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
AVS_CS_UPLANEFIRST = 1 << 4, // I420
AVS_CS_SUB_HEIGHT_MASK = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
AVS_CS_SUB_HEIGHT_1 = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
AVS_CS_SUB_HEIGHT_2 = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
AVS_CS_SUB_HEIGHT_4 = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
AVS_CS_SAMPLE_BITS_MASK = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_8 = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_16 = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_SAMPLE_BITS_32 = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
AVS_CS_PLANAR_MASK = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
AVS_CS_PLANAR_FILTER = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
// AVS_CS_YV12 = 1<<3 Reserved
// AVS_CS_I420 = 1<<4 Reserved
AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
AVS_CS_YV24 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1, // YVU 4:4:4 planar
AVS_CS_YV16 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:2 planar
AVS_CS_YV12 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:0 planar
AVS_CS_I420 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YUV 4:2:0 planar
AVS_CS_IYUV = AVS_CS_I420,
AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:1 planar
AVS_CS_YUV9 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:0 planar
AVS_CS_Y8 = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 // Y 4:0:0 planar
};
enum {
AVS_IT_BFF = 1<<0,
AVS_IT_TFF = 1<<1,
AVS_IT_FIELDBASED = 1<<2};
enum {
AVS_FILTER_TYPE=1,
AVS_FILTER_INPUT_COLORSPACE=2,
AVS_FILTER_OUTPUT_TYPE=9,
AVS_FILTER_NAME=4,
AVS_FILTER_AUTHOR=5,
AVS_FILTER_VERSION=6,
AVS_FILTER_ARGS=7,
AVS_FILTER_ARGS_INFO=8,
AVS_FILTER_ARGS_DESCRIPTION=10,
AVS_FILTER_DESCRIPTION=11};
enum { //SUBTYPES
AVS_FILTER_TYPE_AUDIO=1,
AVS_FILTER_TYPE_VIDEO=2,
AVS_FILTER_OUTPUT_TYPE_SAME=3,
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
// New 2.6 explicitly defined cache hints.
AVS_CACHE_NOTHING=10, // Do not cache video.
AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
AVS_CACHE_GET_POLICY=30, // Get the current policy.
AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
AVS_CACHE_THREAD_SAFE=243, // Allow all threads in any instance.
AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
};
#ifdef BUILDING_AVSCORE
struct AVS_ScriptEnvironment {
IScriptEnvironment * env;
const char * error;
AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
: env(e), error(0) {}
};
#endif
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_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->image_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
{ return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_BFF); }
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
AVSC_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
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;
}
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
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));
}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
//
// VideoFrameBuffer holds information about a memory block which is used
// for video data. For efficiency, instances of this class are not deleted
// when the refcount reaches zero; instead they're stored in a linked list
// to be reused. The instances are deleted when the corresponding AVS
// file is closed.
// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrameBuffer {
BYTE * data;
int data_size;
// sequence_number is incremented every time the buffer is changed, so
// that stale views can tell they're no longer valid.
volatile long sequence_number;
volatile long refcount;
} AVS_VideoFrameBuffer;
// VideoFrame holds a "window" into a VideoFrameBuffer.
// AVS_VideoFrame is layed out identicly to IVideoFrame
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrame {
volatile long refcount;
AVS_VideoFrameBuffer * vfb;
int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture.
int row_sizeUV, heightUV;
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return avs_get_pitch_p(p, 0);}
#endif
AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
return avs_get_read_ptr_p(p, 0);}
#endif
AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
return avs_get_write_ptr_p(p, 0);}
#endif
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
{avs_release_video_frame(f);}
AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
{return avs_copy_video_frame(f);}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_Value
//
// Treat AVS_Value as a fat pointer. That is use avs_copy_value
// and avs_release_value appropiaty as you would if AVS_Value was
// a pointer.
// To maintain source code compatibility with future versions of the
// avisynth_c API don't use the AVS_Value directly. Use the helper
// functions below.
// AVS_Value is layed out identicly to AVSValue
typedef struct AVS_Value AVS_Value;
struct AVS_Value {
short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
// for some function e'rror
short array_size;
union {
void * clip; // do not use directly, use avs_take_clip
char boolean;
int integer;
float floating_pt;
const char * string;
const AVS_Value * array;
} d;
};
// AVS_Value should be initilized with avs_void.
// Should also set to avs_void after the value is released
// with avs_copy_value. Consider it the equalvent of setting
// a pointer to NULL
static const AVS_Value avs_void = {'v'};
AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
AVSC_API(void, avs_release_value)(AVS_Value);
AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
AVSC_INLINE int avs_as_bool(AVS_Value v)
{ return v.d.boolean; }
AVSC_INLINE int avs_as_int(AVS_Value v)
{ return v.d.integer; }
AVSC_INLINE const char * avs_as_string(AVS_Value v)
{ return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
AVSC_INLINE double avs_as_float(AVS_Value v)
{ return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
AVSC_INLINE const char * avs_as_error(AVS_Value v)
{ return avs_is_error(v) ? v.d.string : 0; }
AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
{ return v.d.array; }
AVSC_INLINE int avs_array_size(AVS_Value v)
{ return avs_is_array(v) ? v.array_size : 1; }
AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
{ return avs_is_array(v) ? v.d.array[index] : v; }
// only use these functions on an AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
// AVS_Clip
//
AVSC_API(void, avs_release_clip)(AVS_Clip *);
AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
AVSC_API(int, avs_get_version)(AVS_Clip *);
AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
// The returned video frame must be released with avs_release_video_frame
AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
// return field parity if field_based, else parity of first field in frame
AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
INT64 start, INT64 count);
// start and count are in samples
AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
int cachehints, int frame_range);
// This is the callback type used by avs_add_function
typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
(AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
typedef struct AVS_FilterInfo AVS_FilterInfo;
struct AVS_FilterInfo
{
// these members should not be modified outside of the AVS_ApplyFunc callback
AVS_Clip * child;
AVS_VideoInfo vi;
AVS_ScriptEnvironment * env;
AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
INT64 start, INT64 count);
int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
int frame_range);
void (AVSC_CC * free_filter)(AVS_FilterInfo *);
// Should be set when ever there is an error to report.
// It is cleared before any of the above methods are called
const char * error;
// this is to store whatever and may be modified at will
void * user_data;
};
// Create a new filter
// fi is set to point to the AVS_FilterInfo so that you can
// modify it once it is initilized.
// store_child should generally be set to true. If it is not
// set than ALL methods (the function pointers) must be defined
// If it is set than you do not need to worry about freeing the child
// clip.
AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
AVS_FilterInfo * * fi,
AVS_Value child, int store_child);
/////////////////////////////////////////////////////////////////////
//
// AVS_ScriptEnvironment
//
// For GetCPUFlags. These are backwards-compatible with those in VirtualDub.
enum {
/* slowest CPU to support extension */
AVS_CPU_FORCE = 0x01, // N/A
AVS_CPU_FPU = 0x02, // 386/486DX
AVS_CPU_MMX = 0x04, // P55C, K6, PII
AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon
AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP
AVS_CPU_SSE2 = 0x20, // PIV, Hammer
AVS_CPU_3DNOW = 0x40, // K6-2
AVS_CPU_3DNOW_EXT = 0x80, // Athlon
AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2,
// which only Hammer will have anyway)
AVS_CPUF_SSE3 = 0x100, // PIV+, K8 Venice
AVS_CPUF_SSSE3 = 0x200, // Core 2
AVS_CPUF_SSE4 = 0x400, // Penryn, Wolfdale, Yorkfield
AVS_CPUF_SSE4_1 = 0x400,
//AVS_CPUF_AVX = 0x800, // Sandy Bridge, Bulldozer
AVS_CPUF_SSE4_2 = 0x1000, // Nehalem
//AVS_CPUF_AVX2 = 0x2000, // Haswell
//AVS_CPUF_AVX512 = 0x4000, // Knights Landing
};
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(int, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
// note: val is really a va_list; I hope everyone typedefs va_list to a pointer
AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
const char * name, const char * params,
AVS_ApplyFunc apply, void * user_data);
AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
AVS_Value args, const char** arg_names);
// The returned value must be be released with avs_release_value
AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
// The returned value must be be released with avs_release_value
AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
//void avs_pop_context(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
const AVS_VideoInfo * vi, int align);
// align should be at least 16
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
#endif
AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
// The returned video frame must be be released
AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
// avisynth.dll exports this; it's a way to use it as a library, without
// writing an AVS script or without going through AVIFile.
AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
// this symbol is the entry point for the plugin and must
// be defined
AVSC_EXPORT
const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
// The returned video frame must be be released
#ifdef AVSC_NO_DECLSPEC
// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
/*
The following functions needs to have been declared, probably from windows.h
void* malloc(size_t)
void free(void*);
HMODULE LoadLibrary(const char*);
void* GetProcAddress(HMODULE, const char*);
FreeLibrary(HMODULE);
*/
typedef struct AVS_Library AVS_Library;
#define AVSC_DECLARE_FUNC(name) name##_func name
struct AVS_Library {
HMODULE handle;
AVSC_DECLARE_FUNC(avs_add_function);
AVSC_DECLARE_FUNC(avs_at_exit);
AVSC_DECLARE_FUNC(avs_bit_blt);
AVSC_DECLARE_FUNC(avs_check_version);
AVSC_DECLARE_FUNC(avs_clip_get_error);
AVSC_DECLARE_FUNC(avs_copy_clip);
AVSC_DECLARE_FUNC(avs_copy_value);
AVSC_DECLARE_FUNC(avs_copy_video_frame);
AVSC_DECLARE_FUNC(avs_create_script_environment);
AVSC_DECLARE_FUNC(avs_delete_script_environment);
AVSC_DECLARE_FUNC(avs_function_exists);
AVSC_DECLARE_FUNC(avs_get_audio);
AVSC_DECLARE_FUNC(avs_get_cpu_flags);
AVSC_DECLARE_FUNC(avs_get_frame);
AVSC_DECLARE_FUNC(avs_get_parity);
AVSC_DECLARE_FUNC(avs_get_var);
AVSC_DECLARE_FUNC(avs_get_version);
AVSC_DECLARE_FUNC(avs_get_video_info);
AVSC_DECLARE_FUNC(avs_invoke);
AVSC_DECLARE_FUNC(avs_make_writable);
AVSC_DECLARE_FUNC(avs_new_c_filter);
AVSC_DECLARE_FUNC(avs_new_video_frame_a);
AVSC_DECLARE_FUNC(avs_release_clip);
AVSC_DECLARE_FUNC(avs_release_value);
AVSC_DECLARE_FUNC(avs_release_video_frame);
AVSC_DECLARE_FUNC(avs_save_string);
AVSC_DECLARE_FUNC(avs_set_cache_hints);
AVSC_DECLARE_FUNC(avs_set_global_var);
AVSC_DECLARE_FUNC(avs_set_memory_max);
AVSC_DECLARE_FUNC(avs_set_to_clip);
AVSC_DECLARE_FUNC(avs_set_var);
AVSC_DECLARE_FUNC(avs_set_working_dir);
AVSC_DECLARE_FUNC(avs_sprintf);
AVSC_DECLARE_FUNC(avs_subframe);
AVSC_DECLARE_FUNC(avs_subframe_planar);
AVSC_DECLARE_FUNC(avs_take_clip);
AVSC_DECLARE_FUNC(avs_vsprintf);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_is_yv24);
AVSC_DECLARE_FUNC(avs_is_yv16);
AVSC_DECLARE_FUNC(avs_is_yv12);
AVSC_DECLARE_FUNC(avs_is_yv411);
AVSC_DECLARE_FUNC(avs_is_y8);
AVSC_DECLARE_FUNC(avs_is_color_space);
AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
AVSC_DECLARE_FUNC(avs_bits_per_pixel);
AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
AVSC_DECLARE_FUNC(avs_row_size);
AVSC_DECLARE_FUNC(avs_bmp_size);
AVSC_DECLARE_FUNC(avs_get_pitch_p);
AVSC_DECLARE_FUNC(avs_get_row_size_p);
AVSC_DECLARE_FUNC(avs_get_height_p);
AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
AVSC_DECLARE_FUNC(avs_is_writable);
AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
};
#undef AVSC_DECLARE_FUNC
AVSC_INLINE AVS_Library * avs_load_library() {
AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
if (library == NULL)
return NULL;
library->handle = LoadLibrary("avisynth");
if (library->handle == NULL)
goto fail;
#define __AVSC_STRINGIFY(x) #x
#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
#define AVSC_LOAD_FUNC(name) {\
library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
if (library->name == NULL)\
goto fail;\
}
AVSC_LOAD_FUNC(avs_add_function);
AVSC_LOAD_FUNC(avs_at_exit);
AVSC_LOAD_FUNC(avs_bit_blt);
AVSC_LOAD_FUNC(avs_check_version);
AVSC_LOAD_FUNC(avs_clip_get_error);
AVSC_LOAD_FUNC(avs_copy_clip);
AVSC_LOAD_FUNC(avs_copy_value);
AVSC_LOAD_FUNC(avs_copy_video_frame);
AVSC_LOAD_FUNC(avs_create_script_environment);
AVSC_LOAD_FUNC(avs_delete_script_environment);
AVSC_LOAD_FUNC(avs_function_exists);
AVSC_LOAD_FUNC(avs_get_audio);
AVSC_LOAD_FUNC(avs_get_cpu_flags);
AVSC_LOAD_FUNC(avs_get_frame);
AVSC_LOAD_FUNC(avs_get_parity);
AVSC_LOAD_FUNC(avs_get_var);
AVSC_LOAD_FUNC(avs_get_version);
AVSC_LOAD_FUNC(avs_get_video_info);
AVSC_LOAD_FUNC(avs_invoke);
AVSC_LOAD_FUNC(avs_make_writable);
AVSC_LOAD_FUNC(avs_new_c_filter);
AVSC_LOAD_FUNC(avs_new_video_frame_a);
AVSC_LOAD_FUNC(avs_release_clip);
AVSC_LOAD_FUNC(avs_release_value);
AVSC_LOAD_FUNC(avs_release_video_frame);
AVSC_LOAD_FUNC(avs_save_string);
AVSC_LOAD_FUNC(avs_set_cache_hints);
AVSC_LOAD_FUNC(avs_set_global_var);
AVSC_LOAD_FUNC(avs_set_memory_max);
AVSC_LOAD_FUNC(avs_set_to_clip);
AVSC_LOAD_FUNC(avs_set_var);
AVSC_LOAD_FUNC(avs_set_working_dir);
AVSC_LOAD_FUNC(avs_sprintf);
AVSC_LOAD_FUNC(avs_subframe);
AVSC_LOAD_FUNC(avs_subframe_planar);
AVSC_LOAD_FUNC(avs_take_clip);
AVSC_LOAD_FUNC(avs_vsprintf);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_is_yv24);
AVSC_LOAD_FUNC(avs_is_yv16);
AVSC_LOAD_FUNC(avs_is_yv12);
AVSC_LOAD_FUNC(avs_is_yv411);
AVSC_LOAD_FUNC(avs_is_y8);
AVSC_LOAD_FUNC(avs_is_color_space);
AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
AVSC_LOAD_FUNC(avs_bits_per_pixel);
AVSC_LOAD_FUNC(avs_bytes_from_pixels);
AVSC_LOAD_FUNC(avs_row_size);
AVSC_LOAD_FUNC(avs_bmp_size);
AVSC_LOAD_FUNC(avs_get_pitch_p);
AVSC_LOAD_FUNC(avs_get_row_size_p);
AVSC_LOAD_FUNC(avs_get_height_p);
AVSC_LOAD_FUNC(avs_get_read_ptr_p);
AVSC_LOAD_FUNC(avs_is_writable);
AVSC_LOAD_FUNC(avs_get_write_ptr_p);
#undef __AVSC_STRINGIFY
#undef AVSC_STRINGIFY
#undef AVSC_LOAD_FUNC
return library;
fail:
free(library);
return NULL;
}
AVSC_INLINE void avs_free_library(AVS_Library *library) {
if (library == NULL)
return;
FreeLibrary(library->handle);
free(library);
}
#endif
#endif

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@@ -0,0 +1,62 @@
// 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
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef BUILDING_AVSCORE
# 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
#endif //AVS_CAPI_H

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@@ -0,0 +1,55 @@
// 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 32
#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
#endif //AVS_CONFIG_H

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// 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
// 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

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// 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__

View File

@@ -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__

View File

@@ -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__

View File

@@ -1,131 +0,0 @@
/*
* Minimum CUDA compatibility definitions header
*
* Copyright (c) 2019 Rodger Combs
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_CUDA_CUDA_RUNTIME_H
#define COMPAT_CUDA_CUDA_RUNTIME_H
// Common macros
#define __global__ __attribute__((global))
#define __device__ __attribute__((device))
#define __device_builtin__ __attribute__((device_builtin))
#define __align__(N) __attribute__((aligned(N)))
#define __inline__ __inline__ __attribute__((always_inline))
#define max(a, b) ((a) > (b) ? (a) : (b))
#define min(a, b) ((a) < (b) ? (a) : (b))
#define abs(x) ((x) < 0 ? -(x) : (x))
#define atomicAdd(a, b) (__atomic_fetch_add(a, b, __ATOMIC_SEQ_CST))
// Basic typedefs
typedef __device_builtin__ unsigned long long cudaTextureObject_t;
typedef struct __device_builtin__ __align__(2) uchar2
{
unsigned char x, y;
} uchar2;
typedef struct __device_builtin__ __align__(4) ushort2
{
unsigned short x, y;
} ushort2;
typedef struct __device_builtin__ uint3
{
unsigned int x, y, z;
} 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;
} uchar4;
typedef struct __device_builtin__ __align__(8) ushort4
{
unsigned char x, y, z, w;
} ushort4;
typedef struct __device_builtin__ __align__(16) int4
{
int x, y, z, w;
} int4;
// Accessors for special registers
#define GETCOMP(reg, comp) \
asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
ret.comp = tmp;
#define GET(name, reg) static inline __device__ uint3 name() {\
uint3 ret; \
unsigned tmp; \
GETCOMP(reg, x) \
GETCOMP(reg, y) \
GETCOMP(reg, z) \
return ret; \
}
GET(getBlockIdx, ctaid)
GET(getBlockDim, ntid)
GET(getThreadIdx, tid)
// Instead of externs for these registers, we turn access to them into calls into trivial ASM
#define blockIdx (getBlockIdx())
#define blockDim (getBlockDim())
#define threadIdx (getThreadIdx())
// Basic initializers (simple macros rather than inline functions)
#define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
#define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
#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})
// 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(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
// Template calling tex instruction and converting the output to the selected type
template <class T>
static inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
{
T ret;
unsigned ret1, ret2, ret3, ret4;
asm("tex.2d.v4.u32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
"=r"(ret1), "=r"(ret2), "=r"(ret3), "=r"(ret4) :
"l"(texObject), "f"(x), "f"(y));
conv(&ret, ret1, ret2, ret3, ret4);
return ret;
}
#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

View File

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

View File

@@ -1,34 +0,0 @@
#!/bin/sh
# Copyright (c) 2017, NVIDIA CORPORATION. All rights reserved.
#
# Permission is hereby granted, free of charge, to any person obtaining a
# copy of this software and associated documentation files (the "Software"),
# to deal in the Software without restriction, including without limitation
# the rights to use, copy, modify, merge, publish, distribute, sublicense,
# and/or sell copies of the Software, and to permit persons to whom the
# Software is furnished to do so, subject to the following conditions:
#
# The above copyright notice and this permission notice shall be included in
# all copies or substantial portions of the Software.
#
# THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
# IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
# FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
# THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
# LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
# FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
# DEALINGS IN THE SOFTWARE.
set -e
OUT="$1"
IN="$2"
NAME="$(basename "$IN" | sed 's/\..*//')"
printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
echo >> "$OUT"
sed -e "$(printf 's/\r//g')" -e 's/["\\]/\\&/g' -e "$(printf 's/^/\t"/')" -e 's/$/\\n"/' < "$IN" >> "$OUT"
echo ";" >> "$OUT"
exit 0

