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236 Commits

Author SHA1 Message Date
Michael Niedermayer
205e2264c3 update for FFmpeg 2.0.5
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 17:08:42 +02:00
Michael Niedermayer
4a23765704 avutil/lzo: add asserts to be double sure against overflows
These asserts cannot fail since d6af26c55c1ea30f85a7d9edbc373f53be1743ee

Based-on: ccda51b14c0fcae2fad73a24872dce75a7964996
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cf2b7c01f81c1fb3283a1390c0ca9a2f81f4f4a8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:25 +02:00
Michael Niedermayer
083d0b7b2d avutil/lzo: Fix integer overflow
Embargoed-till: 2014-06-27 requested by researcher, but embargo broken by libav today (git and mailing list)

Fixes: LMS-2014-06-16-4
Found-by: "Don A. Bailey" <donb@securitymouse.com>
See: ccda51b14c0fcae2fad73a24872dce75a7964996
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d6af26c55c1ea30f85a7d9edbc373f53be1743ee)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:25 +02:00
Michael Niedermayer
1e9ae5dbbd avformat/mpc: attempt to allocate a packet that is not smaller than the data inside it
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 86a9370e2b91d67375e66a06d6eb573b5a017775)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:25 +02:00
Michael Niedermayer
01e2dd53a0 cavsdec: fix qp fixed slice handling
Fixes Ticket3400

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0accf24b15ac5a01a67768f41c896ef4e4b8b4a2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:25 +02:00
Michael Niedermayer
b93659db5b avformat/cavsvideodec: Fix probing when the file extension is avs
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 52e563bb2f7897d615391520c3c4acba1ee7dcb4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:25 +02:00
Michael Niedermayer
b94621ac6b avcodec/alsdec: Clear MPEG4AudioConfig so that no use of uninitialized memory is possible
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6e6bd5481cf42a9765c492c77754d4633092cece)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
a2b4e38729 avformat/flvenc: Do not allow creating h263/mpeg4 in flv without unofficial format extensions being enabled.
Found-by: Jean-Baptiste Kempf <jb@videolan.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 74760883fcb4443d105814ed246b3cf51d7e9dca)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
9fe7e45030 avcodec/aic: fix quantization table permutation
Fixes Ticket3700

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0a2004b6d11ff962361420c3150fe760cf1f7115)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
3c169251ee avcodec/mss4: Fix () in MKVAL() macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cf7ff0146c76b93c32edf5230a28b9590acf5105)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
81a2740919 avcodec/mss34dsp: fix () in SOP* macros
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6e720c5c815e510188a0bda654662383f2c48050)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
a144d305dd avcodec/mlpdec: fix () in MSB_MASK() macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fa160af08b6f42f17e93124aef86e3f6eec70d51)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
4965da92ef avcodec/ivi_dsp: add some missing () to macros
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f276bf303cbb7a8fed3c388135007bc29f45f8d5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
a83304814c avcodec/h264: fix () in macros
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit af62b42736c00332d39965168b5cc966a06f07d6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
db6f3be941 avcodec/golomb-test: fix () in EXTEND() macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 97e6b5ee3a16fee7d130f19f4dcee030f14d91cf)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
2f1f407044 avcodec/aac: fix () in IS_CODEBOOK_UNSIGNED macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fa915d4193e13187773c500b80c7df6baeb22c3b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
bdd2f7c7a7 avcodec/libvorbisenc: dont add the duration to AV_NOPTS_VALUE
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 19e66c7232d96e4ae8f05b52da2b84dfaa4e4da3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
332777970c avutil/cpu: force mmx on selection of higher x86 SIMD features
Fixes various runtime failures with manually set flags that represent no
existing CPU

Fixes Ticket3653

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6310eb8010b7a3b3016e297132380cbd4e3d2d10)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
62b9c99aec avcodec/diracdec: move mc buffer allocation to per frame
Fixes out of array accesses for non default buffers with large strides

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4a30f08505a4e85718896ff233c97be41a9754ca)
(cherry picked from commit 9c9fc79d9237d28e33161cb2e75082d8ad232b2e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
0e18480fc3 sws: dont use the optimized 410->420 unscaled conversion when height%4
Fixes Ticket3594
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 421b21ca8a02a346ba03cea3bb2ecc33f791fc30)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
99d4d3532f avfilter/graphdump: Fix pointer to local outside scope
Fixes CID1194435
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 18af0ce62da322176f7bd283b85314d2f41bee2c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
d9d66d2d16 avcodec/mjpegdec: Fix undefined shift
Fixes CID1194388

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b4329605289e25bb071ec1c1182bf25fc83b09aa)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
db33fb1c48 ffmpeg_filter: fix pointer to local outside scope
Fixes CID1206652
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 09b16619d33ddf93005060d0782f28a1c1cbb7f6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
0be9a2e9d9 swresample: fix AV_CH_LAYOUT_STEREO_DOWNMIX input
Fixes Ticket 3542

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 291d464161a5bf3b566bc147f83e4242b0c18d74)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
ee282be99b avformat/h263dec: Fix h263 probe
The code was missing 1 bit in the src format

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fc145e576a443bfc89efdf35b91fd3c9ca0d8388)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
d3bcc9cfa9 avformat/mux: Check for and remove invalid packet durations
Fixes assertion failure
Fixes Ticket3575

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dc6a17cf74a90e41d70ea1753cdb70c0a5b2ced8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
eaa6d79105 avfilter/filtfmts: Support dynamically allocated in/outputs
Fixes crash
Fixes Ticket3468

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 59c7615d58b5b7ea9caff2c8c774677973eb4f1c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
4d9aad95b5 iavcodec/vc1dec: Fix missing {}
Fixes part of Ticket3466
Found-by: Andrey_Karpov / PVS-Studio
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cb53beb81a5b9192c79de401f1e1e13fadddc429)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:24 +02:00
Michael Niedermayer
6882a04706 avfilter/vf_deshake: fix loss of precission with odd resolutions
Fixes part of Ticket3466
Found-by: Andrey_Karpov / PVS-Studio
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 73734282e0e4df92269984ee1671424e39249481)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:23 +02:00
Michael Niedermayer
98805cbac5 avfilter/f_select: fix loss of precission in SAD calculation
Fixes part of Ticket3466
Found-by: Andrey_Karpov / PVS-Studio
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5a8ef3c66b762f265b05aa096105555f1d26879c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:35:23 +02:00
Michael Niedermayer
66ca450fcb avcodec/g723_1: add assert to help static code analyzers
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1457f3fd90e17745791354fbb87899fc4803085a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
8162b69091 avcodec/diracdec: fix undefined behavior with shifts
Fixes part of Ticket3466
Found-by: Andrey_Karpov / PVS-Studio
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b8598f6ce61ccda3f2ff0c730b009fb650e42986)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
ed38ed3b62 avformat/mpegts: Remove redundant check
Fixes part of Ticket3466
Found-by: Andrey_Karpov / PVS-Studio
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ff6fa0b4b980fc5b9f7653d7b159ae02c3d95210)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
6f6f5f9c8b avcodec/mjpegen: Fix declared argument size
Fixes part of Ticket3466
Found-by: Andrey_Karpov / PVS-Studio
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 256f530603ef3838a712a4fcd737b46b7bce455e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
690a3c4fb7 swscale/swscale: fix srcStride/srcSlice typo
Fixes part of Ticket3466
Found by: Andrey_Karpov / PVS-Studio
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 14fa7fc6a81d5e59e05243cdc92108eab1b138ac)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
2ca68b98ba avcodec/x86/idct_sse2_xvid: fix non C99 inline function
Found-by: Matt Oliver <protogonoi@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 46d5625f44185271862337d61cd246fd569c42a4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Anthoine Bourgeois
853bbe1aed avcodec/dirac_arith: Fix build with PIC and stack-check options
Fixes Ticket3540

The function dirac_get_arith_bit in libavcodec/dirac_arith.h can't be
built with PIC and check-stack because the asm code needs 6 registers
and PIC and check-stack options take 1 each and x86 is quite limited
in this area.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d8ab7f31dd819f7b3e0d460a2fa4261aaae87b98)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
218c853774 swscale/x86/swscale_template: loose hardcoded dstw_offset
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f6759d9ad4a8b71e6f212ca4f1e7da9fa56d3298)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
e315549ffa swresample/resample: use av_malloc_array() where appropriate
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5027f39712fdce25b9008e72d52e5abfeefd5fe6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
c9ed7a4fc1 swresample/dither: use av_malloc_array()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a5290cb1ac047851563da7aca06569e3ada55f79)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
1c321b7c31 swresample/resample: Limit filter length
Related to CID1197063

The limit choosen is arbitrary and much larger than what makes sense.
It avoids the need for checking arithmetic operations with the length for overflow

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f9158b01d0f3effb58e87fb07db0382bc1e47de5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
491e1ec95d avcodec/msrle: check return code for success before use
The check is possibly redundant, but better to check for errors
that dont occur than to skip the check and crash

Fixes CID1197060

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 754f84663e8b3a88fa2e953b195d59230393fb8d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
5805c0c2b6 avcodec/wma: use av_freep(), do not leave stale pointers in memory
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d167faafe9dfa0b82bebb267c3c4e5fa5286bd67)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Lukasz Marek
c61c66cfb1 lavu/opt: validate range before dereference
This change make error handling simplier.
av_opt_freep_ranges may be called when some ranges are NULL,
for example after memory allocation fail.

Signed-off-by: Lukasz Marek <lukasz.m.luki@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3aac5fcfa9d3748659d78ab2a66d0ccce22cfd4f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
5ac53d07a2 avcodec/h264: clear cur_pic structure instead of duplicating it in ff_h264_update_thread_context()
Fixes crash

Found-by: iive
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8710ee11d75eebc17e7d63bc6ffb91766933bd68)

Conflicts:

	libavcodec/h264_slice.c
(cherry picked from commit cb44d69665596d2f19e032fc852162b0a6de0562)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
02055da69a avcodec/libx264: move where x264opts is applied down so it isnt overridden by avctx & defaults
fixes x264opts opengop=1

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 64b79141bdfdffaa9fda69eecce140473d0a9a18)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
c7f419efd1 avcodec/h264_mp4toannexb_bsf: prepend global headers before any in stream parameter sets
Fixes h264_mp4toannexb_bsf_failure.mkv

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 289b149cecb381522cc9ccdf382825330169c655)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
b556432f5a avcodec/x86/mpegvideoenc_template: fix integer overflow
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:40 +02:00
Michael Niedermayer
102df43f49 dox/scaler:fix bicubiclin typo
See Ticket3486

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 575b957758670d6094e9095acfcc24e4e32fc4a7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
36fc9bff08 swscale/swscale: fix integer overflow
Should fix fate failure with clang ftrapv

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c9c0451224fd7bc38b4e135e99f114f80c1ae67f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
3ae71dd33e avcodec: Add padding after the remaining AVFrames
This limits ABI issues in case libavcodec is linked to a libavutil with larger AVFrame
Which can happen if they are shiped in seperate binary packages and libavutil is upgraded

A cleaner alternative would be to replace them by pointers but this would likely cause
a small speedloss

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fc567ac49e17151f00f31b59030cd10f952612ef)

Conflicts:

	libavcodec/h264.h
2014-06-25 16:18:39 +02:00
Michael Niedermayer
673f679c8a avformat/mp3enc: use av_copy_packet()
Fixes double free
Fixes Ticket3476

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d003a0cd2e587a47627fd328f9fc5a484adc29f2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
f65501f80c ffmpeg: dont call exit_program() from a signal hander
This is unsafe and can deadlock amongth other things

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9dca02ee541120de2a96c387faed9a4e033a60fd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
6387aa94d6 avcodec/h263dec: Fix use of uninitialized memory from the bitstream buffer
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f07cebcd910c97ff6012085c21493231752990e9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
d581567e09 avcodec/utils: fix sizeof(AVFrame) dependence in avcodec_encode_audio2()
This is a bit tricky, we allocate a correctly sized AVFrame but then only
copy the compile time AVFrame size, this is to ensure that user applications
which do not use the correct av frame API dont end with out of array reads.
Note, applications using the correct API have set extended_data and the
changed code will never be executed for them.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8ab80707841a73ca7708e1e1aa97f3513fff3d35)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
9933e06595 swscale/x86/swscale: fix missing xmm clobbers in yuv2yuvX_sse3()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6c47a4e972485e5f0c812159373f703c6f1d089f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
91d024d75d avutil/timestamp: Warn about missing __STDC_FORMAT_MACROS for C++ use
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8b02dfd37cb3bc9521fc6e1f5b5f13c80d144cd2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-25 16:18:39 +02:00
Michael Niedermayer
5ee384c4eb avformat/mov: fix keyframe flags for sample from chromium Issue 340865
Fixes ticket #3362.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a0911b059763b8f13c70adcbbe71e10382855104)
2014-03-24 23:36:07 +01:00
Hendrik Leppkes
9ecd0f34ad avformat/mov: only force parsing for video tracks if stss is empty
Fixes playback of some AAC streams, which are otherwise mangled by the
parser, and stss is typically only valid for video anyway.

Fixes a regression since e41ea866.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 019247bdc326a90bf20d3ce5d2413cc642e8bb08)
2014-03-24 23:36:04 +01:00
Michael Niedermayer
7de7bd4f56 update for 2.0.4
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-03-04 01:20:37 +01:00
Michael Niedermayer
e939c7b7f5 avcodec/utvideoenc: fix slice_bits size
Fixes assertion failure

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0a8c90202bb906747168a698b6837496f82c717c)

Conflicts:

	libavcodec/utvideoenc.c
(cherry picked from commit 57522ca79cc38c279123596d3288ddbf56fa8903)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-03-01 20:28:49 +01:00
Timothy Gu
612ef09a18 configure: use pkg-config to detect libbluray
The current configure fails when static libbluray is compiled with libxml2
support.

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit baa650cc7946a9eb1cf5a083f61a581a97122f03)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-27 00:33:00 +01:00
Michael Niedermayer
f0ee0fcbfc avcodec/snow: split block clipping checks
Fixes out of array read
Fixes: d4476f68ca1c1c57afbc45806f581963-asan_heap-oob_2266b27_8607_cov_4044577381_snow_chroma_bug.avi
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 61d59703c91869f4e5cdacd8d6be52f8b89d4ba4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-18 05:10:13 +01:00
Michael Niedermayer
d41f4e8dc8 avcodec/ansi: fix integer overflow
Fixes out of array read
Fixes: 5f9698e86d92f19bb08d54ff0d57027f-signal_sigsegv_b30756_3795_cov_2693691257_ansi256.ans
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d42ec8433c687fcbccefa51a7716d81920218e4f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-17 20:56:24 +01:00
Michael Niedermayer
bc1c8ec5e6 avcodec/msrle: use av_image_get_linesize() to calculate the linesize
Fixes out of array access
Fixes: 14a74a0a2dc67ede543f0e35d834fbbe-asan_heap-oob_49572c_556_cov_215466444_44_001_engine_room.mov
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c919e1ca2ecfc47d796382973ba0e48b8f6f92a2)

Conflicts:

	libavcodec/msrle.c
2014-02-16 23:50:50 +01:00
Michael Niedermayer
67b943ad66 avcodec/utils: set AVFrame format unconditional
Fixes inconsistency and out of array accesses
Fixes: 10cdd7e63e7f66e3e66273939e0863dd-asan_heap-oob_1a4ff32_7078_cov_4056274555_mov_h264_aac__mp4box_frag.mp4
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e5c7229999182ad1cef13b9eca050dba7a5a08da)

Conflicts:

	libavcodec/utils.c
(cherry picked from commit 4f94e1901a1ff0073c64122c577b6efd3dee22d4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-16 04:55:02 +01:00
Michael Niedermayer
5e7e43c33e avformat/mpegtsenc: Check data array size in mpegts_write_pmt()
Prevents out of array writes

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 842b6c14bcfc1c5da1a2d288fd65386eb8c158ad)

Conflicts:

	libavformat/mpegtsenc.c
(cherry picked from commit e87de3f50b765134588d0b048c32ed4b8acc16fb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-13 15:18:26 +01:00
Michael Niedermayer
13ce367368 avcodec/wmalosslessdec: fix mclms_coeffs* array size
Fixes corruption of context
Fixes: 8835659dde6a4f7dcdf341de6a45c6c8-signal_sigsegv_1dce67b_4564_cov_2504444599_classical_22_16_1_14000_v3c_0_extend_0_29.wma
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ec9578d54d09b64bf112c2bf7a34b1ef3b93dbd3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-07 15:43:07 +01:00
Justin Ruggles
0962c26b6b samplefmt: avoid integer overflow in av_samples_get_buffer_size()
CC:libav-stable@libav.org
(cherry picked from commit 0e830094ad0dc251613a0aa3234d9c5c397e02e6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-05 01:36:49 +01:00
Michael Niedermayer
33c47d3976 avcodec/vc1: reset fcm/field_mode in non advanced header parsing
Fixes NULL pointer dereference
Fixes: signal_sigsegv_1ab8bf4_2847_cov_4254117347_SA10091.vc1
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b51e9354772de446e8196dabf9aad1567b22f74d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-01 19:22:23 +01:00
Michael Niedermayer
d0d0924947 avcodec/takdec: always check bits_per_raw_sample
Fixes out of array access
Fixes: asan_heap-oob_19c7a94_6470_cov_1453611734_luckynight-partial.tak
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f58eab151214d2d35ff0973f2b3e51c5eb372da4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-31 19:55:25 +01:00
Michael Niedermayer
ca5d6c615e avcodec/jpeg2000dec: fix error detection in pix_fmt_match()
Fixes out of array accesses with CODEC_FLAG_EMU_EDGE

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8001e9f7d17e90b4b0898ba64e3b8bbd716c513c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 17:05:16 +01:00
Michael Niedermayer
7c17207ab9 avcodec/vmnc: Check that rectangles are within the picture
Prevents out of array accesses with CODEC_FLAG_EMU_EDGE

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6ba02602aa7fc7d38db582e75b8b093fb3c1608d)

Conflicts:

	libavcodec/vmnc.c

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 17:02:19 +01:00
Michael Niedermayer
9847f02faf dnxhdenc: fix mb_rc size
Fixes out of array access with RC_VARIANCE set to 0

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f1caaa1c61310beba705957e6366f0392a0b005b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-18 01:13:22 +01:00
Michael Niedermayer
dfdeabadca cmdutils: update year
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-16 16:03:25 +01:00
Dale Curtis
33aa2c5d6b h264: Clear ERContext.cur_pic when unref'ing current picture.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
(cherry picked from commit 4feca2214a0b69dcbe4d1c7cd145c3881459e867)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-03 00:44:29 +01:00
Michael Niedermayer
b4552cc9b8 update for FFmpeg 2.0.3
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 03:22:00 +01:00
Michael Niedermayer
172f929767 avutil/log: skip IO calls on empty strings
These occur when no context is set for example, thus they are common

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a044a183a3fb90b20a8deaa3ea1158510bcdd420)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
66a9edfcf6 do O(1) instead of O(n) atomic operations in register functions
about 1ms faster startup time

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 133fbfc7811ffae7b97dd129fcd0b5e646742362)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Paul B Mahol
a9382fc15c avcodec/libopusenc: change default frame duration to 20 ms
20 ms is used by libopus encoder.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit 74906d3727ec3bd9b7b28dfa7a98ff6e8cf8b6d7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
bd9dcb411d avcodec/jpeg2000dec: Check precno before using it in JPEG2000_PGOD_CPRL
Fixes out of array reads
Fixes: asan_heap-oob_f0de57_6823_mjp2.mov

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3d5a5e86be2a65e33c34ab3ad7923f54e8e49c1d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
ae81a0e32d avcodec: move end zeroing code from av_packet_split_side_data() to avcodec_decode_subtitle2()
This code changes the input packet, which is read only and can in
rare circumstances lead to decoder errors. (i run into one of these in
the audio decoder, which corrupted the packet during av_find_stream_info()
so that actual decoding that single packet failed later)
Until a better fix is implemented, this commit limits the problem.
A better fix might be to make the subtitle decoders not depend on
data[size] = 0 or to copy their input when this is not the case.
(cherry picked from commit 01923bab98506b1e98b4cbf08419364ce6ffea6d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
4f93400db1 avutil: reintroduce lls1 as the 52 ABI needs it
lls1 taken from ff130d7

This is incompatible with libavcodec version
55.18.100 to 55.43.100 except 55.39.101
This incompatibility is caused by these libavcodec versions depending on
a libavutil 52 which is ABI incompatible with the previous ABI 52

you can avoid this incompatibility by upgrading your libavcodec so it
does no longer depend on the invalid ABI

See: 502ab21af0ca68f76d6112722c46d2f35c004053
See: cc6714bb16b1f0716ba43701d47273dbe9657b8b
See: 41578f70cf8aec8e7565fba1ca7e07f3dc46c3d2
See: Ticket3136
Tested-by: marillat
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b382d09d29be90e0947295a70cdcbaa60b9030b8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
0cd61c7f7d rename new lls code to lls2 to avoid conflict with the old which has a different ABI
also remove failed attempt at a compatibility layer, the code simply cannot work

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c3814ab654a993723b0e5f14cc252d68f233ad79)

Conflicts:

	libavcodec/version.h
2013-12-24 01:05:47 +01:00
Michael Niedermayer
28ac4e91dc avutil: rename lls to lls2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bbe66ef912470007f7cc424badde2ccec500b36b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Diego Biurrun
6b683be641 mpeg12dec: Remove incomplete and wrong UV swapping code for VCR2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 321514042534a2501a9f6223b88f0d2b8060f858)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Kostya Shishkov
3a3b5ae4c0 mpegvideo: Fix swapping of UV planes for VCR2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bae14f38d992f326c94d93f01197ccd84ea62053)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
70fcea3b77 h264: Do not treat the initial frame special in handling of frame gaps
The not handling of frame gaps has lead to the lack of a dummy reference
frame, which has lead to the failure of decode_slice_header() which has
lead to one SEI recovery message being skiped which had introduced a
slightly suboptimal recovery point for at least 1 h264 file compared to
JM.

Found-by: Carl & BugMaster
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9e5ef1c5c37208326c59d642e2dc7afd3f10b09b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
b545d11d49 avcodec/jpeg2000dec: non zero image offsets are not supported
Fixes out of array accesses
Fixes Ticket3080
Found-by: ami_stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 780669ef7c23c00836a24921fcc6b03be2b8ca4a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
4324d7bade avformat/thp: force moving forward
Fixes infinite loop
Fixes Ticket3098

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6c4b87d3d6ae08a6da16b4616626b4d2a726afbf)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
c88bdac460 avformat/thp: fix variable types to avoid overflows
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2b1056e4e27b046af3777e8bd65a5145abff878f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
bdf6e6fff4 avcodec/jpeglsdec: check err value for ls_get_code_runterm()
Fixes infinite loop
Fixes Ticket3086

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cc0e47b55096361723b364afa43b79a3f5619cdc)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
83e4aa3e7c avutil/opt: initialize ret
Fixes CID1108610
Fixes use of uninitialized variable

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2d8ccf0adcae09cb9e14b01cfe20e4d77c3bbf5d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
fce2cfbdcf avcodec/utils: add some saftey checks to add_metadata_from_side_data()
This fixes potential overreads with crafted files.

Found-by: wm4
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 838f461b0716393a1b5c70efd03de1e8bc197380)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:47 +01:00
Michael Niedermayer
72f1907c96 avcodec/avpacket/av_packet_split_side_data: ensure that side data padding is initialized
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 240fd8c96f59ebe9dcfc4152a1086cd3f63400c0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:46 +01:00
Michael Niedermayer
074ebfacf4 avfilter/ff_insert_pad: fix order of operations
Fixes out of bounds access
Fixes CID732170
Fixes CID732169

No filter is known to use this function in a way so the issue can be reproduced.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ab2bfb85d49b2f8aa505816f93e75fd18ad0a361)

Conflicts:

	libavfilter/avfilter.c
2013-12-24 01:05:46 +01:00
Michael Niedermayer
47f8497837 avcodec/jpeg2000dec: fix context consistency with too large lowres
Fixes out of array accesses
Fixes Ticket2898

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a1b9004b768bef606ee98d417bceb9392ceb788d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:46 +01:00
Michael Niedermayer
93f26b7992 avcodec/jpeg2000dec: prevent out of array accesses in pixel addressing
Fixes Ticket2921

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fe448cd28d674c3eff3072552eae366d0b659ce9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:46 +01:00
Michael Niedermayer
f684bbf224 avcodec/jpeg2000dec: check transform equality in MCT
Fixes null pointer dereference
Fixes Ticket2843

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ac3b01a9c0607961f4540fe62040833870f5deb1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:46 +01:00
Michael Niedermayer
edca16f1af ffserver: strip odd chars from html error messages before sending them back
Fixes Ticket3034

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 885739f3b4ca3fb60abf417120845e3fcfb99b53)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-12-24 01:05:46 +01:00
Mason Carter
aeac212fda VC1: Fix intensity compensation performance regression
Fix https://trac.ffmpeg.org/ticket/3204

The problem was that intensity compensation was always used once it was
encountered. This is because v->next_use_ic was never set back to zero.
To fix this, when resetting v->next_luty/uv, also reset v->next_use_ic.

This improved (restored) performance by 85% when decoding
http://bit.ly/bbbwmv

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ed5bed4152203aed8cce01a679bed67bbda8903f)
2013-12-22 16:09:44 +01:00
Martin Storsjö
62f05d6309 arm: Don't clobber callee saved registers in scalarproduct
q4-q7/d8-d15 are supposed to not be clobbered by the callee.

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit d307e408d4a9ada22df443cc38be77cc5e492694)
2013-12-21 09:58:12 +01:00
Michael Niedermayer
bd339d4882 swscale/utils: check chroma width for fast bilinear scaler
Fixes artifacts where fast bilinear was used for downscaling chroma

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 037fc3b054b10aee0f11fdbe835e5dffa8e95b37)
2013-12-16 02:24:13 +01:00
Michael Niedermayer
9de71b0eb2 swscale/utils: remove useless ()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 554e913fd7acc9da02ddac2c5ce9487f7f633c92)
2013-12-16 02:24:12 +01:00
Michael Niedermayer
7e73760950 avcodec/cabac: force get_cabac to be not inlined
works around bug in gccs inline asm register assignment
Fixes Ticket3177

gcc from 4.4 to 4.6 is affected at least, no non affected gccs known
clang seems not affected

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0538b29ae8002c44f27bae8a1a6fc6e646998be5)
2013-12-09 10:36:00 +01:00
Michael Niedermayer
0639e403be avcodec/error_resilience: check that er is supported before attempting to read the status of the previous slice
Fixes incorrectly set error_occured and improves speed

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 90539cea336fd513c47295a03c164cb4a851166f)
2013-12-07 11:34:42 +01:00
Michael Niedermayer
5c7d6be5f9 avcodec/error_resilience: factor er_supported() check out
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit afb18c55783362546b5e512ce01b7fe7bf5744d9)
2013-12-07 11:34:25 +01:00
Nicolas George
2005887707 lavfi/af_pan: support unknown layouts on input.
Fix trac ticket #2899.
(cherry picked from commit 7b0a587393e03dab552d66450d43ab82bda0a5a1)
2013-11-28 01:09:19 +01:00
Nicolas George
cf8462ce00 lavfi/af_pan: support unknown layouts on output.
(cherry picked from commit 4e9adc9b7363cc336e3d47c98455e1508902fd29)
2013-11-28 01:09:12 +01:00
Nicolas George
02c8c064ea lswr: fix assert failure on unknown layouts.
(cherry picked from commit 4a640a6ac89099bfb02d6d3d3ada04e321a37476)
2013-11-28 01:09:06 +01:00
Nicolas George
9a4acedf31 lavfi: parsing helper for unknown channel layouts.
Make ff_parse_channel_layout() accept unknown layouts too.
(cherry picked from commit 6e2473edfda26a556c615ebc04d8aeba800bef7e)
2013-11-28 01:08:59 +01:00
Nicolas George
c2d37e7364 lavfi/avfiltergraph: do not reduce incompatible lists.
A list of "all channel layouts" but not "all channel counts"
can not be reduced to a single unknown channel count.
(cherry picked from commit d300f5f6f570659e4b58567b35c9e8600c9f2956)
2013-11-28 01:08:52 +01:00
Nicolas George
62baf22ec0 lavfi/avfiltergraph: suggest a solution when format selection fails.
Format selection can fail if unknown channel layouts are used
with filters that do not support it.
(cherry picked from commit f775eb3fb4c7b716107355e428e40cb63f71ee7a)
2013-11-28 01:08:46 +01:00
Nicolas George
8f7d839e15 lavd/lavfi: support unknown channel layouts.
(cherry picked from commit 863fb11f63f7f60feec390f3c54dd13606e07d05)
2013-11-28 01:08:37 +01:00
Clément Bœsch
93716f7bea avformat/image2: allow muxing gif files.
Fixes Ticket #2936.
(cherry picked from commit f70db22999d713da3306bf29ec763d670b9bf1ea)

Conflicts:
	libavformat/img2enc.c
2013-11-18 14:34:59 +01:00
Clément Bœsch
ff4c53e8b3 build: avoid stdin stall with GNU AS probing.
a758c5e added probing for various tools, such as AS. Unfortunately, GNU
AS is reading stdin with -v, and thus configure is stalled with
configure arguments such as --as=as.

Fixes Ticket #1898.
(cherry picked from commit dbb41f93c16cbc65a899a75723c95da51c851cd5)
2013-11-18 14:33:46 +01:00
Michael Niedermayer
9a22d6dd63 avformat/utils: dont count attached pics toward the probesize
Such pics behave more like headers which we also dont count.
Fixes Ticket3146

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a8dec360c5db15e8da4b44ff3c0f02a6c57e8ac0)
2013-11-18 14:33:09 +01:00
Michael Niedermayer
beb28bc55d avcodec/bink: fix seeking to frame 0
Fixes Ticket3088

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cb52d6da0a9c88c584a38a9a7a94825565854b7e)

Conflicts:
	libavcodec/bink.c
2013-10-31 00:57:11 +01:00
Michael Niedermayer
1fa7ad2e20 ffmpeg_filter: Fix non jpeg yuv in jpeg support
This is a regression, did not bisect so dont know what caused it but
likely some changes to the command line handling code.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d9bc251d3978cf0b9722991f7c23edebca7abb18)
2013-10-27 20:24:31 +01:00
Michael Niedermayer
f514834917 avformat/utils: do not override pts in h264 when they are provided from the demuxer
Fixes Ticket2143

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1e5271a9fd6ddcceb083f2185a4bbd8d44c9a813)
2013-10-27 19:40:25 +01:00
Michael Niedermayer
1d0e583728 h264: make flush_change() set mmco_reset
This ensures that frames do not get mixed on context reinits

Fixes Ticket2836

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3c9dd93faa9f3c250428dd0548c075583aa07cc3)
2013-10-26 02:41:31 +02:00
Michael Niedermayer
782331be1e avcodec/h264: reduce noisiness of "mmco: unref short failure"
Do not consider it an error if we have no frames and should discard one.
This condition can easily happen when decoding is started from an I frame

Fixes Ticket2811

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 08a89761964bdd0a023eff6d37a1131fb7e1d7a0)
2013-10-26 00:52:34 +02:00
Michael Niedermayer
94c7ee4d9e avformat/mp3dec: perform seek resync in the correct direction
Fixes seeking to the last frame in CBR files
Fixes Ticket2773

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ba8716df7fb541fb690d1a898cda0e12f9011faf)
2013-10-26 00:51:50 +02:00
Michael Niedermayer
b7154758de avformat/wavdec: Fix smv packet interleaving
This strips the relative timestamp "flag" off.

Fixes Ticket2849

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6abb9eb525239f954ec12e52fc209e6dfce2a6d4)
2013-10-26 00:50:44 +02:00
Michael Niedermayer
cd7d575e90 avcodec/h264: do not trust last_pic_droppable when marking pictures as done
This simplifies the code and fixes a deadlock

Fixes Ticket2927
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 29ffeef5e73b8f41ff3a3f2242d356759c66f91f)
2013-10-26 00:50:00 +02:00
Stefano Sabatini
17d169ce0f doc/Makefile: fix man pages uninstall path
Fix trac ticket #3054.
(cherry picked from commit af1c5388501e80e6d92704bb70610998226e5e51)
2013-10-26 00:49:18 +02:00
Michael Niedermayer
fb1fb462e5 avformat/mov: force parsing of headers if stts is absent
Fixes Ticket2991

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e41ea866fc26f38d770bbc1ad67703e7f4400ae1)
2013-10-24 10:35:34 +02:00
Michael Niedermayer
7da810e68b avformat/gifdec: make GIF_APP_EXT_LABEL parsing more robust
Fixes Ticket3021

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e1f8184a1a973fd7de1bf53578d09661ec7bad75)
2013-10-24 10:33:43 +02:00
Michael Niedermayer
6de6d9e2d3 avformat/matroskadec: only set r_frame_rate if the value is within reasonable limits
Fixes Ticket2451

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6853e40106cac769f0641183ea0bdd530ae9a0a1)
2013-10-24 10:30:33 +02:00
Michael Niedermayer
736851264b avformat/wavdec: Dont trust the fact chunk for PCM
Fixes Ticket3033

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 83fc6c822b06688e572333299927d93eb3c6c426)
2013-10-24 10:29:37 +02:00
Michael Niedermayer
59431fc841 avcodec/h264_refs: modify key frame detection heuristic to detect more cases
Fixes Ticket2968

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5ac6b6028f17b64723884c9fa72cfcbd369a1ba2)
2013-10-24 10:28:20 +02:00
mrlika
f581e25a69 lavd/v4l2: do not fail when VIDIOC_ENUMSTD returns EINVAL without a valid match
With some (buggy) drivers, the VIDIOC_G_STD ioctl returns a std_id that cannot
be matched with any of the enumerated v4l2_standard structures (for example
std_id = 0 or std_id = 0xffffff). Do not fail when we reach the end of the
enumeration without a valid match.

