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Author SHA1 Message Date
Michael Niedermayer 390d6853d0 RELEASE_NOTES: Based on the version from 4.3
Name suggested by jb and Reto

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2022-01-14 19:43:55 +01:00
Vittorio Giovara e540605d42 vf_tonemap: Fix order of planes
This resulted in a dimmed tonemapping due to bad resulting luma
calculation.

Found by: Derek Buitenhuis

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
(cherry picked from commit 7d377558a6)
2022-01-14 16:34:01 +01:00
softworkz 4a9ec4d4e3 avfilter/vpp_qsv: fix regression on older api versions (e.g. 1.11)
Commit 8b83dad825 introduced a
regression in a way that scaling via vpp_qsv doesn't work any longer
for devices with an MSDK runtime version lower than 1.19. This is true
for older CPUs which are stuck at 1.11.
The commit added checks for the compile-sdk version but it didn't test
for the runtime version.

Signed-off-by: softworkz <softworkz@hotmail.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
(cherry picked from commit 479f3c6598)
2022-01-14 08:50:03 +08:00
Michael Niedermayer 171802a1ba avformat/rawvideodec: check packet size
Fixes: division by zero
Fixes: integer overflow
Fixes: 43347/clusterfuzz-testcase-minimized-ffmpeg_dem_V210X_fuzzer-5846911637127168

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: lance.lmwang@gmail.com
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c36a5dfc8f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2022-01-13 23:24:22 +01:00
Michael Niedermayer af06b9255f avcodec/apedec: Fix integer overflows in predictor_update_3930()
Fixes: signed integer overflow: 1074134419 - -1075212485 cannot be represented in type 'int'
Fixes: 43273/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-4706880883130368

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0c9c9bbd01)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2022-01-13 23:23:18 +01:00
Limin Wang 99d6ab7154 avutil/parseutils: use quadhd for Quad HD
qHD is 960x540 (q stands for quarter) and QHD is 2560x1440 (Q is quad).
use quadhd for QHD for abbreviation.

Fix ticket#9591

Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2022-01-12 17:54:19 +08:00
Anton Khirnov 92c57aece4 lavf/udp: do not return an uninitialized value from udp_open()
(cherry picked from commit 3c2b674468)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2022-01-11 09:11:00 +01:00
Andreas Rheinhardt 58922dc565 fftools/cmdutils: Fix undefined 1 << 31
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit bbf00916e4)
2022-01-11 00:24:23 +01:00
James Almer fa1328babf avcodec/libdav1d: explicitly set Dav1dSettings.apply_grain
Don't depend on its default value being 1, as that could change anytime.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 6c4074e423)
2022-01-10 12:21:52 -03:00
Timo Rothenpieler cd74c838fc avcodec/nvenc: zero-initialize NV_ENC_REGISTER_RESOURCE struct 2022-01-10 15:53:15 +01:00
Zhao Zhili ce1558e66b avformat/movenc: fix duration in mdhd box
mvhd and tkhd present the post-editlist duration, while mdhd should
have the pre-editlist duration. Regression since c2424b1f3.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit f37e66b393)
2022-01-10 12:42:51 +02:00
Niklas Haas fb840a79b9 lavfi/vf_libplacebo: fix side data stripping logic
This was accidentally comparing s->colorspace against out->colorspace,
which is wrong - the intent was to compare in->colorspace against
out->colorspace.

We also forgot to strip mastering metadata. Finally, the order is sort
of wrong - we should strip this side data *before* process_frames,
because otherwise it may end up being seen and used by libplacebo.

