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306 Commits

Author SHA1 Message Date
Michael Niedermayer
e7873dfcca update for FFmpeg 2.1.5
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 17:46:48 +02:00
Michael Niedermayer
9790ed63fc 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 86a9370e2b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
0e3a1d60d3 cavsdec: fix qp fixed slice handling
Fixes Ticket3400

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0accf24b15)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
a45a665991 avformat/cavsvideodec: Fix probing when the file extension is avs
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 52e563bb2f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
c2c922460a 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 6e6bd5481c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
d192ff0ece 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 74760883fc)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
4d1346d46a avcodec/aic: fix quantization table permutation
Fixes Ticket3700

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0a2004b6d1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
66a9c50d42 avcodec/mss4: Fix () in MKVAL() macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cf7ff0146c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
3065919b65 avcodec/mss34dsp: fix () in SOP* macros
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6e720c5c81)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
f1feb7e388 avcodec/mlpdec: fix () in MSB_MASK() macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fa160af08b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
ccbcb17960 avcodec/ivi_dsp: add some missing () to macros
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f276bf303c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
a71b4209c2 avcodec/h264: fix () in macros
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit af62b42736)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
60ed439334 avcodec/golomb-test: fix () in EXTEND() macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 97e6b5ee3a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
76838fb171 avcodec/aac: fix () in IS_CODEBOOK_UNSIGNED macro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fa915d4193)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
e3b6e23a0f avcodec/libvorbisenc: dont add the duration to AV_NOPTS_VALUE
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 19e66c7232)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
6a51bd7360 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 6310eb8010)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
2572372f90 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 4a30f08505)
(cherry picked from commit 9c9fc79d9237d28e33161cb2e75082d8ad232b2e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
2a543e58ec avfilter/graphdump: Fix pointer to local outside scope
Fixes CID1194435
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 18af0ce62d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
aaf6ddfa4d avcodec/mjpegdec: Fix undefined shift
Fixes CID1194388

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b432960528)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
2b45eab4c9 ffmpeg_filter: fix pointer to local outside scope
Fixes CID1206652
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 09b16619d3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
d89e5b20f2 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 fc145e576a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:09 +02:00
Michael Niedermayer
ac182d6d8b 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 dc6a17cf74)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
9cde200ba7 avfilter/filtfmts: Support dynamically allocated in/outputs
Fixes crash
Fixes Ticket3468

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 59c7615d58)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
df208d6ab9 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 cb53beb81a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
4eb0b6c590 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 73734282e0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
90a384cde1 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 5a8ef3c66b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
af109ff125 avcodec/g723_1: add assert to help static code analyzers
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1457f3fd90)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
ec2ce83913 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 b8598f6ce6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
1d91251370 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 ff6fa0b4b9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
861e453996 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 256f530603)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
5228655438 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 14fa7fc6a8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
d2e3afbbd1 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 46d5625f44)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
c33770e74b avutil/lzo: add asserts to be double sure against overflows
These asserts cannot fail since d6af26c55c

Based-on: ccda51b14c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cf2b7c01f8)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Michael Niedermayer
9c358c6e3b 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: ccda51b14c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d6af26c55c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 16:51:08 +02:00
Anthoine Bourgeois
26d72b3df0 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 d8ab7f31dd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
6729193886 swscale/x86/swscale_template: loose hardcoded dstw_offset
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f6759d9ad4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
9d8e231459 swresample/resample: use av_malloc_array() where appropriate
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5027f39712)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
15871a90aa swresample/dither: use av_malloc_array()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a5290cb1ac)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
37f854f5b9 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 f9158b01d0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
9c83914a1f 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 754f84663e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
861732246f avcodec/wma: use av_freep(), do not leave stale pointers in memory
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d167faafe9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Lukasz Marek
46b64d9818 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 3aac5fcfa9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
c5085a0301 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 8710ee11d7)

Conflicts:

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

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
4c9c76af85 avfilter/vf_pullup: fix gray8
Fixes segfault
Fixes Ticket3469

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e818ee090a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
e4f9a5efa1 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 64b79141bd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
61e5001b26 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 289b149cec)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
380587cc34 avcodec/x86/mpegvideoenc_template: fix integer overflow
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
998b5ac3a0 dox/scaler:fix bicubiclin typo
See Ticket3486

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 575b957758)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
b8f6069bf3 swscale/swscale: fix integer overflow
Should fix fate failure with clang ftrapv

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c9c0451224)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:07 +02:00
Michael Niedermayer
2b5b21dbac 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 fc567ac49e)

Conflicts:

	libavcodec/h264.h
(cherry picked from commit 618d062bd553a7d7fad194d4236913f2b0a0251e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:06 +02:00
Michael Niedermayer
4201c74545 avformat/mp3enc: use av_copy_packet()
Fixes double free
Fixes Ticket3476

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d003a0cd2e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:06 +02:00
Michael Niedermayer
946810d1da 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 9dca02ee54)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-06-23 15:10:06 +02:00
Carl Eugen Hoyos
f3802aa325 Show duration for large asf files as written in the file header.
Fixes ticket #3428.
(cherry picked from commit e3fd263f0b)
2014-06-21 22:11:00 +02:00
Martin Storsjö
c5606c04cf adpcm: Write the proper predictor in trellis mode in IMA QT
The actual predictor value, set by the trellis code, never
was written back into the variable that was written into
the block header. This was accidentally removed in b304244b.

This significantly improves the audio quality of the trellis
case, which was plain broken since b304244b.

Encoding IMA QT with trellis still actually gives a slightly
worse quality than without trellis, since the trellis encoder
doesn't use the exact same way of rounding as in
adpcm_ima_qt_compress_sample and adpcm_ima_qt_expand_nibble.

Fixes part of Ticket3701

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fa8f060b75)

Conflicts:
	tests/ref/acodec/adpcm-ima_qt-trellis
2014-06-10 00:13:28 +02:00
Martin Storsjö
39256e54d1 adpcm: Fix trellis encoding of IMA QT
This was broken in 095be4fb - samples+ch (for the previous
non-planar case) equals &samples_p[ch][0]. The confusion
probably stemmed from the IMA WAV case where it originally
was &samples[avctx->channels + ch], which was correctly
changed into &samples_p[ch][1].

Fixes part of Ticket3701

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a32765c425)
2014-06-10 00:13:21 +02:00
Michael Niedermayer
992cdbc56a avcodec/mjpegdec: Improve intel jpeg flip heuristic
Fixes Ticket3698

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0545ef7116)
2014-06-10 00:12:37 +02:00
Michael Niedermayer
e8bb67c2a9 avcodec/h264: in the absence of recovery points, be more tolerant on accepting plain I frames
Fixes: Ticket3652
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 19c9d1e8e7)
2014-06-10 00:10:40 +02:00
Carl Eugen Hoyos
60e2f9bf8b avcodec/utvideodec: Increase vlc len
Fixes a regression since fb3e380 similar to ticket #2661,
reported by fluffrabbit at aol dot com.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 673716c54b)
2014-06-10 00:10:02 +02:00
Carl Eugen Hoyos
bdba204bd0 Do not limit blocksize when reading PCM from aiff.
Fixes ticket #3695.
(cherry picked from commit 763e714442)
2014-06-10 00:09:50 +02:00
Michael Niedermayer
be4d4a61b5 avformat/avidec: allow rounding errors between scale/rate and timebase
Fixes Ticket3670

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 571ab8344a)
2014-06-10 00:09:03 +02:00
Anshul
83f58778cb ffprobe: fix crash happening because of new streams occuring
Fix trac ticket #3603.

Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
(cherry picked from commit 73a6063314)

Conflicts:
	ffprobe.c
2014-05-21 22:58:24 +02:00
Carl Eugen Hoyos
c5a2c26c55 Use Matroska document version 2 for WebM files if possible.
Google's plugin for the Internet Explorer refuses to play
files with another document version.

Fixes ticket #3583.
(cherry picked from commit ab21acecc7)
2014-05-12 23:21:06 +02:00
Michael Niedermayer
03c5ccb4ee 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 421b21ca8a)
2014-05-01 18:28:37 +02:00
Michael Niedermayer
172a5cab87 avcodec/vorbisdec: try to workaround libvorbisenc bug
Fixes Ticket3590

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0a266cb55a)
2014-05-01 18:27:42 +02:00
Michael Niedermayer
8d6de87709 swresample: fix AV_CH_LAYOUT_STEREO_DOWNMIX input
Fixes Ticket 3542

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 291d464161)
2014-04-24 09:02:34 +02:00
Stefano Sabatini
af04f0d1ce ffprobe: fix scaling of vali in value_string() in case -prefix is selected
Fix trac ticket #3523.
(cherry picked from commit 1ba59b1cbe)
2014-04-23 17:17:44 +02:00
Michael Niedermayer
116b959565 avformat/avidec: Speed up keyframe detection code
Fixes Ticket3531

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 57fb570908)
2014-04-23 17:16:36 +02:00
Peter Ross
7269ab10c5 ff_id3v2_read: add option to limit ID3 magic number search
Several chunked formats (AIFF, IFF,DSF) store ID3 metadata within an 'ID3 '
chunk tag. If such chunks are stored sequentially, it is possible for the
ID3v2 parser to confuse the chunk tag for the ID3 magic number. e.g.

[1st chunk tag ('ID3 ') | chunk size] [ID3 magic number | metadata ...]
[2nd chunk tag ('ID3 ') | chunk size] [ID3 magic number | metadata ...]

Fixes ticket #3530.

Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5331773cc3)

Conflicts:
	libavformat/dsfdec.c
	libavformat/id3v2.c
2014-04-19 13:34:28 +02:00
Peter Ross
caeed48982 ff_id3v2_free_extra_meta: set the pointer pointing to extra_meta to NULL
Fixes ticket #3530.

Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c94305ae23)
2014-04-19 13:31:12 +02:00
Michael Niedermayer
353f79defc avformat/matroskadec: support SVQ3 as generated by mkvtoolnix-6.6.0
Fixes part of Ticket3256

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5800b08572)
2014-04-16 19:42:46 +02:00
Michael Niedermayer
92d336670f avformat/matroskaenc: wrap V_QUICKTIME codec private in something that looks like its part of quicktime stsd
This is needed for matroska spec compliance
Fixes playback of SVQ3 in matroska with vlc
Fixes Ticket 3256

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8456bd2c0f)
2014-04-16 19:41:45 +02:00
Michael Niedermayer
3c880d2765 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 a0911b0597)
2014-03-24 23:32:11 +01:00
Hendrik Leppkes
63dcf7c77c 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 019247bdc3)
2014-03-24 23:32:05 +01:00
Michael Niedermayer
610a064607 avcodec/h263dec: Fix use of uninitialized memory from the bitstream buffer
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f07cebcd91)

Conflicts:

	libavcodec/h263dec.c
2014-03-18 18:26:02 +01:00
Michael Niedermayer
65c915d0c3 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 8ab8070784)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-03-18 18:26:01 +01:00
Michael Niedermayer
8c8db86c0d swscale/x86/swscale: fix missing xmm clobbers in yuv2yuvX_sse3()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6c47a4e972)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-03-18 18:26:01 +01:00
Michael Niedermayer
44799d269f avutil/timestamp: Warn about missing __STDC_FORMAT_MACROS for C++ use
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8b02dfd37c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-03-18 18:26:01 +01:00
Michael Niedermayer
3c8a12be2f avcodec/g2meet: fix error returns
Fixes out of array accesses

This should not affect any release

Fixes: 8ab69af9e5a7a7e20fe04cdd25c0d6e7-asan_heap-oob_e72b82_5505_cov_2278389485_g2m4.wmv
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6b53c1aa82)

Conflicts:

	libavcodec/g2meet.c
2014-03-18 18:26:01 +01:00
Michael Niedermayer
0c5f166e6f avcodec/utvideoenc: fix slice_bits size
Fixes assertion failure

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0a8c90202b)

Conflicts:

	libavcodec/utvideoenc.c
2014-03-18 18:26:01 +01:00
Timothy Gu
5a45c644e8 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 baa650cc79)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-03-18 18:26:01 +01:00
Peter Ross
02d46127f5 avcodec/adpcm: squelch 'mismatch in coded sample count' warning for AV_CODEC_ID_ADPCM_EA_R2/3
These ADPCM codecs include a per-frame flag that enables a raw 16-bit mode. Therefore
the the number of samples returned by get_nb_samples() is only ever approximate.

Fixes ticket #3460.

Signed-off-by: Peter Ross <pross@xvid.org>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7380201451)
2014-03-17 15:53:46 +01:00
Peter Ross
2e99fab15c avcodec/adpcm: ADPCM_IMA_DK3 packets are padded to 16-bit packet boundary
Fixes ticket #3461.

Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d1bb17940d)
2014-03-17 15:53:37 +01:00
Michael Niedermayer
9098da0329 avcodec/h264: be more tolerant on what pixel format changes trigger reinits
Fixes Ticket3260

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8e92ff2546)
2014-03-17 15:53:21 +01:00
Michael Niedermayer
db53b2b2c0 ffmpeg_opt: check that a subtitle encoder is available before auto mapping streams
Fixes Ticket3470

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 25bcf24d4d)
2014-03-17 15:52:48 +01:00
Michael Niedermayer
c7a854af19 mvformat/movenc: fix IMX
fixes Ticket3351

Tested-by: carl
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 72d44f1583)
2014-03-13 07:41:32 +01:00
Michael Niedermayer
3e73bea05f avformat/avidec: Check required demuxing buffer sizes in guess_ni_flag()
Fixes Ticket3421

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5d75730c58)
2014-03-13 07:31:59 +01:00
Michael Niedermayer
3712b398e9 avidec: calculate missing bitrates from index
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 01000064c9)
2014-03-13 07:31:56 +01:00
Michael Niedermayer
4362a222e2 avformat/flvdec: discard inconsistent timestamps
Fixes Ticket3425

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dbc3e1109c)
2014-03-12 23:54:56 +01:00
Michael Niedermayer
a9753049b5 avcodec/vorbisdec: use the stored previous window type only when the actual previous is not known
Fixes Ticket3432

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5171ae781a)
2014-03-12 23:53:51 +01:00
Nicolas George
910fd77967 lavfi/af_atempo: clear references before returning error.
Once the frame has been given to ff_filter_frame(), it can
no longer be used, even on error.

Fix trac ticket #3430.
(cherry picked from commit bc6901c949)
2014-03-08 15:19:11 +01:00
Michael Niedermayer
d87ac93bca avformat/oggparsevorbis: dont use invalid granules
Fixes Ticket3437

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 12b97dd375)
2014-03-06 09:34:09 +01:00
Anton Khirnov
96603d5753 pthread_frame: flush all threads on flush, not just the first one
avcodec_flush_buffers() must release all internally held references
according to its documentation, for which all the threads need to be
flushed.

Bug-Id: vlc/9665
(cherry picked from commit d1f9563d50)

Conflicts:
	libavcodec/pthread_frame.c
2014-03-04 01:28:29 +01:00
Michael Niedermayer
c7c724056e avcodec/h264: clear chroma planes when flags gray is used
Fixes Ticket3397
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 72e6913140)
2014-02-24 09:45:48 +01:00
Michael Niedermayer
d3139c9733 update for 2.1.4
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 17:48:16 +01:00
Michael Niedermayer
0909b8acf8 avcodec/hevc: Simplify get_qPy_pred()
Fixes use of uninitialized memory
Fixes: 93728afd9aa074ba14a09bfd93a632fd-asan_static-oob_124a17d_1445_cov_1021181966_DBLK_D_VIXS_1.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 64278039e5)

Conflicts:

	libavcodec/hevc_filter.c
2014-02-23 16:44:37 +01:00
Michael Niedermayer
2368d08e70 Merge commit 'e22ebd04bcab7f86548794556c28ecca46d9c2ac'
* commit 'e22ebd04bcab7f86548794556c28ecca46d9c2ac':
  hevc: Bound check cu_qp_delta

Conflicts:
	libavcodec/hevc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a69dd1163b)

Conflicts:

	libavcodec/hevc.c
2014-02-23 16:42:21 +01:00
Michael Niedermayer
ea7ccf3748 avcodec/mpeg4videodec: Check for bitstream overread in decode_vol_header()
Fixes out of array read
Fixes: 08e48e9daae7d8f8ab6dbe3919e797e5-asan_heap-oob_157461c_5295_cov_1266798650_firefing.mpg
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3edc3b1595)

Conflicts:

	libavcodec/mpeg4videodec.c
2014-02-23 16:36:01 +01:00
Michael Niedermayer
846a9c67ff avcodec/h264: use subsample factors of the used pixel format
Fixes out of array read
Fixes: 1cb91c36c4e55463f14aacb9bdf55b38-asan_heap-oob_106cbce_5617_cov_11212800_h264_mmx_chroma_intra_lf.mp4
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8c55ff3933)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 15:04:07 +01:00
Michael Niedermayer
c8d363a359 avformat/bink: Check return value of av_add_index_entry()
Fixes null pointer dereference
Fixes: cdbf15cbd0a27cee958dd0b8800e452e-signal_sigsegv_737991_2083_cov_317652874_LBSTART.BIK
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c57fc97e95)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 15:03:03 +01:00
Michael Niedermayer
d0d441b350 avcodec/h264: more completely check the loop filter parameters
Fixes out of array read
Fixes: caa65cc01655505705129b677189f036-signal_sigsegv_fdcc43_2681_cov_3043376737_PPH422I5_Panasonic_A.264
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 91253839e1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 15:02:14 +01:00
Michael Niedermayer
6341a7006d avcodec/alsdec: check predictor order against block length
Fixes out of array access
Fixes: abd3c041acbcb816be113455d138166b-asan_heap-oob_b11634_3707_cov_1707137151_als_05_2ch48k16b.mp4
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 18f94df8af)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 15:00:02 +01:00
Michael Niedermayer
7034e808f6 avcodec/hevc_ps: Use get_bits_long() in decode_vui()
Fix assertion failure
Fixes: a225222ef88a0f5b1e93e1d0432debc3-asan_static-oob_124a17d_1448_cov_77608227_DBLK_E_VIXS_1.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b818637b84)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:58:13 +01:00
Michael Niedermayer
ce5d9a2b4b avcodec/hevc: make check for previous slice segment tighter
This ensures the previous one is matching the curent and not just any

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1a3ed056c5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:53:59 +01:00
Michael Niedermayer
aa672f5e6a avcodec/hevc: clear tab_slice_address of ctb on error.
This allows us to detect which areas have failed to decode

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a18f111582)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:53:23 +01:00
Michael Niedermayer
b959e6393e avcodec/hevc: hls_decode_entry: check that the previous slice segment is available before decoding the next
Fixes use of uninitialized memory
Fixes out of array read
Fixes assertion failure
Fixes part of cb307d24befbd109c6f054008d6777b5/asan_static-oob_124a175_1445_cov_2355279992_DBLK_D_VIXS_1.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6ef57f4d9a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:51:46 +01:00
Michael Niedermayer
69f99f80d1 avcodec/hevc: clear tab_slice_address in hevc_frame_start()
Fixes inconsistencies
Fixes use of uninitilaized memory
Fixes part of  cb307d24befbd109c6f054008d6777b5/asan_static-oob_124a175_1445_cov_2355279992_DBLK_D_VIXS_1.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 56985d26d7)

Conflicts:

	libavcodec/hevc.c
2014-02-23 14:51:22 +01:00
Michael Niedermayer
e7b7e69416 avcodec/h264: update current_sps & sps->new only after the whole slice header decoder and init code finished
This avoids them being cleared before the full initialization finished

Fixes out of array read
Fixes: asan_heap-oob_f0c5e6_7071_cov_1605985132_mov_h264_aac__Demo_FlagOfOurFathers.mov
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8a3b85f3a7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:49:47 +01:00
Michael Niedermayer
9330bcff9b avcodec/h264: Disallow pps_id changing between slices
Such changes are forbidden in H.264 and lead to race conditions

Fixes out of array read
Fixes: signal_sigsegv_f9796a_1613_cov_3114610371_FM1_BT_B.h264
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e708424b70)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:49:37 +01:00
Michael Niedermayer
9fb364babd avcodec/aacdec: Fix pulse position checks in decode_pulses()
Fixes out of array read
Fixes: asan_static-oob_1efed25_1887_cov_2013541199_HeyYa_RA10_AAC_192K_30s.rm
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6e42ccb9db)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:49:29 +01:00
Michael Niedermayer
d79419d0f9 avcodec/hevc: propagate error code from hls_coding_quadtree()
Fixes use of uninitialized memory
Fixes out of array read
Fixes: asan_static-oob_123cee5_2630_cov_1869071233_PICSIZE_A_Bossen_1.bin
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 96c4ba2392)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:48:30 +01:00
Michael Niedermayer
ebc490e744 avcodec/tiff: reset geotag_count in free_geotags()
Fixes null pointer dereference
Fixes: signal_sigsegv_19d922e_3688_cov_1577641655_aletrek_tiff.mov
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a744064c41)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:48:07 +01:00
Michael Niedermayer
ab1c7113f9 avcodec/vc1: Check bfraction_lut_index
Fixes: out of array read
Fixes: asan_static-oob_1b40507_2849_SA10143.vc1
Fixes: asan_static-oob_1b40a15_2849_cov_1182297305_SA10143.vc1
Fixes: asan_static-oob_1b40f15_2849_cov_2159513432_SA10143.vc1
Fixes: asan_static-oob_1b40f15_2849_cov_3230311510_SA10143.vc1
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dcf5bfbdb6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:47:46 +01:00
Michael Niedermayer
10a30e4de5 avcodec/vc1: factor read_bfraction() out
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 388b4cf86e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:47:41 +01:00
Michael Niedermayer
9368b91834 avcodec/vc1dec: field pictures with direct mode MBs, followed by frame pictures are not supported
This case could occur when cuting and concatenating bitstreams

Fixes out of array read
Fixes: asan_heap-oob_1b33fdd_2849_cov_478905890_SA10143.vc1
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 85d51d8e32)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:43:52 +01:00
Michael Niedermayer
f22e88c177 avcodec/mjpegdec: pass into ff_mjpeg_decode_sos() and check bitmask size
Fixes: heap array overread
Fixes: asan_heap-oob_149b2bc_6577_m1.mxg
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2884688bd5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:43:28 +01:00
Michael Niedermayer
e266fcf083 avformat/flac_picture: clear padding area
aviod use of uninitialized memory

Fixes: asan_heap-oob_1487fa4_4706_cov_364534849_cover_art.flac
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 13aa82bbbb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:43:18 +01:00
Michael Niedermayer
c9b961748f avformat/flac_picture: allocate buffer padding for picture
Fixes: heap array overread
Fixes: asan_heap-oob_14876d9_4706_cov_815472558_cover_art.flac
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fff2953163)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:43:07 +01:00
Michael Niedermayer
a94f367424 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 61d59703c9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Michael Niedermayer
18eac12c6d 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 d42ec8433c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Michael Niedermayer
4cc18ee5da 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 c919e1ca2e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Michael Niedermayer
f8985cb9d9 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 e5c7229999)

Conflicts:

	libavcodec/utils.c
2014-02-23 14:28:10 +01:00
Michael Niedermayer
656770e2aa avcodec/hevc: make *ps_id unsigned
Fixes integer overflow
Fixes out of array accesses
Fixes 2f65e7dbd02a12f426a423bd7bf880b4-signal_sigsegv_127c952_2793_cov_2517424539_RPLM_A_qualcomm_4.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d1e6602665)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Michael Niedermayer
4a28a3ddc4 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 842b6c14bc)

Conflicts:

	libavformat/mpegtsenc.c
2014-02-23 14:28:10 +01:00
Michael Niedermayer
f91ef98c9d 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 ec9578d54d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Justin Ruggles
a644272a4a samplefmt: avoid integer overflow in av_samples_get_buffer_size()
CC:libav-stable@libav.org
(cherry picked from commit 0e830094ad)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Michael Niedermayer
325feb8e0d 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 b51e935477)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Michael Niedermayer
8c6a976fee 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 f58eab1512)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-23 14:28:10 +01:00
Carl Eugen Hoyos
ac38860ec9 Add decoder dependency to the HEVC parser.
Fixes compilation with --disable-everything --enable-parser=hevc
(cherry picked from commit d4a6133ab8)
2014-02-19 16:59:21 +01:00
Martin Storsjö
d0e0329e9d rtpdec_asf: Copy the need_parsing field from the chained demuxer
This fixes playback of mp3 streams in rtp/asf. This used to work
until 950482bf, but mostly by coincidence.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 2aec9e228c)

Fixes ticket #3223
2014-01-27 15:45:10 +01:00
Alex Sukhanov
9ca79d2849 avformat/matroskadec: Fix start_time
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 251c96a70b)

Fixes ticket #3337.
2014-01-26 16:22:47 +01:00
Lukasz Marek
c5a2a65e1e lavf/libssh: fix seek with whence==SEEK_CUR
Signed-off-by: Lukasz Marek <lukasz.m.luki@gmail.com>
(cherry picked from commit e0d124a920)

Conflicts:

	libavformat/libssh.c
2014-01-22 01:36:31 +01:00
Michael Niedermayer
a8ed3685e1 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 8001e9f7d1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:59:12 +01:00
Michael Niedermayer
32262ca7d7 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 6ba02602aa)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:59:06 +01:00
Michael Niedermayer
5f56e495ae avcodec/apedec: more checks for k
Fixes assertion failure
Fixes part of msan_uninit-mem_7fa0d8c8bd58_8417_sh3.ape
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d5128fce38)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Michael Niedermayer
fedbba5ea0 avformat/rmdec: move packet allocation down
Fixes memleak
Fixes: msan_uninit-mem_7fc5d73327d4_6192_kuerti.ra
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 165f96cd2d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Michael Niedermayer
d35916f6ea avformat/mpegts: check sl.timestamp_len
Fixes: msan_uninit-mem_7ff4404547ba_4883_dmbts.ts
Fixes assertion failure
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e630ca5111)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Michael Niedermayer
94e2673f4e avformat/ape: free packet on avio_read() failure
Fixes memleak
Fixes: msan_uninit-mem_7fcc198b365b_8417_sh3.ape
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 459db51271)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Michael Niedermayer
23ae7bfb4e 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 f1caaa1c61)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Michael Niedermayer
9e8464e81b Merge commit '9eef9eb3014b2ed9c3ff4aac510a9f04edb555cf'
* commit '9eef9eb3014b2ed9c3ff4aac510a9f04edb555cf':
  h264: check that execute_decode_slices() is not called too many times

Conflicts:
	libavcodec/h264.c

The check is replaced by an assert() as the mb index should not ever go out
of bounds.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 64591f8f86)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Michael Niedermayer
c9a8dfa5ae Merge commit '6892d145a0c80249bd61ee7dd31ec851c5076bcd'
* commit '6892d145a0c80249bd61ee7dd31ec851c5076bcd':
  segafilm: fix leaks if reading the header fails

Conflicts:
	libavformat/segafilm.c

See: ca5456db7f
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2a58d5fc0e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Michael Niedermayer
e6299a4cf9 cmdutils: update year
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-21 16:42:32 +01:00
Carl Eugen Hoyos
ee3822af63 Allow decoding of slightly broken Nikon avi files.
Fixes ticket #3330.
(cherry picked from commit f9c2d4d17e)
2014-01-20 22:37:18 +01:00
Nicolas George
fc5261c219 lavfi/dualinput: fix shortest option.
Fix trac ticket #3315.
(cherry picked from commit 2dc5980d61)
2014-01-20 22:28:01 +01:00
Michael Niedermayer
30a94f1159 avcodec/mjpegdec: Dont skip picture allocation if theres no picture allocated
Fixes Ticket 3245
(cherry picked from commit ad8d063f23)
2014-01-20 22:26:00 +01:00
Michael Niedermayer
9d83cff1f1 avcodec/aacdec: Dont fail if channels arent known yet
Fixes Ticket3312

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 676a395ab9)
2014-01-19 14:28:18 +01:00
Michael Niedermayer
756cd1a305 avcodec/mjpegdec: Dont treat the lack of a startcode differently from end of the bitstream
Fixes Ticket3303

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 31e703e899)
2014-01-19 14:27:56 +01:00
Michael Niedermayer
83dc8f044d avcodec/mjpegdec: only run EOI emulation code when there was a scan
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 361e27a3d8)
2014-01-19 14:27:48 +01:00
Michael Niedermayer
2c5c6affb1 avcodec/mjpegdec: update cur_scan also for non-LS jpeg
This should make no difference but the variable will be used in a subsequent commit

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8893f31e20)
2014-01-19 14:27:40 +01:00
Carl Eugen Hoyos
f4e051680e Fix libxvid crash on failing initialisation.
Fixes ticket #3297.
(cherry picked from commit ee3fc8aa86)
2014-01-19 14:26:15 +01:00
Carl Eugen Hoyos
2f4b781e4d Do not read mkv audio bit_depth if bits_per_coded_sample is already set.
This allows decoding broken mkv files containing G.726 audio.
(cherry picked from commit 1132937077)
2014-01-16 00:26:00 +01:00
Carl Eugen Hoyos
0d45e821dc Do not set mkv bit_depth to av_get_bytes_per_sample() for G.726.
The value is wrong and leads to broken files.
(cherry picked from commit 565102dcac)
2014-01-16 00:25:34 +01:00
Michael Niedermayer
eda6effcab Update for 2.1.3
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-15 18:29:16 +01:00
Michael Niedermayer
45900618ae library.mak: only run asm strip if ASMSTRIP flags are set
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e975c147e1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 01:14:43 +01:00
Michael Niedermayer
16e49d85b6 configure: remove code that disables striping in the absence of some flags
This prevents breaking (non asm) striping in that case

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 191454f26e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 01:03:48 +01:00
Michael Niedermayer
bfdfeadf11 build sys: rename STRIPFLAGS to ASMSTRIPFLAGS
This more closely matches the actual use, also we use plain
strip without these flags for striping

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e283c26c35)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 01:03:48 +01:00
Michael Niedermayer
7ae6229b97 Merge commit '0673ede985a6560e7efb86dab1c58fb7f95ce587'
* commit '0673ede985a6560e7efb86dab1c58fb7f95ce587':
  configure: add strip flags checks

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 71b95f2ab6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 01:02:28 +01:00
Michael Niedermayer
2156d9bd7d avformat/utils/av_probe_input_buffer2: fix buffer passed to ffio_rewind_with_probe_data()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 05886c9d4e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 00:32:02 +01:00
Michael Niedermayer
0a055cc62e avformat/utils/av_probe_input_buffer2: fix offset check
The check could fail if avio_read() read less than requested

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8c3b026a0e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 00:32:01 +01:00
Michael Niedermayer
c8d43c22db avformat/utils/av_probe_input_buffer2: Fix pd.buf_size
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6a2064820b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 00:32:00 +01:00
Clément Bœsch
d9b7557732 avcodec/libxavs: 2nd attempt to fix compilation after b18c7c8d.
(cherry picked from commit 260fc0d95b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 00:29:13 +01:00
Clément Bœsch
15d7b7d7cc avcodec/libxavs: attempt to fix compilation after b18c7c8d.
(cherry picked from commit 71cd83e34c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-14 00:29:09 +01:00
Michael Niedermayer
29353dd3f8 Update for 2.1.2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 17:21:42 +01:00
Carl Eugen Hoyos
b336daa952 Fix a crash on oom when decoding hevc.
(cherry picked from commit 5ab1efb9d0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 17:20:19 +01:00
Michael Niedermayer
a0aa5c34a9 avcodec/hevc: Check entry point arrays for malloc failure
Fixes null pointer dereference
Fixes: signal_sigsegv_e1d3b6_2192_DBLK_F_VIXS_2.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 22bfb4be28)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:17:08 +01:00
Luca Barbato
d63476347a hevc: Bound check slice_qp
The T-REC-H.265-2013044 page 79 states they have to be into the range
[-s->sps->qp_bd_offset, 51].

Fixes: asan_stack-oob_eae8e3_9522_WP_MAIN10_B_Toshiba_3.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit aead772b58)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:17:04 +01:00
Luca Barbato
39545c5482 hevc: Reject impossible dependent tile
The tile 0 cannot depend on a previous one.
Prevent an out of array bound load in ff_hevc_cabac_init().

Fixes: asan_heap-oob_e3a924_1630_DBLK_A_MAIN10_VIXS_2.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

Reviewed-by: Guillaume Martres <smarter@ubuntu.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 48a5b15543)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:16:58 +01:00
Guillaume Martres
e43805d401 hevc: remove useless clip in FUNC(sao_band_filter)()
The src buffer should only contain values in the interval
[0, (1 << BIT_DEPTH) - 1]. Since shift = (BIT_DEPTH - 5), src[x] >> shift
must be in the interval [0, 31], so no clip is needed.

This removes the code that was changed in 5856bca360
as the clip that was repositioned in that commit is removed

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b00a8b4d19)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:15:07 +01:00
Guillaume Martres
ea21b7b68c hevc: clip pixels when transquant bypass is used
Fixes: asan_stack-oob_eae8e3_7333_WPP_B_ericsson_MAIN10_2.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind

This is a more proper fix than 5856bca360

The reconstructed picture should always be clipped (see section 8.6.5),
previously we did not clip coding units where
cu_transquant_bypass_flag == 1

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c9fe0caf7a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:12:17 +01:00
Luca Barbato
738a2a04b6 hevc: Clip the pixel before shifting
Prevent an out of array bound read.

Fixes: asan_stack-oob_eae8e3_7333_WPP_B_ericsson_MAIN10_2.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5856bca360)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:12:01 +01:00
Michael Niedermayer
706dca18d0 avcodec/hevc: use av_mallocz() for allocating tab_ipm
Fixes use of uninitialized memory and out of stack array read
Fixes: signal_sigsegv_ecc526_7846_WPP_C_ericsson_MAIN_2.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0999f1613b)

Conflicts:

	libavcodec/hevc.c
2014-01-13 16:11:50 +01:00
Michael Niedermayer
b3c3dc54a5 avcodec/alac: only set *got_frame_ptr when all channels have been decoded
Fixes use of uninitialized memory
Fixes: msan_uninit-mem_7f8b64436530_7895_quicktime_newcodec_applelosslessaudiocodec.m4a
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e11983bda0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:10:34 +01:00
Clément Bœsch
b6af89be6a avformat/pjsdec: dont increase pointer when its already at the end in read_ts()
Fixes use of uninitialized memory
Fixes: msan_uninit-mem_7f91f2de7764_2649_PJS_capability_tester.pjs
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b84a7330af)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:10:30 +01:00
Michael Niedermayer
57824d1ba6 avcodec/wmalosslessdec: shrink output on error so no uninitialized data is returned
Fixes use of uninitialized memory
partly fixes: msan_uninit-mem_7f7834b6a530_6473_luckynight-partial.wma
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6b18a6839b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:10:25 +01:00
Michael Niedermayer
cd753ee36b avcodec/wmalosslessdec: deallocate uninitialized frame on decode_tilehdr() failure
Fixes use of uninitialized memory
partly fixes: msan_uninit-mem_7f7834b6a530_6473_luckynight-partial.wma
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ae3856dcaf)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:10:12 +01:00
Michael Niedermayer
aaaafc300d avcodec/wmalosslessdec: Pass on error code from decode_tilehdr()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 03fff09b32)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:10:09 +01:00
Michael Niedermayer
6da213ce20 avformat/matroskadec: check generic audio deinterleaver sub_packet_size against frame_size
Fixes use of uninitialized memory
Fixes: msan_uninit-mem_7f67d052a530_7517_nosound.mkv
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a1ed1c2193)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:09:26 +01:00
Michael Niedermayer
763a808c96 avformat/flvdec: initialize context before reading from it
Fixes use of uninitialized memory
Fixes: msan_uninit-mem_7f9b8387069e_5377_flv_with_pcm_s16be_audio_track.flv
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 396ddcf22d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:09:22 +01:00
Michael Niedermayer
4f3b0afaa2 avformat/mxfdec: check avio_read(UID) result
Fixes use of uninitialized memory
Fixes: msan_uninit-mem_7fc9ba2fd98e_82_02785736.mxf
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4162ceea93)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:09:15 +01:00
Michael Niedermayer
050d8d727a avformat/rmdec: when reading audio blocks, dont leave holes when reading fails
The fate test is changed because the reference file depends on the use of
non cleared data at the very
end. Alternatively we could upload a new reference file, though that would
then have to be changed every time the handling of a truncated frame changes
or theres a change to error concealment, each time adding a new file ...

Fixes use of uninitialized memory
Fixed: msan_uninit-mem_7f3c02b81363_2787_RLG2_19.rm
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 77d2a1ca59)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-13 16:08:26 +01:00
Michael Niedermayer
6fb0f37def avcodec/h264: fix code that blindly dereferences NULL DPB
Fixes mixed flushing and decoding NULL packets
Found-by: wm4

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d9339ab553)

Conflicts:

	libavcodec/h264.c
2014-01-13 16:06:13 +01:00
Hendrik Leppkes
70028e917c vdpau: restore compatibility with deprecated fields in AVVDPAUContext
Fixes ticket #3133.

Signed-off-by: Hendrik Leppkes <h.leppkes@gmail.com>
Tested-by: EricV
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 23bc1351ad)
2014-01-11 00:21:06 +01:00
Carl Eugen Hoyos
6ce835d77f Fix libopenjpeg colour range adjust for 8<bpp<16.
Fixes ticket #3284.

Reviewed-by: Michael Bradshaw
(cherry picked from commit 8298b54179)
2014-01-10 14:07:36 +01:00
Michael Niedermayer
d45a724192 avcodec/msvideo1enc: fix SKIPS_MAX
Fixes Ticket3270

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fb8f5d0510)
2014-01-09 11:43:21 +01:00
Carl Eugen Hoyos
a48440d857 Use the h264 parser when decoding VSSH in avi.
Fixes ticket #3261 visually.

Analyzed-by: Michael Doilnitsyn
(cherry picked from commit 94cf4f8bac)
2014-01-08 10:19:52 +01:00
Michael Niedermayer
de14fe7e29 avformat/mxfdec: detect loops during header parsing
The header parser uses forward and backward parsing, making the
bulletproof prevention of loops difficult, thus this simple
detection code.
If someone improves the forward/backward parsing so it cannot loop
then this commit should be reverted

Fixes Ticket3278

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1c010fd035)
2014-01-08 10:19:36 +01:00
Michael Niedermayer
580d3ef6fe avformat/mov: Check that we have a stream before accessing it in mov_read_ares()
Fixes out of array read
Fixes: signal_sigsegv_6f1855_3910_avid_test_alpha.mov

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a7f27453f6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:25:25 +01:00
Michael Niedermayer
2864c987d8 avformat/ipmovie: check OPCODE_INIT_VIDEO_BUFFERS size more completely
Fixes use of uninitialized data

Fixes: signal_sigsegv_1571228_5930_ipmovie_interplayvideo_interplay_dpcm__bislogo.mve

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2e97e24409)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:25:20 +01:00
Michael Niedermayer
d1a9195863 avcodec/mjpegdec: check len in mjpeg_decode_app() more completely
Avoids len from becoming negative and causing assertion failure

Fixes: signal_sigabrt_7ffff7126425_5140_fd44dc63fa7bdd12ee34fc602231ef02.jpg

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6060234d43)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:25:07 +01:00
Michael Niedermayer
898ab02557 avformat/avidec: Dont assert the existence of an index for video streams.
Its possible in various rare cases that an index cannot be created or allocated.
Fixes assertion failure
Fixes: signal_sigabrt_7ffff7126425_7712_pokem.avi

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7865759409)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:25:02 +01:00
Michael Niedermayer
69aa3d5b88 avcodec/hevc: clear HEVClc when its deallocated in hevc_decode_free()
Fixes reading freed memory
Fixes: asan_heap-uaf_1abf8ef_3987_NUT_A_ericsson_4.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 21a2fb7e05)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:23:49 +01:00
Michael Niedermayer
dcecca0758 avutil/log: check that len is within the buffer before reading it
Fixes out of array read
Fixes: asan_heap-oob_19d6979_6857_mmw_deadzy.ogg
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 808c10e728)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:23:43 +01:00
Michael Niedermayer
413065aff4 avcodec/g2meet: check available space before copying palette
Fixes out of array read
Fixes: asan_heap-uaf_ae6067_5415_g2m4.wmv

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6d9dad6a7c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:23:35 +01:00
Michael Niedermayer
c094aec76e avcodec/ac3dec: check bap before use.
Fixes out of array read
Fixes assertion failure
Fixes asan_static-oob_16431c0_8036_rio_bravo_mono_64_spx.ac3

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4782c4284f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:23:23 +01:00
Michael Niedermayer
5e21989de4 Revert "Merge remote-tracking branch 'qatar/master'" (43dec5ef9a)
Fixes out of array accesses
Fixes asan_static-oob_eb9812_5961_iv41.avi
This reverts the merge of c9ef6b0932

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
(cherry picked from commit c3d5cd1ebf)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-08 00:23:18 +01:00
Michael Niedermayer
908b951b4e avcodec/hevc: Fix modulo operations
Fixes qp fields becoming out of range
Fixes: asan_static-oob_e393a3_6998_WPP_A_ericsson_MAIN10_2.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4ced5d7780)

Conflicts:

	libavcodec/hevc_filter.c
2014-01-07 23:48:10 +01:00
Michael Niedermayer
4b0cecb457 avcodec/hevc_ps: check that VPS referenced from SPS exists
This matches how its done for SPS/PPS.
An alternative to this is to check it when its used.

Fixes null pointer dereference
Fixes: signal_sigsegv_e30a43_1437_CIP_A_Panasonic_3.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d66bab0a69)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 23:47:39 +01:00
Nicolas George
3dae9d13e5 lavc/mjpegenc: use proper error codes.
(cherry picked from commit 2ebaadf35c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Nicolas George
9189a0a71b lavc/mjpegenc: check av_frame_alloc() failure.
(cherry picked from commit 19a2d101ac)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Nicolas George
4b25b5a8a3 lavc/libopenjpegenc: check av_frame_alloc() failure.
(cherry picked from commit 97af2faaba)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Nicolas George
4c3cd88144 lavc/diracdec: check av_frame_alloc() failure.
(cherry picked from commit a91394f4de)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Nicolas George
4d70639d53 lavc/utils: check av_frame_alloc() failure.
(cherry picked from commit 38004051b5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Nicolas George
a6ba0f7be9 ffprobe: check av_frame_alloc() failure.
(cherry picked from commit a55692a960)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Nicolas George
89205b637e lavc/ffwavesynth: fix dependency sizeof(AVFrame).
(cherry picked from commit bcfcb8b852)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
88058b4650 ffprobe: Dont clear AVFrame between uses.
The old API required this clearing in the past, the new API does not
require it.

Fixes memleak
Regression introduced by 37a749012a

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 565f786d1d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
ca22a2dec5 avcodec/utils: drop 2 dependancies on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c90f31146e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
d058583510 avcodec/libvorbisenc: drop dependancy on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3c8b085764)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
e0c3c612eb ffprobe: drop dependancy on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bf1c87ee7a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
4d4a10cfa1 avcodec/flashsv2enc: drop dependancy on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b8f4410ff6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
ce675bd54a avcodec/j2kenc: drop dependancy on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8443b27072)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
965eb42be0 avcodec/libopenjpegenc: drop dependancy on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1458f0647c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
d2578f8152 avcodec/mjpegenc: drop dependancy on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5b3f4b3ef5)

Conflicts:

	libavcodec/mjpegenc.c
2014-01-07 21:28:41 +01:00
Michael Niedermayer
31c52cd442 avcodec/msvideo1enc: drop dependancy on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c81234651f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
fa220e7307 avcodec/diracdec: avoid depending on sizeof(AVFrame)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fca7943850)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
304260a572 avcodec/utils: implement avcodec_alloc_frame() through av_alloc_frame()
This ensures that theres just one AVFrame allocation function and libs dont
produce multiple AVFrame variants after a minor lib update

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5abdda214d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
ed621efb36 avcodec/libutvideodec: use av_frame_move_ref()
AVFrames cannot be copied literally, their definition is in
avutil and their extended_data can point to their data[]

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4c1b4ae1ba)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
9f864bd324 Merge commit 'd4f1188d1a662fed5347e70016da49e01563e8a8'
* commit 'd4f1188d1a662fed5347e70016da49e01563e8a8':
  dv: use AVFrame API properly

Conflicts:
	libavcodec/dvdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 94a849b8b6)

Conflicts:

	libavcodec/dv.h
	libavcodec/dvdec.c
	libavcodec/dvenc.c

Author of the merged code: Anton Khirnov
2014-01-07 21:28:41 +01:00
Michael Niedermayer
117728cf8f Merge commit 'd351ef47d0e0ccb7de96b37f137c16b2885580ac'
* commit 'd351ef47d0e0ccb7de96b37f137c16b2885580ac':
  pthread_frame: use the AVFrame API properly.

Conflicts:
	libavcodec/pthread_frame.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 45fd4ec9ef)

Conflicts:

	libavcodec/pthread_frame.c
Author of the merged code: Anton Khirnov
2014-01-07 21:28:41 +01:00
Michael Niedermayer
caf7db0c35 Merge commit 'b605b123ef1d3bac0e7c221d8d7fa74cd8c7253c'
* commit 'b605b123ef1d3bac0e7c221d8d7fa74cd8c7253c':
  mxpegdec: use the AVFrame API properly.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8947f47fdf)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:41 +01:00
Michael Niedermayer
09965ae7d8 Merge commit 'afa21a12bf084f905187615706b0a8d92bc98661'
* commit 'afa21a12bf084f905187615706b0a8d92bc98661':
  p*menc: use the AVFrame API properly.

Conflicts:
	libavcodec/Makefile
	libavcodec/pamenc.c
	libavcodec/pnmenc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 37945584bf)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:40 +01:00
anatoly
f448478a31 Add support for picture_ptr field in MJpegDecodeContext
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit e0e3b8b297)

Conflicts:

	libavcodec/jpeglsdec.c
	libavcodec/mjpegbdec.c
	libavcodec/mjpegdec.c

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2fade10cb0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:28:40 +01:00
Michael Niedermayer
82ec6183bc Merge commit 'e2274aa555f023e4f4e4819bf29b2d7e0adec7d5'
* commit 'e2274aa555f023e4f4e4819bf29b2d7e0adec7d5':
  mjpegdec: use the AVFrame API properly.

Conflicts:
	libavcodec/mjpegdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6393038676)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 21:23:32 +01:00
Michael Niedermayer
345e2a2b43 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  mpegvideo_enc: use the AVFrame API properly.
  ffv1: use the AVFrame API properly.
  jpegls: use the AVFrame API properly.
  huffyuv: use the AVFrame API properly.

Conflicts:
	libavcodec/ffv1.c
	libavcodec/ffv1.h
	libavcodec/ffv1dec.c
	libavcodec/ffv1enc.c

Changes to ffv1 are more redone than merged due to them being based on
an ancient codebase and a good part of that having being done already
as well.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit be1e6e7503)

Conflicts:

	libavcodec/ffv1enc.c
	libavcodec/mpegvideo.h
	libavcodec/mpegvideo_enc.c
Author of the merged code: Anton Khirnov
2014-01-07 21:23:25 +01:00
Michael Niedermayer
7442aa20fa Merge commit 'd48c20630214a4effcc920e93a5044bee4e2002e'
* commit 'd48c20630214a4effcc920e93a5044bee4e2002e':
  qtrleenc: use the AVFrame API properly.
  ulti: use the AVFrame API properly.
  vc1: use the AVFrame API properly.
  flashsv: use the AVFrame API properly.

Conflicts:
	libavcodec/flashsv.c
	libavcodec/qtrleenc.c
	libavcodec/ulti.c
	libavcodec/vc1dec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 92cbd77568)

Conflicts:

	libavcodec/flashsv.c
Author of the merged code: Anton Khirnov
2014-01-07 21:23:19 +01:00
Michael Niedermayer
9918296a2d Merge commit 'ffe04c330335add4c6d70ab0bb98e6b3f4f7abfa'
* commit 'ffe04c330335add4c6d70ab0bb98e6b3f4f7abfa':
  libxvid: use the AVFrame API properly.
  pcxenc: use the AVFrame API properly.
  roqvideo: remove unused variables
  libschroedingerenc: use the AVFrame API properly.

Conflicts:
	libavcodec/pcxenc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f4f7888bab)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:32:48 +01:00
Michael Niedermayer
4d373ee8e4 Merge commit '97168b204a0b6b79bb6c5f0d40efdf7fc2262476'
* commit '97168b204a0b6b79bb6c5f0d40efdf7fc2262476':
  eatgv: use the AVFrame API properly.
  libxavs: use the AVFrame API properly.
  nuv: use the AVFrame API properly.
  flashsvenc: use the AVFrame API properly.

Conflicts:
	libavcodec/eatgv.c
	libavcodec/nuv.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a0c0629dd9)

Conflicts:

	libavcodec/eatgv.c
Author of the merged code: Anton Khirnov
2014-01-07 03:32:43 +01:00
Michael Niedermayer
6133f450bb Merge commit '57e7b3a89f5a0879ad039e8f04273b48649799a8'
* commit '57e7b3a89f5a0879ad039e8f04273b48649799a8':
  dnxhdenc: use the AVFrame API properly.
  libx264: use the AVFrame API properly.
  svq1enc: use the AVFrame API properly.
  gif: use the AVFrame API properly.

Conflicts:
	libavcodec/gif.c
	libavcodec/svq1enc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5b0c70c249)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:32:38 +01:00
Michael Niedermayer
8a9f4f8800 Merge commit '45bde93eefa78c1bdb0936109fbd2e2fb27fbfe7'
* commit '45bde93eefa78c1bdb0936109fbd2e2fb27fbfe7':
  sunrastenc: use the AVFrame API properly.
  targaenc: use the AVFrame API properly.
  tiffenc: use the AVFrame API properly.
  pngenc: use the AVFrame API properly.

Conflicts:
	libavcodec/pngenc.c
	libavcodec/sunrastenc.c
	libavcodec/targaenc.c
	libavcodec/tiffenc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3ea168edeb)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:32:32 +01:00
Michael Niedermayer
9ebe344166 Merge commit '0ea430c75b8d90449d2878ad84669a2da2ad3cbc'
* commit '0ea430c75b8d90449d2878ad84669a2da2ad3cbc':
  lclenc: use the AVFrame API properly.