View File

@@ -1,47 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <math.h>
#define FUN(name, type, op) \
type name(type x, type y) \
{ \
if (fpclassify(x) == FP_NAN) return y; \
if (fpclassify(y) == FP_NAN) return x; \
return x op y ? x : y; \
}
FUN(fmin, double, <)
FUN(fmax, double, >)
FUN(fminf, float, <)
FUN(fmaxf, float, >)
long double fmodl(long double x, long double y)
{
return fmod(x, y);
}
long double scalbnl(long double x, int exp)
{
return scalbn(x, exp);
}
long double copysignl(long double x, long double y)
{
return copysign(x, y);
}

View File

@@ -1,25 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
double fmin(double, double);
double fmax(double, double);
float fminf(float, float);
float fmaxf(float, float);
long double fmodl(long double, long double);
long double scalbnl(long double, int);
long double copysignl(long double, long double);

View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011-2017 KO Myung-Hun <komh@chollian.net>
* Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
*
* This file is part of FFmpeg.
*
@@ -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;
@@ -50,11 +46,9 @@ typedef struct {
typedef void pthread_attr_t;
typedef _fmutex pthread_mutex_t;
typedef HMTX pthread_mutex_t;
typedef void pthread_mutexattr_t;
#define PTHREAD_MUTEX_INITIALIZER _FMUTEX_INITIALIZER
typedef struct {
HEV event_sem;
HEV ack_sem;
@@ -104,28 +98,28 @@ static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
const pthread_mutexattr_t *attr)
{
_fmutex_create(mutex, 0);
DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
return 0;
}
static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
{
_fmutex_close(mutex);
DosCloseMutexSem(*(PHMTX)mutex);
return 0;
}
static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
{
_fmutex_request(mutex, 0);
DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
return 0;
}
static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
{
_fmutex_release(mutex);
DosReleaseMutexSem(*(PHMTX)mutex);
return 0;
}
@@ -167,28 +161,6 @@ 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)
{

10
compat/plan9/head Executable file
View File

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

View File

@@ -16,24 +16,19 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
int plan9_main(int argc, char **argv);
#include <pthread.h>
#include <stdint.h>
#include "stdatomic.h"
static pthread_mutex_t atomic_lock = PTHREAD_MUTEX_INITIALIZER;
void avpriv_atomic_lock(void)
#undef main
int main(int argc, char **argv)
{
pthread_mutex_lock(&atomic_lock);
}
/* The setfcr() function in lib9 is broken, must use asm. */
#ifdef __i386
short fcr;
__asm__ volatile ("fstcw %0 \n"
"or $63, %0 \n"
"fldcw %0 \n"
: "=m"(fcr));
#endif
void avpriv_atomic_unlock(void)
{
pthread_mutex_unlock(&atomic_lock);
return plan9_main(argc, argv);
}

2
compat/plan9/printf Executable file
View File

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

View File

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

View File

@@ -16,12 +16,15 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_VAAPI_HEVC_H
#define AVCODEC_VAAPI_HEVC_H
#ifndef COMPAT_TMS470_MATH_H
#define COMPAT_TMS470_MATH_H
#include <va/va.h>
#include "avcodec.h"
#include_next <math.h>
VAProfile ff_vaapi_parse_hevc_rext_profile(AVCodecContext *avctx);
#undef INFINITY
#undef NAN
#endif /* AVCODEC_VAAPI_HEVC_H */
#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
#define NAN (*(const float*)((const unsigned []){ 0x7fc00000 }))
#endif /* COMPAT_TMS470_MATH_H */

View File

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

View File

@@ -38,13 +38,16 @@
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <process.h>
#include <time.h>
#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
#undef MemoryBarrier
#define MemoryBarrier __sync_synchronize
#endif
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/internal.h"
#include "libavutil/mem.h"
#include "libavutil/time.h"
typedef struct pthread_t {
void *handle;
@@ -53,22 +56,28 @@ typedef struct pthread_t {
void *ret;
} pthread_t;
/* use light weight mutex/condition variable API for Windows Vista and later */
typedef SRWLOCK pthread_mutex_t;
/* the conditional variable api for windows 6.0+ uses critical sections and
* not mutexes */
typedef CRITICAL_SECTION pthread_mutex_t;
/* This is the CONDITION_VARIABLE typedef for using Windows' native
* conditional variables on kernels 6.0+. */
#if HAVE_CONDITION_VARIABLE_PTR
typedef CONDITION_VARIABLE pthread_cond_t;
#else
typedef struct pthread_cond_t {
void *Ptr;
} pthread_cond_t;
#endif
#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
#if _WIN32_WINNT >= 0x0600
#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
#define PTHREAD_CANCEL_ENABLE 1
#define PTHREAD_CANCEL_DISABLE 0
#endif
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t *h = (pthread_t*)arg;
pthread_t *h = arg;
h->ret = h->func(h->arg);
return 0;
}
@@ -105,25 +114,26 @@ static av_unused int pthread_join(pthread_t thread, void **value_ptr)
static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
{
InitializeSRWLock(m);
InitializeCriticalSection(m);
return 0;
}
static inline int pthread_mutex_destroy(pthread_mutex_t *m)
{
/* Unlocked SWR locks use no resources */
DeleteCriticalSection(m);
return 0;
}
static inline int pthread_mutex_lock(pthread_mutex_t *m)
{
AcquireSRWLockExclusive(m);
EnterCriticalSection(m);
return 0;
}
static inline int pthread_mutex_unlock(pthread_mutex_t *m)
{
ReleaseSRWLockExclusive(m);
LeaveCriticalSection(m);
return 0;
}
#if _WIN32_WINNT >= 0x0600
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
@@ -157,23 +167,7 @@ static inline int pthread_cond_broadcast(pthread_cond_t *cond)
static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
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;
}
SleepConditionVariableCS(cond, mutex, INFINITE);
return 0;
}
@@ -183,9 +177,242 @@ static inline int pthread_cond_signal(pthread_cond_t *cond)
return 0;
}
static inline int pthread_setcancelstate(int state, int *oldstate)
#else // _WIN32_WINNT < 0x0600
/* atomic init state of dynamically loaded functions */
static LONG w32thread_init_state = 0;
static av_unused void w32thread_init(void);
/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
* compatible to the one used in the native API */
typedef union pthread_once_t {
void * Ptr; ///< For the Windows 6.0+ native functions
LONG state; ///< For the pre-Windows 6.0 compat code
} pthread_once_t;
#define PTHREAD_ONCE_INIT {0}
/* function pointers to init once API on windows 6.0+ kernels */
static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
/* pre-Windows 6.0 compat using a spin-lock */
static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
{
switch (InterlockedCompareExchange(state, 1, 0)) {
/* Initial run */
case 0:
init_routine();
InterlockedExchange(state, 2);
break;
/* Another thread is running init */
case 1:
while (1) {
MemoryBarrier();
if (*state == 2)
break;
Sleep(0);
}
break;
/* Initialization complete */
case 2:
break;
}
}
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
/* Use native functions on Windows 6.0+ */
if (initonce_begin && initonce_complete) {
BOOL pending = FALSE;
initonce_begin(once_control, 0, &pending, NULL);
if (pending)
init_routine();
initonce_complete(once_control, 0, NULL);
return 0;
}
w32thread_once_fallback(&once_control->state, init_routine);
return 0;
}
/* for pre-Windows 6.0 platforms we need to define and use our own condition
* variable and api */
typedef struct win32_cond_t {
pthread_mutex_t mtx_broadcast;
pthread_mutex_t mtx_waiter_count;
volatile int waiter_count;
HANDLE semaphore;
HANDLE waiters_done;
volatile int is_broadcast;
} win32_cond_t;
/* function pointers to conditional variable API on windows 6.0+ kernels */
static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
static void (WINAPI *cond_init)(pthread_cond_t *cond);
static void (WINAPI *cond_signal)(pthread_cond_t *cond);
static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
DWORD milliseconds);
static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
win32_cond_t *win32_cond = NULL;
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
if (cond_init) {
cond_init(cond);
return 0;
}
/* non native condition variables */
win32_cond = av_mallocz(sizeof(win32_cond_t));
if (!win32_cond)
return ENOMEM;
cond->Ptr = win32_cond;
win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
if (!win32_cond->semaphore)
return ENOMEM;
win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
if (!win32_cond->waiters_done)
return ENOMEM;
pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
return 0;
}
static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
/* native condition variables do not destroy */
if (cond_init)
return 0;
/* non native condition variables */
CloseHandle(win32_cond->semaphore);
CloseHandle(win32_cond->waiters_done);
pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
pthread_mutex_destroy(&win32_cond->mtx_broadcast);
av_freep(&win32_cond);
cond->Ptr = NULL;
return 0;
}
static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
int have_waiter;
if (cond_broadcast) {
cond_broadcast(cond);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = 0;
if (win32_cond->waiter_count) {
win32_cond->is_broadcast = 1;
have_waiter = 1;
}
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
win32_cond->is_broadcast = 0;
} else
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
win32_cond_t *win32_cond = cond->Ptr;
int last_waiter;
if (cond_wait) {
cond_wait(cond, mutex, INFINITE);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count++;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
// unlock the external mutex
pthread_mutex_unlock(mutex);
WaitForSingleObject(win32_cond->semaphore, INFINITE);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count--;
last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (last_waiter)
SetEvent(win32_cond->waiters_done);
// lock the external mutex
return pthread_mutex_lock(mutex);
}
static av_unused int pthread_cond_signal(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
int have_waiter;
if (cond_signal) {
cond_signal(cond);
return 0;
}
pthread_mutex_lock(&win32_cond->mtx_broadcast);
/* non-native condition variables */
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = win32_cond->waiter_count;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
}
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
#endif
static av_unused void w32thread_init(void)
{
#if _WIN32_WINNT < 0x0600
HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
/* if one is available, then they should all be available */
cond_init =
(void*)GetProcAddress(kernel_dll, "InitializeConditionVariable");
cond_broadcast =
(void*)GetProcAddress(kernel_dll, "WakeAllConditionVariable");
cond_signal =
(void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
cond_wait =
(void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
initonce_begin =
(void*)GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
initonce_complete =
(void*)GetProcAddress(kernel_dll, "InitOnceComplete");
#endif
}
#endif /* COMPAT_W32PTHREADS_H */

View File

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

View File

@@ -4,6 +4,6 @@ LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
if [ -x "$LINK_EXE_PATH" ]; then
"$LINK_EXE_PATH" $@
else
link.exe $@
link $@
fi
exit $?