Fixes ticket #2370

Note: This commit message has been modified by Giorgio Vazzana, the original
commit message was:

"Fixed regression for mandatory VIDIOC_ENUMSTD support by v4l2"

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ed72542539fb61dc3a6d6280d8a6a956ac04a071)
2013-10-24 10:09:30 +02:00
Stefano Sabatini
9d0bb7fc39 doc/codecs: fix dangling reference to codec-options chapter
(cherry picked from commit b4bd21b7fe2ad8be59b16538448586814e5db65b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-10-08 20:57:07 +02:00
Michael Niedermayer
a53fd4b758 update for 2.0.2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-10-08 19:39:58 +02:00
Paul B Mahol
b4ccdf5e68 avcodec/ffv1dec: fix format detection
Fixes crash with carefuly designed files.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit a27227d401adf12534dc7a26d72e43e2f35f8944)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-10-08 18:13:55 +02:00
Michael Niedermayer
9b02aa2593 avcodec/parser: reset indexes on realloc failure
Fixes Ticket2982

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f31011e9abfb2ae75bb32bc44e2c34194c8dc40a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-10-08 18:13:55 +02:00
Michael Niedermayer
f089e67d51 avcodec/imgconvert/get_color_type: fix type for PAL8
Fixes Ticket3008

Fate changes as PAL8 gets used instead of BGR8

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 95666b22989b9b9f91a27da01b2bdbf4ee8022d3)
2013-10-02 23:47:41 +02:00
Thilo Borgmann
842d7c9b3a configure: fix logic for threads in case of OpenCL is enabled.
Fixes ticket 3004.

Signed-off-by: Thilo Borgmann <thilo.borgmann@mail.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d3a03d90a3d57cdc6f915a05f5f28ee4bb470d00)
2013-10-02 23:44:39 +02:00
Michael Niedermayer
187297b871 Merge remote-tracking branch 'TimothyGu/release/2.0' into release/2.0
* TimothyGu/release/2.0:
  doc/encoders: add doc for AAC encoder
  doc/formats: Add documentation for 3 parameters that have been missing
  doc/encoders: improve libvo-aacenc doc
  doc/encoders: reformat and add some clarification in libtwolame doc
  doc/encoders: reformat libmp3lame doc
  doc/encoders: add libxvid doc
  doc/encoders: partially rewrite and reformat libx264 docs

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 22:31:44 +02:00
Michael Niedermayer
211ad5042a Merge remote-tracking branch 'jamrial/release/2.0' into release/2.0
* jamrial/release/2.0:
  avformat/matroskadec: check out_samplerate before using it in av_rescale()
  matroskadec: Improve TTA duration calculation
  avformat/oggparsevorbis: fix leak of tt
  avformat/oggparsevorbis: fix leak of ct

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 22:30:49 +02:00
Michael Niedermayer
2b06f5f8f1 avcodec/g2meet: Fix framebuf size
Currently the code can in some cases draw tiles that hang outside the
allocated buffer. This patch increases the buffer size to avoid out
of array accesses. An alternative would be to fail if such tiles are
encountered.
I do not know if any valid files use such hanging tiles.

Fixes Ticket2971
Found-by: ami_stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e07ac727c1cc9eed39e7f9117c97006f719864bd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Michael Niedermayer
0a64b25c77 avcodec/g2meet: Fix order of align and pixel size multiplication.
Fixes out of array accesses
Fixes Ticket2922

Found-by: ami_stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 821a5938d100458f4d09d634041b05c860554ce0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Michael Niedermayer
40e52bbb63 avcodec/ffv1enc: update buffer check for 16bps
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3728603f1854b5c79d1a64dd3b41b80640ef1e7f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Michael Niedermayer
aeec1a6430 avcodec/truemotion2: Fix av_freep arguments
Fixes null pointer dereference
Fixes Ticket2944

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c54aa2fb0f869ec025933944cbd1634fffe95d09)

Conflicts:

	libavcodec/truemotion2.c
2013-09-23 21:46:15 +02:00
Michael Niedermayer
ef121a88d5 avcodec/mjpegdec: Add some sanity checks to ljpeg_decode_rgb_scan()
These prevent the rgb ljpeg code from being run on parameters that it doesnt
support. No testcase available but it seems possible to trigger these.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 61c68000eda643dfce96dc46b488d39fd5c4e309)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Michael Niedermayer
db99c41567 avfilter/vf_fps: make sure the fifo is not empty before using it
Fixes Ticket2905

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cdd5df8189ff1537f7abe8defe971f80602cc2d2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Michael Niedermayer
54bdb5fc86 avcodec/dsputil: fix signedness in sizeof() comparissions
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 454a11a1c9c686c78aa97954306fb63453299760)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Michael Niedermayer
2fd824b466 ffv1dec: Check bits_per_raw_sample and colorspace for equality in ver 0/1 headers
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b05cd1ea7e45a836f7f6071a716c38bb30326e0f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Michael Niedermayer
005b38f8f1 avcodec/ffv1dec: check global header version
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 20b965a1a43ae88b7ae95635d5a3570e7dc2bbd4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-23 21:46:15 +02:00
Timothy Gu
6911d9e1b0 doc/encoders: add doc for AAC encoder
Thanks-to: Kostya Shishkov <kostya.shishkov@gmail.com>
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
(cherry picked from commit 0e11790cf7eef3b0f38a64486da1e3fb8c7f14b8)

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2013-09-22 15:29:53 -07:00
Timothy Gu
cecb2b39ce doc/formats: Add documentation for 3 parameters that have been missing
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b7dd4598630fb1f890ae060f21a9ec92a103f22d)

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2013-09-22 15:29:22 -07:00
Timothy Gu
9ebfee7ac0 doc/encoders: improve libvo-aacenc doc
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
(cherry picked from commit 81bbe49a0e588aa899f37a567808ba8926d798d6)

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2013-09-22 15:29:12 -07:00
Timothy Gu
01b39884c7 doc/encoders: reformat and add some clarification in libtwolame doc
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
(cherry picked from commit e45e72f5f89ef5a5791562cfcb935028b46ecd0a)

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2013-09-22 15:28:56 -07:00
Timothy Gu
c08e8ab715 doc/encoders: reformat libmp3lame doc
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
(cherry picked from commit 40b8350b57adaa9aaf6731bac5130d4fec1639c3)

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2013-09-22 15:28:40 -07:00
Timothy Gu
68ae344b5e doc/encoders: add libxvid doc
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
(cherry picked from commit 6b255e5e70c72aa59ff7aed74b4ee976223eb140)

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2013-09-22 15:27:25 -07:00
Timothy Gu
5753d780b4 doc/encoders: partially rewrite and reformat libx264 docs
Format is based on the thread:
"[PATCH] doc/encoders: Add libopus encoder doc" (06-28-2013)
http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/165368/

Also merge the two option sections (Mapping and Private options).

Patch partially edited by Stefano Sabatini.

Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
(cherry picked from commit 11cb697501edf6447a718d6194f0e63ec2310d39)

Signed-off-by: Timothy Gu <timothygu99@gmail.com>
2013-09-22 15:26:26 -07:00
Michael Niedermayer
dbb534cea6 avformat/matroskadec: check out_samplerate before using it in av_rescale()
Prevent assertion failure with damaged input

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 338f8b2eaf36f078eb5cc26ac10e651dc4c48243)
2013-09-18 15:07:49 -03:00
James Almer
414d75b8bc matroskadec: Improve TTA duration calculation
Calculate the duration as accurately as possible to improve decoding of samples
where the last frame is smaller than the rest.

Signed-off-by: James Almer <jamrial@gmail.com>
Approved-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit af248fa1174200acb537a6ab1198bb2fed38e884)
2013-09-18 15:06:31 -03:00
Michael Niedermayer
c52a25e03b avformat/oggparsevorbis: fix leak of tt
Fixes CID1061059
Fixes fate
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

(cherry picked from commit f3b7f470701224086d06012f1d9a31864abe9300)
2013-09-18 14:53:25 -03:00
Michael Niedermayer
c7966bf795 avformat/oggparsevorbis: fix leak of ct
Fixes CID1061058
Fixes fate

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d0a882ab1d2a4197da1edd77450af30e2da3460e)
2013-09-18 14:52:58 -03:00
Clément Bœsch
0f429392cf avcodec/assenc: fix potential overread.
(cherry picked from commit 860a0810583f54ccbde912aebda8711f18eab8eb)

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-09-15 22:21:05 +02:00
Clément Bœsch
2dc6c5d462 avcodec/srtdec: fix potential overread.
(cherry picked from commit 3a54c221d574ec944db1eddf9df895808f32bf9e)

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-09-15 22:21:05 +02:00
Clément Bœsch
dc0403530e avformat/subtitles: add a next line jumper and use it.
This fixes a bunch of possible overread in avformat with the idiom p +=
strcspn(p, "\n") + 1 (strcspn() can focus on the trailing '\0' if no
'\n' is found, so the +1 leads to an overread).

Note on lavf/matroskaenc: no extra subtitles.o Makefile dependency is
added because only the header is required for ff_subtitles_next_line().

Note on lavf/mpsubdec: code gets slightly complex to avoid an infinite
loop in the probing since there is no more forced increment.

NOTE:
Code of function ff_subtitles_next_line fixed by Alexander Strasser.

The original code from master did test the wrong character, but was
corrected by a subsequent commit. That commit however is not backported,
so it had to be fixed in this commit for the backport.

Conflicts:
	libavformat/mpl2dec.c

(cherry picked from commit 90fc00a623de44e137fe1601b91356e8cd8bdd54)

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-09-15 22:21:05 +02:00
Clément Bœsch
59147be24f avformat/srtdec: skip initial random line breaks.
I found a bunch of (recent) SRT files in the wild with 3 to 10 line
breaks at the beginning.

(cherry picked from commit cfcd55db164e0acc0c30b2cf084e6eebe9741d34)

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-09-15 22:21:04 +02:00
Carl Eugen Hoyos
09bc4be3db Use rc_max_rate if no video bit_rate was specified when muxing mxf_d10.
Fixes ticket #2945.

Reviewed-by: Matthieu Bouron
(cherry picked from commit d73565d5ddf41d4b7805327cdb271c59d8c3fc59)
2013-09-12 23:11:47 +02:00
Carl Eugen Hoyos
aaef59d535 Store the video bit_rate in the context when muxing mxf.
This will allow using rc_max_rate if no bit_rate is specified (on remuxing).

Reviewed-by: Matthieu Bouron
(cherry picked from commit 52cf08b4c8859f7cac010a7a59f7aa369384ad85)
2013-09-12 23:06:38 +02:00
Clément Bœsch
283e070877 avformat/subtitles: check lower bound for duration overlap seeking.
(cherry picked from commit 1ca4bf930bab681a79fb591330043675c7cfd798)
2013-09-10 21:32:56 +02:00
Clément Bœsch
4e0c29451b avformat/vobsub: fix seeking.
(cherry picked from commit f8678dcef3c5b0ea82e898e1f419863409fa135f)
2013-09-10 21:32:52 +02:00
Paul B Mahol
f5ae34250a avformat/matroskaenc: remove bogus prores tag
Fixes: ffmpeg -i input -c:v prores output.mkv

Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit 14851ca5f5a3af140085e82589e28e06c7cdefdc)

Conflicts:
	libavformat/matroskaenc.c
2013-09-08 21:44:06 +02:00
Carl Eugen Hoyos
fc4c29bc6e Read h264 headers from v4l2 to allow stream-copying.
Fixes ticket #2882.
Analyzed and tested by William C Bonner.
(cherry picked from commit e337c9d56408dc00a15887309488a1ff5cb06ba3)
2013-09-05 22:49:52 +02:00
Paul B Mahol
6158eec53f w64dec: fix end position of summarylist guid
Noticed-by: James Almer

Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit 3e36dc8626f4721ea749286dae40169ee5cb7d04)
2013-09-05 22:49:28 +02:00
Paul B Mahol
6c0fef5762 w64dec: fix skipping of unknown guids
Regression since 14d50c1.
Fixes #2932.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit 79b70e47a463057a3a48353ee1dd58671c11f86c)
2013-09-05 22:49:16 +02:00
Carl Eugen Hoyos
f5a4bd23e9 Avoid a deadlock when decoding wma.
Fixes ticket #2925.
(cherry picked from commit ec8a4841f7e81040f9a2757f23e70dff5e6b33a4)
2013-09-02 08:54:32 +02:00
Carl Eugen Hoyos
8e1760f37f avcodec/ffv1dec: reorganize thread init/update
This moves some allocations to init, reducing possible failure modes in update.
Always copies from the previous context instead of just during init

Fixes Ticket2923

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 21dc3a3cc23c624630a4379f08a6efe0230de37a)
2013-09-02 08:49:59 +02:00
Michael Niedermayer
1da5ab751f avcodec/ffv1dec: move initial_states init to init_thread_copy()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c72cca5a44d8a83e097a97a13990d421ba7a4c5d)
2013-09-02 08:47:50 +02:00
Michael Niedermayer
237ef710a1 avformat/lxfdec: use a parser to parse video frame headers
lxf needs a parser (or would need to set a few fields explicitly).
Fixes Ticket2917

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8349be852be7f68fe0590584fd46c4d5f1c16b3d)
2013-09-01 09:59:14 +02:00
Michael Niedermayer
be47e93134 avcodec/h264: set er.ref_count earlier
Fixes Ticket2910

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 93cf7b01950b9d8e1646227752b522d0275d32df)
2013-09-01 09:58:55 +02:00
Michael Niedermayer
1821c849da avformat/avidec: match first index and first packet size=0 handling
Fixes Ticket2861

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 227a0eb5a92409572f2cecde6137529b83e7d495)

Conflicts:
	libavformat/avidec.c
2013-08-31 09:47:23 +02:00
Michael Niedermayer
a4522ae516 avcodec/pngdsp: fix (un)signed type in end comparission
Fixes out of array accesses
Fixes Ticket2919

Found_by: ami_stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 86736f59d6a527d8bc807d09b93f971c0fe0bb07)
2013-08-30 23:35:47 +02:00
Michael Niedermayer
c7ee4bc016 ffv1dec: check that global parameters dont change in version 0/1
Such changes are not allowed nor supported

Fixes Ticket2906

Found-by: ami_stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 547d690d676064069d44703a1917e0dab7e33445)
2013-08-30 11:23:25 +02:00
Lukasz Marek
0cabb95811 lavf/ftp: fix possible crash
(cherry picked from commit f3ace37a3b8c93218630a37b7df4dc195f1215a9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-29 02:16:41 +02:00
Michael Niedermayer
a1b32533aa jpeg2000: fix dereferencing invalid pointers
Found-by: Laurent Butti <laurentb@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 912ce9dd2080c5837285a471d750fa311e09b555)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-29 02:13:25 +02:00
Michael Niedermayer
e2eb0d2326 avcodec/jpeg2000dec: Check cdx/y values more carefully
Some invalid values where not handled correctly in the later pixel
format matching code.
Fixes out of array accesses
Fixes Ticket2848

Found-by: Piotr Bandurski <ami_stuff@o2.pl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8bb11c3ca77b52e05a9ed1496a65f8a76e6e2d8f)

Conflicts:

	libavcodec/jpeg2000dec.c
2013-08-29 02:13:25 +02:00
Michael Niedermayer
acac6b0d69 avcodec/rpza: Perform pointer advance and checks before using the pointers
Fixes out of array accesses
Fixes Ticket2850

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3819db745da2ac7fb3faacb116788c32f4753f34)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-29 02:13:25 +02:00
Michael Niedermayer
deb8d0d6a1 avcodec/flashsv: check diff_start/height
Fixes out of array accesses
Fixes Ticket2844

Found-by: ami_stuff
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 880c73cd76109697447fbfbaa8e5ee5683309446)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-29 02:13:25 +02:00
Michael Niedermayer
3e817d91ef jpeg2000: check log2_cblk dimensions
Fixes out of array access
Fixes Ticket2895

Found-by: Piotr Bandurski <ami_stuff@o2.pl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9a271a9368eaabf99e6c2046103acb33957e63b7)
2013-08-25 15:57:13 +02:00
Michael Niedermayer
e231f0fade swr/rematrix: Fix handling of AV_CH_LAYOUT_STEREO_DOWNMIX output
Fixes Ticket2859

Note, testcases related to the downmix channels are welcome.
(id like to make sure this is working correctly now, as obviously it didnt
 work before ...)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c56d4dab039b352961cca298d753b04e2f2fd990)
2013-08-20 18:39:23 +02:00
Michael Niedermayer
f0f55e6726 swr: clean layouts before checking sanity
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6dfffe92004dfd8c79d18791f28a2b1c7e387845)
2013-08-20 18:39:19 +02:00
Michael Niedermayer
61dc8494d7 movenc: ilbc needs audio_vbr set.
Without this the block_align or bitrate value is not available to the decoder

Fixes Ticket2858

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3d64845600c6486a2706b118a81805f3bf4d3db5)
2013-08-20 18:38:05 +02:00
Stephen Hutchinson
423b87d621 Revert "doc/RELEASE_NOTES: add a note about AVISynth"
This reverts commit 3aa2257d240a5a0eb94014b9113dd91730786886.
2013-08-18 23:14:04 -04:00
Stephen Hutchinson
8d9568b4a1 avisynth: Support video input from AviSynth 2.5 properly.
Uses the 2.5 compatibility header included with the variant of
FFMS2 that uses AviSynth's C-interface. A copy of this header is
now provided in compat/avisynth.

avs_get_row_size_p and avs_get_height_p changed between versions
2.5 and 2.6. Since the avisynth_c.h header that avformat uses
assumes AviSynth 2.6, it would cause 2.5 to crash if given any
kind of real video (the Version() function was known to work,
though).

AvxSynth was unaffected by this issue because, despite being based
on AviSynth 2.5.8 and using 2.5.8's interface version number of 3,
it actually uses 2.6's versions of these functions.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-18 23:13:27 -04:00
Michael Niedermayer
acf511de34 avcodec/g2meet: fix src pointer checks in kempf_decode_tile()
Fixes Ticket2842

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2960576378d17d71cc8dccc926352ce568b5eec1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-11 00:14:17 +02:00
Michael Niedermayer
fd2951bb53 update for 2.0.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Michael Niedermayer
fa004f4854 MAINTAINERS: add Alexander Strasser for the server
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c11c180132b3e0038143dc9ba4dfd7287d1509c7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Michael Niedermayer
1155bdb754 MAINTAINERS: remove myself from movenc, 2 maintainers should be enough
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 55a88daf6ff1e09994bff39fac313a38a35b0055)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Matthieu Bouron
01838c5732 MAINTAINERS: add myself as maintainer for lavf/aiff* and lavf/movenc.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 88a1ff22336ebfd62b0d1d920e0e7d49f7bd1ece)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Michael Niedermayer
f09f33031b MAINTAINERS: alphabetical sort
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 81f4d55c3669733318b2f0e61d10d9efe9b00074)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Michael Niedermayer
b2a9f64e1b MAINTAINERS: Add some maintainers for parts of libavutil
Developers added are active and in the copyright of the specified files,

If anyone wants to maintain anything else, send a patch that adds you to
MAINTAINERS.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b45b1d7af93bd5b8b3824edeae3d1d54644a5361)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Michael Niedermayer
15ea618ef6 MAINTAINERS: order libavutil entries alphabetically
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 48188a512068e77fbf94ec1a250870260a1ff1ae)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Michael Niedermayer
ec33423273 MAINTAINERS: drop 1.1 from the releases that i maintain
There seems to be no need to continue maintaining it, people can easily
upgrade to 1.2

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5fe5f020b6cfb0b7137d12aab10db936cc08836f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Michael Niedermayer
d5dd54df69 MAINTAINERS: add myself as maintainer for the interface code to swresample & swscale in libavfilter
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5ad4e29337483bbd3337bc47db4ad181d7e2da00)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 23:58:48 +02:00
Stephen Hutchinson
e7a4c34e7c avisynth: Exit gracefully when trying to serve video from v2.5.8.
'Fixes' ticket #2526 insofar as it stops 2.5.8 from crashing and
tells the user to upgrade to 2.6 if they want to make video input
work. A real solution to #2526 would be to get video input from
2.5.8 to work right.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9db353bc4727d2a184778c110cf4ea0b9d1616cb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 21:45:11 +02:00
Stephen Hutchinson
2881bfbfd6 avisynth: Cosmetics
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f277d6bf42329d481f58b6147f6dc4130f198ba5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 21:45:03 +02:00
Paul B Mahol
80fb38153e sgidec: safer check for buffer overflow
Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit 86e722ab97d7f5f0552c8a0958f7910dfcf3c5b7)

Conflicts:

	libavcodec/sgidec.c

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 02:03:37 +02:00
Paul B Mahol
b79f337f8a ttaenc: fix packet size
Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit bc2187cfdb5eeb82e3caf42a81a00d1ee4c16d8e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-10 02:01:40 +02:00
Michael Niedermayer
f593ac1c21 matroskaenc: simplify mkv_check_tag()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 066111bf19518a9f4d836991b34dbfc5ab72a41a)
2013-08-09 16:28:39 -03:00
James Almer
baf92305a6 lavf/matroskaenc: Check for valid metadata before creating tags
Tags must have at least one SimpleTag element to be spec conformant.
Updated lavf-mkv and seek-lavf-mkv FATE references as the tests were affected by
this.

Fixes ticket #2785

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 088ed5314694031e07e02e9d364c9d486a60e704)
2013-08-09 16:28:31 -03:00
Michael Niedermayer
d6d168e87b avfilter/vf_separatefields: fix ;;
Found-by: llogan
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 74561680cd01f36a2b225efb529bcd5729b65d32)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-09 13:58:45 +02:00
Michael Niedermayer
50f9c4acc3 avformat/paf: Fix integer overflow and out of array read
Found-by:  Laurent Butti <laurentb@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f58cd2867a8af2eed13acdd21d067b48249b14a1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-09 13:53:27 +02:00
Michael Niedermayer
211374e52a avutil/mem: Fix flipped condition
Fixes return code and later null pointer dereference

Found-by: Laurent Butti <laurentb@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c94f9e854228e0ea00e1de8769d8d3f7cab84a55)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-09 13:53:25 +02:00
Michael Niedermayer
1bf2461765 avfilter: fix plane validity checks
Fixes out of array accesses

(cherry picked from commit e43a0a232dbf6d3c161823c2e07c52e76227a1bc)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-09 13:53:14 +02:00
Michael Niedermayer
64444cd578 avcodec/kmvc: fix MV checks
Fixes Ticket2813
Fixes regression since 70b5583

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3cd8aaa2b2e78faf039691e1c31ff4f8d94e3bc6)
2013-07-31 02:54:26 +02:00
Paul B Mahol
0047a31090 Revert "pnm: remove nonsense code"
Breaks decoding pgms with 255 < maxval < 65535.

Found-by: Carl Eugen Hoyos <cehoyos@ag.or.at>.

This reverts commit a0348d0966a81a66f3a1bf061576b24d5296b933.
(cherry picked from commit 768e40b451a459fefaceed6b1b3d6e70c93596ac)
2013-07-29 00:05:01 +02:00
Michael Niedermayer
d73ce6cb56 jpeg2000dec: Support non subsampled 9-16bit planar pixel formats
This applies changes similar to fc6de70c44be05eb0368ab519bfb790431d8dee5
to the >8bit codepath

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1434df3b93fde086be729d174ffbbee1e25792b3)
2013-07-28 04:09:21 +02:00
Michael Niedermayer
9a6d3eee59 jpeg2000dec: silence unused variable warning
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit db33010483a024f1343bfb516415fe9ef77e82d9)
2013-07-28 04:09:11 +02:00
Michael Niedermayer
8b221d60fa jpeg2000dec: Support non subsampled 8bit planar pixel formats
Fixes file2.jp2

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fc6de70c44be05eb0368ab519bfb790431d8dee5)
2013-07-28 04:08:10 +02:00
Michael Niedermayer
9da9b36435 jpeg2000dec: parse CDEF
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Conflicts:

	libavcodec/jpeg2000dec.c

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 99de97cabf35c035b628943b865ab991583a6ea5)
2013-07-28 04:07:36 +02:00
Carl Eugen Hoyos
09b33f9a82 Fix pix_fmt detection in the native jpeg2000 decoder.
Based on b7a928b by Michael Bradshaw.
Fixes ticket #2683.

Reviewed-by: Nicolas Bertrand
(cherry picked from commit b39a6bbe7f43710d8e7163757c5aeef596878712)
2013-07-28 04:07:20 +02:00
Michael Niedermayer
fa6b6dad3d jpeg2000: fix overflow in dequantization
Fixes decoding of file generated with:
ffmpeg -f lavfi -i smptehdbars=hd720 -pix_fmt rgb48 /tmp/o.jp2

Reviewed-by: Nicolas BERTRAND <nicoinattendu@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f57119b8e58cb5437c3ab40d797293ecb9b4a894)
2013-07-28 04:07:10 +02:00
Nicolas Bertrand
e0d88cfd18 jpeg2000: Initialize only once mqc arrays
Increases encoding and decoding speed

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dd1382ac9534bd9b5ec7833eed9ab6f383e68a50)
2013-07-28 04:07:02 +02:00
Michael Niedermayer
18043e3d22 avformat/dtsdec: Improve probe, reject things looking like analoge signals
Fixes Ticket2810

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6663205338f57eedb2392263dde48e2717c6e980)
2013-07-26 12:18:46 +02:00
Rémi Denis-Courmont
ccf470fdb6 mpeg12: Ignore slice threading if hwaccel is active
Slice threading does not work with hardware acceleration, as decoding
is per-picture.  This fixes Bugzilla #542.

Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit 93a51984a27f3ba84d4e6f13d0c704ee9891603e)

Conflicts:
	libavcodec/mpeg12dec.c
2013-07-26 11:35:36 +02:00
Michael Niedermayer
8f9bc6f2ce swscale/input: fix 16bit gbrp input
Fixes Ticket2793

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a4b55bbb6f8ba055bba1493f6872792503e47563)
2013-07-19 14:50:16 +02:00
Lukasz Marek
fcab45f39b ftp: fix interrupt callback misuse
FTP protocol used interrupt callback to simulate nonblock
operation which is a misuse of this callback.

This commit make FTP protocol fully blocking and removes
invalid usage of interrutp callback

Also adds support for multiline responses delimited with dashes
(cherry picked from commit 247e658784ead984f96021acb9c95052ba599f26)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-18 01:54:24 +02:00
Michael Niedermayer
bc44d06c3d avformat/matroskadec: Detect conflicting sample rate/default_duration
Fixes Ticket2508

Thanks-to: Moritz Bunkus
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6158a3bcdf52fafc1d9ae9eb358a56c614b23aa3)
2013-07-16 11:43:21 +02:00
Michael Niedermayer
7740e36a89 mjpegdec: Fix used quant index for gbr
Fixes Ticket1651

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 15cee5e5628a80a51d1bb298c8b838ddefd75a88)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:06:57 +02:00
Michael Niedermayer
6127f792f9 mjpegdec: initialize source variables before gbr remap
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 94e86ae15a4328b22f6f103b9fd9b6a9ee0c676c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:06:57 +02:00
Carl Eugen Hoyos
fd2cf9c45d Suggest recompilation with openssl or gnutls if the https protocol is not found.
Fixes ticket #2765.
(cherry picked from commit 1db88c33f2c0225aae160cc412b62dfaa3a34cbc)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:06:57 +02:00
Carl Eugen Hoyos
fc3dec8b62 lavf/utils.c: Avoid a null pointer dereference on oom after duration_error allocation.
(cherry picked from commit c9eb5c9751c88caaed62af5ffe908fe545022e7e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:06:56 +02:00
Luca Barbato
a7315116dd wmavoice: conceal clearly corrupted blocks
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit d14a26edb7c4487df581f11e5c6911dc0e623d08)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:06:56 +02:00
Michael Niedermayer
37268dcc86 avcodec/qdm2: initialize sign_bits
Fixes non deterministic output

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8f09957194b8d7a3ea909647e22eaf1389b6f5c4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:03:30 +02:00
Michael Niedermayer
ea28e74205 avcodec/qdm2: store bits in an integer instead of float variable
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fbe159e85079f1f94a7201dc210b155dba7ff4a6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:02:56 +02:00
Piotr Bandurski
c1c84f0a55 avformat/utils: avformat_find_stream_info set value for ret in case of oom
without it FFmpeg didn't display any error message when oom event occured

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b0509563347b81d7b0a2474e56abbc9c99043688)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:02:33 +02:00
Paul B Mahol
56bf38859b lavfi/aconvert: unbreak
Even if its deprecated, it should still work correctly.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
(cherry picked from commit bc95b9428950cd672162bcb2bb98fbecad52a5b3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:02:00 +02:00
Piotr Bandurski
1cda4aa1e0 avformat/utils: avformat_find_stream_info fix a crash in case of oom
fixes ticket #2767

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ccf9211e29bdfad02faf93575bf39a8f89c30647)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-11 12:00:39 +02:00
Michael Niedermayer
9711b52739 avfilter/af_earwax: Fix out of array accesses on odd packets
Found-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0a3a0edd52b98aec27d1b8c63c85cb52ff46d40e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-10 19:07:50 +02:00
5772 changed files with 194693 additions and 706882 deletions

1
.gitattributes vendored
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*.pnm -diff -text

57
.gitignore vendored
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@ -1,6 +1,5 @@
*.a
*.o
*.o.*
*.d
*.def
*.dll
@ -16,11 +15,10 @@
*.pdb
*.so
*.so.*
*.swp
*.ver
*-example
*-test
*_g
\#*
.\#*
/.config
/.version
/ffmpeg
@ -29,7 +27,52 @@
/ffserver
/config.*
/coverage.info
/avversion.h
/version.h
/doc/*.1
/doc/*.3
/doc/*.html
/doc/*.pod
/doc/config.texi
/doc/avoptions_codec.texi
/doc/avoptions_format.texi
/doc/examples/decoding_encoding
/doc/examples/demuxing
/doc/examples/filtering_audio
/doc/examples/filtering_video
/doc/examples/metadata
/doc/examples/muxing
/doc/examples/pc-uninstalled
/doc/examples/resampling_audio
/doc/examples/scaling_video
/doc/fate.txt
/doc/doxy/html/
/doc/print_options
/lcov/
/src
/mapfile
/libavcodec/*_tablegen
/libavcodec/*_tables.c
/libavcodec/*_tables.h
/libavutil/avconfig.h
/tests/audiogen
/tests/base64
/tests/data/
/tests/rotozoom
/tests/tiny_psnr
/tests/tiny_ssim
/tests/videogen
/tests/vsynth1/
/tools/aviocat
/tools/ffbisect
/tools/bisect.need
/tools/cws2fws
/tools/fourcc2pixfmt
/tools/ffescape
/tools/ffeval
/tools/ffhash
/tools/graph2dot
/tools/ismindex
/tools/pktdumper
/tools/probetest
/tools/qt-faststart
/tools/trasher
/tools/seek_print
/tools/zmqsend

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@ -1,26 +0,0 @@
language: c
sudo: false
os:
- linux
- osx
addons:
apt:
packages:
- yasm
- diffutils
compiler:
- clang
- gcc
cache:
directories:
- ffmpeg-samples
before_install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install yasm; fi
script:
- mkdir -p ffmpeg-samples
- ./configure --samples=ffmpeg-samples --cc=$CC
- make -j 8
- make fate-rsync
- make check -j 8