Conflicts:
	libavcodec/lclenc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 85b7b0c519)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:32:26 +01:00
Michael Niedermayer
badb8e15ac Merge commit 'e4155f15b35c4272a235f5521d2dc6c2aabdd462'
* commit 'e4155f15b35c4272a235f5521d2dc6c2aabdd462':
  eamad: use the AVFrame API properly.
  dpxenc: use the AVFrame API properly.
  bmpenc: use the AVFrame API properly.
  sgienc: use the AVFrame API properly.

Conflicts:
	libavcodec/bmpenc.c
	libavcodec/dpxenc.c
	libavcodec/eamad.c
	libavcodec/sgienc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9ad477d909)

Conflicts:

	libavcodec/eamad.c

Author of the merged code: Anton Khirnov
2014-01-07 03:32:19 +01:00
Michael Niedermayer
48c192c48c Merge commit '730bac7bab3c7dcd9fcb7c70f154e5f4cfaef9a7'
* commit '730bac7bab3c7dcd9fcb7c70f154e5f4cfaef9a7':
  mss4: use the AVFrame API properly.
  mss3: use the AVFrame API properly.
  mss2: use the AVFrame API properly.
  mss1: use the AVFrame API properly.

Conflicts:
	libavcodec/mss1.c
	libavcodec/mss2.c
	libavcodec/mss3.c
	libavcodec/mss4.c

See: 02fe531afe
See: ff1c13b133
See: 310bf28354
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8d193a24f2)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:32:13 +01:00
Michael Niedermayer
dbb4ff6851 Merge commit '508b37557bf36eae83c18e64d42f27b44a321d81'
* commit '508b37557bf36eae83c18e64d42f27b44a321d81':
  tiertexseqv: use the AVFrame API properly.
  smc: use the AVFrame API properly.
  truemotion2: use the AVFrame API properly.
  truemotion1: use the AVFrame API properly.

Conflicts:
	libavcodec/smc.c
	libavcodec/tiertexseqv.c
	libavcodec/truemotion1.c
	libavcodec/truemotion2.c

See: e999f2339a
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 72df87088c)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:32:08 +01:00
Michael Niedermayer
8c53cacbfd Merge commit '4a4841d4e0f0dc50998511bf6c48b518012024db'
* commit '4a4841d4e0f0dc50998511bf6c48b518012024db':
  fraps: use the AVFrame API properly.
  rpza: use the AVFrame API properly.
  motionpixels: use the AVFrame API properly.
  vmdvideo: use the AVFrame API properly.

Conflicts:
	libavcodec/fraps.c
	libavcodec/motionpixels.c
	libavcodec/rpza.c
	libavcodec/vmdav.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4362f272c0)

Conflicts:

	libavcodec/vmdav.c

Author of the merged code: Anton Khirnov
2014-01-07 03:32:03 +01:00
Michael Niedermayer
dcb91e3dfe Merge commit '3c8ea9d4a74fd4d7493d40c818ca64ee492709f3'
* commit '3c8ea9d4a74fd4d7493d40c818ca64ee492709f3':
  vmnc: use the AVFrame API properly.
  xan: use the AVFrame API properly.
  xxan: use the AVFrame API properly.
  zerocodec: use the AVFrame API properly.

Conflicts:
	libavcodec/vmnc.c
	libavcodec/xan.c
	libavcodec/xxan.c

See: cf5ab8b6f7
See: ad438f450b
See: 67607e20e8
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8af7774c7a)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:31:57 +01:00
Michael Niedermayer
bfe4aa892a Merge commit 'a837c4f2df96a30bf9aa4115b426d608487c7101'
* commit 'a837c4f2df96a30bf9aa4115b426d608487c7101':
  zmbvenc: use the AVFrame API properly.
  flicvideo: use the AVFrame API properly.
  smacker: use the AVFrame API properly.
  mmvideo: use the AVFrame API properly.

Conflicts:
	libavcodec/flicvideo.c
	libavcodec/mmvideo.c
	libavcodec/smacker.c
	libavcodec/zmbvenc.c

See: 76e27b1d05
See: 099e57bc38
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fe3808edde)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:31:47 +01:00
Michael Niedermayer
1dcf9de6e9 Merge commit '2e09096da912f563c4dd889a8f25c314529bbaa6'
* commit '2e09096da912f563c4dd889a8f25c314529bbaa6':
  kgv1: use the AVFrame API properly.
  indeo2: use the AVFrame API properly.
  iff: use the AVFrame API properly.
  msrle: use the AVFrame API properly.

Conflicts:
	libavcodec/iff.c
	libavcodec/indeo2.c
	libavcodec/kgv1dec.c
	libavcodec/msrle.c

See: 451b2ca1b4
See: 80e9e63c94
See: 057dce5f21
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9f890a1656)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:31:34 +01:00
Michael Niedermayer
2fe67ddb24 Merge commit 'b7462a3904d71ff799584faf5b875cad59ca2f31'
* commit 'b7462a3904d71ff799584faf5b875cad59ca2f31':
  jvdec: use the AVFrame API properly.

Conflicts:
	libavcodec/jvdec.c

See: 678431d3f2
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ddfdcd2b5e)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:31:29 +01:00
Michael Niedermayer
b7cb77a322 Merge commit '2d2a92f72199823a92e4e226c32e42a27ec801c0'
* commit '2d2a92f72199823a92e4e226c32e42a27ec801c0':
  dxa: use the AVFrame API properly.
  qpeg: use the AVFrame API properly.
  cin video: use the AVFrame API properly.
  msvideo1: use the AVFrame API properly.

Conflicts:
	libavcodec/dsicinav.c
	libavcodec/dxa.c
	libavcodec/msvideo1.c
	libavcodec/qpeg.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5219afc09d)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:31:20 +01:00
Michael Niedermayer
2d7f20d2bb Merge commit 'a639ea7f4bc44bf6bfa452675558a342924a66a9'
* commit 'a639ea7f4bc44bf6bfa452675558a342924a66a9':
  escape124: use the AVFrame API properly.
  qtrle: use the AVFrame API properly.
  cljr: use the AVFrame API properly.
  cinepak: use the AVFrame API properly.

Conflicts:
	libavcodec/cinepak.c
	libavcodec/cljr.c
	libavcodec/qtrle.c

See: 80e9e63c libavcodec/cinepak.c
See: 71c378984b
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bfb1f44d24)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:31:14 +01:00
Michael Niedermayer
3976c50fc1 Merge commit 'cec5ce49229d61e4eb1f331a6d0dff3aa24f6655'
* commit 'cec5ce49229d61e4eb1f331a6d0dff3aa24f6655':
  cdxl: remove an unused variable
  c93: use the AVFrame API properly.
  bethsoftvid: use the AVFrame API properly.
  avs: use the AVFrame API properly.

Conflicts:
	libavcodec/bethsoftvideo.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 21c41e76d7)

Conflicts:

	libavcodec/avs.c
Author of the merged code: Anton Khirnov
2014-01-07 03:31:02 +01:00
Michael Niedermayer
572ccbd299 Merge commit '6139f481ac9feb1bee4e7d04789fb15d7f24ebbf'
* commit '6139f481ac9feb1bee4e7d04789fb15d7f24ebbf':
  asvenc: use the AVFrame API properly.
  a64multienc: use the AVFrame API properly.

Conflicts:
	libavcodec/vaapi_mpeg.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cc4a643563)

Author of the merged code: Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:30:57 +01:00
Dale Curtis
dcf0f82d08 h264: Clear ERContext.cur_pic when unref'ing current picture.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
(cherry picked from commit 4feca2214a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Michael Niedermayer
061e948153 configure: support raising major version in soname
this allows seperate installation of shared libs that should not conflict with
whatever is already installed.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 102b794e09)

Conflicts:

	libavcodec/utils.c

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Michael Niedermayer
08808084f6 swscale/utils: fill xyz tables only when they will be used
makes the first call to sws_getContext() 1ms faster

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4d18060e56)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Michael Niedermayer
e779595dca 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 a044a183a3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Michael Niedermayer
cf6cf50ac6 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 133fbfc781)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Michael Niedermayer
1ecd1b4aee avcodec/g2meet: fix stride calculation, use correct format field
Fixes out of array accesses
Fixes: asan_heap-oob_ae5f63_5415_g2m4.wmv
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 83f7bd6dcf)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Paul B Mahol
04a4c4144a 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 74906d3727)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Jan Gerber
73aa4518ee lavf/matroskadec ReferenceBlock is a signed integer
according to the Matroska Specification
ReferenceBlock is a signed integer too.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8cc59ec881)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Jan Gerber
aee36a7d16 lavf/matroska*: DiscardPadding is a signed integer
according to the Matriska Specification
 http://matroska.org/technical/specs/index.html
DiscardPadding is a signed integer.

Tested-by: Jan Gerber <j@v2v.cc>
Tested-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f4b1ca99ff)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Michael Niedermayer
d86930b8ff avformat/matroska: simplify signed int access code
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cddd15ba5c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:30 +01:00
Jan Gerber
95b5496dce lavf/matroska*: add support for signed integers
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d03eea36b2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:29 +01:00
Michael Niedermayer
3ffd1c2e40 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 3d5a5e86be)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:29 +01:00
Michael Niedermayer
2d16a88a9c 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 01923bab98)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-07 03:05:29 +01:00
Michael Niedermayer
edc6f3da0e avfilter/vf_format: check that the format list is not empty
Fixes Ticket3210

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ee16e0cacc)
2014-01-07 02:02:52 +01:00
Michael Niedermayer
8763aca389 avformat/oggdec: dont read timestamps from EOS pages of ogm videos
Some muxers store invalid timestamps there, which breaks seeking
Fixes Ticket2739

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5e0c7eab2a)
2014-01-05 17:25:29 +01:00
James Almer
b962157ce3 matroskadec: Fix bug when parsing realaudio codec parameters
flavor can be 0.

This fixes tract ticket #3214

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0d944ee343)
2013-12-30 00:49:35 +01:00
Alexander Strasser
8c79730a8e configure: Special case libfreetype test
Include the freetype header, in-directly through a macro, like it
is done in the drawtext filter. Do not break if the header is moved.

Unfortunately the drawtext filter included the file where the include
macros are defined in a wrong way. This is not needed and breaks the
build. Remove that #include line too.

(cherry picked from commit cea5812fa7)

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-12-29 11:36:01 +01:00
Michael Niedermayer
b432043d55 nutenc/write_index: warn if 2 consecutive keyframes have the same PTS and discard the 2nd
This fixes an assertion failure and regression and restores previous behaviour
Fixes Ticket3197

An alternative would be to fail hard in this case and refuse to mux such data.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit de2a2caf4d)
2013-12-24 07:45:32 +01:00
Peter Ross
94c3f8165c wtvenc: populate VIDEOINFOHEADER2
Fixes ticket #2835.

Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6da21c1f81)
2013-12-24 07:39:31 +01:00
Peter Ross
f27895db0f avformat/riffenc: indent
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f93b0abe40)
2013-12-24 07:39:23 +01:00
Peter Ross
c3f9628407 riffenc: add option to ff_put_bmp_header to ignore extradata
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fcbb94712d)
2013-12-24 07:39:15 +01:00
Mason Carter
7eec11463f 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 ed5bed4152)
2013-12-22 16:07:03 +01:00
Martin Storsjö
0d32483a11 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 d307e408d4)
2013-12-21 09:58:08 +01:00
Michael Niedermayer
85ea846580 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 037fc3b054)
2013-12-16 02:21:37 +01:00
Michael Niedermayer
5b52b4962e swscale/utils: remove useless ()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 554e913fd7)
2013-12-16 02:21:35 +01:00
Michael Niedermayer
69a283e0d2 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 0538b29ae8)
2013-12-09 10:34:52 +01:00
Michael Niedermayer
624b83b3ef 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 90539cea33)
2013-12-07 11:44:18 +01:00
Michael Niedermayer
e8304f4ee0 avcodec/error_resilience: factor er_supported() check out
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit afb18c5578)
2013-12-07 11:44:15 +01:00
Michael Niedermayer
cbcc18bd9f avfilter/vf_pad: fix req_end
Fixes out of array accesses
Fixes Ticket3190

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0cc5011f9a)
2013-12-03 09:52:23 +01:00
Michael Niedermayer
c765b64641 avcodec/h264_refs: improve key frame detection heuristic
Fixes Ticket3186

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e3d7a3978b)
2013-12-02 03:04:04 +01:00
Michael Niedermayer
1141a18e89 avcodec/h264_refs: split conditions of if() up for better readability
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ab6ea7a819)

Conflicts:
	libavcodec/h264_refs.c
2013-12-02 03:03:41 +01:00
Nicolas George
12c2d2ed46 lavc/srtenc: use bprint for text buffers.
Fix trac ticket #3120.
(cherry picked from commit 4b1c9b720e)
2013-11-30 19:22:07 +01:00
Michael Niedermayer
88e368d5a7 ffmpeg: set VCFR when copying timestamps
This fixes unreasonable initial frame repeats
Fixes Ticket3176

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 738ebb4a0e)
2013-11-29 11:31:05 +01:00
Nicolas George
ad19cb3ca7 lavfi/af_pan: support unknown layouts on input.
Fix trac ticket #2899.
(cherry picked from commit 7b0a587393)
2013-11-28 01:05:33 +01:00
Nicolas George
bc04a3a489 lavfi/af_pan: support unknown layouts on output.
(cherry picked from commit 4e9adc9b73)
2013-11-28 01:05:26 +01:00
Nicolas George
cfcb22a77b lswr: fix assert failure on unknown layouts.
(cherry picked from commit 4a640a6ac8)
2013-11-28 01:05:20 +01:00
Nicolas George
838a453e39 lavfi: parsing helper for unknown channel layouts.
Make ff_parse_channel_layout() accept unknown layouts too.
(cherry picked from commit 6e2473edfd)
2013-11-28 01:05:14 +01:00
Nicolas George
1a676881ca 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 d300f5f6f5)
2013-11-28 01:05:07 +01:00
Nicolas George
c2ae9f75d7 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 f775eb3fb4)
2013-11-28 01:05:01 +01:00
Nicolas George
31647c5a46 lavd/lavfi: support unknown channel layouts.
(cherry picked from commit 863fb11f63)
2013-11-28 01:04:54 +01:00
Michael Niedermayer
9422cd85a0 update for 2.1.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-20 03:12:11 +01:00
Michael Niedermayer
87c416d93a avcodec/pcm-dvd: fix 20/24bit 1 channel
Fixes part of ticket3122

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ab184b298d)
2013-11-19 11:46:43 +01:00
Michael Niedermayer
607e5038a9 avcodec/pcm-dvd: fix 20bit 2 channels
Fixes part of ticket3122

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5db49fc38d)
2013-11-19 11:46:32 +01:00
Michael Niedermayer
a289b0b91a avformat/mpegts: fix resync seek
The seek ended up seeking before the begin, which caused problems
Fixes initial sync issues with libbluray
Fixes Ticket3117

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7d0e927a31)
2013-11-18 17:54:15 +01:00
Michael Niedermayer
842def7d78 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 a8dec360c5)
2013-11-18 14:26:49 +01:00
Clément Bœsch
c38af02626 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 dbb41f93c1)
2013-11-18 14:26:18 +01:00
Clément Bœsch
7ce0f4ea3b avformat/image2: allow muxing gif files.
Fixes Ticket #2936.
(cherry picked from commit f70db22999)
2013-11-18 14:23:52 +01:00
Michael Niedermayer
3193b85be3 avcodec/tiff_common: allow count = 0 in ff_tadd_bytes_metadata()
Fixes Ticket3103

Reviewed-by: Thilo Borgmann <thilo.borgmann@mail.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 73d887733d)
2013-11-18 14:21:37 +01:00
Lou Logan
5c8845a554 Fix example in pullup documentation.
The pullup filter does not work well with the fps filter, it
currently needs -r

Signed-off-by: Carl Eugen Hoyos <cehoyos@ag.or.at>
(cherry picked from commit 6d90a5c149)
2013-11-18 14:20:01 +01:00
Michael Niedermayer
e5e048bbf7 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: 502ab21af0
See: cc6714bb16
See: 41578f70cf
See: Ticket3136
Tested-by: marillat
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b382d09d29)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:26:43 +01:00
Michael Niedermayer
b276b913a1 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 c3814ab654)

Conflicts:

	libavcodec/version.h
2013-11-17 19:12:29 +01:00
Michael Niedermayer
d89e14bf54 avutil: rename lls to lls2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bbe66ef912)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:28 +01:00
Michael Niedermayer
4b846f0ccf ffmpeg: Do not fill gap before the first decodable frame on single stream input files unless the user explicitly requests it.
Fixes different behavior to JM and probably several if not all
reference decoders.

We cannot just do this unconditionally as it would ruin AV sync in
some use cases.

Bug-Found-by: BugMaster
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d7ebeba80c)

Conflicts:

	ffmpeg.c
2013-11-17 19:11:07 +01:00
Diego Biurrun
425517eecb mpeg12dec: Remove incomplete and wrong UV swapping code for VCR2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3215140425)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Kostya Shishkov
bc89c2902b mpegvideo: Fix swapping of UV planes for VCR2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bae14f38d9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
1497633924 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 9e5ef1c5c3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
f167511753 avcodec/ffv1enc: Check high bpp RGB against coder type too
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
8c00647982 avformat/utils: never decrease has_b_frames in compute_pkt_fields()
The intent of the original check was to increase has_b_frames when
it was incorrectly set to 0. Later codecs allowed larger values

Found-by: divVerent
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8b73a3f6f6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
e40d01f45d avcodec/cabac: support UNCHECKED_BITSTREAM_READER = 0
Fixes overreads in HEVC
Fixes Ticket3070
Also fixed remaining issues from Ticket3075 and Ticket3076

Some lines of code taken from  0c5f839693da2276c2da23400f67a67be4ea0af1:libavcodec/x86/cabac.h
and                            0c5f839693da2276c2da23400f67a67be4ea0af1:libavcodec/cabac_functions.h

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fa6fa2162b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
51d1e79cc1 avformat/thp: force moving forward
Fixes infinite loop
Fixes Ticket3098

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6c4b87d3d6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
807d85400c avformat/thp: fix variable types to avoid overflows
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2b1056e4e2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
5c1e9d3722 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 cc0e47b550)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Anssi Hannula
b56e9beeb8 lavf/spdifdec: fix demuxing of AAC in IEC 61937
Return value of avpriv_aac_parse_header() is not checked correctly. Fix
it.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f86387b6c2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-17 19:11:07 +01:00
Michael Niedermayer
d8be5bda1b avformat/http: fix cookies
Fixes Ticket3096

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b73900b8a6)
2013-10-31 01:55:00 +01:00
Michael Niedermayer
1cd5797f8e avcodec/bink: fix seeking to frame 0
Fixes Ticket3088

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cb52d6da0a)
2013-10-31 00:50:24 +01:00
Michael Niedermayer
35a7b73590 update for 2.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-10-28 01:35:03 +01:00
2400 changed files with 73051 additions and 179866 deletions

13
.gitignore vendored
View File

@@ -15,7 +15,6 @@
*.pdb
*.so
*.so.*
*.swp
*.ver
*-example
*-test
@@ -28,6 +27,7 @@
/ffserver
/config.*
/coverage.info
/version.h
/doc/*.1
/doc/*.3
/doc/*.html
@@ -35,29 +35,23 @@
/doc/config.texi
/doc/avoptions_codec.texi
/doc/avoptions_format.texi
/doc/doxy/html/
/doc/examples/avio_reading
/doc/examples/decoding_encoding
/doc/examples/demuxing_decoding
/doc/examples/filter_audio
/doc/examples/demuxing
/doc/examples/filtering_audio
/doc/examples/filtering_video
/doc/examples/metadata
/doc/examples/muxing
/doc/examples/pc-uninstalled
/doc/examples/remuxing
/doc/examples/resampling_audio
/doc/examples/scaling_video
/doc/examples/transcode_aac
/doc/examples/transcoding
/doc/fate.txt
/doc/doxy/html/
/doc/print_options
/lcov/
/libavcodec/*_tablegen
/libavcodec/*_tables.c
/libavcodec/*_tables.h
/libavutil/avconfig.h
/libavutil/ffversion.h
/tests/audiogen
/tests/base64
/tests/data/
@@ -82,5 +76,4 @@
/tools/qt-faststart
/tools/trasher
/tools/seek_print
/tools/uncoded_frame
/tools/zmqsend

View File

@@ -1,76 +1,7 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version 2.3.1:
- public AVDCT API/ABI for DCT functions
- g2meet: allow size changes within original sizes
- dv: improved error resilience, fixing Ticket2340 and Ticket2341
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
version <next>
version 2.1:
@@ -115,8 +46,7 @@ version 2.1:
- 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
- HEVC decoder, raw HEVC demuxer, HEVC demuxing in TS, Matroska and MP4
- mergeplanes filter
@@ -132,7 +62,7 @@ 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
@@ -140,7 +70,7 @@ 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
@@ -258,7 +188,7 @@ 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
@@ -302,9 +232,7 @@ 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
@@ -428,7 +356,6 @@ version 0.10:
- ffwavesynth decoder
- aviocat tool
- ffeval tool
- support encoding and decoding 4-channel SGI images
version 0.9:

15
INSTALL Normal file
View File

@@ -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.

View File

@@ -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.

View File

@@ -1,4 +1,4 @@
#FFmpeg:
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
@@ -10,7 +10,7 @@ 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:
Specifically, the GPL parts of FFmpeg are
- libpostproc
- libmpcodecs
@@ -47,6 +47,7 @@ Specifically, the GPL parts of FFmpeg are:
- 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
@@ -79,7 +80,6 @@ The following libraries are under GPL:
- libutvideo
- libvidstab
- libx264
- libx265
- libxavs
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by

View File

@@ -31,7 +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
@@ -44,8 +44,8 @@ Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu
build system (configure, makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
build system (configure,Makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
@@ -54,10 +54,8 @@ release management Michael Niedermayer
Communication
=============
website Robert Swain
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists Michael Niedermayer, Baptiste Coudurier, Lou Logan
website Robert Swain, Lou Logan
mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan
Launchpad Timothy Gu
@@ -75,7 +73,6 @@ Other:
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
@@ -132,7 +129,6 @@ Generic Parts:
tableprint.c, tableprint.h Reimar Doeffinger
fixed point FFT:
fft* Zeljko Lukac
Text Subtitles Clément Bœsch
Codecs:
4xm.c Michael Niedermayer
@@ -146,7 +142,6 @@ Codecs:
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3* Benjamin Larsson
atrac3plus* Maxim Poliakovski
bgmc.c, bgmc.h Thilo Borgmann
bink.c Kostya Shishkov
binkaudio.c Peter Ross
@@ -155,7 +150,6 @@ Codecs:
cdxl.c Paul B Mahol
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cinepakenc.c Rl / Aetey G.T. AB
cljr Alex Beregszaszi
cllc.c Derek Buitenhuis
cook.c, cookdata.h Benjamin Larsson
@@ -166,13 +160,11 @@ Codecs:
dnxhd* Baptiste Coudurier
dpcm.c Mike Melanson
dv.c Roman Shaposhnik
dvbsubdec.c Anshul Maheshwari
dxa.c Kostya Shishkov
eacmv*, eaidct*, eat* Peter Ross
exif.c, exif.h Thilo Borgmann
ffv1* Michael Niedermayer
ffv1.c Michael Niedermayer
ffwavesynth.c Nicolas George
fic.c Derek Buitenhuis
flac* Justin Ruggles
flashsv* Benjamin Larsson
flicvideo.c Mike Melanson
@@ -182,7 +174,7 @@ Codecs:
h261* Michael Niedermayer
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
huffyuv* Michael Niedermayer, Christophe Gisquet
huffyuv.c Michael Niedermayer
idcinvideo.c Mike Melanson
imc* Benjamin Larsson
indeo2* Kostya Shishkov
@@ -205,11 +197,8 @@ Codecs:
libtheoraenc.c David Conrad
libutvideo* Derek Buitenhuis
libvorbis.c David Conrad
libvpx* James Zern
libx264.c Mans Rullgard, Jason Garrett-Glaser
libx265.c Derek Buitenhuis
libxavs.c Stefan Gehrer
libzvbi-teletextdec.c Marton Balint
loco.c Kostya Shishkov
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
@@ -242,12 +231,12 @@ Codecs:
rtjpeg.c, rtjpeg.h Reimar Doeffinger
rv10.c Michael Niedermayer
rv3* Kostya Shishkov
rv4* Kostya Shishkov, Christophe Gisquet
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
@@ -266,18 +255,17 @@ Codecs:
v410*.c Derek Buitenhuis
vb.c Kostya Shishkov
vble.c Derek Buitenhuis
vc1* Kostya Shishkov, Christophe Gisquet
vc1* Kostya Shishkov
vcr1.c Michael Niedermayer
vda_h264_dec.c Xidorn Quan
vima.c Paul B Mahol
vmnc.c Kostya Shishkov
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
vorbis_dec.c Denes Balatoni, David Conrad
vorbis_enc.c Oded Shimon
vp3* Mike Melanson
vp5 Aurelien Jacobs
vp6 Aurelien Jacobs
vp8 David Conrad, Jason Garrett-Glaser, Ronald Bultje
vp9 Ronald Bultje, Clément Bœsch
vqavideo.c Mike Melanson
wavpack.c Kostya Shishkov
wmaprodec.c Sascha Sommer
@@ -308,20 +296,15 @@ libavdevice
libavdevice/avdevice.h
avfoundation.m Thilo Borgmann
dshow.c Roger Pack (CC rogerdpack@gmail.com)
dshow.c Roger Pack
fbdev_enc.c Lukasz Marek
gdigrab.c Roger Pack (CC rogerdpack@gmail.com)
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 Luca Abeni
vfwcap.c Ramiro Polla
xv.c Lukasz Marek
libavfilter
===========
@@ -344,15 +327,12 @@ Filters:
af_ladspa.c Paul B Mahol
af_pan.c Nicolas George
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_colorbalance.c Paul B Mahol
vf_dejudder.c Nicholas Robbins
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_il.c Paul B Mahol
vf_mergeplanes.c Paul B Mahol
vf_psnr.c Paul B Mahol
@@ -405,7 +385,6 @@ Muxers/Demuxers:
flvdec.c, flvenc.c Michael Niedermayer
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hls.c Anssi Hannula
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
@@ -423,7 +402,6 @@ Muxers/Demuxers:
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
@@ -433,15 +411,14 @@ Muxers/Demuxers:
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
@@ -467,7 +444,6 @@ Muxers/Demuxers:
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
@@ -476,7 +452,6 @@ 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
@@ -518,8 +493,6 @@ Amiga / PowerPC Colin Ward
Linux / PowerPC Luca Barbato
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver
Windows ICL Matthew Oliver
ADI/Blackfin DSP Marc Hoffman
Sparc Roman Shaposhnik
x86 Michael Niedermayer
@@ -528,9 +501,8 @@ x86 Michael Niedermayer
Releases
========
2.3 Michael Niedermayer
2.2 Michael Niedermayer
1.2 Michael Niedermayer
2.1 Michael Niedermayer
2.0 Michael Niedermayer
If you want to maintain an older release, please contact us
@@ -546,7 +518,7 @@ 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
@@ -561,14 +533,12 @@ Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
Reimar 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

View File

@@ -4,49 +4,39 @@ include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
vpath %.h $(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)
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)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
OBJS-ffmpeg += ffmpeg_opt.o ffmpeg_filter.o
OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_vda.o
PROGS := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
INSTPROGS = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
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
@@ -60,14 +50,16 @@ include $(SRC_PATH)/common.mak
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
all: $(AVPROGS)
all: $(PROGS)
$(PROGS): %$(EXESUF): %_g$(EXESUF)
$(CP) $< $@
$(STRIP) $@
$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
$(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))
@@ -77,10 +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 \
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 SLIBOBJS HOSTOBJS TESTOBJS
OBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=
@@ -97,10 +90,8 @@ 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))
@@ -108,13 +99,7 @@ $(1)$(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 : 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) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
@@ -129,14 +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)
$(Q)touch .version
# force version.sh to run whenever version might have changed
-include .version
ifdef AVPROGS
ifdef PROGS
install: install-progs install-data
endif
@@ -147,7 +132,7 @@ install-libs: install-libs-yes
install-progs-yes:
install-progs-$(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)"
@@ -159,13 +144,13 @@ 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
@@ -173,13 +158,14 @@ clean::
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
$(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)):

View File

@@ -1,11 +1,10 @@
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

View File

@@ -1 +1 @@
2.3.1
2.1.5

View File

@@ -1,177 +0,0 @@
┌───────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 2.3 "Mandelbrot" │
└───────────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 2.3 "Mandelbrot", a major
release with all the great features committed during the three-month period
since the release of FFmpeg 2.2.
In this release, there are lots of internal overhauls that make FFmpeg a
more accessible project for new developers. Many important new
optimizations and features like QTKit and AVFoundation input devices are
committed. Contributions done by Libav such as a new native Opus decoder
are also merged.
Because of the increasing difficulty to maintain and lack of maintainers,
we are very sorry to say that we have removed all Blackfin and SPARC
architecture assembly optimizations with the cleanups done. If you are
interested in maintaining optimization for these two architecture, feel
free to contact us and we will restore the code!
Oh, and since this release, this modern-looking release note is provided in
addition to the old-style Changelog file, to make it easier for you to
focus on the most important features in this release.
Enjoy!
┌────────────────────────────┐
│ * API Information │
└────────────────────────────┘
FFmpeg 2.3 is completely source-compatible to the FFmpeg 2.2 series. There
are however some API deprecations that you need to take care of. Use `git
diff n2.2 n2.3 doc/APIchanges` to show the list of added and deprecated
APIs. FFmpeg 2.3 includes the following library versions:
• libavutil 52.92.100
• libavcodec 55.69.100
• libavformat 55.48.100
• libavdevice 55.13.102
• libavfilter 4.11.100
• libswscale 2. 6.100
• libswresample 0.19.100
• libpostproc 52. 3.100
Please refer to the doc/APIChanges file for more information.
┌────────────────────────────┐
│ New Optimization │
└────────────────────────────┘
We are excited to announce that we have committed new x86 assembly
optimization for HEVC, and FFmpeg's audio resampler libswresample. ARM
users will get a boost in MLP/TrueHD decoding thanks to new optimization.
Decoding Huffyuv also got a major boost from optimization on the C code.
Of special interest for Microsoft Visual Studio users, we have also
converted some preexisting x86 assembly to NASM/Yasm format compatible
with MSVC setup, especially in the area of audio resampling.
Another major feature in this release is the introduction of AArch64
(ARMv8) assembly optimization. AArch64 is another name for the first
64-bit ARM architecture, used by Apple A7 SoC inside iPhone 5S. Some
32-bit ARM assembly has already been ported to AArch64, but more work is
underway.
┌────────────────────────────┐
│ Native Opus decoder │
└────────────────────────────┘
Opus is an open audio format jointly developed by Xiph.Org, Mozilla,
Skype/Microsoft, and Broadcom. It combines the features of the Skype Cilk
speech codec and the Xiph.Org CELT music codec into one low-latency
codec. Decoding Opus is already possible since FFmpeg 1.0 using the
libopus library, but the new Opus native decoder brings a higher level of
stability and speed.
┌────────────────────────────┐
│ QTKit and AVFoundation │
└────────────────────────────┘
For OS X users, the new QTKit and AVFoundation devices allow you to use
the integrated camera on Macs. AVFoundation is a newer API only available
on OS X 10.7 "Lion" or newer. For users with older OS X systems, the
QTKit device using the older OS X API is for you.
┌────────────────────────────┐
│ API Additions │
└────────────────────────────┘
In this release, stream side data are introduced as AVStream.side_data as
a way to store miscellaneous stream-wide information. The format is
similar to the previously anonymous structure AVPacket.side_data (now
named as AVPacketSideData). With this change, audio ReplayGain
information and video rotation matrix are now exported through this API,
if available in the demuxer.
We also have improved libswresample's Doxygen API documentation, so new
developers wishing to use FFmpeg's excellent libraries can get started
more easily and faster.
┌────────────────────────────┐
│ Last But Not Least │
└────────────────────────────┘
Other interesting new features including hqx video filter, a pixel art
scaling filter; a fixed-point AC-3 decoder contributed by Imagination
Technologies; an On2 TrueMotion VP7 video decoder; an HTML5 WebVTT
subtitle encoder that allows creation of WebVTT from any text-based
subtitles; and an 1-bit Direct Stream Digital audio decoder.
┌────────────────────────────┐
│ ★ List of New Features │
└────────────────────────────┘
┌────────────────────────────┐
│ Command line tools │
└────────────────────────────┘
• Support for decoding through DXVA2 in ffmpeg
┌────────────────────────────┐
│ libavcodec │
└────────────────────────────┘
• AC3 fixed-point decoding
• VP7 video decoder
• Alias PIX image encoder and decoder
• Improvements to the BRender PIX image decoder
• Improvements to the XBM decoder
• Improvements to OpenEXR image decoder
• Support decoding 16-bit RLE SGI images
• Direct Stream Digital (DSD) decoder
• On2 AVC (Audio for Video) decoder
• Native Opus decoder
• WebVTT encoder
┌────────────────────────────┐
│ libavdevice │
└────────────────────────────┘
• QTKit input device
• GDI screen grabbing for Windows
• AVFoundation input device
┌────────────────────────────┐
│ libavformat │
└────────────────────────────┘
• subfile protocol
• Phantom Cine demuxer
• Alternative rendition support for HTTP Live Streaming
• Magic Lantern Video (MLV) demuxer
• Image format auto-detection
• LRC lyric file demuxer and muxer
• Samba protocol (via libsmbclient)
• WebM DASH Manifest muxer
┌────────────────────────────┐
│ libavfilter │
└────────────────────────────┘
• shuffleplanes filter
• libbs2b-based stereo-to-binaural audio filter
• showcqt multimedia filter
• zoompan filter
• signalstats filter
• hqx filter (hq2x, hq3x, hq4x)
• flanger filter
• libfribidi support in drawtext
┌────────────────────────────┐
│ ⚠ Behaviour changes │
└────────────────────────────┘
• libx264 reference frames count is now limited depending on level chosen
• Because of the new image format auto-detection feature, you don't need to
specify image format when decoding an image with no extension.

1
VERSION Normal file
View File

@@ -0,0 +1 @@
2.1.5

View File

@@ -1,6 +1,5 @@
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) $(NEON-OBJS-yes)
@@ -11,5 +10,7 @@ OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS) $(MIPSDSPR2-OBJS-yes)
OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
OBJS-$(HAVE_VIS) += $(VIS-OBJS) $(VIS-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_YASM) += $(YASM-OBJS) $(YASM-OBJS-yes)

View File

@@ -20,7 +20,6 @@
*/
#include <string.h>
#include <stdint.h>
#include <stdlib.h>
#include <errno.h>
#include <math.h>
@@ -49,8 +48,8 @@
#include "libavutil/dict.h"
#include "libavutil/opt.h"
#include "libavutil/cpu.h"
#include "libavutil/ffversion.h"
#include "cmdutils.h"
#include "version.h"
#if CONFIG_NETWORK
#include "libavformat/network.h"
#endif
@@ -58,6 +57,10 @@
#include <sys/time.h>
#include <sys/resource.h>
#endif
#if CONFIG_OPENCL
#include "libavutil/opencl.h"
#endif
static int init_report(const char *env);
@@ -65,9 +68,9 @@ struct SwsContext *sws_opts;
AVDictionary *swr_opts;
AVDictionary *format_opts, *codec_opts, *resample_opts;
const int this_year = 2014;
static FILE *report_file;
static int report_file_level = AV_LOG_DEBUG;
int hide_banner = 0;
void init_opts(void)
{
@@ -105,10 +108,8 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
av_log_default_callback(ptr, level, fmt, vl);
av_log_format_line(ptr, level, fmt, vl2, line, sizeof(line), &print_prefix);
va_end(vl2);
if (report_file_level >= level) {
fputs(line, report_file);
fflush(report_file);
}
fputs(line, report_file);
fflush(report_file);
}
static void (*program_exit)(int ret);
@@ -495,9 +496,6 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
fflush(report_file);
}
}
idx = locate_option(argc, argv, options, "hide_banner");
if (idx)
hide_banner = 1;
}
static const AVOption *opt_find(void *obj, const char *name, const char *unit,
@@ -555,11 +553,6 @@ int opt_default(void *optctx, const char *opt, const char *arg)
}
consumed = 1;
}
#else
if (!consumed && !strcmp(opt, "sws_flags")) {
av_log(NULL, AV_LOG_WARNING, "Ignoring %s %s, due to disabled swscale\n", opt, arg);
consumed = 1;
}
#endif
#if CONFIG_SWRESAMPLE
swr_class = swr_get_class();
@@ -670,7 +663,7 @@ static void init_parse_context(OptionParseContext *octx,
memset(octx, 0, sizeof(*octx));
octx->nb_groups = nb_groups;
octx->groups = av_mallocz_array(octx->nb_groups, sizeof(*octx->groups));
octx->groups = av_mallocz(sizeof(*octx->groups) * octx->nb_groups);
if (!octx->groups)
exit_program(1);
@@ -842,17 +835,10 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
};
char *tail;
int level;
int flags;
int i;
flags = av_log_get_flags();
tail = strstr(arg, "repeat");
if (tail)
flags &= ~AV_LOG_SKIP_REPEATED;
else
flags |= AV_LOG_SKIP_REPEATED;
av_log_set_flags(flags);
av_log_set_flags(tail ? 0 : AV_LOG_SKIP_REPEATED);
if (tail == arg)
arg += 6 + (arg[6]=='+');
if(tail && !*arg)
@@ -934,13 +920,6 @@ static int init_report(const char *env)
av_free(filename_template);
filename_template = val;
val = NULL;
} else if (!strcmp(key, "level")) {
char *tail;
report_file_level = strtol(val, &tail, 10);
if (*tail) {
av_log(NULL, AV_LOG_FATAL, "Invalid report file level\n");
exit_program(1);
}
} else {
av_log(NULL, AV_LOG_ERROR, "Unknown key '%s' in FFREPORT\n", key);
}
@@ -1007,6 +986,26 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg)
return 0;
}
#if CONFIG_OPENCL
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;
}
#endif
void print_error(const char *filename, int err)
{
char errbuf[128];
@@ -1072,7 +1071,7 @@ static void print_program_info(int flags, int level)
av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
if (flags & SHOW_COPYRIGHT)
av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
program_birth_year, CONFIG_THIS_YEAR);
program_birth_year, this_year);
av_log(NULL, level, "\n");
av_log(NULL, level, "%sbuilt on %s %s with %s\n",
indent, __DATE__, __TIME__, CC_IDENT);
@@ -1080,36 +1079,10 @@ static void print_program_info(int flags, int level)
av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
}
static void print_buildconf(int flags, int level)
{
const char *indent = flags & INDENT ? " " : "";
char str[] = { FFMPEG_CONFIGURATION };
char *conflist, *remove_tilde, *splitconf;
// Change all the ' --' strings to '~--' so that
// they can be identified as tokens.
while ((conflist = strstr(str, " --")) != NULL) {
strncpy(conflist, "~--", 3);
}
// Compensate for the weirdness this would cause
// when passing 'pkg-config --static'.
while ((remove_tilde = strstr(str, "pkg-config~")) != NULL) {
strncpy(remove_tilde, "pkg-config ", 11);
}
splitconf = strtok(str, "~");
av_log(NULL, level, "\n%sconfiguration:\n", indent);
while (splitconf != NULL) {
av_log(NULL, level, "%s%s%s\n", indent, indent, splitconf);
splitconf = strtok(NULL, "~");
}
}
void show_banner(int argc, char **argv, const OptionDef *options)
{
int idx = locate_option(argc, argv, options, "version");
if (hide_banner || idx)
if (idx)
return;
print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
@@ -1120,20 +1093,12 @@ void show_banner(int argc, char **argv, const OptionDef *options)
int show_version(void *optctx, const char *opt, const char *arg)
{
av_log_set_callback(log_callback_help);
print_program_info (SHOW_COPYRIGHT, AV_LOG_INFO);
print_program_info (0 , AV_LOG_INFO);
print_all_libs_info(SHOW_VERSION, AV_LOG_INFO);
return 0;
}
int show_buildconf(void *optctx, const char *opt, const char *arg)
{
av_log_set_callback(log_callback_help);
print_buildconf (INDENT|0, AV_LOG_INFO);
return 0;
}
int show_license(void *optctx, const char *opt, const char *arg)
{
#if CONFIG_NONFREE
@@ -1208,29 +1173,16 @@ int show_license(void *optctx, const char *opt, const char *arg)
return 0;
}
static int is_device(const AVClass *avclass)
{
if (!avclass)
return 0;
return avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_OUTPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_INPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_OUTPUT ||
avclass->category == AV_CLASS_CATEGORY_DEVICE_INPUT;
}
static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
int show_formats(void *optctx, const char *opt, const char *arg)
{
AVInputFormat *ifmt = NULL;
AVOutputFormat *ofmt = NULL;
const char *last_name;
int is_dev;
printf("%s\n"
printf("File formats:\n"
" D. = Demuxing supported\n"
" .E = Muxing supported\n"
" --\n", device_only ? "Devices:" : "File formats:");
" --\n");
last_name = "000";
for (;;) {
int decode = 0;
@@ -1239,9 +1191,6 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
const char *long_name = NULL;
while ((ofmt = av_oformat_next(ofmt))) {
is_dev = is_device(ofmt->priv_class);
if (!is_dev && device_only)
continue;
if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
strcmp(ofmt->name, last_name) > 0) {
name = ofmt->name;
@@ -1250,9 +1199,6 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
}
}
while ((ifmt = av_iformat_next(ifmt))) {
is_dev = is_device(ifmt->priv_class);
if (!is_dev && device_only)
continue;
if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
strcmp(ifmt->name, last_name) > 0) {
name = ifmt->name;
@@ -1275,16 +1221,6 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
return 0;
}
int show_formats(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 0);
}
int show_devices(void *optctx, const char *opt, const char *arg)
{
return show_formats_devices(optctx, opt, arg, 1);
}
#define PRINT_CODEC_SUPPORTED(codec, field, type, list_name, term, get_name) \
if (codec->field) { \
const type *p = codec->field; \
@@ -1429,9 +1365,6 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
const AVCodecDescriptor *desc = codecs[i];
const AVCodec *codec = NULL;
if (strstr(desc->name, "_deprecated"))
continue;
printf(" ");
printf(avcodec_find_decoder(desc->id) ? "D" : ".");
printf(avcodec_find_encoder(desc->id) ? "E" : ".");
@@ -1585,7 +1518,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
return 0;
}
int show_colors(void *optctx, const char *opt, const char *arg)
void show_colors(void *optctx, const char *opt, const char *arg)
{
const char *name;
const uint8_t *rgb;
@@ -1595,8 +1528,6 @@ int show_colors(void *optctx, const char *opt, const char *arg)
for (i = 0; name = av_get_known_color_name(i, &rgb); i++)
printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]);
return 0;
}
int show_pix_fmts(void *optctx, const char *opt, const char *arg)
@@ -1858,7 +1789,7 @@ int read_yesno(void)
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
{
int ret;
FILE *f = av_fopen_utf8(filename, "rb");
FILE *f = fopen(filename, "rb");
if (!f) {
av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
@@ -1996,8 +1927,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
}
if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
!codec ||
(codec->priv_class &&
(codec && codec->priv_class &&
av_opt_find(&codec->priv_class, t->key, NULL, flags,
AV_OPT_SEARCH_FAKE_OBJ)))
av_dict_set(&ret, t->key, t->value, 0);
@@ -2020,7 +1950,7 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
if (!s->nb_streams)
return NULL;
opts = av_mallocz_array(s->nb_streams, sizeof(*opts));
opts = av_mallocz(s->nb_streams * sizeof(*opts));
if (!opts) {
av_log(NULL, AV_LOG_ERROR,
"Could not alloc memory for stream options.\n");

View File

@@ -24,13 +24,12 @@
#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 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
@@ -100,12 +103,8 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg);
int opt_codec_debug(void *optctx, const char *opt, const char *arg);
#if CONFIG_OPENCL
int opt_opencl(void *optctx, const char *opt, const char *arg);
int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
#endif
/**
* Limit the execution time.
*/
@@ -415,13 +414,6 @@ 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.
@@ -431,17 +423,10 @@ 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);
int show_formats(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing all the codecs supported by the
@@ -507,7 +492,7 @@ 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);
void show_colors(void *optctx, const char *opt, const char *arg);
/**
* Return a positive value if a line read from standard input

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" },
@@ -22,8 +20,6 @@
{ "report" , 0, {(void*)opt_report}, "generate a report" },
{ "max_alloc" , HAS_ARG, {.func_arg = opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },
{ "cpuflags" , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
{ "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" },
#if CONFIG_OPENCL
{ "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
{ "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, "set OpenCL environment options" },
#endif

View File

@@ -1,274 +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 "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)
{
return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->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;
strcpy(devices[count].device_name, device_node->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

@@ -10,7 +10,7 @@ ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX 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 = $@
@@ -43,7 +43,6 @@ endef
COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.o: %.c
$(COMPILE_C)
@@ -51,21 +50,12 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
%.o: %.cpp
$(COMPILE_CXX)
%.o: %.m
$(COMPILE_C)
%.s: %.c
$(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) $<
@@ -92,15 +82,14 @@ 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)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -124,19 +113,18 @@ 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 *.map *.ver *.ho *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
@@ -151,4 +139,4 @@ endef
$(eval $(RULES))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))

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

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

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;

View File

@@ -32,8 +32,6 @@
#undef __STRICT_ANSI__ /* for _beginthread() */
#include <stdlib.h>
#include "libavutil/mem.h"
typedef TID pthread_t;
typedef void pthread_attr_t;

View File

@@ -24,6 +24,3 @@
#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

View File

@@ -134,29 +134,28 @@ typedef struct win32_cond_t {
volatile int is_broadcast;
} win32_cond_t;
static 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;
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;
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 void pthread_cond_destroy(pthread_cond_t *cond)