4295
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -2,630 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
since the last major version increase or the API was added.
The last version increases were:
libavcodec: 2017-10-21
libavdevice: 2017-10-21
libavfilter: 2017-10-21
libavformat: 2017-10-21
libavresample: 2017-10-21
libpostproc: 2017-10-21
libswresample: 2017-10-21
libswscale: 2017-10-21
libavutil: 2017-10-21
libavcodec: 2015-08-28
libavdevice: 2015-08-28
libavfilter: 2015-08-28
libavformat: 2015-08-28
libavresample: 2015-08-28
libpostproc: 2015-08-28
libswresample: 2015-08-28
libswscale: 2015-08-28
libavutil: 2015-08-28
API changes, most recent first:
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
Add FF_DECODE_ERROR_DECODE_SLICES
2019-06-14 - edfced8c04 - lavu 56.29.100 - frame.h
Add FF_DECODE_ERROR_CONCEALMENT_ACTIVE
2019-05-15 - b79b29ddb1 - lavu 56.28.100 - tx.h
Add av_tx_init(), av_tx_uninit() and related definitions.
2019-04-20 - 3153a6502a - lavc 58.52.100 - avcodec.h
Add AV_CODEC_FLAG_DROPCHANGED to allow avcodec_receive_frame to drop
frames whose parameters differ from first decoded frame in stream.
2019-04-12 - abfeba9724 - lavf 58.27.102
Rename hls,applehttp demuxer to hls
2019-01-27 - 5bcefceec8 - lavc 58.46.100 - avcodec.h
Add discard_damaged_percentage
2019-01-08 - 1ef4828276 - lavu 56.26.100 - frame.h
Add AV_FRAME_DATA_REGIONS_OF_INTEREST
2018-12-21 - 2744d6b364 - lavu 56.25.100 - hdr_dynamic_metadata.h
Add AV_FRAME_DATA_DYNAMIC_HDR_PLUS enum value, av_dynamic_hdr_plus_alloc(),
av_dynamic_hdr_plus_create_side_data() functions, and related structs.
-------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
Add av_read_image_line2(), av_write_image_line2()
2018-10-24 - f9d4126f28 - lavu 56.20.100 - frame.h
Add AV_FRAME_DATA_S12M_TIMECODE
2018-10-11 - f6d48b618a - lavc 58.33.100 - mediacodec.h
Add av_mediacodec_render_buffer_at_time().
2018-09-09 - 35498c124a - lavc 58.29.100 - avcodec.h
Add AV_PKT_DATA_AFD
2018-08-16 - b33f5299a5 - lavc 58.23.100 - avcodec.h
Add av_bsf_flush().
2018-05-18 - 2b2f2f65f3 - lavf 58.15.100 - avformat.h
Add pmt_version field to AVProgram
2018-05-17 - 5dfeb7f081 - lavf 58.14.100 - avformat.h
Add AV_DISPOSITION_STILL_IMAGE
2018-05-10 - c855683427 - lavu 56.18.101 - hwcontext_cuda.h
Add AVCUDADeviceContext.stream.
2018-04-30 - 56b081da57 - lavu 56.18.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_ALPHA to AV_PIX_FMT_PAL8.
2018-04-26 - 5be0410cb3 - lavu 56.17.100 - opt.h
Add AV_OPT_FLAG_DEPRECATED.
2018-04-26 - 71fa82bed6 - lavu 56.16.100 - threadmessage.h
Add av_thread_message_queue_nb_elems().
-------- 8< --------- FFmpeg 4.0 was cut here -------- 8< ---------
2018-04-03 - d6fc031caf - lavu 56.13.100 - pixdesc.h
Deprecate AV_PIX_FMT_FLAG_PSEUDOPAL and make allocating a pseudo palette
optional for API users (see AV_PIX_FMT_FLAG_PSEUDOPAL doxygen for details).
2018-04-01 - 860086ee16 - lavc 58.17.100 - avcodec.h
Add av_packet_make_refcounted().
2018-04-01 - f1805d160d - lavfi 7.14.100 - avfilter.h
Deprecate use of avfilter_register(), avfilter_register_all(),
avfilter_next(). Add av_filter_iterate().
2018-03-25 - b7d0d912ef - lavc 58.16.100 - avcodec.h
Add FF_SUB_CHARENC_MODE_IGNORE.
2018-03-23 - db2a7c947e - lavu 56.12.100 - encryption_info.h
Add AVEncryptionInitInfo and AVEncryptionInfo structures to hold new side-data
for encryption info.
2018-03-21 - f14ca60001 - lavc 58.15.100 - avcodec.h
Add av_packet_make_writable().
2018-03-18 - 4b86ac27a0 - lavu 56.11.100 - frame.h
Add AV_FRAME_DATA_QP_TABLE_PROPERTIES and AV_FRAME_DATA_QP_TABLE_DATA.
2018-03-15 - e0e72539cf - lavu 56.10.100 - opt.h
Add AV_OPT_FLAG_BSF_PARAM
2018-03-07 - 950170bd3b - lavu 56.9.100 - crc.h
Add AV_CRC_8_EBU crc variant.
2018-03-07 - 2a0eb86857 - lavc 58.14.100 - mediacodec.h
Change the default behavior of avcodec_flush() on mediacodec
video decoders. To restore the previous behavior, use the new
delay_flush=1 option.
2018-03-01 - 6731f60598 - lavu 56.8.100 - frame.h
Add av_frame_new_side_data_from_buf().
2018-02-15 - 8a8d0b319a
Change av_ripemd_update(), av_murmur3_update() and av_hash_update() length
parameter type to size_t at next major bump.
2018-02-12 - bcab11a1a2 - lavfi 7.12.100 - avfilter.h
Add AVFilterContext.extra_hw_frames.
2018-02-12 - d23fff0d8a - lavc 58.11.100 - avcodec.h
Add AVCodecContext.extra_hw_frames.
2018-02-06 - 0694d87024 - lavf 58.9.100 - avformat.h
Deprecate use of av_register_input_format(), av_register_output_format(),
av_register_all(), av_iformat_next(), av_oformat_next().
Add av_demuxer_iterate(), and av_muxer_iterate().
2018-02-06 - 36c85d6e77 - lavc 58.10.100 - avcodec.h
Deprecate use of avcodec_register(), avcodec_register_all(),
av_codec_next(), av_register_codec_parser(), and av_parser_next().
Add av_codec_iterate() and av_parser_iterate().
2018-02-04 - ff46124b0d - lavf 58.8.100 - avformat.h
Deprecate the current names of the RTSP "timeout", "stimeout", "user-agent"
options. Introduce "listen_timeout" as replacement for the current "timeout"
option, and "user_agent" as replacement for "user-agent". Once the deprecation
is over, the old "timeout" option will be removed, and "stimeout" will be
renamed to "stimeout" (the "timeout" option will essentially change semantics).
2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
Deprecate AVFormatContext filename field which had limited length, use the
new dynamically allocated url field instead.
2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
Add url field to AVFormatContext and add ff_format_set_url helper function.
2018-01-27 - 6194d7e564 - lavf 58.6.100 - avformat.h
Add AVFMTCTX_UNSEEKABLE (for HLS demuxer).
2018-01-23 - 9f07cf7c00 - lavu 56.9.100 - aes_ctr.h
Add method to set the 16-byte IV.
2018-01-16 - 631c56a8e4 - lavf 58.5.100 - avformat.h
Explicitly make avformat_network_init() and avformat_network_deinit() optional.
If these are not called, network initialization and deinitialization is
automatic, and unlike in older versions, fully supported, unless libavformat
is linked to ancient GnuTLS and OpenSSL.
2018-01-16 - 6512ff72f9 - lavf 58.4.100 - avformat.h
Deprecate AVStream.recommended_encoder_configuration. It was useful only for
FFserver, which has been removed.
2018-01-05 - 798dcf2432 - lavfi 7.11.101 - avfilter.h
Deprecate avfilter_link_get_channels(). Use av_buffersink_get_channels().
2017-01-04 - c29038f304 - lavr 4.0.0 - avresample.h
Deprecate the entire library. Merged years ago to provide compatibility
with Libav, it remained unmaintained by the FFmpeg project and duplicated
functionality provided by libswresample.
In order to improve consistency and reduce attack surface, it has been deprecated.
Users of this library are asked to migrate to libswresample, which, as well as
providing more functionality, is faster and has higher accuracy.
2017-12-26 - a04c2c707d - lavc 58.9.100 - avcodec.h
Deprecate av_lockmgr_register(). You need to build FFmpeg with threading
support enabled to get basic thread-safety (which is the default build
configuration).
2017-12-24 - 8b81eabe57 - lavu 56.7.100 - cpu.h
AVX-512 flags added.
2017-12-16 - 8bf4e6d3ce - lavc 58.8.100 - avcodec.h
The MediaCodec decoders now support AVCodecContext.hw_device_ctx.
2017-12-16 - e4d9f05ca7 - lavu 56.6.100 - hwcontext.h hwcontext_mediacodec.h
Add AV_HWDEVICE_TYPE_MEDIACODEC and a new installed header with
MediaCodec-specific hwcontext definitions.
2017-12-14 - b945fed629 - lavc 58.7.100 - avcodec.h
Add AV_CODEC_CAP_HARDWARE, AV_CODEC_CAP_HYBRID, and AVCodec.wrapper_name,
and mark all AVCodecs accordingly.
2017-11-29 - d268094f88 - lavu 56.4.100 / 56.7.0 - stereo3d.h
Add view field to AVStereo3D structure and AVStereo3DView enum.
2017-11-26 - 3a71bcc213 - lavc 58.6.100 - avcodec.h
Add const to AVCodecContext.hwaccel.
2017-11-26 - 3536a3efb9 - lavc 58.5.100 - avcodec.h
Deprecate user visibility of the AVHWAccel structure and the functions
av_register_hwaccel() and av_hwaccel_next().
2017-11-26 - 24cc0a53e9 - lavc 58.4.100 - avcodec.h
Add AVCodecHWConfig and avcodec_get_hw_config().
2017-11-22 - 3650cb2dfa - lavu 56.3.100 - opencl.h
Remove experimental OpenCL API (av_opencl_*).
2017-11-22 - b25d8ef0a7 - lavu 56.2.100 - hwcontext.h hwcontext_opencl.h
Add AV_HWDEVICE_TYPE_OPENCL and a new installed header with
OpenCL-specific hwcontext definitions.
2017-11-22 - a050f56c09 - lavu 56.1.100 - pixfmt.h
Add AV_PIX_FMT_OPENCL.
2017-11-11 - 48e4eda11d - lavc 58.3.100 - avcodec.h
Add avcodec_get_hw_frames_parameters().
-------- 8< --------- FFmpeg 3.4 was cut here -------- 8< ---------
2017-09-28 - b6cf66ae1c - lavc 57.106.104 - avcodec.h
Add AV_PKT_DATA_A53_CC packet side data, to export closed captions
2017-09-27 - 7aa6b8a68f - lavu 55.77.101 / lavu 55.31.1 - frame.h
Allow passing the value of 0 (meaning "automatic") as the required alignment
to av_frame_get_buffer().
2017-09-27 - 522f877086 - lavu 55.77.100 / lavu 55.31.0 - cpu.h
Add av_cpu_max_align() for querying maximum required data alignment.
2017-09-26 - b1cf151c4d - lavc 57.106.102 - avcodec.h
Deprecate AVCodecContext.refcounted_frames. This was useful for deprecated
API only (avcodec_decode_video2/avcodec_decode_audio4). The new decode APIs
(avcodec_send_packet/avcodec_receive_frame) always work with reference
counted frames.
2017-09-21 - 6f15f1cdc8 - lavu 55.76.100 / 56.6.0 - pixdesc.h
Add av_color_range_from_name(), av_color_primaries_from_name(),
av_color_transfer_from_name(), av_color_space_from_name(), and
av_chroma_location_from_name().
2017-09-13 - 82342cead1 - lavc 57.106.100 - avcodec.h
Add AV_PKT_FLAG_TRUSTED.
2017-09-13 - 9cb23cd9fe - lavu 55.75.100 - hwcontext.h hwcontext_drm.h
Add AV_HWDEVICE_TYPE_DRM and implementation.
2017-09-08 - 5ba2aef6ec - lavfi 6.103.100 - buffersrc.h
Add av_buffersrc_close().
2017-09-04 - 6cadbb16e9 - lavc 57.105.100 - avcodec.h
Add AV_HWACCEL_CODEC_CAP_EXPERIMENTAL, replacing the deprecated
HWACCEL_CODEC_CAP_EXPERIMENTAL flag.
2017-09-01 - 5d76674756 - lavf 57.81.100 - avio.h
Add avio_read_partial().
2017-09-01 - xxxxxxx - lavf 57.80.100 / 57.11.0 - avio.h
Add avio_context_free(). From now on it must be used for freeing AVIOContext.
2017-08-08 - 1460408703 - lavu 55.74.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_FLOAT pixel format flag.
2017-08-08 - 463b81de2b - lavu 55.72.100 - imgutils.h
Add av_image_fill_black().
2017-08-08 - caa12027ba - lavu 55.71.100 - frame.h
Add av_frame_apply_cropping().
2017-07-25 - 24de4fddca - lavu 55.69.100 - frame.h
Add AV_FRAME_DATA_ICC_PROFILE side data type.
2017-06-27 - 70143a3954 - lavc 57.100.100 - avcodec.h
DXVA2 and D3D11 hardware accelerated decoding now supports the new hwaccel API,
which can create the decoder context and allocate hardware frame automatically.
See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx. For D3D11,
the new AV_PIX_FMT_D3D11 pixfmt must be used with the new API.
2017-06-27 - 3303511f33 - lavu 56.67.100 - hwcontext.h
Add AV_HWDEVICE_TYPE_D3D11VA and AV_PIX_FMT_D3D11.
2017-06-24 - 09891c5391 - lavf 57.75.100 - avio.h
Add AVIO_DATA_MARKER_FLUSH_POINT to signal preferred flush points to aviobuf.
2017-06-14 - d59c6a3aeb - lavu 55.66.100 - hwcontext.h
av_hwframe_ctx_create_derived() now takes some AV_HWFRAME_MAP_* combination
as its flags argument (which was previously unused).
2017-06-14 - 49ae8a5e87 - lavc 57.99.100 - avcodec.h
Add AV_HWACCEL_FLAG_ALLOW_PROFILE_MISMATCH.
2017-06-14 - 0b1794a43e - lavu 55.65.100 - hwcontext.h
Add AV_HWDEVICE_TYPE_NONE, av_hwdevice_find_type_by_name(),
av_hwdevice_get_type_name() and av_hwdevice_iterate_types().
2017-06-14 - b22172f6f3 - lavu 55.64.100 - hwcontext.h
Add av_hwdevice_ctx_create_derived().
2017-05-15 - 532b23f079 - lavc 57.96.100 - avcodec.h
VideoToolbox hardware-accelerated decoding now supports the new hwaccel API,
which can create the decoder context and allocate hardware frames automatically.
See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx.
2017-05-15 - 532b23f079 - lavu 57.63.100 - hwcontext.h
Add AV_HWDEVICE_TYPE_VIDEOTOOLBOX and implementation.
2017-05-08 - f089e02fa2 - lavc 57.95.100 / 57.31.0 - avcodec.h
Add AVCodecContext.apply_cropping to control whether cropping
is handled by libavcodec or the caller.
2017-05-08 - a47bd5d77e - lavu 55.62.100 / 55.30.0 - frame.h
Add AVFrame.crop_left/right/top/bottom fields for attaching cropping
information to video frames.
2017-xx-xx - xxxxxxxxxx
Change av_sha_update(), av_sha512_update() and av_md5_sum()/av_md5_update() length
parameter type to size_t at next major bump.
2017-05-05 - c0f17a905f - lavc 57.94.100 - avcodec.h
The cuvid decoders now support AVCodecContext.hw_device_ctx, which removes
the requirement to set an incomplete AVCodecContext.hw_frames_ctx only to
set the Cuda device handle.
2017-04-11 - 8378466507 - lavu 55.61.100 - avstring.h
Add av_strireplace().
2016-04-06 - 157e57a181 - lavc 57.92.100 - avcodec.h
Add AV_PKT_DATA_CONTENT_LIGHT_LEVEL packet side data.
2016-04-06 - b378f5bd64 - lavu 55.60.100 - mastering_display_metadata.h
Add AV_FRAME_DATA_CONTENT_LIGHT_LEVEL value, av_content_light_metadata_alloc()
and av_content_light_metadata_create_side_data() API, and AVContentLightMetadata
type to export content light level video properties.
2017-03-31 - 9033e8723c - lavu 55.57.100 - spherical.h
Add av_spherical_projection_name().
Add av_spherical_from_name().
2017-03-30 - 4cda23f1f1 - lavu 55.53.100 / 55.27.0 - hwcontext.h
Add av_hwframe_map() and associated AV_HWFRAME_MAP_* flags.
Add av_hwframe_ctx_create_derived().
2017-03-29 - bfdcdd6d82 - lavu 55.52.100 - avutil.h
add av_fourcc_make_string() function and av_fourcc2str() macro to replace
av_get_codec_tag_string() from lavc.
2017-03-27 - ddef3d902f - lavf 57.68.100 - avformat.h
Deprecate that demuxers export the stream rotation angle in AVStream.metadata
(via an entry named "rotate"). Use av_stream_get_side_data() with
AV_PKT_DATA_DISPLAYMATRIX instead, and read the rotation angle with
av_display_rotation_get(). The same is done for muxing. Instead of adding a
"rotate" entry to AVStream.metadata, AV_PKT_DATA_DISPLAYMATRIX side data has
to be added to the AVStream.
2017-03-23 - 7e4ba776a2 - lavc 57.85.101 - avcodec.h
vdpau hardware accelerated decoding now supports the new hwaccel API, which
can create the decoder context and allocate hardware frame automatically.
See AVCodecContext.hw_device_ctx and AVCodecContext.hw_frames_ctx.
2017-03-23 - 156bd8278f - lavc 57.85.100 - avcodec.h
Add AVCodecContext.hwaccel_flags field. This will control some hwaccels at
a later point.
2017-03-21 - fc9f14c7de - lavf 57.67.100 / 57.08.0 - avio.h
Add AVIO_SEEKABLE_TIME flag.
2017-03-21 - d682ae70b4 - lavf 57.66.105, lavc 57.83.101 - avformat.h, avcodec.h
Deprecate AVFMT_FLAG_KEEP_SIDE_DATA. It will be ignored after the next major
bump, and libavformat will behave as if it were always set.
Deprecate av_packet_merge_side_data() and av_packet_split_side_data().
2016-03-20 - 8200b16a9c - lavu 55.50.100 / 55.21.0 - imgutils.h
Add av_image_copy_uc_from(), a version of av_image_copy() for copying
from GPU mapped memory.
2017-03-20 - 9c2436e - lavu 55.49.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_BAYER pixel format flag.
2017-03-18 - 3796fb2692 - lavfi 6.77.100 - avfilter.h
Deprecate AVFilterGraph.resample_lavr_opts
It's never been used by avfilter nor passed to anything.
2017-02-10 - 1b7ffddb3a - lavu 55.48.100 / 55.33.0 - spherical.h
Add AV_SPHERICAL_EQUIRECTANGULAR_TILE, av_spherical_tile_bounds(),
and projection-specific properties (bound_left, bound_top, bound_right,
bound_bottom, padding) to AVSphericalMapping.
2017-03-02 - ade7c1a232 - lavc 57.81.104 - videotoolbox.h
AVVideotoolboxContext.cv_pix_fmt_type can now be set to 0 to output the
native decoder format. (The default value is not changed.)
2017-03-02 - 554bc4eea8 - lavu 55.47.101, lavc 57.81.102, lavf 57.66.103
Remove requirement to use AVOption or accessors to access certain fields
in AVFrame, AVCodecContext, and AVFormatContext that were previously
documented as "no direct access" allowed.
2017-02-13 - c1a5fca06f - lavc 57.80.100 - avcodec.h
Add AVCodecContext.hw_device_ctx.
2017-02-11 - e3af49b14b - lavu 55.47.100 - frame.h
Add AVFrame.opaque_ref.
2017-01-31 - 2eab48177d - lavu 55.46.100 / 55.20.0 - cpu.h
Add AV_CPU_FLAG_SSSE3SLOW.
2017-01-24 - c4618f842a - lavu 55.45.100 - channel_layout.h
Add av_get_extended_channel_layout()
2017-01-22 - 76c5a69e26 - lavu 55.44.100 - lfg.h
Add av_lfg_init_from_data().
2017-01-17 - 2a4a8653b6 - lavc 57.74.100 - vaapi.h
Deprecate struct vaapi_context and the vaapi.h installed header.
Callers should set AVCodecContext.hw_frames_ctx instead.
2017-01-12 - dbe9dbed31 - lavfi 6.69.100 - buffersink.h
Add av_buffersink_get_*() functions.
2017-01-06 - 9488032e10 - lavf 57.62.100 - avio.h
Add avio_get_dyn_buf()
2016-12-10 - f542b152aa - lavu 55.43.100 - imgutils.h
Add av_image_check_size2()
2016-12-07 - e7a6f8c972 - lavc 57.67.100 / 57.29.0 - avcodec.h
Add AV_PKT_DATA_SPHERICAL packet side data to export AVSphericalMapping
information from containers.
2016-12-07 - 8f58ecc344 - lavu 55.42.100 / 55.30.0 - spherical.h
Add AV_FRAME_DATA_SPHERICAL value, av_spherical_alloc() API and
AVSphericalMapping type to export and describe spherical video properties.
2016-11-18 - 2ab50647ff - lavf 57.58.100 - avformat.h
Add av_stream_add_side_data().
2016-11-13 - 775a8477b7 - lavu 55.39.100 - hwcontext_vaapi.h
Add AV_VAAPI_DRIVER_QUIRK_ATTRIB_MEMTYPE.
2016-11-13 - a8d51bb424 - lavu 55.38.100 - hwcontext_vaapi.h
Add driver quirks field to VAAPI-specific hwdevice and enum with
members AV_VAAPI_DRIVER_QUIRK_* to represent its values.
2016-11-10 - 638b216d4f - lavu 55.36.100 - pixfmt.h
Add AV_PIX_FMT_GRAY12(LE/BE).
-------- 8< --------- FFmpeg 3.2 was cut here -------- 8< ---------
2016-10-24 - 73ead47 - lavf 57.55.100 - avformat.h
Add AV_DISPOSITION_TIMED_THUMBNAILS
2016-10-24 - a246fef - lavf 57.54.100 - avformat.h
Add avformat_init_output() and AVSTREAM_INIT_IN_ macros
2016-10-22 - f5495c9 - lavu 55.33.100 - avassert.h
Add av_assert0_fpu() / av_assert2_fpu()
2016-10-07 - 3f9137c / 32c8359 - lavc 57.61.100 / 57.24.0 - avcodec.h
Decoders now export the frame timestamp as AVFrame.pts. It was
previously exported as AVFrame.pkt_pts, which is now deprecated.
Note: When decoding, AVFrame.pts uses the stream/packet timebase,
and not the codec timebase.
2016-09-28 - eba0414 - lavu 55.32.100 / 55.16.0 - hwcontext.h hwcontext_qsv.h
Add AV_HWDEVICE_TYPE_QSV and a new installed header with QSV-specific
hwcontext definitions.
2016-09-26 - 32c25f0 - lavc 57.59.100 / 57.23.0 - avcodec.h
AVCodecContext.hw_frames_ctx now may be used by decoders.
2016-09-27 - f0b6f72 - lavf 57.51.100 - avformat.h
Add av_stream_get_codec_timebase()
2016-09-27 - 23c0779 - lswr 2.2.100 - swresample.h
Add swr_build_matrix().
2016-09-23 - 30d3e36 - lavc 57.58.100 - avcodec.h
Add AV_CODEC_CAP_AVOID_PROBING codec capability flag.
2016-09-14 - ae1dd0c - lavf 57.49.100 - avformat.h
Add avformat_transfer_internal_stream_timing_info helper to help with stream
copy.
2016-08-29 - 4493390 - lavfi 6.58.100 - avfilter.h
Add AVFilterContext.nb_threads.
2016-08-15 - c3c4c72 - lavc 57.53.100 - avcodec.h
Add trailing_padding to AVCodecContext to match the corresponding
field in AVCodecParameters.
2016-08-15 - b746ed7 - lavc 57.52.100 - avcodec.h
Add a new API for chained BSF filters and passthrough (null) BSF --
av_bsf_list_alloc(), av_bsf_list_free(), av_bsf_list_append(),
av_bsf_list_append2(), av_bsf_list_finalize(), av_bsf_list_parse_str()
and av_bsf_get_null_filter().
2016-08-04 - 82a33c8 - lavf 57.46.100 - avformat.h
Add av_get_frame_filename2()
2016-07-09 - 775389f / 90f469a - lavc 57.50.100 / 57.20.0 - avcodec.h
Add FF_PROFILE_H264_MULTIVIEW_HIGH and FF_PROFILE_H264_STEREO_HIGH.
2016-06-30 - c1c7e0ab - lavf 57.41.100 - avformat.h
Moved codecpar field from AVStream to the end of the struct, so that
the following private fields are in the same location as in FFmpeg 3.0 (lavf 57.25.100).
@@ -1069,7 +458,7 @@ API changes, most recent first:
Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
dictionary types being set as options.
2014-08-13 - afbd4b7e09 - lavf 56.01.0 - avformat.h
2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
the user when events happen in the file/stream.
@@ -1086,7 +475,7 @@ API changes, most recent first:
2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
Add avio_feof() and deprecate url_feof().
2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
sws_getContext is not going to be removed in the future.
2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h