Signed-off-by: Niklas Haas <git@haasn.dev>
2022-01-10 09:11:18 +01:00
Andreas Rheinhardt 0acca1791e avformat/amr: Return error upon error
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit abc7d1c697)
2022-01-09 21:26:58 +01:00
Marc-Antoine Arnaud e53ab575da avformat/mxfdec: support MCA audio information
Channel reordering is removed from this patch because the new channel layout
API will support it properly.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 47c4df2203)
2022-01-09 18:28:18 +01:00
Ming Qian 832aae6c86 avcodec/v4l2_context: send start decode command after dynamic resolution change event
Fixes decoding of sample https://streams.videolan.org/ffmpeg/incoming/720p60.mp4
on RPi4 after kernel driver commit:
staging: bcm2835-codec: Format changed should trigger drain

Reference:
linux/Documentation/userspace-api/media/v4l/dev-decoder.rst
    "A source change triggers an implicit decoder drain, similar to the
     explicit Drain sequence. The decoder is stopped after it completes.
     The decoding process must be resumed with either a pair of calls to
     VIDIOC_STREAMOFF and VIDIOC_STREAMON on the CAPTURE queue, or a call to
     VIDIOC_DECODER_CMD with the V4L2_DEC_CMD_START command."

Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Ming Qian <ming.qian@nxp.com>
2022-01-09 11:47:26 -05:00
Ming Qian 91b459ab23 avcodec/v4l2_context: don't reinit output queue on dynamic resolution change event
Reference:
linux/Documentation/userspace-api/media/v4l/dev-decoder.rst
    "During the resolution change sequence, the OUTPUT queue must remain
     streaming. Calling VIDIOC_STREAMOFF() on the OUTPUT queue would
     abort the sequence and initiate a seek.

     In principle, the OUTPUT queue operates separately from the CAPTURE
     queue and this remains true for the duration of the entire
     resolution change sequence as well."

Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Ming Qian <ming.qian@nxp.com>
2022-01-09 11:47:09 -05:00
Gyan Doshi 1c2d1d988a avformat/hlsenc: convey stream id to segment streams 2022-01-09 10:38:04 +05:30
Andreas Rheinhardt 4eb182b714 avformat/matroskaenc: Disable MKV-only code if MKV muxer is disabled
The Matroska muxer has quite a lot of dependencies and lots of them
are unnecessary for WebM. By disabling the Matroska-only code
at compile time one can get rid of them.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 52c1e9e530)
2022-01-08 19:11:17 +01:00
Andreas Rheinhardt 201a486a11 fate/subtitles: Fix check for fate-binsub-mksenc test
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit e852b1b063)
2022-01-08 19:11:17 +01:00
Andreas Rheinhardt 48e85918b8 avformat/matroskaenc: Move AAC extradata check to other audio checks
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit d266bf1798)
2022-01-08 19:11:17 +01:00
Andreas Rheinhardt 60604702cd avdevice/iec61883: #if unused code away, fix -O0 compilation
iec61883_parse_queue_hdv() is only called when the mpegts-demuxer
is available and can be optimized away when not. Yet this
optimization is not a given and it fails with e.g. GCC 11 when
using -O0 in which case one will get a compilation error
because the call to the unavailable avpriv_mpegts_parse_packet()
is not optimized away. Therefore #if the offending code away
in this case.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit ad0b4afec5)
2022-01-08 19:11:17 +01:00
Andreas Rheinhardt fbbbe73b0f configure: Let decklink indev suggest libzvbi
Fixes build errors if libzvbi is enabled while libzvbi_teletextdec
is disabled.

Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 2d0b17e820)
2022-01-08 19:11:17 +01:00
Andreas Rheinhardt e9b7e781d7 avformat/rtsp: #if unused functions away, fix -O0 compilation
parse_rtsp_message() is only called if the rtsp demuxer is enabled
and so it is normally compiled away if said demuxer is disabled.
Yet this does not happen when compiling with -O0 and this leads
to a linking failure because parse_rtsp_message() calls functions
that may not be available if the rtsp demuxer is disabled.
Fix this by properly #if'ing the unused functions away.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit aeb5d943c6)
2022-01-08 19:11:17 +01:00
Andreas Rheinhardt 6d2317a611 avformat/Makefile: Add entries for CRI, GEM and PGX image pipe demuxers
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 99f3fb8ea1)
2022-01-08 19:01:10 +01:00
Andreas Rheinhardt 237268f6df avformat/Makefile: Add missing alp-muxer->rawenc.o dependency
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 1beeeaf23d)
2022-01-08 19:01:10 +01:00
Andreas Rheinhardt 519e22b31e avformat/amr: Fix writing AMR header
Regression since f282c34c00.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit a22a71eb2c)
2022-01-08 19:00:26 +01:00
Andreas Rheinhardt 24f1997822 avformat/amr: Don't reset AVFormatContext.priv_data
The AMR muxer doesn't have a private context, so it's priv_data
will be NULL. If it weren't, simply setting it to NULL would lead
to a memleak.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit a5a99cc11c)
2022-01-08 19:00:26 +01:00
Andreas Rheinhardt a20d6eb223 avformat/Makefile, amr: Add missing amr-demuxers->rawdec.o dependency
Forgotten in 1f447fd954.
Also only enable amr_probe() and amr_read_header() in case
the AMR demuxer is enabled; this avoids having to add
a rawdec.o dependency to the muxer.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 78a840e7a3)
2022-01-08 19:00:26 +01:00
Andreas Rheinhardt 359836ce6c avformat/Makefile: Add missing libamqp->urldecode dependency
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 5bdd8e67e6)
2022-01-08 19:00:26 +01:00
Fei Wang 9b099a97f0 lavc/av1dec: use frame split bsf
Split packed data in case of its contains multiple show frame in some
non-standard bitstream. This can benefit decoder which can decode
continuously instead of interrupt with unexpected error.

Signed-off-by: Fei Wang <fei.w.wang@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
(cherry picked from commit 7787cca180)
2022-01-08 11:09:17 -03:00
Martin Storsjö 6ff38630e1 aarch64: Disable ff_hevc_sao_band_filter_8x8_8_neon out of precaution
While this function on its own passes all of fate-hevc, there's
indications that the function might need to handle widths that
aren't a multiple of 8 (noted in commit
f63f9be37c, which later was
reverted).

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 24b93022fe)
2022-01-07 22:36:47 +02:00
Andreas Rheinhardt 60d6efe218 avformat/matroskaenc: Fix build with only WebM muxer enabled
In this case ff_isom_put_dvcc_dvvc() might not be available, leading
to linking failures. Given that WebM currently doesn't support DOVI,
this is fixed by #if'ing the offending code away if the Matroska
muxer is not enabled.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 217c90aac7)
2022-01-07 13:27:41 +01:00
Andreas Rheinhardt 3975f9328a configure: Add missing AMV muxer->riffenc dependency
Reviewed-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 41c62207f6)
2022-01-07 13:26:40 +01:00
Andreas Rheinhardt 743d26ca04 avformat/Makefile: Fix name of PhotoCD demuxer
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 6e163619e3)
2022-01-07 13:26:26 +01:00
James Almer c8df72fce5 avcodec/libdav1d: honor the requested strict_std_compliance level on supported builds
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 3e17e0e5ef)
2022-01-06 23:05:12 -03:00
Andreas Rheinhardt 761a65106b configure: Add missing libshine->mpegaudioheader dependency
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit e228d7b0db)
2022-01-06 08:31:12 +01:00
Andreas Rheinhardt f59e8666f9 avcodec/Makefile: Add missing entry for ADPCM_IMA_AMV_ENCODER
Forgotten in 555f5c1fc5.

Reviewed-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit df4cb384fb)
2022-01-06 08:31:12 +01:00
Andreas Rheinhardt b04b475917 avcodec/Makefile: Only compile nvenc.o if needed
This fixes compilation errors in case nvenc is enabled
(e.g. autodected) with both nvenc-based encoders disabled
because nvenc uses ff_alloc_a53_sei(), yet only the nvenc-based
encoders require atsc_a53.
(This error does not manifest itself in case of static linking
(nothing pulls in nvenc.o), but it exists with shared builds.)