409
Changelog
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@ -1,390 +1,6 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>:
- 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
- 16-bit support in selectivecolor filter
- OpenH264 decoder wrapper
- MediaCodec hwaccel
version 3.1:
- DXVA2-accelerated HEVC Main10 decoding
- fieldhint filter
- loop video filter and aloop audio filter
- Bob Weaver deinterlacing filter
- firequalizer filter
- datascope filter
- bench and abench filters
- ciescope filter
- protocol blacklisting API
- MediaCodec H264 decoding
- VC-2 HQ RTP payload format (draft v1) depacketizer and packetizer
- VP9 RTP payload format (draft v2) packetizer
- AudioToolbox audio decoders
- AudioToolbox audio encoders
- coreimage filter (GPU based image filtering on OSX)
- libdcadec removed
- bitstream filter for extracting DTS core
- ADPCM IMA DAT4 decoder
- musx demuxer
- aix demuxer
- remap filter
- hash and framehash muxers
- colorspace filter
- hdcd filter
- readvitc filter
- VAAPI-accelerated format conversion and scaling
- libnpp/CUDA-accelerated format conversion and scaling
- Duck TrueMotion 2.0 Real Time decoder
- Wideband Single-bit Data (WSD) demuxer
- VAAPI-accelerated H.264/HEVC/MJPEG encoding
- DTS Express (LBR) decoder
- Generic OpenMAX IL encoder with support for Raspberry Pi
- IFF ANIM demuxer & decoder
- Direct Stream Transfer (DST) decoder
- loudnorm filter
- MTAF demuxer and decoder
- MagicYUV decoder
- OpenExr improvements (tile data and B44/B44A support)
- BitJazz SheerVideo decoder
- CUDA CUVID H264/HEVC decoder
- 10-bit depth support in native utvideo decoder
- libutvideo wrapper removed
- YUY2 Lossless Codec decoder
- VideoToolbox H.264 encoder
version 3.0:
- Common Encryption (CENC) MP4 encoding and decoding support
- DXV decoding
- extrastereo filter
- ocr filter
- alimiter filter
- stereowiden filter
- stereotools filter
- rubberband filter
- tremolo filter
- agate filter
- chromakey filter
- maskedmerge filter
- Screenpresso SPV1 decoding
- chromaprint fingerprinting muxer
- ffplay dynamic volume control
- displace filter
- selectivecolor filter
- extensive native AAC encoder improvements and removal of experimental flag
- ADPCM PSX decoder
- 3dostr, dcstr, fsb, genh, vag, xvag, ads, msf, svag & vpk demuxer
- zscale filter
- wve demuxer
- zero-copy Intel QSV transcoding in ffmpeg
- shuffleframes filter
- SDX2 DPCM decoder
- vibrato filter
- innoHeim/Rsupport Screen Capture Codec decoder
- ADPCM AICA decoder
- Interplay ACM demuxer and audio decoder
- XMA1 & XMA2 decoder
- realtime filter
- anoisesrc audio filter source
- IVR demuxer
- compensationdelay filter
- acompressor filter
- support encoding 16-bit RLE SGI images
- apulsator filter
- sidechaingate audio filter
- mipsdspr1 option has been renamed to mipsdsp
- aemphasis filter
- mips32r5 option has been removed
- mips64r6 option has been removed
- DXVA2-accelerated VP9 decoding
- SOFAlizer: virtual binaural acoustics filter
- VAAPI VP9 hwaccel
- audio high-order multiband parametric equalizer
- automatic bitstream filtering
- showspectrumpic filter
- libstagefright support removed
- spectrumsynth filter
- ahistogram filter
- only seek with the right mouse button in ffplay
- toggle full screen when double-clicking with the left mouse button in ffplay
- afftfilt filter
- convolution filter
- libquvi support removed
- support for dvaudio in wav and avi
- libaacplus and libvo-aacenc support removed
- Cineform HD decoder
- new DCA decoder with full support for DTS-HD extensions
- significant performance improvements in Windows Television (WTV) demuxer
- nnedi deinterlacer
- streamselect video and astreamselect audio filter
- swaprect filter
- metadata video and ametadata audio filter
- SMPTE VC-2 HQ profile support for the Dirac decoder
- SMPTE VC-2 native encoder supporting the HQ profile
version 2.8:
- colorkey video filter
- BFSTM/BCSTM demuxer
- little-endian ADPCM_THP decoder
- Hap decoder and encoder
- DirectDraw Surface image/texture decoder
- ssim filter
- optional new ASF demuxer
- showvolume filter
- Many improvements to the JPEG 2000 decoder
- Go2Meeting decoding support
- adrawgraph audio and drawgraph video filter
- removegrain video filter
- Intel QSV-accelerated MPEG-2 video and HEVC encoding
- Intel QSV-accelerated MPEG-2 video and HEVC decoding
- Intel QSV-accelerated VC-1 video decoding
- libkvazaar HEVC encoder
- erosion, dilation, deflate and inflate video filters
- Dynamic Audio Normalizer as dynaudnorm filter
- Reverse video and areverse audio filter
- Random filter
- deband filter
- AAC fixed-point decoding
- sidechaincompress audio filter
- bitstream filter for converting HEVC from MP4 to Annex B
- acrossfade audio filter
- allyuv and allrgb video sources
- atadenoise video filter
- OS X VideoToolbox support
- aphasemeter filter
- showfreqs filter
- vectorscope filter
- waveform filter
- hstack and vstack filter
- Support DNx100 (1440x1080@8)
- VAAPI hevc hwaccel
- VDPAU hevc hwaccel
- framerate filter
- Switched default encoders for webm to VP9 and Opus
- Removed experimental flag from the JPEG 2000 encoder
version 2.7:
- FFT video filter
- TDSC decoder
- DTS lossless extension (XLL) decoding (not lossless, disabled by default)
- showwavespic filter
- DTS decoding through libdcadec
- Drop support for nvenc API before 5.0
- nvenc HEVC encoder
- Detelecine filter
- Intel QSV-accelerated H.264 encoding
- MMAL-accelerated H.264 decoding
- basic APNG encoder and muxer with default extension "apng"
- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
- WebM Live Chunk Muxer
- nvenc level and tier options
- chorus filter
- Canopus HQ/HQA decoder
- Automatically rotate videos based on metadata in ffmpeg
- improved Quickdraw compatibility
- VP9 high bit-depth and extended colorspaces decoding support
- WebPAnimEncoder API when available for encoding and muxing WebP
- Direct3D11-accelerated decoding
- Support Secure Transport
- Multipart JPEG demuxer
version 2.6:
- nvenc encoder
- 10bit spp filter
- colorlevels filter
- RIFX format for *.wav files
- RTP/mpegts muxer
- non continuous cache protocol support
- tblend filter
- cropdetect support for non 8bpp, absolute (if limit >= 1) and relative (if limit < 1.0) threshold
- Camellia symmetric block cipher
- OpenH264 encoder wrapper
- VOC seeking support
- Closed caption Decoder
- fspp, uspp, pp7 MPlayer postprocessing filters ported to native filters
- showpalette filter
- Twofish symmetric block cipher
- Support DNx100 (960x720@8)
- eq2 filter ported from libmpcodecs as eq filter
- removed libmpcodecs
- Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range
- ported softpulldown filter from libmpcodecs as repeatfields filter
- dcshift filter
- RTP depacketizer for loss tolerant payload format for MP3 audio (RFC 5219)
- RTP depacketizer for AC3 payload format (RFC 4184)
- palettegen and paletteuse filters
- VP9 RTP payload format (draft 0) experimental depacketizer
- RTP depacketizer for DV (RFC 6469)
- DXVA2-accelerated HEVC decoding
- AAC ELD 480 decoding
- Intel QSV-accelerated H.264 decoding
- DSS SP decoder and DSS demuxer
- Fix stsd atom corruption in DNxHD QuickTimes
- Canopus HQX decoder
- RTP depacketization of T.140 text (RFC 4103)
- Port MIPS optimizations to 64-bit
version 2.5:
- HEVC/H.265 RTP payload format (draft v6) packetizer
- SUP/PGS subtitle demuxer
- ffprobe -show_pixel_formats option
- CAST128 symmetric block cipher, ECB mode
- STL subtitle demuxer and decoder
- libutvideo YUV 4:2:2 10bit support
- XCB-based screen-grabber
- UDP-Lite support (RFC 3828)
- xBR scaling filter
- AVFoundation screen capturing support
- ffserver supports codec private options
- creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
- WebP muxer with animated WebP support
- zygoaudio decoding support
- APNG demuxer
- postproc visualization support
version 2.4:
- Icecast protocol
- ported lenscorrection filter from frei0r filter
- large optimizations in dctdnoiz to make it usable
- ICY metadata are now requested by default with the HTTP protocol
- support for using metadata in stream specifiers in fftools
- LZMA compression support in TIFF decoder
- H.261 RTP payload format (RFC 4587) depacketizer and experimental packetizer
- HEVC/H.265 RTP payload format (draft v6) depacketizer
- added codecview filter to visualize information exported by some codecs
- Matroska 3D support thorugh side data
- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
- silenceremove filter
version 2.3:
- AC3 fixed-point decoding
- shuffleplanes filter
- subfile protocol
- Phantom Cine demuxer
- replaygain data export
- VP7 video decoder
- Alias PIX image encoder and decoder
- Improvements to the BRender PIX image decoder
- Improvements to the XBM decoder
- QTKit input device
- improvements to OpenEXR image decoder
- support decoding 16-bit RLE SGI images
- GDI screen grabbing for Windows
- alternative rendition support for HTTP Live Streaming
- AVFoundation input device
- Direct Stream Digital (DSD) decoder
- Magic Lantern Video (MLV) demuxer
- On2 AVC (Audio for Video) decoder
- support for decoding through DXVA2 in ffmpeg
- libbs2b-based stereo-to-binaural audio filter
- libx264 reference frames count limiting depending on level
- native Opus decoder
- display matrix export and rotation API
- WebVTT encoder
- showcqt multimedia filter
- zoompan filter
- signalstats filter
- hqx filter (hq2x, hq3x, hq4x)
- flanger filter
- Image format auto-detection
- LRC demuxer and muxer
- Samba protocol (via libsmbclient)
- WebM DASH Manifest muxer
- libfribidi support in drawtext
version 2.2:
- HNM version 4 demuxer and video decoder
- Live HDS muxer
- setsar/setdar filters now support variables in ratio expressions
- elbg filter
- string validation in ffprobe
- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
- complete Voxware MetaSound decoder
- remove mp3_header_compress bitstream filter
- Windows resource files for shared libraries
- aeval filter
- stereoscopic 3d metadata handling
- WebP encoding via libwebp
- ATRAC3+ decoder
- VP8 in Ogg demuxing
- side & metadata support in NUT
- framepack filter
- XYZ12 rawvideo support in NUT
- Exif metadata support in WebP decoder
- OpenGL device
- Use metadata_header_padding to control padding in ID3 tags (currently used in
MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
- Mirillis FIC video decoder
- Support DNx444
- libx265 encoder
- dejudder filter
- Autodetect VDA like all other hardware accelerations
- aliases and defaults for Ogg subtypes (opus, spx)
version 2.1:
- aecho filter
- perspective filter ported from libmpcodecs
- ffprobe -show_programs option
- compand filter
- RTMP seek support
- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
even when used as an input option. Previous behavior can be restored with
the -noaccurate_seek option.
- ffmpeg -t option can now be used for inputs, to limit the duration of
data read from an input file
- incomplete Voxware MetaSound decoder
- read EXIF metadata from JPEG
- DVB teletext decoder
- phase filter ported from libmpcodecs
- w3fdif filter
- Opus support in Matroska
- FFV1 version 1.3 is stable and no longer experimental
- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
more consistent with other muxers.
- adelay filter
- pullup filter ported from libmpcodecs
- ffprobe -read_intervals option
- Lossless and alpha support for WebP decoder
- Error Resilient AAC syntax (ER AAC LC) decoding
- Low Delay AAC (ER AAC LD) decoding
- mux chapters in ASF files
- SFTP protocol (via libssh)
- libx264: add ability to encode in YUVJ422P and YUVJ444P
- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
- make decoding alpha optional for prores, ffv1 and vp6 by setting
the skip_alpha flag.
- ladspa wrapper filter
- native VP9 decoder
- dpx parser
- max_error_rate parameter in ffmpeg
- PulseAudio output device
- ReplayGain scanner
- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
- Linux framebuffer output device
- HEVC decoder
- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
- mergeplanes filter
version 2.0:
- curves filter
@ -397,7 +13,7 @@ version 2.0:
- 10% faster aac encoding on x86 and MIPS
- sine audio filter source
- WebP demuxing and decoding support
- ffmpeg options -filter_script and -filter_complex_script, which allow a
- new ffmpeg options -filter_script and -filter_complex_script, which allow a
filtergraph description to be read from a file
- OpenCL support
- audio phaser filter
@ -405,7 +21,7 @@ version 2.0:
- libquvi demuxer
- uniform options syntax across all filters
- telecine filter
- interlace filter
- new interlace filter
- smptehdbars source
- inverse telecine filters (fieldmatch and decimate)
- colorbalance filter
@ -523,7 +139,7 @@ version 1.1:
- JSON captions for TED talks decoding support
- SOX Resampler support in libswresample
- aselect filter
- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder
- SGI RLE 8-bit decoder
- Silicon Graphics Motion Video Compressor 1 & 2 decoder
- Silicon Graphics Movie demuxer
- apad filter
@ -567,9 +183,7 @@ version 1.0:
- RTMPE protocol support
- RTMPTE protocol support
- showwaves and showspectrum filter
- LucasArts SMUSH SANM playback support
- LucasArts SMUSH VIMA audio decoder (ADPCM)
- LucasArts SMUSH demuxer
- LucasArts SMUSH playback support
- SAMI, RealText and SubViewer demuxers and decoders
- Heart Of Darkness PAF playback support
- iec61883 device
@ -693,7 +307,6 @@ version 0.10:
- ffwavesynth decoder
- aviocat tool
- ffeval tool
- support encoding and decoding 4-channel SGI images
version 0.9:
@ -742,7 +355,7 @@ easier to use. The changes are:
all the stream in the first input file, except for the second audio
stream'.
* There is a new option -c (or -codec) for choosing the decoder/encoder to
use, which makes it possible to precisely specify target stream(s) consistently with
use, which allows to precisely specify target stream(s) consistently with
other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
libvorbis sets the codec for the first audio stream and -c copy copies all
the streams without reencoding. Old -vcodec/-acodec/-scodec options are now
@ -931,8 +544,8 @@ version 0.8:
- showinfo filter added
- SMPTE 302M AES3 audio decoder
- Apple Core Audio Format muxer
- 9 bits and 10 bits per sample support in the H.264 decoder
- 9 bits and 10 bits FFV1 encoding / decoding
- 9bit and 10bit per sample support in the H.264 decoder
- 9bit and 10bit FFV1 encoding / decoding
- split filter added
- select filter added
- sdl output device added
@ -994,7 +607,7 @@ version 0.6:
- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
- WMA Pro decoder
- Core Audio Format demuxer
- ATRAC1 decoder
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder
@ -1094,7 +707,7 @@ version 0.5:
- MXF demuxer
- VC-1/WMV3/WMV9 video decoder
- MacIntel support
- AviSynth support
- AVISynth support
- VMware video decoder
- VP5 video decoder
- VP6 video decoder
@ -1122,7 +735,7 @@ version 0.5:
- Interplay C93 demuxer and video decoder
- Bethsoft VID demuxer and video decoder
- CRYO APC demuxer
- ATRAC3 decoder
- Atrac3 decoder
- V.Flash PTX decoder
- RoQ muxer, RoQ audio encoder
- Renderware TXD demuxer and decoder
@ -1225,7 +838,7 @@ version 0.4.9-pre1:
- rate distorted optimal lambda->qp support
- AAC encoding with libfaac
- Sunplus JPEG codec (SP5X) support
- use Lagrange multiplier instead of QP for ratecontrol
- use Lagrange multipler instead of QP for ratecontrol
- Theora/VP3 decoding support
- XA and ADX ADPCM codecs
- export MPEG-2 active display area / pan scan

15
INSTALL Normal file
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@ -0,0 +1,15 @@
1) Type './configure' to create the configuration. A list of configure
options is printed by running 'configure --help'.
'configure' can be launched from a directory different from the FFmpeg
sources to build the objects out of tree. To do this, use an absolute
path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
3) Type 'make install' to install all binaries and libraries you built.
NOTICE
- Non system dependencies (e.g. libx264, libvpx) are disabled by default.

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@ -1,17 +0,0 @@
#Installing FFmpeg:
1. Type `./configure` to create the configuration. A list of configure
options is printed by running `configure --help`.
`configure` can be launched from a directory different from the FFmpeg
sources to build the objects out of tree. To do this, use an absolute
path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
2. Then type `make` to build FFmpeg. GNU Make 3.81 or later is required.
3. Type `make install` to install all binaries and libraries you built.
NOTICE
------
- Non system dependencies (e.g. libx264, libvpx) are disabled by default.

102
LICENSE Normal file
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FFmpeg:
Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
FFmpeg.
Some optional parts of FFmpeg are licensed under the GNU General Public License
version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
these parts are used by default, you have to explicitly pass --enable-gpl to
configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are
- libpostproc
- libmpcodecs
- optional x86 optimizations in the files
libavcodec/x86/idct_mmx.c
- libutvideo encoding/decoding wrappers in
libavcodec/libutvideo*.cpp
- the X11 grabber in libavdevice/x11grab.c
- the swresample test app in
libswresample/swresample-test.c
- the texi2pod.pl tool
- the following filters in libavfilter:
- f_ebur128.c
- vf_blackframe.c
- vf_boxblur.c
- vf_colormatrix.c
- vf_cropdetect.c
- vf_decimate.c
- vf_delogo.c
- vf_geq.c
- vf_histeq.c
- vf_hqdn3d.c
- vf_hue.c
- vf_kerndeint.c
- vf_mcdeint.c
- vf_mp.c
- vf_noise.c
- vf_owdenoise.c
- vf_pp.c
- vf_sab.c
- vf_smartblur.c
- vf_spp.c
- vf_stereo3d.c
- vf_super2xsai.c
- vf_tinterlace.c
- vf_yadif.c
- vsrc_mptestsrc.c
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
the configure parameter --enable-version3 will activate this licensing option
for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
COPYING.GPLv3 to learn the exact legal terms that apply in this case.
There are a handful of files under other licensing terms, namely:
* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
licensing details. Specifically note that you must credit the IJG in the
documentation accompanying your program if you only distribute executables.
You must also indicate any changes including additions and deletions to
those three files in the documentation.
external libraries
==================
FFmpeg can be combined with a number of external libraries, which sometimes
affect the licensing of binaries resulting from the combination.
compatible libraries
--------------------
The following libraries are under GPL:
- frei0r
- libcdio
- libutvideo
- libvidstab
- libx264
- libxavs
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing --enable-gpl to configure.
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.
incompatible libraries
----------------------
The Fraunhofer AAC library, FAAC and aacplus are under licenses which
are incompatible with the GPLv2 and v3. We do not know for certain if their
licenses are compatible with the LGPL.
If you wish to enable these libraries, pass --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 unredistributeable.

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@ -1,124 +0,0 @@
# License
Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
or later (LGPL v2.1+). Read the file `COPYING.LGPLv2.1` for details. Some other
files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
FFmpeg.
Some optional parts of FFmpeg are licensed under the GNU General Public License
version 2 or later (GPL v2+). See the file `COPYING.GPLv2` for details. None of
these parts are used by default, you have to explicitly pass `--enable-gpl` to
configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are:
- libpostproc
- optional x86 optimization in the files
- `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/swresample-test.c`
- `tests/checkasm/*`
- `tests/tiny_ssim.c`
- the following filters in libavfilter:
- `f_ebur128.c`
- `vf_blackframe.c`
- `vf_boxblur.c`
- `vf_colormatrix.c`
- `vf_cover_rect.c`
- `vf_cropdetect.c`
- `vf_delogo.c`
- `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_mcdeint.c`
- `vf_mpdecimate.c`
- `vf_owdenoise.c`
- `vf_perspective.c`
- `vf_phase.c`
- `vf_pp.c`
- `vf_pp7.c`
- `vf_pullup.c`
- `vf_repeatfields.c`
- `vf_sab.c`
- `vf_smartblur.c`
- `vf_spp.c`
- `vf_stereo3d.c`
- `vf_super2xsai.c`
- `vf_tinterlace.c`
- `vf_uspp.c`
- `vsrc_mptestsrc.c`
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
the configure parameter `--enable-version3` will activate this licensing option
for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
`COPYING.GPLv3` to learn the exact legal terms that apply in this case.
There are a handful of files under other licensing terms, namely:
* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
`libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
licensing details. Specifically note that you must credit the IJG in the
documentation accompanying your program if you only distribute executables.
You must also indicate any changes including additions and deletions to
those three files in the documentation.
* `tests/reference.pnm` is under the expat license.
## External libraries
FFmpeg can be combined with a number of external libraries, which sometimes
affect the licensing of binaries resulting from the combination.
### Compatible libraries
The following libraries are under GPL:
- frei0r
- libcdio
- librubberband
- libvidstab
- libx264
- libx265
- libxavs
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing `--enable-gpl` to configure.
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
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.
### 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. 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

@ -14,6 +14,7 @@ patches and related discussions.
Project Leader
==============
Michael Niedermayer
final design decisions
@ -30,7 +31,7 @@ ffprobe:
ffprobe.c Stefano Sabatini
ffserver:
ffserver.c Reynaldo H. Verdejo Pinochet
ffserver.c, ffserver.h Baptiste Coudurier
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
@ -42,25 +43,16 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
documentation Mike Melanson
website Robert Swain, Lou Logan
build system (configure,Makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
Communication
=============
website Deby Barbara Lepage
fate.ffmpeg.org Timothy Gu
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
libavutil
=========
@ -70,24 +62,20 @@ Internal Interfaces:
libavutil/common.h Michael Niedermayer
Other:
aes_ctr.c, aes_ctr.h Eran Kornblau
bprint Nicolas George
bswap.h
des Reimar Doeffinger
dynarray.h Nicolas George
eval.c, eval.h Michael Niedermayer
float_dsp Loren Merritt
hash Reimar Doeffinger
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
mathematics.c, mathematics.h Michael Niedermayer
mem.c, mem.h Michael Niedermayer
opencl.c, opencl.h Wei Gao
opt.c, opt.h Michael Niedermayer
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
timecode Clément Bœsch
libavcodec
@ -98,6 +86,10 @@ Generic Parts:
avcodec.h Michael Niedermayer
utility code:
utils.c Michael Niedermayer
mem.c Michael Niedermayer
opt.c, opt.h Michael Niedermayer
arithmetic expression evaluator:
eval.c Michael Niedermayer
audio and video frame extraction:
parser.c Michael Niedermayer
bitstream reading:
@ -115,6 +107,8 @@ Generic Parts:
faandct.c, faandct.h Michael Niedermayer
Golomb coding:
golomb.c, golomb.h Michael Niedermayer
LPC:
lpc.c, lpc.h Justin Ruggles
motion estimation:
motion* Michael Niedermayer
rate control:
@ -126,44 +120,43 @@ Generic Parts:
libpostproc/* Michael Niedermayer
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
fixed point FFT:
fft* Zeljko Lukac
Text Subtitles Clément Bœsch
Codecs:
4xm.c Michael Niedermayer
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aacenc*, aaccoder.c Rostislav Pehlivanov
aasc.c Kostya Shishkov
ac3* Justin Ruggles
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann
apedec.c Kostya Shishkov
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3plus* Maxim Poliakovski
atrac3* Benjamin Larsson
bgmc.c, bgmc.h Thilo Borgmann
bink.c Kostya Shishkov
binkaudio.c Peter Ross
bmp.c Mans Rullgard, Kostya Shishkov
cavs* Stefan Gehrer
cdxl.c Paul B Mahol
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cinepakenc.c Rl / Aetey G.T. AB
ccaption_dec.c Anshul Maheshwari, Aman Gupta
cljr Alex Beregszaszi
cllc.c Derek Buitenhuis
cook.c, cookdata.h Benjamin Larsson
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
cuvid.c Timo Rothenpieler
dirac* Rostislav Pehlivanov
dca.c Kostya Shishkov, Benjamin Larsson
dnxhd* Baptiste Coudurier
dpcm.c Mike Melanson
dss_sp.c Oleksij Rempel
dv.c Roman Shaposhnik
dvbsubdec.c Anshul Maheshwari
dxa.c Kostya Shishkov
eacmv*, eaidct*, eat* Peter Ross
evrc* Paul B Mahol
exif.c, exif.h Thilo Borgmann
ffv1* Michael Niedermayer
ffv1.c Michael Niedermayer
ffwavesynth.c Nicolas George
flac* Justin Ruggles
flashsv* Benjamin Larsson
flicvideo.c Mike Melanson
g722.c Martin Storsjo
g726.c Roman Shaposhnik
@ -171,47 +164,55 @@ Codecs:
h261* Michael Niedermayer
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
hap* Tom Butterworth
huffyuv* Michael Niedermayer, Christophe Gisquet
huffyuv.c Michael Niedermayer
idcinvideo.c Mike Melanson
imc* Benjamin Larsson
indeo2* Kostya Shishkov
indeo5* Kostya Shishkov
interplayvideo.c Mike Melanson
jni*, ffjni* Matthieu Bouron
ivi* Kostya Shishkov
jacosub* Clément Bœsch
jpeg2000* Nicolas Bertrand
jpeg_ls.c Kostya Shishkov
jvdec.c Peter Ross
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libdirac* David Conrad
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libschroedinger* David Conrad
libspeexdec.c Justin Ruggles
libtheoraenc.c David Conrad
libutvideo* Derek Buitenhuis
libvorbis.c David Conrad
libvpx* James Zern
libx264.c Mans Rullgard, Jason Garrett-Glaser
libxavs.c Stefan Gehrer
libzvbi-teletextdec.c Marton Balint
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
mimic.c Ramiro Polla
mjpeg*.c Michael Niedermayer
mlp* Ramiro Polla
mmvideo.c Peter Ross
mpc* Kostya Shishkov
mpeg12.c, mpeg12data.h Michael Niedermayer
mpegvideo.c, mpegvideo.h Michael Niedermayer
mqc* Nicolas Bertrand
msmpeg4.c, msmpeg4data.h Michael Niedermayer
msrle.c Mike Melanson
msvideo1.c Mike Melanson
nellymoserdec.c Benjamin Larsson
nuv.c Reimar Doeffinger
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* Ivan Uskov
qdm2.c, qdm2data.h Roberto Togni, Benjamin Larsson
qdrw.c Kostya Shishkov
qpeg.c Kostya Shishkov
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
@ -219,51 +220,63 @@ Codecs:
rpza.c Roberto Togni
rtjpeg.c, rtjpeg.h Reimar Doeffinger
rv10.c Michael Niedermayer
rv4* Christophe Gisquet
rv3* Kostya Shishkov
rv4* Kostya Shishkov
s3tc* Ivo van Poorten
smacker.c Kostya Shishkov
smc.c Mike Melanson
smvjpegdec.c Ash Hughes
snow* Michael Niedermayer, Loren Merritt
snow.c Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
tak* Paul B Mahol
targa.c Kostya Shishkov
tiff.c Kostya Shishkov
truemotion1* Mike Melanson
truemotion2* Kostya Shishkov
truespeech.c Kostya Shishkov
tscc.c Kostya Shishkov
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
vc1* Christophe Gisquet
vc2* Rostislav Pehlivanov
ulti* Kostya Shishkov
v410*.c Derek Buitenhuis
vb.c Kostya Shishkov
vble.c Derek Buitenhuis
vc1* Kostya Shishkov
vcr1.c Michael Niedermayer
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
vmnc.c Kostya Shishkov
vorbis_dec.c Denes Balatoni, David Conrad
vorbis_enc.c Oded Shimon
vp3* Mike Melanson
vp5 Aurelien Jacobs
vp6 Aurelien Jacobs
vp8 David Conrad, Ronald Bultje
vp9 Ronald Bultje
vp8 David Conrad, Jason Garrett-Glaser, Ronald Bultje
vqavideo.c Mike Melanson
wavpack.c Kostya Shishkov
wmaprodec.c Sascha Sommer
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
wnv1.c Kostya Shishkov
xan.c Mike Melanson
xbm* Paul B Mahol
xface Stefano Sabatini
xl.c Kostya Shishkov
xvmc.c Ivan Kalvachev
xwd* Paul B Mahol
zerocodec.c Derek Buitenhuis
zmbv* Kostya Shishkov
Hardware acceleration:
crystalhd.c Philip Langdale
dxva2* Hendrik Leppkes, Laurent Aimar
mediacodec* Matthieu Bouron
dxva2* Laurent Aimar
libstagefright.cpp Mohamed Naufal
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern
vda* Sebastien Zwickert
vdpau* Carl Eugen Hoyos
libavdevice
@ -272,21 +285,11 @@ libavdevice
libavdevice/avdevice.h
avfoundation.m Thilo Borgmann
decklink* Deti Fliegl
dshow.c Roger Pack (CC rogerdpack@gmail.com)
fbdev_enc.c Lukasz Marek
gdigrab.c Roger Pack (CC rogerdpack@gmail.com)
dshow.c Roger Pack
iec61883.c Georg Lippitsch
lavfi Stefano Sabatini
libdc1394.c Roman Shaposhnik
opengl_enc.c Lukasz Marek
pulse_audio_enc.c Lukasz Marek
qtkit.m Thilo Borgmann
sdl Stefano Sabatini
v4l2.c Giorgio Vazzana
v4l2.c Luca Abeni
vfwcap.c Ramiro Polla
xv.c Lukasz Marek
libavfilter
===========
@ -295,57 +298,15 @@ Generic parts:
graphdump.c Nicolas George
Filters:
f_drawgraph.c Paul B Mahol
af_adelay.c Paul B Mahol
af_aecho.c Paul B Mahol
af_afade.c Paul B Mahol
af_amerge.c Nicolas George
af_aphaser.c Paul B Mahol
af_aresample.c Michael Niedermayer
af_astats.c Paul B Mahol
af_astreamsync.c Nicolas George
af_atempo.c Pavel Koshevoy
af_biquads.c Paul B Mahol
af_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
af_sidechaincompress.c Paul B Mahol
af_silenceremove.c Paul B Mahol
avf_aphasemeter.c Paul B Mahol
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_chromakey.c Timo Rothenpieler
vf_colorchannelmixer.c Paul B Mahol
vf_colorbalance.c Paul B Mahol
vf_colorkey.c Timo Rothenpieler
vf_colorlevels.c Paul B Mahol
vf_coreimage.m Thilo Borgmann
vf_deband.c Paul B Mahol
vf_dejudder.c Nicholas Robbins
vf_delogo.c Jean Delvare (CC <jdelvare@suse.com>)
vf_delogo.c Jean Delvare (CC <khali@linux-fr.org>)
vf_drawbox.c/drawgrid Andrey Utkin
vf_extractplanes.c Paul B Mahol
vf_histogram.c Paul B Mahol
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_lenscorrection.c Daniel Oberhoff
vf_mergeplanes.c Paul B Mahol
vf_neighbor.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_random.c Paul B Mahol
vf_readvitc.c Tobias Rapp (CC t.rapp at noa-archive dot com)
vf_scale.c Michael Niedermayer
vf_separatefields.c Paul B Mahol
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c Michael Niedermayer
@ -358,17 +319,15 @@ Generic parts:
libavformat/avformat.h Michael Niedermayer
Utility Code:
libavformat/utils.c Michael Niedermayer
Text Subtitles Clément Bœsch
Muxers/Demuxers:
4xm.c Mike Melanson
aadec.c Vesselin Bontchev (vesselin.bontchev at yandex dot com)
adtsenc.c Robert Swain
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
apngdec.c Benoit Fouet
ape.c Kostya Shishkov
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
@ -381,18 +340,17 @@ Muxers/Demuxers:
cdxl.c Paul B Mahol
crc.c Michael Niedermayer
daud.c Reimar Doeffinger
dss.c Oleksij Rempel
dtshddec.c Paul B Mahol
dv.c Roman Shaposhnik
dxa.c Kostya Shishkov
electronicarts.c Peter Ross
epafdec.c Paul B Mahol
ffm* Baptiste Coudurier
flac* Justin Ruggles
flic.c Mike Melanson
flvdec.c, flvenc.c Michael Niedermayer
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hls.c Anssi Hannula
hls encryption (hlsenc.c) Christian Suloway
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
@ -400,40 +358,37 @@ Muxers/Demuxers:
ipmovie.c Mike Melanson
ircam* Paul B Mahol
iss.c Stefan Gehrer
jacosub* Clément Bœsch
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
libnut.c Oded Shimon
libopenmpt.c Josh de Kock
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
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
microdvd* Aurelien Jacobs
mm.c Peter Ross
mov.c Baptiste Coudurier
mov.c Michael Niedermayer, Baptiste Coudurier
movenc.c Baptiste Coudurier, Matthieu Bouron
movenccenc.c Eran Kornblau
mpc.c Kostya Shishkov
mpeg.c Michael Niedermayer
mpegenc.c Michael Niedermayer
mpegts.c Marton Balint
mpegtsenc.c Baptiste Coudurier
mpegts* Baptiste Coudurier
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier
mxfdec.c Tomas Härdin
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
nut.c Michael Niedermayer
nuv.c Reimar Doeffinger
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
@ -443,23 +398,18 @@ Muxers/Demuxers:
raw.c Michael Niedermayer
rdt.c Ronald S. Bultje
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje
rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov
rtmp* Kostya Shishkov
rtp.c, rtpenc.c Martin Storsjo
rtpdec_ac3.* Gilles Chanteperdrix
rtpdec_dv.* Thomas Volkert
rtpdec_h261.*, rtpenc_h261.* Thomas Volkert
rtpdec_hevc.*, rtpenc_hevc.* Thomas Volkert
rtpdec_mpa_robust.* Gilles Chanteperdrix
rtpdec_asf.* Ronald S. Bultje
rtpdec_vc2hq.*, rtpenc_vc2hq.* Thomas Volkert
rtpdec_vp9.c Thomas Volkert
rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo
rtsp.c Luca Barbato
sbgdec.c Nicolas George
sdp.c Martin Storsjo
segafilm.c Mike Melanson
segment.c Stefano Sabatini
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
smjpeg* Paul B Mahol
spdif* Anssi Hannula
srtdec.c Aurelien Jacobs
swf.c Baptiste Coudurier
takdec.c Paul B Mahol
@ -468,21 +418,18 @@ Muxers/Demuxers:
voc.c Aurelien Jacobs
wav.c Michael Niedermayer
wc3movie.c Mike Melanson
webm dash (matroskaenc.c) Vignesh Venkatasubramanian
webvtt* Matthew J Heaney
westwood.c Mike Melanson
wtv.c Peter Ross
wv.c Kostya Shishkov
wvenc.c Paul B Mahol
Protocols:
async.c Zhang Rui
bluray.c Petri Hintukainen
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libssh.c Lukasz Marek
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c Marvin Scholz
libswresample
@ -502,26 +449,25 @@ Resamplers:
Operating systems / CPU architectures
=====================================
Alpha Falk Hueffner
MIPS Nedeljko Babic
Alpha Mans Rullgard, Falk Hueffner
ARM Mans Rullgard
AVR32 Mans Rullgard
MIPS Mans Rullgard
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Luca Barbato
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver, Hendrik Leppkes
Windows ICL Matthew Oliver
ADI/Blackfin DSP Marc Hoffman
Sparc Roman Shaposhnik
OS/2 KO Myung-Hun
x86 Michael Niedermayer
Releases
========
2.8 Michael Niedermayer
2.7 Michael Niedermayer
2.6 Michael Niedermayer
2.5 Michael Niedermayer
2.0 Michael Niedermayer
1.2 Michael Niedermayer
If you want to maintain an older release, please contact us
@ -529,36 +475,34 @@ If you want to maintain an older release, please contact us
GnuPG Fingerprints of maintainers and contributors
==================================================
Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
Anton Khirnov 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
Ash Hughes 694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
Attila Kinali 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
Baptiste Coudurier 8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
Ben Littler 3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Clément Bœsch 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
Bœsch Clément 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
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
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Justin Ruggles 3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Luca Barbato 6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
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
Reimar Doeffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
Reimar Döffinger 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
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
Tiancheng "Timothy" Gu 9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
Tim Nicholson 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

119
Makefile
View File

@ -4,80 +4,62 @@ include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
vpath %.h $(SRC_PATH)
vpath %.inc $(SRC_PATH)
vpath %.m $(SRC_PATH)
vpath %.S $(SRC_PATH)
vpath %.asm $(SRC_PATH)
vpath %.rc $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %/fate_config.sh.template $(SRC_PATH)
AVPROGS-$(CONFIG_FFMPEG) += ffmpeg
AVPROGS-$(CONFIG_FFPLAY) += ffplay
AVPROGS-$(CONFIG_FFPROBE) += ffprobe
AVPROGS-$(CONFIG_FFSERVER) += ffserver
PROGS-$(CONFIG_FFMPEG) += ffmpeg
PROGS-$(CONFIG_FFPLAY) += ffplay
PROGS-$(CONFIG_FFPROBE) += ffprobe
PROGS-$(CONFIG_FFSERVER) += ffserver
AVPROGS := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
INSTPROGS = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
PROGS += $(AVPROGS)
PROGS := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
INSTPROGS = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
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
OBJS = cmdutils.o $(EXEOBJS)
OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
TOOLS = qt-faststart trasher uncoded_frame
TOOLS = qt-faststart trasher
TOOLS-$(CONFIG_ZLIB) += cws2fws
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
BASENAMES = ffmpeg ffplay ffprobe ffserver
ALLPROGS = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLPROGS_G = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
FFLIBS-$(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 = cmdutils_common_opts.h \
compat/w32pthreads.h
SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h
include $(SRC_PATH)/common.mak
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
all: $(AVPROGS)
all: $(PROGS)
$(PROGS): %$(EXESUF): %_g$(EXESUF)
$(CP) $< $@
$(STRIP) $@
$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
config.h: .config
.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@ -87,10 +69,11 @@ config.h: .config
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 MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS \
MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
ARMV5TE-OBJS ARMV6-OBJS VFP-OBJS NEON-OBJS \
ALTIVEC-OBJS VIS-OBJS \
MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS \
OBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=
@ -102,16 +85,13 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
include $(SRC_PATH)/library.mak
endef
$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
include $(SRC_PATH)/doc/Makefile
define DOPROG
OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
$$(OBJS-$(1)): CFLAGS += $(CFLAGS-$(1))
$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
@ -119,16 +99,10 @@ $(1)$(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)
$(CP) $< $@
$(STRIP) $@
$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
OBJDIRS += tools
@ -140,14 +114,14 @@ GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
.version: M=@
libavutil/ffversion.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
version.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
$(Q)touch .version
# force version.sh to run whenever version might have changed
-include .version
ifdef AVPROGS
ifdef PROGS
install: install-progs install-data
endif
@ -158,7 +132,7 @@ install-libs: install-libs-yes
install-progs-yes:
install-progs-$(CONFIG_SHARED): install-libs
install-progs: install-progs-yes $(AVPROGS)
install-progs: install-progs-yes $(PROGS)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
@ -170,31 +144,28 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
$(RM) $(ALLPROGS) $(ALLPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
$(RM) -rf coverage.info lcov
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
$(RM) -rf doc/examples/pc-uninstalled
$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
check: all alltools examples testprogs fate
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/tests/Makefile
$(sort $(OBJDIRS)):

18
README Normal file
View File

@ -0,0 +1,18 @@
FFmpeg README
-------------
1) Documentation
----------------
* Read the documentation in the doc/ directory in git.
You can also view it online at http://ffmpeg.org/documentation.html
2) Licensing
------------
* See the LICENSE file.
3) Build and Install
--------------------
* See the INSTALL file.