File diff suppressed because it is too large Load Diff

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

View File

@@ -18,7 +18,7 @@ comma-separated list of filters, whose parameters follow the filter
name after a '='.
@example
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
@end example
Below is a description of the currently available bitstream filters,
@@ -26,101 +26,36 @@ with their parameters, if any.
@section aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
bitstream.
Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
bitstream filter.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
ADTS header and removes the ADTS header.
This filter is required for example when copying an AAC stream from a
raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
to MOV/MP4 files and related formats such as 3GP or M4A. Please note
that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
@section av1_metadata
Modify metadata embedded in an AV1 stream.
@table @option
@item td
Insert or remove temporal delimiter OBUs in all temporal units of the
stream.
@table @samp
@item insert
Insert a TD at the beginning of every TU which does not already have one.
@item remove
Remove the TD from the beginning of every TU which has one.
@end table
@item color_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the color description fields in the stream (see AV1 section 6.4.2).
@item color_range
Set the color range in the stream (see AV1 section 6.4.2; note that
this cannot be set for streams using BT.709 primaries, sRGB transfer
characteristic and identity (RGB) matrix coefficients).
@table @samp
@item tv
Limited range.
@item pc
Full range.
@end table
@item chroma_sample_position
Set the chroma sample location in the stream (see AV1 section 6.4.2).
This can only be set for 4:2:0 streams.
@table @samp
@item vertical
Left position (matching the default in MPEG-2 and H.264).
@item colocated
Top-left position.
@end table
@item tick_rate
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
has a fixed framerate. Ignored if @option{tick_rate} is not also set.
@item delete_padding
Deletes Padding OBUs.
@end table
This is required for example when copying an AAC stream from a raw
ADTS AAC container to a FLV or a MOV/MP4 file.
@section chomp
Remove zero padding at the end of a packet.
@section dca_core
Extract the core from a DCA/DTS stream, dropping extensions such as
DTS-HD.
@section dump_extra
Add extradata to the beginning of the filtered packets except when
said packets already exactly begin with the extradata that is intended
to be added.
Add extradata to the beginning of the filtered packets.
@table @option
@item freq
The additional argument specifies which packets should be filtered.
It accepts the values:
@table @samp
@item a
add extradata to all key packets, but only if @var{local_header} is
set in the @option{flags2} codec context field
@item k
@item keyframe
add extradata to all key packets
@item e
@item all
add extradata to all packets
@end table
@end table
If not specified it is assumed @samp{k}.
@@ -132,160 +67,9 @@ the header stored in extradata to the key packets:
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section eac3_core
@section dca_core
Extract the core from a E-AC-3 stream, dropping extra channels.
@section extract_extradata
Extract the in-band extradata.
Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers,
or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part
of the bitstream containing the coded frames) or "out of band" (e.g. on the
container level). This latter form is called "extradata" in FFmpeg terminology.
This bitstream filter detects the in-band headers and makes them available as
extradata.
@table @option
@item remove
When this option is enabled, the long-term headers are removed from the
bitstream after extraction.
@end table
@section filter_units
Remove units with types in or not in a given set from the stream.
@table @option
@item pass_types
List of unit types or ranges of unit types to pass through while removing
all others. This is specified as a '|'-separated list of unit type values
or ranges of values with '-'.
@item remove_types
Identical to @option{pass_types}, except the units in the given set
removed and all others passed through.
@end table
Extradata is unchanged by this transformation, but note that if the stream
contains inline parameter sets then the output may be unusable if they are
removed.
For example, to remove all non-VCL NAL units from an H.264 stream:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
@end example
To remove all AUDs, SEI and filler from an H.265 stream:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
@end example
@section hapqa_extract
Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
@table @option
@item texture
Specifies the texture to keep.
@table @option
@item color
@item alpha
@end table
@end table
Convert HAPQA to HAPQ
@example
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
@end example
Convert HAPQA to HAPAlphaOnly
@example
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
@end example
@section h264_metadata
Modify metadata embedded in an H.264 stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
@item 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
Set the video format in the stream (see H.264 section E.2.1 and
table E-2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.264 section E.2.1
and tables E-3, E-4 and E-5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.264 section
E.2.1 and figure E-1).
@item tick_rate
Set the tick rate (num_units_in_tick / time_scale) in the VUI
parameters. This is the smallest time unit representable in the
stream, and in many cases represents the field rate of the stream
(double the frame rate).
@item fixed_frame_rate_flag
Set whether the stream has fixed framerate - typically this indicates
that the framerate is exactly half the tick rate, but the exact
meaning is dependent on interlacing and the picture structure (see
H.264 section E.2.1 and table E-6).
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the frame cropping offsets in the SPS. These values will replace
the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled or the stream is interlaced
(see H.264 section 7.4.2.1.1).
@item sei_user_data
Insert a string as SEI unregistered user data. The argument must
be of the form @emph{UUID+string}, where the UUID is as hex digits
possibly separated by hyphens, and the string can be anything.
For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
insert the string ``hello'' associated with the given UUID.
@item delete_filler
Deletes both filler NAL units and filler SEI messages.
@item level
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1
to A-5.
The argument must be the name of a level (for example, @samp{4.2}), a
level_idc value (for example, @samp{42}), or the special name @samp{auto}
indicating that the filter should attempt to guess the level from the
input stream properties.
@end table
Extract DCA core from DTS-HD streams.
@section h264_mp4toannexb
@@ -294,7 +78,7 @@ prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer @code{mpegts}).
transport stream format ("mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts
format with @command{ffmpeg}, you can use the command:
@@ -303,101 +87,6 @@ format with @command{ffmpeg}, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
@section h264_redundant_pps
This applies a specific fixup to some Blu-ray streams which contain
redundant PPSs modifying irrelevant parameters of the stream which
confuse other transformations which require correct extradata.
A new single global PPS is created, and all of the redundant PPSs
within the stream are removed.
@section hevc_metadata
Modify metadata embedded in an HEVC stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio in the stream in the VUI parameters.
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.265 section E.3.1 and
table E.2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.265 section E.3.1
and tables E.3, E.4 and E.5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.265 section
E.3.1 and figure E.1).
@item tick_rate
Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
time_scale). Combined with @option{num_ticks_poc_diff_one}, this can
set a constant framerate in the stream. Note that it is likely to be
overridden by container parameters when the stream is in a container.
@item num_ticks_poc_diff_one
Set poc_proportional_to_timing_flag in VPS and VUI and use this value
to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and
E.3.1). Ignored if @option{tick_rate} is not also set.
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the conformance window cropping offsets in the SPS. These values
will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
@item level
Set the level in the VPS and SPS. See H.265 section A.4 and tables
A.6 and A.7.
The argument must be the name of a level (for example, @samp{5.1}), a
@emph{general_level_idc} value (for example, @samp{153} for level 5.1),
or the special name @samp{auto} indicating that the filter should
attempt to guess the level from the input stream properties.
@end table
@section hevc_mp4toannexb
Convert an HEVC/H.265 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.265
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer @code{mpegts}).
For example to remux an MP4 file containing an HEVC stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
@end example
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw HEVC/H.265 (muxer @code{h265} or
@code{hevc}) output formats.
@section imxdump
Modifies the bitstream to fit in MOV and to be usable by the Final Cut
@@ -448,58 +137,11 @@ exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpegadump
@section mjpega_dump_header
Add an MJPEG A header to the bitstream, to enable decoding by
Quicktime.
@section movsub
@anchor{mov2textsub}
@section mov2textsub
Extract a representable text file from MOV subtitles, stripping the
metadata header from each subtitle packet.
See also the @ref{text2movsub} filter.
@section mp3decomp
Decompress non-standard compressed MP3 audio headers.
@section mpeg2_metadata
Modify metadata embedded in an MPEG-2 stream.
@table @option
@item display_aspect_ratio
Set the display aspect ratio in the stream.
The following fixed values are supported:
@table @option
@item 4/3
@item 16/9
@item 221/100
@end table
Any other value will result in square pixels being signalled instead
(see H.262 section 6.3.3 and table 6-3).
@item frame_rate
Set the frame rate in the stream. This is constructed from a table
of known values combined with a small multiplier and divisor - if
the supplied value is not exactly representable, the nearest
representable value will be used instead (see H.262 section 6.3.3
and table 6-4).
@item video_format
Set the video format in the stream (see H.262 section 6.3.6 and
table 6-6).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.262 section 6.3.6
and tables 6-7, 6-8 and 6-9).
@end table
@section mp3_header_decompress
@section mpeg4_unpack_bframes
@@ -523,220 +165,20 @@ ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@section noise
Damages the contents of packets or simply drops them without damaging the
container. Can be used for fuzzing or testing error resilience/concealment.
Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
Parameters:
@table @option
@item amount
A numeral string, whose value is related to how often output bytes will
be modified. Therefore, values below or equal to 0 are forbidden, and
the lower the more frequent bytes will be modified, with 1 meaning
every byte is modified.
@item dropamount
A numeral string, whose value is related to how often packets will be dropped.
Therefore, values below or equal to 0 are forbidden, and the lower the more
frequent packets will be dropped, with 1 meaning every packet is dropped.
@end table
The following example applies the modification to every byte but does not drop
any packets.
@example
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@end example
@section null
This bitstream filter passes the packets through unchanged.
@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 prores_metadata
Modify color property metadata embedded in prores stream.
@table @option
@item color_primaries
Set the color primaries.
Available values are:
@table @samp
@item auto
Keep the same color primaries property (default).
@item unknown
@item bt709
@item bt470bg
BT601 625
@item smpte170m
BT601 525
@item bt2020
@item smpte431
DCI P3
@item smpte432
P3 D65
@end table
@item transfer_characteristics
Set the color transfer.
Available values are:
@table @samp
@item auto
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
@item matrix_coefficients
Set the matrix coefficient.
Available values are:
@table @samp
@item auto
Keep the same colorspace property (default).
@item unknown
@item bt709
@item smpte170m
BT 601
@item bt2020nc
@end table
@end table
Set Rec709 colorspace for each frame of the file
@example
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
applies the modification to every byte.
@section remove_extra
Remove extradata from packets.
It accepts the following parameter:
@table @option
@item freq
Set which frame types to remove extradata from.
@table @samp
@item k
Remove extradata from non-keyframes only.
@item keyframe
Remove extradata from keyframes only.
@item e, all
Remove extradata from all frames.
@end table
@end table
@anchor{text2movsub}
@section text2movsub
Convert text subtitles to MOV subtitles (as used by the @code{mov_text}
codec) with metadata headers.
See also the @ref{mov2textsub} filter.
@section trace_headers
Log trace output containing all syntax elements in the coded stream
headers (everything above the level of individual coded blocks).
This can be useful for debugging low-level stream issues.
Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending
on the build only a subset of these may be available.
@section truehd_core
Extract the core from a TrueHD stream, dropping ATMOS data.
@section vp9_metadata
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 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
@item bt709
@item smpte170
@item smpte240
@item bt2020
@item rgb
@end table
@item color_range
Set the color range value in the frame header. Note that any value
imposed by the color space will take precedence over this value.
@table @samp
@item tv
@item pc
@end table
@end table
@section vp9_superframe
Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
fixes merging of split/segmented VP9 streams where the alt-ref frame
was split from its visible counterpart.
@section vp9_superframe_split
Split VP9 superframes into single frames.
@section vp9_raw_reorder
Given a VP9 stream with correct timestamps but possibly out of order,
insert additional show-existing-frame packets to correct the ordering.
@c man end BITSTREAM FILTERS

View File

@@ -36,20 +36,15 @@ install
examples
Build all examples located in doc/examples.
checkheaders
Check headers dependencies.
libavformat/output-example
Build the libavformat basic example.
alltools
Build all tools in tools directory.
libswscale/swscale-test
Build the swscale self-test (useful also as an example).
config
Reconfigure the project with the current configuration.
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>

View File

@@ -44,6 +44,12 @@ Use 1/4 pel motion compensation.
Use loop filter.
@item qscale
Use fixed qscale.
@item gmc
Use gmc.
@item mv0
Always try a mb with mv=<0,0>.
@item input_preserved
@item pass1
Use internal 2pass ratecontrol in first pass mode.
@item pass2
@@ -55,11 +61,9 @@ 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.
@item naq
Normalize adaptive quantization.
@item ildct
Use interlaced DCT.
@item low_delay
@@ -80,8 +84,6 @@ Deprecated, use mpegvideo private options instead.
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})
@@ -136,8 +138,7 @@ Set audio sampling rate (in Hz).
Set number of audio channels.
@item cutoff @var{integer} (@emph{encoding,audio})
Set cutoff bandwidth. (Supported only by selected encoders, see
their respective documentation sections.)
Set cutoff bandwidth.
@item frame_size @var{integer} (@emph{encoding,audio})
Set audio frame size.
@@ -473,6 +474,8 @@ rate control
macroblock (MB) type
@item qp
per-block quantization parameter (QP)
@item mv
motion vector
@item dct_coeff
@item green_metadata
@@ -482,12 +485,18 @@ display complexity metadata for the upcoming frame, GoP or for a given duration.
@item startcode
@item pts
@item er
error recognition
@item mmco
memory management control operations (H.264)
@item bugs
@item vis_qp
visualize quantization parameter (QP), lower QP are tinted greener
@item vis_mb_type
visualize block types
@item buffers
picture buffer allocations
@item thread_ops
@@ -496,6 +505,21 @@ threading operations
skip motion compensation
@end table
@item vismv @var{integer} (@emph{decoding,video})
Visualize motion vectors (MVs).
This option is deprecated, see the codecview filter instead.
Possible values:
@table @samp
@item pf
forward predicted MVs of P-frames
@item bf
forward predicted MVs of B-frames
@item bb
backward predicted MVs of B-frames
@end table
@item cmp @var{integer} (@emph{encoding,video})
Set full pel me compare function.
@@ -646,24 +670,6 @@ 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.
@@ -750,6 +756,8 @@ Set context model.
@item slice_flags @var{integer}
@item xvmc_acceleration @var{integer}
@item mbd @var{integer} (@emph{encoding,video})
Set macroblock decision algorithm (high quality mode).
@@ -781,12 +789,14 @@ Set noise reduction.
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
@@ -797,25 +807,11 @@ Place global headers at every keyframe instead of in extradata.
Frame data might be split into multiple chunks.
@item showall
Show all frames before the first keyframe.
@item skiprd
Deprecated, use mpegvideo private options instead.
@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.
@end table
@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
Possible values:
@table @samp
@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.
@end table
@item error @var{integer} (@emph{encoding,video})
@@ -855,8 +851,49 @@ 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
profiles are documented in the relevant encoder documentation.
Possible values:
@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})
@@ -957,9 +994,6 @@ Discard all bidirectional frames.
@item nokey
Discard all frames excepts keyframes.
@item nointra
Discard all frames except I frames.
@item all
Discard all frames.
@end table
@@ -984,6 +1018,10 @@ Set chroma qp offset from luma.
@item trellis @var{integer} (@emph{encoding,audio,video})
Set rate-distortion optimal quantization.
@item sc_factor @var{integer} (@emph{encoding,video})
Set value multiplied by qscale for each frame and added to
scene_change_score.
@item mv0_threshold @var{integer} (@emph{encoding,video})
@item b_sensitivity @var{integer} (@emph{encoding,video})
Adjust sensitivity of b_frame_strategy 1.
@@ -1011,34 +1049,7 @@ Possible values:
@item rc_max_vbv_use @var{float} (@emph{encoding,video})
@item rc_min_vbv_use @var{float} (@emph{encoding,video})
@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item bt709
BT.709
@item bt470m
BT.470 M
@item bt470bg
BT.470 BG
@item smpte170m
SMPTE 170 M
@item smpte240m
SMPTE 240 M
@item film
Film
@item bt2020
BT.2020
@item smpte428
@item smpte428_1
SMPTE ST 428-1
@item smpte431
SMPTE 431-2
@item smpte432
SMPTE 432-1
@item jedec-p22
JEDEC P22
@end table
@item color_trc @var{integer} (@emph{decoding/encoding,video})
Possible values:
@@ -1049,98 +1060,35 @@ BT.709
BT.470 M
@item gamma28
BT.470 BG
@item smpte170m
SMPTE 170 M
@item smpte240m
SMPTE 240 M
@item linear
Linear
SMPTE 170 M
@item log
@item log100
Log
SMPTE 240 M
@item log_sqrt
@item log316
Log square root
Linear
@item iec61966_2_4
@item iec61966-2-4
IEC 61966-2-4
Log
@item bt1361
@item bt1361e
BT.1361
Log square root
@item iec61966_2_1
@item iec61966-2-1
IEC 61966-2-1
@item bt2020_10
IEC 61966-2-4
@item bt2020_10bit
BT.2020 - 10 bit
@item bt2020_12
BT.1361
@item bt2020_12bit
BT.2020 - 12 bit
IEC 61966-2-1
@item smpte2084
SMPTE ST 2084
@item smpte428
BT.2020 - 10 bit
@item smpte428_1
SMPTE ST 428-1
@item arib-std-b67
ARIB STD-B67
BT.2020 - 12 bit
@end table
@item colorspace @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item rgb
RGB
@item bt709
BT.709
@item fcc
FCC
@item bt470bg
BT.470 BG
@item smpte170m
SMPTE 170 M
@item smpte240m
SMPTE 240 M
@item ycocg
YCOCG
@item bt2020nc
@item bt2020_ncl
BT.2020 NCL
@item bt2020c
@item bt2020_cl
BT.2020 CL
@item smpte2085
SMPTE 2085
@end table
@item color_range @var{integer} (@emph{decoding/encoding,video})
If used as input parameter, it serves as a hint to the decoder, which
color_range the input has.
Possible values:
@table @samp
@item tv
@item mpeg
MPEG (219*2^(n-8))
@item pc
@item jpeg
JPEG (2^n-1)
@end table
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item left
@item center
@item topleft
@item top
@item bottomleft
@item bottom
@end table
@item log_level_offset @var{integer}
Set the log level offset.
@@ -1219,37 +1167,23 @@ Interlaced video, top coded first, bottom displayed first
Interlaced video, bottom coded first, top displayed first
@end table
@item skip_alpha @var{bool} (@emph{decoding,video})
@item skip_alpha @var{integer} (@emph{decoding,video})
Set to 1 to disable processing alpha (transparency). This works like the
@samp{gray} flag in the @option{flags} option which skips chroma information
instead of alpha. Default is 0.
@item codec_whitelist @var{list} (@emph{input})
"," separated list of allowed decoders. By default all are allowed.
"," separated List of allowed decoders. By default all are allowed.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example, to separate the fields with newlines and indentation:
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@item max_pixels @var{integer} (@emph{decoding/encoding,video})
Maximum number of pixels per image. This value can be used to avoid out of
memory failures due to large images.
@item apply_cropping @var{bool} (@emph{decoding,video})
Enable cropping if cropping parameters are multiples of the required
alignment for the left and top parameters. If the alignment is not met the
cropping will be partially applied to maintain alignment.
Default is 1 (enabled).
Note: The required alignment depends on if @code{AV_CODEC_FLAG_UNALIGNED} is set and the
CPU. @code{AV_CODEC_FLAG_UNALIGNED} cannot be changed from the command line. Also hardware
decoders will not apply left/top Cropping.
@end table
@c man end CODEC OPTIONS

View File

@@ -25,6 +25,13 @@ enabled decoders.
A description of some of the currently available video decoders
follows.
@section hevc
HEVC / H.265 decoder.
Note: the @option{skip_loop_filter} option has effect only at level
@code{all}.
@section rawvideo
Raw video decoder.
@@ -47,45 +54,6 @@ top-field-first is assumed
@end table
@section libdav1d
dav1d AV1 decoder.
libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
Requires the presence of the libdav1d headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libdav1d}.
@subsection Options
The following options are supported by the libdav1d wrapper.
@table @option
@item framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
@item tilethreads
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
@item filmgrain
Apply film grain to the decoded video if present in the bitstream. Defaults to the
internal default of the library.
@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
@section libdavs2
AVS2-P2/IEEE1857.4 video decoder wrapper.
This decoder allows libavcodec to decode AVS2 streams with davs2 library.
@c man end VIDEO DECODERS
@chapter Audio Decoders
@@ -141,7 +109,7 @@ correctly by using lavc's old buggy lpc logic for decoding.
@section ffwavesynth
Internal wave synthesizer.
Internal wave synthetizer.
This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
@@ -227,31 +195,6 @@ without this library.
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section libaribb24
ARIB STD-B24 caption decoder.
Implements profiles A and C of the ARIB STD-B24 standard.
@subsection libaribb24 Decoder Options
@table @option
@item -aribb24-base-path @var{path}
Sets the base path for the libaribb24 library. This is utilized for reading of
configuration files (for custom unicode conversions), and for dumping of
non-text symbols as images under that location.
Unset by default.
@item -aribb24-skip-ruby-text @var{boolean}
Tells the decoder wrapper to skip text blocks that contain half-height ruby
text.
Enabled by default.
@end table
@section dvbsub
@subsection Options
@@ -287,7 +230,7 @@ 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}.
@@ -312,30 +255,18 @@ configuration. You need to explicitly configure the build with
@table @option
@item txt_page
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.
List of teletext page numbers to decode. You may use the special * string to
match all pages. Pages that do not match the specified list are dropped.
Default value is *.
@item txt_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
Specifies the format of the decoded subtitles.
@table @option
@item bitmap
The default format, you should use this for teletext pages, because certain
graphics and colors cannot be expressed in simple text or even ASS.
@item text
Simple text based output without formatting.
@item ass
Formatted ASS output, subtitle pages and teletext pages are returned in
different styles, subtitle pages are stripped down to text, but an effort is
made to keep the text alignment and the formatting.
@end table
Specifies the format of the decoded subtitles. The teletext decoder is capable
of decoding the teletext pages to bitmaps or to simple text, you should use
"bitmap" for teletext pages, because certain graphics and colors cannot be
expressed in simple text. You might use "text" for teletext based subtitles if
your application can handle simple text based subtitles. Default value is
bitmap.
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
@@ -344,12 +275,11 @@ Y offset of generated bitmaps, default is 0.
Chops leading and trailing spaces and removes empty lines from the generated
text. This option is useful for teletext based subtitles where empty spaces may
be present at the start or at the end of the lines or empty lines may be
present between the subtitle lines because of double-sized teletext characters.
present between the subtitle lines because of double-sized teletext charactes.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
milliseconds. Default value is -1 which means infinity or until the next
subtitle event comes.
miliseconds. Default value is 30000 which is 30 seconds.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque background.