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
2022-01-05 19:37:50 +01:00
Wu Jianhua eb091d211f avfilter/vf_blend: fix un-checked potential memory allocation failure
Signed-off-by: Wu Jianhua <jianhua.wu@intel.com>
2022-01-05 15:17:35 +01:00
Andreas Rheinhardt 85fca9f92b avcodec/Makefile: Add missing HEVC decoder->h274.o dependency
Forgotten in 3cc3f5de2a.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit aa8bb05d29)
2022-01-05 14:43:11 +01:00
Wu Jianhua b4d254f2e6 avutil/hwcontext_vulkan: fixed incorrect memory offset
This commit fixed hwupload in Vulkan:

ffmpeg -init_hw_device vulkan -i test.jpg -vf hwupload,hwdownload,format=yuv420p -y out.jpg

Signed-off-by: Wu Jianhua <jianhua.wu@intel.com>
2022-01-05 14:14:54 +01:00
James Almer b9dbb6c114 Changelog: replace <next> by 5.0
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 746df0a19a)
2022-01-05 08:39:53 -03:00
Zane van Iperen d4baad9fe3 Changelog: add IMF demuxer
Suggested-By: Pierre-Anthony Lemieux <pal@palemieux.com>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
(cherry picked from commit 21e25d2fe2)
2022-01-05 21:23:29 +10:00
Pierre-Anthony Lemieux 2c1dd39a98 avformat/imf: Fix indentation
Signed-off-by: Pierre-Anthony Lemieux <pal@palemieux.com>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
(cherry picked from commit 74afc3c650)
2022-01-05 21:23:28 +10:00
Pierre-Anthony Lemieux 8d18c08f0c avformat/imf: fix CPL parsing error handling
Signed-off-by: Pierre-Anthony Lemieux <pal@palemieux.com>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
(cherry picked from commit 76ffe1c2f1)
2022-01-05 21:23:27 +10:00
Pierre-Anthony Lemieux 2186555bc9 avformat/imf: fix bad free() when directory name of the input url is empty
Signed-off-by: Pierre-Anthony Lemieux <pal@palemieux.com>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
(cherry picked from commit 59f1a46048)
2022-01-05 21:23:26 +10:00
Pierre-Anthony Lemieux 5fd7b8e9b2 avformat/imf: fix error CPL root element is absent
Signed-off-by: Pierre-Anthony Lemieux <pal@palemieux.com>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
(cherry picked from commit c1b55cb70c)
2022-01-05 21:23:24 +10:00
Andreas Rheinhardt 696079924a avcodec/Makefile: Add missing mpegaudiodata.o dependency to MPEGAUDIO
mpegaudiodec_template.c uses stuff from mpegaudiodata directly,
yet this dependency was only indirectly fulfilled via mpegaudio-headers
before 33e6d57f01. Since this commit,
the latter only needs (and therefore provides) mpegaudiotabs,
leading to compilation failures.
This commit adds this missing direct dependency directly.
(Sorry for not having checked indirect dependencies.)

Found-by: Zane van Iperen <zane@zanevaniperen.com>
Reviewed-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 07fd34aca7)
2022-01-05 05:17:50 +01:00
Niklas Haas 660858a9e4 lavfi/libplacebo: support dovi metadata application
libplacebo supports automatic dolby vision application, but it requires
us to switch to a new API. Also add some logic to strip the dolby vision
metadata from the output frames in any case where we end up changing the
colorimetry.

The libplacebo dependency bump is justified because neither 184 nor 192
are part of any stable libplacebo release, so users have to build from
git anyways for this filter to exist.