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

1621
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -4,7 +4,7 @@ since the last major version increase or the API was added.
The last version increases were:
libavcodec: 2013-03-xx
libavdevice: 2013-03-xx
libavfilter: 2013-12-xx
libavfilter: 2012-06-22
libavformat: 2013-03-xx
libavresample: 2012-10-05
libpostproc: 2011-04-18
@@ -15,346 +15,31 @@ libavutil: 2012-10-22
API changes, most recent first:
2014-07-14 - 62227a7 - lavf 55.47.100 - avformat.h
Add av_stream_get_parser()
2014-07-09 - c67690f / a54f03b - lavu 52.92.100 / 53.18.0 - display.h
Add av_display_matrix_flip() to flip the transformation matrix.
2014-07-09 - 1b58f13 / f6ee61f - lavc 55.69.100 / 55.56.0 - dv_profile.h
Add a public API for DV profile handling.
2014-06-20 - 0dceefc / 9e500ef - lavu 52.90.100 / 53.17.0 - imgutils.h
Add av_image_check_sar().
2014-06-20 - 4a99333 / 874390e - lavc 55.68.100 / 55.55.0 - avcodec.h
Add av_packet_rescale_ts() to simplify timestamp conversion.
2014-06-18 - ac293b6 / 194be1f - lavf 55.44.100 / 55.20.0 - avformat.h
The proper way for providing a hint about the desired timebase to the muxers
is now setting AVStream.time_base, instead of AVStream.codec.time_base as was
done previously. The old method is now deprecated.
2014-06-11 - 67d29da - lavc 55.66.101 - avcodec.h
Increase FF_INPUT_BUFFER_PADDING_SIZE to 32 due to some corner cases needing
it
2014-06-10 - xxxxxxx - lavf 55.43.100 - avformat.h
New field int64_t max_analyze_duration2 instead of deprecated
int max_analyze_duration.
2014-05-30 - 00759d7 - lavu 52.89.100 - opt.h
Add av_opt_copy()
2014-06-01 - 03bb99a / 0957b27 - lavc 55.66.100 / 55.54.0 - avcodec.h
Add AVCodecContext.side_data_only_packets to allow encoders to output packets
with only side data. This option may become mandatory in the future, so all
users are recommended to update their code and enable this option.
2014-06-01 - 6e8e9f1 / 8c02adc - lavu 52.88.100 / 53.16.0 - frame.h, pixfmt.h
Move all color-related enums (AVColorPrimaries, AVColorSpace, AVColorRange,
AVColorTransferCharacteristic, and AVChromaLocation) inside lavu.
And add AVFrame fields for them.
2014-05-29 - bdb2e80 / b2d4565 - lavr 1.3.0 - avresample.h
Add avresample_max_output_samples
2014-05-24 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
Add strict_std_compliance and related AVOptions to support experimental
muxing.
2014-05-26 - xxxxxxx - lavu 52.87.100 - threadmessage.h
Add thread message queue API.
2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h
Add format_probesize to AVFormatContext.
2014-05-19 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
Add av_stream_get_side_data() to access stream-level side data
in the same way as av_packet_get_side_data().
2014-05-xx - xxxxxxx - lavu 52.86.100 - fifo.h
Add av_fifo_alloc_array() function.
2014-05-19 - ef1d4ee / bddd8cb - lavu 52.85.100 / 53.15.0 - frame.h, display.h
Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level
spatial rendering on video frames for proper display.
2014-05-xx - xxxxxxx - lavc 55.64.100 / 55.53.0 - avcodec.h
Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level
spatial rendering on video frames for proper display.
2014-05-19 - 999a99c / a312f71 - lavf 55.38.101 / 55.17.1 - avformat.h
Deprecate AVStream.pts and the AVFrac struct, which was its only use case.
See use av_stream_get_end_pts()
2014-05-18 - 68c0518 / fd05602 - lavc 55.63.100 / 55.52.0 - avcodec.h
Add avcodec_free_context(). From now on it should be used for freeing
AVCodecContext.
2014-05-17 - 0eec06e - lavu 52.84.100 - time.h
Add av_gettime_relative() av_gettime_relative_is_monotonic()
2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h
Add AVMFT_FLAG_BITEXACT flag. Muxers now use it instead of checking
CODEC_FLAG_BITEXACT on the first stream.
2014-05-15 - 96cb4c8 - lswr 0.19.100 - swresample.h
Add swr_close()
2014-05-11 - 14aef38 / 66e6c8a - lavu 52.83.100 / 53.14.0 - pixfmt.h
Add AV_PIX_FMT_VDA for new-style VDA acceleration.
2014-05-xx - xxxxxxx - lavu 52.82.0 - fifo.h
Add av_fifo_freep() function.
2014-05-02 - ba52fb11 - lavu 52.81.0 - opt.h
Add av_opt_set_dict2() function.
2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h
Deprecate CODEC_FLAG_MV0. It is replaced by the flag "mv0" in the
"mpv_flags" private option of the mpegvideo encoders.
2014-05-01 - e40ae8c / 6484149 - lavc 55.60.102 / 55.50.2 - avcodec.h
Deprecate CODEC_FLAG_GMC. It is replaced by the "gmc" private option of the
libxvid encoder.
2014-05-01 - 1851643 / b2c3171 - lavc 55.60.101 / 55.50.1 - avcodec.h
Deprecate CODEC_FLAG_NORMALIZE_AQP. It is replaced by the flag "naq" in the
"mpv_flags" private option of the mpegvideo encoders.
2014-05-01 - cac07d0 / 5fcceda - avcodec.h
Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing
reference-counted frames to encoders.
2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h
Add AVCodecDescriptor.mime_types field.
2014-04-29 - xxxxxxx - lavu 52.80.0 - hash.h
Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64().
2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h
Add FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO for old Intel GPUs.
2014-04-22 - 502512e /dac7e8a - lavu 53.13.0 / 52.78.100 - avutil.h
Add av_get_time_base_q().
2014-04-17 - a8d01a7 / 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h
Add AV_CRC_16_ANSI_LE crc variant.
2014-04-XX - xxxxxxx - lavf xx.xx.1xx - avformat.h
Add av_format_inject_global_side_data()
2014-04-12 - 4f698be - lavu 52.76.100 - log.h
Add av_log_get_flags()
2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h
Add avdevice_capabilities_create() function.
Add avdevice_capabilities_free() function.
2014-04-07 - 0a1cc04 / 8b17243 - lavu 52.75.100 / 53.11.0 - pixfmt.h
Add AV_PIX_FMT_YVYU422 pixel format.
2014-04-04 - c1d0536 / 8542f9c - lavu 52.74.100 / 53.10.0 - replaygain.h
Full scale for peak values is now 100000 (instead of UINT32_MAX) and values
may overflow.
2014-04-03 - c16e006 / 7763118 - lavu 52.73.100 / 53.9.0 - log.h
Add AV_LOG(c) macro to have 256 color debug messages.
2014-04-03 - eaed4da9 - lavu 52.72.100 - opt.h
Add AV_OPT_MULTI_COMPONENT_RANGE define to allow return
multi-component option ranges.
2014-03-29 - cd50a44b - lavu 52.70.100 - mem.h
Add av_dynarray_add_nofree() function.
2014-02-24 - 3e1f241 / d161ae0 - lavu 52.69.100 / 53.8.0 - frame.h
Add av_frame_remove_side_data() for removing a single side data
instance from a frame.
2014-03-24 - 83e8978 / 5a7e35d - lavu 52.68.100 / 53.7.0 - frame.h, replaygain.h
Add AV_FRAME_DATA_REPLAYGAIN for exporting replaygain tags.
Add a new header replaygain.h with the AVReplayGain struct.
2014-03-24 - 83e8978 / 5a7e35d - lavc 55.54.100 / 55.36.0 - avcodec.h
Add AV_PKT_DATA_REPLAYGAIN for exporting replaygain tags.
2014-03-24 - 595ba3b / 25b3258 - lavf 55.35.100 / 55.13.0 - avformat.h
Add AVStream.side_data and AVStream.nb_side_data for exporting stream-global
side data (e.g. replaygain tags, video rotation)
2014-03-24 - bd34e26 / 0e2c3ee - lavc 55.53.100 / 55.35.0 - avcodec.h
Give the name AVPacketSideData to the previously anonymous struct used for
AVPacket.side_data.
2014-03-18 - 37c07d4 - lsws 2.5.102
Make gray16 full-scale.
2014-03-16 - 6b1ca17 / 1481d24 - lavu 52.67.100 / 53.6.0 - pixfmt.h
Add RGBA64_LIBAV pixel format and variants for compatibility
2014-03-11 - 3f3229c - lavf 55.34.101 - avformat.h
Set AVFormatContext.start_time_realtime when demuxing.
2014-03-03 - 06fed440 - lavd 55.11.100 - avdevice.h
Add av_input_audio_device_next().
Add av_input_video_device_next().
Add av_output_audio_device_next().
Add av_output_video_device_next().
2014-02-24 - fff5262 / 1155fd0 - lavu 52.66.100 / 53.5.0 - frame.h
Add av_frame_copy() for copying the frame data.
2014-02-24 - a66be60 - lswr 0.18.100 - swresample.h
Add swr_is_initialized() for checking whether a resample context is initialized.
2014-02-22 - 5367c0b / 7e86c27 - lavr 1.2.0 - avresample.h
Add avresample_is_open() for checking whether a resample context is open.
2014-02-19 - 6a24d77 / c3ecd96 - lavu 52.65.100 / 53.4.0 - opt.h
Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only)
for reading.
2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h
Deprecate unused AV_OPT_FLAG_METADATA.
2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
Add avdevice_list_devices() and avdevice_free_list_devices()
2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
Add avio_find_protocol_name() to find out the name of the protocol that would
be selected for a given URL.
2014-02-15 - a2bc6c1 / c98f316 - lavu 52.64.100 / 53.3.0 - frame.h
Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and
downmix_info.h API, which identify downmix-related metadata.
2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h
Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame().
2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h
Add AVFormatContext.max_interleave_delta for controlling amount of buffering
when interleaving.
2014-02-02 - 5871ee5 - lavf 55.29.100 - avformat.h
Add output_ts_offset muxing option to AVFormatContext.
2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h
lavf 55.28.100 - avformat.h
Add avdevice_dev_to_app_control_message() function.
2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h
lavf 55.27.100 - avformat.h
Add avdevice_app_to_dev_control_message() function.
2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h
Add AVFormatContext option metadata_header_padding to allow control over the
amount of padding added.
2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h
Edges are not required anymore on video buffers allocated by get_buffer2()
(i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate
CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width().
2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h
Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
Add av_make_q() function.
2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which
identifies AVMatrixEncoding data.
2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h
Add values for various Dolby flags to the AVMatrixEncoding enum.
2014-01-04 - b317f94 - lavu 52.60.100 - mathematics.h
Add av_add_stable() function.
2013-12-22 - 911676c - lavu 52.59.100 - avstring.h
Add av_strnlen() function.
2013-12-09 - 64f73ac - lavu 52.57.100 - opencl.h
Add av_opencl_benchmark() function.
2013-11-30 - 82b2e9c - lavu 52.56.100 - ffversion.h
Moves version.h to libavutil/ffversion.h.
Install ffversion.h and make it public.
2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h
av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be
used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and
avcodec_free_frame() respectively. The latter three functions are deprecated.
2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h
Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and
stereo3d.h API, that identify codec-independent stereo3d information.
2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h
Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which
identifies ATSC A53 Part 4 Closed Captions data.
2013-11-22 - 6859065 - lavu 52.54.100 - avstring.h
Add av_utf8_decode() function.
2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h
Add HEVC profiles
2013-11-20 - c28b61c - lavc 55.44.100 - avcodec.h
Add av_packet_{un,}pack_dictionary()
Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value
strings between a stream and the application.
2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h
Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil.
2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h
Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate'
private option of the mpegvideo encoder family.
2013-11-14 - 31c09b7 / 728c465 - lavc 55.42.100 / 55.26.0 - vdpau.h
Add av_vdpau_get_profile().
Add av_vdpau_alloc_context(). This function must from now on be
used for allocating AVVDPAUContext.
2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h
lavu 52.51.100 - frame.h
Add ITU-R BT.2020 and other not yet included values to color primaries,
transfer characteristics and colorspaces.
2013-11-04 - 85cabf1 - lavu 52.50.100 - avutil.h
Add av_fopen_utf8()
2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h
2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
2013-10-27 - d61617a - lavu 52.48.100 - parseutils.h
2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
Add av_get_known_color_name().
2013-10-17 - 8696e51 - lavu 52.47.100 - opt.h
2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
av_opt_get_channel_layout() and av_opt_set_channel_layout().
2013-10-06 - ccf96f8 -libswscale 2.5.101 - options.c
2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
Change default scaler to bicubic
2013-10-03 - e57dba0 - lavc 55.34.100 - avcodec.h
2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
Add av_codec_get_max_lowres()
2013-10-02 - 5082fcc - lavf 55.19.100 - avformat.h
2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
decoders
2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h
2013-08-xx - xxxxxxx - lavfi 3.11.0 - avfilter.h
Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
implementations.
2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h
2013-09-21 - xxxxxxx - lavu 52.16.0 - pixfmt.h
Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
AV_PIX_FMT_NV20.
@@ -364,7 +49,7 @@ API changes, most recent first:
2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
avcodec_close() argument can be NULL.
2013-09-04 - 36cd017a - lavf 55.16.101 - avformat.h
2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
avformat_close_input() argument can be NULL and point on NULL.
2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
@@ -373,10 +58,10 @@ API changes, most recent first:
2013-08-15 - 1e0e193 - lsws 2.5.100 -
Add a sws_dither AVOption, allowing to set the dither algorithm used
2013-08-11 - d404fe35 - lavc 55.27.100 - vdpau.h
2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
Add a render2 alternative to the render callback function.
2013-08-11 - af05edc - lavc 55.26.100 - vdpau.h
2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
Add allocation function for AVVDPAUContext, allowing
to extend it in the future without breaking ABI/API.
@@ -386,7 +71,7 @@ API changes, most recent first:
2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
Deprecate the bitstream-related members from struct AVVDPAUContext.
The bitstream buffers no longer need to be explicitly freed.
The bistream buffers no longer need to be explicitly freed.
2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
@@ -977,9 +662,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
Add avformat_close_input().
Deprecate av_close_input_file() and av_close_input_stream().
2011-12-09 - c59b80c / b2890f5 - lavu 51.32.0 / 51.20.0 - audioconvert.h
Expand the channel layout list.
2011-12-02 - e4de716 / 0eea212 - lavc 53.40.0 / 53.25.0
Add nb_samples and extended_data fields to AVFrame.
Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.
@@ -993,10 +675,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
Change AVCodecContext.error[4] to [8] at next major bump.
Add AV_NUM_DATA_POINTERS to simplify the bump transition.
2011-11-24 - lavu 51.29.0 / 51.19.0
92afb43 / bd97b2e - add planar RGB pixel formats
92afb43 / 6b0768e - add PIX_FMT_PLANAR and PIX_FMT_RGB pixel descriptions
2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0
Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and
av_samples_alloc(), to samplefmt.h.

View File

@@ -31,7 +31,7 @@ PROJECT_NAME = FFmpeg
# This could be handy for archiving the generated documentation or
# if some version control system is used.
PROJECT_NUMBER = 2.3.1
PROJECT_NUMBER = 2.1.5
# With the PROJECT_LOGO tag one can specify a logo or icon that is included
# in the documentation. The maximum height of the logo should not exceed 55
@@ -759,7 +759,7 @@ ALPHABETICAL_INDEX = YES
# the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
# in which this list will be split (can be a number in the range [1..20])
COLS_IN_ALPHA_INDEX = 5
COLS_IN_ALPHA_INDEX = 2
# In case all classes in a project start with a common prefix, all
# classes will be put under the same header in the alphabetical index.
@@ -793,13 +793,13 @@ HTML_FILE_EXTENSION = .html
# each generated HTML page. If it is left blank doxygen will generate a
# standard header.
HTML_HEADER =
#HTML_HEADER = doc/doxy/header.html
# The HTML_FOOTER tag can be used to specify a personal HTML footer for
# each generated HTML page. If it is left blank doxygen will generate a
# standard footer.
HTML_FOOTER =
#HTML_FOOTER = doc/doxy/footer.html
# The HTML_STYLESHEET tag can be used to specify a user-defined cascading
# style sheet that is used by each HTML page. It can be used to
@@ -808,7 +808,7 @@ HTML_FOOTER =
# the style sheet file to the HTML output directory, so don't put your own
# stylesheet in the HTML output directory as well, or it will be erased!
HTML_STYLESHEET =
#HTML_STYLESHEET = doc/doxy/doxy_stylesheet.css
# The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
# Doxygen will adjust the colors in the stylesheet and background images
@@ -1056,7 +1056,7 @@ FORMULA_TRANSPARENT = YES
# typically be disabled. For large projects the javascript based search engine
# can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution.
SEARCHENGINE = YES
SEARCHENGINE = NO
# When the SERVER_BASED_SEARCH tag is enabled the search engine will be
# implemented using a PHP enabled web server instead of at the web client
@@ -1359,8 +1359,6 @@ PREDEFINED = "__attribute__(x)=" \
"DECLARE_ALIGNED(a,t,n)=t n" \
"offsetof(x,y)=0x42" \
av_alloc_size \
AV_GCC_VERSION_AT_LEAST(x,y)=1 \
__GNUC__=1 \
# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
# this tag can be used to specify a list of macro names that should be expanded.

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) $(LIBRARIES-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
@@ -36,26 +36,6 @@ DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec
DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
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_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
@@ -63,9 +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
@@ -106,14 +84,9 @@ doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
OBJDIRS += doc/examples
DOXY_INPUT = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
doc/doxy/html: TAG = DOXY
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
install-doc: install-html install-man
@@ -147,7 +120,7 @@ uninstall-html:
$(RM) -r "$(DOCDIR)"
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
@@ -155,13 +128,8 @@ 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
=============
* 2.1 "Fourier" October, 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.

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

@@ -74,18 +74,7 @@ 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
@@ -128,17 +117,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@section movsub
@section mp3_header_compress
@section mp3_header_decompress
@section noise
Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
@example
ffmpeg -i INPUT -c copy -bsf noise output.mkv
@end example
@section remove_extra
@c man end BITSTREAM FILTERS

View File

@@ -25,9 +25,6 @@ fate-list
install
Install headers, libraries and programs.
examples
Build all examples located in doc/examples.
libavformat/output-example
Build the libavformat basic example.
@@ -37,9 +34,6 @@ libavcodec/api-example
libswscale/swscale-test
Build the swscale self-test (useful also as example).
config
Reconfigure the project with current configuration.
Useful standard make commands:
make -t <target>

View File

@@ -172,13 +172,7 @@ 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.
@@ -285,11 +279,6 @@ 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
@@ -394,9 +383,6 @@ Possible values:
@item simplemmx
@item simpleauto
Automatically pick a IDCT compatible with the simple one
@item arm
@item altivec
@@ -432,8 +418,6 @@ 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}
@@ -887,9 +871,6 @@ 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.
@@ -1035,26 +1016,15 @@ 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.
@@ -1113,9 +1083,5 @@ instead of alpha. Default is 0.
@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
@@ -52,37 +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 ffwavesynth
Internal wave synthetizer.

View File

@@ -17,8 +17,8 @@ a:visited {
}
#banner img {
margin-bottom: 1px;
margin-top: 5px;
padding-bottom: 1px;
padding-top: 5px;
}
#body {

View File

@@ -74,7 +74,7 @@ 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
@@ -92,22 +92,6 @@ 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{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
@@ -128,14 +112,6 @@ If set to 0, any file name is accepted.
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.
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.
@end table
@section flv
@@ -273,8 +249,6 @@ 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.
@@ -322,7 +296,7 @@ teletext packet PTS and DTS values untouched.
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

@@ -92,7 +92,7 @@ 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 */
@@ -248,7 +248,7 @@ 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://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

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

@@ -2,12 +2,13 @@
SRC_PATH="${1}"
DOXYFILE="${2}"
DOXYGEN="${3}"
shift 3
shift 2
$DOXYGEN - <<EOF
doxygen - <<EOF
@INCLUDE = ${DOXYFILE}
INPUT = $@
EXAMPLE_PATH = ${SRC_PATH}/doc/examples
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

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
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@@ -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">

View File

@@ -14,7 +14,7 @@ You can disable all the encoders with the configure option
with the options @code{--enable-encoder=@var{ENCODER}} /
@code{--disable-encoder=@var{ENCODER}}.
The option @code{-encoders} of the ff* tools will display the list of
The option @code{-codecs} of the ff* tools will display the list of
enabled encoders.
@c man end ENCODERS
@@ -38,8 +38,8 @@ As this encoder is experimental, unexpected behavior may exist from time to
time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
that it has a worse quality reported by some users.
@c todo @ref{libaacplus}
See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.
@c Comment this out until somebody writes the respective documentation.
@c See also @ref{libfaac}, @ref{libaacplus}, and @ref{libfdk-aac-enc}.
@subsection Options
@@ -80,7 +80,7 @@ thresholds with quantizer steps to find the appropriate quantization with
distortion below threshold band by band.
The quality of this method is comparable to the two loop searching method
described below, but somewhat a little better and slower.
descibed below, but somewhat a little better and slower.
@item anmr
Average noise to mask ratio (ANMR) trellis-based solution.
@@ -494,285 +494,6 @@ Selected by Encoder (default)
@end table
@anchor{libfaac}
@section libfaac
libfaac AAC (Advanced Audio Coding) encoder wrapper.
Requires the presence of the libfaac headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libfaac --enable-nonfree}.
This encoder is considered to be of higher quality with respect to the
@ref{aacenc,,the native experimental FFmpeg AAC encoder}.
For more information see the libfaac project at
@url{http://www.audiocoding.com/faac.html/}.
@subsection Options
The following shared FFmpeg codec options are recognized.
The following options are supported by the libfaac wrapper. The
@command{faac}-equivalent of the options are listed in parentheses.
@table @option
@item b (@emph{-b})
Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
is not explicitly specified, it is automatically set to a suitable
value depending on the selected profile. @command{faac} bitrate is
expressed in kilobits/s.
Note that libfaac does not support CBR (Constant Bit Rate) but only
ABR (Average Bit Rate).
If VBR mode is enabled this option is ignored.
@item ar (@emph{-R})
Set audio sampling rate (in Hz).
@item ac (@emph{-c})
Set the number of audio channels.
@item cutoff (@emph{-C})
Set cutoff frequency. If not specified (or explicitly set to 0) it
will use a value automatically computed by the library. Default value
is 0.
@item profile
Set audio profile.
The following profiles are recognized:
@table @samp
@item aac_main
Main AAC (Main)
@item aac_low
Low Complexity AAC (LC)
@item aac_ssr
Scalable Sample Rate (SSR)
@item aac_ltp
Long Term Prediction (LTP)
@end table
If not specified it is set to @samp{aac_low}.
@item flags +qscale
Set constant quality VBR (Variable Bit Rate) mode.
@item global_quality
Set quality in VBR mode as an integer number of lambda units.
Only relevant when VBR mode is enabled with @code{flags +qscale}. The
value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
and used to set the quality value used by libfaac. A reasonable range
for the option value in QP units is [10-500], the higher the value the
higher the quality.
@item q (@emph{-q})
Enable VBR mode when set to a non-negative value, and set constant
quality value as a double floating point value in QP units.
The value sets the quality value used by libfaac. A reasonable range
for the option value is [10-500], the higher the value the higher the
quality.
This option is valid only using the @command{ffmpeg} command-line
tool. For library interface users, use @option{global_quality}.
@end table
@subsection Examples
@itemize
@item
Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
container:
@example
ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
@end example
@item
Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
LTP AAC profile:
@example
ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
@end example
@end itemize
@anchor{libfdk-aac-enc}
@section libfdk_aac
libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.
The libfdk-aac library is based on the Fraunhofer FDK AAC code from
the Android project.
Requires the presence of the libfdk-aac headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libfdk-aac}. The library is also incompatible with GPL,
so if you allow the use of GPL, you should configure with
@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
This encoder is considered to be of higher quality with respect to
both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
@ref{libfaac}.
VBR encoding, enabled through the @option{vbr} or @option{flags
+qscale} options, is experimental and only works with some
combinations of parameters.
Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or
higher.
For more information see the fdk-aac project at
@url{http://sourceforge.net/p/opencore-amr/fdk-aac/}.
@subsection Options
The following options are mapped on the shared FFmpeg codec options.
@table @option
@item b
Set bit rate in bits/s. If the bitrate is not explicitly specified, it
is automatically set to a suitable value depending on the selected
profile.
In case VBR mode is enabled the option is ignored.
@item ar
Set audio sampling rate (in Hz).
@item channels
Set the number of audio channels.
@item flags +qscale
Enable fixed quality, VBR (Variable Bit Rate) mode.
Note that VBR is implicitly enabled when the @option{vbr} value is
positive.
@item cutoff
Set cutoff frequency. If not specified (or explicitly set to 0) it
will use a value automatically computed by the library. Default value
is 0.
@item profile
Set audio profile.
The following profiles are recognized:
@table @samp
@item aac_low
Low Complexity AAC (LC)
@item aac_he
High Efficiency AAC (HE-AAC)
@item aac_he_v2
High Efficiency AAC version 2 (HE-AACv2)
@item aac_ld
Low Delay AAC (LD)
@item aac_eld
Enhanced Low Delay AAC (ELD)
@end table
If not specified it is set to @samp{aac_low}.
@end table
The following are private options of the libfdk_aac encoder.
@table @option
@item afterburner
Enable afterburner feature if set to 1, disabled if set to 0. This
improves the quality but also the required processing power.
Default value is 1.
@item eld_sbr
Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled
if set to 0.
Default value is 0.
@item signaling
Set SBR/PS signaling style.
It can assume one of the following values:
@table @samp
@item default
choose signaling implicitly (explicit hierarchical by default,
implicit if global header is disabled)
@item implicit
implicit backwards compatible signaling
@item explicit_sbr
explicit SBR, implicit PS signaling
@item explicit_hierarchical
explicit hierarchical signaling
@end table
Default value is @samp{default}.
@item latm
Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
Default value is 0.
@item header_period
Set StreamMuxConfig and PCE repetition period (in frames) for sending
in-band configuration buffers within LATM/LOAS transport layer.
Must be a 16-bits non-negative integer.
Default value is 0.
@item vbr
Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty
good) and 5 is highest quality. A value of 0 will disable VBR, and CBR
(Constant Bit Rate) is enabled.
Currently only the @samp{aac_low} profile supports VBR encoding.
VBR modes 1-5 correspond to roughly the following average bit rates:
@table @samp
@item 1
32 kbps/channel
@item 2
40 kbps/channel
@item 3
48-56 kbps/channel
@item 4
64 kbps/channel
@item 5
about 80-96 kbps/channel
@end table
Default value is 0.
@end table
@subsection Examples
@itemize
@item
Use @command{ffmpeg} to convert an audio file to VBR AAC in an M4A (MP4)
container:
@example
ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
@end example
@item
Use @command{ffmpeg} to convert an audio file to CBR 64k kbps AAC, using the
High-Efficiency AAC profile:
@example
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
@end example
@end itemize
@anchor{libmp3lame}
@section libmp3lame
@@ -792,7 +513,7 @@ The following options are supported by the libmp3lame wrapper. The
@table @option
@item b (@emph{-b})
Set bitrate expressed in bits/s for CBR or ABR. LAME @code{bitrate} is
Set bitrate expressed in bits/s for CBR. LAME @code{bitrate} is
expressed in kilobits/s.
@item q (@emph{-V})
@@ -807,18 +528,13 @@ while producing the worst quality.
@item reservoir
Enable use of bit reservoir when set to 1. Default value is 1. LAME
has this enabled by default, but can be overridden by use
has this enabled by default, but can be overriden by use
@option{--nores} option.
@item joint_stereo (@emph{-m j})
Enable the encoder to use (on a frame by frame basis) either L/R
stereo or mid/side stereo. Default value is 1.
@item abr (@emph{--abr})
Enable the encoder to use ABR when set to 1. The @command{lame}
@option{--abr} sets the target bitrate, while this options only
tells FFmpeg to use ABR still relies on @option{b} to set bitrate.
@end table
@section libopencore-amrnb
@@ -1147,111 +863,32 @@ transient response is a higher bitrate.
@end table
@anchor{libwavpack}
@section libwavpack
A wrapper providing WavPack encoding through libwavpack.
Only lossless mode using 32-bit integer samples is supported currently.
Requires the presence of the libwavpack headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libwavpack}.
Note that a libavcodec-native encoder for the WavPack codec exists so users can
encode audios with this codec without using this encoder. See @ref{wavpackenc}.
@subsection Options
@command{wavpack} command line utility's corresponding options are listed in
parentheses, if any.
The @option{compression_level} option can be used to control speed vs.
compression tradeoff, with the values mapped to libwavpack as follows:
@table @option
@item frame_size (@emph{--blocksize})
Default is 32768.
@item compression_level
Set speed vs. compression tradeoff. Acceptable arguments are listed below:
@table @samp
@item 0 (@emph{-f})
Fast mode.
@item 0
Fast mode - corresponding to the wavpack @option{-f} option.
@item 1
Normal (default) settings.
@item 2 (@emph{-h})
High quality.
@item 2
High quality - corresponding to the wavpack @option{-h} option.
@item 3 (@emph{-hh})
Very high quality.
@item 3
Very high quality - corresponding to the wavpack @option{-hh} option.
@item 4-8 (@emph{-hh -x}@var{EXTRAPROC})
Same as @samp{3}, but with extra processing enabled.
@samp{4} is the same as @option{-x2} and @samp{8} is the same as @option{-x6}.
@end table
@end table
@anchor{wavpackenc}
@section wavpack
WavPack lossless audio encoder.
This is a libavcodec-native WavPack encoder. There is also an encoder based on
libwavpack, but there is virtually no reason to use that encoder.
See also @ref{libwavpack}.
@subsection Options
The equivalent options for @command{wavpack} command line utility are listed in
parentheses.
@subsubsection Shared options
The following shared options are effective for this encoder. Only special notes
about this particular encoder will be documented here. For the general meaning
of the options, see @ref{codec-options,,the Codec Options chapter}.
@table @option
@item frame_size (@emph{--blocksize})
For this encoder, the range for this option is between 128 and 131072. Default
is automatically decided based on sample rate and number of channel.
For the complete formula of calculating default, see
@file{libavcodec/wavpackenc.c}.
@item compression_level (@emph{-f}, @emph{-h}, @emph{-hh}, and @emph{-x})
This option's syntax is consistent with @ref{libwavpack}'s.
@end table
@subsubsection Private options
@table @option
@item joint_stereo (@emph{-j})
Set whether to enable joint stereo. Valid values are:
@table @samp
@item on (@emph{1})
Force mid/side audio encoding.
@item off (@emph{0})
Force left/right audio encoding.
@item auto
Let the encoder decide automatically.
@end table
@item optimize_mono
Set whether to enable optimization for mono. This option is only effective for
non-mono streams. Available values:
@table @samp
@item on
enabled
@item off
disabled
@end table
@item 4-8
Same as 3, but with extra processing enabled - corresponding to the wavpack
@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as
@option{-x6}.
@end table
@@ -1265,15 +902,12 @@ follows.
@section libtheora
libtheora Theora encoder wrapper.
Theora format supported through libtheora.
Requires the presence of the libtheora headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libtheora}.
For more information about the libtheora project see
@url{http://www.theora.org/}.
@subsection Options
The following global options are mapped to internal libtheora options
@@ -1281,11 +915,11 @@ which affect the quality and the bitrate of the encoded stream.
@table @option
@item b
Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In
case VBR (Variable Bit Rate) mode is enabled this option is ignored.
Set the video bitrate, only works if the @code{qscale} flag in
@option{flags} is not enabled.
@item flags
Used to enable constant quality mode (VBR) encoding through the
Used to enable constant quality mode encoding through the
@option{qscale} flag, and to enable the @code{pass1} and @code{pass2}
modes.
@@ -1293,41 +927,19 @@ modes.
Set the GOP size.
@item global_quality
Set the global quality as an integer in lambda units.
Set the global quality in lambda units, only works if the
@code{qscale} flag in @option{flags} is enabled. The value is clipped
in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3
to get a value in the native libtheora range [0-63]. A higher value
corresponds to a higher quality.
Only relevant when VBR mode is enabled with @code{flags +qscale}. The
value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
value in the native libtheora range [0-63]. A higher value corresponds
to a higher quality.
@item q
Enable VBR mode when set to a non-negative value, and set constant
quality value as a double floating point value in QP units.
The value is clipped in the [0-10] range, and then multiplied by 6.3
to get a value in the native libtheora range [0-63].
This option is valid only using the @command{ffmpeg} command-line
tool. For library interface users, use @option{global_quality}.
For example, to set maximum constant quality encoding with
@command{ffmpeg}:
@example
ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg
@end example
@end table
@subsection Examples
@itemize
@item
Set maximum constant quality (VBR) encoding with @command{ffmpeg}:
@example
ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
@end example
@item
Use @command{ffmpeg} to convert a CBR 1000 kbps Theora video stream:
@example
ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
@end example
@end itemize
@section libvpx
VP8 format supported through libvpx.
@@ -1447,69 +1059,7 @@ g_error_resilient
For more information about libvpx see:
@url{http://www.webmproject.org/}
@section libwebp
libwebp WebP Image encoder wrapper
libwebp is Google's official encoder for WebP images. It can encode in either
lossy or lossless mode. Lossy images are essentially a wrapper around a VP8
frame. Lossless images are a separate codec developed by Google.
@subsection Pixel Format
Currently, libwebp only supports YUV420 for lossy and RGB for lossless due
to limitations of the format and libwebp. Alpha is supported for either mode.
Because of API limitations, if RGB is passed in when encoding lossy or YUV is
passed in for encoding lossless, the pixel format will automatically be
converted using functions from libwebp. This is not ideal and is done only for
convenience.
@subsection Options
@table @option
@item -lossless @var{boolean}
Enables/Disables use of lossless mode. Default is 0.
@item -compression_level @var{integer}
For lossy, this is a quality/speed tradeoff. Higher values give better quality
for a given size at the cost of increased encoding time. For lossless, this is
a size/speed tradeoff. Higher values give smaller size at the cost of increased
encoding time. More specifically, it controls the number of extra algorithms
and compression tools used, and varies the combination of these tools. This
maps to the @var{method} option in libwebp. The valid range is 0 to 6.
Default is 4.
@item -qscale @var{float}
For lossy encoding, this controls image quality, 0 to 100. For lossless
encoding, this controls the effort and time spent at compressing more. The
default value is 75. Note that for usage via libavcodec, this option is called
@var{global_quality} and must be multiplied by @var{FF_QP2LAMBDA}.
@item -preset @var{type}
Configuration preset. This does some automatic settings based on the general
type of the image.
@table @option
@item none
Do not use a preset.
@item default
Use the encoder default.
@item picture
Digital picture, like portrait, inner shot
@item photo
Outdoor photograph, with natural lighting
@item drawing
Hand or line drawing, with high-contrast details
@item icon
Small-sized colorful images
@item text
Text-like
@end table
@end table
@section libx264, libx264rgb
@section libx264
x264 H.264/MPEG-4 AVC encoder wrapper.
@@ -1525,22 +1075,12 @@ for detail retention (adaptive quantization, psy-RD, psy-trellis).
Many libx264 encoder options are mapped to FFmpeg global codec
options, while unique encoder options are provided through private
options. Additionally the @option{x264opts} and @option{x264-params}
private options allows one to pass a list of key=value tuples as accepted
private options allows to pass a list of key=value tuples as accepted
by the libx264 @code{x264_param_parse} function.
The x264 project website is at
@url{http://www.videolan.org/developers/x264.html}.
The libx264rgb encoder is the same as libx264, except it accepts packed RGB
pixel formats as input instead of YUV.
@subsection Supported Pixel Formats
x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at
x264's configure time. FFmpeg only supports one bit depth in one particular
build. In other words, it is not possible to build one FFmpeg with multiple
versions of x264 with different bit depths.
@subsection Options
The following options are supported by the libx264 wrapper. The
@@ -1566,34 +1106,25 @@ kilobits/s.
@item g (@emph{keyint})
@item qmin (@emph{qpmin})
Minimum quantizer scale.
@item qmax (@emph{qpmax})
Maximum quantizer scale.
@item qmin (@emph{qpmin})
@item qdiff (@emph{qpstep})
Maximum difference between quantizer scales.
@item qblur (@emph{qblur})
Quantizer curve blur
@item qcomp (@emph{qcomp})
Quantizer curve compression factor
@item refs (@emph{ref})
Number of reference frames each P-frame can use. The range is from @var{0-16}.
@item sc_threshold (@emph{scenecut})
Sets the threshold for the scene change detection.
@item trellis (@emph{trellis})
Performs Trellis quantization to increase efficiency. Enabled by default.
@item nr (@emph{nr})
@item me_range (@emph{merange})
Maximum range of the motion search in pixels.
@item me_method (@emph{me})
Set motion estimation method. Possible values in the decreasing order
@@ -1615,13 +1146,10 @@ Hadamard exhaustive search (slowest).
@end table
@item subq (@emph{subme})
Sub-pixel motion estimation method.
@item b_strategy (@emph{b-adapt})
Adaptive B-frame placement decision algorithm. Use only on first-pass.
@item keyint_min (@emph{min-keyint})
Minimum GOP size.
@item coder
Set entropy encoder. Possible values:
@@ -1648,7 +1176,6 @@ Ignore chroma in motion estimation. It generates the same effect as
@end table
@item threads (@emph{threads})
Number of encoding threads.
@item thread_type
Set multithreading technique. Possible values:
@@ -1866,7 +1393,7 @@ Override the x264 configuration using a :-separated list of key=value
parameters.
This option is functionally the same as the @option{x264opts}, but is
duplicated for compatibility with the Libav fork.
duplicated for compability with the Libav fork.
For example to specify libx264 encoding options with @command{ffmpeg}:
@example
@@ -2042,30 +1569,6 @@ fastest.
@end table
@section mpeg2
MPEG-2 video encoder.
@subsection Options
@table @option
@item seq_disp_ext @var{integer}
Specifies if the encoder should write a sequence_display_extension to the
output.
@table @option
@item -1
@itemx auto
Decide automatically to write it or not (this is the default) by checking if
the data to be written is different from the default or unspecified values.
@item 0
@itemx never
Never write it.
@item 1
@itemx always
Always write it.
@end table
@end table
@section png
PNG image encoder.
@@ -2084,7 +1587,7 @@ Set physical density of pixels, in dots per meter, unset by default
Apple ProRes encoder.
FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
The used encoder can be chosen with the @code{-vcodec} option.
The used encoder can be choosen with the @code{-vcodec} option.
@subsection Private Options for prores-ks
@@ -2147,27 +1650,3 @@ For the fastest encoding speed set the @option{qscale} parameter (4 is the
recommended value) and do not set a size constraint.
@c man end VIDEO ENCODERS
@chapter Subtitles Encoders
@c man begin SUBTITLES ENCODERS
@section dvdsub
This codec encodes the bitmap subtitle format that is used in DVDs.
Typically they are stored in VOBSUB file pairs (*.idx + *.sub),
and they can also be used in Matroska files.
@subsection Options
@table @option
@item even_rows_fix
When set to 1, enable a work-around that makes the number of pixel rows
even in all subtitles. This fixes a problem with some players that
cut off the bottom row if the number is odd. The work-around just adds
a fully transparent row if needed. The overhead is low, typically
one byte per subtitle on average.
By default, this work-around is disabled.
@end table
@c man end SUBTITLES ENCODERS

View File

@@ -11,23 +11,19 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_reading \
decoding_encoding \
demuxing_decoding \
EXAMPLES= decoding_encoding \
demuxing \
filtering_video \
filtering_audio \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
OBJS=$(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

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
doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

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, 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;
@@ -231,7 +227,7 @@ static void audio_encode_example(const char *filename)
fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_free_frame(&frame);
avcodec_close(c);
av_free(c);
}
@@ -291,11 +287,12 @@ static void audio_decode_example(const char *outfilename, const char *filename)
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) {
@@ -307,11 +304,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
@@ -337,7 +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);
}
/*
@@ -374,18 +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 */
@@ -400,7 +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);
@@ -419,7 +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;
@@ -427,15 +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;
}
@@ -481,7 +467,7 @@ static void video_encode_example(const char *filename, int codec_id)
avcodec_close(c);
av_free(c);
av_freep(&frame->data[0]);
av_frame_free(&frame);
avcodec_free_frame(&frame);
printf("\n");
}
@@ -495,10 +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);
}
@@ -579,14 +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;
@@ -623,7 +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");
}
@@ -640,7 +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;
}

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>
@@ -53,30 +53,16 @@ static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
/* The different ways of decoding and managing data memory. You are not
* supposed to support all the modes in your application but pick the one most
* appropriate to your needs. Look for the use of api_mode in this example to
* see what are the differences of API usage between them */
enum {
API_MODE_OLD = 0, /* old method, deprecated */
API_MODE_NEW_API_REF_COUNT = 1, /* new method, using the frame reference counting */
API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
};
static int api_mode = API_MODE_OLD;
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;
}
@@ -99,7 +85,7 @@ 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
@@ -127,11 +113,6 @@ static int decode_packet(int *got_frame, int cached)
}
}
/* If we use the new API with reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
av_frame_unref(frame);
return decoded;
}
@@ -142,7 +123,6 @@ static int open_codec_context(int *stream_idx,
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) {
@@ -159,13 +139,10 @@ 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 */
if (api_mode == API_MODE_NEW_API_REF_COUNT)
av_dict_set(&opts, "refcounted_frames", "1", 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;
@@ -208,31 +185,15 @@ int main (int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_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. If unset, it's using\n"
"the classic old method.\n"
"audio frames to a rawaudio file named audio_output_file.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5) {
const char *mode = argv[1] + strlen("-refcount=");
if (!strcmp(mode, "old")) api_mode = API_MODE_OLD;
else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
else if (!strcmp(mode, "new_refcount")) api_mode = API_MODE_NEW_API_REF_COUNT;
else {
fprintf(stderr, "unknow mode '%s'\n", mode);
exit(1);
}
argv++;
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
@@ -279,7 +240,7 @@ int main (int argc, char **argv)
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;
}
@@ -294,12 +255,7 @@ int main (int argc, char **argv)
goto end;
}
/* When using the new API, you need to use the libavutil/frame.h API, while
* the classic frame management is available in libavcodec */
if (api_mode == API_MODE_OLD)
frame = avcodec_alloc_frame();
else
frame = av_frame_alloc();
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
@@ -369,17 +325,16 @@ int main (int argc, char **argv)
}
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);
if (api_mode == API_MODE_OLD)
avcodec_free_frame(&frame);
else
av_frame_free(&frame);
av_free(frame);
av_free(video_dst_data[0]);
return ret < 0;

View File

@@ -1,364 +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/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>
@@ -85,7 +85,7 @@ 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();
@@ -97,10 +97,6 @@ static int init_filters(const char *filters_descr)
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)
@@ -113,7 +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. */
@@ -121,28 +117,28 @@ 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;
}
/* Endpoints for the filter graph. */
@@ -157,11 +153,11 @@ static int init_filters(const char *filters_descr)
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 */
@@ -172,11 +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)
@@ -196,7 +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;
@@ -210,6 +202,7 @@ int main(int argc, char **argv)
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
@@ -219,24 +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 */
@@ -248,31 +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_free_packet(&packet0);
} else {
/* discard non-wanted packets */
av_free_packet(&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 */
@@ -36,7 +36,6 @@
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24";
@@ -71,7 +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) {
@@ -85,18 +83,15 @@ 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();
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),
@@ -109,22 +104,18 @@ static int init_filters(const char *filters_descr)
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;
}
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;
return ret;
}
/* Endpoints for the filter graph. */
@@ -140,16 +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)
@@ -200,6 +186,7 @@ int main(int argc, char **argv)
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
@@ -214,6 +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) {
@@ -240,20 +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_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

@@ -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>
@@ -34,64 +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;
/* pts of the next frame that will be generated */
int64_t next_pts;
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
struct SwsContext *sws_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;
AVStream *st;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
@@ -101,22 +63,20 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
exit(1);
}
ost->st = avformat_new_stream(oc, *codec);
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 = ost->st->codec;
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;
c->channels = 2;
c->channel_layout = AV_CH_LAYOUT_STEREO;
break;
case AVMEDIA_TYPE_VIDEO:
@@ -153,21 +113,32 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
/**************************************************************/
/* audio output */
int samples_count;
static float t, tincr, tincr2;
static uint8_t **src_samples_data;
static int src_samples_linesize;
static int src_nb_samples;
static int max_dst_nb_samples;
uint8_t **dst_samples_data;
int dst_samples_linesize;
int dst_samples_size;
struct SwrContext *swr_ctx = NULL;
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
AVCodecContext *c;
int ret;
c = ost->st->codec;
c = st->codec;
/* open it */
ret = avcodec_open2(c, codec, NULL);
@@ -177,32 +148,20 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
}
/* 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;
ost->frame = av_frame_alloc();
if (!ost->frame)
src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
10000 : c->frame_size;
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 allocate source samples\n");
exit(1);
ost->frame->sample_rate = c->sample_rate;
ost->frame->format = AV_SAMPLE_FMT_S16;
ost->frame->channel_layout = c->channel_layout;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
ost->frame->nb_samples = 10000;
else
ost->frame->nb_samples = c->frame_size;
ost->tmp_frame = av_frame_alloc();
if (!ost->frame)
exit(1);
ost->tmp_frame->sample_rate = c->sample_rate;
ost->tmp_frame->format = c->sample_fmt;
ost->tmp_frame->channel_layout = c->channel_layout;
ost->tmp_frame->nb_samples = ost->frame->nb_samples;
}
/* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
@@ -227,143 +186,121 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
}
}
ret = av_frame_get_buffer(ost->frame, 0);
/* 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 an audio frame.\n");
exit(1);
}
ret = av_frame_get_buffer(ost->tmp_frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate an audio frame.\n");
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)
{
int j, i, v, ret;
int16_t *q = (int16_t*)ost->frame->data[0];
int j, i, v;
int16_t *q;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
for (j = 0; j < ost->frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->st->codec->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;
}
ost->frame->pts = ost->next_pts;
ost->next_pts += ost->frame->nb_samples;
return ost->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->st->codec;
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 */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(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);
/* convert to destination format */
ret = swr_convert(swr_ctx,
ost->tmp_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->tmp_frame;
} else {
dst_nb_samples = frame->nb_samples;
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(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
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)
static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
int ret;
AVCodecContext *c = ost->st->codec;
AVCodecContext *c = st->codec;
/* open the codec */
ret = avcodec_open2(c, codec, NULL);
@@ -373,38 +310,40 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
}
/* 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 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;
@@ -422,77 +361,53 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->st->codec;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->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;
static struct SwsContext *sws_ctx;
AVCodecContext *c = st->codec;
c = ost->st->codec;
frame = get_video_frame(ost);
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 {
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 (oc->oformat->flags & AVFMT_RAWPICTURE) {
/* a hack to avoid data copy with some raw video muxers */
/* Raw video case - directly store the picture in the packet */
AVPacket pkt;
av_init_packet(&pkt);
if (!frame)
return 1;
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = ost->st->index;
pkt.data = (uint8_t *)frame;
pkt.stream_index = st->index;
pkt.data = dst_picture.data[0];
pkt.size = sizeof(AVPicture);
pkt.pts = pkt.dts = frame->pts;
av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
ret = av_interleaved_write_frame(oc, &pkt);
} else {
AVPacket pkt = { 0 };
int got_packet;
av_init_packet(&pkt);
/* encode the image */
@@ -501,28 +416,30 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
/* If size is zero, it means the image was buffered. */
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
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) {
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_close(ost->st->codec);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
avcodec_close(st->codec);
av_free(src_picture.data[0]);
av_free(dst_picture.data[0]);
av_free(frame);
}
/**************************************************************/
@@ -530,14 +447,13 @@ 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;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
@@ -561,31 +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);
if (have_audio)
open_audio(oc, audio_codec, &audio_st);
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);
@@ -607,14 +521,23 @@ int main(int argc, char **argv)
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.st->codec->time_base,
audio_st.next_pts, audio_st.st->codec->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);
}
}
@@ -625,10 +548,10 @@ 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. */

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 |= 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_free_packet(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_close(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);

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>

View File

@@ -1,755 +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 48000
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/** The audio sample output format */
#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
/**
* Convert an error code into a text message.
* @param error Error code to be converted
* @return Corresponding error text (not thread-safe)
*/
static char *const 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)
{
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]->codec->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** Save the decoder context for easier access later. */
*input_codec_context = (*input_format_context)->streams[0]->codec;
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)
{
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, output_codec))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/** Save the encoder context for easiert access later. */
*output_codec_context = stream->codec;
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
(*output_codec_context)->channels = OUTPUT_CHANNELS;
(*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
(*output_codec_context)->sample_rate = input_codec_context->sample_rate;
(*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16;
(*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
/**
* 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)
(*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
/** Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
get_error_text(error));
goto cleanup;
}
return 0;
cleanup:
avio_close((*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)
{
/** Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_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 the 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_free_packet(&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_free_packet(&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;
}
/** 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);
/**
* 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_free_packet(&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_free_packet(&output_packet);
return error;
}
av_free_packet(&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))
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_close(output_codec_context);
if (output_format_context) {
avio_close(output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_close(input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}

View File

@@ -1,597 +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/avcodec.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);
/* 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 */
enc_ctx->pix_fmt = encoder->pix_fmts[0];
/* 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 |= 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;
enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.duration = av_rescale_q(enc_pkt.duration,
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 &
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;
}
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
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 */
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_free_packet(&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_free_packet(&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_close(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;
}

View File

@@ -368,6 +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
@@ -392,7 +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

View File

@@ -14,7 +14,7 @@
The FFmpeg resampler provides a 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

View File

@@ -13,7 +13,7 @@
@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
library image conversion utilities. In particular it allows to perform
image rescaling and pixel format conversion.
@c man end DESCRIPTION

View File

@@ -80,23 +80,11 @@ The transcoding process in @command{ffmpeg} for each output can be described by
the following diagram:
@example
_______ ______________
| | | |
| input | demuxer | encoded data | decoder
| file | ---------> | packets | -----+
|_______| |______________| |
v
_________
| |
| decoded |
| frames |
|_________|
________ ______________ |
| | | | |
| output | <-------- | encoded data | <----+
| file | muxer | packets | encoder
|________| |______________|
_______ ______________ _________ ______________ ________
| | | | | | | | | |
| input | demuxer | encoded data | decoder | decoded | encoder | encoded data | muxer | output |
| file | ---------> | packets | ---------> | frames | ---------> | packets | -------> | file |
|_______| |______________| |_________| |______________| |________|
@end example
@@ -124,16 +112,11 @@ the same type. In the above diagram they can be represented by simply inserting
an additional step between decoding and encoding:
@example
_________ ______________
| | | |
| decoded | | encoded data |
| frames |\ _ | packets |
|_________| \ /||______________|
\ __________ /
simple _\|| | / encoder
filtergraph | filtered |/
| frames |
|__________|
_________ __________ ______________
| | | | | |
| decoded | simple filtergraph | filtered | encoder | encoded data |
| frames | -------------------> | frames | ---------> | packets |
|_________| |__________| |______________|
@end example
@@ -142,10 +125,10 @@ Simple filtergraphs are configured with the per-stream @option{-filter} option
A simple filtergraph for video can look for example like this:
@example
_______ _____________ _______ ________
| | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | output |
|_______| |_____________| |_______| |________|
_______ _____________ _______ _____ ________
| | | | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
|_______| |_____________| |_______| |_____| |________|
@end example
@@ -272,13 +255,8 @@ 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}.
@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.
@@ -307,20 +285,23 @@ input until the timestamps reach @var{position}.
@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 time duration 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.
@@ -367,13 +348,8 @@ 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})
@@ -527,6 +503,9 @@ 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.
@@ -534,7 +513,7 @@ 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})
@@ -637,52 +616,6 @@ 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.
@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.
@end table
@end table
@section Audio Options
@@ -717,7 +650,7 @@ 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})
@@ -732,9 +665,11 @@ 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})
@@ -743,7 +678,7 @@ Disable subtitle recording.
Deprecated, see -bsf
@end table
@section Advanced Subtitle options
@section Advanced Subtitle options:
@table @option
@@ -1063,7 +998,7 @@ 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}
@@ -1137,38 +1072,12 @@ transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
e.g. when copying some streams and transcoding the others.
@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 -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
specified to ffserver but can be to ffmpeg.
@end table

View File

@@ -84,9 +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.
@@ -162,10 +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 presentation metadata. Set by
default, use -noautorotate to disable.
@end table
@section While playing
@@ -181,7 +174,7 @@ Toggle full screen.
Pause.
@item a
Cycle audio channel in the current program.
Cycle audio channel in the curret program.
@item v
Cycle video channel.
@@ -193,13 +186,7 @@ Cycle subtitle channel in the current program.
Cycle program.
@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.
@@ -208,8 +195,6 @@ 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 mouse click
@@ -221,7 +206,6 @@ Seek to percentage in file corresponding to fraction of width.
@include config.texi
@ifset config-all
@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset

View File

@@ -197,11 +197,11 @@ 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
@@ -337,39 +337,6 @@ 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
@@ -599,7 +566,6 @@ 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

@@ -8,15 +8,15 @@
<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="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" 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="programs" type="ffprobe:programsType" 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>
@@ -28,10 +28,7 @@
<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:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
@@ -53,10 +50,6 @@
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
@@ -65,8 +58,6 @@
<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" />
@@ -91,16 +82,6 @@
<xsd:attribute name="repeat_pict" type="xsd:int" />
</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"/>
@@ -151,8 +132,6 @@
<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="timecode" type="xsd:string"/>
<!-- audio attributes -->
@@ -171,7 +150,6 @@
<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="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
@@ -226,7 +204,8 @@
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string" use="required"/>
<xsd:attribute name="build_time" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_ident" 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>

View File

@@ -235,7 +235,7 @@ StartSendOnKey
#<Stream test.ogg>
#Feed feed1.ffm
#Metadata title "Stream title"
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
@@ -280,10 +280,10 @@ StartSendOnKey
#<Stream file.asf>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Metadata author "Me"
#Metadata copyright "Super MegaCorp"
#Metadata title "Test stream from disk"
#Metadata comment "Test comment"
#Author "Me"
#Copyright "Super MegaCorp"
#Title "Test stream from disk"
#Comment "Test comment"
#</Stream>