View File

@@ -13,9 +13,8 @@ You can disable all the demuxers using the configure option
the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
with the option @code{--disable-demuxer=@var{DEMUXER}}.
The option @code{-demuxers} of the ff* tools will display the list of
enabled demuxers. Use @code{-formats} to view a combined list of
enabled demuxers and muxers.
The option @code{-formats} of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
@@ -25,6 +24,17 @@ Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
@section applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section apng
Animated Portable Network Graphics demuxer.
@@ -62,7 +72,7 @@ Do not try to resynchronize by looking for a certain optional start code.
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and
demuxes them one after the other, as if all their packets had been muxed
demuxes them one after the other, as if all their packet had been muxed
together.
The timestamps in the files are adjusted so that the first file starts at 0
@@ -97,7 +107,7 @@ 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
appears exactly as is (no extra space or byte-order-mark) on the very first
line of the script.
@item @code{duration @var{dur}}
@@ -233,39 +243,30 @@ file subdir/file-2.wav
@end example
@end itemize
@section dash
Dynamic Adaptive Streaming over HTTP demuxer.
This demuxer presents all AVStreams found in the manifest.
By setting the discard flags on AVStreams the caller can decide
which streams to actually receive.
Each stream mirrors the @code{id} and @code{bandwidth} properties from the
@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
@section flv, live_flv
@section flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
@example
ffmpeg -f flv -i myfile.flv ...
ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
@end example
This demuxer is used to demux FLV files and RTMP network streams.
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@item -flv_ignore_prevtag @var{bool}
Ignore the size of previous tag value.
@item -flv_full_metadata @var{bool}
Output all context of the onMetadata.
@end table
@section libgme
The Game Music Emu library is a collection of video game music file emulators.
See @url{http://code.google.com/p/game-music-emu/} for more information.
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
For very large files, the @option{max_size} option may have to be adjusted.
@section gif
Animated GIF demuxer.
@@ -305,49 +306,6 @@ used to end the output video at the length of the shortest input file,
which in this case is @file{input.mp4} as the GIF in this example loops
infinitely.
@section hls
HLS demuxer
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
It accepts the following options:
@table @option
@item live_start_index
segment index to start live streams at (negative values are from the end).
@item allowed_extensions
',' separated list of file extensions that hls is allowed to access.
@item max_reload
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@item 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.
@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.
@end table
@section image2
Image file demuxer.
@@ -456,17 +414,6 @@ 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
@@ -494,123 +441,9 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section libgme
@section mov/mp4/3gp/QuickTime
The Game Music Emu library is a collection of video game music file emulators.
See @url{https://bitbucket.org/mpyne/game-music-emu/overview} for more information.
It accepts the following options:
@table @option
@item track_index
Set the index of which track to demux. The demuxer can only export one track.
Track indexes start at 0. Default is to pick the first track. Number of tracks
is exported as @var{tracks} metadata entry.
@item sample_rate
Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
@item max_size @emph{(bytes)}
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read.
Default is 50 MiB.
@end table
@section libmodplug
ModPlug based module demuxer
See @url{https://github.com/Konstanty/libmodplug}
It will export one 2-channel 16-bit 44.1 kHz audio stream.
Optionally, a @code{pal8} 16-color video stream can be exported with or without printed metadata.
It accepts the following options:
@table @option
@item noise_reduction
Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
@item reverb_depth
Set amount of reverb. Range 0-100. Default is 0.
@item reverb_delay
Set delay in ms, clamped to 40-250 ms. Default is 0.
@item bass_amount
Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
@item bass_range
Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
@item surround_depth
Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
@item surround_delay
Set surround delay in ms, clamped to 5-40 ms. Default is 0.
@item max_size
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB.
0 removes buffer size limit (not recommended). Default is 5 MiB.
@item video_stream_expr
String which is evaluated using the eval API to assign colors to the generated video stream.
Variables which can be used are @code{x}, @code{y}, @code{w}, @code{h}, @code{t}, @code{speed},
@code{tempo}, @code{order}, @code{pattern} and @code{row}.
@item video_stream
Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
@item video_stream_w
Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_h
Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_ptxt
Print metadata on video stream. Includes @code{speed}, @code{tempo}, @code{order}, @code{pattern},
@code{row} and @code{ts} (time in ms). Can be 1 (on) or 0 (off). Default is 1.
@end table
@section libopenmpt
libopenmpt based module demuxer
See @url{https://lib.openmpt.org/libopenmpt/} for more information.
Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
option.
It accepts the following options:
@table @option
@item subsong
Set the subsong index. This can be either 'all', 'auto', or the index of the
subsong. Subsong indexes start at 0. The default is 'auto'.
The default value is to let libopenmpt choose.
@item layout
Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
The default value is STEREO.
@item sample_rate
Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section mov/mp4/3gp
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).
Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v
@subsection Options
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
@table @option
@@ -621,73 +454,10 @@ 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 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).
@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.
@@ -698,9 +468,6 @@ This demuxer accepts the following options:
Set size limit for looking up a new synchronization. Default value is
65536.
@item skip_unknown_pmt
Skip PMTs for programs not defined in the PAT. Default value is 0.
@item fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps calculated
from the PCR of the first program which the teletext stream is part of and is
@@ -715,10 +482,6 @@ Show the detected raw packet size, cannot be set by the user.
Scan and combine all PMTs. The value is an integer with value from -1
to 1 (-1 means automatic setting, 1 means enabled, 0 means
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.
@end table
@section mpjpeg
@@ -816,20 +579,4 @@ Example: convert the captions to a format most players understand:
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@section vapoursynth
Vapoursynth wrapper.
Due to security concerns, Vapoursynth scripts will not
be autodetected so the input format has to be forced. For ff* CLI tools,
add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
This demuxer accepts the following option:
@table @option
@item max_script_size
The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of scripts that can be read.
Default is 1 MiB.
@end table
@c man end DEMUXERS

View File

@@ -10,7 +10,9 @@
@contents
@chapter Notes for external developers
@chapter Developers Guide
@section Notes for external developers
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
@@ -28,13 +30,15 @@ For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
@chapter Contributing
@section Contributing
There are 2 ways by which code gets into FFmpeg:
There are 3 ways by which code gets into FFmpeg.
@itemize @bullet
@item Submitting patches to the ffmpeg-devel mailing list.
@item Submitting patches to the main developer mailing list.
See @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
@@ -43,9 +47,9 @@ The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@chapter Coding Rules
@section Coding Rules
@section Code formatting conventions
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@@ -70,7 +74,7 @@ The presentation is one inspired by 'indent -i4 -kr -nut'.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@section Comments
@subsection Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
@@ -110,7 +114,7 @@ int myfunc(int my_parameter)
...
@end example
@section C language features
@subsection C language features
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@@ -127,17 +131,6 @@ 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
Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
@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
@@ -162,7 +155,7 @@ mixing statements and declarations;
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@section Naming conventions
@subsection Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
@@ -186,7 +179,7 @@ e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_report_missing_feature}.
@samp{avpriv_aac_parse_header}.
@item
Each library has its own prefix for public symbols, in addition to the
@@ -206,7 +199,7 @@ letter as they are reserved by the C standard. Names starting with @code{_}
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@section Miscellaneous conventions
@subsection Miscellaneous conventions
@itemize @bullet
@item
@@ -218,7 +211,7 @@ Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@end itemize
@section Editor configuration
@subsection Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
@@ -251,10 +244,10 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
(setq c-default-style "ffmpeg")
@end lisp
@chapter Development Policy
@section Development Policy
@section Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
@enumerate
@item
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
including an "or any later version" clause, or, if you prefer
@@ -267,15 +260,15 @@ preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@subheading You must not commit code which breaks FFmpeg!
This means unfinished code which is enabled and breaks compilation,
or compiles but does not work/breaks the regression tests. Code which
is unfinished but disabled may be permitted under-circumstances, like
missing samples or an implementation with a small subset of features.
Always check the mailing list for any reviewers with issues and test
FATE before you push.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@subheading Keep the main commit message short with an extended description below.
@item
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
@@ -283,24 +276,30 @@ If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
@subheading Testing must be adequate but not excessive.
If it works for you, others, and passes FATE then it should be OK to commit
it, provided it fits the other committing criteria. You should not worry about
over-testing things. If your code has problems (portability, triggers
compiler bugs, unusual environment etc) they will be reported and eventually
fixed.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@subheading Do not commit unrelated changes together.
They should be split them into self-contained pieces. Also do not forget
that if part B depends on part A, but A does not depend on B, then A can
and should be committed first and separate from B. Keeping changes well
split into self-contained parts makes reviewing and understanding them on
the commit log mailing list easier. This also helps in case of debugging
later on.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@subheading Ask before you change the build system (configure, etc).
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
@@ -309,7 +308,7 @@ the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@subheading Cosmetic changes should be kept in separate patches.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
@@ -323,7 +322,7 @@ NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@subheading Commit messages should always be filled out properly.
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
@@ -335,31 +334,47 @@ area changed: Short 1 line description
details describing what and why and giving references.
@end example
@subheading Credit the author of the patch.
@item
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@subheading Complex patches should refer to discussion surrounding them.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@subheading Always wait long enough before pushing changes
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel. If no one answers within a reasonable
time-frame (12h for build failures and security fixes, 3 days small changes,
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@section Code
@subheading API/ABI changes should be discussed before they are made.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove widely used functionality or features (redundant code can be removed).
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@subheading Remember to check if you need to bump versions for libav*.
Depending on the change, you may need to change the version integer.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
@@ -369,7 +384,7 @@ Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@subheading Warnings for correct code may be disabled if there is no other option.
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
@@ -378,54 +393,17 @@ If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@subheading Check untrusted input properly.
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
It is important to be subscribed to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Almost any non-trivial patch is to be sent there for review.
Other developers may have comments about your contribution. We expect you see
those comments, and to improve it if requested. (N.B. Experienced committers
have other channels, and may sometimes skip review for trivial fixes.) Also,
discussion here about bug fixes and FFmpeg improvements by other developers may
be helpful information for you. Finally, by being a list subscriber, your
contribution will be posted immediately to the list, without the moderation
hold which messages from non-subscribers experience.
However, it is more important to the project that we receive your patch than
that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
want to subscribe and discuss the patch, then please do send it to the list
anyway.
@subheading Subscribe to the ffmpeg-cvslog mailing list.
Diffs of all commits are sent to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
mailing list. Some developers read this list to review all code base changes
from all sources. Subscribing to this list is not mandatory.
@subheading Keep the documentation up to date.
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@subheading Important discussions should be accessible to all.
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@subheading Check your entries in MAINTAINERS.
@item
Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help in
finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
@chapter Code of conduct
@section Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
@@ -455,7 +433,7 @@ Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@chapter Submitting patches
@section Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
@@ -488,11 +466,7 @@ Patches should be posted to the
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission. Also ensure the correct mime type is used
(text/x-diff or text/x-patch or at least text/plain) and that only one
patch is inline or attached per mail.
You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
likely was wrong.
transmission.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
@@ -504,7 +478,7 @@ Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@chapter New codecs or formats checklist
@section New codecs or formats checklist
@enumerate
@item
@@ -556,7 +530,7 @@ Did you make sure it compiles standalone, i.e. with
@end enumerate
@chapter Patch submission checklist
@section patch submission checklist
@enumerate
@item
@@ -566,9 +540,9 @@ Does @code{make fate} pass with the patch applied?
Was the patch generated with git format-patch or send-email?
@item
Did you sign-off your patch? (@code{git commit -s})
See @uref{https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst, Sign your work} for the meaning
of @dfn{sign-off}.
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
@@ -625,7 +599,7 @@ 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?
@@ -669,7 +643,7 @@ Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
@end enumerate
@chapter Patch review process
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
@@ -700,7 +674,7 @@ to be reviewed, please consider helping to review other patches, that is a great
way to get everyone's patches reviewed sooner.
@anchor{Regression tests}
@chapter Regression tests
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
@@ -711,7 +685,7 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
this case, the reference results of the regression tests shall be modified
accordingly].
@section Adding files to the fate-suite dataset
@subsection Adding files to the fate-suite dataset
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
@@ -722,7 +696,7 @@ Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@section Visualizing Test Coverage
@subsection Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
@@ -749,7 +723,7 @@ You can use the command @code{make lcov-reset} to reset the coverage
measurements. You will need to rerun @code{make lcov} after running a
new test.
@section Using Valgrind
@subsection Using Valgrind
The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
@@ -763,7 +737,7 @@ In case you need finer control over how valgrind is invoked, use the
your configure line instead.
@anchor{Release process}
@chapter Release process
@section Release process
FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
@@ -795,7 +769,7 @@ adjustments to the symbol versioning file. Please discuss such changes
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
@anchor{Criteria for Point Releases}
@section Criteria for Point Releases
@subsection Criteria for Point Releases
Changes that match the following criteria are valid candidates for
inclusion into a point release:
@@ -819,7 +793,7 @@ point releases of the same release branch.
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@section Release Checklist
@subsection Release Checklist
The release process involves the following steps:

File diff suppressed because it is too large Load Diff

View File

@@ -1,24 +1,16 @@
/avio_list_dir
/avio_dir_cmd
/avio_reading
/decode_audio
/decode_video
/decoding_encoding
/demuxing_decoding
/encode_audio
/encode_video
/extract_mvs
/filter_audio
/filtering_audio
/filtering_video
/http_multiclient
/hw_decode
/metadata
/muxing
/pc-uninstalled
/qsvdec
/remuxing
/resampling_audio
/scaling_video
/transcode_aac
/transcoding
/vaapi_encode
/vaapi_transcode

View File

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

View File

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

View File

@@ -102,15 +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"
"API example program to show how to list files in directory "
"accessed through AVIOContext.\n", program_name);
fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
"API example program to show how to manipulate resources "
"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);
@@ -120,9 +143,36 @@ int main(int argc, char *argv[])
return 1;
}
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
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();

View File

@@ -44,8 +44,6 @@ static int read_packet(void *opaque, uint8_t *buf, int buf_size)
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
if (!buf_size)
return AVERROR_EOF;
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
@@ -74,6 +72,9 @@ int main(int argc, char *argv[])
}
input_filename = argv[1];
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)
@@ -117,12 +118,11 @@ int main(int argc, char *argv[])
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx)
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
avio_context_free(&avio_ctx);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {

View File

@@ -1,236 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* audio decoding with libavcodec API example
*
* @example decode_audio.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavutil/frame.h>
#include <libavutil/mem.h>
#include <libavcodec/avcodec.h>
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static 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)
{
int i, ch;
int ret, data_size;
/* send the packet with the compressed data to the decoder */
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting the packet to the decoder\n");
exit(1);
}
/* read all the output frames (in general there may be any number of them */
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
exit(1);
}
data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i = 0; i < frame->nb_samples; i++)
for (ch = 0; ch < dec_ctx->channels; ch++)
fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
}
}
int main(int argc, char **argv)
{
const char *outfilename, *filename;
const AVCodec *codec;
AVCodecContext *c= NULL;
AVCodecParserContext *parser = NULL;
int len, ret;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t *data;
size_t data_size;
AVPacket *pkt;
AVFrame *decoded_frame = NULL;
enum AVSampleFormat sfmt;
int n_channels = 0;
const char *fmt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(0);
}
filename = argv[1];
outfilename = argv[2];
pkt = av_packet_alloc();
/* find the MPEG audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
parser = av_parser_init(codec->id);
if (!parser) {
fprintf(stderr, "Parser not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
data = inbuf;
data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (data_size > 0) {
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
}
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
data, data_size,
AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
fprintf(stderr, "Error while parsing\n");
exit(1);
}
data += ret;
data_size -= ret;
if (pkt->size)
decode(c, pkt, decoded_frame, outfile);
if (data_size < AUDIO_REFILL_THRESH) {
memmove(inbuf, data, data_size);
data = inbuf;
len = fread(data + data_size, 1,
AUDIO_INBUF_SIZE - data_size, f);
if (len > 0)
data_size += len;
}
}
/* flush the decoder */
pkt->data = NULL;
pkt->size = 0;
decode(c, pkt, decoded_frame, outfile);
/* 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->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);
avcodec_free_context(&c);
av_parser_close(parser);
av_frame_free(&decoded_frame);
av_packet_free(&pkt);
return 0;
}

View File

@@ -1,187 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* video decoding with libavcodec API example
*
* @example decode_video.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#define INBUF_SIZE 4096
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
fclose(f);
}
static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
const char *filename)
{
char buf[1024];
int ret;
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error sending a packet for decoding\n");
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
exit(1);
}
printf("saving frame %3d\n", dec_ctx->frame_number);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
pgm_save(frame->data[0], frame->linesize[0],
frame->width, frame->height, buf);
}
}
int main(int argc, char **argv)
{
const char *filename, *outfilename;
const AVCodec *codec;
AVCodecParserContext *parser;
AVCodecContext *c= NULL;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t *data;
size_t data_size;
int ret;
AVPacket *pkt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n"
"And check your input file is encoded by mpeg1video please.\n", argv[0]);
exit(0);
}
filename = argv[1];
outfilename = argv[2];
pkt = av_packet_alloc();
if (!pkt)
exit(1);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
/* find the MPEG-1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
parser = av_parser_init(codec->id);
if (!parser) {
fprintf(stderr, "parser not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
while (!feof(f)) {
/* read raw data from the input file */
data_size = fread(inbuf, 1, INBUF_SIZE, f);
if (!data_size)
break;
/* use the parser to split the data into frames */
data = inbuf;
while (data_size > 0) {
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
fprintf(stderr, "Error while parsing\n");
exit(1);
}
data += ret;
data_size -= ret;
if (pkt->size)
decode(c, frame, pkt, outfilename);
}
}
/* flush the decoder */
decode(c, frame, NULL, outfilename);
fclose(f);
av_parser_close(parser);
avcodec_free_context(&c);
av_frame_free(&frame);
av_packet_free(&pkt);
return 0;
}

View File

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

View File

@@ -55,93 +55,96 @@ static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int output_video_frame(AVFrame *frame)
{
if (frame->width != width || frame->height != height ||
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;
}
/* 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
* needs. Look for the use of refcount in this example to see what are the
* differences of API usage between them. */
static int refcount = 0;
printf("video_frame n:%d coded_n:%d\n",
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);
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)
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
// submit the packet to the decoder
ret = avcodec_send_packet(dec, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
return ret;
}
*got_frame = 0;
// get all the available frames from the decoder
while (ret >= 0) {
ret = avcodec_receive_frame(dec, frame);
if (pkt.stream_index == video_stream_idx) {
/* decode video 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
// 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));
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
// write the frame data to output file
if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
ret = output_video_frame(frame);
else
ret = output_audio_frame(frame);
if (*got_frame) {
av_frame_unref(frame);
if (ret < 0)
if (frame->width != width || frame->height != height ||
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 pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number,
av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
/* 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,
@@ -184,7 +187,8 @@ static int open_codec_context(int *stream_idx,
return ret;
}
/* Init the decoders */
/* Init the decoders, with or without reference counting */
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));
@@ -227,21 +231,31 @@ 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) {
fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
if (argc != 4 && argc != 5) {
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",
argv[0]);
"audio frames to a rawaudio file named audio_output_file.\n\n"
"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];
/* register all formats and codecs */
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
@@ -315,22 +329,23 @@ int main (int argc, char **argv)
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
// check if the packet belongs to a stream we are interested in, otherwise
// skip it
if (pkt.stream_index == video_stream_idx)
ret = decode_packet(video_dec_ctx, &pkt);
else if (pkt.stream_index == audio_stream_idx)
ret = decode_packet(audio_dec_ctx, &pkt);
av_packet_unref(&pkt);
if (ret < 0)
break;
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_packet_unref(&orig_pkt);
}
/* flush the decoders */
if (video_dec_ctx)
decode_packet(video_dec_ctx, NULL);
if (audio_dec_ctx)
decode_packet(audio_dec_ctx, NULL);
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");

View File

@@ -1,238 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* audio encoding with libavcodec API example.
*
* @example encode_audio.c
*/
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(const AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
best_samplerate = *p;
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
FILE *output)
{
int ret;
/* send the frame for encoding */
ret = avcodec_send_frame(ctx, frame);
if (ret < 0) {
fprintf(stderr, "Error sending the frame to the encoder\n");
exit(1);
}
/* read all the available output packets (in general there may be any
* number of them */
while (ret >= 0) {
ret = avcodec_receive_packet(ctx, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
fwrite(pkt->data, 1, pkt->size, output);
av_packet_unref(pkt);
}
}
int main(int argc, char **argv)
{
const char *filename;
const AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket *pkt;
int i, j, k, ret;
FILE *f;
uint16_t *samples;
float t, tincr;
if (argc <= 1) {
fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
return 0;
}
filename = argv[1];
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* packet for holding encoded output */
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "could not allocate the packet\n");
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* allocate the data buffers */
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate audio data buffers\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
/* make sure the frame is writable -- makes a copy if the encoder
* kept a reference internally */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
samples = (uint16_t*)frame->data[0];
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
encode(c, frame, pkt, f);
}
/* flush the encoder */
encode(c, NULL, pkt, f);
fclose(f);
av_frame_free(&frame);
av_packet_free(&pkt);
avcodec_free_context(&c);
return 0;
}