Signed-off-by: Niklas Haas <git@haasn.dev>
2022-01-05 03:16:35 +01:00
rcombs 1477386ca2 lavfi/drawutils: re-enable P010 and P016 support
These formats now work as expected.
2022-01-04 20:01:10 -06:00
rcombs 052ee78d14 lavfi/drawutils: overhaul to improve pixel format support
- No longer mixes u8 and u16 component accesses (this was UB)
- De-duplicated 8->16 conversion
- De-duplicated component -> plane+offset conversion
- De-duplicated planar + packed RGB
- No longer calls ff_fill_rgba_map
- Removed redundant comp_mask data member
- RGB0 and related formats no longer write an alpha value to the 0 byte
- Non-planar YA formats now work correctly
- High-bit-depth semi-planar YUV now works correctly
2022-01-04 20:01:10 -06:00
rcombs 8688209176 lavfi/drawutils: ensure we don't support formats with non-pixel-sized offsets 2022-01-04 20:01:10 -06:00
rcombs 55ff7356bf lavfi/drawutils: ensure we can't overflow a component 2022-01-04 20:01:10 -06:00
rcombs c327743ee0 lavfi/drawutils: ensure we don't allow mixed-byte-depth formats
These could be hazardous because of FFDrawColor's union
2022-01-04 20:01:10 -06:00
rcombs 74210138a4 lavfi/drawutils: reimplement ff_fill_rgba_map without hardcoding the list
Same outputs, but computed instead of statically known, so new formats will be
supported more easily. Asserts in place to ensure we update this if we add
anything incompatible with its logic.
2022-01-04 20:01:10 -06:00
rcombs b059ded2a9 lavfi/drawutils: reject shift-packed formats
Disables x2bgr10/x2rgb10 (which did not behave correctly before).
2022-01-04 20:01:10 -06:00
rcombs 2641f7338d lavfi/drawutils: remove redundant BE format checks
We already explicitly don't support big-endian in general
2022-01-04 20:01:10 -06:00
rcombs 3cad1f756e lavfi/drawutils: move BE check out of loop 2022-01-04 20:01:10 -06:00
rcombs 9f7875c18e swscale/output: use isSwappedChroma 2022-01-04 20:01:10 -06:00
rcombs 656ffdbd28 swscale/output: use isSemiPlanarYUV for NV12/21/24/42 case 2022-01-04 20:01:10 -06:00
rcombs df94cdde4b swscale: introduce isSwappedChroma 2022-01-04 20:01:10 -06:00
rcombs 1292a8d91b swscale/output: use isDataInHighBits for 10-bit case
This code will need fleshing-out (probably templating) if we ever add
e.g. a P012 format.
2022-01-04 20:01:10 -06:00
rcombs 795f803a47 swscale/output: use isSemiPlanarYUV for 16-bit case 2022-01-04 20:01:10 -06:00
rcombs a5f1f0558f swscale: introduce isDataInHighBits 2022-01-04 20:01:10 -06:00
rcombs ff9bb93145 swscale/output: template-ize yuv2nv12cX 10-bit and 16-bit cases
Fixes incorrect big-endian output introduced in 88d804b7ff

Avoids making the filter-time BE check more expensive
2022-01-04 20:01:10 -06:00
Limin Wang f582e40595 avcodec/videotoolbox: Fix undefined symbol with minimal configuration
Please reproduced with the following minimal configure command:
./configure --enable-shared --disable-all --enable-avcodec --enable-decoder=h264 --enable-hwaccel=h264_videotoolbox

You'll get below error:

Undefined symbols for architecture x86_64:
  "_ff_videotoolbox_vpcc_extradata_create", referenced from:
      _videotoolbox_start in videotoolbox.o
ld: symbol(s) not found for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)

Reported-by: Cameron Gutman <aicommander@gmail.com>
Tested-by: Cameron Gutman <aicommander@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2022-01-05 09:34:22 +08:00
Niklas Haas a02d3054ea lavfi/showinfo: fix printf precision for dovi metadata
Fix warning caused by this field changing from uint64_t to uint16_t.

Signed-off-by: Niklas Haas <git@haasn.dev>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 10e4b2b1d2)
2022-01-04 14:43:53 +01:00
Andreas Rheinhardt 312060ecfc Merge branch 'master' into release/5.0
This is necessary to have the recent DOVI additions
in the 5.0 release.