View File

@ -1,49 +0,0 @@
FFmpeg README
=============
FFmpeg is a collection of libraries and tools to process multimedia content
such as audio, video, subtitles and related metadata.
## Libraries
* `libavcodec` provides implementation of a wider range of codecs.
* `libavformat` implements streaming protocols, container formats and basic I/O access.
* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
* `libavdevice` provides an abstraction to access capture and playback devices.
* `libswresample` implements audio mixing and resampling routines.
* `libswscale` implements color conversion and scaling routines.
## Tools
* [ffmpeg](https://ffmpeg.org/ffmpeg.html) is a command line toolbox to
manipulate, convert and stream multimedia content.
* [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
The offline documentation is available in the **doc/** directory.
The online documentation is available in the main [website](https://ffmpeg.org)
and in the [wiki](https://trac.ffmpeg.org).
### Examples
Coding examples are available in the **doc/examples** directory.
## License
FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
GPL. Please refer to the LICENSE file for detailed information.
## Contributing
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. Few developers
follow pull requests so they will likely be ignored.

View File

@ -1 +1 @@
3.0.git
2.0.5

1
VERSION Normal file
View File

@ -0,0 +1 @@
2.0.5

View File

@ -1,17 +1,16 @@
OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
OBJS-$(HAVE_ARMV6) += $(ARMV6-OBJS) $(ARMV6-OBJS-yes)
OBJS-$(HAVE_ARMV8) += $(ARMV8-OBJS) $(ARMV8-OBJS-yes)
OBJS-$(HAVE_VFP) += $(VFP-OBJS) $(VFP-OBJS-yes)
OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes)
OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes)
OBJS-$(HAVE_MIPSDSP) += $(MIPSDSP-OBJS) $(MIPSDSP-OBJS-yes)
OBJS-$(HAVE_MIPS32R2) += $(MIPS32R2-OBJS) $(MIPS32R2-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS) $(MIPSDSPR1-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS) $(MIPSDSPR2-OBJS-yes)
OBJS-$(HAVE_MSA) += $(MSA-OBJS) $(MSA-OBJS-yes)
OBJS-$(HAVE_MMI) += $(MMI-OBJS) $(MMI-OBJS-yes)
OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
OBJS-$(HAVE_VSX) += $(VSX-OBJS) $(VSX-OBJS-yes)
OBJS-$(HAVE_VIS) += $(VIS-OBJS) $(VIS-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_YASM) += $(YASM-OBJS) $(YASM-OBJS-yes)

File diff suppressed because it is too large Load Diff

View File

@ -19,18 +19,17 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef CMDUTILS_H
#define CMDUTILS_H
#ifndef FFMPEG_CMDUTILS_H
#define FFMPEG_CMDUTILS_H
#include <stdint.h>
#include "config.h"
#include "libavcodec/avcodec.h"
#include "libavfilter/avfilter.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#ifdef _WIN32
#ifdef __MINGW32__
#undef main /* We don't want SDL to override our main() */
#endif
@ -44,12 +43,16 @@ extern const char program_name[];
*/
extern const int program_birth_year;
/**
* this year, defined by the program for show_banner()
*/
extern const int this_year;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern AVDictionary *sws_dict;
extern struct SwsContext *sws_opts;
extern AVDictionary *swr_opts;
extern AVDictionary *format_opts, *codec_opts, *resample_opts;
extern int hide_banner;
/**
* Register a program-specific cleanup routine.
@ -59,7 +62,7 @@ void register_exit(void (*cb)(int ret));
/**
* Wraps exit with a program-specific cleanup routine.
*/
void exit_program(int ret) av_noreturn;
void exit_program(int ret);
/**
* Initialize the cmdutils option system, in particular
@ -78,11 +81,6 @@ void uninit_opts(void);
*/
void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
/**
* Override the cpuflags.
*/
int opt_cpuflags(void *optctx, const char *opt, const char *arg);
/**
* Fallback for options that are not explicitly handled, these will be
* parsed through AVOptions.
@ -98,14 +96,12 @@ int opt_report(const char *opt);
int opt_max_alloc(void *optctx, const char *opt, const char *arg);
int opt_cpuflags(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.
*/
@ -277,7 +273,7 @@ typedef struct OptionGroup {
AVDictionary *codec_opts;
AVDictionary *format_opts;
AVDictionary *resample_opts;
AVDictionary *sws_dict;
struct SwsContext *sws_opts;
AVDictionary *swr_opts;
} OptionGroup;
@ -415,13 +411,6 @@ void show_banner(int argc, char **argv, const OptionDef *options);
*/
int show_version(void *optctx, const char *opt, const char *arg);
/**
* Print the build configuration of the program to stdout. The contents
* depend on the definition of FFMPEG_CONFIGURATION.
* This option processing function does not utilize the arguments.
*/
int show_buildconf(void *optctx, const char *opt, const char *arg);
/**
* Print the license of the program to stdout. The license depends on
* the license of the libraries compiled into the program.
@ -431,31 +420,10 @@ int show_license(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the formats supported by the
* program (including devices).
* This option processing function does not utilize the arguments.
*/
int show_formats(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the devices supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int show_devices(void *optctx, const char *opt, const char *arg);
#if CONFIG_AVDEVICE
/**
* 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 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);
#endif
int show_formats(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the codecs supported by the
@ -517,18 +485,24 @@ int show_layouts(void *optctx, const char *opt, const char *arg);
*/
int show_sample_fmts(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the color names and values recognized
* by the program.
*/
int show_colors(void *optctx, const char *opt, const char *arg);
/**
* Return a positive value if a line read from standard input
* starts with [yY], otherwise return 0.
*/
int read_yesno(void);
/**
* Read the file with name filename, and put its content in a newly
* allocated 0-terminated buffer.
*
* @param filename file to read from
* @param bufptr location where pointer to buffer is returned
* @param size location where size of buffer is returned
* @return 0 in case of success, a negative value corresponding to an
* AVERROR error code in case of failure.
*/
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
/**
* Get a file corresponding to a preset file.
*
@ -585,6 +559,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
char name[128];\
av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
double get_rotation(AVStream *st);
#endif /* CMDUTILS_H */

View File

@ -4,9 +4,7 @@
{ "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" },
@ -16,20 +14,11 @@
{ "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" },
{ "cpuflags" , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },
#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

View File

@ -1,278 +0,0 @@
/*
* 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

@ -5,20 +5,12 @@
# first so "all" becomes default target
all: all-yes
DEFAULT_YASMD=.dbg
ifeq ($(DBG),1)
YASMD=$(DEFAULT_YASMD)
else
YASMD=
endif
ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
BRIEF = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
MSG = $@
@ -32,14 +24,12 @@ endif
ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_LINK)/
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CPPFLAGS) $(CFLAGS)
OBJCFLAGS += $(EOBJCFLAGS)
OBJCCFLAGS = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS := $(CPPFLAGS) $(CFLAGS) $(CXXFLAGS)
CXXFLAGS += $(CPPFLAGS) $(CFLAGS)
YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
@ -47,14 +37,12 @@ LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
define COMPILE
$(call $(1)DEP,$(1))
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<)
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
endef
COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
COMPILE_M = $(call COMPILE,OBJCC)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.o: %.c
$(COMPILE_C)
@ -62,28 +50,22 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.o: %.cpp
$(COMPILE_CXX)
%.o: %.m
$(COMPILE_M)
%.s: %.c
$(CC) $(CCFLAGS) -S -o $@ $<
$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
%.o: %.S
$(COMPILE_S)
%_host.o: %.c
$(COMPILE_HOSTC)
%.o: %.rc
$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
%.i: %.c
$(CC) $(CCFLAGS) $(CC_E) $<
%.h.c:
$(Q)echo '#include "$*.h"' >$@
%.c %.h %.ver: TAG = GEN
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
%.c %.h: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@ -100,17 +82,16 @@ endif
include $(SRC_PATH)/arch.mak
OBJS += $(OBJS-yes)
SLIBOBJS += $(SLIBOBJS-yes)
FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
LDLIBS = $(FFLIBS:%=%$(BUILDSUF))
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
EXAMPLES := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
TOOLS += $(TOOLS-yes)
@ -118,9 +99,7 @@ TOOLOBJS := $(TOOLS:%=tools/%.o)
TOOLS := $(TOOLS:%=tools/%$(EXESUF))
HEADERS += $(HEADERS-yes)
PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
SRC_DIR := $(SRC_PATH)/lib$(NAME)
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
@ -133,29 +112,30 @@ checkheaders: $(HOBJS)
alltools: $(TOOLS)
$(HOSTOBJS): %.o: %.c
$(COMPILE_HOSTC)
$(call COMPILE,HOSTCC)
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.o *~ *.h.c *.gcda *.gcno *.map *.ver *.ho *$(DEFAULT_YASMD).asm
CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
define RULES
clean::
$(RM) $(HOSTPROGS) $(TESTPROGS) $(TOOLS)
$(RM) $(OBJS) $(OBJS:.o=.d)
$(RM) $(HOSTPROGS)
$(RM) $(TOOLS)
endef
$(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))

View File

@ -1,22 +1,5 @@
/*
* Work around the class() function in AIX math.h clashing with
* identifiers named "class".
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
* Workaround aix-specific class() function clashing with ffmpeg class usage
*/
#ifndef COMPAT_AIX_MATH_H

View File

@ -13,8 +13,7 @@
//
// 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
// 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
@ -38,9 +37,40 @@
#ifndef __AVISYNTH_C__
#define __AVISYNTH_C__
#include "avs/config.h"
#include "avs/capi.h"
#include "avs/types.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
typedef unsigned char BYTE;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
@ -48,8 +78,8 @@
// Constants
//
#ifndef __AVISYNTH_6_H__
enum { AVISYNTH_INTERFACE_VERSION = 6 };
#ifndef __AVISYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 4 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
@ -81,8 +111,8 @@ enum {AVS_CS_BGR = 1<<28,
AVS_CS_PLANAR = 1<<31,
AVS_CS_SHIFT_SUB_WIDTH = 0,
AVS_CS_SHIFT_SUB_HEIGHT = 8,
AVS_CS_SHIFT_SAMPLE_BITS = 16,
AVS_CS_SHIFT_SUB_HEIGHT = 1 << 3,
AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
@ -149,66 +179,15 @@ enum { //SUBTYPES
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)
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
};
#ifdef BUILDING_AVSCORE
struct AVS_ScriptEnvironment {
IScriptEnvironment * env;
const char * error;
AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
: env(e), error(0) {}
};
#endif
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
@ -258,23 +237,29 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->image_type & property)==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_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
@ -288,18 +273,25 @@ AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
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_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_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
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_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_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; }
@ -357,13 +349,11 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
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
/////////////////////////////////////////////////////////////////////
//
@ -400,38 +390,89 @@ typedef struct AVS_VideoFrame {
} 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
return p->pitch;}
AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
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_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_sizeUV;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_sizeUV;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->heightUV;
return 0;
}
return p->height;}
#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
return p->vfb->data + p->offset;}
AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
#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
@ -616,16 +657,12 @@ enum {
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
AVS_CPUF_SSE4_2 = 0x800, // Nehalem
};
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(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);
@ -662,12 +699,12 @@ AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
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);}
{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,FRAME_ALIGN);}
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
@ -735,6 +772,7 @@ struct AVS_Library {
AVSC_DECLARE_FUNC(avs_function_exists);
AVSC_DECLARE_FUNC(avs_get_audio);
AVSC_DECLARE_FUNC(avs_get_cpu_flags);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_get_frame);
AVSC_DECLARE_FUNC(avs_get_parity);
AVSC_DECLARE_FUNC(avs_get_var);
@ -759,27 +797,6 @@ struct AVS_Library {
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
@ -814,6 +831,7 @@ AVSC_INLINE AVS_Library * avs_load_library() {
AVSC_LOAD_FUNC(avs_function_exists);
AVSC_LOAD_FUNC(avs_get_audio);
AVSC_LOAD_FUNC(avs_get_cpu_flags);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_get_frame);
AVSC_LOAD_FUNC(avs_get_parity);
AVSC_LOAD_FUNC(avs_get_var);
@ -839,27 +857,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
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

View File

@ -0,0 +1,68 @@
// Copyright (c) 2011 FFmpegSource Project
//
// Permission is hereby granted, free of charge, to any person obtaining a copy
// of this software and associated documentation files (the "Software"), to deal
// in the Software without restriction, including without limitation the rights
// to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
// copies of the Software, and to permit persons to whom the Software is
// furnished to do so, subject to the following conditions:
//
// The above copyright notice and this permission notice shall be included in
// all copies or substantial portions of the Software.
//
// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
// FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
// AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
// LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
// OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
// THE SOFTWARE.
/* these are defines/functions that are used and were changed in the switch to 2.6
* and are needed to maintain full compatility with 2.5 */
enum {
AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar
AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar
};
AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
switch (plane)
{
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV)
return p->height>>1;
return 0;
}
return p->height;}
AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane)
{
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV)
return p->row_size>>1;
else
return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV)
{
r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_size>>1;
}
else
return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
{ return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }

View File

@ -1,62 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef 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

View File

@ -1,55 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef 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

View File

@ -1,51 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef 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

View File

@ -13,8 +13,7 @@
//
// 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
// 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
@ -513,21 +512,21 @@ AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
// 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; }
{ 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 = {0}; v.type = 'i'; v.d.integer = v0; return v; }
{ 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 = {0}; v.type = 's'; v.d.string = v0; return v; }
{ 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 = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
{ 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 = {0}; v.type = 'e'; v.d.string = v0; return v; }
{ 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 = {0}; avs_set_to_clip(&v, v0); return v; }
{ 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 = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
{ AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//

View File

@ -52,8 +52,8 @@ namespace avxsynth {
//
// Functions
//
#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
#define MAKEWORD(a,b) (((a) << 8) | (b))
#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

View File

@ -1,42 +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_DISPATCH_SEMAPHORE_SEMAPHORE_H
#define COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H
#include <dispatch/dispatch.h>
#include <errno.h>
#define sem_t dispatch_semaphore_t
#define sem_post(psem) dispatch_semaphore_signal(*psem)
#define sem_wait(psem) dispatch_semaphore_wait(*psem, DISPATCH_TIME_FOREVER)
#define sem_timedwait(psem, val) dispatch_semaphore_wait(*psem, dispatch_walltime(val, 0))
#define sem_destroy(psem) dispatch_release(*psem)
static inline int compat_sem_init(dispatch_semaphore_t *psem,
int unused, int val)
{
int ret = !!(*psem = dispatch_semaphore_create(val)) - 1;
if (ret < 0)
errno = ENOMEM;
return ret;
}
#define sem_init compat_sem_init
#endif /* COMPAT_DISPATCH_SEMAPHORE_SEMAPHORE_H */

View File

@ -1,35 +0,0 @@
/*
* Work around broken floating point limits on some systems.
*
* 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_next <float.h>
#ifdef FLT_MAX
#undef FLT_MAX
#define FLT_MAX 3.40282346638528859812e+38F
#undef FLT_MIN
#define FLT_MIN 1.17549435082228750797e-38F
#undef DBL_MAX
#define DBL_MAX ((double)1.79769313486231570815e+308L)
#undef DBL_MIN
#define DBL_MIN ((double)2.22507385850720138309e-308L)
#endif

View File

@ -1,22 +0,0 @@
/*
* Work around broken floating point limits on some systems.
*
* 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_next <limits.h>
#include <float.h>

View File

@ -38,6 +38,8 @@ static int optind = 1;
static int optopt;
static char *optarg;
#undef fprintf
static int getopt(int argc, char *argv[], char *opts)
{
static int sp = 1;
@ -54,7 +56,7 @@ static int getopt(int argc, char *argv[], char *opts)
}
}
optopt = c = argv[optind][sp];
if (c == ':' || !(cp = strchr(opts, c))) {
if (c == ':' || (cp = strchr(opts, c)) == NULL) {
fprintf(stderr, ": illegal option -- %c\n", c);
if (argv[optind][++sp] == '\0') {
optind++;

View File

@ -19,8 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_MSVCRT_SNPRINTF_H
#define COMPAT_MSVCRT_SNPRINTF_H
#ifndef COMPAT_SNPRINTF_H
#define COMPAT_SNPRINTF_H
#include <stdarg.h>
#include <stdio.h>
@ -35,4 +35,4 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap);
#define _snprintf avpriv_snprintf
#define vsnprintf avpriv_vsnprintf
#endif /* COMPAT_MSVCRT_SNPRINTF_H */
#endif /* COMPAT_SNPRINTF_H */

View File

@ -23,8 +23,8 @@
* os2threads to pthreads wrapper
*/
#ifndef COMPAT_OS2THREADS_H
#define COMPAT_OS2THREADS_H
#ifndef AVCODEC_OS2PTHREADS_H
#define AVCODEC_OS2PTHREADS_H
#define INCL_DOS
#include <os2.h>
@ -32,71 +32,55 @@
#undef __STRICT_ANSI__ /* for _beginthread() */
#include <stdlib.h>
#include <sys/builtin.h>
#include <sys/fmutex.h>
#include "libavutil/attributes.h"
typedef struct {
TID tid;
void *(*start_routine)(void *);
void *arg;
void *result;
} pthread_t;
typedef TID pthread_t;
typedef void pthread_attr_t;
typedef HMTX pthread_mutex_t;
typedef void pthread_mutexattr_t;
typedef struct {
HEV event_sem;
HEV ack_sem;
volatile unsigned wait_count;
HEV event_sem;
int wait_count;
} pthread_cond_t;
typedef void pthread_condattr_t;
typedef struct {
volatile int done;
_fmutex mtx;
} pthread_once_t;
#define PTHREAD_ONCE_INIT {0, _FMUTEX_INITIALIZER}
struct thread_arg {
void *(*start_routine)(void *);
void *arg;
};
static void thread_entry(void *arg)
{
pthread_t *thread = arg;
struct thread_arg *thread_arg = arg;
thread->result = thread->start_routine(thread->arg);
thread_arg->start_routine(thread_arg->arg);
av_free(thread_arg);
}
static av_always_inline int pthread_create(pthread_t *thread,
const pthread_attr_t *attr,
void *(*start_routine)(void*),
void *arg)
static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg)
{
thread->start_routine = start_routine;
thread->arg = arg;
thread->result = NULL;
struct thread_arg *thread_arg;
thread->tid = _beginthread(thread_entry, NULL, 1024 * 1024, thread);
thread_arg = av_mallocz(sizeof(struct thread_arg));
thread_arg->start_routine = start_routine;
thread_arg->arg = arg;
*thread = _beginthread(thread_entry, NULL, 256 * 1024, thread_arg);
return 0;
}
static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
{
DosWaitThread(&thread.tid, DCWW_WAIT);
if (value_ptr)
*value_ptr = thread.result;
DosWaitThread((PTID)&thread, DCWW_WAIT);
return 0;
}
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
const pthread_mutexattr_t *attr)
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr)
{
DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
@ -124,11 +108,9 @@ static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
return 0;
}
static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
const pthread_condattr_t *attr)
static av_always_inline int pthread_cond_init(pthread_cond_t *cond, const pthread_condattr_t *attr)
{
DosCreateEventSem(NULL, &cond->event_sem, DCE_POSTONE, FALSE);
DosCreateEventSem(NULL, &cond->ack_sem, DCE_POSTONE, FALSE);
cond->wait_count = 0;
@ -138,16 +120,16 @@ static av_always_inline int pthread_cond_init(pthread_cond_t *cond,
static av_always_inline int pthread_cond_destroy(pthread_cond_t *cond)
{
DosCloseEventSem(cond->event_sem);
DosCloseEventSem(cond->ack_sem);
return 0;
}
static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
{
if (!__atomic_cmpxchg32(&cond->wait_count, 0, 0)) {
if (cond->wait_count > 0) {
DosPostEventSem(cond->event_sem);
DosWaitEventSem(cond->ack_sem, SEM_INDEFINITE_WAIT);
cond->wait_count--;
}
return 0;
@ -155,47 +137,26 @@ static av_always_inline int pthread_cond_signal(pthread_cond_t *cond)
static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
{
while (!__atomic_cmpxchg32(&cond->wait_count, 0, 0))
pthread_cond_signal(cond);
while (cond->wait_count > 0) {
DosPostEventSem(cond->event_sem);
cond->wait_count--;
}
return 0;
}
static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
pthread_mutex_t *mutex)
static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
__atomic_increment(&cond->wait_count);
cond->wait_count++;
pthread_mutex_unlock(mutex);
DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT);
__atomic_decrement(&cond->wait_count);
DosPostEventSem(cond->ack_sem);
pthread_mutex_lock(mutex);
return 0;
}
static av_always_inline int pthread_once(pthread_once_t *once_control,
void (*init_routine)(void))
{
if (!once_control->done)
{
_fmutex_request(&once_control->mtx, 0);
if (!once_control->done)
{
init_routine();
once_control->done = 1;
}
_fmutex_release(&once_control->mtx);
}
return 0;
}
#endif /* COMPAT_OS2THREADS_H */
#endif /* AVCODEC_OS2PTHREADS_H */

View File

@ -1,352 +0,0 @@
#!/usr/bin/env perl
# make_sunver.pl
#
# Copyright (C) 2010, 2011, 2012, 2013
# Free Software Foundation, Inc.
#
# This file 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 3 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; see the file COPYING.GPLv3. If not see
# <http://www.gnu.org/licenses/>.
# This script takes at least two arguments, a GNU style version script and
# a list of object and archive files, and generates a corresponding Sun
# style version script as follows:
#
# Each glob pattern, C++ mangled pattern or literal in the input script is
# matched against all global symbols in the input objects, emitting those
# that matched (or nothing if no match was found).
# A comment with the original pattern and its type is left in the output
# file to make it easy to understand the matches.
#
# It uses elfdump when present (native), GNU readelf otherwise.
# It depends on the GNU version of c++filt, since it must understand the
# GNU mangling style.
use FileHandle;
use IPC::Open2;
# Enforce C locale.
$ENV{'LC_ALL'} = "C";
$ENV{'LANG'} = "C";
# Input version script, GNU style.
my $symvers = shift;
##########
# Get all the symbols from the library, match them, and add them to a hash.
my %sym_hash = ();
# List of objects and archives to process.
my @OBJECTS = ();
# List of shared objects to omit from processing.
my @SHAREDOBJS = ();
# Filter out those input archives that have corresponding shared objects to
# avoid adding all symbols matched in the archive to the output map.
foreach $file (@ARGV) {
if (($so = $file) =~ s/\.a$/.so/ && -e $so) {
printf STDERR "omitted $file -> $so\n";
push (@SHAREDOBJS, $so);
} else {
push (@OBJECTS, $file);
}
}
# We need to detect and ignore hidden symbols. Solaris nm can only detect
# this in the harder to parse default output format, and GNU nm not at all,
# so use elfdump -s in the native case and GNU readelf -s otherwise.
# GNU objdump -t cannot be used since it produces a variable number of
# columns.
# The path to elfdump.
my $elfdump = "/usr/ccs/bin/elfdump";
if (-f $elfdump) {
open ELFDUMP,$elfdump.' -s '.(join ' ',@OBJECTS).'|' or die $!;
my $skip_arsym = 0;
while (<ELFDUMP>) {
chomp;
# Ignore empty lines.
if (/^$/) {
# End of archive symbol table, stop skipping.
$skip_arsym = 0 if $skip_arsym;
next;
}
# Keep skipping until end of archive symbol table.
next if ($skip_arsym);
# Ignore object name header for individual objects and archives.
next if (/:$/);
# Ignore table header lines.
next if (/^Symbol Table Section:/);
next if (/index.*value.*size/);
# Start of archive symbol table: start skipping.
if (/^Symbol Table: \(archive/) {
$skip_arsym = 1;
next;
}
# Split table.
(undef, undef, undef, undef, $bind, $oth, undef, $shndx, $name) = split;
# Error out for unknown input.
die "unknown input line:\n$_" unless defined($bind);
# Ignore local symbols.
next if ($bind eq "LOCL");
# Ignore hidden symbols.
next if ($oth eq "H");
# Ignore undefined symbols.
next if ($shndx eq "UNDEF");
# Error out for unhandled cases.
if ($bind !~ /^(GLOB|WEAK)/ or $oth ne "D") {
die "unhandled symbol:\n$_";
}
# Remember symbol.
$sym_hash{$name}++;
}
close ELFDUMP or die "$elfdump error";
} else {
open READELF, 'readelf -s -W '.(join ' ',@OBJECTS).'|' or die $!;
# Process each symbol.
while (<READELF>) {
chomp;
# Ignore empty lines.
next if (/^$/);
# Ignore object name header.
next if (/^File: .*$/);
# Ignore table header lines.
next if (/^Symbol table.*contains.*:/);
next if (/Num:.*Value.*Size/);
# Split table.
(undef, undef, undef, undef, $bind, $vis, $ndx, $name) = split;
# Error out for unknown input.
die "unknown input line:\n$_" unless defined($bind);
# Ignore local symbols.
next if ($bind eq "LOCAL");
# Ignore hidden symbols.
next if ($vis eq "HIDDEN");
# Ignore undefined symbols.
next if ($ndx eq "UND");
# Error out for unhandled cases.
if ($bind !~ /^(GLOBAL|WEAK)/ or $vis ne "DEFAULT") {
die "unhandled symbol:\n$_";
}
# Remember symbol.
$sym_hash{$name}++;
}
close READELF or die "readelf error";
}
##########
# The various types of glob patterns.
#
# A glob pattern that is to be applied to the demangled name: 'cxx'.
# A glob patterns that applies directly to the name in the .o files: 'glob'.
# This pattern is ignored; used for local variables (usually just '*'): 'ign'.
# The type of the current pattern.
my $glob = 'glob';
# We're currently inside `extern "C++"', which Sun ld doesn't understand.
my $in_extern = 0;
# The c++filt command to use. This *must* be GNU c++filt; the Sun Studio
# c++filt doesn't handle the GNU mangling style.
my $cxxfilt = $ENV{'CXXFILT'} || "c++filt";
# The current version name.
my $current_version = "";
# Was there any attempt to match a symbol to this version?
my $matches_attempted;
# The number of versions which matched this symbol.
my $matched_symbols;
open F,$symvers or die $!;
# Print information about generating this file
print "# This file was generated by make_sunver.pl. DO NOT EDIT!\n";
print "# It was generated by:\n";
printf "# %s %s %s\n", $0, $symvers, (join ' ',@ARGV);
printf "# Omitted archives with corresponding shared libraries: %s\n",
(join ' ', @SHAREDOBJS) if $#SHAREDOBJS >= 0;
print "#\n\n";
print "\$mapfile_version 2\n";
while (<F>) {
# Lines of the form '};'
if (/^([ \t]*)(\}[ \t]*;[ \t]*)$/) {
$glob = 'glob';
if ($in_extern) {
$in_extern--;
print "$1##$2\n";
} else {
print;
}
next;
}
# Lines of the form '} SOME_VERSION_NAME_1.0;'
if (/^[ \t]*\}[ \tA-Z0-9_.a-z]+;[ \t]*$/) {
$glob = 'glob';
# We tried to match symbols agains this version, but none matched.
# Emit dummy hidden symbol to avoid marking this version WEAK.
if ($matches_attempted && $matched_symbols == 0) {
print " hidden:\n";
print " .force_WEAK_off_$current_version = DATA S0x0 V0x0;\n";
}
print; next;
}
# Comment and blank lines
if (/^[ \t]*\#/) { print; next; }
if (/^[ \t]*$/) { print; next; }
# Lines of the form '{'
if (/^([ \t]*){$/) {
if ($in_extern) {
print "$1##{\n";
} else {
print;
}
next;
}
# Lines of the form 'SOME_VERSION_NAME_1.1 {'
if (/^([A-Z0-9_.]+)[ \t]+{$/) {
# Record version name.
$current_version = $1;
# Reset match attempts, #matched symbols for this version.
$matches_attempted = 0;
$matched_symbols = 0;
print "SYMBOL_VERSION $1 {\n";
next;
}
# Ignore 'global:'
if (/^[ \t]*global:$/) { print; next; }
# After 'local:', globs should be ignored, they won't be exported.
if (/^[ \t]*local:$/) {
$glob = 'ign';
print;
next;
}
# After 'extern "C++"', globs are C++ patterns
if (/^([ \t]*)(extern \"C\+\+\"[ \t]*)$/) {
$in_extern++;
$glob = 'cxx';
# Need to comment, Sun ld cannot handle this.
print "$1##$2\n"; next;
}
# Chomp newline now we're done with passing through the input file.
chomp;
# Catch globs. Note that '{}' is not allowed in globs by this script,
# so only '*' and '[]' are available.
if (/^([ \t]*)([^ \t;{}#]+);?[ \t]*$/) {
my $ws = $1;
my $ptn = $2;
# Turn the glob into a regex by replacing '*' with '.*', '?' with '.'.
# Keep $ptn so we can still print the original form.
($pattern = $ptn) =~ s/\*/\.\*/g;
$pattern =~ s/\?/\./g;
if ($glob eq 'ign') {
# We're in a local: * section; just continue.
print "$_\n";
next;
}
# Print the glob commented for human readers.
print "$ws##$ptn ($glob)\n";
# We tried to match a symbol to this version.
$matches_attempted++;
if ($glob eq 'glob') {
my %ptn_syms = ();
# Match ptn against symbols in %sym_hash.
foreach my $sym (keys %sym_hash) {
# Maybe it matches one of the patterns based on the symbol in
# the .o file.
$ptn_syms{$sym}++ if ($sym =~ /^$pattern$/);
}
foreach my $sym (sort keys(%ptn_syms)) {
$matched_symbols++;
print "$ws$sym;\n";
}
} elsif ($glob eq 'cxx') {
my %dem_syms = ();
# Verify that we're actually using GNU c++filt. Other versions
# most likely cannot handle GNU style symbol mangling.
my $cxxout = `$cxxfilt --version 2>&1`;
$cxxout =~ m/GNU/ or die "$0 requires GNU c++filt to function";
# Talk to c++filt through a pair of file descriptors.
# Need to start a fresh instance per pattern, otherwise the
# process grows to 500+ MB.
my $pid = open2(*FILTIN, *FILTOUT, $cxxfilt) or die $!;
# Match ptn against symbols in %sym_hash.
foreach my $sym (keys %sym_hash) {
# No? Well, maybe its demangled form matches one of those
# patterns.
printf FILTOUT "%s\n",$sym;
my $dem = <FILTIN>;
chomp $dem;
$dem_syms{$sym}++ if ($dem =~ /^$pattern$/);
}
close FILTOUT or die "c++filt error";
close FILTIN or die "c++filt error";
# Need to wait for the c++filt process to avoid lots of zombies.
waitpid $pid, 0;
foreach my $sym (sort keys(%dem_syms)) {
$matched_symbols++;
print "$ws$sym;\n";
}
} else {
# No? Well, then ignore it.
}
next;
}
# Important sanity check. This script can't handle lots of formats
# that GNU ld can, so be sure to error out if one is seen!
die "strange line `$_'";
}
close F;

View File

@ -1,24 +1,3 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_TMS470_MATH_H
#define COMPAT_TMS470_MATH_H
#include_next <math.h>
#undef INFINITY
@ -26,5 +5,3 @@
#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
#define NAN (*(const float*)((const unsigned []){ 0x7fc00000 }))
#endif /* COMPAT_TMS470_MATH_H */

View File

@ -19,16 +19,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_VA_COPY_H
#define COMPAT_VA_COPY_H
#include <stdarg.h>
#if !defined(va_copy) && defined(_MSC_VER)
#define va_copy(dst, src) ((dst) = (src))
#endif
#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
#define va_copy(dst, src) __va_copy(dst, src)
#endif
#endif /* COMPAT_VA_COPY_H */