View File

@@ -1,198 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* video encoding with libavcodec API example
*
* @example encode_video.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include <libavutil/imgutils.h>
static void encode(AVCodecContext *enc_ctx, AVFrame *frame, AVPacket *pkt,
FILE *outfile)
{
int ret;
/* send the frame to the encoder */
if (frame)
printf("Send frame %3"PRId64"\n", frame->pts);
ret = avcodec_send_frame(enc_ctx, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame for encoding\n");
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_packet(enc_ctx, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during encoding\n");
exit(1);
}
printf("Write packet %3"PRId64" (size=%5d)\n", pkt->pts, pkt->size);
fwrite(pkt->data, 1, pkt->size, outfile);
av_packet_unref(pkt);
}
}
int main(int argc, char **argv)
{
const char *filename, *codec_name;
const AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y;
FILE *f;
AVFrame *frame;
AVPacket *pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
if (argc <= 2) {
fprintf(stderr, "Usage: %s <output file> <codec name>\n", argv[0]);
exit(0);
}
filename = argv[1];
codec_name = argv[2];
/* find the mpeg1video encoder */
codec = avcodec_find_encoder_by_name(codec_name);
if (!codec) {
fprintf(stderr, "Codec '%s' not found\n", codec_name);
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
pkt = av_packet_alloc();
if (!pkt)
exit(1);
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1, 25};
c->framerate = (AVRational){25, 1};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec->id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate the video frame data\n");
exit(1);
}
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
fflush(stdout);
/* make sure the frame data is writable */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < c->height/2; y++) {
for (x = 0; x < c->width/2; x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
encode(c, frame, pkt, f);
}
/* flush the encoder */
encode(c, NULL, pkt, f);
/* add sequence end code to have a real MPEG file */
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);
av_frame_free(&frame);
av_packet_free(&pkt);
return 0;
}

View File

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

View File

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

View File

@@ -32,6 +32,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -68,12 +69,8 @@ static int open_input_file(const char *filename)
return ret;
}
audio_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -88,8 +85,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
@@ -199,7 +196,7 @@ end:
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
@@ -214,9 +211,10 @@ static void print_frame(const AVFrame *frame)
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVPacket packet0, packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
@@ -227,58 +225,63 @@ int main(int argc, char **argv)
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
packet0.data = NULL;
packet.data = NULL;
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (!packet0.data) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
packet0 = packet;
}
if (packet.stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
packet.size -= ret;
packet.data += ret;
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
if (ret >= 0) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
if (packet.size <= 0)
av_packet_unref(&packet0);
} else {
/* discard non-wanted packets */
av_packet_unref(&packet0);
}
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);

View File

@@ -29,11 +29,10 @@
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -73,12 +72,8 @@ static int open_input_file(const char *filename)
return ret;
}
video_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -93,8 +88,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
@@ -211,20 +206,21 @@ int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame;
AVFrame *filt_frame;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
@@ -237,22 +233,15 @@ int main(int argc, char **argv)
break;
if (packet.stream_index == video_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
frame->pts = frame->best_effort_timestamp;
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
@@ -277,7 +266,7 @@ int main(int argc, char **argv)
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);

View File

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

View File

@@ -1,252 +0,0 @@
/*
* Copyright (c) 2017 Jun Zhao
* Copyright (c) 2017 Kaixuan Liu
*
* HW Acceleration API (video decoding) decode 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
* 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
* HW-Accelerated decoding example.
*
* @example hw_decode.c
* This example shows how to do HW-accelerated decoding with output
* frames from the HW video surfaces.
*/
#include <stdio.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
#include <libavutil/opt.h>
#include <libavutil/avassert.h>
#include <libavutil/imgutils.h>
static AVBufferRef *hw_device_ctx = NULL;
static enum AVPixelFormat hw_pix_fmt;
static FILE *output_file = NULL;
static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
{
int err = 0;
if ((err = av_hwdevice_ctx_create(&hw_device_ctx, type,
NULL, NULL, 0)) < 0) {
fprintf(stderr, "Failed to create specified HW device.\n");
return err;
}
ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
return err;
}
static enum AVPixelFormat get_hw_format(AVCodecContext *ctx,
const enum AVPixelFormat *pix_fmts)
{
const enum AVPixelFormat *p;
for (p = pix_fmts; *p != -1; p++) {
if (*p == hw_pix_fmt)
return *p;
}
fprintf(stderr, "Failed to get HW surface format.\n");
return AV_PIX_FMT_NONE;
}
static int decode_write(AVCodecContext *avctx, AVPacket *packet)
{
AVFrame *frame = NULL, *sw_frame = NULL;
AVFrame *tmp_frame = NULL;
uint8_t *buffer = NULL;
int size;
int ret = 0;
ret = avcodec_send_packet(avctx, packet);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
while (1) {
if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
fprintf(stderr, "Can not alloc frame\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ret = avcodec_receive_frame(avctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
av_frame_free(&sw_frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding\n");
goto fail;
}
if (frame->format == hw_pix_fmt) {
/* retrieve data from GPU to CPU */
if ((ret = av_hwframe_transfer_data(sw_frame, frame, 0)) < 0) {
fprintf(stderr, "Error transferring the data to system memory\n");
goto fail;
}
tmp_frame = sw_frame;
} else
tmp_frame = frame;
size = av_image_get_buffer_size(tmp_frame->format, tmp_frame->width,
tmp_frame->height, 1);
buffer = av_malloc(size);
if (!buffer) {
fprintf(stderr, "Can not alloc buffer\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ret = av_image_copy_to_buffer(buffer, size,
(const uint8_t * const *)tmp_frame->data,
(const int *)tmp_frame->linesize, tmp_frame->format,
tmp_frame->width, tmp_frame->height, 1);
if (ret < 0) {
fprintf(stderr, "Can not copy image to buffer\n");
goto fail;
}
if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
fprintf(stderr, "Failed to dump raw data.\n");
goto fail;
}
fail:
av_frame_free(&frame);
av_frame_free(&sw_frame);
av_freep(&buffer);
if (ret < 0)
return ret;
}
}
int main(int argc, char *argv[])
{
AVFormatContext *input_ctx = NULL;
int video_stream, ret;
AVStream *video = NULL;
AVCodecContext *decoder_ctx = NULL;
AVCodec *decoder = NULL;
AVPacket packet;
enum AVHWDeviceType type;
int i;
if (argc < 4) {
fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
return -1;
}
type = av_hwdevice_find_type_by_name(argv[1]);
if (type == AV_HWDEVICE_TYPE_NONE) {
fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
fprintf(stderr, "Available device types:");
while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
fprintf(stderr, "\n");
return -1;
}
/* open the input file */
if (avformat_open_input(&input_ctx, argv[2], NULL, NULL) != 0) {
fprintf(stderr, "Cannot open input file '%s'\n", argv[2]);
return -1;
}
if (avformat_find_stream_info(input_ctx, NULL) < 0) {
fprintf(stderr, "Cannot find input stream information.\n");
return -1;
}
/* find the video stream information */
ret = av_find_best_stream(input_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file\n");
return -1;
}
video_stream = ret;
for (i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
if (!config) {
fprintf(stderr, "Decoder %s does not support device type %s.\n",
decoder->name, av_hwdevice_get_type_name(type));
return -1;
}
if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
config->device_type == type) {
hw_pix_fmt = config->pix_fmt;
break;
}
}
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
video = input_ctx->streams[video_stream];
if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
return -1;
decoder_ctx->get_format = get_hw_format;
if (hw_decoder_init(decoder_ctx, type) < 0)
return -1;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0) {
fprintf(stderr, "Failed to open codec for stream #%u\n", video_stream);
return -1;
}
/* open the file to dump raw data */
output_file = fopen(argv[3], "w+");
/* actual decoding and dump the raw data */
while (ret >= 0) {
if ((ret = av_read_frame(input_ctx, &packet)) < 0)
break;
if (video_stream == packet.stream_index)
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(&packet);
}
/* flush the decoder */
packet.data = NULL;
packet.size = 0;
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(&packet);
if (output_file)
fclose(output_file);
avcodec_free_context(&decoder_ctx);
avformat_close_input(&input_ctx);
av_buffer_unref(&hw_device_ctx);
return 0;
}

View File

@@ -44,14 +44,10 @@ int main (int argc, char **argv)
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);

View File

@@ -78,45 +78,15 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
AVStream *st, AVFrame *frame)
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, 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
ret = avcodec_send_frame(c, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame to the encoder: %s\n",
av_err2str(ret));
exit(1);
}
while (ret >= 0) {
AVPacket pkt = { 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);
av_packet_unref(&pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
exit(1);
}
}
return ret == AVERROR_EOF ? 1 : 0;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* Add an output stream. */
@@ -315,7 +285,7 @@ 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++) {
@@ -339,10 +309,13 @@ 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);
@@ -362,21 +335,36 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
return write_frame(oc, c, ost->st, frame);
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;
}
/**************************************************************/
@@ -396,7 +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);
@@ -452,7 +440,15 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i;
int x, y, i, ret;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(pict);
if (ret < 0)
exit(1);
i = frame_index;
@@ -476,14 +472,9 @@ 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
* internally; make sure we do not overwrite it here */
if (av_frame_make_writable(ost->frame) < 0)
exit(1);
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
@@ -500,9 +491,9 @@ static AVFrame *get_video_frame(OutputStream *ost)
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
ost->frame->linesize);
sws_scale(ost->sws_ctx,
(const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
0, c->height, ost->frame->data, ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
@@ -518,8 +509,37 @@ 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));
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)
@@ -547,6 +567,9 @@ int main(int argc, char **argv)
AVDictionary *opt = NULL;
int i;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"

View File

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

View File

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

View File

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

View File

@@ -30,6 +30,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -44,12 +45,6 @@ typedef struct FilteringContext {
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
} StreamContext;
static StreamContext *stream_ctx;
static int open_input_file(const char *filename)
{
int ret;
@@ -66,42 +61,22 @@ static int open_input_file(const char *filename)
return ret;
}
stream_ctx = av_mallocz_array(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
if (!stream_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream = ifmt_ctx->streams[i];
AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
AVStream *stream;
AVCodecContext *codec_ctx;
if (!dec) {
av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
return AVERROR_DECODER_NOT_FOUND;
}
codec_ctx = avcodec_alloc_context3(dec);
if (!codec_ctx) {
av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
return AVERROR(ENOMEM);
}
ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
"for stream #%u\n", i);
return ret;
}
stream = ifmt_ctx->streams[i];
codec_ctx = stream->codec;
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
/* Open decoder */
ret = avcodec_open2(codec_ctx, dec, NULL);
ret = avcodec_open2(codec_ctx,
avcodec_find_decoder(codec_ctx->codec_id), NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
stream_ctx[i].dec_ctx = codec_ctx;
}
av_dump_format(ifmt_ctx, 0, filename, 0);
@@ -133,7 +108,8 @@ static int open_output_file(const char *filename)
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = stream_ctx[i].dec_ctx;
dec_ctx = in_stream->codec;
enc_ctx = out_stream->codec;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
@@ -143,11 +119,6 @@ static int open_output_file(const char *filename)
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
enc_ctx = avcodec_alloc_context3(encoder);
if (!enc_ctx) {
av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
return AVERROR(ENOMEM);
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
@@ -162,7 +133,7 @@ static int open_output_file(const char *filename)
else
enc_ctx->pix_fmt = dec_ctx->pix_fmt;
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
enc_ctx->time_base = dec_ctx->time_base;
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
@@ -172,36 +143,28 @@ static int open_output_file(const char *filename)
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
return ret;
}
out_stream->time_base = enc_ctx->time_base;
stream_ctx[i].enc_ctx = enc_ctx;
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
ifmt_ctx->streams[i]->codec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
return ret;
}
out_stream->time_base = in_stream->time_base;
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -228,8 +191,8 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = NULL;
const AVFilter *buffersink = NULL;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -385,17 +348,17 @@ static int init_filters(void)
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
stream_ctx[i].enc_ctx, filter_spec);
ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
ofmt_ctx->streams[i]->codec, filter_spec);
if (ret)
return ret;
}
@@ -407,7 +370,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
(ifmt_ctx->streams[stream_index]->codec->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
@@ -418,7 +381,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
@@ -429,7 +392,7 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
stream_ctx[stream_index].enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
@@ -487,7 +450,7 @@ static int flush_encoder(unsigned int stream_index)
int ret;
int got_frame;
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
@@ -518,6 +481,9 @@ int main(int argc, char **argv)
return 1;
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
@@ -530,7 +496,7 @@ int main(int argc, char **argv)
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
@@ -543,10 +509,10 @@ int main(int argc, char **argv)
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
stream_ctx[stream_index].dec_ctx->time_base);
ifmt_ctx->streams[stream_index]->codec->time_base);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
@@ -555,7 +521,7 @@ int main(int argc, char **argv)
}
if (got_frame) {
frame->pts = frame->best_effort_timestamp;
frame->pts = av_frame_get_best_effort_timestamp(frame);
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
@@ -600,14 +566,13 @@ end:
av_packet_unref(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_free_context(&stream_ctx[i].dec_ctx);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
avcodec_free_context(&stream_ctx[i].enc_ctx);
avcodec_close(ifmt_ctx->streams[i]->codec);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
avcodec_close(ofmt_ctx->streams[i]->codec);
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
}
av_free(filter_ctx);
av_free(stream_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);

View File

@@ -1,224 +0,0 @@
/*
* 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
* 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 VAAPI-accelerated encoding example.
*
* @example vaapi_encode.c
* This example shows how to do VAAPI-accelerated encoding. now only support NV12
* raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
*
*/
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <libavcodec/avcodec.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
static int width, height;
static AVBufferRef *hw_device_ctx = NULL;
static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
{
AVBufferRef *hw_frames_ref;
AVHWFramesContext *frames_ctx = NULL;
int err = 0;
if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
fprintf(stderr, "Failed to create VAAPI frame context.\n");
return -1;
}
frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
frames_ctx->format = AV_PIX_FMT_VAAPI;
frames_ctx->sw_format = AV_PIX_FMT_NV12;
frames_ctx->width = width;
frames_ctx->height = height;
frames_ctx->initial_pool_size = 20;
if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
fprintf(stderr, "Failed to initialize VAAPI frame context."
"Error code: %s\n",av_err2str(err));
av_buffer_unref(&hw_frames_ref);
return err;
}
ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
if (!ctx->hw_frames_ctx)
err = AVERROR(ENOMEM);
av_buffer_unref(&hw_frames_ref);
return err;
}
static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
{
int ret = 0;
AVPacket enc_pkt;
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
fprintf(stderr, "Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(avctx, &enc_pkt);
if (ret)
break;
enc_pkt.stream_index = 0;
ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
av_packet_unref(&enc_pkt);
}
end:
ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
return ret;
}
int main(int argc, char *argv[])
{
int size, err;
FILE *fin = NULL, *fout = NULL;
AVFrame *sw_frame = NULL, *hw_frame = NULL;
AVCodecContext *avctx = NULL;
AVCodec *codec = NULL;
const char *enc_name = "h264_vaapi";
if (argc < 5) {
fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
return -1;
}
width = atoi(argv[1]);
height = atoi(argv[2]);
size = width * height;
if (!(fin = fopen(argv[3], "r"))) {
fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
return -1;
}
if (!(fout = fopen(argv[4], "w+b"))) {
fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
err = -1;
goto close;
}
err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
NULL, NULL, 0);
if (err < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
goto close;
}
if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
fprintf(stderr, "Could not find encoder.\n");
err = -1;
goto close;
}
if (!(avctx = avcodec_alloc_context3(codec))) {
err = AVERROR(ENOMEM);
goto close;
}
avctx->width = width;
avctx->height = height;
avctx->time_base = (AVRational){1, 25};
avctx->framerate = (AVRational){25, 1};
avctx->sample_aspect_ratio = (AVRational){1, 1};
avctx->pix_fmt = AV_PIX_FMT_VAAPI;
/* set hw_frames_ctx for encoder's AVCodecContext */
if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
fprintf(stderr, "Failed to set hwframe context.\n");
goto close;
}
if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
goto close;
}
while (1) {
if (!(sw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
/* read data into software frame, and transfer them into hw frame */
sw_frame->width = width;
sw_frame->height = height;
sw_frame->format = AV_PIX_FMT_NV12;
if ((err = av_frame_get_buffer(sw_frame, 0)) < 0)
goto close;
if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
break;
if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
break;
if (!(hw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
fprintf(stderr, "Error code: %s.\n", av_err2str(err));
goto close;
}
if (!hw_frame->hw_frames_ctx) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
fprintf(stderr, "Error while transferring frame data to surface."
"Error code: %s.\n", av_err2str(err));
goto close;
}
if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
fprintf(stderr, "Failed to encode.\n");
goto close;
}
av_frame_free(&hw_frame);
av_frame_free(&sw_frame);
}
/* flush encoder */
err = encode_write(avctx, NULL, fout);
if (err == AVERROR_EOF)
err = 0;
close:
if (fin)
fclose(fin);
if (fout)
fclose(fout);
av_frame_free(&sw_frame);
av_frame_free(&hw_frame);
avcodec_free_context(&avctx);
av_buffer_unref(&hw_device_ctx);
return err;
}

View File

@@ -1,306 +0,0 @@
/*
* 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
* 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 VAAPI-accelerated transcoding example.
*
* @example vaapi_transcode.c
* This example shows how to do VAAPI-accelerated transcoding.
* Usage: vaapi_transcode input_stream codec output_stream
* e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
* - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
*/
#include <stdio.h>
#include <errno.h>
#include <libavutil/hwcontext.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
static AVBufferRef *hw_device_ctx = NULL;
static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
static int video_stream = -1;
static AVStream *ost;
static int initialized = 0;
static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
const enum AVPixelFormat *pix_fmts)
{
const enum AVPixelFormat *p;
for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
if (*p == AV_PIX_FMT_VAAPI)
return *p;
}
fprintf(stderr, "Unable to decode this file using VA-API.\n");
return AV_PIX_FMT_NONE;
}
static int open_input_file(const char *filename)
{
int ret;
AVCodec *decoder = NULL;
AVStream *video = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
filename, av_err2str(ret));
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
av_err2str(ret));
return ret;
}
ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
video_stream = ret;
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
video = ifmt_ctx->streams[video_stream];
if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
av_err2str(ret));
return ret;
}
decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
if (!decoder_ctx->hw_device_ctx) {
fprintf(stderr, "A hardware device reference create failed.\n");
return AVERROR(ENOMEM);
}
decoder_ctx->get_format = get_vaapi_format;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
av_err2str(ret));
return ret;
}
static int encode_write(AVFrame *frame)
{
int ret = 0;
AVPacket enc_pkt;
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(encoder_ctx, &enc_pkt);
if (ret)
break;
enc_pkt.stream_index = 0;
av_packet_rescale_ts(&enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
ofmt_ctx->streams[0]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
if (ret < 0) {
fprintf(stderr, "Error during writing data to output file. "
"Error code: %s\n", av_err2str(ret));
return -1;
}
}
end:
if (ret == AVERROR_EOF)
return 0;
ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
return ret;
}
static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
{
AVFrame *frame;
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
return ret;
}
while (ret >= 0) {
if (!(frame = av_frame_alloc()))
return AVERROR(ENOMEM);
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
goto fail;
}
if (!initialized) {
/* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
Only after we get a decoded frame, can we obtain its hw_frames_ctx */
encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
if (!encoder_ctx->hw_frames_ctx) {
ret = AVERROR(ENOMEM);
goto fail;
}
/* set AVCodecContext Parameters for encoder, here we keep them stay
* the same as decoder.
* xxx: now the sample can't handle resolution change case.
*/
encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
encoder_ctx->pix_fmt = AV_PIX_FMT_VAAPI;
encoder_ctx->width = decoder_ctx->width;
encoder_ctx->height = decoder_ctx->height;
if ((ret = avcodec_open2(encoder_ctx, enc_codec, NULL)) < 0) {
fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
av_err2str(ret));
goto fail;
}
if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
fprintf(stderr, "Failed to allocate stream for output format.\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ost->time_base = encoder_ctx->time_base;
ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
if (ret < 0) {
fprintf(stderr, "Failed to copy the stream parameters. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
/* write the stream header */
if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
fprintf(stderr, "Error while writing stream header. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
initialized = 1;
}
if ((ret = encode_write(frame)) < 0)
fprintf(stderr, "Error during encoding and writing.\n");
fail:
av_frame_free(&frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0;
AVPacket dec_pkt;
AVCodec *enc_codec;
if (argc != 4) {
fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return -1;
}
ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI, NULL, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(ret));
return -1;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
ret = -1;
goto end;
}
if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
"%s\n", av_err2str(ret));
goto end;
}
if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
ret = AVERROR(ENOMEM);
goto end;
}
ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Cannot open output file. "
"Error code: %s\n", av_err2str(ret));
goto end;
}
/* read all packets and only transcoding video */
while (ret >= 0) {
if ((ret = av_read_frame(ifmt_ctx, &dec_pkt)) < 0)
break;
if (video_stream == dec_pkt.stream_index)
ret = dec_enc(&dec_pkt, enc_codec);
av_packet_unref(&dec_pkt);
}
/* flush decoder */
dec_pkt.data = NULL;
dec_pkt.size = 0;
ret = dec_enc(&dec_pkt, enc_codec);
av_packet_unref(&dec_pkt);
/* flush encoder */
ret = encode_write(NULL);
/* write the trailer for output stream */
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
avformat_close_input(&ofmt_ctx);
avcodec_free_context(&decoder_ctx);
avcodec_free_context(&encoder_ctx);
av_buffer_unref(&hw_device_ctx);
return ret;
}