View File

@@ -16,14 +16,11 @@ ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION
@command{ffserver} is a streaming server for both audio and video.
It supports several live feeds, streaming from files and time shifting
on live feeds. You can seek to positions in the past on each live
feed, provided you specify a big enough feed storage.
@command{ffserver} is configured through a configuration file, which
is read at startup. If not explicitly specified, it will read from
@file{/etc/ffserver.conf}.
@command{ffserver} is a streaming server for both audio and video. It
supports several live feeds, streaming from files and time shifting on
live feeds (you can seek to positions in the past on each live feed,
provided you specify a big enough feed storage in
@file{ffserver.conf}).
@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
@@ -42,126 +39,10 @@ For each feed you can have different output streams in various
formats, each one specified by a @code{<Stream>} section in the
configuration file.
@chapter Detailed description
@command{ffserver} works by forwarding streams encoded by
@command{ffmpeg}, or pre-recorded streams which are read from disk.
Precisely, @command{ffserver} acts as an HTTP server, accepting POST
requests from @command{ffmpeg} to acquire the stream to publish, and
serving RTSP clients or HTTP clients GET requests with the stream
media content.
A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
a port where @command{ffserver} is listening.
Each feed is identified by a unique name, corresponding to the name
of the resource published on @command{ffserver}, and is configured by
a dedicated @code{Feed} section in the configuration file.
The feed publish URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
@end example
where @var{ffserver_ip_address} is the IP address of the machine where
@command{ffserver} is installed, @var{http_port} is the port number of
the HTTP server (configured through the @option{Port} option), and
@var{feed_name} is the name of the corresponding feed defined in the
configuration file.
Each feed is associated to a file which is stored on disk. This stored
file is used to allow to send pre-recorded data to a player as fast as
possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by
@command{ffserver}, and made accessible through the HTTP protocol to
clients.
A stream can be connected to a feed, or to a file. In the first case,
the published stream is forwarded from the corresponding feed
generated by a running instance of @command{ffmpeg}, in the second
case the stream is read from a pre-recorded file.
Each stream is identified by a unique name, corresponding to the name
of the resource served by @command{ffserver}, and is configured by
a dedicated @code{Stream} section in the configuration file.
The stream access HTTP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
@end example
The stream access RTSP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
@end example
@var{stream_name} is the name of the corresponding stream defined in
the configuration file. @var{options} is a list of options specified
after the URL which affects how the stream is served by
@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
and RTSP ports configured with the options @var{Port} and
@var{RTSPPort} respectively.
In case the stream is associated to a feed, the encoding parameters
must be configured in the stream configuration. They are sent to
@command{ffmpeg} when setting up the encoding. This allows
@command{ffserver} to define the encoding parameters used by
the @command{ffmpeg} encoders.
The @command{ffmpeg} @option{override_ffserver} commandline option
allows one to override the encoding parameters set by the server.
Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following
graph:
@example
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
\ |_________|\ |__________|
\ \
\ \ __________
\ \ | |
\ \| stream 2 |
\ |__________|
\
\ _________ __________
\ | | | |
\| feed 2 |-----| stream 3 |
|_________| |__________|
_________ __________
| | | |
ffmpeg 2 -----| feed 3 |-----| stream 4 |
|_________| |__________|
_________ __________
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
@end example
@anchor{FFM}
@section FFM, FFM2 formats
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
@section Status stream
@command{ffserver} supports an HTTP interface which exposes the
current status of the server.
ffserver supports an HTTP interface which exposes the current status
of the server.
Simply point your browser to the address of the special status stream
specified in the configuration file.
@@ -180,8 +61,27 @@ ACL allow 192.168.0.0 192.168.255.255
then the server will post a page with the status information when
the special stream @file{status.html} is requested.
@section What can this do?
When properly configured and running, you can capture video and audio in real
time from a suitable capture card, and stream it out over the Internet to
either Windows Media Player or RealAudio player (with some restrictions).
It can also stream from files, though that is currently broken. Very often, a
web server can be used to serve up the files just as well.
It can stream prerecorded video from .ffm files, though it is somewhat tricky
to make it work correctly.
@section How do I make it work?
First, build the kit. It *really* helps to have installed LAME first. Then when
you run the ffserver ./configure, make sure that you have the
@code{--enable-libmp3lame} flag turned on.
LAME is important as it allows for streaming audio to Windows Media Player.
Don't ask why the other audio types do not work.
As a simple test, just run the following two command lines where INPUTFILE
is some file which you can decode with ffmpeg:
@@ -209,6 +109,35 @@ You should edit the ffserver.conf file to suit your needs (in terms of
frame rates etc). Then install ffserver and ffmpeg, write a script to start
them up, and off you go.
@section Troubleshooting
@subsection I don't hear any audio, but video is fine.
Maybe you didn't install LAME, or got your ./configure statement wrong. Check
the ffmpeg output to see if a line referring to MP3 is present. If not, then
your configuration was incorrect. If it is, then maybe your wiring is not
set up correctly. Maybe the sound card is not getting data from the right
input source. Maybe you have a really awful audio interface (like I do)
that only captures in stereo and also requires that one channel be flipped.
If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
starting ffmpeg.
@subsection The audio and video lose sync after a while.
Yes, they do.
@subsection After a long while, the video update rate goes way down in WMP.
Yes, it does. Who knows why?
@subsection WMP 6.4 behaves differently to WMP 7.
Yes, it does. Any thoughts on this would be gratefully received. These
differences extend to embedding WMP into a web page. [There are two
object IDs that you can use: The old one, which does not play well, and
the new one, which does (both tested on the same system). However,
I suspect that the new one is not available unless you have installed WMP 7].
@section What else can it do?
You can replay video from .ffm files that was recorded earlier.
@@ -248,6 +177,9 @@ specify a time. In addition, ffserver will skip frames until a key_frame
is found. This further reduces the startup delay by not transferring data
that will be discarded.
* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
the amount of bandwidth consumed by live streams.
@section Why does the ?buffer / Preroll stop working after a time?
It turns out that (on my machine at least) the number of frames successfully
@@ -281,6 +213,19 @@ You use this by adding the ?date= to the end of the URL for the stream.
For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end
@section What is FFM, FFM2
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
@chapter Options
@c man begin OPTIONS
@@ -290,540 +235,15 @@ For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@table @option
@item -f @var{configfile}
Read configuration file @file{configfile}. If not specified it will
read by default from @file{/etc/ffserver.conf}.
Use @file{configfile} instead of @file{/etc/ffserver.conf}.
@item -n
Enable no-launch mode. This option disables all the @code{Launch}
directives within the various @code{<Feed>} sections. Since
@command{ffserver} will not launch any @command{ffmpeg} instances, you
will have to launch them manually.
Enable no-launch mode. This option disables all the Launch directives
within the various <Stream> sections. Since ffserver will not launch
any ffmpeg instances, you will have to launch them manually.
@item -d
Enable debug mode. This option increases log verbosity, and directs
log messages to stdout. When specified, the @option{CustomLog} option
is ignored.
Enable debug mode. This option increases log verbosity, directs log
messages to stdout.
@end table
@chapter Configuration file syntax
@command{ffserver} reads a configuration file containing global
options and settings for each stream and feed.
The configuration file consists of global options and dedicated
sections, which must be introduced by "<@var{SECTION_NAME}
@var{ARGS}>" on a separate line and must be terminated by a line in
the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
Currently the following sections are recognized: @samp{Feed},
@samp{Stream}, @samp{Redirect}.
A line starting with @code{#} is ignored and treated as a comment.
Name of options and sections are case-insensitive.
@section ACL syntax
An ACL (Access Control List) specifies the address which are allowed
to access a given stream, or to write a given feed.
It accepts the folling forms
@itemize
@item
Allow/deny access to @var{address}.
@example
ACL ALLOW <address>
ACL DENY <address>
@end example
@item
Allow/deny access to ranges of addresses from @var{first_address} to
@var{last_address}.
@example
ACL ALLOW <first_address> <last_address>
ACL DENY <first_address> <last_address>
@end example
@end itemize
You can repeat the ACL allow/deny as often as you like. It is on a per
stream basis. The first match defines the action. If there are no matches,
then the default is the inverse of the last ACL statement.
Thus 'ACL allow localhost' only allows access from localhost.
'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
allow everybody else.
@section Global options
@table @option
@item Port @var{port_number}
@item RTSPPort @var{port_number}
Set TCP port number on which the HTTP/RTSP server is listening. You
must select a different port from your standard HTTP web server if it
is running on the same computer.
If not specified, no corresponding server will be created.
@item BindAddress @var{ip_address}
@item RTSPBindAddress @var{ip_address}
Set address on which the HTTP/RTSP server is bound. Only useful if you
have several network interfaces.
@item MaxHTTPConnections @var{n}
Set number of simultaneous HTTP connections that can be handled. It
has to be defined @emph{before} the @option{MaxClients} parameter,
since it defines the @option{MaxClients} maximum limit.
Default value is 2000.
@item MaxClients @var{n}
Set number of simultaneous requests that can be handled. Since
@command{ffserver} is very fast, it is more likely that you will want
to leave this high and use @option{MaxBandwidth}.
Default value is 5.
@item MaxBandwidth @var{kbps}
Set the maximum amount of kbit/sec that you are prepared to consume
when streaming to clients.
Default value is 1000.
@item CustomLog @var{filename}
Set access log file (uses standard Apache log file format). '-' is the
standard output.
If not specified @command{ffserver} will produce no log.
In case the commandline option @option{-d} is specified this option is
ignored, and the log is written to standard output.
@item NoDaemon
Set no-daemon mode. This option is currently ignored since now
@command{ffserver} will always work in no-daemon mode, and is
deprecated.
@end table
@section Feed section
A Feed section defines a feed provided to @command{ffserver}.
Each live feed contains one video and/or audio sequence coming from an
@command{ffmpeg} encoder or another @command{ffserver}. This sequence
may be encoded simultaneously with several codecs at several
resolutions.
A feed instance specification is introduced by a line in the form:
@example
<Feed FEED_FILENAME>
@end example
where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
The following options are recognized within a Feed section.
@table @option
@item File @var{filename}
@item ReadOnlyFile @var{filename}
Set the path where the feed file is stored on disk.
If not specified, the @file{/tmp/FEED.ffm} is assumed, where
@var{FEED} is the feed name.
If @option{ReadOnlyFile} is used the file is marked as read-only and
it will not be deleted or updated.
@item Truncate
Truncate the feed file, rather than appending to it. By default
@command{ffserver} will append data to the file, until the maximum
file size value is reached (see @option{FileMaxSize} option).
@item FileMaxSize @var{size}
Set maximum size of the feed file in bytes. 0 means unlimited. The
postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
recognized.
Default value is 5M.
@item Launch @var{args}
Launch an @command{ffmpeg} command when creating @command{ffserver}.
@var{args} must be a sequence of arguments to be provided to an
@command{ffmpeg} instance. The first provided argument is ignored, and
it is replaced by a path with the same dirname of the @command{ffserver}
instance, followed by the remaining argument and terminated with a
path corresponding to the feed.
When the launched process exits, @command{ffserver} will launch
another program instance.
In case you need a more complex @command{ffmpeg} configuration,
e.g. if you need to generate multiple FFM feeds with a single
@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
This option is ignored in case the commandline option @option{-n} is
specified.
@item ACL @var{spec}
Specify the list of IP address which are allowed or denied to write
the feed. Multiple ACL options can be specified.
@end table
@section Stream section
A Stream section defines a stream provided by @command{ffserver}, and
identified by a single name.
The stream is sent when answering a request containing the stream
name.
A stream section must be introduced by the line:
@example
<Stream STREAM_NAME>
@end example
where @var{STREAM_NAME} specifies the unique name of the stream.
The following options are recognized within a Stream section.
Encoding options are marked with the @emph{encoding} tag, and they are
used to set the encoding parameters, and are mapped to libavcodec
encoding options. Not all encoding options are supported, in
particular it is not possible to set encoder private options. In order
to override the encoding options specified by @command{ffserver}, you
can use the @command{ffmpeg} @option{override_ffserver} commandline
option.
Only one of the @option{Feed} and @option{File} options should be set.
@table @option
@item Feed @var{feed_name}
Set the input feed. @var{feed_name} must correspond to an existing
feed defined in a @code{Feed} section.
When this option is set, encoding options are used to setup the
encoding operated by the remote @command{ffmpeg} process.
@item File @var{filename}
Set the filename of the pre-recorded input file to stream.
When this option is set, encoding options are ignored and the input
file content is re-streamed as is.
@item Format @var{format_name}
Set the format of the output stream.
Must be the name of a format recognized by FFmpeg. If set to
@samp{status}, it is treated as a status stream.
@item InputFormat @var{format_name}
Set input format. If not specified, it is automatically guessed.
@item Preroll @var{n}
Set this to the number of seconds backwards in time to start. Note that
most players will buffer 5-10 seconds of video, and also you need to allow
for a keyframe to appear in the data stream.
Default value is 0.
@item StartSendOnKey
Do not send stream until it gets the first key frame. By default
@command{ffserver} will send data immediately.
@item MaxTime @var{n}
Set the number of seconds to run. This value set the maximum duration
of the stream a client will be able to receive.
A value of 0 means that no limit is set on the stream duration.
@item ACL @var{spec}
Set ACL for the stream.
@item DynamicACL @var{spec}
@item RTSPOption @var{option}
@item MulticastAddress @var{address}
@item MulticastPort @var{port}
@item MulticastTTL @var{integer}
@item NoLoop
@item FaviconURL @var{url}
Set favicon (favourite icon) for the server status page. It is ignored
for regular streams.
@item Author @var{value}
@item Comment @var{value}
@item Copyright @var{value}
@item Title @var{value}
Set metadata corresponding to the option. All these options are
deprecated in favor of @option{Metadata}.
@item Metadata @var{key} @var{value}
Set metadata value on the output stream.
@item NoAudio
@item NoVideo
Suppress audio/video.
@item AudioCodec @var{codec_name} (@emph{encoding,audio})
Set audio codec.
@item AudioBitRate @var{rate} (@emph{encoding,audio})
Set bitrate for the audio stream in kbits per second.
@item AudioChannels @var{n} (@emph{encoding,audio})
Set number of audio channels.
@item AudioSampleRate @var{n} (@emph{encoding,audio})
Set sampling frequency for audio. When using low bitrates, you should
lower this frequency to 22050 or 11025. The supported frequencies
depend on the selected audio codec.
@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
Set generic option for audio stream.
@item AVPresetAudio @var{preset} (@emph{encoding,audio})
Set preset for audio stream.
@item VideoCodec @var{codec_name} (@emph{encoding,video})
Set video codec.
@item VideoBitRate @var{n} (@emph{encoding,video})
Set bitrate for the video stream in kbits per second.
@item VideoBitRateRange @var{range} (@emph{encoding,video})
Set video bitrate range.
A range must be specified in the form @var{minrate}-@var{maxrate}, and
specifies the @option{minrate} and @option{maxrate} encoding options
expressed in kbits per second.
@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
Set video bitrate tolerance in kbits per second.
@item PixelFormat @var{pixel_format} (@emph{encoding,video})
Set video pixel format.
@item Debug @var{integer} (@emph{encoding,video})
Set video @option{debug} encoding option.
@item Strict @var{integer} (@emph{encoding,video})
Set video @option{strict} encoding option.
@item VideoBufferSize @var{n} (@emph{encoding,video})
Set ratecontrol buffer size, expressed in KB.
@item VideoFrameRate @var{n} (@emph{encoding,video})
Set number of video frames per second.
@item VideoSize (@emph{encoding,video})
Set size of the video frame, must be an abbreviation or in the form
@var{W}x@var{H}. See @ref{video size syntax,,the Video size section
in the ffmpeg-utils(1) manual,ffmpeg-utils}.
Default value is @code{160x128}.
@item VideoIntraOnly (@emph{encoding,video})
Transmit only intra frames (useful for low bitrates, but kills frame rate).
@item VideoGopSize @var{n} (@emph{encoding,video})
If non-intra only, an intra frame is transmitted every VideoGopSize
frames. Video synchronization can only begin at an intra frame.
@item VideoTag @var{tag} (@emph{encoding,video})
Set video tag.
@item VideoHighQuality (@emph{encoding,video})
@item Video4MotionVector (@emph{encoding,video})
@item BitExact (@emph{encoding,video})
Set bitexact encoding flag.
@item IdctSimple (@emph{encoding,video})
Set simple IDCT algorithm.
@item Qscale @var{n} (@emph{encoding,video})
Enable constant quality encoding, and set video qscale (quantization
scale) value, expressed in @var{n} QP units.
@item VideoQMin @var{n} (@emph{encoding,video})
@item VideoQMax @var{n} (@emph{encoding,video})
Set video qmin/qmax.
@item VideoQDiff @var{integer} (@emph{encoding,video})
Set video @option{qdiff} encoding option.
@item LumiMask @var{float} (@emph{encoding,video})
@item DarkMask @var{float} (@emph{encoding,video})
Set @option{lumi_mask}/@option{dark_mask} encoding options.
@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
Set generic option for video stream.
@item AVPresetVideo @var{preset} (@emph{encoding,video})
Set preset for video stream.
@var{preset} must be the path of a preset file.
@end table
@subsection Server status stream
A server status stream is a special stream which is used to show
statistics about the @command{ffserver} operations.
It must be specified setting the option @option{Format} to
@samp{status}.
@section Redirect section
A redirect section specifies where to redirect the requested URL to
another page.
A redirect section must be introduced by the line:
@example
<Redirect NAME>
@end example
where @var{NAME} is the name of the page which should be redirected.
It only accepts the option @option{URL}, which specify the redirection
URL.
@chapter Stream examples
@itemize
@item
Multipart JPEG
@example
<Stream test.mjpg>
Feed feed1.ffm
Format mpjpeg
VideoFrameRate 2
VideoIntraOnly
NoAudio
Strict -1
</Stream>
@end example
@item
Single JPEG
@example
<Stream test.jpg>
Feed feed1.ffm
Format jpeg
VideoFrameRate 2
VideoIntraOnly
VideoSize 352x240
NoAudio
Strict -1
</Stream>
@end example
@item
Flash
@example
<Stream test.swf>
Feed feed1.ffm
Format swf
VideoFrameRate 2
VideoIntraOnly
NoAudio
</Stream>
@end example
@item
ASF compatible
@example
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
@end example
@item
MP3 audio
@example
<Stream test.mp3>
Feed feed1.ffm
Format mp2
AudioCodec mp3
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Ogg Vorbis audio
@example
<Stream test.ogg>
Feed feed1.ffm
Metadata title "Stream title"
AudioBitRate 64
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Real with audio only at 32 kbits
@example
<Stream test.ra>
Feed feed1.ffm
Format rm
AudioBitRate 32
NoVideo
</Stream>
@end example
@item
Real with audio and video at 64 kbits
@example
<Stream test.rm>
Feed feed1.ffm
Format rm
AudioBitRate 32
VideoBitRate 128
VideoFrameRate 25
VideoGopSize 25
</Stream>
@end example
@item
For stream coming from a file: you only need to set the input filename
and optionally a new format.
@example
<Stream file.rm>
File "/usr/local/httpd/htdocs/tlive.rm"
NoAudio
</Stream>
@end example
@example
<Stream file.asf>
File "/usr/local/httpd/htdocs/test.asf"
NoAudio
Metadata author "Me"
Metadata copyright "Super MegaCorp"
Metadata title "Test stream from disk"
Metadata comment "Test comment"
</Stream>
@end example
@end itemize
@c man end
@include config.texi

View File

@@ -44,8 +44,8 @@ streams of this type.
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 #@var{stream_id}
Matches the stream by a format-specific ID.
@end table
@section Generic options
@@ -189,20 +189,11 @@ following option is recognized:
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 level
@end table
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.
@@ -259,10 +250,6 @@ Possible flags for this option are:
@end table
@end table
@item -opencl_bench
Benchmark all available OpenCL devices and show the results. This option
is only available when FFmpeg has been compiled with @code{--enable-opencl}.
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with @code{--enable-opencl}.

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View File

@@ -125,26 +125,18 @@ Consider things that a sane encoder should not do as an error.
Use wallclock as timestamps.
@item avoid_negative_ts @var{integer} (@emph{output})
Possible values:
@table @samp
@item make_non_negative
Shift timestamps to make them non-negative.
Also note that this affects only leading negative timestamps, and not
non-monotonic negative timestamps.
@item make_zero
Shift timestamps so that the first timestamp is 0.
@item auto (default)
Enables shifting when required by the target format.
@item disabled
Disables shifting of timestamp.
@end table
Shift timestamps to make them non-negative. A value of 1 enables shifting,
a value of 0 disables it, the default value of -1 enables shifting
when required by the target format.
When shifting is enabled, all output timestamps are shifted by the
same amount. Audio, video, and subtitles desynching and relative
timestamp differences are preserved compared to how they would have
been without shifting.
Also note that this affects only leading negative timestamps, and not
non-monotonic negative timestamps.
@item skip_initial_bytes @var{integer} (@emph{input})
Set number of bytes to skip before reading header and frames if set to 1.
Default is 0.
@@ -156,18 +148,6 @@ Correct single timestamp overflows if set to 1. Default is 1.
Flush the underlying I/O stream after each packet. Default 1 enables it, and
has the effect of reducing the latency; 0 disables it and may slightly
increase performance in some cases.
@item output_ts_offset @var{offset} (@emph{output})
Set the output time offset.
@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 by the muxer to the output timestamps.
Specifying a positive offset means that the corresponding streams are
delayed bt the time duration specified in @var{offset}. Default value
is @code{0} (meaning that no offset is applied).
@end table
@c man end FORMAT OPTIONS
@@ -203,10 +183,6 @@ The exact semantics of stream specifiers is defined by the
@code{avformat_match_stream_specifier()} function declared in the
@file{libavformat/avformat.h} header.
@ifclear config-writeonly
@include demuxers.texi
@end ifclear
@ifclear config-readonly
@include muxers.texi
@end ifclear
@include metadata.texi

View File

@@ -122,20 +122,6 @@ x264 is under the GNU Public License Version 2 or later
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section x265
FFmpeg can make use of the x265 library for HEVC encoding.
Go to @url{http://x265.org/developers.html} and follow the instructions
for installing the library. Then pass @code{--enable-libx265} to configure
to enable it.
@float note
x265 is under the GNU Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section libilbc
iLBC is a narrowband speech codec that has been made freely available
@@ -162,27 +148,6 @@ libzvbi is licensed under the GNU General Public License Version 2 or later
you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@section AviSynth
FFmpeg can read AviSynth scripts as input. To enable support, pass
@code{--enable-avisynth} to configure. The correct headers are
included in compat/avisynth/, which allows the user to enable support
without needing to search for these headers themselves.
For Windows, supported AviSynth variants are
@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
For Linux and OS X, the supported AviSynth variant is
@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
@float NOTE
AviSynth and AvxSynth are loaded dynamically. Distributors can build FFmpeg
with @code{--enable-avisynth}, and the binaries will work regardless of the
end user having AviSynth or AvxSynth installed - they'll only need to be
installed to use AviSynth scripts (obviously).
@end float
@chapter Supported File Formats, Codecs or Features
@@ -205,7 +170,7 @@ library:
@item American Laser Games MM @tab @tab X
@tab Multimedia format used in games like Mad Dog McCree.
@item 3GPP AMR @tab X @tab X
@item Amazing Studio Packed Animation File @tab @tab X
@item Amazing Studio Packed Animation File @tab @tab X
@tab Multimedia format used in game Heart Of Darkness.
@item Apple HTTP Live Streaming @tab @tab X
@item Artworx Data Format @tab @tab X
@@ -245,7 +210,6 @@ library:
@tab Multimedia format used by Delphine Software games.
@item CD+G @tab @tab X
@tab Video format used by CD+G karaoke disks
@item Phantom Cine @tab @tab X
@item Commodore CDXL @tab @tab X
@tab Amiga CD video format
@item Core Audio Format @tab X @tab X
@@ -259,7 +223,6 @@ library:
@item Deluxe Paint Animation @tab @tab X
@item DFA @tab @tab X
@tab This format is used in Chronomaster game
@item DSD Stream File (DSF) @tab @tab X
@item DV video @tab X @tab X
@item DXA @tab @tab X
@tab This format is used in the non-Windows version of the Feeble Files
@@ -286,8 +249,6 @@ library:
@item GXF @tab X @tab X
@tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
playout servers.
@item HNM @tab @tab X
@tab Only version 4 supported, used in some games from Cryo Interactive
@item iCEDraw File @tab @tab X
@item ICO @tab X @tab X
@tab Microsoft Windows ICO
@@ -310,11 +271,9 @@ library:
@tab Used by Linux Media Labs MPEG-4 PCI boards
@item LOAS @tab @tab X
@tab contains LATM multiplexed AAC audio
@item LRC @tab X @tab X
@item LVF @tab @tab X
@item LXF @tab @tab X
@tab VR native stream format, used by Leitch/Harris' video servers.
@item Magic Lantern Video (MLV) @tab @tab X
@item Matroska @tab X @tab X
@item Matroska audio @tab X @tab
@item FFmpeg metadata @tab X @tab X
@@ -340,8 +299,6 @@ library:
@tab also known as DVB Transport Stream
@item MPEG-4 @tab X @tab X
@tab MPEG-4 is a variant of QuickTime.
@item Mirillis FIC video @tab @tab X
@tab No cursor rendering.
@item MIME multipart JPEG @tab X @tab
@item MSN TCP webcam @tab @tab X
@tab Used by MSN Messenger webcam streams.
@@ -381,7 +338,6 @@ library:
@item raw H.261 @tab X @tab X
@item raw H.263 @tab X @tab X
@item raw H.264 @tab X @tab X
@item raw HEVC @tab X @tab X
@item raw Ingenient MJPEG @tab @tab X
@item raw MJPEG @tab X @tab X
@item raw MLP @tab @tab X
@@ -492,13 +448,11 @@ following image formats are supported:
@item Name @tab Encoding @tab Decoding @tab Comments
@item .Y.U.V @tab X @tab X
@tab one raw file per component
@item Alias PIX @tab X @tab X
@tab Alias/Wavefront PIX image format
@item animated GIF @tab X @tab X
@item BMP @tab X @tab X
@tab Microsoft BMP image
@item BRender PIX @tab @tab X
@tab Argonaut BRender 3D engine image format.
@item PIX @tab @tab X
@tab PIX is an image format used in the Argonaut BRender engine.
@item DPX @tab X @tab X
@tab Digital Picture Exchange
@item EXR @tab @tab X
@@ -534,8 +488,8 @@ following image formats are supported:
@tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa @tab X @tab X
@tab Targa (.TGA) image format
@item WebP @tab E @tab X
@tab WebP image format, encoding supported through external library libwebp
@item WebP @tab @tab X
@tab WebP image format
@item XBM @tab X @tab X
@tab X BitMap image format
@item XFace @tab X @tab X
@@ -653,9 +607,6 @@ following image formats are supported:
@item H.263+ / H.263-1998 / H.263 version 2 @tab X @tab X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 @tab E @tab X
@tab encoding supported through external library libx264
@item HEVC @tab X @tab X
@tab encoding supported through the external library libx265
@item HNM version 4 @tab @tab X
@item HuffYUV @tab X @tab X
@item HuffYUV FFmpeg variant @tab X @tab X
@item IBM Ultimotion @tab @tab X
@@ -686,8 +637,8 @@ following image formats are supported:
@item LCL (LossLess Codec Library) MSZH @tab @tab X
@item LCL (LossLess Codec Library) ZLIB @tab E @tab E
@item LOCO @tab @tab X
@item LucasArts SANM/Smush @tab @tab X
@tab Used in LucasArts games / SMUSH animations.
@item LucasArts Smush @tab @tab X
@tab Used in LucasArts games.
@item lossless MJPEG @tab X @tab X
@item Microsoft ATC Screen @tab @tab X
@tab Also known as Microsoft Screen 3.
@@ -707,6 +658,7 @@ following image formats are supported:
@item Mobotix MxPEG video @tab @tab X
@item Motion Pixels video @tab @tab X
@item MPEG-1 video @tab X @tab X
@item MPEG-1/2 video XvMC (X-Video Motion Compensation) @tab @tab X
@item MPEG-2 video @tab X @tab X
@item MPEG-4 part 2 @tab X @tab X
@tab libxvidcore can be used alternatively for encoding.
@@ -722,8 +674,6 @@ following image formats are supported:
@tab fourcc: VP50
@item On2 VP6 @tab @tab X
@tab fourcc: VP60,VP61,VP62
@item On2 VP7 @tab @tab X
@tab fourcc: VP70,VP71
@item VP8 @tab E @tab X
@tab fourcc: VP80, encoding supported through external library libvpx
@item VP9 @tab E @tab X
@@ -753,11 +703,11 @@ following image formats are supported:
@tab Texture dictionaries used by the Renderware Engine.
@item RL2 video @tab @tab X
@tab used in some games by Entertainment Software Partners
@item SGI RLE 8-bit @tab @tab X
@item Sierra VMD video @tab @tab X
@tab Used in Sierra VMD files.
@item Silicon Graphics Motion Video Compressor 1 (MVC1) @tab @tab X
@item Silicon Graphics Motion Video Compressor 2 (MVC2) @tab @tab X
@item Silicon Graphics RLE 8-bit video @tab @tab X
@item Smacker video @tab @tab X
@tab Video encoding used in Smacker.
@item SMPTE VC-1 @tab @tab X
@@ -823,7 +773,7 @@ following image formats are supported:
@tab encoding supported through external library libaacplus
@item AAC @tab E @tab X
@tab encoding supported through external library libfaac and libvo-aacenc
@item AC-3 @tab IX @tab IX
@item AC-3 @tab IX @tab X
@item ADPCM 4X Movie @tab @tab X
@item ADPCM CDROM XA @tab @tab X
@item ADPCM Creative Technology @tab @tab X
@@ -867,8 +817,6 @@ following image formats are supported:
@item ADPCM Sound Blaster Pro 2-bit @tab @tab X
@item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X
@item ADPCM Sound Blaster Pro 4-bit @tab @tab X
@item ADPCM VIMA
@tab Used in LucasArts SMUSH animations.
@item ADPCM Westwood Studios IMA @tab @tab X
@tab Used in Westwood Studios games like Command and Conquer.
@item ADPCM Yamaha @tab X @tab X
@@ -881,7 +829,6 @@ following image formats are supported:
@tab QuickTime fourcc 'alac'
@item ATRAC1 @tab @tab X
@item ATRAC3 @tab @tab X
@item ATRAC3+ @tab @tab X
@item Bink Audio @tab @tab X
@tab Used in Bink and Smacker files in many games.
@item CELT @tab @tab E
@@ -901,10 +848,6 @@ following image formats are supported:
@item DPCM Sol @tab @tab X
@item DPCM Xan @tab @tab X
@tab Used in Origin's Wing Commander IV AVI files.
@item DSD (Direct Stream Digitial), least significant bit first @tab @tab X
@item DSD (Direct Stream Digitial), most significant bit first @tab @tab X
@item DSD (Direct Stream Digitial), least significant bit first, planar @tab @tab X
@item DSD (Direct Stream Digitial), most significant bit first, planar @tab @tab X
@item DSP Group TrueSpeech @tab @tab X
@item DV audio @tab @tab X
@item Enhanced AC-3 @tab X @tab X
@@ -927,14 +870,13 @@ following image formats are supported:
@item Monkey's Audio @tab @tab X
@item MP1 (MPEG audio layer 1) @tab @tab IX
@item MP2 (MPEG audio layer 2) @tab IX @tab IX
@tab encoding supported also through external library TwoLAME
@tab libtwolame can be used alternatively for encoding.
@item MP3 (MPEG audio layer 3) @tab E @tab IX
@tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X
@item Musepack SV7 @tab @tab X
@item Musepack SV8 @tab @tab X
@item Nellymoser Asao @tab X @tab X
@item On2 AVC (Audio for Video Codec) @tab @tab X
@item Opus @tab E @tab E
@tab supported through external library libopus
@item PCM A-law @tab X @tab X
@@ -997,6 +939,7 @@ following image formats are supported:
@item Vorbis @tab E @tab X
@tab A native but very primitive encoder exists.
@item Voxware MetaSound @tab @tab X
@tab imperfect and incomplete support
@item WavPack @tab X @tab X
@item Westwood Audio (SND1) @tab @tab X
@item Windows Media Audio 1 @tab X @tab X
@@ -1037,7 +980,7 @@ performance on systems without hardware floating point support).
@item TED Talks captions @tab @tab X @tab @tab X
@item VobSub (IDX+SUB) @tab @tab X @tab @tab X
@item VPlayer @tab @tab X @tab @tab X
@item WebVTT @tab X @tab X @tab X @tab X
@item WebVTT @tab X @tab X @tab @tab X
@item XSUB @tab @tab @tab X @tab X
@end multitable
@@ -1050,7 +993,6 @@ performance on systems without hardware floating point support).
@multitable @columnfractions .4 .1
@item Name @tab Support
@item file @tab X
@item FTP @tab X
@item Gopher @tab X
@item HLS @tab X
@item HTTP @tab X
@@ -1065,9 +1007,7 @@ performance on systems without hardware floating point support).
@item RTMPTE @tab X
@item RTMPTS @tab X
@item RTP @tab X
@item SAMBA @tab E
@item SCTP @tab X
@item SFTP @tab E
@item TCP @tab X
@item TLS @tab X
@item UDP @tab X
@@ -1087,19 +1027,17 @@ performance on systems without hardware floating point support).
@item caca @tab @tab X
@item DV1394 @tab X @tab
@item Lavfi virtual device @tab X @tab
@item Linux framebuffer @tab X @tab X
@item Linux framebuffer @tab X @tab
@item JACK @tab X @tab
@item LIBCDIO @tab X
@item LIBDC1394 @tab X @tab
@item OpenAL @tab X
@item OpenGL @tab @tab X
@item OSS @tab X @tab X
@item PulseAudio @tab X @tab X
@item Pulseaudio @tab X @tab
@item SDL @tab @tab X
@item Video4Linux2 @tab X @tab X
@item VfW capture @tab X @tab
@item X11 grabbing @tab X @tab
@item Win32 grabbing @tab X @tab
@end multitable
@code{X} means that input/output is supported.

View File

@@ -299,7 +299,7 @@ the current branch history.
git commit --amend
@end example
allows one to amend the last commit details quickly.
allows to amend the last commit details quickly.
@example
git rebase -i origin/master

273
doc/git-howto.txt Normal file
View File

@@ -0,0 +1,273 @@
About Git write access:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Before everything else, you should know how to use GIT properly.
Luckily Git comes with excellent documentation.
git --help
man git
shows you the available subcommands,
git <command> --help
man git-<command>
shows information about the subcommand <command>.
The most comprehensive manual is the website Git Reference
http://gitref.org/
For more information about the Git project, visit
http://git-scm.com/
Consult these resources whenever you have problems, they are quite exhaustive.
You do not need a special username or password.
All you need is to provide a ssh public key to the Git server admin.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines. Read it at least once, if you are granted commit privileges to the
FFmpeg project you are expected to be familiar with these rules.
I. BASICS:
==========
0. Get GIT:
Most distributions have a git package, if not
You can get git from http://git-scm.com/
1. Cloning the source tree:
git clone git://source.ffmpeg.org/ffmpeg <target>
This will put the FFmpeg sources into the directory <target>.
git clone git@source.ffmpeg.org:ffmpeg <target>
This will put the FFmpeg sources into the directory <target> and let
you push back your changes to the remote repository.
2. Updating the source tree to the latest revision:
git pull (--ff-only)
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
Caveat: Since merge commits are forbidden at least for the initial
months of git --ff-only or --rebase (see below) are recommended.
--ff-only will fail and not create merge commits if your branch
has diverged (has a different history) from the tracked branch.
2.a Rebasing your local branches:
git pull --rebase
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
3. Adding/removing files/directories:
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
4. Showing modifications:
git diff <filename(s)>
will show all local modifications in your working directory as unified diff.
5. Inspecting the changelog:
git log <filename(s)>
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org
6. Checking source tree status:
git status
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
7. Committing:
git diff --check
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
git diff -w -b <filename(s)>
Also check the output of
git status
to make sure you don't have untracked files or deletions.
git add [-i|-p|-A] <filenames/dirnames>
Make sure you have told git your name and email address, e.g. by running
git config --global user.name "My Name"
git config --global user.email my@email.invalid
(--global to set the global configuration for all your git checkouts).
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
git commit
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
git config core.editor
or set by one of the following environment variables:
GIT_EDITOR, VISUAL or EDITOR.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
8. Renaming/moving/copying files or contents of files:
Git automatically tracks such changes, making those normal commits.
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
Do not move, rename or copy files of which you are not the maintainer without
discussing it on the mailing list first!
9. Reverting broken commits
git revert <commit>
git revert will generate a revert commit. This will not make the faulty
commit disappear from the history.
git reset <commit>
git reset will uncommit the changes till <commit> rewriting the current
branch history.
git commit --amend
allows to amend the last commit details quickly.
git rebase -i origin/master
will replay local commits over the main repository allowing to edit,
merge or remove some of them in the process.
Note that the reset, commit --amend and rebase rewrite history, so you
should use them ONLY on your local or topic branches.
The main repository will reject those changes.
10. Preparing a patchset.
git format-patch <commit> [-o directory]
will generate a set of patches for each commit between <commit> and
current HEAD. E.g.
git format-patch origin/master
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
git format-patch -n
which will generate patches from last n commits.
By default the patches are created in the current directory.
11. Sending patches for review
git send-email <commit list|directory>
will send the patches created by git format-patch or directly generates
them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. git-email
package on Debian-based distros).
12. Pushing changes to remote trees
git push
Will push the changes to the default remote (origin).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to 2 and 2.a to sync the local tree.
git remote add <name> <url>
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
git push <remote> <refspec>
Will push the changes to the remote repository. Omitting refspec makes git
push update all the remote branches matching the local ones.
13. Finding a specific svn revision
Since version 1.7.1 git supports ':/foo' syntax for specifying commits
based on a regular expression. see man gitrevisions
git show :/'as revision 23456'
will show the svn changeset r23456. With older git versions searching in
the git log output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
git checkout -b svn_23456 :/'as revision 23456'
or for git < 1.7.1 with
git checkout -b svn_23456 $SHA1
where $SHA1 is the commit SHA1 from the 'git log' output.
Contact the project admins <root at ffmpeg dot org> if you have technical
problems with the GIT server.

View File

@@ -13,8 +13,8 @@ You can disable all the input devices using the configure option
option "--enable-indev=@var{INDEV}", or you can disable a particular
input device using the option "--disable-indev=@var{INDEV}".
The option "-devices" of the ff* tools will display the list of
supported input devices.
The option "-formats" of the ff* tools will display the list of
supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
@@ -51,41 +51,6 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
For more information see:
@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
@section avfoundation
AVFoundation input device.
AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
The older QTKit framework has been marked deprecated since OSX version 10.7.
The filename passed as input is parsed to contain either a device name or index.
The device index can also be given by using -video_device_index.
A given device index will override any given device name.
If the desired device consists of numbers only, use -video_device_index to identify it.
The default device will be chosen if an empty string or the device name "default" is given.
The available devices can be enumerated by using -list_devices.
The pixel format can be set using -pixel_format.
Available formats:
monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
yuv420p, nv12, yuyv422, gray
@example
ffmpeg -f avfoundation -i "0" out.mpg
@end example
@example
ffmpeg -f avfoundation -video_device_index 0 -i "" out.mpg
@end example
@example
ffmpeg -f avfoundation -pixel_format bgr0 -i "default" out.mpg
@end example
@example
ffmpeg -f avfoundation -list_devices true -i ""
@end example
@section bktr
BSD video input device.
@@ -227,81 +192,6 @@ ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section gdigrab
Win32 GDI-based screen capture device.
This device allows you to capture a region of the display on Windows.
There are two options for the input filename:
@example
desktop
@end example
or
@example
title=@var{window_title}
@end example
The first option will capture the entire desktop, or a fixed region of the
desktop. The second option will instead capture the contents of a single
window, regardless of its position on the screen.
For example, to grab the entire desktop using @command{ffmpeg}:
@example
ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg
@end example
Grab a 640x480 region at position @code{10,20}:
@example
ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg
@end example
Grab the contents of the window named "Calculator"
@example
ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg
@end example
@subsection Options
@table @option
@item draw_mouse
Specify whether to draw the mouse pointer. Use the value @code{0} to
not draw the pointer. Default value is @code{1}.
@item framerate
Set the grabbing frame rate. Default value is @code{ntsc},
corresponding to a frame rate of @code{30000/1001}.
@item show_region
Show grabbed region on screen.
If @var{show_region} is specified with @code{1}, then the grabbing
region will be indicated on screen. With this option, it is easy to
know what is being grabbed if only a portion of the screen is grabbed.
Note that @var{show_region} is incompatible with grabbing the contents
of a single window.
For example:
@example
ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg
@end example
@item video_size
Set the video frame size. The default is to capture the full screen if @file{desktop} is selected, or the full window size if @file{title=@var{window_title}} is selected.
@item offset_x
When capturing a region with @var{video_size}, set the distance from the left edge of the screen or desktop.
Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned to the left of your primary monitor, you will need to use a negative @var{offset_x} value to move the region to that monitor.
@item offset_y
When capturing a region with @var{video_size}, set the distance from the top edge of the screen or desktop.
Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned above your primary monitor, you will need to use a negative @var{offset_y} value to move the region to that monitor.
@end table
@section iec61883
FireWire DV/HDV input device using libiec61883.
@@ -483,21 +373,6 @@ ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
@end itemize
@section libcdio
Audio-CD input device based on cdio.
To enable this input device during configuration you need libcdio
installed on your system.
This device allows playing and grabbing from an Audio-CD.
For example to copy with @command{ffmpeg} the entire Audio-CD in /dev/sr0,
you may run the command:
@example
ffmpeg -f libcdio -i /dev/sr0 cd.wav
@end example
@section libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.
@@ -534,7 +409,7 @@ OpenAL is part of Core Audio, the official Mac OS X Audio interface.
See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
@end table
This device allows one to capture from an audio input device handled
This device allows to capture from an audio input device handled
through OpenAL.
You need to specify the name of the device to capture in the provided
@@ -656,33 +531,6 @@ Record a stream from default device:
ffmpeg -f pulse -i default /tmp/pulse.wav
@end example
@section qtkit
QTKit input device.
The filename passed as input is parsed to contain either a device name or index.
The device index can also be given by using -video_device_index.
A given device index will override any given device name.
If the desired device consists of numbers only, use -video_device_index to identify it.
The default device will be chosen if an empty string or the device name "default" is given.
The available devices can be enumerated by using -list_devices.
@example
ffmpeg -f qtkit -i "0" out.mpg
@end example
@example
ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg
@end example
@example
ffmpeg -f qtkit -i "default" out.mpg
@end example
@example
ffmpeg -f qtkit -list_devices true -i ""
@end example
@section sndio
sndio input device.
@@ -769,7 +617,7 @@ Select the pixel format (only valid for raw video input).
@item input_format
Set the preferred pixel format (for raw video) or a codec name.
This option allows one to select the input format, when several are
This option allows to select the input format, when several are
available.
@item framerate
@@ -830,7 +678,7 @@ other filename will be interpreted as device number 0.
X11 video input device.
This device allows one to capture a region of an X11 display.
This device allows to capture a region of an X11 display.
The filename passed as input has the syntax:
@example

View File

@@ -16,25 +16,7 @@ The libavutil library is a utility library to aid portable
multimedia programming. It contains safe portable string functions,
random number generators, data structures, additional mathematics
functions, cryptography and multimedia related functionality (like
enumerations for pixel and sample formats). It is not a library for
code needed by both libavcodec and libavformat.
The goals for this library is to be:
@table @strong
@item Modular
It should have few interdependencies and the possibility of disabling individual
parts during @command{./configure}.
@item Small
Both sources and objects should be small.
@item Efficient
It should have low CPU and memory usage.
@item Useful
It should avoid useless features that almost no one needs.
@end table
enumerations for pixel and sample formats).
@c man end DESCRIPTION