View File

@ -26,8 +26,8 @@
* w32threads to pthreads wrapper
*/
#ifndef COMPAT_W32PTHREADS_H
#define COMPAT_W32PTHREADS_H
#ifndef LIBAV_W32PTHREADS_H
#define LIBAV_W32PTHREADS_H
/* Build up a pthread-like API using underlying Windows API. Have only static
* methods so as to not conflict with a potentially linked in pthread-win32
@ -39,12 +39,6 @@
#include <windows.h>
#include <process.h>
#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
#undef MemoryBarrier
#define MemoryBarrier __sync_synchronize
#endif
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/internal.h"
#include "libavutil/mem.h"
@ -60,56 +54,45 @@ typedef struct pthread_t {
* 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
/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
* conditional variables on kernels 6.0+.
* MinGW does not currently have this typedef. */
typedef struct pthread_cond_t {
void *Ptr;
void *ptr;
} pthread_cond_t;
#endif
#if _WIN32_WINNT >= 0x0600
#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
#endif
/* 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 unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t *h = arg;
h->ret = h->func(h->arg);
return 0;
}
static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
static int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
{
thread->func = start_routine;
thread->arg = arg;
#if HAVE_WINRT
thread->handle = (void*)CreateThread(NULL, 0, win32thread_worker, thread,
0, NULL);
#else
thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
0, NULL);
#endif
return !thread->handle;
}
static av_unused int pthread_join(pthread_t thread, void **value_ptr)
static void pthread_join(pthread_t thread, void **value_ptr)
{
DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
if (ret != WAIT_OBJECT_0) {
if (ret == WAIT_ABANDONED)
return EINVAL;
else
return EDEADLK;
}
if (ret != WAIT_OBJECT_0)
return;
if (value_ptr)
*value_ptr = thread.ret;
CloseHandle(thread.handle);
return 0;
}
static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
@ -133,115 +116,8 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
return 0;
}
#if _WIN32_WINNT >= 0x0600
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
BOOL pending = FALSE;
InitOnceBeginInitialize(once_control, 0, &pending, NULL);
if (pending)
init_routine();
InitOnceComplete(once_control, 0, NULL);
return 0;
}
static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
InitializeConditionVariable(cond);
return 0;
}
/* native condition variables do not destroy */
static inline int pthread_cond_destroy(pthread_cond_t *cond)
{
return 0;
}
static inline int pthread_cond_broadcast(pthread_cond_t *cond)
{
WakeAllConditionVariable(cond);
return 0;
}
static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
SleepConditionVariableCS(cond, mutex, INFINITE);
return 0;
}
static inline int pthread_cond_signal(pthread_cond_t *cond)
{
WakeConditionVariable(cond);
return 0;
}
#else // _WIN32_WINNT < 0x0600
/* atomic init state of dynamically loaded functions */
static LONG w32thread_init_state = 0;
static av_unused void w32thread_init(void);
/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
* compatible to the one used in the native API */
typedef union pthread_once_t {
void * Ptr; ///< For the Windows 6.0+ native functions
LONG state; ///< For the pre-Windows 6.0 compat code
} pthread_once_t;
#define PTHREAD_ONCE_INIT {0}
/* function pointers to init once API on windows 6.0+ kernels */
static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
/* pre-Windows 6.0 compat using a spin-lock */
static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
{
switch (InterlockedCompareExchange(state, 1, 0)) {
/* Initial run */
case 0:
init_routine();
InterlockedExchange(state, 2);
break;
/* Another thread is running init */
case 1:
while (1) {
MemoryBarrier();
if (*state == 2)
break;
Sleep(0);
}
break;
/* Initialization complete */
case 2:
break;
}
}
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
/* Use native functions on Windows 6.0+ */
if (initonce_begin && initonce_complete) {
BOOL pending = FALSE;
initonce_begin(once_control, 0, &pending, NULL);
if (pending)
init_routine();
initonce_complete(once_control, 0, NULL);
return 0;
}
w32thread_once_fallback(&once_control->state, init_routine);
return 0;
}
/* for pre-Windows 6.0 platforms we need to define and use our own condition
* variable and api */
typedef struct win32_cond_t {
pthread_mutex_t mtx_broadcast;
pthread_mutex_t mtx_waiter_count;
@ -251,47 +127,36 @@ typedef struct win32_cond_t {
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)
static void 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;
return;
}
/* non native condition variables */
win32_cond = av_mallocz(sizeof(win32_cond_t));
if (!win32_cond)
return ENOMEM;
cond->Ptr = win32_cond;
return;
cond->ptr = win32_cond;
win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
if (!win32_cond->semaphore)
return ENOMEM;
return;
win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
if (!win32_cond->waiters_done)
return ENOMEM;
return;
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)
static void pthread_cond_destroy(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
/* native condition variables do not destroy */
if (cond_init)
return 0;
return;
/* non native condition variables */
CloseHandle(win32_cond->semaphore);
@ -299,18 +164,17 @@ static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
pthread_mutex_destroy(&win32_cond->mtx_broadcast);
av_freep(&win32_cond);
cond->Ptr = NULL;
return 0;
cond->ptr = NULL;
}
static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
static void pthread_cond_broadcast(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int have_waiter;
if (cond_broadcast) {
cond_broadcast(cond);
return 0;
return;
}
/* non native condition variables */
@ -332,12 +196,11 @@ static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
} 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)
static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int last_waiter;
if (cond_wait) {
cond_wait(cond, mutex, INFINITE);
@ -367,13 +230,13 @@ static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mu
return pthread_mutex_lock(mutex);
}
static av_unused int pthread_cond_signal(pthread_cond_t *cond)
static void pthread_cond_signal(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int have_waiter;
if (cond_signal) {
cond_signal(cond);
return 0;
return;
}
pthread_mutex_lock(&win32_cond->mtx_broadcast);
@ -390,11 +253,9 @@ static av_unused int pthread_cond_signal(pthread_cond_t *cond)
}
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
#endif
static av_unused void w32thread_init(void)
static void w32thread_init(void)
{
#if _WIN32_WINNT < 0x0600
HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
@ -407,12 +268,13 @@ static av_unused void w32thread_init(void)
(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");
#else
cond_init = InitializeConditionVariable;
cond_broadcast = WakeAllConditionVariable;
cond_signal = WakeConditionVariable;
cond_wait = SleepConditionVariableCS;
#endif
}
#endif /* COMPAT_W32PTHREADS_H */
#endif /* LIBAV_W32PTHREADS_H */

View File

@ -1,132 +0,0 @@
#!/bin/sh
# Copyright (c) 2013, Derek Buitenhuis
#
# Permission to use, copy, modify, and/or distribute this software for any
# purpose with or without fee is hereby granted, provided that the above
# copyright notice and this permission notice appear in all copies.
#
# THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
# WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
# MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
# ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
# WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
# ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
# OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
# mktemp isn't POSIX, so supply an implementation
mktemp() {
echo "${2%%XXX*}.${HOSTNAME}.${UID}.$$"
}
if [ $# -lt 2 ]; then
echo "Usage: makedef <version_script> <objects>" >&2
exit 0
fi
vscript=$1
shift
if [ ! -f "$vscript" ]; then
echo "Version script does not exist" >&2
exit 1
fi
for object in "$@"; do
if [ ! -f "$object" ]; then
echo "Object does not exist: ${object}" >&2
exit 1
fi
done
# Create a lib temporarily to dump symbols from.
# It's just much easier to do it this way
libname=$(mktemp -u "library").lib
trap 'rm -f -- $libname' EXIT
lib -out:${libname} $@ >/dev/null
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
exit 1
fi
IFS='
'
# Determine if we're building for x86 or x86_64 and
# set the symbol prefix accordingly.
prefix=""
arch=$(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"
for line in $(cat ${vscript} | tr '\t' ' '); do
# We only care about global symbols
echo "${line}" | grep -q '^ \+global:'
if [ $? = 0 ]; then
started=1
line=$(echo "${line}" | sed -e 's/^ \{1,\}global: *//')
else
echo "${line}" | grep -q '^ \+local:'
if [ $? = 0 ]; then
started=0
fi
fi
if [ ${started} = 0 ]; then
continue
fi
# Handle multiple symbols on one line
IFS=';'
# Work around stupid expansion to filenames
line=$(echo "${line}" | sed -e 's/\*/.\\+/g')
for exp in ${line}; do
# Remove leading and trailing whitespace
exp=$(echo "${exp}" | sed -e 's/^ *//' -e 's/ *$//')
if [ "${regex}" = "none" ]; then
regex="${exp}"
else
regex="${regex};${exp}"
fi
done
IFS='
'
done
dump=$(dumpbin -linkermember:1 ${libname})
rm ${libname}
IFS=';'
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
echo "EXPORTS"
echo "${list}" | sort | uniq | tail -n +2

View File

@ -1,9 +0,0 @@
#!/bin/sh
LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
if [ -x "$LINK_EXE_PATH" ]; then
"$LINK_EXE_PATH" $@
else
link $@
fi
exit $?

3679
configure vendored

File diff suppressed because it is too large Load Diff

9
doc/.gitignore vendored
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@ -1,9 +0,0 @@
/*.1
/*.3
/*.html
/*.pod
/config.texi
/avoptions_codec.texi
/avoptions_format.texi
/fate.txt
/print_options

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

View File

@ -14,11 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols
COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices
COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters
MANPAGES1 = $(AVPROGS-yes:%=doc/%.1) $(AVPROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1)
MANPAGES1 = $(PROGS-yes:%=doc/%.1) $(PROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1)
MANPAGES3 = $(LIBRARIES-yes:%=doc/%.3)
MANPAGES = $(MANPAGES1) $(MANPAGES3)
PODPAGES = $(AVPROGS-yes:%=doc/%.pod) $(AVPROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod)
HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
PODPAGES = $(PROGS-yes:%=doc/%.pod) $(PROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod)
HTMLPAGES = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
@ -36,30 +36,6 @@ DOCS-$(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
@ -67,9 +43,7 @@ doc: documentation
apidoc: doc/doxy/html
documentation: $(DOCS)
examples: $(DOC_EXAMPLES)
TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
@ -84,25 +58,14 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%-all.html: TAG = HTML
ifdef HAVE_MAKEINFO_HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
else
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%-all.html: TAG = HTML
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
endif
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
@ -116,35 +79,17 @@ doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
doc/%.1 doc/%.3: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
$(M)pod2man --section=3 --center=" " --release=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
OBJDIRS += doc/examples
DOXY_INPUT = $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)
DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) config.mak
doc/doxy/html: TAG = DOXY
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
install-doc: install-html install-man
install-html:
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
install-man:
ifdef CONFIG_HTMLPAGES
install-progs-$(CONFIG_DOC): install-html
install-html: $(HTMLPAGES)
$(Q)mkdir -p "$(DOCDIR)"
$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
endif
ifdef CONFIG_MANPAGES
install-progs-$(CONFIG_DOC): install-man
@ -155,15 +100,10 @@ install-man: $(MANPAGES)
$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
endif
uninstall: uninstall-doc
uninstall-doc: uninstall-html uninstall-man
uninstall-html:
$(RM) -r "$(DOCDIR)"
uninstall: uninstall-man
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
clean:: docclean
@ -171,13 +111,8 @@ clean:: docclean
distclean:: docclean
$(RM) doc/config.texi
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
docclean:
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
$(RM) -r doc/doxy/html
-include $(wildcard $(DOCS:%=%.d))

16
doc/RELEASE_NOTES Normal file
View File

@ -0,0 +1,16 @@
Release Notes
=============
* 2.0 "Nameless" July, 2013
General notes
-------------
See the Changelog file for a list of significant changes. Note, there
are many more new features and bugfixes than whats listed there.
Bugreports against FFmpeg git master or the most recent FFmpeg release are
accepted. If you are experiencing issues with any formally released version of
FFmpeg, please try git master to check if the issue still exists. If it does,
make your report against the development code following the usual bug reporting
guidelines.

View File

@ -3,7 +3,7 @@ representing a number as input, which may be followed by one of the SI
unit prefixes, for example: 'K', 'M', or 'G'.
If 'i' is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiples, which are based on
interpreted as a unit prefix for binary multiplies, which are based on
powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
prefix multiplies the value by 8. This allows using, for example:
'KB', 'MiB', 'G' and 'B' as number suffixes.
@ -36,28 +36,16 @@ Possible forms of stream specifiers are:
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
streams, 'V' only matches video streams which are not attached pictures, video
thumbnails or cover arts. If @var{stream_index} is given, then it matches
@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
stream number @var{stream_index} of this type. Otherwise, it matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
in the program with the id @var{program_id}. Otherwise, it matches all streams in the
program.
@item #@var{stream_id} or i:@var{stream_id}
Match the stream by stream id (e.g. PID in MPEG-TS container).
@item m:@var{key}[:@var{value}]
Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value.
@item u
Matches streams with usable configuration, the codec must be defined and the
essential information such as video dimension or audio sample rate must be present.
Note that in @command{ffmpeg}, matching by metadata will only work properly for
input files.
@item #@var{stream_id}
Matches the stream by a format-specific ID.
@end table
@section Generic options
@ -71,18 +59,10 @@ Show license.
@item -h, -?, -help, --help [@var{arg}]
Show help. An optional parameter may be specified to print help about a specific
item. If no argument is specified, only basic (non advanced) tool
options are shown.
item.
Possible values of @var{arg} are:
@table @option
@item long
Print advanced tool options in addition to the basic tool options.
@item full
Print complete list of options, including shared and private options
for encoders, decoders, demuxers, muxers, filters, etc.
@item decoder=@var{decoder_name}
Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@ -102,16 +82,14 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
@item filter=@var{filter_name}
Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
@end table
@item -version
Show version.
@item -formats
Show available formats (including devices).
@item -devices
Show available devices.
Show available formats.
@item -codecs
Show all codecs known to libavcodec.
@ -143,25 +121,6 @@ Show available sample formats.
@item -layouts
Show channel names and standard channel layouts.
@item -colors
Show recognized color names.
@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
Show autodetected sources of the intput device.
Some devices may provide system-dependent source names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
@example
ffmpeg -sources pulse,server=192.168.0.4
@end example
@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
Show autodetected sinks of the output device.
Some devices may provide system-dependent sink names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
@example
ffmpeg -sinks pulse,server=192.168.0.4
@end example
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
Set the logging level used by the library.
Adding "repeat+" indicates that repeated log output should not be compressed
@ -170,29 +129,28 @@ omitted. "repeat" can also be used alone.
If "repeat" is used alone, and with no prior loglevel set, the default
loglevel will be used. If multiple loglevel parameters are given, using
'repeat' will not change the loglevel.
@var{loglevel} is a string or a number containing one of the following values:
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet, -8
@item quiet
Show nothing at all; be silent.
@item panic, 0
@item panic
Only show fatal errors which could lead the process to crash, such as
and assert failure. This is not currently used for anything.
@item fatal, 8
@item fatal
Only show fatal errors. These are errors after which the process absolutely
cannot continue after.
@item error, 16
@item error
Show all errors, including ones which can be recovered from.
@item warning, 24
@item warning
Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.
@item info, 32
@item info
Show informative messages during processing. This is in addition to
warnings and errors. This is the default value.
@item verbose, 40
@item verbose
Same as @code{info}, except more verbose.
@item debug, 48
@item debug
Show everything, including debugging information.
@item trace, 56
@end table
By default the program logs to stderr, if coloring is supported by the
@ -210,39 +168,22 @@ directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Setting the environment variable @env{FFREPORT} to any value has the
Setting the environment variable @code{FFREPORT} to any value has the
same effect. If the value is a ':'-separated key=value sequence, these
options will affect the report; option values must be escaped if they
options will affect the report; options values must be escaped if they
contain special characters or the options delimiter ':' (see the
``Quoting and escaping'' section in the ffmpeg-utils manual).
The following options are recognized:
``Quoting and escaping'' section in the ffmpeg-utils manual). The
following option is recognized:
@table @option
@item file
set the file name to use for the report; @code{%p} is expanded to the name
of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
to a plain @code{%}
@item level
set the log verbosity level using a numerical value (see @code{-loglevel}).
@end table
For example, to output a report to a file named @file{ffreport.log}
using a log level of @code{32} (alias for log level @code{info}):
@example
FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
@end example
Errors in parsing the environment variable are not fatal, and will not
appear in the report.
@item -hide_banner
Suppress printing banner.
All FFmpeg tools will normally show a copyright notice, build options
and library versions. This option can be used to suppress printing
this information.
@item -cpuflags flags (@emph{global})
Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you're doing.
@ -267,14 +208,10 @@ Possible flags for this option are:
@item sse4.1
@item sse4.2
@item avx
@item avx2
@item xop
@item fma3
@item fma4
@item 3dnow
@item 3dnowext
@item bmi1
@item bmi2
@item cmov
@end table
@item ARM
@ -285,13 +222,6 @@ Possible flags for this option are:
@item vfp
@item vfpv3
@item neon
@item setend
@end table
@item AArch64
@table @samp
@item armv8
@item vfp
@item neon
@end table
@item PowerPC
@table @samp
@ -310,43 +240,6 @@ Possible flags for this option are:
@end table
@end table
@item -opencl_bench
This option is used to benchmark all available OpenCL devices and print the
results. This option is only available when FFmpeg has been compiled with
@code{--enable-opencl}.
When FFmpeg is configured with @code{--enable-opencl}, the options for the
global OpenCL context are set via @option{-opencl_options}. See the
"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
supported options. Amongst others, these options include the ability to select
a specific platform and device to run the OpenCL code on. By default, FFmpeg
will run on the first device of the first platform. While the options for the
global OpenCL context provide flexibility to the user in selecting the OpenCL
device of their choice, most users would probably want to select the fastest
OpenCL device for their system.
This option assists the selection of the most efficient configuration by
identifying the appropriate device for the user's system. The built-in
benchmark is run on all the OpenCL devices and the performance is measured for
each device. The devices in the results list are sorted based on their
performance with the fastest device listed first. The user can subsequently
invoke @command{ffmpeg} using the device deemed most appropriate via
@option{-opencl_options} to obtain the best performance for the OpenCL
accelerated code.
Typical usage to use the fastest OpenCL device involve the following steps.
Run the command:
@example
ffmpeg -opencl_bench
@end example
Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
i.e. fastest device in the list.
Select the platform and device using the command:
@example
ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
@end example
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with @code{--enable-opencl}.
@ -378,12 +271,11 @@ muxer:
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them.
All codec AVOptions are obviously per-stream, so the chapter on stream
specifiers applies to them
Note: the @option{-nooption} syntax cannot be used for boolean
AVOptions, use @option{-option 0}/@option{-option 1}.
Note @option{-nooption} syntax cannot be used for boolean AVOptions,
use @option{-option 0}/@option{-option 1}.
Note: the old undocumented way of specifying per-stream AVOptions by
prepending v/a/s to the options name is now obsolete and will be
removed soon.
Note2 old undocumented way of specifying per-stream AVOptions by prepending
v/a/s to the options name is now obsolete and will be removed soon.

36
doc/avutil.txt Normal file
View File

@ -0,0 +1,36 @@
AVUtil
======
libavutil is a small lightweight library of generally useful functions.
It is not a library for code needed by both libavcodec and libavformat.
Overview:
=========
adler32.c adler32 checksum
aes.c AES encryption and decryption
fifo.c resizeable first in first out buffer
intfloat_readwrite.c portable reading and writing of floating point values
log.c "printf" with context and level
md5.c MD5 Message-Digest Algorithm
rational.c code to perform exact calculations with rational numbers
tree.c generic AVL tree
crc.c generic CRC checksumming code
integer.c 128bit integer math
lls.c
mathematics.c greatest common divisor, integer sqrt, integer log2, ...
mem.c memory allocation routines with guaranteed alignment
Headers:
bswap.h big/little/native-endian conversion code
x86_cpu.h a few useful macros for unifying x86-64 and x86-32 code
avutil.h
common.h
intreadwrite.h reading and writing of unaligned big/little/native-endian integers
Goals:
======
* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
* Small (source and object)
* Efficient (low CPU and memory usage)
* Useful (avoid useless features almost no one needs)

View File

@ -13,16 +13,7 @@ bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
The ff* tools have a -bsf option applied per stream, taking a
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
@end example
Below is a description of the currently available bitstream filters,
with their parameters, if any.
Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
@ -39,37 +30,7 @@ ADTS AAC container to a FLV or a MOV/MP4 file.
Remove zero padding at the end of a packet.
@section dump_extra
Add extradata to the beginning of the filtered packets.
The additional argument specifies which packets should be filtered.
It accepts the values:
@table @samp
@item a
add extradata to all key packets, but only if @var{local_header} is
set in the @option{flags2} codec context field
@item k
add extradata to all key packets
@item e
add extradata to all packets
@end table
If not specified it is assumed @samp{k}.
For example the following @command{ffmpeg} command forces a global
header (thus disabling individual packet headers) in the H.264 packets
generated by the @code{libx264} encoder, but corrects them by adding
the header stored in extradata to the key packets:
@example
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dca_core
Extract DCA core from DTS-HD streams.
@section dump_extradata
@section h264_mp4toannexb
@ -87,18 +48,7 @@ format with @command{ffmpeg}, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
@section imxdump
Modifies the bitstream to fit in MOV and to be usable by the Final Cut
Pro decoder. This filter only applies to the mpeg2video codec, and is
likely not needed for Final Cut Pro 7 and newer with the appropriate
@option{-tag:v}.
For example, to remux 30 MB/sec NTSC IMX to MOV:
@example
ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
@end example
@section imx_dump_header
@section mjpeg2jpeg
@ -141,44 +91,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@section movsub
@section mp3_header_compress
@section mp3_header_decompress
@section mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
DivX-style packed B-frames are not valid MPEG-4 and were only a
workaround for the broken Video for Windows subsystem.
They use more space, can cause minor AV sync issues, require more
CPU power to decode (unless the player has some decoded picture queue
to compensate the 2,0,2,0 frame per packet style) and cause
trouble if copied into a standard container like mp4 or mpeg-ps/ts,
because MPEG-4 decoders may not be able to decode them, since they are
not valid MPEG-4.
For example to fix an AVI file containing an MPEG-4 stream with
DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@end example
@section noise
Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
Parameters:
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.
@example
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@end example
applies the modification to every byte.
@section remove_extra
@section remove_extradata
@c man end BITSTREAM FILTERS

File diff suppressed because one or more lines are too long

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@ -7,55 +7,44 @@ V
Disable the default terse mode, the full command issued by make and its
output will be shown on the screen.
DBG
Preprocess x86 external assembler files to a .dbg.asm file in the object
directory, which then gets compiled. Helps in developing those assembler
files.
DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
GEN
Set to 1 to generate the missing or mismatched references.
Makefile targets:
all
Default target, builds all the libraries and the executables.
fate
Run the fate test suite, note that you must have installed it.
Run the fate test suite, note you must have installed it
fate-list
List all fate/regression test targets.
Will list all fate/regression test targets
install
Install headers, libraries and programs.
examples
Build all examples located in doc/examples.
libavformat/output-example
Build the libavformat basic example.
libswscale/swscale-test
Build the swscale self-test (useful also as an example).
libavcodec/api-example
Build the libavcodec basic example.
config
Reconfigure the project with the current configuration.
libswscale/swscale-test
Build the swscale self-test (useful also as example).
Useful standard make commands:
make -t <target>
Touch all files that otherwise would be built, this is useful to reduce
unneeded rebuilding when changing headers, but note that you must force rebuilds
Touch all files that otherwise would be build, this is useful to reduce
unneeded rebuilding when changing headers, but note you must force rebuilds
of files that actually need it by hand then.
make -j<num>
Rebuild with multiple jobs at the same time. Faster on multi processor systems.
rebuild with multiple jobs at the same time. Faster on multi processor systems
make -k
Continue build in case of errors, this is useful for the regression tests
sometimes but note that it will still not run all reg tests.
continue build in case of errors, this is useful for the regression tests
sometimes but note it will still not run all reg tests.

View File

@ -7,7 +7,7 @@ all the encoders and decoders. In addition each codec may support
so-called private options, which are specific for a given codec.
Sometimes, a global option may only affect a specific kind of codec,
and may be nonsensical or ignored by another, so you need to be aware
and may be unsensical or ignored by another, so you need to be aware
of the meaning of the specified options. Also some options are
meant only for decoding or encoding.
@ -71,9 +71,7 @@ Force low delay.
@item global_header
Place global headers in extradata instead of every keyframe.
@item bitexact
Only write platform-, build- and time-independent data. (except (I)DCT).
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
Use only bitexact stuff (except (I)DCT).
@item aic
Apply H263 advanced intra coding / mpeg4 ac prediction.
@item cbp
@ -86,6 +84,9 @@ Apply interlaced motion estimation.
Use closed gop.
@end table
@item sub_id @var{integer}
Deprecated, currently unused.
@item me_method @var{integer} (@emph{encoding,video})
Set motion estimation method.
@ -129,7 +130,7 @@ should be @code{1 / frame_rate} and timestamp increments should be
identically 1.
@item g @var{integer} (@emph{encoding,video})
Set the group of picture (GOP) size. Default value is 12.
Set the group of picture size. Default value is 12.
@item ar @var{integer} (@emph{decoding/encoding,audio})
Set audio sampling rate (in Hz).
@ -174,13 +175,7 @@ Set max video quantizer scale (VBR). Must be included between -1 and
Set max difference between the quantizer scale (VBR).
@item bf @var{integer} (@emph{encoding,video})
Set max number of B frames between non-B-frames.
Must be an integer between -1 and 16. 0 means that B-frames are
disabled. If a value of -1 is used, it will choose an automatic value
depending on the encoder.
Default value is 0.
Set max number of B frames.
@item b_qfactor @var{float} (@emph{encoding,video})
Set qp factor between P and B frames.
@ -257,7 +252,7 @@ Specify how strictly to follow the standards.
Possible values:
@table @samp
@item very
strictly conform to an older more strict version of the spec or reference software
strictly conform to a older more strict version of the spec or reference software
@item strict
strictly conform to all the things in the spec no matter what consequences
@item normal
@ -287,11 +282,6 @@ detect bitstream specification deviations
detect improper bitstream length
@item explode
abort decoding on minor error detection
@item ignore_err
ignore decoding errors, and continue decoding.
This is useful if you want to analyze the content of a video and thus want
everything to be decoded no matter what. This option will not result in a video
that is pleasing to watch in case of errors.
@item careful
consider things that violate the spec and have not been seen in the wild as errors
@item compliant
@ -396,8 +386,9 @@ Possible values:
@item simplemmx
@item simpleauto
Automatically pick a IDCT compatible with the simple one
@item libmpeg2mmx
@item mmi
@item arm
@ -415,6 +406,10 @@ Automatically pick a IDCT compatible with the simple one
@item simplealpha
@item h264
@item vp3
@item ipp
@item xvidmmx
@ -434,8 +429,6 @@ Possible values:
iterative motion vector (MV) search (slow)
@item deblock
use strong deblock filter for damaged MBs
@item favor_inter
favor predicting from the previous frame instead of the current
@end table
@item bits_per_coded_sample @var{integer}
@ -456,9 +449,6 @@ Possible values:
@item aspect @var{rational number} (@emph{encoding,video})
Set sample aspect ratio.
@item sar @var{rational number} (@emph{encoding,video})
Set sample aspect ratio. Alias to @var{aspect}.
@item debug @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
Print specific debug info.
@ -478,9 +468,6 @@ per-block quantization parameter (QP)
motion vector
@item dct_coeff
@item green_metadata
display complexity metadata for the upcoming frame, GoP or for a given duration.
@item skip
@item startcode
@ -501,15 +488,11 @@ visualize block types
picture buffer allocations
@item thread_ops
threading operations
@item nomc
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
@ -789,29 +772,26 @@ Set noise reduction.
Set number of bits which should be loaded into the rc buffer before
decoding starts.
@item inter_threshold @var{integer} (@emph{encoding,video})
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
Possible values:
@table @samp
@item fast
Allow non spec compliant speedup tricks.
allow non spec compliant speedup tricks
@item sgop
Deprecated, use mpegvideo private options instead.
Deprecated, use mpegvideo private options instead
@item noout
Skip bitstream encoding.
@item ignorecrop
Ignore cropping information from sps.
skip bitstream encoding
@item local_header
Place global headers at every keyframe instead of in extradata.
place global headers at every keyframe instead of in extradata
@item chunks
Frame data might be split into multiple chunks.
Frame data might be split into multiple chunks
@item showall
Show all frames before the first keyframe.
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}.
Deprecated, use mpegvideo private options instead
@end table
@item error @var{integer} (@emph{encoding,video})
@ -820,17 +800,13 @@ for codecs that support it. See also @file{doc/examples/export_mvs.c}.
Deprecated, use mpegvideo private options instead.
@item threads @var{integer} (@emph{decoding/encoding,video})
Set the number of threads to be used, in case the selected codec
implementation supports multi-threading.
Possible values:
@table @samp
@item auto, 0
automatically select the number of threads to set
@item auto
detect a good number of threads
@end table
Default value is @samp{auto}.
@item me_threshold @var{integer} (@emph{encoding,video})
Set motion estimation threshold.
@ -871,18 +847,6 @@ Possible values:
@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
@ -914,9 +878,6 @@ Set frame skip factor.
@item skip_exp @var{integer} (@emph{encoding,video})
Set frame skip exponent.
Negative values behave identical to the corresponding positive ones, except
that the score is normalized.
Positive values exist primarily for compatibility reasons and are not so useful.
@item skipcmp @var{integer} (@emph{encoding,video})
Set frame skip compare function.
@ -1050,44 +1011,9 @@ Possible values:
@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})
@item color_trc @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item bt709
BT.709
@item gamma22
BT.470 M
@item gamma28
BT.470 BG
@item linear
SMPTE 170 M
@item log
SMPTE 240 M
@item log_sqrt
Linear
@item iec61966_2_4
Log
@item bt1361
Log square root
@item iec61966_2_1
IEC 61966-2-4
@item bt2020_10bit
BT.1361
@item bt2020_12bit
IEC 61966-2-1
@item smpte2084
BT.2020 - 10 bit
@item smpte428_1
BT.2020 - 12 bit
@end table
@item colorspace @var{integer} (@emph{decoding/encoding,video})
@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.
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
@item log_level_offset @var{integer}
@ -1097,26 +1023,15 @@ Set the log level offset.
Number of slices, used in parallelized encoding.
@item thread_type @var{flags} (@emph{decoding/encoding,video})
Select which multithreading methods to use.
Use of @samp{frame} will increase decoding delay by one frame per
thread, so clients which cannot provide future frames should not use
it.
Select multithreading type.
Possible values:
@table @samp
@item slice
Decode more than one part of a single frame at once.
Multithreading using slices works only when the video was encoded with
slices.
@item frame
Decode more than one frame at once.
@end table
Default value is @samp{slice+frame}.
@item audio_service_type @var{integer} (@emph{encoding,audio})
Set audio service type.
@ -1150,47 +1065,9 @@ Set sample format audio decoders should prefer. Default value is
@item sub_charenc @var{encoding} (@emph{decoding,subtitles})
Set the input subtitles character encoding.
@item field_order @var{field_order} (@emph{video})
Set/override the field order of the video.
Possible values:
@table @samp
@item progressive
Progressive video
@item tt
Interlaced video, top field coded and displayed first
@item bb
Interlaced video, bottom field coded and displayed first
@item tb
Interlaced video, top coded first, bottom displayed first
@item bt
Interlaced video, bottom coded first, top displayed first
@end table
@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.
@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 indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@end table
@c man end CODEC OPTIONS
@ifclear config-writeonly
@include decoders.texi
@end ifclear
@ifclear config-readonly
@include encoders.texi
@end ifclear

View File

@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
The option @code{-decoders} of the ff* tools will display the list of
The option @code{-codecs} of the ff* tools will display the list of
enabled decoders.
@c man end DECODERS
@ -25,13 +25,6 @@ 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.
@ -59,54 +52,6 @@ top-field-first is assumed
@chapter Audio Decoders
@c man begin AUDIO DECODERS
A description of some of the currently available audio decoders
follows.
@section ac3
AC-3 audio decoder.
This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
@subsection AC-3 Decoder Options
@table @option
@item -drc_scale @var{value}
Dynamic Range Scale Factor. The factor to apply to dynamic range values
from the AC-3 stream. This factor is applied exponentially.
There are 3 notable scale factor ranges:
@table @option
@item drc_scale == 0
DRC disabled. Produces full range audio.
@item 0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value.
Audio reproduction is between full range and full compression.
@item drc_scale > 1
DRC enabled. Applies drc_scale asymmetrically.
Loud sounds are fully compressed. Soft sounds are enhanced.
@end table
@end table
@section flac
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from Xiph.
@subsection FLAC Decoder options
@table @option
@item -use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc values
(like the default value). This option makes it possible to decode such streams
correctly by using lavc's old buggy lpc logic for decoding.
@end table
@section ffwavesynth
Internal wave synthetizer.
@ -187,33 +132,11 @@ Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
An FFmpeg native decoder for Opus exists, so users can decode Opus
without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section dvbsub
@subsection Options
@table @option
@item compute_clut
@table @option
@item -1
Compute clut if no matching CLUT is in the stream.
@item 0
Never compute CLUT
@item 1
Always compute CLUT and override the one provided in the stream.
@end table
@item dvb_substream
Selects the dvb substream, or all substreams if -1 which is default.
@end table
@section dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
@ -233,62 +156,6 @@ The format for this option is a string containing 16 24-bits hexadecimal
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}.
@item ifo_palette
Specify the IFO file from which the global palette is obtained.
(experimental)
@item forced_subs_only
Only decode subtitle entries marked as forced. Some titles have forced
and non-forced subtitles in the same track. Setting this flag to @code{1}
will only keep the forced subtitles. Default value is @code{0}.
@end table
@section libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
subtitles. Requires the presence of the libzvbi headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libzvbi}.
@subsection Options
@table @option
@item txt_page
List of teletext page numbers to decode. You may use the special * string to
match all pages. Pages that do not match the specified list are dropped.
Default value is *.
@item txt_chop_top
Discards the top teletext line. Default value is 1.
@item txt_format
Specifies the format of the decoded subtitles. The teletext decoder is capable
of decoding the teletext pages to bitmaps or to simple text, you should use
"bitmap" for teletext pages, because certain graphics and colors cannot be
expressed in simple text. You might use "text" for teletext based subtitles if
your application can handle simple text based subtitles. Default value is
bitmap.
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
Y offset of generated bitmaps, default is 0.
@item txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the generated
text. This option is useful for teletext based subtitles where empty spaces may
be present at the start or at the end of the lines or empty lines may be
present between the subtitle lines because of double-sized teletext charactes.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
milliseconds. 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.
@item txt_opacity
Sets the opacity (0-255) of the teletext background. If
@option{txt_transparent} is not set, it only affects characters between a start
box and an end box, typically subtitles. Default value is 0 if
@option{txt_transparent} is set, 255 otherwise.
@end table
@c man end SUBTILES DECODERS

View File

@ -1,7 +1,3 @@
a.summary-letter {
text-decoration: none;
}
a {
color: #2D6198;
}
@ -17,8 +13,8 @@ a:visited {
}
#banner img {
margin-bottom: 1px;
margin-top: 5px;
padding-bottom: 1px;
padding-top: 5px;
}
#body {
@ -49,16 +45,11 @@ body {
text-align: center;
}
h1 a, h2 a, h3 a, h4 a {
text-decoration: inherit;
color: inherit;
}
h1, h2, h3, h4 {
h1, h2, h3 {
padding-left: 0.4em;
border-radius: 4px;
padding-bottom: 0.25em;
padding-top: 0.25em;
padding-bottom: 0.2em;
padding-top: 0.2em;
border: 1px solid #6A996A;
}
@ -72,22 +63,15 @@ h1 {
h2 {
color: #313131;
font-size: 1.0em;
font-size: 0.9em;
background-color: #ABE3AB;
}
h3 {
color: #313131;
font-size: 0.9em;
margin-bottom: -6px;
background-color: #BBF3BB;
}
h4 {
color: #313131;
font-size: 0.8em;
margin-bottom: -8px;
background-color: #D1FDD1;
background-color: #BBF3BB;
}
img {