View File

@@ -76,7 +76,7 @@ the gcc developers. Note that we will not add workarounds for gcc bugs.
Also note that (some of) the gcc developers believe this is not a bug or
not a bug they should fix:
@url{https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
@@ -257,13 +257,13 @@ default.
@section Which are good parameters for encoding high quality MPEG-4?
'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
but beware the '-g 100' might cause problems with some decoders.
Things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
@@ -311,18 +311,18 @@ invoking ffmpeg with several @option{-i} options.
For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
@url{ffmpeg-filters.html#amerge, @code{amerge}} filter.
@url{https://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
For audio, to play one on top of the other: this is called to @emph{mix}
them, and can be done by first merging them into a single stream and then
using the @url{ffmpeg-filters.html#pan, @code{pan}} filter to mix
using the @url{https://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
the channels at will.
@item
For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
@url{ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@url{https://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@@ -333,19 +333,19 @@ There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
FFmpeg has a @url{ffmpeg-filters.html#concat,
FFmpeg has a @url{https://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
FFmpeg has a @url{ffmpeg-formats.html#concat,
FFmpeg has a @url{https://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
FFmpeg has a @url{ffmpeg-protocols.html#concat,
FFmpeg has a @url{https://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
@@ -385,7 +385,7 @@ mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -c:a libmp3lame output.avi
ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
@end example
@subsection Concatenating using raw audio and video
@@ -407,13 +407,13 @@ mkfifo temp2.a
mkfifo temp2.v
mkfifo all.a
mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
cat temp1.a temp2.a > all.a &
cat temp1.v temp2.v > all.v &
ffmpeg -f u16le -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v \
-y output.flv
rm temp[12].[av] all.[av]
@@ -485,7 +485,7 @@ scaling adjusts the SAR to keep the DAR constant.
If you want to stretch, or “unstretch”, the image, you need to override the
information with the
@url{ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
@url{https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
Do not forget to examine carefully the original video to check whether the
stretching comes from the image or from the aspect ratio information.
@@ -501,71 +501,6 @@ ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
@end example
@anchor{background task}
@section How do I run ffmpeg as a background task?
ffmpeg normally checks the console input, for entries like "q" to stop
and "?" to give help, while performing operations. ffmpeg does not have a way of
detecting when it is running as a background task.
When it checks the console input, that can cause the process running ffmpeg
in the background to suspend.
To prevent those input checks, allowing ffmpeg to run as a background task,
use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
or invoke ffmpeg in its own process via an operating system API.
As an alternative, when you are running ffmpeg in a shell, you can redirect
standard input to @code{/dev/null} (on Linux and macOS)
or @code{NUL} (on Windows). You can do this redirect either
on the ffmpeg invocation, or from a shell script which calls ffmpeg.
For example:
@example
ffmpeg -nostdin -i INPUT OUTPUT
@end example
or (on Linux, macOS, and other UNIX-like shells):
@example
ffmpeg -i INPUT OUTPUT </dev/null
@end example
or (on Windows):
@example
ffmpeg -i INPUT OUTPUT <NUL
@end example
@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
If you run ffmpeg in the background, you may find that its process suspends.
There may be a message like @emph{suspended (tty output)}. The question is how
to prevent the process from being suspended.
For example:
@example
% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
[1] 93352
%
[1] + suspended (tty output) ffmpeg -i INPUT OUTPUT &>
@end example
The message "tty output" notwithstanding, the problem here is that
ffmpeg normally checks the console input when it runs. The operating system
detects this, and suspends the process until you can bring it to the
foreground and attend to it.
The solution is to use the right techniques to tell ffmpeg not to consult
console input. You can use the
@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
or redirect standard input with @code{< /dev/null}.
See FAQ
@ref{background task, @emph{How do I run ffmpeg as a background task?}}
for details.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -601,7 +536,7 @@ No. These tools are too bloated and they complicate the build.
FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read @uref{https://web.archive.org/web/20111004021423/http://kernel.org/pub/linux/docs/lkml/#s15, "Programming Religion"}.
read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?

View File

@@ -147,32 +147,6 @@ process.
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@chapter Uploading new samples to the fate suite
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.
Also keep in mind older checkouts use existing sample files, that means in
practice generally do not replace, remove or overwrite files as it likely would
break older checkouts or releases.
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:
rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
#Then do a dry run checking what would be uploaded:
rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
#Upload the files:
rsync -vaL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
@end example
@chapter FATE makefile targets and variables
@@ -223,16 +197,6 @@ through @command{ssh}.
@item GEN
Set to @samp{1} to generate the missing or mismatched references.
@item HWACCEL
Specify which hardware acceleration to use while running regression tests,
by default @samp{none} is used.
@item KEEP
Set to @samp{1} to keep temp files generated by fate test(s) when test is successful.
Default is @samp{0}, which removes these files. Files are always kept when a test
fails.
@end table
@section Examples