View File

@@ -23,8 +23,6 @@ A description of some of the currently available muxers follows.
Audio Interchange File Format muxer.
@subsection Options
It accepts the following options:
@table @option
@@ -51,10 +49,6 @@ The output of the muxer consists of a single line of the form:
CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
8 digits containing the CRC for all the decoded input frames.
See also the @ref{framecrc} muxer.
@subsection Examples
For example to compute the CRC of the input, and store it in the file
@file{out.crc}:
@example
@@ -74,6 +68,8 @@ and the input video converted to MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
@end example
See also the @ref{framecrc} muxer.
@anchor{framecrc}
@section framecrc
@@ -93,8 +89,6 @@ packet of the form:
@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
CRC of the packet.
@subsection Examples
For example to compute the CRC of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
in the file @file{out.crc}:
@@ -138,8 +132,6 @@ packet of the form:
@var{MD5} is a hexadecimal number representing the computed MD5 hash
for the packet.
@subsection Examples
For example to compute the MD5 of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
in the file @file{out.md5}:
@@ -154,93 +146,30 @@ ffmpeg -i INPUT -f framemd5 -
See also the @ref{md5} muxer.
@anchor{gif}
@section gif
Animated GIF muxer.
It accepts the following options:
@table @option
@item loop
Set the number of times to loop the output. Use @code{-1} for no loop, @code{0}
for looping indefinitely (default).
@item final_delay
Force the delay (expressed in centiseconds) after the last frame. Each frame
ends with a delay until the next frame. The default is @code{-1}, which is a
special value to tell the muxer to re-use the previous delay. In case of a
loop, you might want to customize this value to mark a pause for instance.
@end table
For example, to encode a gif looping 10 times, with a 5 seconds delay between
the loops:
@example
ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif
@end example
Note 1: if you wish to extract the frames in separate GIF files, you need to
force the @ref{image2} muxer:
@example
ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
@end example
Note 2: the GIF format has a very small time base: the delay between two frames
can not be smaller than one centi second.
@anchor{hls}
@section hls
Apple HTTP Live Streaming muxer that segments MPEG-TS according to
the HTTP Live Streaming (HLS) specification.
the HTTP Live Streaming specification.
It creates a playlist file and numbered segment files. The output
filename specifies the playlist filename; the segment filenames
receive the same basename as the playlist, a sequential number and
a .ts extension.
For example, to convert an input file with @command{ffmpeg}:
@example
ffmpeg -i in.nut out.m3u8
@end example
See also the @ref{segment} muxer, which provides a more generic and
flexible implementation of a segmenter, and can be used to perform HLS
segmentation.
@subsection Options
This muxer supports the following options:
@table @option
@item hls_time @var{seconds}
Set the segment length in seconds. Default value is 2.
@item hls_list_size @var{size}
Set the maximum number of playlist entries. If set to 0 the list file
will contain all the segments. Default value is 5.
@item hls_wrap @var{wrap}
Set the number after which the segment filename number (the number
specified in each segment file) wraps. If set to 0 the number will be
never wrapped. Default value is 0.
This option is useful to avoid to fill the disk with many segment
files, and limits the maximum number of segment files written to disk
to @var{wrap}.
@item start_number @var{number}
Start the playlist sequence number from @var{number}. Default value is
0.
@item hls_base_url @var{baseurl}
Append @var{baseurl} to every entry in the playlist.
Useful to generate playlists with absolute paths.
Note that the playlist sequence number must be unique for each segment
and it is not to be confused with the segment filename sequence number
which can be cyclic, for example if the @option{wrap} option is
specified.
@item -hls_time @var{seconds}
Set the segment length in seconds.
@item -hls_list_size @var{size}
Set the maximum number of playlist entries.
@item -hls_wrap @var{wrap}
Set the number after which index wraps.
@item -start_number @var{number}
Start the sequence from @var{number}.
@end table
@anchor{ico}
@@ -306,8 +235,6 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
etc.
@subsection Examples
The following example shows how to use @command{ffmpeg} for creating a
sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
taking one image every second from the input video:
@@ -330,32 +257,16 @@ Note also that the pattern must not necessarily contain "%d" or
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example
The @option{strftime} option allows you to expand the filename with
date and time information. Check the documentation of
the @code{strftime()} function for the syntax.
For example to generate image files from the @code{strftime()}
"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command
can be used:
@example
ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
@end example
@subsection Options
@table @option
@item start_number
Start the sequence from the specified number. Default value is 1. Must
be a non-negative number.
@item start_number @var{number}
Start the sequence from @var{number}. Default value is 1. Must be a
non-negative number.
@item update
If set to 1, the filename will always be interpreted as just a
filename, not a pattern, and the corresponding file will be continuously
overwritten with new images. Default value is 0.
@item -update @var{number}
If @var{number} is nonzero, the filename will always be interpreted as just a
filename, not a pattern, and this file will be continuously overwritten with new
images.
@item strftime
If set to 1, expand the filename with date and time information from
@code{strftime()}. Default value is 0.
@end table
The image muxer supports the .Y.U.V image file format. This format is
@@ -370,27 +281,25 @@ Matroska container muxer.
This muxer implements the matroska and webm container specs.
@subsection Metadata
The recognized metadata settings in this muxer are:
@table @option
@item title
Set title name provided to a single track.
@item language
Specify the language of the track in the Matroska languages form.
@item title=@var{title name}
Name provided to a single track
@end table
The language can be either the 3 letters bibliographic ISO-639-2 (ISO
639-2/B) form (like "fre" for French), or a language code mixed with a
country code for specialities in languages (like "fre-ca" for Canadian
French).
@table @option
@item stereo_mode
Set stereo 3D video layout of two views in a single video track.
@item language=@var{language name}
Specifies the language of the track in the Matroska languages form
@end table
The following values are recognized:
@table @samp
@table @option
@item stereo_mode=@var{mode}
Stereo 3D video layout of two views in a single video track
@table @option
@item mono
video is not stereo
@item left_right
@@ -429,11 +338,10 @@ For example a 3D WebM clip can be created using the following command line:
ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
@end example
@subsection Options
This muxer supports the following options:
@table @option
@item reserve_index_space
By default, this muxer writes the index for seeking (called cues in Matroska
terms) at the end of the file, because it cannot know in advance how much space
@@ -448,6 +356,7 @@ for most use cases should be about 50kB per hour of video.
Note that cues are only written if the output is seekable and this option will
have no effect if it is not.
@end table
@anchor{md5}
@@ -477,9 +386,7 @@ ffmpeg -i INPUT -f md5 -
See also the @ref{framemd5} muxer.
@section mov, mp4, ismv
MOV/MP4/ISMV (Smooth Streaming) muxer.
@section MOV/MP4/ISMV
The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
file has all the metadata about all packets stored in one location
@@ -495,8 +402,6 @@ very long files (since writing normal MOV/MP4 files stores info about
every single packet in memory until the file is closed). The downside
is that it is less compatible with other applications.
@subsection Options
Fragmentation is enabled by setting one of the AVOptions that define
how to cut the file into fragments:
@@ -553,8 +458,6 @@ as fragmented output, thus it is not enabled by default.
Add RTP hinting tracks to the output file.
@end table
@subsection Example
Smooth Streaming content can be pushed in real time to a publishing
point on IIS with this muxer. Example:
@example
@@ -565,15 +468,12 @@ ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe
The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
@code{id3v2_version} option controls which one is used. Setting
@code{id3v2_version} to 0 will disable the ID3v2 header completely. The legacy
ID3v1 tag is not written by default, but may be enabled with the
@code{write_id3v1} option.
@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
not written by default, but may be enabled with the @code{write_id3v1} option.
The muxer may also write a Xing frame at the beginning, which contains the
number of frames in the file. It is useful for computing duration of VBR files.
The Xing frame is written if the output stream is seekable and if the
@code{write_xing} option is set to 1 (the default).
For seekable output the muxer also writes a Xing frame at the beginning, which
contains the number of frames in the file. It is useful for computing duration
of VBR files.
The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
are supplied to the muxer in form of a video stream with a single packet. There
@@ -600,24 +500,12 @@ ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
@end example
Write a "clean" MP3 without any extra features:
@example
ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3
@end example
@section mpegts
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The recognized metadata settings in mpegts muxer are @code{service_provider}
and @code{service_name}. If they are not set the default for
@code{service_provider} is "FFmpeg" and the default for
@code{service_name} is "Service01".
@subsection Options
The muxer options are:
@table @option
@@ -637,10 +525,7 @@ Set the first PID for data packets (default 0x0100, max 0x0f00).
@item -mpegts_m2ts_mode @var{number}
Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
@item -muxrate @var{number}
Set a constant muxrate (default VBR).
@item -pcr_period @var{numer}
Override the default PCR retransmission time (default 20ms), ignored
if variable muxrate is selected.
Set muxrate.
@item -pes_payload_size @var{number}
Set minimum PES packet payload in bytes.
@item -mpegts_flags @var{flags}
@@ -673,7 +558,10 @@ Reemit PAT/PMT before writing the next packet.
Use LATM packetization for AAC.
@end table
@subsection Example
The recognized metadata settings in mpegts muxer are @code{service_provider}
and @code{service_name}. If they are not set the default for
@code{service_provider} is "FFmpeg" and the default for
@code{service_name} is "Service01".
@example
ffmpeg -i file.mpg -c copy \
@@ -709,30 +597,6 @@ Alternatively you can write the command as:
ffmpeg -benchmark -i INPUT -f null -
@end example
@section nut
@table @option
@item -syncpoints @var{flags}
Change the syncpoint usage in nut:
@table @option
@item @var{default} use the normal low-overhead seeking aids.
@item @var{none} do not use the syncpoints at all, reducing the overhead but making the stream non-seekable;
Use of this option is not recommended, as the resulting files are very damage
sensitive and seeking is not possible. Also in general the overhead from
syncpoints is negligible. Note, -@code{write_index} 0 can be used to disable
all growing data tables, allowing to mux endless streams with limited memory
and wihout these disadvantages.
@item @var{timestamped} extend the syncpoint with a wallclock field.
@end table
The @var{none} and @var{timestamped} flags are experimental.
@item -write_index @var{bool}
Write index at the end, the default is to write an index.
@end table
@example
ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor
@end example
@section ogg
Ogg container muxer.
@@ -748,12 +612,11 @@ situations, giving a small seek granularity at the cost of additional container
overhead.
@end table
@anchor{segment}
@section segment, stream_segment, ssegment
Basic stream segmenter.
This muxer outputs streams to a number of separate files of nearly
The segmenter muxer outputs streams to a number of separate files of nearly
fixed duration. Output filename pattern can be set in a fashion similar to
@ref{image2}.
@@ -775,21 +638,14 @@ The segment muxer works best with a single constant frame rate video.
Optionally it can generate a list of the created segments, by setting
the option @var{segment_list}. The list type is specified by the
@var{segment_list_type} option. The entry filenames in the segment
list are set by default to the basename of the corresponding segment
files.
See also the @ref{hls} muxer, which provides a more specific
implementation for HLS segmentation.
@subsection Options
@var{segment_list_type} option.
The segment muxer supports the following options:
@table @option
@item reference_stream @var{specifier}
Set the reference stream, as specified by the string @var{specifier}.
If @var{specifier} is set to @code{auto}, the reference is chosen
If @var{specifier} is set to @code{auto}, the reference is choosen
automatically. Otherwise it must be a stream specifier (see the ``Stream
specifiers'' chapter in the ffmpeg manual) which specifies the
reference stream. The default value is @code{auto}.
@@ -814,21 +670,15 @@ Allow caching (only affects M3U8 list files).
Allow live-friendly file generation.
@end table
@item segment_list_type @var{type}
Select the listing format.
@table @option
@item @var{flat} use a simple flat list of entries.
@item @var{hls} use a m3u8-like structure.
@end table
Default value is @code{samp}.
@item segment_list_size @var{size}
Update the list file so that it contains at most the last @var{size}
segments. If 0 the list file will contain all the segments. Default
value is 0.
@item segment_list_entry_prefix @var{prefix}
Prepend @var{prefix} to each entry. Useful to generate absolute paths.
By default no prefix is applied.
@item segment_list_type @var{type}
Specify the format for the segment list file.
The following values are recognized:
@table @samp
@@ -879,16 +729,6 @@ Note that splitting may not be accurate, unless you force the
reference stream key-frames at the given time. See the introductory
notice and the examples below.
@item segment_atclocktime @var{1|0}
If set to "1" split at regular clock time intervals starting from 00:00
o'clock. The @var{time} value specified in @option{segment_time} is
used for setting the length of the splitting interval.
For example with @option{segment_time} set to "900" this makes it possible
to create files at 12:00 o'clock, 12:15, 12:30, etc.
Default value is "0".
@item segment_time_delta @var{delta}
Specify the accuracy time when selecting the start time for a
segment, expressed as a duration specification. Default value is "0".
@@ -908,7 +748,7 @@ In particular may be used in combination with the @file{ffmpeg} option
@var{force_key_frames} may not be set accurately because of rounding
issues, with the consequence that a key frame time may result set just
before the specified time. For constant frame rate videos a value of
1/(2*@var{frame_rate}) should address the worst case mismatch between
1/2*@var{frame_rate} should address the worst case mismatch between
the specified time and the time set by @var{force_key_frames}.
@item segment_times @var{times}
@@ -994,28 +834,6 @@ ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
@end example
@end itemize
@section smoothstreaming
Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server.
@table @option
@item window_size
Specify the number of fragments kept in the manifest. Default 0 (keep all).
@item extra_window_size
Specify the number of fragments kept outside of the manifest before removing from disk. Default 5.
@item lookahead_count
Specify the number of lookahead fragments. Default 2.
@item min_frag_duration
Specify the minimum fragment duration (in microseconds). Default 5000000.
@item remove_at_exit
Specify whether to remove all fragments when finished. Default 0 (do not remove).
@end table
@section tee
The tee muxer can be used to write the same data to several files or any
@@ -1031,8 +849,8 @@ to feed the same packets to several muxers directly.
The slave outputs are specified in the file name given to the muxer,
separated by '|'. If any of the slave name contains the '|' separator,
leading or trailing spaces or any special character, it must be
escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
manual).
Muxer options can be specified for each slave by prepending them as a list of
@var{key}=@var{value} pairs separated by ':', between square brackets. If
@@ -1047,13 +865,10 @@ output name suffix.
@item bsfs[/@var{spec}]
Specify a list of bitstream filters to apply to the specified
output.
It is possible to specify to which streams a given bitstream filter
applies, by appending a stream specifier to the option separated by
@code{/}. @var{spec} must be a stream specifier (see @ref{Format
stream specifiers}). If the stream specifier is not specified, the
bitstream filters will be applied to all streams in the output.
output. It is possible to specify to which streams a given bitstream
filter applies, by appending a stream specifier to the option
separated by @code{/}. If the stream specifier is not specified, the
bistream filters will be applied to all streams in the output.
Several bitstream filters can be specified, separated by ",".
@@ -1063,8 +878,7 @@ specified by a stream specifier. If not specified, this defaults to
all the input streams.
@end table
@subsection Examples
Some examples follow.
@itemize
@item
Encode something and both archive it in a WebM file and stream it
@@ -1085,49 +899,10 @@ audio packets.
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
@end example
@item
As below, but select only stream @code{a:1} for the audio output. Note
that a second level escaping must be performed, as ":" is a special
character used to separate options.
@example
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
@end example
@end itemize
Note: some codecs may need different options depending on the output format;
the auto-detection of this can not work with the tee muxer. The main example
is the @option{global_header} flag.
@section webm_dash_manifest
WebM DASH Manifest muxer.
This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
@subsection Options
This muxer supports the following options:
@table @option
@item adaptation_sets
This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
audio and video streams. Any number of adaptation sets can be added using this option.
@end table
@subsection Example
@example
ffmpeg -f webm_dash_manifest -i video1.webm \
-f webm_dash_manifest -i video2.webm \
-f webm_dash_manifest -i audio1.webm \
-f webm_dash_manifest -i audio2.webm \
-map 0 -map 1 -map 2 -map 3 \
-c copy \
-f webm_dash_manifest \
-adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
manifest.xml
@end example
@c man end MUXERS

View File

@@ -21,27 +21,6 @@ The official nut specification is at svn://svn.mplayerhq.hu/nut
In case of any differences between this text and the official specification,
the official specification shall prevail.
@chapter Modes
NUT has some variants signaled by using the flags field in its main header.
@multitable @columnfractions .4 .4
@item BROADCAST @tab Extend the syncpoint to report the sender wallclock
@item PIPE @tab Omit completely the syncpoint
@end multitable
@section BROADCAST
The BROADCAST variant provides a secondary time reference to facilitate
detecting endpoint latency and network delays.
It assumes all the endpoint clocks are syncronized.
To be used in real-time scenarios.
@section PIPE
The PIPE variant assumes NUT is used as non-seekable intermediate container,
by not using syncpoint removes unneeded overhead and reduces the overall
memory usage.
@chapter Container-specific codec tags
@section Generic raw YUVA formats

View File

@@ -79,6 +79,9 @@ qpel{8,16}_mc??_old_c / *pixels{8,16}_l4
Just used to work around a bug in an old libavcodec encoder version.
Don't optimize them.
tpel_mc_func {put,avg}_tpel_pixels_tab
Used only for SVQ3, so only optimize them if you need fast SVQ3 decoding.
add_bytes/diff_bytes
For huffyuv only, optimize if you want a faster ffhuffyuv codec.
@@ -136,6 +139,9 @@ dct_unquantize_mpeg2
dct_unquantize_h263
Used in MPEG-4/H.263 en/decoding.
FIXME remaining functions?
BTW, most of these functions are in dsputil.c/.h, some are in mpegvideo.c/.h.
Alignment:
@@ -262,6 +268,17 @@ CELL/SPU:
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf
SPARC-specific:
---------------
SPARC Joint Programming Specification (JPS1): Commonality
http://www.fujitsu.com/downloads/PRMPWR/JPS1-R1.0.4-Common-pub.pdf
UltraSPARC III Processor User's Manual (contains instruction timings)
http://www.sun.com/processors/manuals/USIIIv2.pdf
VIS Whitepaper (contains optimization guidelines)
http://www.sun.com/processors/vis/download/vis/vis_whitepaper.pdf
GCC asm links:
--------------
official doc but quite ugly

View File

@@ -13,8 +13,8 @@ You can disable all the output devices using the configure option
option "--enable-outdev=@var{OUTDEV}", or you can disable a particular
input device using the option "--disable-outdev=@var{OUTDEV}".
The option "-devices" of the ff* tools will display the list of
enabled output devices.
The option "-formats" of the ff* tools will display the list of
enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
@@ -22,27 +22,11 @@ A description of the currently available output devices follows.
ALSA (Advanced Linux Sound Architecture) output device.
@subsection Examples
@itemize
@item
Play a file on default ALSA device:
@example
ffmpeg -i INPUT -f alsa default
@end example
@item
Play a file on soundcard 1, audio device 7:
@example
ffmpeg -i INPUT -f alsa hw:1,7
@end example
@end itemize
@section caca
CACA output device.
This output device allows one to show a video stream in CACA window.
This output device allows to show a video stream in CACA window.
Only one CACA window is allowed per application, so you can
have only one instance of this output device in an application.
@@ -120,68 +104,6 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
@end example
@end itemize
@section decklink
The decklink output device provides playback capabilities for Blackmagic
DeckLink devices.
To enable this output device, you need the Blackmagic DeckLink SDK and you
need to configure with the appropriate @code{--extra-cflags}
and @code{--extra-ldflags}.
On Windows, you need to run the IDL files through @command{widl}.
DeckLink is very picky about the formats it supports. Pixel format is always
uyvy422, framerate and video size must be determined for your device with
@command{-list_formats 1}. Audio sample rate is always 48 kHz.
@subsection Options
@table @option
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
Defaults to @option{false}.
@item preroll
Amount of time to preroll video in seconds.
Defaults to @option{0.5}.
@end table
@subsection Examples
@itemize
@item
List output devices:
@example
ffmpeg -i test.avi -f decklink -list_devices 1 dummy
@end example
@item
List supported formats:
@example
ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
@end example
@item
Play video clip:
@example
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
@end example
@item
Play video clip with non-standard framerate or video size:
@example
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
@end example
@end itemize
@section fbdev
Linux framebuffer output device.
@@ -211,45 +133,6 @@ ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section opengl
OpenGL output device.
To enable this output device you need to configure FFmpeg with @code{--enable-opengl}.
This output device allows one to render to OpenGL context.
Context may be provided by application or default SDL window is created.
When device renders to external context, application must implement handlers for following messages:
@code{AV_DEV_TO_APP_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread.
@code{AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER} - make OpenGL context current.
@code{AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER} - swap buffers.
@code{AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER} - destroy OpenGL context.
Application is also required to inform a device about current resolution by sending @code{AV_APP_TO_DEV_WINDOW_SIZE} message.
@subsection Options
@table @option
@item background
Set background color. Black is a default.
@item no_window
Disables default SDL window when set to non-zero value.
Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set.
@item window_title
Set the SDL window title, if not specified default to the filename specified for the output device.
Ignored when @option{no_window} is set.
@item window_size
Set preferred window size, can be a string of the form widthxheight or a video size abbreviation.
If not specified it defaults to the size of the input video, downscaled according to the aspect ratio.
Mostly usable when @option{no_window} is not set.
@end table
@subsection Examples
Play a file on SDL window using OpenGL rendering:
@example
ffmpeg -i INPUT -f opengl "window title"
@end example
@section oss
OSS (Open Sound System) output device.
@@ -281,33 +164,6 @@ by default it is set to the specified output name.
Specify the device to use. Default device is used when not provided.
List of output devices can be obtained with command @command{pactl list sinks}.
@item buffer_size
@item buffer_duration
Control the size and duration of the PulseAudio buffer. A small buffer
gives more control, but requires more frequent updates.
@option{buffer_size} specifies size in bytes while
@option{buffer_duration} specifies duration in milliseconds.
When both options are provided then the highest value is used
(duration is recalculated to bytes using stream parameters). If they
are set to 0 (which is default), the device will use the default
PulseAudio duration value. By default PulseAudio set buffer duration
to around 2 seconds.
@item prebuf
Specify pre-buffering size in bytes. The server does not start with
playback before at least @option{prebuf} bytes are available in the
buffer. By default this option is initialized to the same value as
@option{buffer_size} or @option{buffer_duration} (whichever is bigger).
@item minreq
Specify minimum request size in bytes. The server does not request less
than @option{minreq} bytes from the client, instead waits until the buffer
is free enough to request more bytes at once. It is recommended to not set
this option, which will initialize this to a value that is deemed sensible
by the server.
@end table
@subsection Examples
@@ -320,7 +176,7 @@ ffmpeg -i INPUT -f pulse "stream name"
SDL (Simple DirectMedia Layer) output device.
This output device allows one to show a video stream in an SDL
This output device allows to show a video stream in an SDL
window. Only one SDL window is allowed per application, so you can
have only one instance of this output device in an application.
@@ -350,17 +206,7 @@ downscaled according to the aspect ratio.
@item window_fullscreen
Set fullscreen mode when non-zero value is provided.
Default value is zero.
@end table
@subsection Interactive commands
The window created by the device can be controlled through the
following interactive commands.
@table @key
@item q, ESC
Quit the device immediately.
Zero is a default.
@end table
@subsection Examples
@@ -379,7 +225,7 @@ sndio audio output device.
XV (XVideo) output device.
This output device allows one to show a video stream in a X Window System
This output device allows to show a video stream in a X Window System
window.
@subsection Options
@@ -406,26 +252,19 @@ For example, @code{dual-headed:0.1} would specify screen 1 of display
Check the X11 specification for more detailed information about the
display name format.
@item window_id
When set to non-zero value then device doesn't create new window,
but uses existing one with provided @var{window_id}. By default
this options is set to zero and device creates its own window.
@item window_size
Set the created window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation. If not
specified it defaults to the size of the input video.
Ignored when @var{window_id} is set.
@item window_x
@item window_y
Set the X and Y window offsets for the created window. They are both
set to 0 by default. The values may be ignored by the window manager.
Ignored when @var{window_id} is set.
@item window_title
Set the window title, if not specified default to the filename
specified for the output device. Ignored when @var{window_id} is set.
specified for the output device.
@end table
For more information about XVideo see @url{http://www.x.org/}.

View File

@@ -24,17 +24,6 @@ If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass @code{--disable-asm}
to configure.
@section Advanced linking configuration
If you compiled FFmpeg libraries statically and you want to use them to
build your own shared library, you may need to force PIC support (with
@code{--enable-pic} during FFmpeg configure) and add the following option
to your project LDFLAGS:
@example
-Wl,-Bsymbolic
@end example
@section BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
@@ -62,15 +51,14 @@ The toolchain provided with Xcode is sufficient to build the basic
unacelerated code.
Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
@url{https://github.com/FFmpeg/gas-preprocessor} or
@url{https://github.com/yuvi/gas-preprocessor}(currently outdated) to build the optimized
assembly functions. Put the Perl script somewhere
@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
assembler functions. Just download the Perl script and put it somewhere
in your PATH, FFmpeg's configure will pick it up automatically.
Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
optimized assembly functions. @uref{http://www.finkproject.org/, Fink},
optimized assembler functions. @uref{http://www.finkproject.org/, Fink},
@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix},
@uref{https://mxcl.github.com/homebrew/, Homebrew}
@uref{http://mxcl.github.com/homebrew/, Homebrew}
or @uref{http://www.macports.org, MacPorts} can easily provide it.
@@ -120,16 +108,14 @@ libavformat) as DLLs.
@section Microsoft Visual C++ or Intel C++ Compiler for Windows
FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
and wrapper, or with MSVC 2013 and ICL natively.
FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
wrapper. For ICL, only the wrapper is used, since ICL supports C99.
You will need the following prerequisites:
@itemize
@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
(if using MSVC 2012 or earlier)
@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
(if using MSVC 2012 or earlier)
@item @uref{http://www.mingw.org/, MSYS}
@item @uref{http://yasm.tortall.net/, YASM}
@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
@@ -139,16 +125,14 @@ you want to run @uref{fate.html, FATE}.
To set up a proper environment in MSYS, you need to run @code{msys.bat} from
the Visual Studio or Intel Compiler command prompt.
Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
@code{PATH} as well.
Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
somewhere in your @code{PATH}.
Next, make sure any other headers and libs you want to use, such as zlib, are
located in a spot that the compiler can see. Do so by modifying the @code{LIB}
and @code{INCLUDE} environment variables to include the @strong{Windows-style}
paths to these directories. Alternatively, you can try and use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options. If using MSVC
2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.
Next, make sure @code{inttypes.h} and any other headers and libs you want to use
are located in a spot that the compiler can see. Do so by modifying the @code{LIB}
and @code{INCLUDE} environment variables to include the @strong{Windows} paths to
these directories. Alternatively, you can try and use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options.
Finally, run:
@@ -198,9 +182,7 @@ can see.
@itemize
@item Visual Studio 2010 Pro and Express
@item Visual Studio 2012 Pro and Express
@item Visual Studio 2013 Pro and Express
@item Intel Composer XE 2013
@item Intel Composer XE 2013 SP1
@end itemize
Anything else is not officially supported.

View File

@@ -117,19 +117,7 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////
File access protocol.
Allow to read from or write to a file.
A file URL can have the form:
@example
file:@var{filename}
@end example
where @var{filename} is the path of the file to read.
An URL that does not have a protocol prefix will be assumed to be a
file URL. Depending on the build, an URL that looks like a Windows
path with the drive letter at the beginning will also be assumed to be
a file URL (usually not the case in builds for unix-like systems).
Allow to read from or read to a file.
For example to read from a file @file{input.mpeg} with @command{ffmpeg}
use the command:
@@ -137,6 +125,10 @@ use the command:
ffmpeg -i file:input.mpeg output.mpeg
@end example
The ff* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".
This protocol accepts the following options:
@table @option
@@ -166,7 +158,7 @@ This protocol accepts the following options.
@table @option
@item timeout
Set timeout in microseconds of socket I/O operations used by the underlying low level
Set timeout of socket I/O operations used by the underlying low level
operation. By default it is set to -1, which means that the timeout is
not specified.
@@ -213,7 +205,7 @@ m3u8 files.
HTTP (Hyper Text Transfer Protocol).
This protocol accepts the following options:
This protocol accepts the following options.
@table @option
@item seekable
@@ -223,33 +215,32 @@ if set to -1 it will try to autodetect if it is seekable. Default
value is -1.
@item chunked_post
If set to 1 use chunked Transfer-Encoding for posts, default is 1.
@item content_type
Set a specific content type for the POST messages.
If set to 1 use chunked transfer-encoding for posts, default is 1.
@item headers
Set custom HTTP headers, can override built in default headers. The
value must be a string encoding the headers.
@item content_type
Force a content type.
@item user-agent
Override User-Agent header. If not specified the protocol will use a
string describing the libavformat build.
@item multiple_requests
Use persistent connections if set to 1, default is 0.
Use persistent connections if set to 1. By default it is 0.
@item post_data
Set custom HTTP post data.
@item user-agent
@item user_agent
Override the User-Agent header. If not specified the protocol will use a
string describing the libavformat build. ("Lavf/<version>")
@item timeout
Set timeout in microseconds of socket I/O operations used by the underlying low level
Set timeout of socket I/O operations used by the underlying low level
operation. By default it is set to -1, which means that the timeout is
not specified.
@item mime_type
Export the MIME type.
Set MIME type.
@item icy
If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
@@ -258,25 +249,17 @@ the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
The default is 0.
@item icy_metadata_headers
If the server supports ICY metadata, this contains the ICY-specific HTTP reply
headers, separated by newline characters.
If the server supports ICY metadata, this contains the ICY specific HTTP reply
headers, separated with newline characters.
@item icy_metadata_packet
If the server supports ICY metadata, and @option{icy} was set to 1, this
contains the last non-empty metadata packet sent by the server. It should be
polled in regular intervals by applications interested in mid-stream metadata
updates.
contains the last non-empty metadata packet sent by the server.
@item cookies
Set the cookies to be sent in future requests. The format of each cookie is the
same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
delimited by a newline character.
@item offset
Set initial byte offset.
@item end_offset
Try to limit the request to bytes preceding this offset.
@end table
@subsection HTTP Cookies
@@ -537,35 +520,6 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
for streaming multimedia content within HTTPS requests to traverse
firewalls.
@section libsmbclient
libsmbclient permits to manipulate CIFS/SMB network resources.
Following syntax is required.
@example
smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
@end example
This protocol accepts the following options.
@table @option
@item timeout
Set timeout in miliseconds of socket I/O operations used by the underlying
low level operation. By default it is set to -1, which means that the timeout
is not specified.
@item truncate
Truncate existing files on write, if set to 1. A value of 0 prevents
truncating. Default value is 1.
@item workgroup
Set the workgroup used for making connections. By default workgroup is not specified.
@end table
For more information see: @url{http://www.samba.org/}.
@section libssh
Secure File Transfer Protocol via libssh
@@ -590,10 +544,6 @@ is not specified.
Truncate existing files on write, if set to 1. A value of 0 prevents
truncating. Default value is 1.
@item private_key
Specify the path of the file containing private key to use during authorization.
By default libssh searches for keys in the @file{~/.ssh/} directory.
@end table
Example: Play a file stored on remote server.
@@ -711,8 +661,6 @@ set to the the local RTP port value plus 1.
@section rtsp
Real-Time Streaming Protocol.
RTSP is not technically a protocol handler in libavformat, it is a demuxer
and muxer. The demuxer supports both normal RTSP (with data transferred
over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
@@ -720,29 +668,21 @@ data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server
supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
@uref{https://github.com/revmischa/rtsp-server, RTSP server}).
@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
The required syntax for a RTSP url is:
@example
rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example
Options can be set on the @command{ffmpeg}/@command{ffplay} command
line, or set in code via @code{AVOption}s or in
@code{avformat_open_input}.
The following options (set on the @command{ffmpeg}/@command{ffplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:
The following options are supported.
Flags for @code{rtsp_transport}:
@table @option
@item initial_pause
Do not start playing the stream immediately if set to 1. Default value
is 0.
@item rtsp_transport
Set RTSP transport protocols.
It accepts the following values:
@table @samp
@item udp
Use UDP as lower transport protocol.
@@ -760,56 +700,15 @@ passing proxies.
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
For the muxer, only the @code{tcp} and @code{udp} options are supported.
@item rtsp_flags
Set RTSP flags.
Flags for @code{rtsp_flags}:
The following values are accepted:
@table @samp
@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
@item listen
Act as a server, listening for an incoming connection.
@item prefer_tcp
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
@end table
Default value is @samp{none}.
@item allowed_media_types
Set media types to accept from the server.
The following flags are accepted:
@table @samp
@item video
@item audio
@item data
@end table
By default it accepts all media types.
@item min_port
Set minimum local UDP port. Default value is 5000.
@item max_port
Set maximum local UDP port. Default value is 65000.
@item timeout
Set maximum timeout (in seconds) to wait for incoming connections.
A value of -1 means infinite (default). This option implies the
@option{rtsp_flags} set to @samp{listen}.
@item reorder_queue_size
Set number of packets to buffer for handling of reordered packets.
@item stimeout
Set socket TCP I/O timeout in microseconds.
@item user-agent
Override User-Agent header. If not specified, it defaults to the
libavformat identifier string.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
@@ -822,36 +721,36 @@ streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
on the fly by pressing @code{v} and @code{a}.
@subsection Examples
Example command lines:
The following examples all make use of the @command{ffplay} and
@command{ffmpeg} tools.
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@itemize
@item
Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@example
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example
@item
Watch a stream tunneled over HTTP:
To watch a stream tunneled over HTTP:
@example
ffplay -rtsp_transport http rtsp://server/video.mp4
@end example
@item
Send a stream in realtime to a RTSP server, for others to watch:
To send a stream in realtime to a RTSP server, for others to watch:
@example
ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example
@item
Receive a stream in realtime:
To receive a stream in realtime:
@example
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@end example
@end itemize
@table @option
@item stimeout
Socket IO timeout in micro seconds.
@end table
@section sap
@@ -989,65 +888,32 @@ this binary block are used as master key, the following 14 bytes are
used as master salt.
@end table
@section subfile
Virtually extract a segment of a file or another stream.
The underlying stream must be seekable.
Accepted options:
@table @option
@item start
Start offset of the extracted segment, in bytes.
@item end
End offset of the extracted segment, in bytes.
@end table
Examples:
Extract a chapter from a DVD VOB file (start and end sectors obtained
externally and multiplied by 2048):
@example
subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
@end example
Play an AVI file directly from a TAR archive:
subfile,,start,183241728,end,366490624,,:archive.tar
@section tcp
Transmission Control Protocol.
Trasmission Control Protocol.
The required syntax for a TCP url is:
@example
tcp://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}.
The list of supported options follows.
@table @option
@item listen=@var{1|0}
Listen for an incoming connection. Default value is 0.
@item listen
Listen for an incoming connection
@item timeout=@var{microseconds}
Set raise error timeout, expressed in microseconds.
In read mode: if no data arrived in more than this time interval, raise error.
In write mode: if socket cannot be written in more than this time interval, raise error.
This also sets timeout on TCP connection establishing.
This option is only relevant in read mode: if no data arrived in more
than this time interval, raise error.
@item listen_timeout=@var{microseconds}
Set listen timeout, expressed in microseconds.
@end table
The following example shows how to setup a listening TCP connection
with @command{ffmpeg}, which is then accessed with @command{ffplay}:
@example
ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
ffplay tcp://@var{hostname}:@var{port}
@end example
@end table
@section tls
Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
@@ -1113,7 +979,7 @@ ffplay tls://@var{hostname}:@var{port}
User Datagram Protocol.
The required syntax for an UDP URL is:
The required syntax for a UDP url is:
@example
udp://@var{hostname}:@var{port}[?@var{options}]
@end example
@@ -1121,17 +987,17 @@ udp://@var{hostname}:@var{port}[?@var{options}]
@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
In case threading is enabled on the system, a circular buffer is used
to store the incoming data, which allows one to reduce loss of data due to
to store the incoming data, which allows to reduce loss of data due to
UDP socket buffer overruns. The @var{fifo_size} and
@var{overrun_nonfatal} options are related to this buffer.
The list of supported options follows.
@table @option
@item buffer_size=@var{size}
Set the UDP maximum socket buffer size in bytes. This is used to set either
the receive or send buffer size, depending on what the socket is used for.
Default is 64KB. See also @var{fifo_size}.
Set the UDP socket buffer size in bytes. This is used both for the
receiving and the sending buffer size.
@item localport=@var{port}
Override the local UDP port to bind with.
@@ -1178,40 +1044,25 @@ Survive in case of UDP receiving circular buffer overrun. Default
value is 0.
@item timeout=@var{microseconds}
Set raise error timeout, expressed in microseconds.
This option is only relevant in read mode: if no data arrived in more
than this time interval, raise error.
@item broadcast=@var{1|0}
Explicitly allow or disallow UDP broadcasting.
Note that broadcasting may not work properly on networks having
a broadcast storm protection.
In read mode: if no data arrived in more than this time interval, raise error.
@end table
@subsection Examples
Some usage examples of the UDP protocol with @command{ffmpeg} follow.
@itemize
@item
Use @command{ffmpeg} to stream over UDP to a remote endpoint:
To stream over UDP to a remote endpoint:
@example
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example
@item
Use @command{ffmpeg} to stream in mpegts format over UDP using 188
sized UDP packets, using a large input buffer:
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
@example
ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
@end example
@item
Use @command{ffmpeg} to receive over UDP from a remote endpoint:
To receive over UDP from a remote endpoint:
@example
ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
ffmpeg -i udp://[@var{multicast-address}]:@var{port}
@end example
@end itemize
@section unix

View File

@@ -112,14 +112,6 @@ bayer dither
@item ed
error diffusion dither
@item a_dither
arithmetic dither, based using addition
@item x_dither
arithmetic dither, based using xor (more random/less apparent patterning that
a_dither).
@end table
@end table

24
doc/soc.txt Normal file
View File

@@ -0,0 +1,24 @@
Google Summer of Code and similar project guidelines
Summer of Code is a project by Google in which students are paid to implement
some nice new features for various participating open source projects ...
This text is a collection of things to take care of for the next soc as
it's a little late for this year's soc (2006).
The Goal:
Our goal in respect to soc is and must be of course exactly one thing and
that is to improve FFmpeg, to reach this goal, code must
* conform to the development policy and patch submission guidelines
* must improve FFmpeg somehow (faster, smaller, "better",
more codecs supported, fewer bugs, cleaner, ...)
for mentors and other developers to help students to reach that goal it is
essential that changes to their codebase are publicly visible, clean and
easy reviewable that again leads us to:
* use of a revision control system like git
* separation of cosmetic from non-cosmetic changes (this is almost entirely
ignored by mentors and students in soc 2006 which might lead to a surprise
when the code will be reviewed at the end before a possible inclusion in
FFmpeg, individual changes were generally not reviewable due to cosmetics).
* frequent commits, so that comments can be provided early

23
doc/texi2pod.pl Normal file → Executable file
View File

@@ -1,4 +1,4 @@
#!/usr/bin/env perl
#! /usr/bin/perl
# Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
@@ -282,14 +282,6 @@ INF: while(<$inf>) {
$_ = "\n=over 4\n";
};
/^\@(multitable)\s+{.*/ and do {
push @endwstack, $endw;
push @icstack, $ic;
$endw = $1;
$ic = "";
$_ = "\n=over 4\n";
};
/^\@((?:small)?example|display)/ and do {
push @endwstack, $endw;
$endw = $1;
@@ -306,10 +298,10 @@ INF: while(<$inf>) {
/^\@tab\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
my $columns = $1;
$columns =~ s/\@tab//;
$columns =~ s/\@tab/ : /;
$_ = $columns;
$chapter =~ s/$//;
$_ = " : ". $columns;
$chapter =~ s/\n+\s+$//;
};
/^\@itemx?\s*(.+)?$/ and do {
@@ -335,9 +327,6 @@ die "No filename or title\n" unless defined $fn && defined $tl;
$chapters{NAME} = "$fn \- $tl\n";
$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};
# always use utf8
print "=encoding utf8\n\n";
unshift @chapters_sequence, "NAME";
for $chapter (@chapters_sequence) {
if (exists $chapters{$chapter}) {
@@ -388,8 +377,8 @@ sub postprocess
s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
s/;\s+\@pxref\{(?:[^\}]*)\}//g;
s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/B<$1>/g;
s/\@ref\{([^\}]*)\}/B<$1>/g;
s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g;
s/\@ref\{([^\}]*)\}/$1/g;
s/\@noindent\s*//g;
s/\@refill//g;
s/\@gol//g;

View File

@@ -1,32 +0,0 @@
#! /usr/bin/env perl
# This script will print the dependency of a Texinfo file to stdout.
# texidep.pl <src-path> <input.texi> <output.ext>
use warnings;
use strict;
die unless @ARGV == 3;
my ($src_path, $root, $target) = @ARGV;
sub print_deps {
my ($file, $deps) = @_;
$deps->{$file} = 1;
open(my $fh, "<", "$file") or die "Cannot open file '$file': $!";
while (<$fh>) {
if (my ($i) = /^\@(?:verbatim)?include\s+(\S+)/) {
die "Circular dependency found in file $root\n" if exists $deps->{"doc/$1"};
print "$target: doc/$1\n";
# skip looking for config.texi dependencies, since it has
# none, and is not located in the source tree
if ("$1" ne "config.texi") {
print_deps("$src_path/doc/$1", {%$deps});
}
}
}
}
print_deps($root, {});

View File

@@ -1031,7 +1031,7 @@ indication of the corresponding powers of 10 and of 2.
10^24 / 2^70
@end table
@c man end EXPRESSION EVALUATION
@c man end
@chapter OpenCL Options
@c man begin OPENCL OPTIONS
@@ -1051,13 +1051,13 @@ See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
Select the index of the platform to run OpenCL code.
The specified index must be one of the indexes in the device list
which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
which can be obtained with @code{av_opencl_get_device_list()}.
@item device_idx
Select the index of the device used to run OpenCL code.
The specified index must be one of the indexes in the device list which
can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
The specifed index must be one of the indexes in the device list which
can be obtained with @code{av_opencl_get_device_list()}.
@end table

109
doc/viterbi.txt Normal file
View File

@@ -0,0 +1,109 @@
This is a quick description of the viterbi aka dynamic programing
algorthm.
Its reason for existence is that wikipedia has become very poor on
describing algorithms in a way that makes it useable for understanding
them or anything else actually. It tends now to describe the very same
algorithm under 50 different names and pages with few understandable
by even people who fully understand the algorithm and the theory behind.
Problem description: (that is what it can solve)
assume we have a 2d table, or you could call it a graph or matrix if you
prefer
O O O O O O O
O O O O O O O
O O O O O O O
O O O O O O O
That table has edges connecting points from each column to the next column
and each edge has a score like: (only some edge and scores shown to keep it
readable)
O--5--O-----O-----O-----O-----O
2 / 7 / \ / \ / \ /
\ / \ / \ / \ / \ /
O7-/--O--/--O--/--O--/--O--/--O
\/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
/\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
O3-/--O--/--O--/--O--/--O--/--O
/ \ / \ / \ / \ / \
1 \ 9 \ / \ / \ / \
O--2--O--1--O--5--O--3--O--8--O
Our goal is to find a path from left to right through it which
minimizes the sum of the score of all edges.
(and of course left/right is just a convention here it could be top down too)
Similarly the minimum could be the maximum by just fliping the sign,
Example of a path with scores:
O O O O O O O
>---O. O O .O-2-O O O
5. .7 .
O O-1-O O O 8 O O
.
O O O O O O-1-O---> (sum here is 24)
The viterbi algorthm now solves this simply column by column
For the previous column each point has a best path and a associated
score:
O-----5 O
\
\
O \ 1 O
\/
/\
O / 2 O
/
/
O-----2 O
To move one column forward we just need to find the best path and associated
scores for the next column
here are some edges we could choose from:
O-----5--3--O
\ \8
\ \
O \ 1--9--O
\/ \3
/\ \
O / 2--1--O
/ \2
/ \
O-----2--4--O
Finding the new best paths and scores for each point of our new column is
trivial given we know the previous column best paths and scores:
O-----0-----8
\
\
O \ 0----10
\/
/\
O / 0-----3
/ \
/ \
O 0 4
the viterbi algorthm continues exactly like this column for column until the
end and then just picks the path with the best score (above that would be the
one with score 3)
Author: Michael niedermayer
Copyright LGPL