View File

@ -1,7 +1,7 @@
@chapter Demuxers
@c man begin DEMUXERS
Demuxers are configured elements in FFmpeg that can read the
Demuxers are configured elements in FFmpeg which allow to read the
multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
@ -18,12 +18,6 @@ enabled demuxers.
The description of some of the currently available demuxers follows.
@section aa
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.
@ -35,37 +29,6 @@ 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.
This demuxer is used to demux APNG files.
All headers, but the PNG signature, up to (but not including) the first
fcTL chunk are transmitted as extradata.
Frames are then split as being all the chunks between two fcTL ones, or
between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
Ignore the loop variable in the file if set.
@item -max_fps @var{int}
Maximum framerate in frames per second (0 for no limit).
@item -default_fps @var{int}
Default framerate in frames per second when none is specified in the file
(0 meaning as fast as possible).
@end table
@section asf
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
@table @option
@item -no_resync_search @var{bool}
Do not try to resynchronize by looking for a certain optional start code.
@end table
@anchor{concat}
@section concat
@ -100,11 +63,11 @@ following directive is recognized:
Path to a file to read; special characters and spaces must be escaped with
backslash or single quotes.
All subsequent file-related directives apply to that file.
All subsequent directives apply to that file.
@item @code{ffconcat version 1.0}
Identify the script type and version. It also sets the @option{safe} option
to 1 if it was -1.
to 1 if it was to its default -1.
To make FFmpeg recognize the format automatically, this directive must
appears exactly as is (no extra space or byte-order-mark) on the very first
@ -118,63 +81,6 @@ file is not available or accurate.
If the duration is set for all files, then it is possible to seek in the
whole concatenated video.
@item @code{inpoint @var{timestamp}}
In point of the file. When the demuxer opens the file it instantly seeks to the
specified timestamp. Seeking is done so that all streams can be presented
successfully at In point.
This directive works best with intra frame codecs, because for non-intra frame
ones you will usually get extra packets before the actual In point and the
decoded content will most likely contain frames before In point too.
For each file, packets before the file In point will have timestamps less than
the calculated start timestamp of the file (negative in case of the first
file), and the duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified In point.
Because of potential packets before the specified In point, packet timestamps
may overlap between two concatenated files.
@item @code{outpoint @var{timestamp}}
Out point of the file. When the demuxer reaches the specified decoding
timestamp in any of the streams, it handles it as an end of file condition and
skips the current and all the remaining packets from all streams.
Out point is exclusive, which means that the demuxer will not output packets
with a decoding timestamp greater or equal to Out point.
This directive works best with intra frame codecs and formats where all streams
are tightly interleaved. For non-intra frame codecs you will usually get
additional packets with presentation timestamp after Out point therefore the
decoded content will most likely contain frames after Out point too. If your
streams are not tightly interleaved you may not get all the packets from all
streams before Out point and you may only will be able to decode the earliest
stream until Out point.
The duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified Out point.
@item @code{file_packet_metadata @var{key=value}}
Metadata of the packets of the file. The specified metadata will be set for
each file packet. You can specify this directive multiple times to add multiple
metadata entries.
@item @code{stream}
Introduce a stream in the virtual file.
All subsequent stream-related directives apply to the last introduced
stream.
Some streams properties must be set in order to allow identifying the
matching streams in the subfiles.
If no streams are defined in the script, the streams from the first file are
copied.
@item @code{exact_stream_id @var{id}}
Set the id of the stream.
If this directive is given, the string with the corresponding id in the
subfiles will be used.
This is especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.
@end table
@subsection Options
@ -192,66 +98,9 @@ component.
If set to 0, any file name is accepted.
The default is 1.
-1 is equivalent to 1 if the format was automatically
The default is -1, it is equivalent to 1 if the format was automatically
probed and 0 otherwise.
@item auto_convert
If set to 1, try to perform automatic conversions on packet data to make the
streams concatenable.
The default is 1.
Currently, the only conversion is adding the h264_mp4toannexb bitstream
filter to H.264 streams in MP4 format. This is necessary in particular if
there are resolution changes.
@item segment_time_metadata
If set to 1, every packet will contain the @var{lavf.concat.start_time} and the
@var{lavf.concat.duration} packet metadata values which are the start_time and
the duration of the respective file segments in the concatenated output
expressed in microseconds. The duration metadata is only set if it is known
based on the concat file.
The default is 0.
@end table
@subsection Examples
@itemize
@item
Use absolute filenames and include some comments:
@example
# my first filename
file /mnt/share/file-1.wav
# my second filename including whitespace
file '/mnt/share/file 2.wav'
# my third filename including whitespace plus single quote
file '/mnt/share/file 3'\''.wav'
@end example
@item
Allow for input format auto-probing, use safe filenames and set the duration of
the first file:
@example
ffconcat version 1.0
file file-1.wav
duration 20.0
file subdir/file-2.wav
@end example
@end itemize
@section flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams.
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@end table
@section libgme
@ -267,71 +116,17 @@ track. Track indexes start at 0. The demuxer exports the number of tracks as
For very large files, the @option{max_size} option may have to be adjusted.
@section libopenmpt
@section libquvi
libopenmpt based module demuxer
Play media from Internet services using the quvi project.
See @url{https://lib.openmpt.org/libopenmpt/} for more information.
The demuxer accepts a @option{format} option to request a specific quality. It
is by default set to @var{best}.
Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
option.
See @url{http://quvi.sourceforge.net/} for more information.
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 gif
Animated GIF demuxer.
It accepts the following options:
@table @option
@item min_delay
Set the minimum valid delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 2.
@item max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds.
Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
the maximum value allowed by the specification.
@item default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
@item ignore_loop
GIF files can contain information to loop a certain number of times (or
infinitely). If @option{ignore_loop} is set to 1, then the loop setting
from the input will be ignored and looping will not occur. If set to 0,
then looping will occur and will cycle the number of times according to
the GIF. Default value is 1.
@end table
For example, with the overlay filter, place an infinitely looping GIF
over another video:
@example
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
Note that in the above example the shortest option for overlay filter is
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.
FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
enabled.
@section image2
@ -358,10 +153,6 @@ Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item none
Disable pattern matching, therefore the video will only contain the specified
image. You should use this option if you do not want to create sequences from
multiple images and your filenames may contain special pattern characters.
@item sequence
Select a sequence pattern type, used to specify a sequence of files
indexed by sequential numbers.
@ -436,8 +227,6 @@ is 5.
If set to 1, will set frame timestamp to modification time of image file. Note
that monotonity of timestamps is not provided: images go in the same order as
without this option. Default value is 0.
If set to 2, will set frame timestamp to the modification time of the image file in
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.
@ -451,86 +240,28 @@ Use @command{ffmpeg} for creating a video from the images in the file
sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
input frame rate of 10 frames per second:
@example
ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
@end example
@item
As above, but start by reading from a file with index 100 in the sequence:
@example
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
@end example
@item
Read images matching the "*.png" glob pattern , that is all the files
terminating with the ".png" suffix:
@example
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
@end example
@end itemize
@section mov/mp4/3gp/QuickTime
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
@table @option
@item enable_drefs
Enable loading of external tracks, disabled by default.
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.
@end table
@section mpegts
MPEG-2 transport stream demuxer.
This demuxer accepts the following options:
@table @option
@item resync_size
Set size limit for looking up a new synchronization. Default value is
65536.
@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
not discarded. Default value is 1, set this option to 0 if you want your
teletext packet PTS and DTS values untouched.
@item ts_packetsize
Output option carrying the raw packet size in bytes.
Show the detected raw packet size, cannot be set by the user.
@item scan_all_pmts
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.
@end table
@section mpjpeg
MJPEG encapsulated in multi-part MIME demuxer.
This demuxer allows reading of MJPEG, where each frame is represented as a part of
multipart/x-mixed-replace stream.
@table @option
@item strict_mime_boundary
Default implementation applies a relaxed standard to multi-part MIME boundary detection,
to prevent regression with numerous existing endpoints not generating a proper MIME
MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check
of the boundary value.
@end table
@section rawvideo
Raw video demuxer.
This demuxer allows one to read raw video data. Since there is no header
This demuxer allows to read raw video data. Since there is no header
specifying the assumed video parameters, the user must specify them
in order to be able to decode the data correctly.

View File

@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Developer Documentation
@titlepage
@ -12,30 +11,36 @@
@chapter Developers Guide
@section Notes for external developers
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
it.
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in @file{doc/examples} and in the source code to
see how the public API is employed.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{doc/examples/muxing.c} to use it to generate audio or video
streams.
You can use the FFmpeg libraries in your commercial program, but you
are encouraged to @emph{publish any patch you make}. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
@end itemize
For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@section Contributing
There are 3 ways by which code gets into FFmpeg.
There are 3 ways by which code gets into ffmpeg.
@itemize @bullet
@item Submitting patches to the main developer mailing list.
See @ref{Submitting patches} for details.
@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.
@ -52,22 +57,16 @@ and should try to fix issues their commit causes.
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
Indent size is 4.
@item
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@item
K&R coding style is used.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
@ -96,7 +95,7 @@ for markup commands, i.e. use @code{@@param} and not @code{\param}.
* more text ...
* ...
*/
typedef struct Foobar @{
typedef struct Foobar@{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
@ -118,19 +117,15 @@ int myfunc(int my_parameter)
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
These features are supported by all compilers we care about, so we will not
@ -140,17 +135,13 @@ clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@ -159,28 +150,23 @@ GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in CamelCase.
for example structs and enums; they should always be in the CamelCase
There are the following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For file-scope variables and functions declared as @code{static}, no prefix
is required.
@item
For variables and functions visible outside of file scope, but only used
internally by a library, an @code{ff_} prefix should be used,
e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_aac_parse_header}.
@item
Each library has its own prefix for public symbols, in addition to the
commonly used @code{av_} (@code{avformat_} for libavformat,
@ -200,12 +186,10 @@ are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@subsection Miscellaneous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@ -231,7 +215,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
@example
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
@ -242,196 +226,141 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
)
)
(setq c-default-style "ffmpeg")
@end lisp
@end example
@section Development Policy
@enumerate
@item
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
including an "or any later version" clause, or, if you prefer
a gift-style license, the
@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
@uref{http://mit-license.org/, MIT} license.
@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
including an "or any later version" clause, or, if you prefer
a gift-style license, the
@uref{http://www.isc.org/software/license/, ISC} or
@uref{http://mit-license.org/, MIT} license.
@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
@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.
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
Do not 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
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
@example
area changed: Short 1 line description
details describing what and why and giving references.
@end example
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
area changed: Short 1 line description
details describing what and why and giving references.
@item
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.
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
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.
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
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.
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.
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.
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
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help in
finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
@section Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it's rarely good to start with that as initial assumption.
Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
Try to help other team members and cooperate if you can.
The goal of software development is to create technical excellence, not for any
individual to be better and "win" against the others. Large software projects
are only possible and successful through teamwork.
If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@section Submitting patches
@ -439,7 +368,7 @@ First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-).
@code{git send-email}. We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
@ -462,7 +391,7 @@ Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Patches should be posted to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
@ -482,51 +411,40 @@ send a reminder by email. Your patch should eventually be dealt with.
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the AVCodecID to @file{avcodec.h}?
When adding new codec IDs, also add an entry to the codec descriptor
list in @file{libavcodec/codec_desc.c}.
Did you add the AVCodecID to @file{avcodec.h}?
When adding new codec IDs, also add an entry to the codec descriptor
list in @file{libavcodec/codec_desc.c}.
@item
If it has a FourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
If it has a FourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you @code{git add} the appropriate files before committing?
Did you @code{git add} the appropriate files before committing?
@item
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate
@ -534,113 +452,82 @@ Did you make sure it compiles standalone, i.e. with
@enumerate
@item
Does @code{make fate} pass with the patch applied?
Does @code{make fate} pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
Was the patch generated with git format-patch or send-email?
@item
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
Did you provide a clear git commit log message?
@item
Is the patch against latest FFmpeg git master branch?
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
@item
Did you test your decoder or demuxer against sample files?
Samples may be obtained at @url{https://samples.ffmpeg.org}.
Does the patch not mix functional and cosmetic changes?
@item
Does the patch not mix functional and cosmetic changes?
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
Is the patch attached to the email you send?
@item
Is the patch attached to the email you send?
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide 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 ftp://upload.ffmpeg.org
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org.
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose summary about what the patch does change?
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose explanation why it changes things like it does?
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
Consider to add a regression test for your code.
@item
Consider adding a regression test for your code.
If you added YASM code please check that things still work with --disable-yasm
@item
If you added YASM code please check that things still work with --disable-yasm.
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@item
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@item
Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
@end enumerate
@section Patch review process
@ -669,10 +556,6 @@ not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
Everyone is welcome to review patches. Also if you are waiting for your patch
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}
@section Regression tests
@ -688,14 +571,15 @@ accordingly].
@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.
specific test then the media has to be inlcuded in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
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
message or introductionary message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@subsection Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
@ -706,15 +590,12 @@ the following steps:
@item
Configure to compile with instrumentation enabled:
@code{configure --toolchain=gcov}.
@item
Run your test case, either manually or via FATE. This can be either
the full FATE regression suite, or any arbitrary invocation of any
front-end tool provided by FFmpeg, in any combination.
@item
Run @code{make lcov} to generate coverage data in HTML format.
@item
View @code{lcov/index.html} in your preferred HTML viewer.
@end enumerate
@ -743,19 +624,18 @@ FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a @strong{release
manager} prepares, tests and publishes tarballs on the
@url{https://ffmpeg.org} website.
@url{http://ffmpeg.org} website.
There are two kinds of releases:
@enumerate
@item
@strong{Major releases} always include the latest and greatest
features and functionality.
@strong{Major releases} always include the latest and greatest
features and functionality.
@item
@strong{Point releases} are cut from @strong{release} branches,
which are named @code{release/X}, with @code{X} being the release
version number.
@strong{Point releases} are cut from @strong{release} branches,
which are named @code{release/X}, with @code{X} being the release
version number.
@end enumerate
Note that we promise to our users that shared libraries from any FFmpeg
@ -776,18 +656,15 @@ inclusion into a point release:
@enumerate
@item
Fixes a security issue, preferably identified by a @strong{CVE
number} issued by @url{http://cve.mitre.org/}.
Fixes a security issue, preferably identified by a @strong{CVE
number} issued by @url{http://cve.mitre.org/}.
@item
Fixes a documented bug in @url{https://trac.ffmpeg.org}.
Fixes a documented bug in @url{https://trac.ffmpeg.org}.
@item
Improves the included documentation.
Improves the included documentation.
@item
Retains both source code and binary compatibility with previous
point releases of the same release branch.
Retains both source code and binary compatibility with previous
point releases of the same release branch.
@end enumerate
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@ -799,42 +676,33 @@ The release process involves the following steps:
@enumerate
@item
Ensure that the @file{RELEASE} file contains the version number for
the upcoming release.
Ensure that the @file{RELEASE} file contains the version number for
the upcoming release.
@item
Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
@item
Announce the intent to do a release to the mailing list.
Announce the intent to do a release to the mailing list.
@item
Make sure all relevant security fixes have been backported. See
@url{https://ffmpeg.org/security.html}.
Make sure all relevant security fixes have been backported. See
@url{https://ffmpeg.org/security.html}.
@item
Ensure that the FATE regression suite still passes in the release
branch on at least @strong{i386} and @strong{amd64}
(cf. @ref{Regression tests}).
Ensure that the FATE regression suite still passes in the release
branch on at least @strong{i386} and @strong{amd64}
(cf. @ref{Regression tests}).
@item
Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
supplementing files that contain @code{gpg} signatures
Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
supplementing files that contain @code{gpg} signatures
@item
Publish the tarballs at @url{https://ffmpeg.org/releases}. Create and
push an annotated tag in the form @code{nX}, with @code{X}
containing the version number.
Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
push an annotated tag in the form @code{nX}, with @code{X}
containing the version number.
@item
Propose and send a patch to the @strong{ffmpeg-devel} mailing list
with a news entry for the website.
Propose and send a patch to the @strong{ffmpeg-devel} mailing list
with a news entry for the website.
@item
Publish the news entry.
Publish the news entry.
@item
Send an announcement to the mailing list.
Send announcement to the mailing list.
@end enumerate
@bye

View File

@ -17,9 +17,5 @@ for programmatic use.
@c man end DEVICE OPTIONS
@ifclear config-writeonly
@include indevs.texi
@end ifclear
@ifclear config-readonly
@include outdevs.texi
@end ifclear

View File

@ -1,21 +1,14 @@
#!/bin/sh
OUT_DIR="${1}"
SRC_PATH="${1}"
DOXYFILE="${2}"
DOXYGEN="${3}"
shift 3
shift 2
if [ -e "VERSION" ]; then
VERSION=`cat "VERSION"`
else
VERSION=`git describe`
fi
$DOXYGEN - <<EOF
doxygen - <<EOF
@INCLUDE = ${DOXYFILE}
INPUT = $@
HTML_TIMESTAMP = NO
PROJECT_NUMBER = $VERSION
OUTPUT_DIRECTORY = $OUT_DIR
HTML_HEADER = ${SRC_PATH}/doc/doxy/header.html
HTML_FOOTER = ${SRC_PATH}/doc/doxy/footer.html
HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
EOF

1
doc/doxy/.gitignore vendored
View File

@ -1 +0,0 @@
/html/

2019
doc/doxy/doxy_stylesheet.css Normal file

File diff suppressed because it is too large Load Diff

9
doc/doxy/footer.html Normal file
View File

@ -0,0 +1,9 @@
<footer class="footer pagination-right">
<span class="label label-info">
Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
</span>
</footer>
</div>
</body>
</html>

16
doc/doxy/header.html Normal file
View File

@ -0,0 +1,16 @@
<!DOCTYPE html>
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
<meta http-equiv="X-UA-Compatible" content="IE=9"/>
<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
<!--Header replace -->
</head>
<div class="container">
<!--Header replace -->
<div class="menu">

File diff suppressed because it is too large Load Diff

View File

@ -76,7 +76,7 @@ EMFILE POSIX ++++++ Too many open files
EMLINK POSIX ++++++ Too many links
EMSGSIZE POSIX +++..+ Message too long
EMULTIHOP POSIX ++4... Multihop attempted
ENAMETOOLONG POSIX - ++++++ File name too long
ENAMETOOLONG POSIX - ++++++ Filen ame too long
ENAVAIL +..... No XENIX semaphores available
ENEEDAUTH .++... Need authenticator
ENETDOWN POSIX +++..+ Network is down

View File

@ -1,16 +0,0 @@
/avio_dir_cmd
/avio_reading
/decoding_encoding
/demuxing_decoding
/extract_mvs
/filter_audio
/filtering_audio
/filtering_video
/metadata
/muxing
/pc-uninstalled
/remuxing
/resampling_audio
/scaling_video
/transcode_aac
/transcoding

View File

@ -11,29 +11,20 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_dir_cmd \
avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \
EXAMPLES= decoding_encoding \
demuxing \
filtering_video \
filtering_audio \
http_multiclient \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
.phony: all clean-test clean

View File

@ -5,19 +5,14 @@ Both following use cases rely on pkg-config and make, thus make sure
that you have them installed and working on your system.
Method 1: build the installed examples in a generic read/write user directory
1) Build the installed examples in a generic read/write user directory
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
Method 2: build the examples in-tree
2) Build the examples in-tree
Assuming you are in the source FFmpeg checkout directory, you need to build
FFmpeg (no need to make install in any prefix). Then just run "make examples".
This will build the examples using the FFmpeg build system. You can clean those
examples using "make examplesclean"
If you want to try the dedicated Makefile examples (to emulate the first
method), go into doc/examples and run a command such as
PKG_CONFIG_PATH=pc-uninstalled make.
FFmpeg (no need to make install in any prefix). Then you can go into the
doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

View File

@ -1,180 +0,0 @@
/*
* Copyright (c) 2014 Lukasz Marek
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
static const char *type_string(int type)
{
switch (type) {
case AVIO_ENTRY_DIRECTORY:
return "<DIR>";
case AVIO_ENTRY_FILE:
return "<FILE>";
case AVIO_ENTRY_BLOCK_DEVICE:
return "<BLOCK DEVICE>";
case AVIO_ENTRY_CHARACTER_DEVICE:
return "<CHARACTER DEVICE>";
case AVIO_ENTRY_NAMED_PIPE:
return "<PIPE>";
case AVIO_ENTRY_SYMBOLIC_LINK:
return "<LINK>";
case AVIO_ENTRY_SOCKET:
return "<SOCKET>";
case AVIO_ENTRY_SERVER:
return "<SERVER>";
case AVIO_ENTRY_SHARE:
return "<SHARE>";
case AVIO_ENTRY_WORKGROUP:
return "<WORKGROUP>";
case AVIO_ENTRY_UNKNOWN:
default:
break;
}
return "<UNKNOWN>";
}
static int list_op(const char *input_dir)
{
AVIODirEntry *entry = NULL;
AVIODirContext *ctx = NULL;
int cnt, ret;
char filemode[4], uid_and_gid[20];
if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
goto fail;
}
cnt = 0;
for (;;) {
if ((ret = avio_read_dir(ctx, &entry)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
goto fail;
}
if (!entry)
break;
if (entry->filemode == -1) {
snprintf(filemode, 4, "???");
} else {
snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
}
snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
if (cnt == 0)
av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
"TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
"ACCESSED", "STATUS_CHANGED");
av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
type_string(entry->type),
entry->size,
entry->name,
uid_and_gid,
filemode,
entry->modification_timestamp,
entry->access_timestamp,
entry->status_change_timestamp);
avio_free_directory_entry(&entry);
cnt++;
};
fail:
avio_close_dir(&ctx);
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 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);
if (argc < 2) {
usage(argv[0]);
return 1;
}
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
avformat_network_init();
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();
return ret < 0 ? 1 : 0;
}

View File

@ -1,134 +0,0 @@
/*
* Copyright (c) 2014 Stefano Sabatini
*
* 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
* libavformat AVIOContext API example.
*
* Make libavformat demuxer access media content through a custom
* AVIOContext read callback.
* @example avio_reading.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
struct buffer_data {
uint8_t *ptr;
size_t size; ///< size left in the buffer
};
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);
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
memcpy(buf, bd->ptr, buf_size);
bd->ptr += buf_size;
bd->size -= buf_size;
return buf_size;
}
int main(int argc, char *argv[])
{
AVFormatContext *fmt_ctx = NULL;
AVIOContext *avio_ctx = NULL;
uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
size_t buffer_size, avio_ctx_buffer_size = 4096;
char *input_filename = NULL;
int ret = 0;
struct buffer_data bd = { 0 };
if (argc != 2) {
fprintf(stderr, "usage: %s input_file\n"
"API example program to show how to read from a custom buffer "
"accessed through AVIOContext.\n", argv[0]);
return 1;
}
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)
goto end;
/* fill opaque structure used by the AVIOContext read callback */
bd.ptr = buffer;
bd.size = buffer_size;
if (!(fmt_ctx = avformat_alloc_context())) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
if (!avio_ctx_buffer) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
0, &bd, &read_packet, NULL, NULL);
if (!avio_ctx) {
ret = AVERROR(ENOMEM);
goto end;
}
fmt_ctx->pb = avio_ctx;
ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open input\n");
goto end;
}
ret = avformat_find_stream_info(fmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Could not find stream information\n");
goto end;
}
av_dump_format(fmt_ctx, 0, input_filename, 0);
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

View File

@ -24,10 +24,10 @@
* @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
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
* @example doc/examples/decoding_encoding.c
*/
#include <math.h>
@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
}
/* frame containing input raw audio */
frame = av_frame_alloc();
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
@ -170,10 +170,6 @@ static void audio_encode_example(const char *filename)
* 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",
@ -191,7 +187,7 @@ static void audio_encode_example(const char *filename)
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
for(i=0;i<200;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
@ -211,7 +207,7 @@ static void audio_encode_example(const char *filename)
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
@ -225,13 +221,13 @@ static void audio_encode_example(const char *filename)
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_free_frame(&frame);
avcodec_close(c);
av_free(c);
}
@ -245,7 +241,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
@ -253,7 +249,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
printf("Decode audio file %s to %s\n", filename, outfilename);
/* find the MPEG audio decoder */
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
@ -288,15 +284,15 @@ static void audio_decode_example(const char *outfilename, const char *filename)
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())) {
if (!(decoded_frame = avcodec_alloc_frame())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
}
} else
avcodec_get_frame_defaults(decoded_frame);
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
@ -305,15 +301,10 @@ static void audio_decode_example(const char *outfilename, const char *filename)
}
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);
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
@ -338,7 +329,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
avcodec_close(c);
av_free(c);
av_frame_free(&decoded_frame);
avcodec_free_frame(&decoded_frame);
}
/*
@ -356,7 +347,7 @@ static void video_encode_example(const char *filename, int codec_id)
printf("Encode video file %s\n", filename);
/* find the video encoder */
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
@ -375,18 +366,12 @@ static void video_encode_example(const char *filename, int codec_id)
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->time_base= (AVRational){1,25};
c->gop_size = 10; /* emit one intra frame every ten frames */
c->max_b_frames=1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
if(codec_id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
@ -401,7 +386,7 @@ static void video_encode_example(const char *filename, int codec_id)
exit(1);
}
frame = av_frame_alloc();
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
@ -420,7 +405,7 @@ static void video_encode_example(const char *filename, int codec_id)
}
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
for(i=0;i<25;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
@ -428,15 +413,15 @@ static void video_encode_example(const char *filename, int codec_id)
fflush(stdout);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
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++) {
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;
}
@ -454,7 +439,7 @@ static void video_encode_example(const char *filename, int codec_id)
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
@ -471,18 +456,18 @@ static void video_encode_example(const char *filename, int codec_id)
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_packet_unref(&pkt);
av_free_packet(&pkt);
}
}
/* add sequence end code to have a real MPEG file */
/* 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);
avcodec_free_frame(&frame);
printf("\n");
}
@ -496,10 +481,10 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
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);
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);
}
@ -521,7 +506,7 @@ static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
/* 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);
avctx->width, avctx->height, buf);
(*frame_count)++;
}
if (pkt->data) {
@ -538,17 +523,17 @@ static void video_decode_example(const char *outfilename, const char *filename)
int frame_count;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t inbuf[INBUF_SIZE + FF_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);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Decode video file %s to %s\n", filename, outfilename);
/* find the MPEG-1 video decoder */
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
@ -561,8 +546,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
exit(1);
}
if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= 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
@ -580,14 +565,14 @@ static void video_decode_example(const char *outfilename, const char *filename)
exit(1);
}
frame = av_frame_alloc();
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame_count = 0;
for (;;) {
for(;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
@ -613,9 +598,9 @@ static void video_decode_example(const char *outfilename, const char *filename)
exit(1);
}
/* Some codecs, such as MPEG, transmit the I- and P-frame with a
/* 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. */
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);
@ -624,7 +609,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
avcodec_close(c);
av_free(c);
av_frame_free(&frame);
avcodec_free_frame(&frame);
printf("\n");
}
@ -641,7 +626,7 @@ int main(int argc, char **argv)
"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",
"output_type must be choosen between 'h264', 'mp2', 'mpg'.\n",
argv[0]);
return 1;
}
@ -651,7 +636,7 @@ int main(int argc, char **argv)
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");
audio_decode_example("test.sw", "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");

View File

@ -22,11 +22,11 @@
/**
* @file
* Demuxing and decoding example.
* libavformat demuxing API use example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example demuxing_decoding.c
* @example doc/examples/demuxing.c
*/
#include <libavutil/imgutils.h>
@ -36,8 +36,6 @@
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static int width, height;
static enum AVPixelFormat pix_fmt;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
@ -49,50 +47,29 @@ static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static uint8_t **audio_dst_data = NULL;
static int audio_dst_linesize;
static int audio_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
/* 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;
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
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));
fprintf(stderr, "Error decoding video frame\n");
return ret;
}
if (*got_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;
}
printf("video_frame%s n:%d coded_n:%d pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number,
@ -102,7 +79,7 @@ static int decode_packet(int *got_frame, int cached)
* 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);
video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
@ -111,50 +88,49 @@ static int decode_packet(int *got_frame, int cached)
/* 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));
fprintf(stderr, "Error decoding audio frame\n");
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
frame->nb_samples, frame->format, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate audio buffer\n");
return AVERROR(ENOMEM);
}
/* TODO: extend return code of the av_samples_* functions so that this call is not needed */
audio_dst_bufsize =
av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
frame->nb_samples, frame->format, 1);
/* copy audio data to destination buffer:
* this is required since rawaudio expects non aligned data */
av_samples_copy(audio_dst_data, frame->data, 0, 0,
frame->nb_samples, av_frame_get_channels(frame), frame->format);
/* write to rawaudio file */
fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
av_freep(&audio_dst_data[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;
return ret;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret, stream_index;
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
@ -162,8 +138,8 @@ static int open_codec_context(int *stream_idx,
av_get_media_type_string(type), src_filename);
return ret;
} else {
stream_index = ret;
st = fmt_ctx->streams[stream_index];
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
@ -171,17 +147,14 @@ static int open_codec_context(int *stream_idx,
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
return ret;
}
/* 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) {
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
*stream_idx = stream_index;
}
return 0;
@ -220,22 +193,15 @@ int main (int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
if (argc != 4) {
fprintf(stderr, "usage: %s 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\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"
"audio frames to a rawaudio file named audio_output_file.\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];
@ -267,11 +233,9 @@ int main (int argc, char **argv)
}
/* allocate image where the decoded image will be put */
width = video_dec_ctx->width;
height = video_dec_ctx->height;
pix_fmt = video_dec_ctx->pix_fmt;
ret = av_image_alloc(video_dst_data, video_dst_linesize,
width, height, pix_fmt, 1);
video_dec_ctx->width, video_dec_ctx->height,
video_dec_ctx->pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
@ -280,14 +244,25 @@ int main (int argc, char **argv)
}
if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
int nb_planes;
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dec_ctx = audio_stream->codec;
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
audio_dec_ctx->channels : 1;
audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
if (!audio_dst_data) {
fprintf(stderr, "Could not allocate audio data buffers\n");
ret = AVERROR(ENOMEM);
goto end;
}
}
/* dump input information to stderr */
@ -299,7 +274,7 @@ int main (int argc, char **argv)
goto end;
}
frame = av_frame_alloc();
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
@ -318,15 +293,8 @@ int main (int argc, char **argv)
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_packet_unref(&orig_pkt);
decode_packet(&got_frame, 0);
av_free_packet(&pkt);
}
/* flush cached frames */
@ -341,43 +309,34 @@ int main (int argc, char **argv)
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(pix_fmt), width, height,
av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_close(video_dec_ctx);
avcodec_close(audio_dec_ctx);
if (video_dec_ctx)
avcodec_close(video_dec_ctx);
if (audio_dec_ctx)
avcodec_close(audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_frame_free(&frame);
av_free(frame);
av_free(video_dst_data[0]);
av_free(audio_dst_data);
return ret < 0;
}

View File

@ -1,185 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
* Copyright (c) 2014 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavutil/motion_vector.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL;
static AVStream *video_stream = NULL;
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(int *got_frame, int cached)
{
int decoded = pkt.size;
*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 (*got_frame) {
int i;
AVFrameSideData *sd;
video_frame_count++;
sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
if (sd) {
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags);
}
}
}
}
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
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 {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* 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);
}
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
exit(1);
}
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);
}
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
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);
if (!video_stream) {
fprintf(stderr, "Could not find video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
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) {
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 */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
end:
avcodec_close(video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
return ret < 0;
}

View File

@ -1,365 +0,0 @@
/*
* copyright (c) 2013 Andrew Kelley
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* libavfilter API usage example.
*
* @example filter_audio.c
* This example will generate a sine wave audio,
* pass it through a simple filter chain, and then compute the MD5 checksum of
* the output data.
*
* The filter chain it uses is:
* (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
*
* abuffer: This provides the endpoint where you can feed the decoded samples.
* volume: In this example we hardcode it to 0.90.
* aformat: This converts the samples to the samplefreq, channel layout,
* and sample format required by the audio device.
* abuffersink: This provides the endpoint where you can read the samples after
* they have passed through the filter chain.
*/
#include <inttypes.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"
#define INPUT_SAMPLERATE 48000
#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
AVFilterContext **sink)
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
AVFilter *abuffer;
AVFilterContext *volume_ctx;
AVFilter *volume;
AVFilterContext *aformat_ctx;
AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
uint8_t ch_layout[64];
int err;
/* Create a new filtergraph, which will contain all the filters. */
filter_graph = avfilter_graph_alloc();
if (!filter_graph) {
fprintf(stderr, "Unable to create filter graph.\n");
return AVERROR(ENOMEM);
}
/* Create the abuffer filter;
* it will be used for feeding the data into the graph. */
abuffer = avfilter_get_by_name("abuffer");
if (!abuffer) {
fprintf(stderr, "Could not find the abuffer filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
if (!abuffer_ctx) {
fprintf(stderr, "Could not allocate the abuffer instance.\n");
return AVERROR(ENOMEM);
}
/* Set the filter options through the AVOptions API. */
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", INPUT_SAMPLERATE, AV_OPT_SEARCH_CHILDREN);
/* Now initialize the filter; we pass NULL options, since we have already
* set all the options above. */
err = avfilter_init_str(abuffer_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffer filter.\n");
return err;
}
/* Create volume filter. */
volume = avfilter_get_by_name("volume");
if (!volume) {
fprintf(stderr, "Could not find the volume filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
if (!volume_ctx) {
fprintf(stderr, "Could not allocate the volume instance.\n");
return AVERROR(ENOMEM);
}
/* A different way of passing the options is as key/value pairs in a
* dictionary. */
av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
err = avfilter_init_dict(volume_ctx, &options_dict);
av_dict_free(&options_dict);
if (err < 0) {
fprintf(stderr, "Could not initialize the volume filter.\n");
return err;
}
/* Create the aformat filter;
* it ensures that the output is of the format we want. */
aformat = avfilter_get_by_name("aformat");
if (!aformat) {
fprintf(stderr, "Could not find the aformat filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
if (!aformat_ctx) {
fprintf(stderr, "Could not allocate the aformat instance.\n");
return AVERROR(ENOMEM);
}
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
(uint64_t)AV_CH_LAYOUT_STEREO);
err = avfilter_init_str(aformat_ctx, options_str);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
return err;
}
/* Finally create the abuffersink filter;
* it will be used to get the filtered data out of the graph. */
abuffersink = avfilter_get_by_name("abuffersink");
if (!abuffersink) {
fprintf(stderr, "Could not find the abuffersink filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
if (!abuffersink_ctx) {
fprintf(stderr, "Could not allocate the abuffersink instance.\n");
return AVERROR(ENOMEM);
}
/* This filter takes no options. */
err = avfilter_init_str(abuffersink_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffersink instance.\n");
return err;
}
/* Connect the filters;
* in this simple case the filters just form a linear chain. */
err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
if (err >= 0)
err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
if (err >= 0)
err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
if (err < 0) {
fprintf(stderr, "Error connecting filters\n");
return err;
}
/* Configure the graph. */
err = avfilter_graph_config(filter_graph, NULL);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
return err;
}
*graph = filter_graph;
*src = abuffer_ctx;
*sink = abuffersink_ctx;
return 0;
}
/* Do something useful with the filtered data: this simple
* example just prints the MD5 checksum of each plane to stdout. */
static int process_output(struct AVMD5 *md5, AVFrame *frame)
{
int planar = av_sample_fmt_is_planar(frame->format);
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
int i, j;
for (i = 0; i < planes; i++) {
uint8_t checksum[16];
av_md5_init(md5);
av_md5_sum(checksum, frame->extended_data[i], plane_size);
fprintf(stdout, "plane %d: 0x", i);
for (j = 0; j < sizeof(checksum); j++)
fprintf(stdout, "%02X", checksum[j]);
fprintf(stdout, "\n");
}
fprintf(stdout, "\n");
return 0;
}
/* Construct a frame of audio data to be filtered;
* this simple example just synthesizes a sine wave. */
static int get_input(AVFrame *frame, int frame_num)
{
int err, i, j;
#define FRAME_SIZE 1024
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;
err = av_frame_get_buffer(frame, 0);
if (err < 0)
return err;
/* Fill the data for each channel. */
for (i = 0; i < 5; i++) {
float *data = (float*)frame->extended_data[i];
for (j = 0; j < frame->nb_samples; j++)
data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
}
return 0;
}
int main(int argc, char *argv[])
{
struct AVMD5 *md5;
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
if (argc < 2) {
fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
return 1;
}
duration = atof(argv[1]);
nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
if (nb_frames <= 0) {
fprintf(stderr, "Invalid duration: %s\n", argv[1]);
return 1;
}
avfilter_register_all();
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Error allocating the frame\n");
return 1;
}
md5 = av_md5_alloc();
if (!md5) {
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
/* Set up the filtergraph. */
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
fprintf(stderr, "Unable to init filter graph:");
goto fail;
}
/* the main filtering loop */
for (i = 0; i < nb_frames; i++) {
/* get an input frame to be filtered */
err = get_input(frame, i);
if (err < 0) {
fprintf(stderr, "Error generating input frame:");
goto fail;
}
/* Send the frame to the input of the filtergraph. */
err = av_buffersrc_add_frame(src, frame);
if (err < 0) {
av_frame_unref(frame);
fprintf(stderr, "Error submitting the frame to the filtergraph:");
goto fail;
}
/* Get all the filtered output that is available. */
while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
/* now do something with our filtered frame */
err = process_output(md5, frame);
if (err < 0) {
fprintf(stderr, "Error processing the filtered frame:");
goto fail;
}
av_frame_unref(frame);
}
if (err == AVERROR(EAGAIN)) {
/* Need to feed more frames in. */
continue;
} else if (err == AVERROR_EOF) {
/* Nothing more to do, finish. */
break;
} else if (err < 0) {
/* An error occurred. */
fprintf(stderr, "Error filtering the data:");
goto fail;
}
}
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
return 0;
fail:
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}