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

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

View File

@@ -12,7 +12,7 @@
@chapter Synopsis
ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_url}@} ... @{[@var{output_file_options}] @file{output_url}@} ...
ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ...
@chapter Description
@c man begin DESCRIPTION
@@ -24,10 +24,10 @@ rates and resize video on the fly with a high quality polyphase filter.
@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
files, pipes, network streams, grabbing devices, etc.), specified by the
@code{-i} option, and writes to an arbitrary number of output "files", which are
specified by a plain output url. Anything found on the command line which
cannot be interpreted as an option is considered to be an output url.
specified by a plain output filename. Anything found on the command line which
cannot be interpreted as an option is considered to be an output filename.
Each input or output url can, in principle, contain any number of streams of
Each input or output file can, in principle, contain any number of streams of
different types (video/audio/subtitle/attachment/data). The allowed number and/or
types of streams may be limited by the container format. Selecting which
streams from which inputs will go into which output is either done automatically
@@ -216,208 +216,16 @@ filters is obviously also impossible, since filters work on uncompressed data.
@chapter Stream selection
@c man begin STREAM SELECTION
@command{ffmpeg} provides the @code{-map} option for manual control of stream selection in each
output file. Users can skip @code{-map} and let ffmpeg perform automatic stream selection as
described below. The @code{-vn / -an / -sn / -dn} options can be used to skip inclusion of
video, audio, subtitle and data streams respectively, whether manually mapped or automatically
selected, except for those streams which are outputs of complex filtergraphs.
By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
present in the input files and adds them to each output file. It picks the
"best" of each based upon the following criteria: for video, it is the stream
with the highest resolution, for audio, it is the stream with the most channels, for
subtitles, it is the first subtitle stream. In the case where several streams of
the same type rate equally, the stream with the lowest index is chosen.
@section Description
The sub-sections that follow describe the various rules that are involved in stream selection.
The examples that follow next show how these rules are applied in practice.
While every effort is made to accurately reflect the behavior of the program, FFmpeg is under
continuous development and the code may have changed since the time of this writing.
@subsection Automatic stream selection
In the absence of any map options for a particular output file, ffmpeg inspects the output
format to check which type of streams can be included in it, viz. video, audio and/or
subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available,
from among all the inputs.
It will select that stream based upon the following criteria:
@itemize
@item
for video, it is the stream with the highest resolution,
@item
for audio, it is the stream with the most channels,
@item
for subtitles, it is the first subtitle stream found but there's a caveat.
The output format's default subtitle encoder can be either text-based or image-based,
and only a subtitle stream of the same type will be chosen.
@end itemize
In the case where several streams of the same type rate equally, the stream with the lowest
index is chosen.
Data or attachment streams are not automatically selected and can only be included
using @code{-map}.
@subsection Manual stream selection
When @code{-map} is used, only user-mapped streams are included in that output file,
with one possible exception for filtergraph outputs described below.
@subsection Complex filtergraphs
If there are any complex filtergraph output streams with unlabeled pads, they will be added
to the first output file. This will lead to a fatal error if the stream type is not supported
by the output format. In the absence of the map option, the inclusion of these streams leads
to the automatic stream selection of their types being skipped. If map options are present,
these filtergraph streams are included in addition to the mapped streams.
Complex filtergraph output streams with labeled pads must be mapped once and exactly once.
@subsection Stream handling
Stream handling is independent of stream selection, with an exception for subtitles described
below. Stream handling is set via the @code{-codec} option addressed to streams within a
specific @emph{output} file. In particular, codec options are applied by ffmpeg after the
stream selection process and thus do not influence the latter. If no @code{-codec} option is
specified for a stream type, ffmpeg will select the default encoder registered by the output
file muxer.
An exception exists for subtitles. If a subtitle encoder is specified for an output file, the
first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate
if the specified encoder can convert the selected stream or if the converted stream is acceptable
within the output format. This applies generally as well: when the user sets an encoder manually,
the stream selection process cannot check if the encoded stream can be muxed into the output file.
If it cannot, ffmpeg will abort and @emph{all} output files will fail to be processed.
@section Examples
The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream
selection methods.
They assume the following three input files.
@verbatim
input file 'A.avi'
stream 0: video 640x360
stream 1: audio 2 channels
input file 'B.mp4'
stream 0: video 1920x1080
stream 1: audio 2 channels
stream 2: subtitles (text)
stream 3: audio 5.1 channels
stream 4: subtitles (text)
input file 'C.mkv'
stream 0: video 1280x720
stream 1: audio 2 channels
stream 2: subtitles (image)
@end verbatim
@subsubheading Example: automatic stream selection
@example
ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov
@end example
There are three output files specified, and for the first two, no @code{-map} options
are set, so ffmpeg will select streams for these two files automatically.
@file{out1.mkv} is a Matroska container file and accepts video, audio and subtitle streams,
so ffmpeg will try to select one of each type.@*
For video, it will select @code{stream 0} from @file{B.mp4}, which has the highest
resolution among all the input video streams.@*
For audio, it will select @code{stream 3} from @file{B.mp4}, since it has the greatest
number of channels.@*
For subtitles, it will select @code{stream 2} from @file{B.mp4}, which is the first subtitle
stream from among @file{A.avi} and @file{B.mp4}.
@file{out2.wav} accepts only audio streams, so only @code{stream 3} from @file{B.mp4} is
selected.
For @file{out3.mov}, since a @code{-map} option is set, no automatic stream selection will
occur. The @code{-map 1:a} option will select all audio streams from the second input
@file{B.mp4}. No other streams will be included in this output file.
For the first two outputs, all included streams will be transcoded. The encoders chosen will
be the default ones registered by each output format, which may not match the codec of the
selected input streams.
For the third output, codec option for audio streams has been set
to @code{copy}, so no decoding-filtering-encoding operations will occur, or @emph{can} occur.
Packets of selected streams shall be conveyed from the input file and muxed within the output
file.
@subsubheading Example: automatic subtitles selection
@example
ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv
@end example
Although @file{out1.mkv} is a Matroska container file which accepts subtitle streams, only a
video and audio stream shall be selected. The subtitle stream of @file{C.mkv} is image-based
and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation
for the subtitles is expected to fail and hence the stream isn't selected. However, in
@file{out2.mkv}, a subtitle encoder is specified in the command and so, the subtitle stream is
selected, in addition to the video stream. The presence of @code{-an} disables audio stream
selection for @file{out2.mkv}.
@subsubheading Example: unlabeled filtergraph outputs
@example
ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt
@end example
A filtergraph is setup here using the @code{-filter_complex} option and consists of a single
video filter. The @code{overlay} filter requires exactly two video inputs, but none are
specified, so the first two available video streams are used, those of @file{A.avi} and
@file{C.mkv}. The output pad of the filter has no label and so is sent to the first output file
@file{out1.mp4}. Due to this, automatic selection of the video stream is skipped, which would
have selected the stream in @file{B.mp4}. The audio stream with most channels viz. @code{stream 3}
in @file{B.mp4}, is chosen automatically. No subtitle stream is chosen however, since the MP4
format has no default subtitle encoder registered, and the user hasn't specified a subtitle encoder.
The 2nd output file, @file{out2.srt}, only accepts text-based subtitle streams. So, even though
the first subtitle stream available belongs to @file{C.mkv}, it is image-based and hence skipped.
The selected stream, @code{stream 2} in @file{B.mp4}, is the first text-based subtitle stream.
@subsubheading Example: labeled filtergraph outputs
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-map '[outv]' -an out1.mp4 \
out2.mkv \
-map '[outv]' -map 1:a:0 out3.mkv
@end example
The above command will fail, as the output pad labelled @code{[outv]} has been mapped twice.
None of the output files shall be processed.
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-an out1.mp4 \
out2.mkv \
-map 1:a:0 out3.mkv
@end example
This command above will also fail as the hue filter output has a label, @code{[outv]},
and hasn't been mapped anywhere.
The command should be modified as follows,
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
-map '[outv1]' -an out1.mp4 \
out2.mkv \
-map '[outv2]' -map 1:a:0 out3.mkv
@end example
The video stream from @file{B.mp4} is sent to the hue filter, whose output is cloned once using
the split filter, and both outputs labelled. Then a copy each is mapped to the first and third
output files.
The overlay filter, requiring two video inputs, uses the first two unused video streams. Those
are the streams from @file{A.avi} and @file{C.mkv}. The overlay output isn't labelled, so it is
sent to the first output file @file{out1.mp4}, regardless of the presence of the @code{-map} option.
The aresample filter is sent the first unused audio stream, that of @file{A.avi}. Since this filter
output is also unlabelled, it too is mapped to the first output file. The presence of @code{-an}
only suppresses automatic or manual stream selection of audio streams, not outputs sent from
filtergraphs. Both these mapped streams shall be ordered before the mapped stream in @file{out1.mp4}.
The video, audio and subtitle streams mapped to @code{out2.mkv} are entirely determined by
automatic stream selection.
@file{out3.mkv} consists of the cloned video output from the hue filter and the first audio
stream from @file{B.mp4}.
@*
You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For
full manual control, use the @code{-map} option, which disables the defaults just
described.
@c man end STREAM SELECTION
@@ -435,8 +243,8 @@ Force input or output file format. The format is normally auto detected for inpu
files and guessed from the file extension for output files, so this option is not
needed in most cases.
@item -i @var{url} (@emph{input})
input file url
@item -i @var{filename} (@emph{input})
input file name
@item -y (@emph{global})
Overwrite output files without asking.
@@ -473,7 +281,7 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
When used as an input option (before @code{-i}), limit the @var{duration} of
data read from the input file.
When used as an output option (before an output url), stop writing the
When used as an output option (before an output filename), stop writing the
output after its duration reaches @var{duration}.
@var{duration} must be a time duration specification,
@@ -481,8 +289,8 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
-to and -t are mutually exclusive and -t has priority.
@item -to @var{position} (@emph{input/output})
Stop writing the output or reading the input at @var{position}.
@item -to @var{position} (@emph{output})
Stop writing the output at @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@@ -502,13 +310,13 @@ extra segment between the seek point and @var{position} will be decoded and
discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
will be preserved.
When used as an output option (before an output url), decodes but discards
When used as an output option (before an output filename), decodes but discards
input until the timestamps reach @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -sseof @var{position} (@emph{input})
@item -sseof @var{position} (@emph{input/output})
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.
@@ -523,9 +331,6 @@ The offset is added to the timestamps of the input files. Specifying
a positive offset means that the corresponding streams are delayed by
the time duration specified in @var{offset}.
@item -itsscale @var{scale} (@emph{input,per-stream})
Rescale input timestamps. @var{scale} should be a floating point number.
@item -timestamp @var{date} (@emph{output})
Set the recording timestamp in the container.
@@ -552,49 +357,6 @@ To set the language of the first audio stream:
ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
@end example
@item -disposition[:stream_specifier] @var{value} (@emph{output,per-stream})
Sets the disposition for a stream.
This option overrides the disposition copied from the input stream. It is also
possible to delete the disposition by setting it to 0.
The following dispositions are recognized:
@table @option
@item default
@item dub
@item original
@item comment
@item lyrics
@item karaoke
@item forced
@item hearing_impaired
@item visual_impaired
@item clean_effects
@item attached_pic
@item captions
@item descriptions
@item dependent
@item metadata
@end table
For example, to make the second audio stream the default stream:
@example
ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv
@end example
To make the second subtitle stream the default stream and remove the default
disposition from the first subtitle stream:
@example
ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv
@end example
To add an embedded cover/thumbnail:
@example
ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4
@end example
Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.
@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@@ -617,18 +379,8 @@ they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dn (@emph{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}
option to disable streams individually.
As an output option, disables data recording i.e. automatic selection or
mapping of any data stream. For full manual control see the @code{-map}
option.
@item -dframes @var{number} (@emph{output})
Set the number of data frames to output. This is an obsolete alias for
@code{-frames:d}, which you should use instead.
Set the number of data frames to output. This is an alias for @code{-frames:d}.
@item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
Stop writing to the stream after @var{framecount} frames.
@@ -663,11 +415,6 @@ This option is similar to @option{-filter}, the only difference is that its
argument is the name of the file from which a filtergraph description is to be
read.
@item -filter_threads @var{nb_threads} (@emph{global})
Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel processing.
The default is the number of available CPUs.
@item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
Specify the preset for matching stream(s).
@@ -683,7 +430,6 @@ the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
consists of only alphanumeric characters. The last key of a sequence of
progress information is always "progress".
@anchor{stdin option}
@item -stdin
Enable interaction on standard input. On by default unless standard input is
used as an input. To explicitly disable interaction you need to specify
@@ -744,8 +490,7 @@ Disable automatically rotating video based on file metadata.
@table @option
@item -vframes @var{number} (@emph{output})
Set the number of video frames to output. This is an obsolete alias for
@code{-frames:v}, which you should use instead.
Set the number of video frames to output. This is an alias for @code{-frames:v}.
@item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
Set frame rate (Hz value, fraction or abbreviation).
@@ -783,14 +528,8 @@ 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 -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}
option to disable streams individually.
As an output option, disables video recording i.e. automatic selection or
mapping of any video stream. For full manual control see the @code{-map}
option.
@item -vn (@emph{output})
Disable video recording.
@item -vcodec @var{codec} (@emph{output})
Set the video codec. This is an alias for @code{-codec:v}.
@@ -837,6 +576,8 @@ as the input (or graph output) and automatic conversions are disabled.
@item -sws_flags @var{flags} (@emph{input/output})
Set SwScaler flags.
@item -vdt @var{n}
Discard threshold.
@item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
Rate control override for specific intervals, formatted as "int,int,int"
@@ -856,16 +597,6 @@ Calculate PSNR of compressed frames.
Dump video coding statistics to @file{vstats_HHMMSS.log}.
@item -vstats_file @var{file}
Dump video coding statistics to @var{file}.
@item -vstats_version @var{file}
Specifies which version of the vstats format to use. Default is 2.
version = 1 :
@code{frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
version > 1:
@code{out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s}
@item -top[:@var{stream_specifier}] @var{n} (@emph{output,per-stream})
top=1/bottom=0/auto=-1 field first
@item -dc @var{precision}
@@ -879,19 +610,12 @@ 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
frames after each specified time.
@var{force_key_frames} can take arguments of the following form:
@table @option
@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 the argument is prefixed with @code{expr:}, the string @var{expr}
is interpreted like an expression and is evaluated for each frame. A
key frame is forced in case the evaluation is non-zero.
If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
the time of the beginning of all chapters in the file, shifted by
@@ -905,11 +629,6 @@ 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
@@ -937,12 +656,6 @@ 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.
@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.
@@ -951,133 +664,6 @@ would be more efficient.
When doing stream copy, copy also non-key frames found at the
beginning.
@item -init_hw_device @var{type}[=@var{name}][:@var{device}[,@var{key=value}...]]
Initialise a new hardware device of type @var{type} called @var{name}, using the
given device parameters.
If no name is specified it will receive a default name of the form "@var{type}%d".
The meaning of @var{device} and the following arguments depends on the
device type:
@table @option
@item cuda
@var{device} is the number of the CUDA device.
@item dxva2
@var{device} is the number of the Direct3D 9 display adapter.
@item vaapi
@var{device} is either an X11 display name or a DRM render node.
If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY})
and then the first DRM render node (@emph{/dev/dri/renderD128}).
@item vdpau
@var{device} is an X11 display name.
If not specified, it will attempt to open the default X11 display (@emph{$DISPLAY}).
@item qsv
@var{device} selects a value in @samp{MFX_IMPL_*}. Allowed values are:
@table @option
@item auto
@item sw
@item hw
@item auto_any
@item hw_any
@item hw2
@item hw3
@item hw4
@end table
If not specified, @samp{auto_any} is used.
(Note that it may be easier to achieve the desired result for QSV by creating the
platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
QSV device from that.)
@item opencl
@var{device} selects the platform and device as @emph{platform_index.device_index}.
The set of devices can also be filtered using the key-value pairs to find only
devices matching particular platform or device strings.
The strings usable as filters are:
@table @option
@item platform_profile
@item platform_version
@item platform_name
@item platform_vendor
@item platform_extensions
@item device_name
@item device_vendor
@item driver_version
@item device_version
@item device_profile
@item device_extensions
@item device_type
@end table
The indices and filters must together uniquely select a device.
Examples:
@table @emph
@item -init_hw_device opencl:0.1
Choose the second device on the first platform.
@item -init_hw_device opencl:,device_name=Foo9000
Choose the device with a name containing the string @emph{Foo9000}.
@item -init_hw_device opencl:1,device_type=gpu,device_extensions=cl_khr_fp16
Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
extension.
@end table
@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}
Initialise a new hardware device of type @var{type} called @var{name},
deriving it from the existing device with the name @var{source}.
@item -init_hw_device list
List all hardware device types supported in this build of ffmpeg.
@item -filter_hw_device @var{name}
Pass the hardware device called @var{name} to all filters in any filter graph.
This can be used to set the device to upload to with the @code{hwupload} filter,
or the device to map to with the @code{hwmap} filter. Other filters may also
make use of this parameter when they require a hardware device. Note that this
is typically only required when the input is not already in hardware frames -
when it is, filters will derive the device they require from the context of the
frames they receive as input.
This is a global setting, so all filters will receive the same device.
@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
Use hardware acceleration to decode the matching stream(s). The allowed values
of @var{hwaccel} are:
@@ -1088,15 +674,15 @@ Do not use any hardware acceleration (the default).
@item auto
Automatically select the hardware acceleration method.
@item vda
Use Apple VDA hardware acceleration.
@item vdpau
Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
@item dxva2
Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
@item vaapi
Use VAAPI (Video Acceleration API) hardware acceleration.
@item qsv
Use the Intel QuickSync Video acceleration for video transcoding.
@@ -1120,11 +706,33 @@ useful for testing.
@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
Select a device to use for hardware acceleration.
This option only makes sense when the @option{-hwaccel} option is also specified.
It can either refer to an existing device created with @option{-init_hw_device}
by name, or it can create a new device as if
@samp{-init_hw_device} @var{type}:@var{hwaccel_device}
were called immediately before.
This option only makes sense when the @option{-hwaccel} option is also
specified. Its exact meaning depends on the specific hardware acceleration
method chosen.
@table @option
@item vdpau
For VDPAU, this option specifies the X11 display/screen to use. If this option
is not specified, the value of the @var{DISPLAY} environment variable is used
@item dxva2
For DXVA2, this option should contain the number of the display adapter to use.
If this option is not specified, the default adapter is used.
@item qsv
For QSV, this option corresponds to the values of MFX_IMPL_* . Allowed values
are:
@table @option
@item auto
@item sw
@item hw
@item auto_any
@item hw_any
@item hw2
@item hw3
@item hw4
@end table
@end table
@item -hwaccels
List all hardware acceleration methods supported in this build of ffmpeg.
@@ -1135,8 +743,7 @@ List all hardware acceleration methods supported in this build of ffmpeg.
@table @option
@item -aframes @var{number} (@emph{output})
Set the number of audio frames to output. This is an obsolete alias for
@code{-frames:a}, which you should use instead.
Set the number of audio frames to output. This is an alias for @code{-frames:a}.
@item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
Set the audio sampling frequency. For output streams it is set by
default to the frequency of the corresponding input stream. For input
@@ -1149,14 +756,8 @@ Set the number of audio channels. For output streams it is set by
default to the number of input audio channels. For input streams
this option only makes sense for audio grabbing devices and raw demuxers
and is mapped to the corresponding demuxer options.
@item -an (@emph{input/output})
As an input option, blocks all audio streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables audio recording i.e. automatic selection or
mapping of any audio stream. For full manual control see the @code{-map}
option.
@item -an (@emph{output})
Disable audio recording.
@item -acodec @var{codec} (@emph{input/output})
Set the audio codec. This is an alias for @code{-codec:a}.
@item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
@@ -1190,14 +791,8 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
@table @option
@item -scodec @var{codec} (@emph{input/output})
Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{input/output})
As an input option, blocks all subtitle streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables subtitle recording i.e. automatic selection or
mapping of any subtitle stream. For full manual control see the @code{-map}
option.
@item -sn (@emph{output})
Disable subtitle recording.
@item -sbsf @var{bitstream_filter}
Deprecated, see -bsf
@end table
@@ -1227,7 +822,7 @@ Set the size of the canvas used to render subtitles.
@section Advanced options
@table @option
@item -map [-]@var{input_file_id}[:@var{stream_specifier}][?][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
Designate one or more input streams as a source for the output file. Each input
stream is identified by the input file index @var{input_file_id} and
@@ -1243,11 +838,6 @@ the source for output stream 1, etc.
A @code{-} character before the stream identifier creates a "negative" mapping.
It disables matching streams from already created mappings.
A trailing @code{?} after the stream index will allow the map to be
optional: if the map matches no streams the map will be ignored instead
of failing. Note the map will still fail if an invalid input file index
is used; such as if the map refers to a non-existent input.
An alternative @var{[linklabel]} form will map outputs from complex filter
graphs (see the @option{-filter_complex} option) to the output file.
@var{linklabel} must correspond to a defined output link label in the graph.
@@ -1285,13 +875,6 @@ To map all the streams except the second audio, use negative mappings
ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
@end example
To map the video and audio streams from the first input, and using the
trailing @code{?}, ignore the audio mapping if no audio streams exist in
the first input:
@example
ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
@end example
To pick the English audio stream:
@example
ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
@@ -1307,7 +890,7 @@ such streams is attempted.
Allow input streams with unknown type to be copied instead of failing if copying
such streams is attempted.
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][?][:@var{output_file_id}.@var{stream_specifier}]
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
Map an audio channel from a given input to an output. If
@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
be mapped on all the audio streams.
@@ -1316,10 +899,6 @@ Using "-1" instead of
@var{input_file_id}.@var{stream_specifier}.@var{channel_id} will map a muted
channel.
A trailing @code{?} will allow the map_channel to be
optional: if the map_channel matches no channel the map_channel will be ignored instead
of failing.
For example, assuming @var{INPUT} is a stereo audio file, you can switch the
two audio channels with the following command:
@example
@@ -1367,13 +946,6 @@ video stream), you can use the following command:
ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv
@end example
To map the first two audio channels from the first input, and using the
trailing @code{?}, ignore the audio channel mapping if the first input is
mono instead of stereo:
@example
ffmpeg -i INPUT -map_channel 0.0.0 -map_channel 0.0.1? OUTPUT
@end example
@item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata})
Set metadata information of the next output file from @var{infile}. Note that
those are file indices (zero-based), not filenames.
@@ -1423,26 +995,34 @@ disable any chapter copying.
@item -benchmark (@emph{global})
Show benchmarking information at the end of an encode.
Shows real, system and user time used and maximum memory consumption.
Shows CPU time used and maximum memory consumption.
Maximum memory consumption is not supported on all systems,
it will usually display as 0 if not supported.
@item -benchmark_all (@emph{global})
Show benchmarking information during the encode.
Shows real, system and user time used in various steps (audio/video encode/decode).
Shows CPU time used in various steps (audio/video encode/decode).
@item -timelimit @var{duration} (@emph{global})
Exit after ffmpeg has been running for @var{duration} seconds 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 -re (@emph{input})
Read input at native frame rate. Mainly used to simulate a grab device,
Read input at native frame rate. Mainly used to simulate a grab device.
or live input stream (e.g. when reading from a file). Should not be used
with actual grab devices or live input streams (where it can cause packet
loss).
By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
This option will slow down the reading of the input(s) to the native frame rate
of the input(s). It is useful for real-time output (e.g. live streaming).
@item -loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
This option is deprecated, use -loop 1.
@item -loop_output @var{number_of_times}
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
This option is deprecated, use -loop.
@item -vsync @var{parameter}
Video sync method.
For compatibility reasons old values can be specified as numbers.
@@ -1532,43 +1112,13 @@ Try to make the choice automatically, in order to generate a sane output.
Default value is -1.
@item -enc_time_base[:@var{stream_specifier}] @var{timebase} (@emph{output,per-stream})
Set the encoder timebase. @var{timebase} is a floating point number,
and can assume one of the following values:
@table @option
@item 0
Assign a default value according to the media type.
For video - use 1/framerate, for audio - use 1/samplerate.
@item -1
Use the input stream timebase when possible.
If an input stream is not available, the default timebase will be used.
@item >0
Use the provided number as the timebase.
This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000)
or as a floating point number (e.g. 0.04166, 2.0833e-5)
@end table
Default value is 0.
@item -bitexact (@emph{input/output})
Enable bitexact mode for (de)muxer and (de/en)coder
@item -shortest (@emph{output})
Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
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})
@item -muxdelay @var{seconds} (@emph{input})
Set the maximum demux-decode delay.
@item -muxpreload @var{seconds} (@emph{output})
@item -muxpreload @var{seconds} (@emph{input})
Set the initial demux-decode delay.
@item -streamid @var{output-stream-index}:@var{new-value} (@emph{output})
Assign a new stream-id value to an output stream. This option should be
@@ -1579,7 +1129,7 @@ may be reassigned to a different value.
For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for
an output mpegts file:
@example
ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
@end example
@item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
@@ -1651,11 +1201,6 @@ To generate 5 seconds of pure red video using lavfi @code{color} source:
ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
@end example
@item -filter_complex_threads @var{nb_threads} (@emph{global})
Defines how many threads are used to process a filter_complex graph.
Similar to filter_threads but used for @code{-filter_complex} graphs only.
The default is the number of available CPUs.
@item -lavfi @var{filtergraph} (@emph{global})
Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
outputs. Equivalent to @option{-filter_complex}.
@@ -1684,16 +1229,24 @@ file or device. With low latency / high rate live streams, packets may be
discarded if they are not read in a timely manner; raising this value can
avoid it.
@item -override_ffserver (@emph{global})
Overrides the input specifications from @command{ffserver}. Using this
option you can map any input stream to @command{ffserver} and control
many aspects of the encoding from @command{ffmpeg}. Without this
option @command{ffmpeg} will transmit to @command{ffserver} what is
requested by @command{ffserver}.
The option is intended for cases where features are needed that cannot be
specified to @command{ffserver} but can be to @command{ffmpeg}.
@item -sdp_file @var{file} (@emph{global})
Print sdp information for an output stream to @var{file}.
This allows dumping sdp information when at least one output isn't an
rtp stream. (Requires at least one of the output formats to be rtp).
@item -discard (@emph{input})
Allows discarding specific streams or frames from streams.
Any input stream can be fully discarded, using value @code{all} whereas
selective discarding of frames from a stream occurs at the demuxer
and is not supported by all demuxers.
Allows discarding specific streams or frames of streams at the demuxer.
Not all demuxers support this.
@table @option
@item none
@@ -1721,22 +1274,11 @@ 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 -xerror (@emph{global})
Stop and exit on error
@item -max_muxing_queue_size @var{packets} (@emph{output,per-stream})
When transcoding audio and/or video streams, ffmpeg will not begin writing into
the output until it has one packet for each such stream. While waiting for that
to happen, packets for other streams are buffered. This option sets the size of
this buffer, in packets, for the matching output stream.
The default value of this option should be high enough for most uses, so only
touch this option if you are sure that you need it.
@end table
As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1942,7 +1484,7 @@ to enable LAME support by passing @code{--enable-libmp3lame} to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use @code{ffmpeg -demuxers}.
NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
@item
You can extract images from a video, or create a video from many images:
@@ -1957,8 +1499,8 @@ output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the
above command in combination with the @code{-frames:v} or @code{-t} option,
or in combination with -ss to start extracting from a certain point in time.
above command in combination with the -vframes or -t option, or in
combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
@example
@@ -2042,7 +1584,7 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@ifset config-not-all
@url{ffmpeg-all.html,ffmpeg-all},
@end ifset
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -2061,7 +1603,7 @@ ffmpeg(1),
@ifset config-not-all
ffmpeg-all(1),
@end ifset
ffplay(1), ffprobe(1),
ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -12,7 +12,7 @@
@chapter Synopsis
ffplay [@var{options}] [@file{input_url}]
ffplay [@var{options}] [@file{input_file}]
@chapter Description
@c man begin DESCRIPTION
@@ -60,26 +60,12 @@ Play @var{duration} seconds of audio/video.
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -bytes
Seek by bytes.
@item -seek_interval
Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
@item -nodisp
Disable graphical display.
@item -noborder
Borderless window.
@item -alwaysontop
Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.
@item -volume
Set the startup volume. 0 means silence, 100 means no volume reduction or
amplification. Negative values are treated as 0, values above 100 are treated
as 100.
@item -f @var{fmt}
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -left @var{title}
Set the x position for the left of the window (default is a centered window).
@item -top @var{title}
Set the y position for the top of the window (default is a centered window).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
@@ -120,8 +106,8 @@ the input audio.
Use the option "-filters" to show all the available filters (including
sources and sinks).
@item -i @var{input_url}
Read @var{input_url}.
@item -i @var{input_file}
Read @var{input_file}.
@end table
@section Advanced options
@@ -133,9 +119,8 @@ 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
log level is lower than @code{info}. Its display can be forced by manually
specifying this option. To disable it, you need to specify @code{-nostats}.
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
@item -fast
Non-spec-compliant optimizations.
@@ -198,12 +183,6 @@ input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use @option{-noinfbuf} to disable it.
@item -filter_threads @var{nb_threads}
Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel
processing. The default is 0 which means that the thread count will be
determined by the number of available CPUs.
@end table
@section While playing
@@ -306,7 +285,7 @@ Toggle full screen.
@ifset config-not-all
@url{ffplay-all.html,ffmpeg-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -325,7 +304,7 @@ ffplay(1),
@ifset config-not-all
ffplay-all(1),
@end ifset
ffmpeg(1), ffprobe(1),
ffmpeg(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -12,7 +12,7 @@
@chapter Synopsis
ffprobe [@var{options}] [@file{input_url}]
ffprobe [@var{options}] [@file{input_file}]
@chapter Description
@c man begin DESCRIPTION
@@ -24,8 +24,8 @@ For example it can be used to check the format of the container used
by a multimedia stream and the format and type of each media stream
contained in it.
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
If a filename is specified in input, ffprobe will try to open and
probe the file content. If the file cannot be opened or recognized as
a multimedia file, a positive exit code is returned.
ffprobe may be employed both as a standalone application or in
@@ -208,13 +208,6 @@ multimedia stream.
The information for each single frame is printed within a dedicated
section with name "FRAME" or "SUBTITLE".
@item -show_log @var{loglevel}
Show logging information from the decoder about each frame according to
the value set in @var{loglevel}, (see @code{-loglevel}). This option requires @code{-show_frames}.
The information for each log message is printed within a dedicated
section with name "LOG".
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
@@ -252,7 +245,7 @@ continue reading from that.
Each interval is specified by two optional parts, separated by "%".
The first part specifies the interval start position. It is
interpreted as an absolute position, or as a relative offset from the
interpreted as an abolute position, or as a relative offset from the
current position if it is preceded by the "+" character. If this first
part is not specified, no seeking will be performed when reading this
interval.
@@ -339,8 +332,8 @@ with name "PIXEL_FORMAT".
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@item -i @var{input_url}
Read @var{input_url}.
@item -i @var{input_file}
Read @var{input_file}.
@end table
@c man end
@@ -425,7 +418,7 @@ The @code{csv} writer is equivalent to @code{compact}, but supports
different defaults.
Each section is printed on a single line.
If no option is specified, the output has the form:
If no option is specifid, the output has the form:
@example
section|key1=val1| ... |keyN=valN
@end example
@@ -471,7 +464,7 @@ Perform no escaping.
@end table
@item print_section, p
Print the section name at the beginning of each line if the value is
Print the section name at the begin of each line if the value is
@code{1}, disable it with value set to @code{0}. Default value is
@code{1}.
@@ -584,14 +577,14 @@ value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_strict, x
@item xsd_compliant, x
If set to 1 perform more checks for ensuring that the output is XSD
compliant. Default value is 0.
This option automatically sets @option{fully_qualified} to 1.
@end table
For more information about the XML format, see
@url{https://www.w3.org/XML/}.
@url{http://www.w3.org/XML/}.
@c man end WRITERS
@chapter Timecode
@@ -653,7 +646,7 @@ DV, GXF and AVI timecodes are available in format metadata
@ifset config-not-all
@url{ffprobe-all.html,ffprobe-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -672,7 +665,7 @@ ffprobe(1),
@ifset config-not-all
ffprobe-all(1),
@end ifset
ffmpeg(1), ffplay(1),
ffmpeg(1), ffplay(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -83,7 +83,6 @@
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
@@ -120,25 +119,6 @@
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="logsType">
<xsd:sequence>
<xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="logType">
<xsd:attribute name="context" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int" />
<xsd:attribute name="category" type="xsd:int" />
<xsd:attribute name="parent_context" type="xsd:string"/>
<xsd:attribute name="parent_category" type="xsd:int" />
<xsd:attribute name="message" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
@@ -147,23 +127,8 @@
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:sequence>
<xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
<xsd:attribute name="timecode" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeList">
<xsd:sequence>
<xsd:element name="timecode" type="ffprobe:frameSideDataTimecodeType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeType">
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="subtitleType">
@@ -200,7 +165,6 @@
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
<xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
@@ -226,7 +190,6 @@
<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="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -237,7 +200,6 @@
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
<xsd:attribute name="field_order" type="xsd:string"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<xsd:attribute name="refs" type="xsd:int"/>

372
doc/ffserver.conf Normal file
View File

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

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

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