View File

@@ -1,424 +0,0 @@
This document is a tutorial/initiation for writing simple filters in
libavfilter.
Foreword: just like everything else in FFmpeg, libavfilter is monolithic, which
means that it is highly recommended that you submit your filters to the FFmpeg
development mailing-list and make sure it is applied. Otherwise, your filter is
likely to have a very short lifetime due to more a less regular internal API
changes, and a limited distribution, review, and testing.
Bootstrap
=========
Let's say you want to write a new simple video filter called "foobar" which
takes one frame in input, changes the pixels in whatever fashion you fancy, and
outputs the modified frame. The most simple way of doing this is to take a
similar filter. We'll pick edgedetect, but any other should do. You can look
for others using the `./ffmpeg -v 0 -filters|grep ' V->V '` command.
- cp libavfilter/vf_{edgedetect,foobar}.c
- sed -i s/edgedetect/foobar/g -i libavfilter/vf_foobar.c
- sed -i s/EdgeDetect/Foobar/g -i libavfilter/vf_foobar.c
- edit libavfilter/Makefile, and add an entry for "foobar" following the
pattern of the other filters.
- edit libavfilter/allfilters.c, and add an entry for "foobar" following the
pattern of the other filters.
- ./configure ...
- make -j<whatever> ffmpeg
- ./ffmpeg -i tests/lena.pnm -vf foobar foobar.png
If everything went right, you should get a foobar.png with Lena edge-detected.
That's it, your new playground is ready.
Some little details about what's going on:
libavfilter/allfilters.c:avfilter_register_all() is called at runtime to create
a list of the available filters, but it's important to know that this file is
also parsed by the configure script, which in turn will define variables for
the build system and the C:
--- after running configure ---
$ grep FOOBAR config.mak
CONFIG_FOOBAR_FILTER=yes
$ grep FOOBAR config.h
#define CONFIG_FOOBAR_FILTER 1
CONFIG_FOOBAR_FILTER=yes from the config.mak is later used to enable the filter in
libavfilter/Makefile and CONFIG_FOOBAR_FILTER=1 from the config.h will be used
for registering the filter in libavfilter/allfilters.c.
Filter code layout
==================
You now need some theory about the general code layout of a filter. Open your
libavfilter/vf_foobar.c. This section will detail the important parts of the
code you need to understand before messing with it.
Copyright
---------
First chunk is the copyright. Most filters are LGPL, and we are assuming
vf_foobar is as well. We are also assuming vf_foobar is not an edge detector
filter, so you can update the boilerplate with your credits.
Doxy
----
Next chunk is the Doxygen about the file. See http://ffmpeg.org/doxygen/trunk/.
Detail here what the filter is, does, and add some references if you feel like
it.
Context
-------
Skip the headers and scroll down to the definition of FoobarContext. This is
your local state context. It is already filled with 0 when you get it so do not
worry about uninitialized read into this context. This is where you put every
"global" information you need, typically the variable storing the user options.
You'll notice the first field "const AVClass *class"; it's the only field you
need to keep assuming you have a context. There are some magic you don't care
about around this field, just let it be (in first position) for now.
Options
-------
Then comes the options array. This is what will define the user accessible
options. For example, -vf foobar=mode=colormix:high=0.4:low=0.1. Most options
have the following pattern:
name, description, offset, type, default value, minimum value, maximum value, flags
- name is the option name, keep it simple, lowercase
- description are short, in lowercase, without period, and describe what they
do, for example "set the foo of the bar"
- offset is the offset of the field in your local context, see the OFFSET()
macro; the option parser will use that information to fill the fields
according to the user input
- type is any of AV_OPT_TYPE_* defined in libavutil/opt.h
- default value is an union where you pick the appropriate type; "{.dbl=0.3}",
"{.i64=0x234}", "{.str=NULL}", ...
- min and max values define the range of available values, inclusive
- flags are AVOption generic flags. See AV_OPT_FLAG_* definitions
In doubt, just look at the other AVOption definitions all around the codebase,
there are tons of examples.
Class
-----
AVFILTER_DEFINE_CLASS(foobar) will define a unique foobar_class with some kind
of signature referencing the options, etc. which will be referenced in the
definition of the AVFilter.
Filter definition
-----------------
At the end of the file, you will find foobar_inputs, foobar_outputs and
the AVFilter ff_vf_foobar. Don't forget to update the AVFilter.description with
a description of what the filter does, starting with a capitalized letter and
ending with a period. You'd better drop the AVFilter.flags entry for now, and
re-add them later depending on the capabilities of your filter.
Callbacks
---------
Let's now study the common callbacks. Before going into details, note that all
these callbacks are explained in details in libavfilter/avfilter.h, so in
doubt, refer to the doxy in that file.
init()
~~~~~~
First one to be called is init(). It's flagged as cold because not called
often. Look for "cold" on
http://gcc.gnu.org/onlinedocs/gcc/Function-Attributes.html for more
information.
As the name suggests, init() is where you eventually initialize and allocate
your buffers, pre-compute your data, etc. Note that at this point, your local
context already has the user options initialized, but you still haven't any
clue about the kind of data input you will get, so this function is often
mainly used to sanitize the user options.
Some init()s will also define the number of inputs or outputs dynamically
according to the user options. A good example of this is the split filter, but
we won't cover this here since vf_foobar is just a simple 1:1 filter.
uninit()
~~~~~~~~
Similarly, there is the uninit() callback, doing what the name suggest. Free
everything you allocated here.
query_formats()
~~~~~~~~~~~~~~~
This is following the init() and is used for the format negotiation, basically
where you say what pixel format(s) (gray, rgb 32, yuv 4:2:0, ...) you accept
for your inputs, and what you can output. All pixel formats are defined in
libavutil/pixfmt.h. If you don't change the pixel format between the input and
the output, you just have to define a pixel formats array and call
ff_set_common_formats(). For more complex negotiation, you can refer to other
filters such as vf_scale.
config_props()
~~~~~~~~~~~~~~
This callback is not necessary, but you will probably have one or more
config_props() anyway. It's not a callback for the filter itself but for its
inputs or outputs (they're called "pads" - AVFilterPad - in libavfilter's
lexicon).
Inside the input config_props(), you are at a point where you know which pixel
format has been picked after query_formats(), and more information such as the
video width and height (inlink->{w,h}). So if you need to update your internal
context state depending on your input you can do it here. In edgedetect you can
see that this callback is used to allocate buffers depending on these
information. They will be destroyed in uninit().
Inside the output config_props(), you can define what you want to change in the
output. Typically, if your filter is going to double the size of the video, you
will update outlink->w and outlink->h.
filter_frame()
~~~~~~~~~~~~~~
This is the callback you are waiting from the beginning: it is where you
process the received frames. Along with the frame, you get the input link from
where the frame comes from.
static int filter_frame(AVFilterLink *inlink, AVFrame *in) { ... }
You can get the filter context through that input link:
AVFilterContext *ctx = inlink->dst;
Then access your internal state context:
FoobarContext *foobar = ctx->priv;
And also the output link where you will send your frame when you are done:
AVFilterLink *outlink = ctx->outputs[0];
Here, we are picking the first output. You can have several, but in our case we
only have one since we are in a 1:1 input-output situation.
If you want to define a simple pass-through filter, you can just do:
return ff_filter_frame(outlink, in);
But of course, you probably want to change the data of that frame.
This can be done by accessing frame->data[] and frame->linesize[]. Important
note here: the width does NOT match the linesize. The linesize is always
greater or equal to the width. The padding created should not be changed or
even read. Typically, keep in mind that a previous filter in your chain might
have altered the frame dimension but not the linesize. Imagine a crop filter
that halves the video size: the linesizes won't be changed, just the width.
<-------------- linesize ------------------------>
+-------------------------------+----------------+ ^
| | | |
| | | |
| picture | padding | | height
| | | |
| | | |
+-------------------------------+----------------+ v
<----------- width ------------->
Before modifying the "in" frame, you have to make sure it is writable, or get a
new one. Multiple scenarios are possible here depending on the kind of
processing you are doing.
Let's say you want to change one pixel depending on multiple pixels (typically
the surrounding ones) of the input. In that case, you can't do an in-place
processing of the input so you will need to allocate a new frame, with the same
properties as the input one, and send that new frame to the next filter:
AVFrame *out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
// out->data[...] = foobar(in->data[...])
av_frame_free(&in);
return ff_filter_frame(outlink, out);
In-place processing
~~~~~~~~~~~~~~~~~~~
If you can just alter the input frame, you probably just want to do that
instead:
av_frame_make_writable(in);
// in->data[...] = foobar(in->data[...])
return ff_filter_frame(outlink, in);
You may wonder why a frame might not be writable. The answer is that for
example a previous filter might still own the frame data: imagine a filter
prior to yours in the filtergraph that needs to cache the frame. You must not
alter that frame, otherwise it will make that previous filter buggy. This is
where av_frame_make_writable() helps (it won't have any effect if the frame
already is writable).
The problem with using av_frame_make_writable() is that in the worst case it
will copy the whole input frame before you change it all over again with your
filter: if the frame is not writable, av_frame_make_writable() will allocate
new buffers, and copy the input frame data. You don't want that, and you can
avoid it by just allocating a new buffer if necessary, and process from in to
out in your filter, saving the memcpy. Generally, this is done following this
scheme:
int direct = 0;
AVFrame *out;
if (av_frame_is_writable(in)) {
direct = 1;
out = in;
} else {
out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
// out->data[...] = foobar(in->data[...])
if (!direct)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
Of course, this will only work if you can do in-place processing. To test if
your filter handles well the permissions, you can use the perms filter. For
example with:
-vf perms=random,foobar
Make sure no automatic pixel conversion is inserted between perms and foobar,
otherwise the frames permissions might change again and the test will be
meaningless: add av_log(0,0,"direct=%d\n",direct) in your code to check that.
You can avoid the issue with something like:
-vf format=rgb24,perms=random,foobar
...assuming your filter accepts rgb24 of course. This will make sure the
necessary conversion is inserted before the perms filter.
Timeline
~~~~~~~~
Adding timeline support
(http://ffmpeg.org/ffmpeg-filters.html#Timeline-editing) is often an easy
feature to add. In the most simple case, you just have to add
AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC to the AVFilter.flags. You can typically
do this when your filter does not need to save the previous context frames, or
basically if your filter just alter whatever goes in and doesn't need
previous/future information. See for instance commit 86cb986ce that adds
timeline support to the fieldorder filter.
In some cases, you might need to reset your context somehow. This is handled by
the AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL flag which is used if the filter
must not process the frames but still wants to keep track of the frames going
through (to keep them in cache for when it's enabled again). See for example
commit 69d72140a that adds timeline support to the phase filter.
Threading
~~~~~~~~~
libavfilter does not yet support frame threading, but you can add slice
threading to your filters.
Let's say the foobar filter has the following frame processing function:
dst = out->data[0];
src = in ->data[0];
for (y = 0; y < inlink->h; y++) {
for (x = 0; x < inlink->w; x++)
dst[x] = foobar(src[x]);
dst += out->linesize[0];
src += in ->linesize[0];
}
The first thing is to make this function work into slices. The new code will
look like this:
for (y = slice_start; y < slice_end; y++) {
for (x = 0; x < inlink->w; x++)
dst[x] = foobar(src[x]);
dst += out->linesize[0];
src += in ->linesize[0];
}
The source and destination pointers, and slice_start/slice_end will be defined
according to the number of jobs. Generally, it looks like this:
const int slice_start = (in->height * jobnr ) / nb_jobs;
const int slice_end = (in->height * (jobnr+1)) / nb_jobs;
uint8_t *dst = out->data[0] + slice_start * out->linesize[0];
const uint8_t *src = in->data[0] + slice_start * in->linesize[0];
This new code will be isolated in a new filter_slice():
static int filter_slice(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { ... }
Note that we need our input and output frame to define slice_{start,end} and
dst/src, which are not available in that callback. They will be transmitted
through the opaque void *arg. You have to define a structure which contains
everything you need:
typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
If you need some more information from your local context, put them here.
In you filter_slice function, you access it like that:
const ThreadData *td = arg;
Then in your filter_frame() callback, you need to call the threading
distributor with something like this:
ThreadData td;
// ...
td.in = in;
td.out = out;
ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ctx->graph->nb_threads));
// ...
return ff_filter_frame(outlink, out);
Last step is to add AVFILTER_FLAG_SLICE_THREADS flag to AVFilter.flags.
For more example of slice threading additions, you can try to run git log -p
--grep 'slice threading' libavfilter/
Finalization
~~~~~~~~~~~~
When your awesome filter is finished, you have a few more steps before you're
done:
- write its documentation in doc/filters.texi, and test the output with make
doc/ffmpeg-filters.html.
- add a FATE test, generally by adding an entry in
tests/fate/filter-video.mak, add running make fate-filter-foobar GEN=1 to
generate the data.
- add an entry in the Changelog
- edit libavfilter/version.h and increase LIBAVFILTER_VERSION_MINOR by one
(and reset LIBAVFILTER_VERSION_MICRO to 100)
- git add ... && git commit -m "avfilter: add foobar filter." && git format-patch -1
When all of this is done, you can submit your patch to the ffmpeg-devel
mailing-list for review. If you need any help, feel free to come on our IRC
channel, #ffmpeg-devel on irc.freenode.net.

1242
ffmpeg.c

File diff suppressed because it is too large Load Diff

View File

@@ -44,7 +44,6 @@
#include "libavutil/fifo.h"
#include "libavutil/pixfmt.h"
#include "libavutil/rational.h"
#include "libavutil/threadmessage.h"
#include "libswresample/swresample.h"
@@ -57,21 +56,6 @@
#define MAX_STREAMS 1024 /* arbitrary sanity check value */
enum HWAccelID {
HWACCEL_NONE = 0,
HWACCEL_AUTO,
HWACCEL_VDPAU,
HWACCEL_DXVA2,
HWACCEL_VDA,
};
typedef struct HWAccel {
const char *name;
int (*init)(AVCodecContext *s);
enum HWAccelID id;
enum AVPixelFormat pix_fmt;
} HWAccel;
/* select an input stream for an output stream */
typedef struct StreamMap {
int disabled; /* 1 is this mapping is disabled by a negative map */
@@ -116,10 +100,6 @@ typedef struct OptionsContext {
int nb_ts_scale;
SpecifierOpt *dump_attachment;
int nb_dump_attachment;
SpecifierOpt *hwaccels;
int nb_hwaccels;
SpecifierOpt *hwaccel_devices;
int nb_hwaccel_devices;
/* output options */
StreamMap *stream_maps;
@@ -174,8 +154,6 @@ typedef struct OptionsContext {
int nb_intra_matrices;
SpecifierOpt *inter_matrices;
int nb_inter_matrices;
SpecifierOpt *chroma_intra_matrices;
int nb_chroma_intra_matrices;
SpecifierOpt *top_field_first;
int nb_top_field_first;
SpecifierOpt *metadata_map;
@@ -204,8 +182,6 @@ typedef struct OptionsContext {
int nb_guess_layout_max;
SpecifierOpt *apad;
int nb_apad;
SpecifierOpt *discard;
int nb_discard;
} OptionsContext;
typedef struct InputFilter {
@@ -242,9 +218,7 @@ typedef struct InputStream {
int file_index;
AVStream *st;
int discard; /* true if stream data should be discarded */
int user_set_discard;
int decoding_needed; /* true if the packets must be decoded in 'raw_fifo' */
AVCodecContext *dec_ctx;
AVCodec *dec;
AVFrame *decoded_frame;
AVFrame *filter_frame; /* a ref of decoded_frame, to be sent to filters */
@@ -262,9 +236,10 @@ typedef struct InputStream {
int64_t filter_in_rescale_delta_last;
double ts_scale;
int is_start; /* is 1 at the start and after a discontinuity */
int saw_first_ts;
int showed_multi_packet_warning;
AVDictionary *decoder_opts;
AVDictionary *opts;
AVRational framerate; /* framerate forced with -r */
int top_field_first;
int guess_layout_max;
@@ -300,28 +275,6 @@ typedef struct InputStream {
int nb_filters;
int reinit_filters;
/* hwaccel options */
enum HWAccelID hwaccel_id;
char *hwaccel_device;
/* hwaccel context */
enum HWAccelID active_hwaccel_id;
void *hwaccel_ctx;
void (*hwaccel_uninit)(AVCodecContext *s);
int (*hwaccel_get_buffer)(AVCodecContext *s, AVFrame *frame, int flags);
int (*hwaccel_retrieve_data)(AVCodecContext *s, AVFrame *frame);
enum AVPixelFormat hwaccel_pix_fmt;
enum AVPixelFormat hwaccel_retrieved_pix_fmt;
/* stats */
// combined size of all the packets read
uint64_t data_size;
/* number of packets successfully read for this stream */
uint64_t nb_packets;
// number of frames/samples retrieved from the decoder
uint64_t frames_decoded;
uint64_t samples_decoded;
} InputStream;
typedef struct InputFile {
@@ -341,10 +294,12 @@ typedef struct InputFile {
int accurate_seek;
#if HAVE_PTHREADS
AVThreadMessageQueue *in_thread_queue;
pthread_t thread; /* thread reading from this file */
int non_blocking; /* reading packets from the thread should not block */
int finished; /* the thread has exited */
int joined; /* the thread has been joined */
pthread_mutex_t fifo_lock; /* lock for access to fifo */
pthread_cond_t fifo_cond; /* the main thread will signal on this cond after reading from fifo */
AVFifoBuffer *fifo; /* demuxed packets are stored here; freed by the main thread */
#endif
} InputFile;
@@ -359,11 +314,6 @@ enum forced_keyframes_const {
extern const char *const forced_keyframes_const_names[];
typedef enum {
ENCODER_FINISHED = 1,
MUXER_FINISHED = 2,
} OSTFinished ;
typedef struct OutputStream {
int file_index; /* file index */
int index; /* stream index in the output file */
@@ -381,7 +331,6 @@ typedef struct OutputStream {
/* dts of the last packet sent to the muxer */
int64_t last_mux_dts;
AVBitStreamFilterContext *bitstream_filters;
AVCodecContext *enc_ctx;
AVCodec *enc;
int64_t max_frames;
AVFrame *filtered_frame;
@@ -410,15 +359,13 @@ typedef struct OutputStream {
OutputFilter *filter;
char *avfilter;
char *filters; ///< filtergraph associated to the -filter option
char *filters_script; ///< filtergraph script associated to the -filter_script option
int64_t sws_flags;
AVDictionary *encoder_opts;
AVDictionary *opts;
AVDictionary *swr_opts;
AVDictionary *resample_opts;
char *apad;
OSTFinished finished; /* no more packets should be written for this stream */
int finished; /* no more packets should be written for this stream */
int unavailable; /* true if the steram is unavailable (possibly temporarily) */
int stream_copy;
const char *attachment_filename;
@@ -426,17 +373,6 @@ typedef struct OutputStream {
int copy_prior_start;
int keep_pix_fmt;
AVCodecParserContext *parser;
/* stats */
// combined size of all the packets written
uint64_t data_size;
// number of packets send to the muxer
uint64_t packets_written;
// number of frames/samples sent to the encoder
uint64_t frames_encoded;
uint64_t samples_encoded;
} OutputStream;
typedef struct OutputFile {
@@ -491,8 +427,6 @@ extern float max_error_rate;
extern const AVIOInterruptCB int_cb;
extern const OptionDef options[];
extern const HWAccel hwaccels[];
void term_init(void);
void term_exit(void);
@@ -506,7 +440,7 @@ void assert_avoptions(AVDictionary *m);
int guess_input_channel_layout(InputStream *ist);
enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodecContext *avctx, AVCodec *codec, enum AVPixelFormat target);
enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFormat target);
void choose_sample_fmt(AVStream *st, AVCodec *codec);
int configure_filtergraph(FilterGraph *fg);
@@ -516,8 +450,4 @@ FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
int ffmpeg_parse_options(int argc, char **argv);
int vdpau_init(AVCodecContext *s);
int dxva2_init(AVCodecContext *s);
int vda_init(AVCodecContext *s);
#endif /* FFMPEG_H */

View File

@@ -1,620 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <windows.h>
#ifdef _WIN32_WINNT
#undef _WIN32_WINNT
#endif
#define _WIN32_WINNT 0x0600
#define DXVA2API_USE_BITFIELDS
#define COBJMACROS
#include <stdint.h>
#include <d3d9.h>
#include <dxva2api.h>
#include "ffmpeg.h"
#include "libavcodec/dxva2.h"
#include "libavutil/avassert.h"
#include "libavutil/buffer.h"
#include "libavutil/frame.h"
#include "libavutil/imgutils.h"
#include "libavutil/pixfmt.h"
/* define all the GUIDs used directly here,
to avoid problems with inconsistent dxva2api.h versions in mingw-w64 and different MSVC version */
#include <initguid.h>
DEFINE_GUID(IID_IDirectXVideoDecoderService, 0xfc51a551,0xd5e7,0x11d9,0xaf,0x55,0x00,0x05,0x4e,0x43,0xff,0x02);
DEFINE_GUID(DXVA2_ModeMPEG2_VLD, 0xee27417f, 0x5e28,0x4e65,0xbe,0xea,0x1d,0x26,0xb5,0x08,0xad,0xc9);
DEFINE_GUID(DXVA2_ModeMPEG2and1_VLD, 0x86695f12, 0x340e,0x4f04,0x9f,0xd3,0x92,0x53,0xdd,0x32,0x74,0x60);
DEFINE_GUID(DXVA2_ModeH264_E, 0x1b81be68, 0xa0c7,0x11d3,0xb9,0x84,0x00,0xc0,0x4f,0x2e,0x73,0xc5);
DEFINE_GUID(DXVA2_ModeH264_F, 0x1b81be69, 0xa0c7,0x11d3,0xb9,0x84,0x00,0xc0,0x4f,0x2e,0x73,0xc5);
DEFINE_GUID(DXVADDI_Intel_ModeH264_E, 0x604F8E68, 0x4951,0x4C54,0x88,0xFE,0xAB,0xD2,0x5C,0x15,0xB3,0xD6);
DEFINE_GUID(DXVA2_ModeVC1_D, 0x1b81beA3, 0xa0c7,0x11d3,0xb9,0x84,0x00,0xc0,0x4f,0x2e,0x73,0xc5);
DEFINE_GUID(DXVA2_ModeVC1_D2010, 0x1b81beA4, 0xa0c7,0x11d3,0xb9,0x84,0x00,0xc0,0x4f,0x2e,0x73,0xc5);
DEFINE_GUID(DXVA2_NoEncrypt, 0x1b81beD0, 0xa0c7,0x11d3,0xb9,0x84,0x00,0xc0,0x4f,0x2e,0x73,0xc5);
DEFINE_GUID(GUID_NULL, 0x00000000, 0x0000,0x0000,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00);
typedef IDirect3D9* WINAPI pDirect3DCreate9(UINT);
typedef HRESULT WINAPI pCreateDeviceManager9(UINT *, IDirect3DDeviceManager9 **);
typedef struct dxva2_mode {
const GUID *guid;
enum AVCodecID codec;
} dxva2_mode;
static const dxva2_mode dxva2_modes[] = {
/* MPEG-2 */
{ &DXVA2_ModeMPEG2_VLD, AV_CODEC_ID_MPEG2VIDEO },
{ &DXVA2_ModeMPEG2and1_VLD, AV_CODEC_ID_MPEG2VIDEO },
/* H.264 */
{ &DXVA2_ModeH264_F, AV_CODEC_ID_H264 },
{ &DXVA2_ModeH264_E, AV_CODEC_ID_H264 },
/* Intel specific H.264 mode */
{ &DXVADDI_Intel_ModeH264_E, AV_CODEC_ID_H264 },
/* VC-1 / WMV3 */
{ &DXVA2_ModeVC1_D2010, AV_CODEC_ID_VC1 },
{ &DXVA2_ModeVC1_D2010, AV_CODEC_ID_WMV3 },
{ &DXVA2_ModeVC1_D, AV_CODEC_ID_VC1 },
{ &DXVA2_ModeVC1_D, AV_CODEC_ID_WMV3 },
{ NULL, 0 },
};
typedef struct surface_info {
int used;
uint64_t age;
} surface_info;
typedef struct DXVA2Context {
HMODULE d3dlib;
HMODULE dxva2lib;
HANDLE deviceHandle;
IDirect3D9 *d3d9;
IDirect3DDevice9 *d3d9device;
IDirect3DDeviceManager9 *d3d9devmgr;
IDirectXVideoDecoderService *decoder_service;
IDirectXVideoDecoder *decoder;
GUID decoder_guid;
DXVA2_ConfigPictureDecode decoder_config;
LPDIRECT3DSURFACE9 *surfaces;
surface_info *surface_infos;
uint32_t num_surfaces;
uint64_t surface_age;
AVFrame *tmp_frame;
} DXVA2Context;
typedef struct DXVA2SurfaceWrapper {
DXVA2Context *ctx;
LPDIRECT3DSURFACE9 surface;
IDirectXVideoDecoder *decoder;
} DXVA2SurfaceWrapper;
static void dxva2_destroy_decoder(AVCodecContext *s)
{
InputStream *ist = s->opaque;
DXVA2Context *ctx = ist->hwaccel_ctx;
int i;
if (ctx->surfaces) {
for (i = 0; i < ctx->num_surfaces; i++) {
if (ctx->surfaces[i])
IDirect3DSurface9_Release(ctx->surfaces[i]);
}
}
av_freep(&ctx->surfaces);
av_freep(&ctx->surface_infos);
ctx->num_surfaces = 0;
ctx->surface_age = 0;
if (ctx->decoder) {
IDirectXVideoDecoder_Release(ctx->decoder);
ctx->decoder = NULL;
}
}
static void dxva2_uninit(AVCodecContext *s)
{
InputStream *ist = s->opaque;
DXVA2Context *ctx = ist->hwaccel_ctx;
ist->hwaccel_uninit = NULL;
ist->hwaccel_get_buffer = NULL;
ist->hwaccel_retrieve_data = NULL;
if (ctx->decoder)
dxva2_destroy_decoder(s);
if (ctx->decoder_service)
IDirectXVideoDecoderService_Release(ctx->decoder_service);
if (ctx->d3d9devmgr && ctx->deviceHandle != INVALID_HANDLE_VALUE)
IDirect3DDeviceManager9_CloseDeviceHandle(ctx->d3d9devmgr, ctx->deviceHandle);
if (ctx->d3d9devmgr)
IDirect3DDeviceManager9_Release(ctx->d3d9devmgr);
if (ctx->d3d9device)
IDirect3DDevice9_Release(ctx->d3d9device);
if (ctx->d3d9)
IDirect3D9_Release(ctx->d3d9);
if (ctx->d3dlib)
FreeLibrary(ctx->d3dlib);
if (ctx->dxva2lib)
FreeLibrary(ctx->dxva2lib);
av_frame_free(&ctx->tmp_frame);
av_freep(&ist->hwaccel_ctx);
av_freep(&s->hwaccel_context);
}
static void dxva2_release_buffer(void *opaque, uint8_t *data)
{
DXVA2SurfaceWrapper *w = opaque;
DXVA2Context *ctx = w->ctx;
int i;
for (i = 0; i < ctx->num_surfaces; i++) {
if (ctx->surfaces[i] == w->surface) {
ctx->surface_infos[i].used = 0;
break;
}
}
IDirect3DSurface9_Release(w->surface);
IDirectXVideoDecoder_Release(w->decoder);
av_free(w);
}
static int dxva2_get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
{
InputStream *ist = s->opaque;
DXVA2Context *ctx = ist->hwaccel_ctx;
int i, old_unused = -1;
LPDIRECT3DSURFACE9 surface;
DXVA2SurfaceWrapper *w = NULL;
av_assert0(frame->format == AV_PIX_FMT_DXVA2_VLD);
for (i = 0; i < ctx->num_surfaces; i++) {
surface_info *info = &ctx->surface_infos[i];
if (!info->used && (old_unused == -1 || info->age < ctx->surface_infos[old_unused].age))
old_unused = i;
}
if (old_unused == -1) {
av_log(NULL, AV_LOG_ERROR, "No free DXVA2 surface!\n");
return AVERROR(ENOMEM);
}
i = old_unused;
surface = ctx->surfaces[i];
w = av_mallocz(sizeof(*w));
if (!w)
return AVERROR(ENOMEM);
frame->buf[0] = av_buffer_create((uint8_t*)surface, 0,
dxva2_release_buffer, w,
AV_BUFFER_FLAG_READONLY);
if (!frame->buf[0]) {
av_free(w);
return AVERROR(ENOMEM);
}
w->ctx = ctx;
w->surface = surface;
IDirect3DSurface9_AddRef(w->surface);
w->decoder = ctx->decoder;
IDirectXVideoDecoder_AddRef(w->decoder);
ctx->surface_infos[i].used = 1;
ctx->surface_infos[i].age = ctx->surface_age++;
frame->data[3] = (uint8_t *)surface;
return 0;
}
static int dxva2_retrieve_data(AVCodecContext *s, AVFrame *frame)
{
LPDIRECT3DSURFACE9 surface = (LPDIRECT3DSURFACE9)frame->data[3];
InputStream *ist = s->opaque;
DXVA2Context *ctx = ist->hwaccel_ctx;
D3DSURFACE_DESC surfaceDesc;
D3DLOCKED_RECT LockedRect;
HRESULT hr;
int ret;
IDirect3DSurface9_GetDesc(surface, &surfaceDesc);
ctx->tmp_frame->width = frame->width;
ctx->tmp_frame->height = frame->height;
ctx->tmp_frame->format = AV_PIX_FMT_NV12;
ret = av_frame_get_buffer(ctx->tmp_frame, 32);
if (ret < 0)
return ret;
hr = IDirect3DSurface9_LockRect(surface, &LockedRect, NULL, D3DLOCK_READONLY);
if (FAILED(hr)) {
av_log(NULL, AV_LOG_ERROR, "Unable to lock DXVA2 surface\n");
return AVERROR_UNKNOWN;
}
av_image_copy_plane(ctx->tmp_frame->data[0], ctx->tmp_frame->linesize[0],
(uint8_t*)LockedRect.pBits,
LockedRect.Pitch, frame->width, frame->height);
av_image_copy_plane(ctx->tmp_frame->data[1], ctx->tmp_frame->linesize[1],
(uint8_t*)LockedRect.pBits + LockedRect.Pitch * surfaceDesc.Height,
LockedRect.Pitch, frame->width, frame->height / 2);
IDirect3DSurface9_UnlockRect(surface);
ret = av_frame_copy_props(ctx->tmp_frame, frame);
if (ret < 0)
goto fail;
av_frame_unref(frame);
av_frame_move_ref(frame, ctx->tmp_frame);
return 0;
fail:
av_frame_unref(ctx->tmp_frame);
return ret;
}
static int dxva2_alloc(AVCodecContext *s)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
DXVA2Context *ctx;
pDirect3DCreate9 *createD3D = NULL;
pCreateDeviceManager9 *createDeviceManager = NULL;
HRESULT hr;
D3DPRESENT_PARAMETERS d3dpp = {0};
D3DDISPLAYMODE d3ddm;
unsigned resetToken = 0;
UINT adapter = D3DADAPTER_DEFAULT;
ctx = av_mallocz(sizeof(*ctx));
if (!ctx)
return AVERROR(ENOMEM);
ctx->deviceHandle = INVALID_HANDLE_VALUE;
ist->hwaccel_ctx = ctx;
ist->hwaccel_uninit = dxva2_uninit;
ist->hwaccel_get_buffer = dxva2_get_buffer;
ist->hwaccel_retrieve_data = dxva2_retrieve_data;
ctx->d3dlib = LoadLibrary("d3d9.dll");
if (!ctx->d3dlib) {
av_log(NULL, loglevel, "Failed to load D3D9 library\n");
goto fail;
}
ctx->dxva2lib = LoadLibrary("dxva2.dll");
if (!ctx->dxva2lib) {
av_log(NULL, loglevel, "Failed to load DXVA2 library\n");
goto fail;
}
createD3D = (pDirect3DCreate9 *)GetProcAddress(ctx->d3dlib, "Direct3DCreate9");
if (!createD3D) {
av_log(NULL, loglevel, "Failed to locate Direct3DCreate9\n");
goto fail;
}
createDeviceManager = (pCreateDeviceManager9 *)GetProcAddress(ctx->dxva2lib, "DXVA2CreateDirect3DDeviceManager9");
if (!createDeviceManager) {
av_log(NULL, loglevel, "Failed to locate DXVA2CreateDirect3DDeviceManager9\n");
goto fail;
}
ctx->d3d9 = createD3D(D3D_SDK_VERSION);
if (!ctx->d3d9) {
av_log(NULL, loglevel, "Failed to create IDirect3D object\n");
goto fail;
}
if (ist->hwaccel_device) {
adapter = atoi(ist->hwaccel_device);
av_log(NULL, AV_LOG_INFO, "Using HWAccel device %d\n", adapter);
}
IDirect3D9_GetAdapterDisplayMode(ctx->d3d9, adapter, &d3ddm);
d3dpp.Windowed = TRUE;
d3dpp.BackBufferWidth = 640;
d3dpp.BackBufferHeight = 480;
d3dpp.BackBufferCount = 0;
d3dpp.BackBufferFormat = d3ddm.Format;
d3dpp.SwapEffect = D3DSWAPEFFECT_DISCARD;
d3dpp.Flags = D3DPRESENTFLAG_VIDEO;
hr = IDirect3D9_CreateDevice(ctx->d3d9, adapter, D3DDEVTYPE_HAL, GetShellWindow(),
D3DCREATE_SOFTWARE_VERTEXPROCESSING | D3DCREATE_MULTITHREADED | D3DCREATE_FPU_PRESERVE,
&d3dpp, &ctx->d3d9device);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to create Direct3D device\n");
goto fail;
}
hr = createDeviceManager(&resetToken, &ctx->d3d9devmgr);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to create Direct3D device manager\n");
goto fail;
}
hr = IDirect3DDeviceManager9_ResetDevice(ctx->d3d9devmgr, ctx->d3d9device, resetToken);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to bind Direct3D device to device manager\n");
goto fail;
}
hr = IDirect3DDeviceManager9_OpenDeviceHandle(ctx->d3d9devmgr, &ctx->deviceHandle);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to open device handle\n");
goto fail;
}
hr = IDirect3DDeviceManager9_GetVideoService(ctx->d3d9devmgr, ctx->deviceHandle, &IID_IDirectXVideoDecoderService, (void **)&ctx->decoder_service);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to create IDirectXVideoDecoderService\n");
goto fail;
}
ctx->tmp_frame = av_frame_alloc();
if (!ctx->tmp_frame)
goto fail;
s->hwaccel_context = av_mallocz(sizeof(struct dxva_context));
if (!s->hwaccel_context)
goto fail;
return 0;
fail:
dxva2_uninit(s);
return AVERROR(EINVAL);
}
static int dxva2_get_decoder_configuration(AVCodecContext *s, const GUID *device_guid,
const DXVA2_VideoDesc *desc,
DXVA2_ConfigPictureDecode *config)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
DXVA2Context *ctx = ist->hwaccel_ctx;
unsigned cfg_count = 0, best_score = 0;
DXVA2_ConfigPictureDecode *cfg_list = NULL;
DXVA2_ConfigPictureDecode best_cfg = {{0}};
HRESULT hr;
int i;
hr = IDirectXVideoDecoderService_GetDecoderConfigurations(ctx->decoder_service, device_guid, desc, NULL, &cfg_count, &cfg_list);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Unable to retrieve decoder configurations\n");
return AVERROR(EINVAL);
}
for (i = 0; i < cfg_count; i++) {
DXVA2_ConfigPictureDecode *cfg = &cfg_list[i];
unsigned score;
if (cfg->ConfigBitstreamRaw == 1)
score = 1;
else if (s->codec_id == AV_CODEC_ID_H264 && cfg->ConfigBitstreamRaw == 2)
score = 2;
else
continue;
if (IsEqualGUID(&cfg->guidConfigBitstreamEncryption, &DXVA2_NoEncrypt))
score += 16;
if (score > best_score) {
best_score = score;
best_cfg = *cfg;
}
}
CoTaskMemFree(cfg_list);
if (!best_score) {
av_log(NULL, loglevel, "No valid decoder configuration available\n");
return AVERROR(EINVAL);
}
*config = best_cfg;
return 0;
}
static int dxva2_create_decoder(AVCodecContext *s)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
DXVA2Context *ctx = ist->hwaccel_ctx;
struct dxva_context *dxva_ctx = s->hwaccel_context;
GUID *guid_list = NULL;
unsigned guid_count = 0, i, j;
GUID device_guid = GUID_NULL;
D3DFORMAT target_format = 0;
DXVA2_VideoDesc desc = { 0 };
DXVA2_ConfigPictureDecode config;
HRESULT hr;
int surface_alignment;
int ret;
hr = IDirectXVideoDecoderService_GetDecoderDeviceGuids(ctx->decoder_service, &guid_count, &guid_list);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to retrieve decoder device GUIDs\n");
goto fail;
}
for (i = 0; dxva2_modes[i].guid; i++) {
D3DFORMAT *target_list = NULL;
unsigned target_count = 0;
const dxva2_mode *mode = &dxva2_modes[i];
if (mode->codec != s->codec_id)
continue;
for (j = 0; j < guid_count; j++) {
if (IsEqualGUID(mode->guid, &guid_list[j]))
break;
}
if (j == guid_count)
continue;
hr = IDirectXVideoDecoderService_GetDecoderRenderTargets(ctx->decoder_service, mode->guid, &target_count, &target_list);
if (FAILED(hr)) {
continue;
}
for (j = 0; j < target_count; j++) {
const D3DFORMAT format = target_list[j];
if (format == MKTAG('N','V','1','2')) {
target_format = format;
break;
}
}
CoTaskMemFree(target_list);
if (target_format) {
device_guid = *mode->guid;
break;
}
}
CoTaskMemFree(guid_list);
if (IsEqualGUID(&device_guid, &GUID_NULL)) {
av_log(NULL, loglevel, "No decoder device for codec found\n");
goto fail;
}
desc.SampleWidth = s->coded_width;
desc.SampleHeight = s->coded_height;
desc.Format = target_format;
ret = dxva2_get_decoder_configuration(s, &device_guid, &desc, &config);
if (ret < 0) {
goto fail;
}
/* decoding MPEG-2 requires additional alignment on some Intel GPUs,
but it causes issues for H.264 on certain AMD GPUs..... */
if (s->codec_id == AV_CODEC_ID_MPEG2VIDEO)
surface_alignment = 32;
else
surface_alignment = 16;
/* 4 base work surfaces */
ctx->num_surfaces = 4;
/* add surfaces based on number of possible refs */
if (s->codec_id == AV_CODEC_ID_H264)
ctx->num_surfaces += 16;
else
ctx->num_surfaces += 2;
/* add extra surfaces for frame threading */
if (s->active_thread_type & FF_THREAD_FRAME)
ctx->num_surfaces += s->thread_count;
ctx->surfaces = av_mallocz(ctx->num_surfaces * sizeof(*ctx->surfaces));
ctx->surface_infos = av_mallocz(ctx->num_surfaces * sizeof(*ctx->surface_infos));
if (!ctx->surfaces || !ctx->surface_infos) {
av_log(NULL, loglevel, "Unable to allocate surface arrays\n");
goto fail;
}
hr = IDirectXVideoDecoderService_CreateSurface(ctx->decoder_service,
FFALIGN(s->coded_width, surface_alignment),
FFALIGN(s->coded_height, surface_alignment),
ctx->num_surfaces - 1,
target_format, D3DPOOL_DEFAULT, 0,
DXVA2_VideoDecoderRenderTarget,
ctx->surfaces, NULL);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to create %d video surfaces\n", ctx->num_surfaces);
goto fail;
}
hr = IDirectXVideoDecoderService_CreateVideoDecoder(ctx->decoder_service, &device_guid,
&desc, &config, ctx->surfaces,
ctx->num_surfaces, &ctx->decoder);
if (FAILED(hr)) {
av_log(NULL, loglevel, "Failed to create DXVA2 video decoder\n");
goto fail;
}
ctx->decoder_guid = device_guid;
ctx->decoder_config = config;
dxva_ctx->cfg = &ctx->decoder_config;
dxva_ctx->decoder = ctx->decoder;
dxva_ctx->surface = ctx->surfaces;
dxva_ctx->surface_count = ctx->num_surfaces;
if (IsEqualGUID(&ctx->decoder_guid, &DXVADDI_Intel_ModeH264_E))
dxva_ctx->workaround |= FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO;
return 0;
fail:
dxva2_destroy_decoder(s);
return AVERROR(EINVAL);
}
int dxva2_init(AVCodecContext *s)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
DXVA2Context *ctx;
int ret;
if (!ist->hwaccel_ctx) {
ret = dxva2_alloc(s);
if (ret < 0)
return ret;
}
ctx = ist->hwaccel_ctx;
if (s->codec_id == AV_CODEC_ID_H264 &&
(s->profile & ~FF_PROFILE_H264_CONSTRAINED) > FF_PROFILE_H264_HIGH) {
av_log(NULL, loglevel, "Unsupported H.264 profile for DXVA2 HWAccel: %d\n", s->profile);
return AVERROR(EINVAL);
}
if (ctx->decoder)
dxva2_destroy_decoder(s);
ret = dxva2_create_decoder(s);
if (ret < 0) {
av_log(NULL, loglevel, "Error creating the DXVA2 decoder\n");
return ret;
}
return 0;
}

View File

@@ -18,8 +18,6 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "ffmpeg.h"
#include "libavfilter/avfilter.h"
@@ -37,23 +35,21 @@
#include "libavutil/imgutils.h"
#include "libavutil/samplefmt.h"
enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodecContext *enc_ctx, AVCodec *codec, enum AVPixelFormat target)
enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFormat target)
{
if (codec && codec->pix_fmts) {
const enum AVPixelFormat *p = codec->pix_fmts;
const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
enum AVPixelFormat best= AV_PIX_FMT_NONE;
static const enum AVPixelFormat mjpeg_formats[] =
{ AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
static const enum AVPixelFormat ljpeg_formats[] =
{ AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
const enum AVPixelFormat mjpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
const enum AVPixelFormat ljpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
if (enc_ctx->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
if (enc_ctx->codec_id == AV_CODEC_ID_MJPEG) {
if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
if (st->codec->codec_id == AV_CODEC_ID_MJPEG) {
p = mjpeg_formats;
} else if (enc_ctx->codec_id == AV_CODEC_ID_LJPEG) {
} else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) {
p =ljpeg_formats;
}
}
@@ -99,21 +95,21 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)
static char *choose_pix_fmts(OutputStream *ost)
{
AVDictionaryEntry *strict_dict = av_dict_get(ost->encoder_opts, "strict", NULL, 0);
AVDictionaryEntry *strict_dict = av_dict_get(ost->opts, "strict", NULL, 0);
if (strict_dict)
// used by choose_pixel_fmt() and below
av_opt_set(ost->enc_ctx, "strict", strict_dict->value, 0);
av_opt_set(ost->st->codec, "strict", strict_dict->value, 0);
if (ost->keep_pix_fmt) {
if (ost->filter)
avfilter_graph_set_auto_convert(ost->filter->graph->graph,
AVFILTER_AUTO_CONVERT_NONE);
if (ost->enc_ctx->pix_fmt == AV_PIX_FMT_NONE)
if (ost->st->codec->pix_fmt == AV_PIX_FMT_NONE)
return NULL;
return av_strdup(av_get_pix_fmt_name(ost->enc_ctx->pix_fmt));
return av_strdup(av_get_pix_fmt_name(ost->st->codec->pix_fmt));
}
if (ost->enc_ctx->pix_fmt != AV_PIX_FMT_NONE) {
return av_strdup(av_get_pix_fmt_name(choose_pixel_fmt(ost->st, ost->enc_ctx, ost->enc, ost->enc_ctx->pix_fmt)));
if (ost->st->codec->pix_fmt != AV_PIX_FMT_NONE) {
return av_strdup(av_get_pix_fmt_name(choose_pixel_fmt(ost->st, ost->enc, ost->st->codec->pix_fmt)));
} else if (ost->enc && ost->enc->pix_fmts) {
const enum AVPixelFormat *p;
AVIOContext *s = NULL;
@@ -124,10 +120,10 @@ static char *choose_pix_fmts(OutputStream *ost)
exit_program(1);
p = ost->enc->pix_fmts;
if (ost->enc_ctx->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
if (ost->enc_ctx->codec_id == AV_CODEC_ID_MJPEG) {
if (ost->st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
if (ost->st->codec->codec_id == AV_CODEC_ID_MJPEG) {
p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
} else if (ost->enc_ctx->codec_id == AV_CODEC_ID_LJPEG) {
} else if (ost->st->codec->codec_id == AV_CODEC_ID_LJPEG) {
p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
}
@@ -149,8 +145,8 @@ static char *choose_pix_fmts(OutputStream *ost)
#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name) \
static char *choose_ ## var ## s(OutputStream *ost) \
{ \
if (ost->enc_ctx->var != none) { \
get_name(ost->enc_ctx->var); \
if (ost->st->codec->var != none) { \
get_name(ost->st->codec->var); \
return av_strdup(name); \
} else if (ost->enc && ost->enc->supported_list) { \
const type *p; \
@@ -262,7 +258,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
/* find the first unused stream of corresponding type */
for (i = 0; i < nb_input_streams; i++) {
ist = input_streams[i];
if (ist->dec_ctx->codec_type == type && ist->discard)
if (ist->st->codec->codec_type == type && ist->discard)
break;
}
if (i == nb_input_streams) {
@@ -344,7 +340,7 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
char *pix_fmts;
OutputStream *ost = ofilter->ost;
OutputFile *of = output_files[ost->file_index];
AVCodecContext *codec = ost->enc_ctx;
AVCodecContext *codec = ost->st->codec;
AVFilterContext *last_filter = out->filter_ctx;
int pad_idx = out->pad_idx;
int ret;
@@ -434,7 +430,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
{
OutputStream *ost = ofilter->ost;
OutputFile *of = output_files[ost->file_index];
AVCodecContext *codec = ost->enc_ctx;
AVCodecContext *codec = ost->st->codec;
AVFilterContext *last_filter = out->filter_ctx;
int pad_idx = out->pad_idx;
char *sample_fmts, *sample_rates, *channel_layouts;
@@ -595,8 +591,8 @@ static int sub2video_prepare(InputStream *ist)
/* Compute the size of the canvas for the subtitles stream.
If the subtitles codec has set a size, use it. Otherwise use the
maximum dimensions of the video streams in the same file. */
w = ist->dec_ctx->width;
h = ist->dec_ctx->height;
w = ist->st->codec->width;
h = ist->st->codec->height;
if (!(w && h)) {
for (i = 0; i < avf->nb_streams; i++) {
if (avf->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
@@ -610,12 +606,12 @@ static int sub2video_prepare(InputStream *ist)
}
av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h);
}
ist->sub2video.w = ist->dec_ctx->width = ist->resample_width = w;
ist->sub2video.h = ist->dec_ctx->height = ist->resample_height = h;
ist->sub2video.w = ist->st->codec->width = ist->resample_width = w;
ist->sub2video.h = ist->st->codec->height = ist->resample_height = h;
/* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the
palettes for all rectangles are identical or compatible */
ist->resample_pix_fmt = ist->dec_ctx->pix_fmt = AV_PIX_FMT_RGB32;
ist->resample_pix_fmt = ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32;
ist->sub2video.frame = av_frame_alloc();
if (!ist->sub2video.frame)
@@ -638,7 +634,7 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
char name[255];
int ret, pad_idx = 0;
if (ist->dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
av_log(NULL, AV_LOG_ERROR, "Cannot connect video filter to audio input\n");
return AVERROR(EINVAL);
}
@@ -646,7 +642,7 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
if (!fr.num)
fr = av_guess_frame_rate(input_files[ist->file_index]->ctx, ist->st, NULL);
if (ist->dec_ctx->codec_type == AVMEDIA_TYPE_SUBTITLE) {
if (ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
ret = sub2video_prepare(ist);
if (ret < 0)
return ret;
@@ -654,17 +650,16 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
sar = ist->st->sample_aspect_ratio.num ?
ist->st->sample_aspect_ratio :
ist->dec_ctx->sample_aspect_ratio;
ist->st->codec->sample_aspect_ratio;
if(!sar.den)
sar = (AVRational){0,1};
av_bprint_init(&args, 0, 1);
av_bprintf(&args,
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
"pixel_aspect=%d/%d:sws_param=flags=%d", ist->resample_width,
ist->resample_height,
ist->hwaccel_retrieve_data ? ist->hwaccel_retrieved_pix_fmt : ist->resample_pix_fmt,
ist->resample_height, ist->resample_pix_fmt,
tb.num, tb.den, sar.num, sar.den,
SWS_BILINEAR + ((ist->dec_ctx->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
SWS_BILINEAR + ((ist->st->codec->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
if (fr.num && fr.den)
av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
@@ -732,21 +727,21 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
char name[255];
int ret, pad_idx = 0;
if (ist->dec_ctx->codec_type != AVMEDIA_TYPE_AUDIO) {
if (ist->st->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
av_log(NULL, AV_LOG_ERROR, "Cannot connect audio filter to non audio input\n");
return AVERROR(EINVAL);
}
av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
av_bprintf(&args, "time_base=%d/%d:sample_rate=%d:sample_fmt=%s",
1, ist->dec_ctx->sample_rate,
ist->dec_ctx->sample_rate,
av_get_sample_fmt_name(ist->dec_ctx->sample_fmt));
if (ist->dec_ctx->channel_layout)
1, ist->st->codec->sample_rate,
ist->st->codec->sample_rate,
av_get_sample_fmt_name(ist->st->codec->sample_fmt));
if (ist->st->codec->channel_layout)
av_bprintf(&args, ":channel_layout=0x%"PRIx64,
ist->dec_ctx->channel_layout);
ist->st->codec->channel_layout);
else
av_bprintf(&args, ":channels=%d", ist->dec_ctx->channels);
av_bprintf(&args, ":channels=%d", ist->st->codec->channels);
snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
ist->file_index, ist->st->index);
@@ -874,7 +869,7 @@ int configure_filtergraph(FilterGraph *fg)
args[strlen(args) - 1] = '\0';
fg->graph->resample_lavr_opts = av_strdup(args);
e = av_dict_get(ost->encoder_opts, "threads", NULL, 0);
e = av_dict_get(ost->opts, "threads", NULL, 0);
if (e)
av_opt_set(fg->graph, "threads", e->value, 0);
}

View File

@@ -62,20 +62,6 @@
outvar = o->name[i].u.type;\
}\
}
const HWAccel hwaccels[] = {
#if HAVE_VDPAU_X11
{ "vdpau", vdpau_init, HWACCEL_VDPAU, AV_PIX_FMT_VDPAU },
#endif
#if HAVE_DXVA2_LIB
{ "dxva2", dxva2_init, HWACCEL_DXVA2, AV_PIX_FMT_DXVA2_VLD },
#endif
#if CONFIG_VDA
{ "vda", vda_init, HWACCEL_VDA, AV_PIX_FMT_VDA },
#endif
{ 0 },
};
char *vstats_filename;
float audio_drift_threshold = 0.1;
@@ -104,6 +90,8 @@ float max_error_rate = 2.0/3;
static int intra_only = 0;
static int file_overwrite = 0;
static int no_file_overwrite = 0;
static int video_discard = 0;
static int intra_dc_precision = 8;
static int do_psnr = 0;
static int input_sync;
static int override_ffserver = 0;
@@ -562,17 +550,14 @@ static AVCodec *choose_decoder(OptionsContext *o, AVFormatContext *s, AVStream *
* list of input streams. */
static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
{
int i, ret;
int i;
char *next, *codec_tag = NULL;
for (i = 0; i < ic->nb_streams; i++) {
AVStream *st = ic->streams[i];
AVCodecContext *dec = st->codec;
InputStream *ist = av_mallocz(sizeof(*ist));
char *framerate = NULL, *hwaccel = NULL, *hwaccel_device = NULL;
char *codec_tag = NULL;
char *next;
char *discard_str = NULL;
const AVOption *discard_opt = av_opt_find(dec, "skip_frame", NULL, 0, 0);
char *framerate = NULL;
if (!ist)
exit_program(1);
@@ -597,33 +582,13 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
}
ist->dec = choose_decoder(o, ic, st);
ist->decoder_opts = filter_codec_opts(o->g->codec_opts, ist->st->codec->codec_id, ic, st, ist->dec);
ist->opts = filter_codec_opts(o->g->codec_opts, ist->st->codec->codec_id, ic, st, ist->dec);
ist->reinit_filters = -1;
MATCH_PER_STREAM_OPT(reinit_filters, i, ist->reinit_filters, ic, st);
MATCH_PER_STREAM_OPT(discard, str, discard_str, ic, st);
ist->user_set_discard = AVDISCARD_NONE;