View File

@ -25,7 +25,7 @@
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
* @example doc/examples/filtering_audio.c
*/
#include <unistd.h>
@ -33,12 +33,13 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
@ -65,7 +66,7 @@ static int open_input_file(const char *filename)
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
@ -84,22 +85,18 @@ static int open_input_file(const char *filename)
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
int ret;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
@ -112,7 +109,7 @@ static int init_filters(const char *filters_descr)
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
return ret;
}
/* buffer audio sink: to terminate the filter chain. */
@ -120,63 +117,47 @@ static int init_filters(const char *filters_descr)
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
return ret;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
return ret;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
@ -187,11 +168,7 @@ static int init_filters(const char *filters_descr)
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
return 0;
}
static void print_frame(const AVFrame *frame)
@ -211,7 +188,7 @@ static void print_frame(const AVFrame *frame)
int main(int argc, char **argv)
{
int ret;
AVPacket packet0, packet;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
@ -225,6 +202,7 @@ int main(int argc, char **argv)
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
@ -234,24 +212,18 @@ int main(int argc, char **argv)
goto end;
/* read all packets */
packet0.data = NULL;
packet.data = NULL;
while (1) {
if (!packet0.data) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
packet0 = packet;
}
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == audio_stream_index) {
avcodec_get_frame_defaults(frame);
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
packet.size -= ret;
packet.data += ret;
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
@ -263,31 +235,29 @@ int main(int argc, char **argv)
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
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_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
if (dec_ctx)
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}

View File

@ -24,7 +24,7 @@
/**
* @file
* API example for decoding and filtering
* @example filtering_video.c
* @example doc/examples/filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
@ -33,14 +33,11 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24,transpose=cclock";
/* other way:
scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
*/
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
@ -73,7 +70,6 @@ static int open_input_file(const char *filename)
}
video_stream_index = ret;
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) {
@ -87,71 +83,47 @@ static int open_input_file(const char *filename)
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
int ret;
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;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
AVBufferSinkParams *buffersink_params;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
return ret;
}
/* buffer video sink: to terminate the filter chain. */
buffersink_params = av_buffersink_params_alloc();
buffersink_params->pixel_fmts = pix_fmts;
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
NULL, buffersink_params, filter_graph);
av_free(buffersink_params);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
@ -159,16 +131,11 @@ static int init_filters(const char *filters_descr)
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
return ret;
return 0;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
@ -219,6 +186,7 @@ int main(int argc, char **argv)
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
@ -233,6 +201,7 @@ int main(int argc, char **argv)
break;
if (packet.stream_index == video_stream_index) {
avcodec_get_frame_defaults(frame);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
@ -259,20 +228,22 @@ int main(int argc, char **argv)
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_packet_unref(&packet);
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
if (dec_ctx)
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}

View File

@ -1,155 +0,0 @@
/*
* Copyright (c) 2015 Stephan Holljes
*
* 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
* libavformat multi-client network API usage example.
*
* @example http_multiclient.c
* This example will serve a file without decoding or demuxing it over http.
* Multiple clients can connect and will receive the same file.
*/
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <unistd.h>
void process_client(AVIOContext *client, const char *in_uri)
{
AVIOContext *input = NULL;
uint8_t buf[1024];
int ret, n, reply_code;
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;
}
if (ret < 0)
goto end;
av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
reply_code = 200;
} else {
reply_code = AVERROR_HTTP_NOT_FOUND;
}
if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
goto end;
}
av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
while ((ret = avio_handshake(client)) > 0);
if (ret < 0)
goto end;
fprintf(stderr, "Handshake performed.\n");
if (reply_code != 200)
goto end;
fprintf(stderr, "Opening input file.\n");
if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
av_err2str(ret));
goto end;
}
for(;;) {
n = avio_read(input, buf, sizeof(buf));
if (n < 0) {
if (n == AVERROR_EOF)
break;
av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
av_err2str(n));
break;
}
avio_write(client, buf, n);
avio_flush(client);
}
end:
fprintf(stderr, "Flushing client\n");
avio_flush(client);
fprintf(stderr, "Closing client\n");
avio_close(client);
fprintf(stderr, "Closing input\n");
avio_close(input);
}
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;
if (argc < 3) {
printf("usage: %s input http://hostname[:port]\n"
"API example program to serve http to multiple clients.\n"
"\n", argv[0]);
return 1;
}
in_uri = argv[1];
out_uri = argv[2];
av_register_all();
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
return ret;
}
if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
return ret;
}
fprintf(stderr, "Entering main loop.\n");
for(;;) {
if ((ret = avio_accept(server, &client)) < 0)
goto end;
fprintf(stderr, "Accepted client, forking process.\n");
// XXX: Since we don't reap our children and don't ignore signals
// this produces zombie processes.
pid = fork();
if (pid < 0) {
perror("Fork failed");
ret = AVERROR(errno);
goto end;
}
if (pid == 0) {
fprintf(stderr, "In child.\n");
process_client(client, in_uri);
avio_close(server);
exit(0);
}
if (pid > 0)
avio_close(client);
}
end:
avio_close(server);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

View File

@ -23,7 +23,7 @@
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example metadata.c
* @example doc/examples/metadata.c
*/
#include <stdio.h>

View File

@ -24,9 +24,9 @@
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format. The default
* codecs are used.
* @example muxing.c
* Output a media file in any supported libavformat format.
* The default codecs are used.
* @example doc/examples/muxing.c
*/
#include <stdlib.h>
@ -34,68 +34,26 @@
#include <string.h>
#include <math.h>
#include <libavutil/avassert.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#define STREAM_DURATION 10.0
/* 5 seconds stream duration */
#define STREAM_DURATION 200.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
AVCodecContext *enc;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* 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;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
static int sws_flags = SWS_BICUBIC;
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
AVStream *st;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
@ -105,43 +63,20 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
if (!ost->st) {
st = avformat_new_stream(oc, *codec);
if (!st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = avcodec_alloc_context3(*codec);
if (!c) {
fprintf(stderr, "Could not alloc an encoding context\n");
exit(1);
}
ost->enc = c;
st->id = oc->nb_streams-1;
c = st->codec;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->sample_fmt = AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
c->channels = 2;
break;
case AVMEDIA_TYPE_VIDEO:
@ -155,13 +90,12 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
@ -178,277 +112,238 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
static float t, tincr, tincr2;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
static uint8_t **src_samples_data;
static int src_samples_linesize;
static int src_nb_samples;
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
static int max_dst_nb_samples;
uint8_t **dst_samples_data;
int dst_samples_linesize;
int dst_samples_size;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
struct SwrContext *swr_ctx = NULL;
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
c = st->codec;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
t = 0;
tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
10000 : c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
src_nb_samples, c->sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
fprintf(stderr, "Could not allocate source samples\n");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = src_nb_samples;
ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
max_dst_nb_samples, c->sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
exit(1);
}
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
c->sample_fmt, 0);
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
int16_t *q;
/* 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)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->enc->channels; i++)
q = samples;
for (j = 0; j < frame_size; j++) {
v = (int)(sin(t) * 10000);
for (i = 0; i < nb_channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
t += tincr;
tincr += tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
AVFrame *frame = avcodec_alloc_frame();
int got_packet, ret, dst_nb_samples;
av_init_packet(&pkt);
c = ost->enc;
c = st->codec;
frame = get_audio_frame(ost);
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* 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(ost->frame);
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");
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_samples_data[0]);
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
dst_nb_samples, c->sample_fmt, 0);
if (ret < 0)
exit(1);
}
frame = ost->frame;
max_dst_nb_samples = dst_nb_samples;
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
c->sample_fmt, 0);
}
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
/* convert to destination format */
ret = swr_convert(swr_ctx,
dst_samples_data, dst_nb_samples,
(const uint8_t **)src_samples_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
} else {
dst_samples_data[0] = src_samples_data[0];
dst_nb_samples = src_nb_samples;
}
frame->nb_samples = dst_nb_samples;
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
dst_samples_data[0], dst_samples_size, 0);
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);
}
}
if (!got_packet)
return;
return (frame || got_packet) ? 0 : 1;
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &pkt);
if (ret != 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
avcodec_free_frame(&frame);
}
static void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
av_free(src_samples_data[0]);
av_free(dst_samples_data[0]);
}
/**************************************************************/
/* video output */
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
int ret;
static AVFrame *frame;
static AVPicture src_picture, dst_picture;
static int frame_count;
picture = av_frame_alloc();
if (!picture)
return NULL;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
AVCodecContext *c = st->codec;
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
/* Allocate the encoded raw picture. */
ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
if (ret < 0) {
fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary picture\n");
ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
if (ret < 0) {
fprintf(stderr, "Could not allocate temporary picture: %s\n",
av_err2str(ret));
exit(1);
}
}
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
/* copy data and linesize picture pointers to frame */
*((AVPicture *)frame) = dst_picture;
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
static void fill_yuv_image(AVPicture *pict, int frame_index,
int width, int height)
{
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);
int x, y, i;
i = frame_index;
@ -466,89 +361,85 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->enc;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
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 */
if (!ost->sws_ctx) {
ost->sws_ctx = sws_getContext(c->width, c->height,
AV_PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
SCALE_FLAGS, NULL, NULL, NULL);
if (!ost->sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
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);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
ost->frame->pts = ost->next_pts++;
return ost->frame;
}
/*
* encode one video frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
static void write_video_frame(AVFormatContext *oc, AVStream *st)
{
int ret;
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
AVPacket pkt = { 0 };
static struct SwsContext *sws_ctx;
AVCodecContext *c = st->codec;
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);
if (frame_count >= STREAM_NB_FRAMES) {
/* No more frames to compress. The codec has a latency of a few
* frames if using B-frames, so we get the last frames by
* passing the same picture again. */
} else {
ret = 0;
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 */
if (!sws_ctx) {
sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
c->width, c->height, c->pix_fmt,
sws_flags, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(&src_picture, frame_count, c->width, c->height);
sws_scale(sws_ctx,
(const uint8_t * const *)src_picture.data, src_picture.linesize,
0, c->height, dst_picture.data, dst_picture.linesize);
} else {
fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
}
}
if (ret < 0) {
if (oc->oformat->flags & AVFMT_RAWPICTURE) {
/* Raw video case - directly store the picture in the packet */
AVPacket pkt;
av_init_packet(&pkt);
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = st->index;
pkt.data = dst_picture.data[0];
pkt.size = sizeof(AVPicture);
ret = av_interleaved_write_frame(oc, &pkt);
} else {
AVPacket pkt = { 0 };
int got_packet;
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
/* If size is zero, it means the image was buffered. */
if (!ret && got_packet && pkt.size) {
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &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;
frame_count++;
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
static void close_video(AVFormatContext *oc, AVStream *st)
{
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
avcodec_close(st->codec);
av_free(src_picture.data[0]);
av_free(dst_picture.data[0]);
av_free(frame);
}
/**************************************************************/
@ -556,21 +447,18 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVStream *audio_st, *video_st;
AVCodec *audio_codec, *video_codec;
double audio_time, video_time;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc < 2) {
if (argc != 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
@ -582,10 +470,6 @@ int main(int argc, char **argv)
}
filename = argv[1];
for (i = 2; i+1 < argc; i+=2) {
if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@ -593,31 +477,29 @@ int main(int argc, char **argv)
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
if (!oc) {
return 1;
}
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
video_st = NULL;
audio_st = NULL;
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
video_st = add_stream(oc, &video_codec, fmt->video_codec);
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
if (video_st)
open_video(oc, video_codec, video_st);
if (audio_st)
open_audio(oc, audio_codec, audio_st);
av_dump_format(oc, 0, filename, 1);
@ -632,21 +514,30 @@ int main(int argc, char **argv)
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
if (frame)
frame->pts = 0;
for (;;) {
/* Compute current audio and video time. */
audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
if ((!audio_st || audio_time >= STREAM_DURATION) &&
(!video_st || video_time >= STREAM_DURATION))
break;
/* write interleaved audio and video frames */
if (!video_st || (video_st && audio_st && audio_time < video_time)) {
write_audio_frame(oc, audio_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
write_video_frame(oc, video_st);
frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
}
}
@ -657,14 +548,14 @@ int main(int argc, char **argv)
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (video_st)
close_video(oc, video_st);
if (audio_st)
close_audio(oc, audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
avio_close(oc->pb);
/* free the stream */
avformat_free_context(oc);

View File

@ -1,487 +0,0 @@
/*
* Copyright (c) 2015 Anton Khirnov
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel QSV-accelerated H.264 decoding example.
*
* @example qsvdec.c
* This example shows how to do QSV-accelerated H.264 decoding with output
* 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/error.h"
#include "libavutil/mem.h"
typedef struct DecodeContext {
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) {
if (!avctx->hwaccel_context) {
DecodeContext *decode = avctx->opaque;
AVQSVContext *qsv = av_qsv_alloc_context();
if (!qsv)
return AV_PIX_FMT_NONE;
qsv->session = decode->mfx_session;
qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
avctx->hwaccel_context = qsv;
}
return AV_PIX_FMT_QSV;
}
pix_fmts++;
}
fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
return AV_PIX_FMT_NONE;
}
static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
AVFrame *frame, AVPacket *pkt,
AVIOContext *output_ctx)
{
int ret = 0;
int got_frame = 1;
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. */
if (got_frame) {
mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
VASurfaceID surface = *(VASurfaceID*)surf->Data.MemId;
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:
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;
}
}
return 0;
}
int main(int argc, char **argv)
{
AVFormatContext *input_ctx = NULL;
AVStream *video_st = NULL;
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder;
AVPacket pkt = { 0 };
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, err;
av_register_all();
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
/* open the input file */
ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
goto finish;
}
/* find the first H.264 video stream */
for (i = 0; i < input_ctx->nb_streams; i++) {
AVStream *st = input_ctx->streams[i];
if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
video_st = st;
else
st->discard = AVDISCARD_ALL;
}
if (!video_st) {
fprintf(stderr, "No H.264 video stream in the input file\n");
goto finish;
}
/* 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");
if (!decoder) {
fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
goto finish;
}
decoder_ctx = avcodec_alloc_context3(decoder);
if (!decoder_ctx) {
ret = AVERROR(ENOMEM);
goto finish;
}
decoder_ctx->codec_id = AV_CODEC_ID_H264;
if (video_st->codecpar->extradata_size) {
decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!decoder_ctx->extradata) {
ret = AVERROR(ENOMEM);
goto finish;
}
memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
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);
if (ret < 0) {
fprintf(stderr, "Error opening the decoder: ");
goto finish;
}
/* open the output stream */
ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Error opening the output context: ");
goto finish;
}
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
goto finish;
}
/* actual decoding */
while (ret >= 0) {
ret = av_read_frame(input_ctx, &pkt);
if (ret < 0)
break;
if (pkt.stream_index == video_st->index)
ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
av_packet_unref(&pkt);
}
/* flush the decoder */
pkt.data = NULL;
pkt.size = 0;
ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
finish:
if (ret < 0) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "%s\n", buf);
}
avformat_close_input(&input_ctx);
av_frame_free(&frame);
if (decoder_ctx)
av_freep(&decoder_ctx->hwaccel_context);
avcodec_free_context(&decoder_ctx);
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);
return ret;
}

View File

@ -1,165 +0,0 @@
/*
* Copyright (c) 2013 Stefano Sabatini
*
* 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
* libavformat/libavcodec demuxing and muxing API example.
*
* Remux streams from one container format to another.
* @example remuxing.c
*/
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
tag,
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
int main(int argc, char **argv)
{
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
if (argc < 3) {
printf("usage: %s input output\n"
"API example program to remux a media file with libavformat and libavcodec.\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return 1;
}
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;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
fprintf(stderr, "Failed to retrieve input stream information");
goto end;
}
av_dump_format(ifmt_ctx, 0, in_filename, 0);
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
if (!ofmt_ctx) {
fprintf(stderr, "Could not create output context\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *in_stream = ifmt_ctx->streams[i];
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_copy_context(out_stream->codec, in_stream->codec);
if (ret < 0) {
fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
goto end;
}
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);
if (!(ofmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open output file '%s'", out_filename);
goto end;
}
}
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file\n");
goto end;
}
while (1) {
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, &pkt);
if (ret < 0)
break;
in_stream = ifmt_ctx->streams[pkt.stream_index];
out_stream = ofmt_ctx->streams[pkt.stream_index];
log_packet(ifmt_ctx, &pkt, "in");
/* copy packet */
pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
pkt.pos = -1;
log_packet(ofmt_ctx, &pkt, "out");
ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
av_packet_unref(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

View File

@ -21,7 +21,7 @@
*/
/**
* @example resampling_audio.c
* @example doc/examples/resampling_audio.c
* libswresample API use example.
*/
@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
@ -168,7 +168,7 @@ int main(int argc, char **argv)
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
av_free(dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
@ -184,10 +184,6 @@ int main(int argc, char **argv)
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
@ -199,7 +195,8 @@ int main(int argc, char **argv)
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (dst_file)
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);

View File

@ -23,7 +23,7 @@
/**
* @file
* libswscale API use example.
* @example scaling_video.c
* @example doc/examples/scaling_video.c
*/
#include <libavutil/imgutils.h>
@ -132,7 +132,8 @@ int main(int argc, char **argv)
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
if (dst_file)
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);

View File

@ -1,802 +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
*/
/**
* @file
* simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 96000
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/**
* 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)
{
AVCodecContext *avctx;
AVCodec *input_codec;
int error;
/** 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, get_error_text(error));
*input_format_context = NULL;
return error;
}
/** 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",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** 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);
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** 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 */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate a decoding context\n");
avformat_close_input(input_format_context);
return AVERROR(ENOMEM);
}
/** 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);
avcodec_free_context(&avctx);
return error;
}
/** 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",
get_error_text(error));
avcodec_free_context(&avctx);
avformat_close_input(input_format_context);
return error;
}
/** Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
}
/**
* 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.
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
AVCodecContext *avctx = NULL;
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
int error;
/** 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, get_error_text(error));
return error;
}
/** 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. */
(*output_format_context)->pb = output_io_context;
/** 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;
}
av_strlcpy((*output_format_context)->filename, filename,
sizeof((*output_format_context)->filename));
/** 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. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
avctx = avcodec_alloc_context3(output_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate an encoding context\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/**
* 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 */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/** Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
/**
* 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. */
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
get_error_text(error));
goto cleanup;
}
error = avcodec_parameters_from_context(stream->codecpar, avctx);
if (error < 0) {
fprintf(stderr, "Could not initialize stream parameters\n");
goto cleanup;
}
/** Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
cleanup:
avcodec_free_context(&avctx);
avio_closep(&(*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
}
/** 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. */
packet->data = NULL;
packet->size = 0;
}
/** Initialize one audio frame for reading from the input file */
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate input frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* 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.
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext **resample_context)
{
int error;
/**
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!*resample_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. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/** 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. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
/** 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",
get_error_text(error));
return error;
}
return 0;
}
/** 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. */
AVPacket input_packet;
int error;
init_packet(&input_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 (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
get_error_text(error));
return 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;
}
/**
* 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 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.
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/**
* 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).
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/**
* 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",
get_error_text(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
}
/**
* 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.
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
SwrContext *resample_context)
{
int error;
/** 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",
get_error_text(error));
return error;
}
return 0;
}
/** 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.
*/
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. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
/**
* Read one audio frame from the input file, decodes, converts and stores
* it in the FIFO buffer.
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext *resampler_context,
int *finished)
{
/** Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/** Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int ret = AVERROR_EXIT;
/** Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/** 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
* in the decoder which are delayed, we are actually 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 (data_present) {
/** 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.
*/
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. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/** 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.
* 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.
*/
(*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.
*/
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
get_error_text(error));
av_frame_free(frame);
return error;
}
return 0;
}
/** Global timestamp for the audio frames */
static int64_t pts = 0;
/** 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. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
/** Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
/**
* 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;
}
/** 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);
}
return 0;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
*/
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. */
AVFrame *output_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
*/
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. */
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.
*/
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. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
av_frame_free(&output_frame);
return 0;
}
/** 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",
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;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/** 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. */
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. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/** 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. */
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.
*/
while (1) {
/** 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
* 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.
*/
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.
*/
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 (finished)
break;
}
/**
* 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.
*/
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.
*/
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 (finished) {
int data_written;
/** Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/** Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_free_context(&output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_free_context(&input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}

View File

@ -1,585 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2014 Andrey Utkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for demuxing, decoding, filtering, encoding and muxing
* @example transcoding.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
} FilteringContext;
static FilteringContext *filter_ctx;
static int open_input_file(const char *filename)
{
int ret;
unsigned int i;
ifmt_ctx = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream;
AVCodecContext *codec_ctx;
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) {
/* Open decoder */
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;
}
}
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
AVCodec *encoder;
int ret;
unsigned int i;
ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return AVERROR_UNKNOWN;
}
in_stream = ifmt_ctx->streams[i];
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) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
if (encoder->pix_fmts)
enc_ctx->pix_fmt = encoder->pix_fmts[0];
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 = dec_ctx->time_base;
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
/* 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;
}
} 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_copy_context(ofmt_ctx->streams[i]->codec,
ifmt_ctx->streams[i]->codec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
return ret;
}
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, filename, 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
return ret;
}
}
/* init muxer, write output file header */
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return ret;
}
return 0;
}
static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
AVCodecContext *enc_ctx, const char *filter_spec)
{
char args[512];
int ret = 0;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVFilterGraph *filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
(uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout =
av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
(uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
(uint8_t*)&enc_ctx->channel_layout,
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
(uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if (!outputs->name || !inputs->name) {
ret = AVERROR(ENOMEM);
goto end;
}
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Fill FilteringContext */
fctx->buffersrc_ctx = buffersrc_ctx;
fctx->buffersink_ctx = buffersink_ctx;
fctx->filter_graph = filter_graph;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static int init_filters(void)
{
const char *filter_spec;
unsigned int i;
int ret;
filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
if (!filter_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
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]->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], ifmt_ctx->streams[i]->codec,
ofmt_ctx->streams[i]->codec, filter_spec);
if (ret)
return ret;
}
return 0;
}
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codec->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
got_frame = &got_frame_local;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&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)
return ret;
if (!(*got_frame))
return 0;
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
ret = AVERROR(ENOMEM);
break;
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
* rewrite retcode to 0 to show it as normal procedure completion
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
int ret;
int got_frame;
if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
while (1) {
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index, &got_frame);
if (ret < 0)
break;
if (!got_frame)
return 0;
}
return ret;
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
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)
goto end;
if ((ret = init_filters()) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
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);
if (filter_ctx[stream_index].filter_graph) {
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->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(ifmt_ctx->streams[stream_index]->codec, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
goto end;
} else {
av_frame_free(&frame);
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_packet_unref(&packet);
}
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_packet_unref(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
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);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}

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@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg FAQ
@titlepage
@ -91,63 +90,13 @@ To build FFmpeg, you need to install the development package. It is usually
called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
build is finished, but be sure to keep the main package.
@section How do I make @command{pkg-config} find my libraries?
Somewhere along with your libraries, there is a @file{.pc} file (or several)
in a @file{pkgconfig} directory. You need to set environment variables to
point @command{pkg-config} to these files.
If you need to @emph{add} directories to @command{pkg-config}'s search list
(typical use case: library installed separately), add it to
@code{$PKG_CONFIG_PATH}:
@example
export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
@end example
If you need to @emph{replace} @command{pkg-config}'s search list
(typical use case: cross-compiling), set it in
@code{$PKG_CONFIG_LIBDIR}:
@example
export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
@end example
If you need to know the library's internal dependencies (typical use: static
linking), add the @code{--static} option to @command{pkg-config}:
@example
./configure --pkg-config-flags=--static
@end example
@section How do I use @command{pkg-config} when cross-compiling?
The best way is to install @command{pkg-config} in your cross-compilation
environment. It will automatically use the cross-compilation libraries.
You can also use @command{pkg-config} from the host environment by
specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
In that case, you must point @command{pkg-config} to the correct directories
using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
As an intermediate solution, you can place in your cross-compilation
environment a script that calls the host @command{pkg-config} with
@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
@example
#!/bin/sh
PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
export PKG_CONFIG_LIBDIR
exec /usr/bin/pkg-config "$@@"
@end example
@chapter Usage
@section ffmpeg does not work; what is wrong?
Try a @code{make distclean} in the ffmpeg source directory before the build.
If this does not help see
(@url{https://ffmpeg.org/bugreports.html}).
(@url{http://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
@ -156,7 +105,7 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
Then you may run:
@example
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example
Notice that @samp{%d} is replaced by the image number.
@ -169,7 +118,7 @@ the sequence. This is useful if your sequence does not start with
example will start with @file{img100.jpg}:
@example
ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example
If you have large number of pictures to rename, you can use the
@ -179,7 +128,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@ -188,7 +137,7 @@ If you want to sequence them by oldest modified first, substitute
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
The same logic is used for any image format that ffmpeg reads.
@ -196,7 +145,7 @@ The same logic is used for any image format that ffmpeg reads.
You can also use @command{cat} to pipe images to ffmpeg:
@example
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example
@section How do I encode movie to single pictures?
@ -204,7 +153,7 @@ cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
Use:
@example
ffmpeg -i movie.mpg movie%d.jpg
ffmpeg -i movie.mpg movie%d.jpg
@end example
The @file{movie.mpg} used as input will be converted to
@ -220,7 +169,7 @@ to force the encoding.
Applying that to the previous example:
@example
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@ -278,15 +227,15 @@ then you may use any file that DirectShow can read as input.
Just create an "input.avs" text file with this single line ...
@example
DirectShowSource("C:\path to your file\yourfile.asf")
DirectShowSource("C:\path to your file\yourfile.asf")
@end example
... and then feed that text file to ffmpeg:
@example
ffmpeg -i input.avs
ffmpeg -i input.avs
@end example
For ANY other help on AviSynth, please visit the
@uref{http://www.avisynth.org/, AviSynth homepage}.
For ANY other help on Avisynth, please visit the
@uref{http://www.avisynth.org/, Avisynth homepage}.
@section How can I join video files?
@ -311,18 +260,18 @@ 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{https://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@url{http://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{https://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
using the @url{http://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{https://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@ -333,23 +282,23 @@ There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
FFmpeg has a @url{https://ffmpeg.org/ffmpeg-filters.html#concat,
FFmpeg has a @url{http://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{https://www.ffmpeg.org/ffmpeg-formats.html#concat,
FFmpeg has a @url{http://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{https://ffmpeg.org/ffmpeg-protocols.html#concat,
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
video by merely concatenating the files containing them.
Hence you may concatenate your multimedia files by first transcoding them to
@ -419,6 +368,26 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
rm temp[12].[av] all.[av]
@end example
@section -profile option fails when encoding H.264 video with AAC audio
@command{ffmpeg} prints an error like
@example
Undefined constant or missing '(' in 'baseline'
Unable to parse option value "baseline"
Error setting option profile to value baseline.
@end example
Short answer: write @option{-profile:v} instead of @option{-profile}.
Long answer: this happens because the @option{-profile} option can apply to both
video and audio. Specifically the AAC encoder also defines some profiles, none
of which are named @var{baseline}.
The solution is to apply the @option{-profile} option to the video stream only
by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
Appending @code{:v} to it will do exactly that.
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
Use @option{-dumpgraph -} to find out exactly where the channel layout is
@ -443,7 +412,7 @@ VOB and a few other formats do not have a global header that describes
everything present in the file. Instead, applications are supposed to scan
the file to see what it contains. Since VOB files are frequently large, only
the beginning is scanned. If the subtitles happen only later in the file,
they will not be initially detected.
they will not be initally detected.
Some applications, including the @code{ffmpeg} command-line tool, can only
work with streams that were detected during the initial scan; streams that
@ -467,40 +436,6 @@ point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.
@section I have a stretched video, why does scaling does not fix it?
A lot of video codecs and formats can store the @emph{aspect ratio} of the
video: this is the ratio between the width and the height of either the full
image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
SAR.
Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
of video standards, especially from the analogic-numeric transition era, use
non-square pixels.
Most processing filters in FFmpeg handle the aspect ratio to avoid
stretching the image: cropping adjusts the DAR to keep the SAR constant,
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{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.
For example, to fix a badly encoded EGA capture, use the following commands,
either the first one to upscale to square pixels or the second one to set
the correct aspect ratio or the third one to avoid transcoding (may not work
depending on the format / codec / player / phase of the moon):
@example
ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
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
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@ -540,10 +475,9 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?
The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
contain full debug information. Those binaries are stripped to create
@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
the *_g versions.
The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
you need the debug information, use the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?
@ -563,7 +497,7 @@ An easy way to get the full list of required libraries in dependency order
is to use @code{pkg-config}.
@example
c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example
See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
@ -587,9 +521,13 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
You have to create a custom AVIOContext using @code{avio_alloc_context},
see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
@section Where can I find libav* headers for Pascal/Delphi?
see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
see @url{https://www.ffmpeg.org/~michael/}
see @url{http://www.ffmpeg.org/~michael/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
@ -599,12 +537,11 @@ In this specific case please look at RFC 4629 to see how it should be done.
@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
r_frame_rate is NOT the average frame rate, it is the smallest frame rate
that can accurately represent all timestamps. So no, it is not
wrong if it is larger than the average!
For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
will be 150 (it is the least common multiple).
If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
For example, if you have mixed 25 and 30 fps content, then r_frame_rate
will be 150.
@section Why is @code{make fate} not running all tests?