Merged-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-01-04 14:26:04 +01:00
Pierre-Anthony Lemieux 311ea9c529 avformat/imf: Fix error handling in set_context_streams_from_tracks()
Signed-off-by: Pierre-Anthony Lemieux <pal@palemieux.com>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
(cherry picked from commit 4c03928f4d)
2022-01-04 19:40:32 +10:00
Andreas Rheinhardt b01ac2d863 avformat/tests/imf: Don't use uninitialized value
Reviewed-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(cherry picked from commit 20b0b2be6c)
2022-01-04 10:37:53 +01:00
Michael Niedermayer 1f447cf9c4 Update versions for 5.0
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2022-01-03 22:21:12 +01:00
7806 changed files with 344595 additions and 797254 deletions
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# This file describes the expected reviewers for a PR based on the changed
# files. Unlike what the name of the file suggests they don't own the code, but
# merely have a good understanding of that area of the codebase and therefore
# are usually suited as a reviewer.
# Lines in this file match changed paths via Go-Style regular expressions:
# https://pkg.go.dev/regexp/syntax
# Mind the alphabetical order
# avcodec
# =======
libavcodec/.*aac.* @lynne
libavcodec/.*ac3.* @lynne
libavcodec/.*adpcm.* @zane @pross
libavcodec/anm.* @pross
libavcodec/amf.* @OvchinnikovDmitrii @ArazIusubov
libavcodec/ansi.* @pross
libavcodec/aom_film_grain.* @haasn
libavcodec/.*atrac9.* @lynne
libavcodec/bink.* @pross
libavcodec/bintext.* @pross
libavcodec/.*bitpacked.* @lynne
libavcodec/.*d3d12va.* @jianhuaw @tong1wu @younengxiao
libavcodec/.*dirac.* @lynne
libavcodec/.*dovi_rpu.* @haasn
libavcodec/dpx.* @pross
libavcodec/dsd.* @pross
libavcodec/eacmv.* @pross
libavcodec/eaidct.* @pross
libavcodec/eamad.* @pross
libavcodec/eat.* @pross
libavcodec/.*exif.* @Traneptora
libavcodec/.*ffv1.* @lynne @michaelni
libavcodec/g728.* @pross
libavcodec/gem.* @pross
libavcodec/golomb.* @michaelni
libavcodec/.*h266.* @frankplow @NuoMi @jianhuaw
libavcodec/h26x/.* @frankplow @NuoMi @jianhuaw
libavcodec/.*h274.* @haasn
libavcodec/iff.* @pross
libavcodec/.*jpegxl.* @lynne @Traneptora
libavcodec/jpeg2000htdec.* @cae
libavcodec/jv.* @pross
libavcodec/.*jxl.* @lynne @Traneptora
libavcodec/.*lcms2.* @haasn
libavcodec/lead.* @pross
libavcodec/mediacodec* @quink
libavcodec/mjpeg* @michaelni
libavcodec/mmvideo.* @pross
libavcodec/msp2.* @pross
libavcodec/mvc.* @pross
libavcodec/oh* @quink
libavcodec/.*opus.* @lynne
libavcodec/pictor.* @pross
libavcodec/.*png.* @Traneptora
libavcodec/.*prores.* @lynne
libavcodec/rangecoder.* @michaelni
libavcodec/ratecontrol.* @michaelni
libavcodec/rkmpp* @quink
libavcodec/rv60.* @pross
libavcodec/sgirle.* @pross
libavcodec/.*siren.* @lynne
libavcodec/smpte_436m.* @programmerjake
libavcodec/svq1.* @pross
libavcodec/svq3.* @pross
libavcodec/.*vc2.* @lynne
libavcodec/videotoolbox.* @ePirat
libavcodec/vp3.* @pross
libavcodec/vp4.* @pross
libavcodec/vp5.* @pross
libavcodec/vp6.* @pross
libavcodec/vp8.* @rbultje @pross