if (discard_str && av_opt_eval_int(dec, discard_opt, discard_str, &ist->user_set_discard) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error parsing discard %s.\n",
discard_str);
exit_program(1);
}
ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE;
ist->dec_ctx = avcodec_alloc_context3(ist->dec);
if (!ist->dec_ctx) {
av_log(NULL, AV_LOG_ERROR, "Error allocating the decoder context.\n");
exit_program(1);
}
ret = avcodec_copy_context(ist->dec_ctx, dec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error initializing the decoder context.\n");
exit_program(1);
}
switch (dec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
if(!ist->dec)
@@ -632,9 +597,9 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
dec->flags |= CODEC_FLAG_EMU_EDGE;
}
ist->resample_height = ist->dec_ctx->height;
ist->resample_width = ist->dec_ctx->width;
ist->resample_pix_fmt = ist->dec_ctx->pix_fmt;
ist->resample_height = dec->height;
ist->resample_width = dec->width;
ist->resample_pix_fmt = dec->pix_fmt;
MATCH_PER_STREAM_OPT(frame_rates, str, framerate, ic, st);
if (framerate && av_parse_video_rate(&ist->framerate,
@@ -647,51 +612,16 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
ist->top_field_first = -1;
MATCH_PER_STREAM_OPT(top_field_first, i, ist->top_field_first, ic, st);
MATCH_PER_STREAM_OPT(hwaccels, str, hwaccel, ic, st);
if (hwaccel) {
if (!strcmp(hwaccel, "none"))
ist->hwaccel_id = HWACCEL_NONE;
else if (!strcmp(hwaccel, "auto"))
ist->hwaccel_id = HWACCEL_AUTO;
else {
int i;
for (i = 0; hwaccels[i].name; i++) {
if (!strcmp(hwaccels[i].name, hwaccel)) {
ist->hwaccel_id = hwaccels[i].id;
break;
}
}
if (!ist->hwaccel_id) {
av_log(NULL, AV_LOG_FATAL, "Unrecognized hwaccel: %s.\n",
hwaccel);
av_log(NULL, AV_LOG_FATAL, "Supported hwaccels: ");
for (i = 0; hwaccels[i].name; i++)
av_log(NULL, AV_LOG_FATAL, "%s ", hwaccels[i].name);
av_log(NULL, AV_LOG_FATAL, "\n");
exit_program(1);
}
}
}
MATCH_PER_STREAM_OPT(hwaccel_devices, str, hwaccel_device, ic, st);
if (hwaccel_device) {
ist->hwaccel_device = av_strdup(hwaccel_device);
if (!ist->hwaccel_device)
exit_program(1);
}
ist->hwaccel_pix_fmt = AV_PIX_FMT_NONE;
break;
case AVMEDIA_TYPE_AUDIO:
ist->guess_layout_max = INT_MAX;
MATCH_PER_STREAM_OPT(guess_layout_max, i, ist->guess_layout_max, ic, st);
guess_input_channel_layout(ist);
ist->resample_sample_fmt = ist->dec_ctx->sample_fmt;
ist->resample_sample_rate = ist->dec_ctx->sample_rate;
ist->resample_channels = ist->dec_ctx->channels;
ist->resample_channel_layout = ist->dec_ctx->channel_layout;
ist->resample_sample_fmt = dec->sample_fmt;
ist->resample_sample_rate = dec->sample_rate;
ist->resample_channels = dec->channels;
ist->resample_channel_layout = dec->channel_layout;
break;
case AVMEDIA_TYPE_DATA:
@@ -724,9 +654,10 @@ static void assert_file_overwrite(const char *filename)
exit_program(1);
}
if (!file_overwrite) {
const char *proto_name = avio_find_protocol_name(filename);
if (proto_name && !strcmp(proto_name, "file") && avio_check(filename, 0) == 0) {
if (!file_overwrite &&
(strchr(filename, ':') == NULL || filename[1] == ':' ||
av_strstart(filename, "file:", NULL))) {
if (avio_check(filename, 0) == 0) {
if (stdin_interaction && !no_file_overwrite) {
fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename);
fflush(stderr);
@@ -887,10 +818,8 @@ static int open_input_file(OptionsContext *o, const char *filename)
ret = avformat_find_stream_info(ic, opts);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "%s: could not find codec parameters\n", filename);
if (ic->nb_streams == 0) {
avformat_close_input(&ic);
exit_program(1);
}
avformat_close_input(&ic);
exit_program(1);
}
timestamp = (o->start_time == AV_NOPTS_VALUE) ? 0 : o->start_time;
@@ -933,7 +862,7 @@ static int open_input_file(OptionsContext *o, const char *filename)
unused_opts = strip_specifiers(o->g->codec_opts);
for (i = f->ist_index; i < nb_input_streams; i++) {
e = NULL;
while ((e = av_dict_get(input_streams[i]->decoder_opts, "", e,
while ((e = av_dict_get(input_streams[i]->opts, "", e,
AV_DICT_IGNORE_SUFFIX)))
av_dict_set(&unused_opts, e->key, NULL, 0);
}
@@ -943,13 +872,8 @@ static int open_input_file(OptionsContext *o, const char *filename)
const AVClass *class = avcodec_get_class();
const AVOption *option = av_opt_find(&class, e->key, NULL, 0,
AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
const AVClass *fclass = avformat_get_class();
const AVOption *foption = av_opt_find(&fclass, e->key, NULL, 0,
AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
if (!option || foption)
if (!option)
continue;
if (!(option->flags & AV_OPT_FLAG_DECODING_PARAM)) {
av_log(NULL, AV_LOG_ERROR, "Codec AVOption %s (%s) specified for "
"input file #%d (%s) is not a decoding option.\n", e->key,
@@ -1006,14 +930,14 @@ static uint8_t *get_line(AVIOContext *s)
static int get_preset_file_2(const char *preset_name, const char *codec_name, AVIOContext **s)
{
int i, ret = -1;
int i, ret = 1;
char filename[1000];
const char *base[3] = { getenv("AVCONV_DATADIR"),
getenv("HOME"),
AVCONV_DATADIR,
};
for (i = 0; i < FF_ARRAY_ELEMS(base) && ret < 0; i++) {
for (i = 0; i < FF_ARRAY_ELEMS(base) && ret; i++) {
if (!base[i])
continue;
if (codec_name) {
@@ -1021,7 +945,7 @@ static int get_preset_file_2(const char *preset_name, const char *codec_name, AV
i != 1 ? "" : "/.avconv", codec_name, preset_name);
ret = avio_open2(s, filename, AVIO_FLAG_READ, &int_cb, NULL);
}
if (ret < 0) {
if (ret) {
snprintf(filename, sizeof(filename), "%s%s/%s.avpreset", base[i],
i != 1 ? "" : "/.avconv", preset_name);
ret = avio_open2(s, filename, AVIO_FLAG_READ, &int_cb, NULL);
@@ -1075,19 +999,11 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
ost->st = st;
st->codec->codec_type = type;
choose_encoder(o, oc, ost);
ost->enc_ctx = avcodec_alloc_context3(ost->enc);
if (!ost->enc_ctx) {
av_log(NULL, AV_LOG_ERROR, "Error allocating the encoding context.\n");
exit_program(1);
}
ost->enc_ctx->codec_type = type;
if (ost->enc) {
AVIOContext *s = NULL;
char *buf = NULL, *arg = NULL, *preset = NULL;
ost->encoder_opts = filter_codec_opts(o->g->codec_opts, ost->enc->id, oc, st, ost->enc);
ost->opts = filter_codec_opts(o->g->codec_opts, ost->enc->id, oc, st, ost->enc);
MATCH_PER_STREAM_OPT(presets, str, preset, oc, st);
if (preset && (!(ret = get_preset_file_2(preset, ost->enc->name, &s)))) {
@@ -1102,7 +1018,7 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
exit_program(1);
}
*arg++ = 0;
av_dict_set(&ost->encoder_opts, buf, arg, AV_DICT_DONT_OVERWRITE);
av_dict_set(&ost->opts, buf, arg, AV_DICT_DONT_OVERWRITE);
av_free(buf);
} while (!s->eof_reached);
avio_close(s);
@@ -1114,9 +1030,12 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
exit_program(1);
}
} else {
ost->encoder_opts = filter_codec_opts(o->g->codec_opts, AV_CODEC_ID_NONE, oc, st, NULL);
ost->opts = filter_codec_opts(o->g->codec_opts, AV_CODEC_ID_NONE, oc, st, NULL);
}
avcodec_get_context_defaults3(st->codec, ost->enc);
st->codec->codec_type = type; // XXX hack, avcodec_get_context_defaults2() sets type to unknown for stream copy
ost->max_frames = INT64_MAX;
MATCH_PER_STREAM_OPT(max_frames, i64, ost->max_frames, oc, st);
for (i = 0; i<o->nb_max_frames; i++) {
@@ -1152,17 +1071,17 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
uint32_t tag = strtol(codec_tag, &next, 0);
if (*next)
tag = AV_RL32(codec_tag);
ost->enc_ctx->codec_tag = tag;
st->codec->codec_tag = tag;
}
MATCH_PER_STREAM_OPT(qscale, dbl, qscale, oc, st);
if (qscale >= 0) {
ost->enc_ctx->flags |= CODEC_FLAG_QSCALE;
ost->enc_ctx->global_quality = FF_QP2LAMBDA * qscale;
st->codec->flags |= CODEC_FLAG_QSCALE;
st->codec->global_quality = FF_QP2LAMBDA * qscale;
}
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
ost->enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
st->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
av_opt_get_int(o->g->sws_opts, "sws_flags", 0, &ost->sws_flags);
@@ -1176,7 +1095,7 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
if (source_index >= 0) {
ost->sync_ist = input_streams[source_index];
input_streams[source_index]->discard = 0;
input_streams[source_index]->st->discard = input_streams[source_index]->user_set_discard;
input_streams[source_index]->st->discard = AVDISCARD_NONE;
}
ost->last_mux_dts = AV_NOPTS_VALUE;
@@ -1233,36 +1152,26 @@ static char *get_ost_filters(OptionsContext *o, AVFormatContext *oc,
OutputStream *ost)
{
AVStream *st = ost->st;
char *filter = NULL, *filter_script = NULL;
if (ost->filters_script && ost->filters) {
MATCH_PER_STREAM_OPT(filter_scripts, str, filter_script, oc, st);
MATCH_PER_STREAM_OPT(filters, str, filter, oc, st);
if (filter_script && filter) {
av_log(NULL, AV_LOG_ERROR, "Both -filter and -filter_script set for "
"output stream #%d:%d.\n", nb_output_files, st->index);
exit_program(1);
}
if (ost->filters_script)
return read_file(ost->filters_script);
else if (ost->filters)
return av_strdup(ost->filters);
if (filter_script)
return read_file(filter_script);
else if (filter)
return av_strdup(filter);
return av_strdup(st->codec->codec_type == AVMEDIA_TYPE_VIDEO ?
"null" : "anull");
}
static void check_streamcopy_filters(OptionsContext *o, AVFormatContext *oc,
const OutputStream *ost, enum AVMediaType type)
{
if (ost->filters_script || ost->filters) {
av_log(NULL, AV_LOG_ERROR,
"%s '%s' was defined for %s output stream %d:%d but codec copy was selected.\n"
"Filtering and streamcopy cannot be used together.\n",
ost->filters ? "Filtergraph" : "Filtergraph script",
ost->filters ? ost->filters : ost->filters_script,
av_get_media_type_string(type), ost->file_index, ost->index);
exit_program(1);
}
}
static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
{
AVStream *st;
@@ -1272,7 +1181,7 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
ost = new_output_stream(o, oc, AVMEDIA_TYPE_VIDEO, source_index);
st = ost->st;
video_enc = ost->enc_ctx;
video_enc = st->codec;
MATCH_PER_STREAM_OPT(frame_rates, str, frame_rate, oc, st);
if (frame_rate && av_parse_video_rate(&ost->frame_rate, frame_rate) < 0) {
@@ -1291,15 +1200,11 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
ost->frame_aspect_ratio = q;
}
MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st);
MATCH_PER_STREAM_OPT(filters, str, ost->filters, oc, st);
if (!ost->stream_copy) {
const char *p = NULL;
char *frame_size = NULL;
char *frame_pix_fmt = NULL;
char *intra_matrix = NULL, *inter_matrix = NULL;
char *chroma_intra_matrix = NULL;
int do_pass = 0;
int i;
@@ -1332,16 +1237,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
}
parse_matrix_coeffs(video_enc->intra_matrix, intra_matrix);
}
MATCH_PER_STREAM_OPT(chroma_intra_matrices, str, chroma_intra_matrix, oc, st);
if (chroma_intra_matrix) {
uint16_t *p = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64);
if (!p) {
av_log(NULL, AV_LOG_FATAL, "Could not allocate memory for intra matrix.\n");
exit_program(1);
}
av_codec_set_chroma_intra_matrix(video_enc, p);
parse_matrix_coeffs(p, chroma_intra_matrix);
}
MATCH_PER_STREAM_OPT(inter_matrices, str, inter_matrix, oc, st);
if (inter_matrix) {
if (!(video_enc->inter_matrix = av_mallocz(sizeof(*video_enc->inter_matrix) * 64))) {
@@ -1377,6 +1272,7 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
if (p) p++;
}
video_enc->rc_override_count = i;
video_enc->intra_dc_precision = intra_dc_precision - 8;
if (do_psnr)
video_enc->flags|= CODEC_FLAG_PSNR;
@@ -1386,11 +1282,11 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
if (do_pass) {
if (do_pass & 1) {
video_enc->flags |= CODEC_FLAG_PASS1;
av_dict_set(&ost->encoder_opts, "flags", "+pass1", AV_DICT_APPEND);
av_dict_set(&ost->opts, "flags", "+pass1", AV_DICT_APPEND);
}
if (do_pass & 2) {
video_enc->flags |= CODEC_FLAG_PASS2;
av_dict_set(&ost->encoder_opts, "flags", "+pass2", AV_DICT_APPEND);
av_dict_set(&ost->opts, "flags", "+pass2", AV_DICT_APPEND);
}
}
@@ -1416,9 +1312,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
MATCH_PER_STREAM_OPT(copy_initial_nonkeyframes, i, ost->copy_initial_nonkeyframes, oc ,st);
}
if (ost->stream_copy)
check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_VIDEO);
return ost;
}
@@ -1432,12 +1325,9 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in
ost = new_output_stream(o, oc, AVMEDIA_TYPE_AUDIO, source_index);
st = ost->st;
audio_enc = ost->enc_ctx;
audio_enc = st->codec;
audio_enc->codec_type = AVMEDIA_TYPE_AUDIO;
MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st);
MATCH_PER_STREAM_OPT(filters, str, ost->filters, oc, st);
if (!ost->stream_copy) {
char *sample_fmt = NULL;
@@ -1462,34 +1352,19 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in
/* check for channel mapping for this audio stream */
for (n = 0; n < o->nb_audio_channel_maps; n++) {
AudioChannelMap *map = &o->audio_channel_maps[n];
if ((map->ofile_idx == -1 || ost->file_index == map->ofile_idx) &&
InputStream *ist = input_streams[ost->source_index];
if ((map->channel_idx == -1 || (ist->file_index == map->file_idx && ist->st->index == map->stream_idx)) &&
(map->ofile_idx == -1 || ost->file_index == map->ofile_idx) &&
(map->ostream_idx == -1 || ost->st->index == map->ostream_idx)) {
InputStream *ist;
if (map->channel_idx == -1) {
ist = NULL;
} else if (ost->source_index < 0) {
av_log(NULL, AV_LOG_FATAL, "Cannot determine input stream for channel mapping %d.%d\n",
if (ost->audio_channels_mapped < FF_ARRAY_ELEMS(ost->audio_channels_map))
ost->audio_channels_map[ost->audio_channels_mapped++] = map->channel_idx;
else
av_log(NULL, AV_LOG_FATAL, "Max channel mapping for output %d.%d reached\n",
ost->file_index, ost->st->index);
continue;
} else {
ist = input_streams[ost->source_index];
}
if (!ist || (ist->file_index == map->file_idx && ist->st->index == map->stream_idx)) {
if (ost->audio_channels_mapped < FF_ARRAY_ELEMS(ost->audio_channels_map))
ost->audio_channels_map[ost->audio_channels_mapped++] = map->channel_idx;
else
av_log(NULL, AV_LOG_FATAL, "Max channel mapping for output %d.%d reached\n",
ost->file_index, ost->st->index);
}
}
}
}
if (ost->stream_copy)
check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_AUDIO);
return ost;
}
@@ -1522,7 +1397,7 @@ static OutputStream *new_subtitle_stream(OptionsContext *o, AVFormatContext *oc,
ost = new_output_stream(o, oc, AVMEDIA_TYPE_SUBTITLE, source_index);
st = ost->st;
subtitle_enc = ost->enc_ctx;
subtitle_enc = st->codec;
subtitle_enc->codec_type = AVMEDIA_TYPE_SUBTITLE;
@@ -1640,8 +1515,7 @@ static int read_ffserver_streams(OptionsContext *o, AVFormatContext *s, const ch
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && !ost->stream_copy)
choose_sample_fmt(st, codec);
else if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO && !ost->stream_copy)
choose_pixel_fmt(st, st->codec, codec, st->codec->pix_fmt);
avcodec_copy_context(ost->enc_ctx, st->codec);
choose_pixel_fmt(st, codec, st->codec->pix_fmt);
}
avformat_close_input(&ic);
@@ -1675,18 +1549,6 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
exit_program(1);
}
if (ost->avfilter && (ost->filters || ost->filters_script)) {
const char *opt = ost->filters ? "-vf/-af/-filter" : "-filter_script";
av_log(NULL, AV_LOG_ERROR,
"%s '%s' was specified through the %s option "
"for output stream %d:%d, which is fed from a complex filtergraph.\n"
"%s and -filter_complex cannot be used together for the same stream.\n",
ost->filters ? "Filtergraph" : "Filtergraph script",
ost->filters ? ost->filters : ost->filters_script,
opt, ost->file_index, ost->index, opt);
exit_program(1);
}
if (configure_output_filter(ofilter->graph, ofilter, ofilter->out_tmp) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error configuring filter.\n");
exit_program(1);
@@ -1814,7 +1676,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
if(ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) ost->avfilter = av_strdup("anull");
if(ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) ost->avfilter = av_strdup("null");
ist->discard = 0;
ist->st->discard = ist->user_set_discard;
ist->st->discard = AVDISCARD_NONE;
break;
}
}
@@ -1959,8 +1821,8 @@ loop_end:
ost->stream_copy = 0;
ost->attachment_filename = o->attachments[i];
ost->finished = 1;
ost->enc_ctx->extradata = attachment;
ost->enc_ctx->extradata_size = len;
ost->st->codec->extradata = attachment;
ost->st->codec->extradata_size = len;
p = strrchr(o->attachments[i], '/');
av_dict_set(&ost->st->metadata, "filename", (p && *p) ? p + 1 : o->attachments[i], AV_DICT_DONT_OVERWRITE);
@@ -1982,7 +1844,7 @@ loop_end:
unused_opts = strip_specifiers(o->g->codec_opts);
for (i = of->ost_index; i < nb_output_streams; i++) {
e = NULL;
while ((e = av_dict_get(output_streams[i]->encoder_opts, "", e,
while ((e = av_dict_get(output_streams[i]->opts, "", e,
AV_DICT_IGNORE_SUFFIX)))
av_dict_set(&unused_opts, e->key, NULL, 0);
}
@@ -2093,8 +1955,6 @@ loop_end:
continue;
ist = input_streams[output_streams[i]->source_index];
av_dict_copy(&output_streams[i]->st->metadata, ist->st->metadata, AV_DICT_DONT_OVERWRITE);
if (!output_streams[i]->stream_copy)
av_dict_set(&output_streams[i]->st->metadata, "encoder", NULL, 0);
}
/* process manually set metadata */
@@ -2196,22 +2056,23 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
opt_video_codec(o, "c:v", "mpeg1video");
opt_audio_codec(o, "c:a", "mp2");
parse_option(o, "f", "vcd", options);
av_dict_set(&o->g->codec_opts, "b:v", arg, 0);
parse_option(o, "s", norm == PAL ? "352x288" : "352x240", options);
parse_option(o, "r", frame_rates[norm], options);
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
av_dict_set(&o->g->codec_opts, "b:v", "1150000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "maxrate", "1150000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "minrate", "1150000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "bufsize", "327680", AV_DICT_DONT_OVERWRITE); // 40*1024*8;
av_dict_set(&o->g->codec_opts, "b:v", "1150000", 0);
av_dict_set(&o->g->codec_opts, "maxrate", "1150000", 0);
av_dict_set(&o->g->codec_opts, "minrate", "1150000", 0);
av_dict_set(&o->g->codec_opts, "bufsize", "327680", 0); // 40*1024*8;
av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "b:a", "224000", 0);
parse_option(o, "ar", "44100", options);
parse_option(o, "ac", "2", options);
av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->format_opts, "muxrate", "1411200", AV_DICT_DONT_OVERWRITE); // 2352 * 75 * 8;
av_dict_set(&o->g->format_opts, "packetsize", "2324", 0);
av_dict_set(&o->g->format_opts, "muxrate", "1411200", 0); // 2352 * 75 * 8;
/* We have to offset the PTS, so that it is consistent with the SCR.
SCR starts at 36000, but the first two packs contain only padding
@@ -2228,18 +2089,18 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
parse_option(o, "s", norm == PAL ? "480x576" : "480x480", options);
parse_option(o, "r", frame_rates[norm], options);
parse_option(o, "pix_fmt", "yuv420p", options);
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
av_dict_set(&o->g->codec_opts, "b:v", "2040000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "maxrate", "2516000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1145000;
av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8;
av_dict_set(&o->g->codec_opts, "scan_offset", "1", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "b:v", "2040000", 0);
av_dict_set(&o->g->codec_opts, "maxrate", "2516000", 0);
av_dict_set(&o->g->codec_opts, "minrate", "0", 0); // 1145000;
av_dict_set(&o->g->codec_opts, "bufsize", "1835008", 0); // 224*1024*8;
av_dict_set(&o->g->codec_opts, "scan_offset", "1", 0);
av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "b:a", "224000", 0);
parse_option(o, "ar", "44100", options);
av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->format_opts, "packetsize", "2324", 0);
} else if (!strcmp(arg, "dvd")) {
@@ -2250,17 +2111,17 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
parse_option(o, "s", norm == PAL ? "720x576" : "720x480", options);
parse_option(o, "r", frame_rates[norm], options);
parse_option(o, "pix_fmt", "yuv420p", options);
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
av_dict_set(&o->g->codec_opts, "b:v", "6000000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "maxrate", "9000000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1500000;
av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8;
av_dict_set(&o->g->codec_opts, "b:v", "6000000", 0);
av_dict_set(&o->g->codec_opts, "maxrate", "9000000", 0);
av_dict_set(&o->g->codec_opts, "minrate", "0", 0); // 1500000;
av_dict_set(&o->g->codec_opts, "bufsize", "1835008", 0); // 224*1024*8;
av_dict_set(&o->g->format_opts, "packetsize", "2048", AV_DICT_DONT_OVERWRITE); // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
av_dict_set(&o->g->format_opts, "muxrate", "10080000", AV_DICT_DONT_OVERWRITE); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
av_dict_set(&o->g->format_opts, "packetsize", "2048", 0); // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
av_dict_set(&o->g->format_opts, "muxrate", "10080000", 0); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
av_dict_set(&o->g->codec_opts, "b:a", "448000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "b:a", "448000", 0);
parse_option(o, "ar", "48000", options);
} else if (!strncmp(arg, "dv", 2)) {
@@ -2398,11 +2259,7 @@ static int opt_old2new(void *optctx, const char *opt, const char *arg)
static int opt_bitrate(void *optctx, const char *opt, const char *arg)
{
OptionsContext *o = optctx;
if(!strcmp(opt, "ab")){
av_dict_set(&o->g->codec_opts, "b:a", arg, 0);
return 0;
} else if(!strcmp(opt, "b")){
if(!strcmp(opt, "b")){
av_log(NULL, AV_LOG_WARNING, "Please use -b:a or -b:v, -b is ambiguous\n");
av_dict_set(&o->g->codec_opts, "b:v", arg, 0);
return 0;
@@ -2832,7 +2689,7 @@ const OptionDef options[] = {
{ "frames", OPT_INT64 | HAS_ARG | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(max_frames) },
"set the number of frames to record", "number" },
{ "tag", OPT_STRING | HAS_ARG | OPT_SPEC |
OPT_EXPERT | OPT_OUTPUT | OPT_INPUT, { .off = OFFSET(codec_tags) },
OPT_EXPERT | OPT_OUTPUT, { .off = OFFSET(codec_tags) },
"force codec tag/fourcc", "fourcc/tag" },
{ "q", HAS_ARG | OPT_EXPERT | OPT_DOUBLE |
OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(qscale) },
@@ -2866,9 +2723,6 @@ const OptionDef options[] = {
"print timestamp debugging info" },
{ "max_error_rate", HAS_ARG | OPT_FLOAT, { &max_error_rate },
"maximum error rate", "ratio of errors (0.0: no errors, 1.0: 100% errors) above which ffmpeg returns an error instead of success." },
{ "discard", OPT_STRING | HAS_ARG | OPT_SPEC |
OPT_INPUT, { .off = OFFSET(discard) },
"discard", "" },
/* video options */
{ "vframes", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_video_frames },
@@ -2891,6 +2745,8 @@ const OptionDef options[] = {
"deprecated use -g 1" },
{ "vn", OPT_VIDEO | OPT_BOOL | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT,{ .off = OFFSET(video_disable) },
"disable video" },
{ "vdt", OPT_VIDEO | OPT_INT | HAS_ARG | OPT_EXPERT , { &video_discard },
"discard threshold", "n" },
{ "rc_override", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_STRING | OPT_SPEC |
OPT_OUTPUT, { .off = OFFSET(rc_overrides) },
"rate control override for specific intervals", "override" },
@@ -2924,12 +2780,11 @@ const OptionDef options[] = {
{ "inter_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_STRING | OPT_SPEC |
OPT_OUTPUT, { .off = OFFSET(inter_matrices) },
"specify inter matrix coeffs", "matrix" },
{ "chroma_intra_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_STRING | OPT_SPEC |
OPT_OUTPUT, { .off = OFFSET(chroma_intra_matrices) },
"specify intra matrix coeffs", "matrix" },
{ "top", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_INT| OPT_SPEC |
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(top_field_first) },
"top=1/bottom=0/auto=-1 field first", "" },
{ "dc", OPT_VIDEO | OPT_INT | HAS_ARG | OPT_EXPERT , { &intra_dc_precision },
"intra_dc_precision", "precision" },
{ "vtag", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_PERFILE |
OPT_OUTPUT, { .func_arg = opt_old2new },
"force video tag/fourcc", "fourcc/tag" },
@@ -2944,16 +2799,8 @@ const OptionDef options[] = {
{ "force_key_frames", OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(forced_key_frames) },
"force key frames at specified timestamps", "timestamps" },
{ "ab", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_bitrate },
"audio bitrate (please use -b:a)", "bitrate" },
{ "b", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_bitrate },
"video bitrate (please use -b:v)", "bitrate" },
{ "hwaccel", OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
OPT_SPEC | OPT_INPUT, { .off = OFFSET(hwaccels) },
"use HW accelerated decoding", "hwaccel name" },
{ "hwaccel_device", OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
OPT_SPEC | OPT_INPUT, { .off = OFFSET(hwaccel_devices) },
"select a device for HW acceleration" "devicename" },
/* audio options */
{ "aframes", OPT_AUDIO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_audio_frames },

View File

@@ -1,134 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavcodec/avcodec.h"
#include "libavcodec/vda.h"
#include "libavutil/imgutils.h"
#include "ffmpeg.h"
typedef struct VDAContext {
AVFrame *tmp_frame;
} VDAContext;
static int vda_retrieve_data(AVCodecContext *s, AVFrame *frame)
{
InputStream *ist = s->opaque;
VDAContext *vda = ist->hwaccel_ctx;
CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
CVReturn err;
uint8_t *data[4] = { 0 };
int linesize[4] = { 0 };
int planes, ret, i;
av_frame_unref(vda->tmp_frame);
switch (pixel_format) {
case kCVPixelFormatType_420YpCbCr8Planar: vda->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
case kCVPixelFormatType_422YpCbCr8: vda->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
default:
av_log(NULL, AV_LOG_ERROR,
"Unsupported pixel format: %u\n", pixel_format);
return AVERROR(ENOSYS);
}
vda->tmp_frame->width = frame->width;
vda->tmp_frame->height = frame->height;
ret = av_frame_get_buffer(vda->tmp_frame, 32);
if (ret < 0)
return ret;
err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
if (err != kCVReturnSuccess) {
av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
return AVERROR_UNKNOWN;
}
if (CVPixelBufferIsPlanar(pixbuf)) {
planes = CVPixelBufferGetPlaneCount(pixbuf);
for (i = 0; i < planes; i++) {
data[i] = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
}
} else {
data[0] = CVPixelBufferGetBaseAddress(pixbuf);
linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
}
av_image_copy(vda->tmp_frame->data, vda->tmp_frame->linesize,
data, linesize, vda->tmp_frame->format,
frame->width, frame->height);
ret = av_frame_copy_props(vda->tmp_frame, frame);
if (ret < 0)
return ret;
av_frame_unref(frame);
av_frame_move_ref(frame, vda->tmp_frame);
return 0;
}
static void vda_uninit(AVCodecContext *s)
{
InputStream *ist = s->opaque;
VDAContext *vda = ist->hwaccel_ctx;
ist->hwaccel_uninit = NULL;
ist->hwaccel_retrieve_data = NULL;
av_frame_free(&vda->tmp_frame);
av_vda_default_free(s);
av_freep(&ist->hwaccel_ctx);
}
int vda_init(AVCodecContext *s)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
VDAContext *vda;
int ret;
vda = av_mallocz(sizeof(*vda));
if (!vda)
return AVERROR(ENOMEM);
ist->hwaccel_ctx = vda;
ist->hwaccel_uninit = vda_uninit;
ist->hwaccel_retrieve_data = vda_retrieve_data;
vda->tmp_frame = av_frame_alloc();
if (!vda->tmp_frame) {
ret = AVERROR(ENOMEM);
goto fail;
}
ret = av_vda_default_init(s);
if (ret < 0) {
av_log(NULL, loglevel, "Error creating VDA decoder.\n");
goto fail;
}
return 0;
fail:
vda_uninit(s);
return ret;
}

View File

@@ -1,335 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include <vdpau/vdpau.h>
#include <vdpau/vdpau_x11.h>
#include <X11/Xlib.h>
#include "ffmpeg.h"
#include "libavcodec/vdpau.h"
#include "libavutil/avassert.h"
#include "libavutil/buffer.h"
#include "libavutil/frame.h"
#include "libavutil/pixfmt.h"
typedef struct VDPAUContext {
Display *dpy;
VdpDevice device;
VdpDecoder decoder;
VdpGetProcAddress *get_proc_address;
VdpGetErrorString *get_error_string;
VdpGetInformationString *get_information_string;
VdpDeviceDestroy *device_destroy;
VdpDecoderCreate *decoder_create;
VdpDecoderDestroy *decoder_destroy;
VdpDecoderRender *decoder_render;
VdpVideoSurfaceCreate *video_surface_create;
VdpVideoSurfaceDestroy *video_surface_destroy;
VdpVideoSurfaceGetBitsYCbCr *video_surface_get_bits;
VdpVideoSurfaceGetParameters *video_surface_get_parameters;
VdpVideoSurfaceQueryGetPutBitsYCbCrCapabilities *video_surface_query;
AVFrame *tmp_frame;
enum AVPixelFormat pix_fmt;
VdpYCbCrFormat vdpau_format;
} VDPAUContext;
static void vdpau_uninit(AVCodecContext *s)
{
InputStream *ist = s->opaque;
VDPAUContext *ctx = ist->hwaccel_ctx;
ist->hwaccel_uninit = NULL;
ist->hwaccel_get_buffer = NULL;
ist->hwaccel_retrieve_data = NULL;
if (ctx->decoder_destroy)
ctx->decoder_destroy(ctx->decoder);
if (ctx->device_destroy)
ctx->device_destroy(ctx->device);
if (ctx->dpy)
XCloseDisplay(ctx->dpy);
av_frame_free(&ctx->tmp_frame);
av_freep(&ist->hwaccel_ctx);
av_freep(&s->hwaccel_context);
}
static void vdpau_release_buffer(void *opaque, uint8_t *data)
{
VdpVideoSurface surface = *(VdpVideoSurface*)data;
VDPAUContext *ctx = opaque;
ctx->video_surface_destroy(surface);
av_freep(&data);
}
static int vdpau_get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
{
InputStream *ist = s->opaque;
VDPAUContext *ctx = ist->hwaccel_ctx;
VdpVideoSurface *surface;
VdpStatus err;
av_assert0(frame->format == AV_PIX_FMT_VDPAU);
surface = av_malloc(sizeof(*surface));
if (!surface)
return AVERROR(ENOMEM);
frame->buf[0] = av_buffer_create((uint8_t*)surface, sizeof(*surface),
vdpau_release_buffer, ctx,
AV_BUFFER_FLAG_READONLY);
if (!frame->buf[0]) {
av_freep(&surface);
return AVERROR(ENOMEM);
}
// properly we should keep a pool of surfaces instead of creating
// them anew for each frame, but since we don't care about speed
// much in this code, we don't bother
err = ctx->video_surface_create(ctx->device, VDP_CHROMA_TYPE_420,
frame->width, frame->height, surface);
if (err != VDP_STATUS_OK) {
av_log(NULL, AV_LOG_ERROR, "Error allocating a VDPAU video surface: %s\n",
ctx->get_error_string(err));
av_buffer_unref(&frame->buf[0]);
return AVERROR_UNKNOWN;
}
frame->data[3] = (uint8_t*)(uintptr_t)*surface;
return 0;
}
static int vdpau_retrieve_data(AVCodecContext *s, AVFrame *frame)
{
VdpVideoSurface surface = (VdpVideoSurface)(uintptr_t)frame->data[3];
InputStream *ist = s->opaque;
VDPAUContext *ctx = ist->hwaccel_ctx;
VdpStatus err;
int ret, chroma_type;
err = ctx->video_surface_get_parameters(surface, &chroma_type,
&ctx->tmp_frame->width,
&ctx->tmp_frame->height);
if (err != VDP_STATUS_OK) {
av_log(NULL, AV_LOG_ERROR, "Error getting surface parameters: %s\n",
ctx->get_error_string(err));
return AVERROR_UNKNOWN;
}
ctx->tmp_frame->format = ctx->pix_fmt;
ret = av_frame_get_buffer(ctx->tmp_frame, 32);
if (ret < 0)
return ret;
ctx->tmp_frame->width = frame->width;
ctx->tmp_frame->height = frame->height;
err = ctx->video_surface_get_bits(surface, ctx->vdpau_format,
(void * const *)ctx->tmp_frame->data,
ctx->tmp_frame->linesize);
if (err != VDP_STATUS_OK) {
av_log(NULL, AV_LOG_ERROR, "Error retrieving frame data from VDPAU: %s\n",
ctx->get_error_string(err));
ret = AVERROR_UNKNOWN;
goto fail;
}
if (ctx->vdpau_format == VDP_YCBCR_FORMAT_YV12)
FFSWAP(uint8_t*, ctx->tmp_frame->data[1], ctx->tmp_frame->data[2]);
ret = av_frame_copy_props(ctx->tmp_frame, frame);
if (ret < 0)
goto fail;
av_frame_unref(frame);
av_frame_move_ref(frame, ctx->tmp_frame);
return 0;
fail:
av_frame_unref(ctx->tmp_frame);
return ret;
}
static const int vdpau_formats[][2] = {
{ VDP_YCBCR_FORMAT_YV12, AV_PIX_FMT_YUV420P },
{ VDP_YCBCR_FORMAT_NV12, AV_PIX_FMT_NV12 },
{ VDP_YCBCR_FORMAT_YUYV, AV_PIX_FMT_YUYV422 },
{ VDP_YCBCR_FORMAT_UYVY, AV_PIX_FMT_UYVY422 },
};
static int vdpau_alloc(AVCodecContext *s)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
AVVDPAUContext *vdpau_ctx;
VDPAUContext *ctx;
const char *display, *vendor;
VdpStatus err;
int i;
ctx = av_mallocz(sizeof(*ctx));
if (!ctx)
return AVERROR(ENOMEM);
ist->hwaccel_ctx = ctx;
ist->hwaccel_uninit = vdpau_uninit;
ist->hwaccel_get_buffer = vdpau_get_buffer;
ist->hwaccel_retrieve_data = vdpau_retrieve_data;
ctx->tmp_frame = av_frame_alloc();
if (!ctx->tmp_frame)
goto fail;
ctx->dpy = XOpenDisplay(ist->hwaccel_device);
if (!ctx->dpy) {
av_log(NULL, loglevel, "Cannot open the X11 display %s.\n",
XDisplayName(ist->hwaccel_device));
goto fail;
}
display = XDisplayString(ctx->dpy);
err = vdp_device_create_x11(ctx->dpy, XDefaultScreen(ctx->dpy), &ctx->device,
&ctx->get_proc_address);
if (err != VDP_STATUS_OK) {
av_log(NULL, loglevel, "VDPAU device creation on X11 display %s failed.\n",
display);
goto fail;
}
#define GET_CALLBACK(id, result) \
do { \
void *tmp; \
err = ctx->get_proc_address(ctx->device, id, &tmp); \
if (err != VDP_STATUS_OK) { \
av_log(NULL, loglevel, "Error getting the " #id " callback.\n"); \
goto fail; \
} \
ctx->result = tmp; \
} while (0)
GET_CALLBACK(VDP_FUNC_ID_GET_ERROR_STRING, get_error_string);
GET_CALLBACK(VDP_FUNC_ID_GET_INFORMATION_STRING, get_information_string);
GET_CALLBACK(VDP_FUNC_ID_DEVICE_DESTROY, device_destroy);
GET_CALLBACK(VDP_FUNC_ID_DECODER_CREATE, decoder_create);
GET_CALLBACK(VDP_FUNC_ID_DECODER_DESTROY, decoder_destroy);
GET_CALLBACK(VDP_FUNC_ID_DECODER_RENDER, decoder_render);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_CREATE, video_surface_create);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_DESTROY, video_surface_destroy);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_GET_BITS_Y_CB_CR, video_surface_get_bits);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_GET_PARAMETERS, video_surface_get_parameters);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_QUERY_GET_PUT_BITS_Y_CB_CR_CAPABILITIES,
video_surface_query);
for (i = 0; i < FF_ARRAY_ELEMS(vdpau_formats); i++) {
VdpBool supported;
err = ctx->video_surface_query(ctx->device, VDP_CHROMA_TYPE_420,
vdpau_formats[i][0], &supported);
if (err != VDP_STATUS_OK) {
av_log(NULL, loglevel,
"Error querying VDPAU surface capabilities: %s\n",
ctx->get_error_string(err));
goto fail;
}
if (supported)
break;
}
if (i == FF_ARRAY_ELEMS(vdpau_formats)) {
av_log(NULL, loglevel,
"No supported VDPAU format for retrieving the data.\n");
return AVERROR(EINVAL);
}
ctx->vdpau_format = vdpau_formats[i][0];
ctx->pix_fmt = vdpau_formats[i][1];
vdpau_ctx = av_vdpau_alloc_context();
if (!vdpau_ctx)
goto fail;
vdpau_ctx->render = ctx->decoder_render;
s->hwaccel_context = vdpau_ctx;
ctx->get_information_string(&vendor);
av_log(NULL, AV_LOG_VERBOSE, "Using VDPAU -- %s -- on X11 display %s, "
"to decode input stream #%d:%d.\n", vendor,
display, ist->file_index, ist->st->index);
return 0;
fail:
av_log(NULL, loglevel, "VDPAU init failed for stream #%d:%d.\n",
ist->file_index, ist->st->index);
vdpau_uninit(s);
return AVERROR(EINVAL);
}
int vdpau_init(AVCodecContext *s)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
AVVDPAUContext *vdpau_ctx;
VDPAUContext *ctx;
VdpStatus err;
int profile, ret;
if (!ist->hwaccel_ctx) {
ret = vdpau_alloc(s);
if (ret < 0)
return ret;
}
ctx = ist->hwaccel_ctx;
vdpau_ctx = s->hwaccel_context;
ret = av_vdpau_get_profile(s, &profile);
if (ret < 0) {
av_log(NULL, loglevel, "No known VDPAU decoder profile for this stream.\n");
return AVERROR(EINVAL);
}
if (ctx->decoder)
ctx->decoder_destroy(ctx->decoder);
err = ctx->decoder_create(ctx->device, profile,
s->coded_width, s->coded_height,
16, &ctx->decoder);
if (err != VDP_STATUS_OK) {
av_log(NULL, loglevel, "Error creating the VDPAU decoder: %s\n",
ctx->get_error_string(err));
return AVERROR_UNKNOWN;
}
vdpau_ctx->decoder = ctx->decoder;
ist->hwaccel_get_buffer = vdpau_get_buffer;
ist->hwaccel_retrieve_data = vdpau_retrieve_data;
return 0;
}

422
ffplay.c
View File

@@ -28,8 +28,6 @@
#include <math.h>
#include <limits.h>
#include <signal.h>
#include <stdint.h>
#include "libavutil/avstring.h"
#include "libavutil/colorspace.h"
#include "libavutil/mathematics.h"
@@ -67,13 +65,12 @@ const int program_birth_year = 2003;
#define MAX_QUEUE_SIZE (15 * 1024 * 1024)
#define MIN_FRAMES 5
/* Minimum SDL audio buffer size, in samples. */
#define SDL_AUDIO_MIN_BUFFER_SIZE 512
/* Calculate actual buffer size keeping in mind not cause too frequent audio callbacks */
#define SDL_AUDIO_MAX_CALLBACKS_PER_SEC 30
/* SDL audio buffer size, in samples. Should be small to have precise
A/V sync as SDL does not have hardware buffer fullness info. */
#define SDL_AUDIO_BUFFER_SIZE 1024
/* no AV sync correction is done if below the minimum AV sync threshold */
#define AV_SYNC_THRESHOLD_MIN 0.04
#define AV_SYNC_THRESHOLD_MIN 0.01
/* AV sync correction is done if above the maximum AV sync threshold */
#define AV_SYNC_THRESHOLD_MAX 0.1
/* If a frame duration is longer than this, it will not be duplicated to compensate AV sync */
@@ -120,11 +117,10 @@ typedef struct PacketQueue {
} PacketQueue;
#define VIDEO_PICTURE_QUEUE_SIZE 3
#define SUBPICTURE_QUEUE_SIZE 16
#define SUBPICTURE_QUEUE_SIZE 4
typedef struct VideoPicture {
double pts; // presentation timestamp for this picture
double duration; // estimated duration based on frame rate
int64_t pos; // byte position in file
SDL_Overlay *bmp;
int width, height; /* source height & width */
@@ -146,8 +142,6 @@ typedef struct AudioParams {
int channels;
int64_t channel_layout;
enum AVSampleFormat fmt;
int frame_size;
int bytes_per_sec;
} AudioParams;
typedef struct Clock {
@@ -203,7 +197,7 @@ typedef struct VideoState {
AVStream *audio_st;
PacketQueue audioq;
int audio_hw_buf_size;
uint8_t silence_buf[SDL_AUDIO_MIN_BUFFER_SIZE];
uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE];
uint8_t *audio_buf;
uint8_t *audio_buf1;
unsigned int audio_buf_size; /* in bytes */
@@ -248,15 +242,20 @@ typedef struct VideoState {
SDL_cond *subpq_cond;
double frame_timer;
double frame_last_pts;
double frame_last_duration;
double frame_last_dropped_pts;
double frame_last_returned_time;
double frame_last_filter_delay;
int64_t frame_last_dropped_pos;
int frame_last_dropped_serial;
int video_stream;
AVStream *video_st;
PacketQueue videoq;
int64_t video_current_pos; // current displayed file pos
double max_frame_duration; // maximum duration of a frame - above this, we consider the jump a timestamp discontinuity
VideoPicture pictq[VIDEO_PICTURE_QUEUE_SIZE];
int pictq_size, pictq_rindex, pictq_windex, pictq_rindex_shown;
int pictq_size, pictq_rindex, pictq_windex;
SDL_mutex *pictq_mutex;
SDL_cond *pictq_cond;
#if !CONFIG_AVFILTER
@@ -269,7 +268,6 @@ typedef struct VideoState {
int step;
#if CONFIG_AVFILTER
int vfilter_idx;
AVFilterContext *in_video_filter; // the first filter in the video chain
AVFilterContext *out_video_filter; // the last filter in the video chain
AVFilterContext *in_audio_filter; // the first filter in the audio chain
@@ -310,6 +308,7 @@ static int workaround_bugs = 1;
static int fast = 0;
static int genpts = 0;
static int lowres = 0;
static int error_concealment = 3;
static int decoder_reorder_pts = -1;
static int autoexit;
static int exit_on_keydown;
@@ -325,11 +324,9 @@ double rdftspeed = 0.02;
static int64_t cursor_last_shown;
static int cursor_hidden = 0;
#if CONFIG_AVFILTER
static const char **vfilters_list = NULL;
static int nb_vfilters = 0;
static char *vfilters = NULL;
static char *afilters = NULL;
#endif
static int autorotate = 1;
/* current context */
static int is_full_screen;
@@ -342,15 +339,6 @@ static AVPacket flush_pkt;
static SDL_Surface *screen;
#if CONFIG_AVFILTER
static int opt_add_vfilter(void *optctx, const char *opt, const char *arg)
{
GROW_ARRAY(vfilters_list, nb_vfilters);
vfilters_list[nb_vfilters - 1] = arg;
return 0;
}
#endif
static inline
int cmp_audio_fmts(enum AVSampleFormat fmt1, int64_t channel_count1,
enum AVSampleFormat fmt2, int64_t channel_count2)
@@ -371,6 +359,8 @@ int64_t get_valid_channel_layout(int64_t channel_layout, int channels)
return 0;
}
static int packet_queue_put(PacketQueue *q, AVPacket *pkt);
static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
{
MyAVPacketList *pkt1;
@@ -808,21 +798,19 @@ static void free_subpicture(SubPicture *sp)
avsubtitle_free(&sp->sub);
}
static void calculate_display_rect(SDL_Rect *rect,
int scr_xleft, int scr_ytop, int scr_width, int scr_height,
int pic_width, int pic_height, AVRational pic_sar)
static void calculate_display_rect(SDL_Rect *rect, int scr_xleft, int scr_ytop, int scr_width, int scr_height, VideoPicture *vp)
{
float aspect_ratio;
int width, height, x, y;
if (pic_sar.num == 0)
if (vp->sar.num == 0)
aspect_ratio = 0;
else
aspect_ratio = av_q2d(pic_sar);
aspect_ratio = av_q2d(vp->sar);
if (aspect_ratio <= 0.0)
aspect_ratio = 1.0;
aspect_ratio *= (float)pic_width / (float)pic_height;
aspect_ratio *= (float)vp->width / (float)vp->height;
/* XXX: we suppose the screen has a 1.0 pixel ratio */
height = scr_height;
@@ -847,7 +835,7 @@ static void video_image_display(VideoState *is)
SDL_Rect rect;
int i;
vp = &is->pictq[(is->pictq_rindex + is->pictq_rindex_shown) % VIDEO_PICTURE_QUEUE_SIZE];
vp = &is->pictq[is->pictq_rindex];
if (vp->bmp) {
if (is->subtitle_st) {
if (is->subpq_size > 0) {
@@ -873,7 +861,7 @@ static void video_image_display(VideoState *is)
}
}
calculate_display_rect(&rect, is->xleft, is->ytop, is->width, is->height, vp->width, vp->height, vp->sar);
calculate_display_rect(&rect, is->xleft, is->ytop, is->width, is->height, vp);
SDL_DisplayYUVOverlay(vp->bmp, &rect);
@@ -913,7 +901,7 @@ static void video_audio_display(VideoState *s)
/* to be more precise, we take into account the time spent since
the last buffer computation */
if (audio_callback_time) {
time_diff = av_gettime_relative() - audio_callback_time;
time_diff = av_gettime() - audio_callback_time;
delay -= (time_diff * s->audio_tgt.freq) / 1000000;
}
@@ -1065,7 +1053,7 @@ static void do_exit(VideoState *is)
av_lockmgr_register(NULL);
uninit_opts();
#if CONFIG_AVFILTER
av_freep(&vfilters_list);
av_freep(&vfilters);
#endif
avformat_network_deinit();
if (show_status)
@@ -1080,24 +1068,20 @@ static void sigterm_handler(int sig)
exit(123);
}
static void set_default_window_size(int width, int height, AVRational sar)
{
SDL_Rect rect;
calculate_display_rect(&rect, 0, 0, INT_MAX, height, width, height, sar);
default_width = rect.w;
default_height = rect.h;
}
static int video_open(VideoState *is, int force_set_video_mode, VideoPicture *vp)
{
int flags = SDL_HWSURFACE | SDL_ASYNCBLIT | SDL_HWACCEL;
int w,h;
SDL_Rect rect;
if (is_full_screen) flags |= SDL_FULLSCREEN;
else flags |= SDL_RESIZABLE;
if (vp && vp->width)
set_default_window_size(vp->width, vp->height, vp->sar);
if (vp && vp->width) {
calculate_display_rect(&rect, 0, 0, INT_MAX, vp->height, vp);
default_width = rect.w;
default_height = rect.h;
}
if (is_full_screen && fs_screen_width) {
w = fs_screen_width;
@@ -1146,7 +1130,7 @@ static double get_clock(Clock *c)
if (c->paused) {
return c->pts;
} else {
double time = av_gettime_relative() / 1000000.0;
double time = av_gettime() / 1000000.0;
return c->pts_drift + time - (time - c->last_updated) * (1.0 - c->speed);
}
}
@@ -1161,7 +1145,7 @@ static void set_clock_at(Clock *c, double pts, int serial, double time)
static void set_clock(Clock *c, double pts, int serial)
{
double time = av_gettime_relative() / 1000000.0;
double time = av_gettime() / 1000000.0;
set_clock_at(c, pts, serial, time);
}
@@ -1254,7 +1238,7 @@ static void stream_seek(VideoState *is, int64_t pos, int64_t rel, int seek_by_by
static void stream_toggle_pause(VideoState *is)
{
if (is->paused) {
is->frame_timer += av_gettime_relative() / 1000000.0 + is->vidclk.pts_drift - is->vidclk.pts;
is->frame_timer += av_gettime() / 1000000.0 + is->vidclk.pts_drift - is->vidclk.pts;
if (is->read_pause_return != AVERROR(ENOSYS)) {
is->vidclk.paused = 0;
}
@@ -1308,35 +1292,7 @@ static double compute_target_delay(double delay, VideoState *is)
return delay;
}
static double vp_duration(VideoState *is, VideoPicture *vp, VideoPicture *nextvp) {
if (vp->serial == nextvp->serial) {
double duration = nextvp->pts - vp->pts;
if (isnan(duration) || duration <= 0 || duration > is->max_frame_duration)
return vp->duration;
else
return duration;
} else {
return 0.0;
}
}
/* return the number of undisplayed pictures in the queue */
static int pictq_nb_remaining(VideoState *is) {
return is->pictq_size - is->pictq_rindex_shown;
}
/* jump back to the previous picture if available by resetting rindex_shown */
static int pictq_prev_picture(VideoState *is) {
int ret = is->pictq_rindex_shown;
is->pictq_rindex_shown = 0;
return ret;
}
static void pictq_next_picture(VideoState *is) {
if (!is->pictq_rindex_shown) {
is->pictq_rindex_shown = 1;
return;
}
/* update queue size and signal for next picture */
if (++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE)
is->pictq_rindex = 0;
@@ -1347,17 +1303,38 @@ static void pictq_next_picture(VideoState *is) {
SDL_UnlockMutex(is->pictq_mutex);
}
static int pictq_prev_picture(VideoState *is) {
VideoPicture *prevvp;
int ret = 0;
/* update queue size and signal for the previous picture */
prevvp = &is->pictq[(is->pictq_rindex + VIDEO_PICTURE_QUEUE_SIZE - 1) % VIDEO_PICTURE_QUEUE_SIZE];
if (prevvp->allocated && prevvp->serial == is->videoq.serial) {
SDL_LockMutex(is->pictq_mutex);
if (is->pictq_size < VIDEO_PICTURE_QUEUE_SIZE) {
if (--is->pictq_rindex == -1)
is->pictq_rindex = VIDEO_PICTURE_QUEUE_SIZE - 1;
is->pictq_size++;
ret = 1;
}
SDL_CondSignal(is->pictq_cond);
SDL_UnlockMutex(is->pictq_mutex);
}
return ret;
}
static void update_video_pts(VideoState *is, double pts, int64_t pos, int serial) {
/* update current video pts */
set_clock(&is->vidclk, pts, serial);
sync_clock_to_slave(&is->extclk, &is->vidclk);
is->video_current_pos = pos;
is->frame_last_pts = pts;
}
/* called to display each frame */
static void video_refresh(void *opaque, double *remaining_time)
{
VideoState *is = opaque;
VideoPicture *vp;
double time;
SubPicture *sp, *sp2;
@@ -1366,7 +1343,7 @@ static void video_refresh(void *opaque, double *remaining_time)
check_external_clock_speed(is);
if (!display_disable && is->show_mode != SHOW_MODE_VIDEO && is->audio_st) {
time = av_gettime_relative() / 1000000.0;
time = av_gettime() / 1000000.0;
if (is->force_refresh || is->last_vis_time + rdftspeed < time) {
video_display(is);
is->last_vis_time = time;
@@ -1379,37 +1356,40 @@ static void video_refresh(void *opaque, double *remaining_time)
if (is->force_refresh)
redisplay = pictq_prev_picture(is);
retry:
if (pictq_nb_remaining(is) == 0) {
if (is->pictq_size == 0) {
SDL_LockMutex(is->pictq_mutex);
if (is->frame_last_dropped_pts != AV_NOPTS_VALUE && is->frame_last_dropped_pts > is->frame_last_pts) {
update_video_pts(is, is->frame_last_dropped_pts, is->frame_last_dropped_pos, is->frame_last_dropped_serial);
is->frame_last_dropped_pts = AV_NOPTS_VALUE;
}
SDL_UnlockMutex(is->pictq_mutex);
// nothing to do, no picture to display in the queue
} else {
double last_duration, duration, delay;
VideoPicture *vp, *lastvp;
/* dequeue the picture */
lastvp = &is->pictq[is->pictq_rindex];
vp = &is->pictq[(is->pictq_rindex + is->pictq_rindex_shown) % VIDEO_PICTURE_QUEUE_SIZE];
vp = &is->pictq[is->pictq_rindex];
if (vp->serial != is->videoq.serial) {
pictq_next_picture(is);
is->video_current_pos = -1;
redisplay = 0;
goto retry;
}
if (lastvp->serial != vp->serial && !redisplay)
is->frame_timer = av_gettime_relative() / 1000000.0;
if (is->paused)
goto display;
/* compute nominal last_duration */
last_duration = vp_duration(is, lastvp, vp);
last_duration = vp->pts - is->frame_last_pts;
if (!isnan(last_duration) && last_duration > 0 && last_duration < is->max_frame_duration) {
/* if duration of the last frame was sane, update last_duration in video state */
is->frame_last_duration = last_duration;
}
if (redisplay)
delay = 0.0;
else
delay = compute_target_delay(last_duration, is);