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@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Automated Testing Environment
@titlepage
@ -13,36 +12,36 @@
@chapter Introduction
FATE is an extended regression suite on the client-side and a means
FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
The first part of this document explains how you can use FATE from
The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg's
FATE server.
In any way you can have a look at the publicly viewable FATE results
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@url{http://fate.ffmpeg.org/}
@url{http://fate.ffmpeg.org/}
This is especially recommended for all people contributing source
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
the developers could not test on.
The second part of this document describes how you can run FATE to
The second part of this document describes how you can run FATE to
submit your results to FFmpeg's FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
In the third part you can find a comprehensive listing of FATE makefile
In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine you need to have the samples
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@ -51,11 +50,11 @@ make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
The above commands set the samples location by passing a makefile
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
@option{--samples=<path to the samples directory>}. Afterwards you can
invoke the makefile targets without setting the @var{SAMPLES} makefile
`--samples=<path to the samples directory>'. Afterwards you can
invoke the makefile targets without setting the SAMPLES makefile
variable. This is illustrated by the following commands:
@example
@ -64,7 +63,7 @@ make fate-rsync
make fate
@end example
Yet another way to tell FATE about the location of the sample
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
@ -85,7 +84,7 @@ To use a custom wrapper to run the test, pass @option{--target-exec} to
@chapter Submitting the results to the FFmpeg result aggregation server
To submit your results to the server you should run fate through the
To submit your results to the server you should run fate through the
shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
@ -93,23 +92,23 @@ to be invoked with a configuration file as its first argument.
tests/fate.sh /path/to/fate_config
@end example
A configuration file template with comments describing the individual
A configuration file template with comments describing the individual
configuration variables can be found at @file{doc/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
template. The @env{slot} configuration variable can be any string that is not
Create a configuration that suits your needs, based on the configuration
template. The `slot' configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
configuration file itself will be sourced in a shell script, therefore all
shell features may be used. This enables you to setup the environment as you
need it for your build.
pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
itself will be sourced in a shell script, therefore all shell features may
be used. This enables you to setup the environment as you need it for your
build.
For your first test runs the @env{fate_recv} variable should be empty or
For your first test runs the `fate_recv' variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
@ -122,29 +121,29 @@ present in $workdir as specified in the configuration file:
@item version
@end itemize
When you have everything working properly you can create an SSH key pair
When you have everything working properly you can create an SSH key pair
and send the public key to the FATE server administrator who can be contacted
at the email address @email{fate-admin@@ffmpeg.org}.
Configure your SSH client to use public key authentication with that key
Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server's fingerprint is:
@table @samp
@table @option
@item RSA
d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table
If you have problems connecting to the FATE server, it may help to try out
If you have problems connecting to the FATE server, it may help to try out
the @command{ssh} command with one or more @option{-v} options. You should
get detailed output concerning your SSH configuration and the authentication
process.
The only thing left is to automate the execution of the fate.sh script and
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@ -154,20 +153,20 @@ the synchronisation of the samples directory.
@table @option
@item fate-rsync
Download/synchronize sample files to the configured samples directory.
Download/synchronize sample files to the configured samples directory.
@item fate-list
Will list all fate/regression test targets.
Will list all fate/regression test targets.
@item fate
Run the FATE test suite (requires the fate-suite dataset).
Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@table @env
@table @option
@item V
Verbosity level, can be set to 0, 1 or 2.
Verbosity level, can be set to 0, 1 or 2.
@itemize
@item 0: show just the test arguments
@item 1: show just the command used in the test
@ -175,28 +174,24 @@ Verbosity level, can be set to 0, 1 or 2.
@end itemize
@item SAMPLES
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
@item THREADS
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@item THREAD_TYPE
Specify which threading strategy test, either @samp{slice} or @samp{frame},
by default @samp{slice+frame}
Specify which threading strategy test, either @var{slice} or @var{frame},
by default @var{slice+frame}
@item CPUFLAGS
Specify CPU flags.
Specify CPU flags.
@item TARGET_EXEC
Specify or override the wrapper used to run the tests.
The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
through @command{ssh}.
Specify or override the wrapper used to run the tests.
The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
through @command{ssh}.
@item GEN
Set to @samp{1} to generate the missing or mismatched references.
Set to @var{1} to generate the missing or mismatched references.
@end table
@section Examples

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@ -1,6 +1,5 @@
slot= # some unique identifier
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
#branch=release/2.6 # the branch to test
samples= # path to samples directory
workdir= # directory in which to do all the work
#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report

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@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Bitstream Filters Documentation
@titlepage

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Codecs Documentation
@titlepage

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Devices Documentation
@titlepage

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Filters Documentation
@titlepage

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Formats Documentation
@titlepage

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Protocols Documentation
@titlepage

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Resampler Documentation
@titlepage
@ -13,9 +12,9 @@
@chapter Description
@c man begin DESCRIPTION
The FFmpeg resampler provides a high-level interface to the
The FFmpeg resampler provides an high-level interface to the
libswresample library audio resampling utilities. In particular it
allows one to perform audio resampling, audio channel layout rematrixing,
allows to perform audio resampling, audio channel layout rematrixing,
and convert audio format and packing layout.
@c man end DESCRIPTION

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@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Scaler Documentation
@titlepage
@ -13,8 +12,8 @@
@chapter Description
@c man begin DESCRIPTION
The FFmpeg rescaler provides a high-level interface to the libswscale
library image conversion utilities. In particular it allows one to perform
The FFmpeg rescaler provides an high-level interface to the libswscale
library image conversion utilities. In particular it allows to perform
image rescaling and pixel format conversion.
@c man end DESCRIPTION

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@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Utilities Documentation
@titlepage

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@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffmpeg Documentation
@titlepage
@ -80,26 +79,14 @@ The format option may be needed for raw input files.
The transcoding process in @command{ffmpeg} for each output can be described by
the following diagram:
@verbatim
_______ ______________
| | | |
| input | demuxer | encoded data | decoder
| file | ---------> | packets | -----+
|_______| |______________| |
v
_________
| |
| decoded |
| frames |
|_________|
________ ______________ |
| | | | |
| output | <-------- | encoded data | <----+
| file | muxer | packets | encoder
|________| |______________|
@example
_______ ______________ _________ ______________ ________
| | | | | | | | | |
| input | demuxer | encoded data | decoder | decoded | encoder | encoded data | muxer | output |
| file | ---------> | packets | ---------> | frames | ---------> | packets | -------> | file |
|_______| |______________| |_________| |______________| |________|
@end verbatim
@end example
@command{ffmpeg} calls the libavformat library (containing demuxers) to read
input files and get packets containing encoded data from them. When there are
@ -124,31 +111,26 @@ Simple filtergraphs are those that have exactly one input and output, both of
the same type. In the above diagram they can be represented by simply inserting
an additional step between decoding and encoding:
@verbatim
_________ ______________
| | | |
| decoded | | encoded data |
| frames |\ _ | packets |
|_________| \ /||______________|
\ __________ /
simple _\|| | / encoder
filtergraph | filtered |/
| frames |
|__________|
@example
_________ __________ ______________
| | | | | |
| decoded | simple filtergraph | filtered | encoder | encoded data |
| frames | -------------------> | frames | ---------> | packets |
|_________| |__________| |______________|
@end verbatim
@end example
Simple filtergraphs are configured with the per-stream @option{-filter} option
(with @option{-vf} and @option{-af} aliases for video and audio respectively).
A simple filtergraph for video can look for example like this:
@verbatim
_______ _____________ _______ ________
| | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | output |
|_______| |_____________| |_______| |________|
@example
_______ _____________ _______ _____ ________
| | | | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
|_______| |_____________| |_______| |_____| |________|
@end verbatim
@end example
Note that some filters change frame properties but not frame contents. E.g. the
@code{fps} filter in the example above changes number of frames, but does not
@ -161,7 +143,7 @@ processing chain applied to one stream. This is the case, for example, when the
more than one input and/or output, or when output stream type is different from
input. They can be represented with the following diagram:
@verbatim
@example
_________
| |
| input 0 |\ __________
@ -179,7 +161,7 @@ input. They can be represented with the following diagram:
| input 2 |/
|_________|
@end verbatim
@end example
Complex filtergraphs are configured with the @option{-filter_complex} option.
Note that this option is global, since a complex filtergraph, by its nature,
@ -198,14 +180,14 @@ step for the specified stream, so it does only demuxing and muxing. It is useful
for changing the container format or modifying container-level metadata. The
diagram above will, in this case, simplify to this:
@verbatim
@example
_______ ______________ ________
| | | | | |
| input | demuxer | encoded data | muxer | output |
| file | ---------> | packets | -------> | file |
|_______| |______________| |________|
@end verbatim
@end example
Since there is no decoding or encoding, it is very fast and there is no quality
loss. However, it might not work in some cases because of many factors. Applying
@ -232,7 +214,7 @@ described.
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@ -253,10 +235,6 @@ Overwrite output files without asking.
Do not overwrite output files, and exit immediately if a specified
output file already exists.
@item -stream_loop @var{number} (@emph{input})
Set number of times input stream shall be looped. Loop 0 means no loop,
loop -1 means infinite loop.
@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
Select an encoder (when used before an output file) or a decoder (when used
@ -277,72 +255,54 @@ ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT
will copy all the streams except the second video, which will be encoded with
libx264, and the 138th audio, which will be encoded with libvorbis.
@item -t @var{duration} (@emph{input/output})
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 filename), stop writing the
output after its duration reaches @var{duration}.
@var{duration} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -t @var{duration} (@emph{output})
Stop writing the output after its duration reaches @var{duration}.
@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
-to and -t are mutually exclusive and -t has priority.
@item -to @var{position} (@emph{output})
Stop writing the output at @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
-to and -t are mutually exclusive and -t has priority.
@item -fs @var{limit_size} (@emph{output})
Set the file size limit, expressed in bytes. No further chunk of bytes is written
after the limit is exceeded. The size of the output file is slightly more than the
requested file size.
Set the file size limit, expressed in bytes.
@item -ss @var{position} (@emph{input/output})
When used as an input option (before @code{-i}), seeks in this input file to
@var{position}. Note that in most formats it is not possible to seek exactly,
so @command{ffmpeg} will seek to the closest seek point before @var{position}.
When transcoding and @option{-accurate_seek} is enabled (the default), this
extra segment between the seek point and @var{position} will be decoded and
discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
will be preserved.
@var{position}. When used as an output option (before an output filename),
decodes but discards input until the timestamps reach @var{position}. This is
slower, but more accurate.
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/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.
@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
@item -itsoffset @var{offset} (@emph{input})
Set the input time offset.
Set the input time offset in seconds.
@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by @var{offset} seconds.
@var{offset} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
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 -timestamp @var{date} (@emph{output})
@item -timestamp @var{time} (@emph{output})
Set the recording timestamp in the container.
@var{date} must be a date specification,
see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
The syntax for @var{time} is:
@example
now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
@end example
If the value is "now" it takes the current time.
Time is local time unless 'Z' or 'z' is appended, in which case it is
interpreted as UTC.
If the year-month-day part is not specified it takes the current
year-month-day.
@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
Set a metadata key/value pair.
An optional @var{metadata_specifier} may be given to set metadata
on streams, chapters or programs. See @code{-map_metadata}
documentation for details.
on streams or chapters. See @code{-map_metadata} documentation for
details.
This option overrides metadata set with @code{-map_metadata}. It is
also possible to delete metadata by using an empty value.
@ -354,14 +314,9 @@ ffmpeg -i in.avi -metadata title="my title" out.flv
To set the language of the first audio stream:
@example
ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
@end example
@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@var{stream}(s) to it.
@item -target @var{type} (@emph{output})
Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv},
@code{dv50}). @var{type} may be prefixed with @code{pal-}, @code{ntsc-} or
@ -380,20 +335,15 @@ ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dframes @var{number} (@emph{output})
Set the number of data frames to output. This is an alias for @code{-frames:d}.
Set the number of data frames to record. 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.
@item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
@itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
Use fixed quality scale (VBR). The meaning of @var{q} is
codec-dependent.
If @var{qscale} is used without a @var{stream_specifier} then it applies only
to the video stream, this is to maintain compatibility with previous behavior
and as specifying the same codec specific value to 2 different codecs that is
audio and video generally is not what is intended when no stream_specifier is
used.
@anchor{filter_option}
@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
@ -481,24 +431,18 @@ Technical note -- attachments are implemented as codec extradata, so this
option can actually be used to extract extradata from any stream, not just
attachments.
@item -noautorotate
Disable automatically rotating video based on file metadata.
@end table
@section Video Options
@table @option
@item -vframes @var{number} (@emph{output})
Set the number of video frames to output. This is an alias for @code{-frames:v}.
Set the number of video frames to record. This is an alias for @code{-frames:v}.
@item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
Set frame rate (Hz value, fraction or abbreviation).
As an input option, ignore any timestamps stored in the file and instead
generate timestamps assuming constant frame rate @var{fps}.
This is not the same as the @option{-framerate} option used for some input formats
like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
If in doubt use @option{-framerate} instead of the input option @option{-r}.
As an output option, duplicate or drop input frames to achieve constant output
frame rate @var{fps}.
@ -553,6 +497,9 @@ prefix is ``ffmpeg2pass''. The complete file name will be
@file{PREFIX-N.log}, where N is a number specific to the output
stream
@item -vlang @var{code}
Set the ISO 639 language code (3 letters) of the current video stream.
@item -vf @var{filtergraph} (@emph{output})
Create the filtergraph specified by @var{filtergraph} and use it to
filter the stream.
@ -560,7 +507,7 @@ filter the stream.
This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
@end table
@section Advanced Video options
@section Advanced Video Options
@table @option
@item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
@ -663,87 +610,13 @@ would be more efficient.
@item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
When doing stream copy, copy also non-key frames found at the
beginning.
@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:
@table @option
@item none
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 qsv
Use the Intel QuickSync Video acceleration for video transcoding.
Unlike most other values, this option does not enable accelerated decoding (that
is used automatically whenever a qsv decoder is selected), but accelerated
transcoding, without copying the frames into the system memory.
For it to work, both the decoder and the encoder must support QSV acceleration
and no filters must be used.
@end table
This option has no effect if the selected hwaccel is not available or not
supported by the chosen decoder.
Note that most acceleration methods are intended for playback and will not be
faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
will usually need to copy the decoded frames from the GPU memory into the system
memory, resulting in further performance loss. This option is thus mainly
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. 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.
@end table
@section Audio Options
@table @option
@item -aframes @var{number} (@emph{output})
Set the number of audio frames to output. This is an alias for @code{-frames:a}.
Set the number of audio frames to record. 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
@ -771,7 +644,7 @@ filter the stream.
This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
@end table
@section Advanced Audio options
@section Advanced Audio options:
@table @option
@item -atag @var{fourcc/tag} (@emph{output})
@ -786,9 +659,11 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
0 to disable all guessing.
@end table
@section Subtitle options
@section Subtitle options:
@table @option
@item -slang @var{code}
Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -scodec @var{codec} (@emph{input/output})
Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{output})
@ -797,7 +672,7 @@ Disable subtitle recording.
Deprecated, see -bsf
@end table
@section Advanced Subtitle options
@section Advanced Subtitle options:
@table @option
@ -875,21 +750,8 @@ 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 pick the English audio stream:
@example
ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
@end example
Note that using this option disables the default mappings for this output file.
@item -ignore_unknown
Ignore input streams with unknown type instead of failing if copying
such streams is attempted.
@item -copy_unknown
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}]
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
@ -1009,12 +871,13 @@ Dump each input packet to stderr.
When dumping packets, also dump the payload.
@item -re (@emph{input})
Read input at native frame rate. Mainly used to simulate a grab device.
or live input stream (e.g. when reading from a file). Should not be used
with actual grab devices or live input streams (where it can cause packet
loss).
By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
This option will slow down the reading of the input(s) to the native frame rate
of the input(s). It is useful for real-time output (e.g. live streaming).
of the input(s). It is useful for real-time output (e.g. live streaming). If
your input(s) is coming from some other live streaming source (through HTTP or
UDP for example) the server might already be in real-time, thus the option will
likely not be required. On the other hand, this is meaningful if your input(s)
is a file you are trying to push in real-time.
@item -loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
@ -1053,13 +916,6 @@ With -map you can select from which stream the timestamps should be
taken. You can leave either video or audio unchanged and sync the
remaining stream(s) to the unchanged one.
@item -frame_drop_threshold @var{parameter}
Frame drop threshold, which specifies how much behind video frames can
be before they are dropped. In frame rate units, so 1.0 is one frame.
The default is -1.1. One possible usecase is to avoid framedrops in case
of noisy timestamps or to increase frame drop precision in case of exact
timestamps.
@item -async @var{samples_per_second}
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
the parameter is the maximum samples per second by which the audio is changed.
@ -1082,12 +938,6 @@ processing (e.g. in case the format option @option{avoid_negative_ts}
is enabled) the output timestamps may mismatch with the input
timestamps even when this option is selected.
@item -start_at_zero
When used with @option{copyts}, shift input timestamps so they start at zero.
This means that using e.g. @code{-ss 50} will make output timestamps start at
50 seconds, regardless of what timestamp the input file started at.
@item -copytb @var{mode}
Specify how to set the encoder timebase when stream copying. @var{mode} is an
integer numeric value, and can assume one of the following values:
@ -1143,7 +993,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
@end example
@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
Force a tag/fourcc for matching streams.
@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@ -1210,74 +1060,13 @@ This option is similar to @option{-filter_complex}, the only difference is that
its argument is the name of the file from which a complex filtergraph
description is to be read.
@item -accurate_seek (@emph{input})
This option enables or disables accurate seeking in input files with the
@option{-ss} option. It is enabled by default, so seeking is accurate when
transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
e.g. when copying some streams and transcoding the others.
@item -seek_timestamp (@emph{input})
This option enables or disables seeking by timestamp in input files with the
@option{-ss} option. It is disabled by default. If enabled, the argument
to the @option{-ss} option is considered an actual timestamp, and is not
offset by the start time of the file. This matters only for files which do
not start from timestamp 0, such as transport streams.
@item -thread_queue_size @var{size} (@emph{input})
This option sets the maximum number of queued packets when reading from the
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}.
Overrides the input specifications from ffserver. Using this option you can
map any input stream to ffserver and control many aspects of the encoding from
ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
ffserver.
The option is intended for cases where features are needed that cannot be
specified to @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 of streams at the demuxer.
Not all demuxers support this.
@table @option
@item none
Discard no frame.
@item default
Default, which discards no frames.
@item noref
Discard all non-reference frames.
@item bidir
Discard all bidirectional frames.
@item nokey
Discard all frames excepts keyframes.
@item all
Discard all frames.
@end table
@item -abort_on @var{flags} (@emph{global})
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.
@end table
@item -xerror (@emph{global})
Stop and exit on error
specified to ffserver but can be to ffmpeg.
@end table
@ -1304,10 +1093,7 @@ awkward to specify on the command line. Lines starting with the hash
('#') character are ignored and are used to provide comments. Check
the @file{presets} directory in the FFmpeg source tree for examples.
There are two types of preset files: ffpreset and avpreset files.
@subsection ffpreset files
ffpreset files are specified with the @code{vpre}, @code{apre},
Preset files are specified with the @code{vpre}, @code{apre},
@code{spre}, and @code{fpre} options. The @code{fpre} option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the @code{vpre}, @code{apre}, and
@ -1332,31 +1118,67 @@ directories, where @var{codec_name} is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p},
then it will search for the file @file{libvpx-1080p.ffpreset}.
@subsection avpreset files
avpreset files are specified with the @code{pre} option. They work similar to
ffpreset files, but they only allow encoder- specific options. Therefore, an
@var{option}=@var{value} pair specifying an encoder cannot be used.
When the @code{pre} option is specified, ffmpeg will look for files with the
suffix .avpreset in the directories @file{$AVCONV_DATADIR} (if set), and
@file{$HOME/.avconv}, and in the datadir defined at configuration time (usually
@file{PREFIX/share/ffmpeg}), in that order.
First ffmpeg searches for a file named @var{codec_name}-@var{arg}.avpreset in
the above-mentioned directories, where @var{codec_name} is the name of the codec
to which the preset file options will be applied. For example, if you select the
video codec with @code{-vcodec libvpx} and use @code{-pre 1080p}, then it will
search for the file @file{libvpx-1080p.avpreset}.
If no such file is found, then ffmpeg will search for a file named
@var{arg}.avpreset in the same directories.
@c man end OPTIONS
@chapter Tips
@c man begin TIPS
@itemize
@item
For streaming at very low bitrate application, use a low frame rate
and a small GOP size. This is especially true for RealVideo where
the Linux player does not seem to be very fast, so it can miss
frames. An example is:
@example
ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm
@end example
@item
The parameter 'q' which is displayed while encoding is the current
quantizer. The value 1 indicates that a very good quality could
be achieved. The value 31 indicates the worst quality. If q=31 appears
too often, it means that the encoder cannot compress enough to meet
your bitrate. You must either increase the bitrate, decrease the
frame rate or decrease the frame size.
@item
If your computer is not fast enough, you can speed up the
compression at the expense of the compression ratio. You can use
'-me zero' to speed up motion estimation, and '-g 0' to disable
motion estimation completely (you have only I-frames, which means it
is about as good as JPEG compression).
@item
To have very low audio bitrates, reduce the sampling frequency
(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
@item
To have a constant quality (but a variable bitrate), use the option
'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
quality).
@end itemize
@c man end TIPS
@chapter Examples
@c man begin EXAMPLES
@section Preset files
A preset file contains a sequence of @var{option=value} pairs, one for
each line, specifying a sequence of options which can be specified also on
the command line. Lines starting with the hash ('#') character are ignored and
are used to provide comments. Empty lines are also ignored. Check the
@file{presets} directory in the FFmpeg source tree for examples.
Preset files are specified with the @code{pre} option, this option takes a
preset name as input. FFmpeg searches for a file named @var{preset_name}.avpreset in
the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg})
in that order. For example, if the argument is @code{libx264-max}, it will
search for the file @file{libx264-max.avpreset}.
@section Video and Audio grabbing
If you specify the input format and device then ffmpeg can grab video
@ -1382,14 +1204,14 @@ standard mixer.
Grab the X11 display with ffmpeg via
@example
ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
@end example
0.0 is display.screen number of your X11 server, same as
the DISPLAY environment variable.
@example
ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
@end example
0.0 is display.screen number of your X11 server, same as the DISPLAY environment
@ -1504,7 +1326,7 @@ combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
@example
ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example
The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
@ -1519,18 +1341,18 @@ image2-specific @code{-pattern_type glob} option.
For example, for creating a video from filenames matching the glob pattern
@code{foo-*.jpeg}:
@example
ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
@end example
@item
You can put many streams of the same type in the output:
@example
ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
@end example
The resulting output file @file{test12.nut} will contain the first four streams
from the input files in reverse order.
The resulting output file @file{test12.avi} will contain first four streams from
the input file in reverse order.
@item
To force CBR video output:

View File

@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffplay Documentation
@titlepage
@ -25,7 +24,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@ -38,26 +37,14 @@ Force displayed height.
Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
@item -fs
Start in fullscreen mode.
@item -an
Disable audio.
@item -vn
Disable video.
@item -sn
Disable subtitles.
@item -ss @var{pos}
Seek to @var{pos}. Note that in most formats it is not possible to seek
exactly, so @command{ffplay} will seek to the nearest seek point to
@var{pos}.
@var{pos} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
Seek to a given position in seconds.
@item -t @var{duration}
Play @var{duration} seconds of audio/video.
@var{duration} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
play <duration> seconds of audio/video
@item -bytes
Seek by bytes.
@item -nodisp
@ -97,9 +84,6 @@ output. In the filtergraph, the input is associated to the label
ffmpeg-filters manual for more information about the filtergraph
syntax.
You can specify this parameter multiple times and cycle through the specified
filtergraphs along with the show modes by pressing the key @key{w}.
@item -af @var{filtergraph}
@var{filtergraph} is a description of the filtergraph to apply to
the input audio.
@ -122,10 +106,15 @@ duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
@item -bug
Work around bugs.
@item -fast
Non-spec-compliant optimizations.
@item -genpts
Generate pts.
@item -rtp_tcp
Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
if you are streaming with the RTSP protocol.
@item -sync @var{type}
Set the master clock to audio (@code{type=audio}), video
(@code{type=video}) or external (@code{type=ext}). Default is audio. The
@ -133,20 +122,23 @@ master clock is used to control audio-video synchronization. Most media
players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
@item -ast @var{audio_stream_specifier}
Select the desired audio stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" audio stream is selected in the program of the
already selected video stream.
@item -vst @var{video_stream_specifier}
Select the desired video stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" video stream is selected.
@item -sst @var{subtitle_stream_specifier}
Select the desired subtitle stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" subtitle stream is selected in the program of the
already selected video or audio stream.
@item -threads @var{count}
Set the thread count.
@item -ast @var{audio_stream_number}
Select the desired audio stream number, counting from 0. The number
refers to the list of all the input audio streams. If it is greater
than the number of audio streams minus one, then the last one is
selected, if it is negative the audio playback is disabled.
@item -vst @var{video_stream_number}
Select the desired video stream number, counting from 0. The number
refers to the list of all the input video streams. If it is greater
than the number of video streams minus one, then the last one is
selected, if it is negative the video playback is disabled.
@item -sst @var{subtitle_stream_number}
Select the desired subtitle stream number, counting from 0. The number
refers to the list of all the input subtitle streams. If it is greater
than the number of subtitle streams minus one, then the last one is
selected, if it is negative the subtitle rendering is disabled.
@item -autoexit
Exit when video is done playing.
@item -exitonkeydown
@ -167,22 +159,6 @@ Force a specific video decoder.
@item -scodec @var{codec_name}
Force a specific subtitle decoder.
@item -autorotate
Automatically rotate the video according to file metadata. Enabled by
default, use @option{-noautorotate} to disable it.
@item -framedrop
Drop video frames if video is out of sync. Enabled by default if the master
clock is not set to video. Use this option to enable frame dropping for all
master clock sources, use @option{-noframedrop} to disable it.
@item -infbuf
Do not limit the input buffer size, read as much data as possible from the
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.
@end table
@section While playing
@ -197,35 +173,17 @@ Toggle full screen.
@item p, SPC
Pause.
@item m
Toggle mute.
@item 9, 0
Decrease and increase volume respectively.
@item /, *
Decrease and increase volume respectively.
@item a
Cycle audio channel in the current program.
Cycle audio channel.
@item v
Cycle video channel.
@item t
Cycle subtitle channel in the current program.
@item c
Cycle program.
Cycle subtitle channel.
@item w
Cycle video filters or show modes.
@item s
Step to the next frame.
Pause if the stream is not already paused, step to the next video
frame, and pause.
Show audio waves.
@item left/right
Seek backward/forward 10 seconds.
@ -234,23 +192,17 @@ Seek backward/forward 10 seconds.
Seek backward/forward 1 minute.
@item page down/page up
Seek to the previous/next chapter.
or if there are no chapters
Seek backward/forward 10 minutes.
@item right mouse click
@item mouse click
Seek to percentage in file corresponding to fraction of width.
@item left mouse double-click
Toggle full screen.
@end table
@c man end
@include config.texi
@ifset config-all
@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset

View File

@ -1,5 +1,4 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffprobe Documentation
@titlepage
@ -45,15 +44,14 @@ name (which may be shared by other sections), and an unique
name. See the output of @option{sections}.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
section.
and printed in the corresponding "FORMAT" or "STREAM" section.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@include avtools-common-opts.texi
@section Main options
@ -114,16 +112,12 @@ ffprobe -show_packets -select_streams v:1 INPUT
@end example
@item -show_data
Show payload data, as a hexadecimal and ASCII dump. Coupled with
Show payload data, as an hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.
The dump is printed as the "data" field. It may contain newlines.
@item -show_data_hash @var{algorithm}
Show a hash of payload data, for packets with @option{-show_packets} and for
codec extradata with @option{-show_streams}.
@item -show_error
Show information about the error found when trying to probe the input.
@ -185,7 +179,7 @@ format : stream=codec_type
To show all the tags in the stream and format sections:
@example
stream_tags : format_tags
format_tags : format_tags
@end example
To show only the @code{title} tag (if available) in the stream
@ -202,11 +196,11 @@ The information for each single packet is printed within a dedicated
section with name "PACKET".
@item -show_frames
Show information about each frame and subtitle contained in the input
multimedia stream.
Show information about each frame contained in the input multimedia
stream.
The information for each single frame is printed within a dedicated
section with name "FRAME" or "SUBTITLE".
section with name "FRAME".
@item -show_streams
Show information about each media stream contained in the input
@ -215,13 +209,6 @@ multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM".
@item -show_programs
Show information about programs and their streams contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "PROGRAM_STREAM".
@item -show_chapters
Show information about chapters stored in the format.
@ -235,70 +222,6 @@ corresponding stream section.
Count the number of packets per stream and report it in the
corresponding stream section.
@item -read_intervals @var{read_intervals}
Read only the specified intervals. @var{read_intervals} must be a
sequence of interval specifications separated by ",".
@command{ffprobe} will seek to the interval starting point, and will
continue reading from that.
Each interval is specified by two optional parts, separated by "%".
The first part specifies the interval start position. It is
interpreted as an abolute position, or as a relative offset from the
current position if it is preceded by the "+" character. If this first
part is not specified, no seeking will be performed when reading this
interval.
The second part specifies the interval end position. It is interpreted
as an absolute position, or as a relative offset from the current
position if it is preceded by the "+" character. If the offset
specification starts with "#", it is interpreted as the number of
packets to read (not including the flushing packets) from the interval
start. If no second part is specified, the program will read until the
end of the input.
Note that seeking is not accurate, thus the actual interval start
point may be different from the specified position. Also, when an
interval duration is specified, the absolute end time will be computed
by adding the duration to the interval start point found by seeking
the file, rather than to the specified start value.
The formal syntax is given by:
@example
@var{INTERVAL} ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
@end example
A few examples follow.
@itemize
@item
Seek to time 10, read packets until 20 seconds after the found seek
point, then seek to position @code{01:30} (1 minute and thirty
seconds) and read packets until position @code{01:45}.
@example
10%+20,01:30%01:45
@end example
@item
Read only 42 packets after seeking to position @code{01:23}:
@example
01:23%+#42
@end example
@item
Read only the first 20 seconds from the start:
@example
%+20
@end example
@item
Read from the start until position @code{02:30}:
@example
%02:30
@end example
@end itemize
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
@ -322,12 +245,6 @@ Show information related to program and library versions. This is the
equivalent of setting both @option{-show_program_version} and
@option{-show_library_versions} options.
@item -show_pixel_formats
Show information about all pixel formats supported by FFmpeg.
Pixel format information for each format is printed within a section
with name "PIXEL_FORMAT".
@item -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@ -348,39 +265,6 @@ A writer may accept one or more arguments, which specify the options
to adopt. The options are specified as a list of @var{key}=@var{value}
pairs, separated by ":".
All writers support the following options:
@table @option
@item string_validation, sv
Set string validation mode.
The following values are accepted.
@table @samp
@item fail
The writer will fail immediately in case an invalid string (UTF-8)
sequence or code point is found in the input. This is especially
useful to validate input metadata.
@item ignore
Any validation error will be ignored. This will result in possibly
broken output, especially with the json or xml writer.
@item replace
The writer will substitute invalid UTF-8 sequences or code points with
the string specified with the @option{string_validation_replacement}.
@end table
Default value is @samp{replace}.
@item string_validation_replacement, svr
Set replacement string to use in case @option{string_validation} is
set to @samp{replace}.
In case the option is not specified, the writer will assume the empty
string, that is it will remove the invalid sequences from the input
strings.
@end table
A description of the currently available writers follows.
@section default
@ -395,8 +279,8 @@ keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
PROGRAM_STREAM section, and are prefixed by the string "TAG:".
Metadata tags are printed as a line in the corresponding FORMAT or
STREAM section, and are prefixed by the string "TAG:".
A description of the accepted options follows.
@ -447,17 +331,17 @@ writer).
It can assume one of the following values:
@table @option
@item c
Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
character (@samp{\}) or the item separator character @var{SEP} are escaped
using C-like fashioned escaping, so that a newline is converted to the
sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
the separator @var{SEP} is converted to @samp{\@var{SEP}}.
Perform C-like escaping. Strings containing a newline ('\n'), carriage
return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
escaping, so that a newline is converted to the sequence "\n", a
carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
converted to "\@var{SEP}".
@item csv
Perform CSV-like escaping, as described in RFC4180. Strings
containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
(@samp{"}), or @var{SEP} are enclosed in double-quotes.
containing a newline ('\n'), a carriage return ('\r'), a double quote
('"'), or @var{SEP} are enclosed in double-quotes.
@item none
Perform no escaping.
@ -485,7 +369,7 @@ The description of the accepted options follows.
Separator character used to separate the chapter, the section name, IDs and
potential tags in the printed field key.
Default value is @samp{.}.
Default value is '.'.
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
@ -507,22 +391,21 @@ The following conventions are adopted:
@item
all key and values are UTF-8
@item
@samp{.} is the subgroup separator
'.' is the subgroup separator
@item
newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
escaped
newline, '\t', '\f', '\b' and the following characters are escaped
@item
@samp{\} is the escape character
'\' is the escape character
@item
@samp{#} is the comment indicator
'#' is the comment indicator
@item
@samp{=} is the key/value separator
'=' is the key/value separator
@item
@samp{:} is not used but usually parsed as key/value separator
':' is not used but usually parsed as key/value separator
@end itemize
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by @samp{:}.
separated by ":".
The description of the accepted options follows.
@ -611,7 +494,6 @@ DV, GXF and AVI timecodes are available in format metadata
@include config.texi
@ifset config-all
@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset

View File

@ -8,17 +8,14 @@
<xsd:complexType name="ffprobeType">
<xsd:sequence>
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="pixel_formats" type="ffprobe:pixelFormatsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
</xsd:complexType>
@ -30,29 +27,11 @@
<xsd:complexType name="framesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsAndFramesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
@ -67,27 +46,10 @@
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="packetSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="stream_index" type="xsd:int" />
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
@ -95,8 +57,6 @@
<xsd:attribute name="pkt_pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_dts" type="xsd:long" />
<xsd:attribute name="pkt_dts_time" type="xsd:float"/>
<xsd:attribute name="best_effort_timestamp" type="xsd:long" />
<xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
<xsd:attribute name="pkt_duration" type="xsd:long" />
<xsd:attribute name="pkt_duration_time" type="xsd:float"/>
<xsd:attribute name="pkt_pos" type="xsd:long" />
@ -121,39 +81,12 @@
<xsd:attribute name="repeat_pict" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
<xsd:attribute name="timecode" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="format" type="xsd:int" />
<xsd:attribute name="start_display_time" type="xsd:int" />
<xsd:attribute name="end_display_time" type="xsd:int" />
<xsd:attribute name="num_rects" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="streamsType">
<xsd:sequence>
<xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="programsType">
<xsd:sequence>
<xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamDispositionType">
<xsd:attribute name="default" type="xsd:int" use="required" />
<xsd:attribute name="dub" type="xsd:int" use="required" />
@ -170,9 +103,8 @@
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
@ -184,31 +116,21 @@
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<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="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<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="timecode" type="xsd:string"/>
<xsd:attribute name="refs" type="xsd:int"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="channel_layout" type="xsd:string"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
@ -220,30 +142,11 @@
<xsd:attribute name="duration_ts" type="xsd:long"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="max_bit_rate" type="xsd:int"/>
<xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="programType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="program_id" type="xsd:int" use="required"/>
<xsd:attribute name="program_num" type="xsd:int" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="end_time" type="xsd:float"/>
<xsd:attribute name="end_pts" type="xsd:long"/>
<xsd:attribute name="pmt_pid" type="xsd:int" use="required"/>
<xsd:attribute name="pcr_pid" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@ -251,14 +154,12 @@
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="nb_programs" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="size" type="xsd:long"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
<xsd:attribute name="probe_score" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="tagType">
@ -274,9 +175,10 @@
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string"/>
<xsd:attribute name="build_time" type="xsd:string"/>
<xsd:attribute name="compiler_ident" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string" use="required"/>
<xsd:attribute name="build_time" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_type" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
@ -313,45 +215,4 @@
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatFlagsType">
<xsd:attribute name="big_endian" type="xsd:int" use="required"/>
<xsd:attribute name="palette" type="xsd:int" use="required"/>
<xsd:attribute name="bitstream" type="xsd:int" use="required"/>
<xsd:attribute name="hwaccel" type="xsd:int" use="required"/>
<xsd:attribute name="planar" type="xsd:int" use="required"/>
<xsd:attribute name="rgb" type="xsd:int" use="required"/>
<xsd:attribute name="pseudopal" type="xsd:int" use="required"/>
<xsd:attribute name="alpha" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentType">
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="bit_depth" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentsType">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatType">
<xsd:sequence>
<xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="nb_components" type="xsd:int" use="required"/>
<xsd:attribute name="log2_chroma_w" type="xsd:int"/>
<xsd:attribute name="log2_chroma_h" type="xsd:int"/>
<xsd:attribute name="bits_per_pixel" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatsType">
<xsd:sequence>
<xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

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