delay = compute_target_delay(is->frame_last_duration, is);
time= av_gettime_relative()/1000000.0;
time= av_gettime()/1000000.0;
if (time < is->frame_timer + delay && !redisplay) {
*remaining_time = FFMIN(is->frame_timer + delay - time, *remaining_time);
return;
@@ -1424,9 +1404,9 @@ retry:
update_video_pts(is, vp->pts, vp->pos, vp->serial);
SDL_UnlockMutex(is->pictq_mutex);
if (pictq_nb_remaining(is) > 1) {
VideoPicture *nextvp = &is->pictq[(is->pictq_rindex + is->pictq_rindex_shown + 1) % VIDEO_PICTURE_QUEUE_SIZE];
duration = vp_duration(is, vp, nextvp);
if (is->pictq_size > 1) {
VideoPicture *nextvp = &is->pictq[(is->pictq_rindex + 1) % VIDEO_PICTURE_QUEUE_SIZE];
duration = nextvp->pts - vp->pts;
if(!is->step && (redisplay || framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) && time > is->frame_timer + duration){
if (!redisplay)
is->frame_drops_late++;
@@ -1483,7 +1463,7 @@ display:
int aqsize, vqsize, sqsize;
double av_diff;
cur_time = av_gettime_relative();
cur_time = av_gettime();
if (!last_time || (cur_time - last_time) >= 30000) {
aqsize = 0;
vqsize = 0;
@@ -1569,7 +1549,7 @@ static void duplicate_right_border_pixels(SDL_Overlay *bmp) {
}
}
static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double duration, int64_t pos, int serial)
static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t pos, int serial)
{
VideoPicture *vp;
@@ -1581,7 +1561,8 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double
/* wait until we have space to put a new picture */
SDL_LockMutex(is->pictq_mutex);
while (is->pictq_size >= VIDEO_PICTURE_QUEUE_SIZE &&
/* keep the last already displayed picture in the queue */
while (is->pictq_size >= VIDEO_PICTURE_QUEUE_SIZE - 1 &&
!is->videoq.abort_request) {
SDL_CondWait(is->pictq_cond, is->pictq_mutex);
}
@@ -1618,7 +1599,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double
}
/* if the queue is aborted, we have to pop the pending ALLOC event or wait for the allocation to complete */
if (is->videoq.abort_request && SDL_PeepEvents(&event, 1, SDL_GETEVENT, SDL_EVENTMASK(FF_ALLOC_EVENT)) != 1) {
while (!vp->allocated && !is->abort_request) {
while (!vp->allocated) {
SDL_CondWait(is->pictq_cond, is->pictq_mutex);
}
}
@@ -1665,7 +1646,6 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double
SDL_UnlockYUVOverlay(vp->bmp);
vp->pts = pts;
vp->duration = duration;
vp->pos = pos;
vp->serial = serial;
@@ -1688,6 +1668,18 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
if (pkt->data == flush_pkt.data) {
avcodec_flush_buffers(is->video_st->codec);
SDL_LockMutex(is->pictq_mutex);
// Make sure there are no long delay timers (ideally we should just flush the queue but that's harder)
while (is->pictq_size && !is->videoq.abort_request) {
SDL_CondWait(is->pictq_cond, is->pictq_mutex);
}
is->video_current_pos = -1;
is->frame_last_pts = AV_NOPTS_VALUE;
is->frame_last_duration = 0;
is->frame_timer = (double)av_gettime() / 1000000.0;
is->frame_last_dropped_pts = AV_NOPTS_VALUE;
SDL_UnlockMutex(is->pictq_mutex);
return 0;
}
@@ -1715,17 +1707,23 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
frame->sample_aspect_ratio = av_guess_sample_aspect_ratio(is->ic, is->video_st, frame);
if (framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) {
if (frame->pts != AV_NOPTS_VALUE) {
double diff = dpts - get_master_clock(is);
if (!isnan(diff) && fabs(diff) < AV_NOSYNC_THRESHOLD &&
diff - is->frame_last_filter_delay < 0 &&
*serial == is->vidclk.serial &&
SDL_LockMutex(is->pictq_mutex);
if (is->frame_last_pts != AV_NOPTS_VALUE && frame->pts != AV_NOPTS_VALUE) {
double clockdiff = get_clock(&is->vidclk) - get_master_clock(is);
double ptsdiff = dpts - is->frame_last_pts;
if (!isnan(clockdiff) && fabs(clockdiff) < AV_NOSYNC_THRESHOLD &&
!isnan(ptsdiff) && ptsdiff > 0 && ptsdiff < AV_NOSYNC_THRESHOLD &&
clockdiff + ptsdiff - is->frame_last_filter_delay < 0 &&
is->videoq.nb_packets) {
is->frame_last_dropped_pos = av_frame_get_pkt_pos(frame);
is->frame_last_dropped_pts = dpts;
is->frame_last_dropped_serial = *serial;
is->frame_drops_early++;
av_frame_unref(frame);
ret = 0;
}
}
SDL_UnlockMutex(is->pictq_mutex);
}
return ret;
@@ -1737,8 +1735,7 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
static int configure_filtergraph(AVFilterGraph *graph, const char *filtergraph,
AVFilterContext *source_ctx, AVFilterContext *sink_ctx)
{
int ret, i;
int nb_filters = graph->nb_filters;
int ret;
AVFilterInOut *outputs = NULL, *inputs = NULL;
if (filtergraph) {
@@ -1766,10 +1763,6 @@ static int configure_filtergraph(AVFilterGraph *graph, const char *filtergraph,
goto fail;
}
/* Reorder the filters to ensure that inputs of the custom filters are merged first */
for (i = 0; i < graph->nb_filters - nb_filters; i++)
FFSWAP(AVFilterContext*, graph->filters[i], graph->filters[i + nb_filters]);
ret = avfilter_graph_config(graph, NULL);
fail:
avfilter_inout_free(&outputs);
@@ -1783,7 +1776,7 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
char sws_flags_str[128];
char buffersrc_args[256];
int ret;
AVFilterContext *filt_src = NULL, *filt_out = NULL, *last_filter = NULL;
AVFilterContext *filt_src = NULL, *filt_out = NULL, *filt_crop;
AVCodecContext *codec = is->video_st->codec;
AVRational fr = av_guess_frame_rate(is->ic, is->video_st, NULL);
@@ -1814,49 +1807,16 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
if ((ret = av_opt_set_int_list(filt_out, "pix_fmts", pix_fmts, AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN)) < 0)
goto fail;
last_filter = filt_out;
/* Note: this macro adds a filter before the lastly added filter, so the
* processing order of the filters is in reverse */
#define INSERT_FILT(name, arg) do { \
AVFilterContext *filt_ctx; \
\
ret = avfilter_graph_create_filter(&filt_ctx, \
avfilter_get_by_name(name), \
"ffplay_" name, arg, NULL, graph); \
if (ret < 0) \
goto fail; \
\
ret = avfilter_link(filt_ctx, 0, last_filter, 0); \
if (ret < 0) \
goto fail; \
\
last_filter = filt_ctx; \
} while (0)
/* SDL YUV code is not handling odd width/height for some driver
* combinations, therefore we crop the picture to an even width/height. */
INSERT_FILT("crop", "floor(in_w/2)*2:floor(in_h/2)*2");
if ((ret = avfilter_graph_create_filter(&filt_crop,
avfilter_get_by_name("crop"),
"ffplay_crop", "floor(in_w/2)*2:floor(in_h/2)*2", NULL, graph)) < 0)
goto fail;
if ((ret = avfilter_link(filt_crop, 0, filt_out, 0)) < 0)
goto fail;
if (autorotate) {
AVDictionaryEntry *rotate_tag = av_dict_get(is->video_st->metadata, "rotate", NULL, 0);
if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
if (!strcmp(rotate_tag->value, "90")) {
INSERT_FILT("transpose", "clock");
} else if (!strcmp(rotate_tag->value, "180")) {
INSERT_FILT("hflip", NULL);
INSERT_FILT("vflip", NULL);
} else if (!strcmp(rotate_tag->value, "270")) {
INSERT_FILT("transpose", "cclock");
} else {
char rotate_buf[64];
snprintf(rotate_buf, sizeof(rotate_buf), "%s*PI/180", rotate_tag->value);
INSERT_FILT("rotate", rotate_buf);
}
}
}
if ((ret = configure_filtergraph(graph, vfilters, filt_src, last_filter)) < 0)
if ((ret = configure_filtergraph(graph, vfilters, filt_src, filt_crop)) < 0)
goto fail;
is->in_video_filter = filt_src;
@@ -1949,11 +1909,8 @@ static int video_thread(void *arg)
VideoState *is = arg;
AVFrame *frame = av_frame_alloc();
double pts;
double duration;
int ret;
int serial = 0;
AVRational tb = is->video_st->time_base;
AVRational frame_rate = av_guess_frame_rate(is->ic, is->video_st, NULL);
#if CONFIG_AVFILTER
AVFilterGraph *graph = avfilter_graph_alloc();
@@ -1962,13 +1919,13 @@ static int video_thread(void *arg)
int last_h = 0;
enum AVPixelFormat last_format = -2;
int last_serial = -1;
int last_vfilter_idx = 0;
#endif
for (;;) {
while (is->paused && !is->videoq.abort_request)
SDL_Delay(10);
avcodec_get_frame_defaults(frame);
av_free_packet(&pkt);
ret = get_video_frame(is, frame, &pkt, &serial);
@@ -1981,8 +1938,7 @@ static int video_thread(void *arg)
if ( last_w != frame->width
|| last_h != frame->height
|| last_format != frame->format
|| last_serial != serial
|| last_vfilter_idx != is->vfilter_idx) {
|| last_serial != serial) {
av_log(NULL, AV_LOG_DEBUG,
"Video frame changed from size:%dx%d format:%s serial:%d to size:%dx%d format:%s serial:%d\n",
last_w, last_h,
@@ -1991,11 +1947,12 @@ static int video_thread(void *arg)
(const char *)av_x_if_null(av_get_pix_fmt_name(frame->format), "none"), serial);
avfilter_graph_free(&graph);
graph = avfilter_graph_alloc();
if ((ret = configure_video_filters(graph, is, vfilters_list ? vfilters_list[is->vfilter_idx] : NULL, frame)) < 0) {
if ((ret = configure_video_filters(graph, is, vfilters, frame)) < 0) {
SDL_Event event;
event.type = FF_QUIT_EVENT;
event.user.data1 = is;
SDL_PushEvent(&event);
av_free_packet(&pkt);
goto the_end;
}
filt_in = is->in_video_filter;
@@ -2004,16 +1961,17 @@ static int video_thread(void *arg)
last_h = frame->height;
last_format = frame->format;
last_serial = serial;
last_vfilter_idx = is->vfilter_idx;
frame_rate = filt_out->inputs[0]->frame_rate;
}
ret = av_buffersrc_add_frame(filt_in, frame);
if (ret < 0)
goto the_end;
av_frame_unref(frame);
avcodec_get_frame_defaults(frame);
av_free_packet(&pkt);
while (ret >= 0) {
is->frame_last_returned_time = av_gettime_relative() / 1000000.0;
is->frame_last_returned_time = av_gettime() / 1000000.0;
ret = av_buffersink_get_frame_flags(filt_out, frame, 0);
if (ret < 0) {
@@ -2023,23 +1981,25 @@ static int video_thread(void *arg)
break;
}
is->frame_last_filter_delay = av_gettime_relative() / 1000000.0 - is->frame_last_returned_time;
is->frame_last_filter_delay = av_gettime() / 1000000.0 - is->frame_last_returned_time;
if (fabs(is->frame_last_filter_delay) > AV_NOSYNC_THRESHOLD / 10.0)
is->frame_last_filter_delay = 0;
tb = filt_out->inputs[0]->time_base;
#endif
duration = (frame_rate.num && frame_rate.den ? av_q2d((AVRational){frame_rate.den, frame_rate.num}) : 0);
pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);
ret = queue_picture(is, frame, pts, duration, av_frame_get_pkt_pos(frame), serial);
pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(filt_out->inputs[0]->time_base);
ret = queue_picture(is, frame, pts, av_frame_get_pkt_pos(frame), serial);
av_frame_unref(frame);
#if CONFIG_AVFILTER
}
#else
pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(is->video_st->time_base);
ret = queue_picture(is, frame, pts, av_frame_get_pkt_pos(frame), serial);
av_frame_unref(frame);
#endif
if (ret < 0)
goto the_end;
}
the_end:
avcodec_flush_buffers(is->video_st->codec);
#if CONFIG_AVFILTER
avfilter_graph_free(&graph);
#endif
@@ -2209,10 +2169,11 @@ static int audio_decode_frame(VideoState *is)
/* NOTE: the audio packet can contain several frames */
while (pkt_temp->stream_index != -1 || is->audio_buf_frames_pending) {
if (!is->frame) {
if (!(is->frame = av_frame_alloc()))
if (!(is->frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
} else {
av_frame_unref(is->frame);
avcodec_get_frame_defaults(is->frame);
}
if (is->audioq.serial != is->audio_pkt_temp_serial)
@@ -2287,6 +2248,7 @@ static int audio_decode_frame(VideoState *is)
if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
return ret;
av_frame_unref(is->frame);
#endif
}
#if CONFIG_AVFILTER
@@ -2423,8 +2385,10 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
VideoState *is = opaque;
int audio_size, len1;
int bytes_per_sec;
int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, 1, is->audio_tgt.fmt, 1);
audio_callback_time = av_gettime_relative();
audio_callback_time = av_gettime();
while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
@@ -2432,7 +2396,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
if (audio_size < 0) {
/* if error, just output silence */
is->audio_buf = is->silence_buf;
is->audio_buf_size = sizeof(is->silence_buf) / is->audio_tgt.frame_size * is->audio_tgt.frame_size;
is->audio_buf_size = sizeof(is->silence_buf) / frame_size * frame_size;
} else {
if (is->show_mode != SHOW_MODE_VIDEO)
update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
@@ -2448,10 +2412,11 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
stream += len1;
is->audio_buf_index += len1;
}
bytes_per_sec = is->audio_tgt.freq * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
/* Let's assume the audio driver that is used by SDL has two periods. */
if (!isnan(is->audio_clock)) {
set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
sync_clock_to_slave(&is->extclk, &is->audclk);
}
}
@@ -2461,8 +2426,6 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
SDL_AudioSpec wanted_spec, spec;
const char *env;
static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
static const int next_sample_rates[] = {0, 44100, 48000, 96000, 192000};
int next_sample_rate_idx = FF_ARRAY_ELEMS(next_sample_rates) - 1;
env = SDL_getenv("SDL_AUDIO_CHANNELS");
if (env) {
@@ -2473,32 +2436,24 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
}
wanted_nb_channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
wanted_spec.channels = wanted_nb_channels;
wanted_spec.channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
wanted_spec.freq = wanted_sample_rate;
if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
av_log(NULL, AV_LOG_ERROR, "Invalid sample rate or channel count!\n");
return -1;
}
while (next_sample_rate_idx && next_sample_rates[next_sample_rate_idx] >= wanted_spec.freq)
next_sample_rate_idx--;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.silence = 0;
wanted_spec.samples = FFMAX(SDL_AUDIO_MIN_BUFFER_SIZE, 2 << av_log2(wanted_spec.freq / SDL_AUDIO_MAX_CALLBACKS_PER_SEC));
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
wanted_spec.callback = sdl_audio_callback;
wanted_spec.userdata = opaque;
while (SDL_OpenAudio(&wanted_spec, &spec) < 0) {
av_log(NULL, AV_LOG_WARNING, "SDL_OpenAudio (%d channels, %d Hz): %s\n",
wanted_spec.channels, wanted_spec.freq, SDL_GetError());
av_log(NULL, AV_LOG_WARNING, "SDL_OpenAudio (%d channels): %s\n", wanted_spec.channels, SDL_GetError());
wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)];
if (!wanted_spec.channels) {
wanted_spec.freq = next_sample_rates[next_sample_rate_idx--];
wanted_spec.channels = wanted_nb_channels;
if (!wanted_spec.freq) {
av_log(NULL, AV_LOG_ERROR,
"No more combinations to try, audio open failed\n");
return -1;
}
av_log(NULL, AV_LOG_ERROR,
"No more channel combinations to try, audio open failed\n");
return -1;
}
wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels);
}
@@ -2520,12 +2475,6 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
audio_hw_params->freq = spec.freq;
audio_hw_params->channel_layout = wanted_channel_layout;
audio_hw_params->channels = spec.channels;
audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1);
audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {
av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");
return -1;
}
return spec.size;
}
@@ -2572,6 +2521,7 @@ static int stream_component_open(VideoState *is, int stream_index)
stream_lowres = av_codec_get_max_lowres(codec);
}
av_codec_set_lowres(avctx, stream_lowres);
avctx->error_concealment = error_concealment;
if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
if (fast) avctx->flags2 |= CODEC_FLAG2_FAST;
@@ -2629,7 +2579,7 @@ static int stream_component_open(VideoState *is, int stream_index)
is->audio_diff_avg_count = 0;
/* since we do not have a precise anough audio fifo fullness,
we correct audio sync only if larger than this threshold */
is->audio_diff_threshold = (double)(is->audio_hw_buf_size) / is->audio_tgt.bytes_per_sec;
is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / av_samples_get_buffer_size(NULL, is->audio_tgt.channels, is->audio_tgt.freq, is->audio_tgt.fmt, 1);
memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
memset(&is->audio_pkt_temp, 0, sizeof(is->audio_pkt_temp));
@@ -2807,8 +2757,6 @@ static int read_thread(void *arg)
if (genpts)
ic->flags |= AVFMT_FLAG_GENPTS;
av_format_inject_global_side_data(ic);
opts = setup_find_stream_info_opts(ic, codec_opts);
orig_nb_streams = ic->nb_streams;
@@ -2876,13 +2824,6 @@ static int read_thread(void *arg)
}
is->show_mode = show_mode;
if (st_index[AVMEDIA_TYPE_VIDEO] >= 0) {
AVStream *st = ic->streams[st_index[AVMEDIA_TYPE_VIDEO]];
AVCodecContext *avctx = st->codec;
AVRational sar = av_guess_sample_aspect_ratio(ic, st, NULL);
if (avctx->width)
set_default_window_size(avctx->width, avctx->height, sar);
}
/* open the streams */
if (st_index[AVMEDIA_TYPE_AUDIO] >= 0) {
@@ -2992,7 +2933,7 @@ static int read_thread(void *arg)
}
if (!is->paused &&
(!is->audio_st || is->audio_finished == is->audioq.serial) &&
(!is->video_st || (is->video_finished == is->videoq.serial && pictq_nb_remaining(is) == 0))) {
(!is->video_st || (is->video_finished == is->videoq.serial && is->pictq_size == 0))) {
if (loop != 1 && (!loop || --loop)) {
stream_seek(is, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0);
} else if (autoexit) {
@@ -3005,8 +2946,6 @@ static int read_thread(void *arg)
packet_queue_put_nullpacket(&is->videoq, is->video_stream);
if (is->audio_stream >= 0)
packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
if (is->subtitle_stream >= 0)
packet_queue_put_nullpacket(&is->subtitleq, is->subtitle_stream);
SDL_Delay(10);
eof=0;
continue;
@@ -3177,11 +3116,6 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
the_end:
if (p && stream_index != -1)
stream_index = p->stream_index[stream_index];
av_log(NULL, AV_LOG_INFO, "Switch %s stream from #%d to #%d\n",
av_get_media_type_string(codec_type),
old_index,
stream_index);
stream_component_close(is, old_index);
stream_component_open(is, stream_index);
}
@@ -3219,7 +3153,7 @@ static void refresh_loop_wait_event(VideoState *is, SDL_Event *event) {
double remaining_time = 0.0;
SDL_PumpEvents();
while (!SDL_PeepEvents(event, 1, SDL_GETEVENT, SDL_ALLEVENTS)) {
if (!cursor_hidden && av_gettime_relative() - cursor_last_shown > CURSOR_HIDE_DELAY) {
if (!cursor_hidden && av_gettime() - cursor_last_shown > CURSOR_HIDE_DELAY) {
SDL_ShowCursor(0);
cursor_hidden = 1;
}
@@ -3232,33 +3166,6 @@ static void refresh_loop_wait_event(VideoState *is, SDL_Event *event) {
}
}
static void seek_chapter(VideoState *is, int incr)
{
int64_t pos = get_master_clock(is) * AV_TIME_BASE;
int i;
if (!is->ic->nb_chapters)
return;
/* find the current chapter */
for (i = 0; i < is->ic->nb_chapters; i++) {
AVChapter *ch = is->ic->chapters[i];
if (av_compare_ts(pos, AV_TIME_BASE_Q, ch->start, ch->time_base) < 0) {
i--;
break;
}
}
i += incr;
i = FFMAX(i, 0);
if (i >= is->ic->nb_chapters)
return;
av_log(NULL, AV_LOG_VERBOSE, "Seeking to chapter %d.\n", i);
stream_seek(is, av_rescale_q(is->ic->chapters[i]->start, is->ic->chapters[i]->time_base,
AV_TIME_BASE_Q), 0, 0);
}
/* handle an event sent by the GUI */
static void event_loop(VideoState *cur_stream)
{
@@ -3305,32 +3212,14 @@ static void event_loop(VideoState *cur_stream)
stream_cycle_channel(cur_stream, AVMEDIA_TYPE_SUBTITLE);
break;
case SDLK_w:
#if CONFIG_AVFILTER
if (cur_stream->show_mode == SHOW_MODE_VIDEO && cur_stream->vfilter_idx < nb_vfilters - 1) {
if (++cur_stream->vfilter_idx >= nb_vfilters)
cur_stream->vfilter_idx = 0;
} else {
cur_stream->vfilter_idx = 0;
toggle_audio_display(cur_stream);
}
#else
toggle_audio_display(cur_stream);
#endif
break;
case SDLK_PAGEUP:
if (cur_stream->ic->nb_chapters <= 1) {
incr = 600.0;
goto do_seek;
}
seek_chapter(cur_stream, 1);
break;
incr = 600.0;
goto do_seek;
case SDLK_PAGEDOWN:
if (cur_stream->ic->nb_chapters <= 1) {
incr = -600.0;
goto do_seek;
}
seek_chapter(cur_stream, -1);
break;
incr = -600.0;
goto do_seek;
case SDLK_LEFT:
incr = -10.0;
goto do_seek;
@@ -3383,7 +3272,7 @@ static void event_loop(VideoState *cur_stream)
SDL_ShowCursor(1);
cursor_hidden = 0;
}
cursor_last_shown = av_gettime_relative();
cursor_last_shown = av_gettime();
if (event.type == SDL_MOUSEBUTTONDOWN) {
x = event.button.x;
} else {
@@ -3571,6 +3460,7 @@ static const OptionDef options[] = {
{ "genpts", OPT_BOOL | OPT_EXPERT, { &genpts }, "generate pts", "" },
{ "drp", OPT_INT | HAS_ARG | OPT_EXPERT, { &decoder_reorder_pts }, "let decoder reorder pts 0=off 1=on -1=auto", ""},
{ "lowres", OPT_INT | HAS_ARG | OPT_EXPERT, { &lowres }, "", "" },
{ "ec", OPT_INT | HAS_ARG | OPT_EXPERT, { &error_concealment }, "set error concealment options", "bit_mask" },
{ "sync", HAS_ARG | OPT_EXPERT, { .func_arg = opt_sync }, "set audio-video sync. type (type=audio/video/ext)", "type" },
{ "autoexit", OPT_BOOL | OPT_EXPERT, { &autoexit }, "exit at the end", "" },
{ "exitonkeydown", OPT_BOOL | OPT_EXPERT, { &exit_on_keydown }, "exit on key down", "" },
@@ -3580,7 +3470,7 @@ static const OptionDef options[] = {
{ "infbuf", OPT_BOOL | OPT_EXPERT, { &infinite_buffer }, "don't limit the input buffer size (useful with realtime streams)", "" },
{ "window_title", OPT_STRING | HAS_ARG, { &window_title }, "set window title", "window title" },
#if CONFIG_AVFILTER
{ "vf", OPT_EXPERT | HAS_ARG, { .func_arg = opt_add_vfilter }, "set video filters", "filter_graph" },
{ "vf", OPT_STRING | HAS_ARG, { &vfilters }, "set video filters", "filter_graph" },
{ "af", OPT_STRING | HAS_ARG, { &afilters }, "set audio filters", "filter_graph" },
#endif
{ "rdftspeed", OPT_INT | HAS_ARG| OPT_AUDIO | OPT_EXPERT, { &rdftspeed }, "rdft speed", "msecs" },
@@ -3591,7 +3481,6 @@ static const OptionDef options[] = {
{ "acodec", HAS_ARG | OPT_STRING | OPT_EXPERT, { &audio_codec_name }, "force audio decoder", "decoder_name" },
{ "scodec", HAS_ARG | OPT_STRING | OPT_EXPERT, { &subtitle_codec_name }, "force subtitle decoder", "decoder_name" },
{ "vcodec", HAS_ARG | OPT_STRING | OPT_EXPERT, { &video_codec_name }, "force video decoder", "decoder_name" },
{ "autorotate", OPT_BOOL, { &autorotate }, "automatically rotate video", "" },
{ NULL, },
};
@@ -3624,7 +3513,7 @@ void show_help_default(const char *opt, const char *arg)
"v cycle video channel\n"
"t cycle subtitle channel in the current program\n"
"c cycle program\n"
"w cycle video filters or show modes\n"
"w show audio waves\n"
"s activate frame-step mode\n"
"left/right seek backward/forward 10 seconds\n"
"down/up seek backward/forward 1 minute\n"
@@ -3663,6 +3552,7 @@ int main(int argc, char **argv)
parse_loglevel(argc, argv, options);
/* register all codecs, demux and protocols */
avcodec_register_all();
#if CONFIG_AVDEVICE
avdevice_register_all();
#endif
@@ -3697,7 +3587,7 @@ int main(int argc, char **argv)
flags &= ~SDL_INIT_AUDIO;
if (display_disable)
SDL_putenv(dummy_videodriver); /* For the event queue, we always need a video driver. */
#if !defined(_WIN32) && !defined(__APPLE__)
#if !defined(__MINGW32__) && !defined(__APPLE__)
flags |= SDL_INIT_EVENTTHREAD; /* Not supported on Windows or Mac OS X */
#endif
if (SDL_Init (flags)) {

407
ffprobe.c
View File

@@ -24,7 +24,7 @@
*/
#include "config.h"
#include "libavutil/ffversion.h"
#include "version.h"
#include <string.h>
@@ -66,12 +66,6 @@ static int do_show_data = 0;
static int do_show_program_version = 0;
static int do_show_library_versions = 0;
static int do_show_chapter_tags = 0;
static int do_show_format_tags = 0;
static int do_show_frame_tags = 0;
static int do_show_program_tags = 0;
static int do_show_stream_tags = 0;
static int show_value_unit = 0;
static int use_value_prefix = 0;
static int use_byte_value_binary_prefix = 0;
@@ -141,7 +135,6 @@ typedef enum {
SECTION_ID_STREAM_DISPOSITION,
SECTION_ID_STREAMS,
SECTION_ID_STREAM_TAGS,
SECTION_ID_SUBTITLE,
} SectionID;
static struct section sections[] = {
@@ -151,7 +144,7 @@ static struct section sections[] = {
[SECTION_ID_ERROR] = { SECTION_ID_ERROR, "error", 0, { -1 } },
[SECTION_ID_FORMAT] = { SECTION_ID_FORMAT, "format", 0, { SECTION_ID_FORMAT_TAGS, -1 } },
[SECTION_ID_FORMAT_TAGS] = { SECTION_ID_FORMAT_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "format_tags" },
[SECTION_ID_FRAMES] = { SECTION_ID_FRAMES, "frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME, SECTION_ID_SUBTITLE, -1 } },
[SECTION_ID_FRAMES] = { SECTION_ID_FRAMES, "frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME, -1 } },
[SECTION_ID_FRAME] = { SECTION_ID_FRAME, "frame", 0, { SECTION_ID_FRAME_TAGS, -1 } },
[SECTION_ID_FRAME_TAGS] = { SECTION_ID_FRAME_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "frame_tags" },
[SECTION_ID_LIBRARY_VERSIONS] = { SECTION_ID_LIBRARY_VERSIONS, "library_versions", SECTION_FLAG_IS_ARRAY, { SECTION_ID_LIBRARY_VERSION, -1 } },
@@ -174,7 +167,6 @@ static struct section sections[] = {
[SECTION_ID_STREAM] = { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
[SECTION_ID_STREAM_DISPOSITION] = { SECTION_ID_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_disposition" },
[SECTION_ID_STREAM_TAGS] = { SECTION_ID_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "stream_tags" },
[SECTION_ID_SUBTITLE] = { SECTION_ID_SUBTITLE, "subtitle", 0, { -1 } },
};
static const OptionDef *options;
@@ -268,13 +260,6 @@ typedef struct WriterContext WriterContext;
#define WRITER_FLAG_DISPLAY_OPTIONAL_FIELDS 1
#define WRITER_FLAG_PUT_PACKETS_AND_FRAMES_IN_SAME_CHAPTER 2
typedef enum {
WRITER_STRING_VALIDATION_FAIL,
WRITER_STRING_VALIDATION_REPLACE,
WRITER_STRING_VALIDATION_IGNORE,
WRITER_STRING_VALIDATION_NB
} StringValidation;
typedef struct Writer {
const AVClass *priv_class; ///< private class of the writer, if any
int priv_size; ///< private size for the writer context
@@ -315,10 +300,6 @@ struct WriterContext {
unsigned int nb_section_packet; ///< number of the packet section in case we are in "packets_and_frames" section
unsigned int nb_section_frame; ///< number of the frame section in case we are in "packets_and_frames" section
unsigned int nb_section_packet_frame; ///< nb_section_packet or nb_section_frame according if is_packets_and_frames
StringValidation string_validation;
char *string_validation_replacement;
unsigned int string_validation_utf8_flags;
};
static const char *writer_get_name(void *p)
@@ -327,35 +308,11 @@ static const char *writer_get_name(void *p)
return wctx->writer->name;
}
#define OFFSET(x) offsetof(WriterContext, x)
static const AVOption writer_options[] = {
{ "string_validation", "set string validation mode",
OFFSET(string_validation), AV_OPT_TYPE_INT, {.i64=WRITER_STRING_VALIDATION_REPLACE}, 0, WRITER_STRING_VALIDATION_NB-1, .unit = "sv" },
{ "sv", "set string validation mode",
OFFSET(string_validation), AV_OPT_TYPE_INT, {.i64=WRITER_STRING_VALIDATION_REPLACE}, 0, WRITER_STRING_VALIDATION_NB-1, .unit = "sv" },
{ "ignore", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = WRITER_STRING_VALIDATION_IGNORE}, .unit = "sv" },
{ "replace", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = WRITER_STRING_VALIDATION_REPLACE}, .unit = "sv" },
{ "fail", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = WRITER_STRING_VALIDATION_FAIL}, .unit = "sv" },
{ "string_validation_replacement", "set string validation replacement string", OFFSET(string_validation_replacement), AV_OPT_TYPE_STRING, {.str=""}},
{ "svr", "set string validation replacement string", OFFSET(string_validation_replacement), AV_OPT_TYPE_STRING, {.str="\xEF\xBF\xBD"}},
{ NULL }
};
static void *writer_child_next(void *obj, void *prev)
{
WriterContext *ctx = obj;
if (!prev && ctx->writer && ctx->writer->priv_class && ctx->priv)
return ctx->priv;
return NULL;
}
static const AVClass writer_class = {
.class_name = "Writer",
.item_name = writer_get_name,
.option = writer_options,
.version = LIBAVUTIL_VERSION_INT,
.child_next = writer_child_next,
"Writer",
writer_get_name,
NULL,
LIBAVUTIL_VERSION_INT,
};
static void writer_close(WriterContext **wctx)
@@ -372,19 +329,9 @@ static void writer_close(WriterContext **wctx)
if ((*wctx)->writer->priv_class)
av_opt_free((*wctx)->priv);
av_freep(&((*wctx)->priv));
av_opt_free(*wctx);
av_freep(wctx);
}
static void bprint_bytes(AVBPrint *bp, const uint8_t *ubuf, size_t ubuf_size)
{
int i;
av_bprintf(bp, "0X");
for (i = 0; i < ubuf_size; i++)
av_bprintf(bp, "%02X", ubuf[i]);
}
static int writer_open(WriterContext **wctx, const Writer *writer, const char *args,
const struct section *sections, int nb_sections)
{
@@ -406,55 +353,14 @@ static int writer_open(WriterContext **wctx, const Writer *writer, const char *a
(*wctx)->sections = sections;
(*wctx)->nb_sections = nb_sections;
av_opt_set_defaults(*wctx);
if (writer->priv_class) {
void *priv_ctx = (*wctx)->priv;
*((const AVClass **)priv_ctx) = writer->priv_class;
av_opt_set_defaults(priv_ctx);
}
/* convert options to dictionary */
if (args) {
AVDictionary *opts = NULL;
AVDictionaryEntry *opt = NULL;
if ((ret = av_dict_parse_string(&opts, args, "=", ":", 0)) < 0) {
av_log(*wctx, AV_LOG_ERROR, "Failed to parse option string '%s' provided to writer context\n", args);
av_dict_free(&opts);
if (args &&
(ret = av_set_options_string(priv_ctx, args, "=", ":")) < 0)
goto fail;
}
while ((opt = av_dict_get(opts, "", opt, AV_DICT_IGNORE_SUFFIX))) {
if ((ret = av_opt_set(*wctx, opt->key, opt->value, AV_OPT_SEARCH_CHILDREN)) < 0) {
av_log(*wctx, AV_LOG_ERROR, "Failed to set option '%s' with value '%s' provided to writer context\n",
opt->key, opt->value);
av_dict_free(&opts);
goto fail;
}
}
av_dict_free(&opts);
}
/* validate replace string */
{
const uint8_t *p = (*wctx)->string_validation_replacement;
const uint8_t *endp = p + strlen(p);
while (*p) {
const uint8_t *p0 = p;
int32_t code;
ret = av_utf8_decode(&code, &p, endp, (*wctx)->string_validation_utf8_flags);
if (ret < 0) {
AVBPrint bp;
av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
bprint_bytes(&bp, p0, p-p0),
av_log(wctx, AV_LOG_ERROR,
"Invalid UTF8 sequence %s found in string validation replace '%s'\n",
bp.str, (*wctx)->string_validation_replacement);
return ret;
}
}
}
for (i = 0; i < SECTION_MAX_NB_LEVELS; i++)
@@ -524,98 +430,18 @@ static inline void writer_print_integer(WriterContext *wctx,
}
}
static inline int validate_string(WriterContext *wctx, char **dstp, const char *src)
{
const uint8_t *p, *endp;
AVBPrint dstbuf;
int invalid_chars_nb = 0, ret = 0;
av_bprint_init(&dstbuf, 0, AV_BPRINT_SIZE_UNLIMITED);
endp = src + strlen(src);
for (p = (uint8_t *)src; *p;) {
uint32_t code;
int invalid = 0;
const uint8_t *p0 = p;
if (av_utf8_decode(&code, &p, endp, wctx->string_validation_utf8_flags) < 0) {
AVBPrint bp;
av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
bprint_bytes(&bp, p0, p-p0);
av_log(wctx, AV_LOG_DEBUG,
"Invalid UTF-8 sequence %s found in string '%s'\n", bp.str, src);
invalid = 1;
}
if (invalid) {
invalid_chars_nb++;
switch (wctx->string_validation) {
case WRITER_STRING_VALIDATION_FAIL:
av_log(wctx, AV_LOG_ERROR,
"Invalid UTF-8 sequence found in string '%s'\n", src);
ret = AVERROR_INVALIDDATA;
goto end;
break;
case WRITER_STRING_VALIDATION_REPLACE:
av_bprintf(&dstbuf, "%s", wctx->string_validation_replacement);
break;
}
}
if (!invalid || wctx->string_validation == WRITER_STRING_VALIDATION_IGNORE)
av_bprint_append_data(&dstbuf, p0, p-p0);
}
if (invalid_chars_nb && wctx->string_validation == WRITER_STRING_VALIDATION_REPLACE) {
av_log(wctx, AV_LOG_WARNING,
"%d invalid UTF-8 sequence(s) found in string '%s', replaced with '%s'\n",
invalid_chars_nb, src, wctx->string_validation_replacement);
}
end:
av_bprint_finalize(&dstbuf, dstp);
return ret;
}
#define PRINT_STRING_OPT 1
#define PRINT_STRING_VALIDATE 2
static inline int writer_print_string(WriterContext *wctx,
const char *key, const char *val, int flags)
static inline void writer_print_string(WriterContext *wctx,
const char *key, const char *val, int opt)
{
const struct section *section = wctx->section[wctx->level];
int ret = 0;
if ((flags & PRINT_STRING_OPT)
&& !(wctx->writer->flags & WRITER_FLAG_DISPLAY_OPTIONAL_FIELDS))
return 0;
if (opt && !(wctx->writer->flags & WRITER_FLAG_DISPLAY_OPTIONAL_FIELDS))
return;
if (section->show_all_entries || av_dict_get(section->entries_to_show, key, NULL, 0)) {
if (flags & PRINT_STRING_VALIDATE) {
char *key1 = NULL, *val1 = NULL;
ret = validate_string(wctx, &key1, key);
if (ret < 0) goto end;
ret = validate_string(wctx, &val1, val);
if (ret < 0) goto end;
wctx->writer->print_string(wctx, key1, val1);
end:
if (ret < 0) {
av_log(wctx, AV_LOG_ERROR,
"Invalid key=value string combination %s=%s in section %s\n",
key, val, section->unique_name);
}
av_free(key1);
av_free(val1);
} else {
wctx->writer->print_string(wctx, key, val);
}
wctx->writer->print_string(wctx, key, val);
wctx->nb_item[wctx->level]++;
}
return ret;
}
static inline void writer_print_rational(WriterContext *wctx,
@@ -633,7 +459,7 @@ static void writer_print_time(WriterContext *wctx, const char *key,
char buf[128];
if ((!is_duration && ts == AV_NOPTS_VALUE) || (is_duration && ts == 0)) {
writer_print_string(wctx, key, "N/A", PRINT_STRING_OPT);
writer_print_string(wctx, key, "N/A", 1);
} else {
double d = ts * av_q2d(*time_base);
struct unit_value uv;
@@ -647,7 +473,7 @@ static void writer_print_time(WriterContext *wctx, const char *key,
static void writer_print_ts(WriterContext *wctx, const char *key, int64_t ts, int is_duration)
{
if ((!is_duration && ts == AV_NOPTS_VALUE) || (is_duration && ts == 0)) {
writer_print_string(wctx, key, "N/A", PRINT_STRING_OPT);
writer_print_string(wctx, key, "N/A", 1);
} else {
writer_print_integer(wctx, key, ts);
}
@@ -716,9 +542,9 @@ static const char *name##_get_name(void *ctx) \
return #name ; \
} \
static const AVClass name##_class = { \
.class_name = #name, \
.item_name = name##_get_name, \
.option = name##_options \
#name, \
name##_get_name, \
name##_options \
}
/* Default output */
@@ -730,7 +556,6 @@ typedef struct DefaultContext {
int nested_section[SECTION_MAX_NB_LEVELS];
} DefaultContext;
#undef OFFSET
#define OFFSET(x) offsetof(DefaultContext, x)
static const AVOption default_options[] = {
@@ -1617,8 +1442,7 @@ static void writer_register_all(void)
#define print_int(k, v) writer_print_integer(w, k, v)
#define print_q(k, v, s) writer_print_rational(w, k, v, s)
#define print_str(k, v) writer_print_string(w, k, v, 0)
#define print_str_opt(k, v) writer_print_string(w, k, v, PRINT_STRING_OPT)
#define print_str_validate(k, v) writer_print_string(w, k, v, PRINT_STRING_VALIDATE)
#define print_str_opt(k, v) writer_print_string(w, k, v, 1)
#define print_time(k, v, tb) writer_print_time(w, k, v, tb, 0)
#define print_ts(k, v) writer_print_ts(w, k, v, 0)
#define print_duration_time(k, v, tb) writer_print_time(w, k, v, tb, 1)
@@ -1641,22 +1465,16 @@ static void writer_register_all(void)
memset( (ptr) + (cur_n), 0, ((new_n) - (cur_n)) * sizeof(*(ptr)) ); \
}
static inline int show_tags(WriterContext *w, AVDictionary *tags, int section_id)
static inline void show_tags(WriterContext *wctx, AVDictionary *tags, int section_id)
{
AVDictionaryEntry *tag = NULL;
int ret = 0;
if (!tags)
return 0;
writer_print_section_header(w, section_id);
while ((tag = av_dict_get(tags, "", tag, AV_DICT_IGNORE_SUFFIX))) {
if ((ret = print_str_validate(tag->key, tag->value)) < 0)
break;
}
writer_print_section_footer(w);
return ret;
return;
writer_print_section_header(wctx, section_id);
while ((tag = av_dict_get(tags, "", tag, AV_DICT_IGNORE_SUFFIX)))
writer_print_string(wctx, tag->key, tag->value, 0);
writer_print_section_footer(wctx);
}
static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pkt, int packet_idx)
@@ -1694,29 +1512,6 @@ static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pk
fflush(stdout);
}
static void show_subtitle(WriterContext *w, AVSubtitle *sub, AVStream *stream,
AVFormatContext *fmt_ctx)
{
AVBPrint pbuf;
av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);
writer_print_section_header(w, SECTION_ID_SUBTITLE);
print_str ("media_type", "subtitle");
print_ts ("pts", sub->pts);
print_time("pts_time", sub->pts, &AV_TIME_BASE_Q);
print_int ("format", sub->format);
print_int ("start_display_time", sub->start_display_time);
print_int ("end_display_time", sub->end_display_time);
print_int ("num_rects", sub->num_rects);
writer_print_section_footer(w);
av_bprint_finalize(&pbuf, NULL);
fflush(stdout);
}
static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
AVFormatContext *fmt_ctx)
{
@@ -1735,8 +1530,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
print_time("pkt_pts_time", frame->pkt_pts, &stream->time_base);
print_ts ("pkt_dts", frame->pkt_dts);
print_time("pkt_dts_time", frame->pkt_dts, &stream->time_base);
print_ts ("best_effort_timestamp", av_frame_get_best_effort_timestamp(frame));
print_time("best_effort_timestamp_time", av_frame_get_best_effort_timestamp(frame), &stream->time_base);
print_duration_ts ("pkt_duration", av_frame_get_pkt_duration(frame));
print_duration_time("pkt_duration_time", av_frame_get_pkt_duration(frame), &stream->time_base);
if (av_frame_get_pkt_pos (frame) != -1) print_fmt ("pkt_pos", "%"PRId64, av_frame_get_pkt_pos(frame));
@@ -1782,8 +1575,7 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
print_str_opt("channel_layout", "unknown");
break;
}
if (do_show_frame_tags)
show_tags(w, av_frame_get_metadata(frame), SECTION_ID_FRAME_TAGS);
show_tags(w, av_frame_get_metadata(frame), SECTION_ID_FRAME_TAGS);
writer_print_section_footer(w);
@@ -1796,7 +1588,6 @@ static av_always_inline int process_frame(WriterContext *w,
AVFrame *frame, AVPacket *pkt)
{
AVCodecContext *dec_ctx = fmt_ctx->streams[pkt->stream_index]->codec;
AVSubtitle sub;
int ret = 0, got_frame = 0;
if (dec_ctx->codec) {
@@ -1808,10 +1599,6 @@ static av_always_inline int process_frame(WriterContext *w,
case AVMEDIA_TYPE_AUDIO:
ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, pkt);
break;
case AVMEDIA_TYPE_SUBTITLE:
ret = avcodec_decode_subtitle2(dec_ctx, &sub, &got_frame, pkt);
break;
}
}
@@ -1821,15 +1608,9 @@ static av_always_inline int process_frame(WriterContext *w,
pkt->data += ret;
pkt->size -= ret;
if (got_frame) {
int is_sub = (dec_ctx->codec_type == AVMEDIA_TYPE_SUBTITLE);
nb_streams_frames[pkt->stream_index]++;
if (do_show_frames)
if (is_sub)
show_subtitle(w, &sub, fmt_ctx->streams[pkt->stream_index], fmt_ctx);
else
show_frame(w, frame, fmt_ctx->streams[pkt->stream_index], fmt_ctx);
if (is_sub)
avsubtitle_free(&sub);
show_frame(w, frame, fmt_ctx->streams[pkt->stream_index], fmt_ctx);
}
return got_frame;
}
@@ -1963,7 +1744,7 @@ end:
return ret;
}
static int read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
static void read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
{
int i, ret = 0;
int64_t cur_ts = fmt_ctx->start_time;
@@ -1978,11 +1759,9 @@ static int read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
break;
}
}
return ret;
}
static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx, int in_program)
static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx, int in_program)
{
AVStream *stream = fmt_ctx->streams[stream_idx];
AVCodecContext *dec_ctx;
@@ -1991,8 +1770,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
const char *s;
AVRational sar, dar;
AVBPrint pbuf;
const AVCodecDescriptor *cd;
int ret = 0;
av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);
@@ -2009,12 +1786,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
if (dec->long_name) print_str ("codec_long_name", dec->long_name);
else print_str_opt("codec_long_name", "unknown");
}
} else if ((cd = avcodec_descriptor_get(stream->codec->codec_id))) {
print_str_opt("codec_name", cd->name);
if (!do_bitexact) {
print_str_opt("codec_long_name",
cd->long_name ? cd->long_name : "unknown");
}
} else {
print_str_opt("codec_name", "unknown");
if (!do_bitexact) {
@@ -2058,13 +1829,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
if (s) print_str ("pix_fmt", s);
else print_str_opt("pix_fmt", "unknown");
print_int("level", dec_ctx->level);
if (dec_ctx->color_range != AVCOL_RANGE_UNSPECIFIED)
print_str ("color_range", dec_ctx->color_range == AVCOL_RANGE_MPEG ? "tv": "pc");
else
print_str_opt("color_range", "N/A");
s = av_get_colorspace_name(dec_ctx->colorspace);
if (s) print_str ("color_space", s);
else print_str_opt("color_space", "unknown");
if (dec_ctx->timecode_frame_start >= 0) {
char tcbuf[AV_TIMECODE_STR_SIZE];
av_timecode_make_mpeg_tc_string(tcbuf, dec_ctx->timecode_frame_start);
@@ -2129,8 +1893,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
print_time("duration", stream->duration, &stream->time_base);
if (dec_ctx->bit_rate > 0) print_val ("bit_rate", dec_ctx->bit_rate, unit_bit_per_second_str);
else print_str_opt("bit_rate", "N/A");
if (dec_ctx->rc_max_rate > 0) print_val ("max_bit_rate", dec_ctx->rc_max_rate, unit_bit_per_second_str);
else print_str_opt("max_bit_rate", "N/A");
if (stream->nb_frames) print_fmt ("nb_frames", "%"PRId64, stream->nb_frames);
else print_str_opt("nb_frames", "N/A");
if (nb_streams_frames[stream_idx]) print_fmt ("nb_read_frames", "%"PRIu64, nb_streams_frames[stream_idx]);
@@ -2162,35 +1924,26 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
writer_print_section_footer(w);
}
if (do_show_stream_tags)
ret = show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
writer_print_section_footer(w);
av_bprint_finalize(&pbuf, NULL);
fflush(stdout);
return ret;
}
static int show_streams(WriterContext *w, AVFormatContext *fmt_ctx)
static void show_streams(WriterContext *w, AVFormatContext *fmt_ctx)
{
int i, ret = 0;
int i;
writer_print_section_header(w, SECTION_ID_STREAMS);
for (i = 0; i < fmt_ctx->nb_streams; i++)
if (selected_streams[i]) {
ret = show_stream(w, fmt_ctx, i, 0);
if (ret < 0)
break;
}
if (selected_streams[i])
show_stream(w, fmt_ctx, i, 0);
writer_print_section_footer(w);
return ret;
}
static int show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *program)
static void show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *program)
{
int i, ret = 0;
int i;
writer_print_section_header(w, SECTION_ID_PROGRAM);
print_int("program_id", program->id);
@@ -2202,46 +1955,35 @@ static int show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *p
print_time("start_time", program->start_time, &AV_TIME_BASE_Q);
print_ts("end_pts", program->end_time);
print_time("end_time", program->end_time, &AV_TIME_BASE_Q);
if (do_show_program_tags)
ret = show_tags(w, program->metadata, SECTION_ID_PROGRAM_TAGS);
if (ret < 0)
goto end;
show_tags(w, program->metadata, SECTION_ID_PROGRAM_TAGS);
writer_print_section_header(w, SECTION_ID_PROGRAM_STREAMS);
for (i = 0; i < program->nb_stream_indexes; i++) {
if (selected_streams[program->stream_index[i]]) {
ret = show_stream(w, fmt_ctx, program->stream_index[i], 1);
if (ret < 0)
break;
}
if (selected_streams[program->stream_index[i]])
show_stream(w, fmt_ctx, program->stream_index[i], 1);
}
writer_print_section_footer(w);
end:
writer_print_section_footer(w);
return ret;
}
static int show_programs(WriterContext *w, AVFormatContext *fmt_ctx)
static void show_programs(WriterContext *w, AVFormatContext *fmt_ctx)
{
int i, ret = 0;
int i;
writer_print_section_header(w, SECTION_ID_PROGRAMS);
for (i = 0; i < fmt_ctx->nb_programs; i++) {
AVProgram *program = fmt_ctx->programs[i];
if (!program)
continue;
ret = show_program(w, fmt_ctx, program);
if (ret < 0)
break;
show_program(w, fmt_ctx, program);
}
writer_print_section_footer(w);
return ret;
}
static int show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
static void show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
{
int i, ret = 0;
int i;
writer_print_section_header(w, SECTION_ID_CHAPTERS);
for (i = 0; i < fmt_ctx->nb_chapters; i++) {
@@ -2254,23 +1996,19 @@ static int show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
print_time("start_time", chapter->start, &chapter->time_base);
print_int("end", chapter->end);
print_time("end_time", chapter->end, &chapter->time_base);
if (do_show_chapter_tags)
ret = show_tags(w, chapter->metadata, SECTION_ID_CHAPTER_TAGS);
show_tags(w, chapter->metadata, SECTION_ID_CHAPTER_TAGS);
writer_print_section_footer(w);
}
writer_print_section_footer(w);
return ret;
}
static int show_format(WriterContext *w, AVFormatContext *fmt_ctx)
static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
{
char val_str[128];
int64_t size = fmt_ctx->pb ? avio_size(fmt_ctx->pb) : -1;
int ret = 0;
writer_print_section_header(w, SECTION_ID_FORMAT);
print_str_validate("filename", fmt_ctx->filename);
print_str("filename", fmt_ctx->filename);
print_int("nb_streams", fmt_ctx->nb_streams);
print_int("nb_programs", fmt_ctx->nb_programs);
print_str("format_name", fmt_ctx->iformat->name);
@@ -2285,12 +2023,10 @@ static int show_format(WriterContext *w, AVFormatContext *fmt_ctx)
if (fmt_ctx->bit_rate > 0) print_val ("bit_rate", fmt_ctx->bit_rate, unit_bit_per_second_str);
else print_str_opt("bit_rate", "N/A");
print_int("probe_score", av_format_get_probe_score(fmt_ctx));
if (do_show_format_tags)
ret = show_tags(w, fmt_ctx->metadata, SECTION_ID_FORMAT_TAGS);
show_tags(w, fmt_ctx->metadata, SECTION_ID_FORMAT_TAGS);
writer_print_section_footer(w);
fflush(stdout);
return ret;
}
static void show_error(WriterContext *w, int err)
@@ -2396,8 +2132,6 @@ static int probe_file(WriterContext *wctx, const char *filename)
if (ret < 0)
return ret;
#define CHECK_END if (ret < 0) goto end
nb_streams = fmt_ctx->nb_streams;
REALLOCZ_ARRAY_STREAM(nb_streams_frames,0,fmt_ctx->nb_streams);
REALLOCZ_ARRAY_STREAM(nb_streams_packets,0,fmt_ctx->nb_streams);
@@ -2408,7 +2142,8 @@ static int probe_file(WriterContext *wctx, const char *filename)
ret = avformat_match_stream_specifier(fmt_ctx,
fmt_ctx->streams[i],
stream_specifier);
CHECK_END;
if (ret < 0)
goto end;
else
selected_streams[i] = ret;
ret = 0;
@@ -2427,29 +2162,18 @@ static int probe_file(WriterContext *wctx, const char *filename)
section_id = SECTION_ID_FRAMES;
if (do_show_frames || do_show_packets)
writer_print_section_header(wctx, section_id);
ret = read_packets(wctx, fmt_ctx);
read_packets(wctx, fmt_ctx);
if (do_show_frames || do_show_packets)
writer_print_section_footer(wctx);
CHECK_END;
}
if (do_show_programs) {
ret = show_programs(wctx, fmt_ctx);
CHECK_END;
}
if (do_show_streams) {
ret = show_streams(wctx, fmt_ctx);
CHECK_END;
}
if (do_show_chapters) {
ret = show_chapters(wctx, fmt_ctx);
CHECK_END;
}
if (do_show_format) {
ret = show_format(wctx, fmt_ctx);
CHECK_END;
}
if (do_show_programs)
show_programs(wctx, fmt_ctx);
if (do_show_streams)
show_streams(wctx, fmt_ctx);
if (do_show_chapters)
show_chapters(wctx, fmt_ctx);
if (do_show_format)
show_format(wctx, fmt_ctx);
end:
close_input_file(&fmt_ctx);
@@ -2475,7 +2199,7 @@ static void ffprobe_show_program_version(WriterContext *w)
writer_print_section_header(w, SECTION_ID_PROGRAM_VERSION);
print_str("version", FFMPEG_VERSION);
print_fmt("copyright", "Copyright (c) %d-%d the FFmpeg developers",
program_birth_year, CONFIG_THIS_YEAR);
program_birth_year, this_year);
print_str("build_date", __DATE__);
print_str("build_time", __TIME__);
print_str("compiler_ident", CC_IDENT);
@@ -2749,7 +2473,7 @@ static int parse_read_intervals(const char *intervals_spec)
n++;
n++;
read_intervals = av_malloc_array(n, sizeof(*read_intervals));
read_intervals = av_malloc(n * sizeof(*read_intervals));
if (!read_intervals) {
ret = AVERROR(ENOMEM);
goto end;
@@ -2946,12 +2670,6 @@ int main(int argc, char **argv)
SET_DO_SHOW(STREAM_DISPOSITION, stream_disposition);
SET_DO_SHOW(PROGRAM_STREAM_DISPOSITION, stream_disposition);
SET_DO_SHOW(CHAPTER_TAGS, chapter_tags);
SET_DO_SHOW(FORMAT_TAGS, format_tags);
SET_DO_SHOW(FRAME_TAGS, frame_tags);
SET_DO_SHOW(PROGRAM_TAGS, program_tags);
SET_DO_SHOW(STREAM_TAGS, stream_tags);
if (do_bitexact && (do_show_program_version || do_show_library_versions)) {
av_log(NULL, AV_LOG_ERROR,
"-bitexact and -show_program_version or -show_library_versions "
@@ -2980,9 +2698,6 @@ int main(int argc, char **argv)
if ((ret = writer_open(&wctx, w, w_args,
sections, FF_ARRAY_ELEMS(sections))) >= 0) {
if (w == &xml_writer)
wctx->string_validation_utf8_flags |= AV_UTF8_FLAG_EXCLUDE_XML_INVALID_CONTROL_CODES;
writer_print_section_header(wctx, SECTION_ID_ROOT);
if (do_show_program_version)

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