libavcodec/vp9.* @rbultje
libavcodec/vpx.* @rbultje @pross
libavcodec/vqc.* @pross
libavcodec/.*vvc.* @frankplow @NuoMi @jianhuaw
libavcodec/wmavoice.* @rbultje
libavcodec/wbmp.* @pross
# bitstream filters
libavcodec/bsf/eia608_to_smpte436m.* @programmerjake
libavcodec/bsf/smpte436m_to_eia608.* @programmerjake
# architecture-specific
libavcodec/aarch64/.* @lynne @mstorsjo
libavcodec/arm/.* @mstorsjo
libavcodec/ppc/.* @sean_mcg
libavcodec/riscv/.* @Courmisch
libavcodec/wasm/hevc/.* @quink
libavcodec/x86/.* @lynne
libavcodec/x86/vp8.* @rbultje
libavcodec/x86/vp9.* @rbultje
libavcodec/x86/vpx.* @rbultje
# avfilter
# =======
libavfilter/af_loudnorm.* @kylophone
libavfilter/af_whisper.* @vpalmisano
libavfilter/.*_amf* @OvchinnikovDmitrii @ArazIusubov
libavfilter/avfiltergraph.* @haasn
libavfilter/colorspace.* @rbultje
libavfilter/formats.* @haasn
libavfilter/.*f_ebur128.* @haasn
libavfilter/vf_blackdetect.* @haasn
libavfilter/vf_colordetect.* @haasn
libavfilter/vf_colorspace.* @rbultje
libavfilter/.*drawvg.* @ayosec
libavfilter/vf_find_rect.* @michaelni
libavfilter/vf_icc.* @haasn
libavfilter/vf_libplacebo.* @haasn
libavfilter/vf_libvmaf.* @kylophone
libavfilter/vf_mpdecimate.* @dana-feng
libavfilter/vf_premultiply.* @haasn
libavfilter/vf_scale.* @haasn
libavfilter/vf_scale_vt.* @quink
libavfilter/vf_thumbnail.* @haasn
libavfilter/vf_transpose_vt.* @quink
libavfilter/vf_yadif.* @michaelni
libavfilter/vsrc_mandelbrot.* @michaelni
libavfilter/aarch64/.* @mstorsjo
libavfilter/riscv/.* @Courmisch
libavfilter/x86/colorspace.* @rbultje
libavfilter/x86/scene_sad.* @haasn
# avformat
# =======
libavformat/alp.* @zane
libavformat/amv.* @zane
libavformat/anm.* @pross
libavformat/apm.* @zane
libavformat/argo_.* @zane
libavformat/bink.* @pross
libavformat/bintext.* @pross
libavformat/caf.* @pross
libavformat/cine.* @pross
libavformat/dsf.* @pross
libavformat/eacdata.* @pross
libavformat/electronicarts.* @pross
libavformat/.*exif.* @Traneptora
libavformat/filmstrip.* @pross
libavformat/frm.* @pross
libavformat/hls.* @kasper93
libavformat/hxvs.* @quink
libavformat/iamf.* @jamrial
libavformat/icecast.c @ePirat
libavformat/ico.* @pross
libavformat/iff.* @pross
libavformat/.*jpegxl.* @Traneptora
libavformat/jv.* @pross
libavformat/.*jxl.* @Traneptora
libavformat/kvag.* @zane
libavformat/mccdec.* @programmerjake
libavformat/mccenc.* @programmerjake
libavformat/mlv.* @pross
libavformat/mm.* @pross
libavformat/msp.* @pross
libavformat/mv.* @pross
libavformat/ogg.* @toots
libavformat/pp_bnk.* @zane
libavformat/rm.* @pross
libavformat/sauce.* @pross
libavformat/scd.* @zane
libavformat/tty.* @pross
libavformat/whip.c @JackLau
libavformat/wsd.* @pross
libavformat/wtv.* @pross
# avutil
# ======
libavutil/.*_amf* @OvchinnikovDmitrii @ArazIusubov
libavutil/.*crc.* @lynne @michaelni
libavutil/.*d3d12va.* @jianhuaw @tong1wu @younengxiao
libavutil/csp.* @rbultje @haasn
libavutil/eval.* @michaelni
libavutil/film_grain.* @haasn
libavutil/dovi_meta.* @haasn
libavutil/hwcontext_oh.* @quink
libavutil/hwcontext_mediacodec.* @quink
libavutil/hwcontext_videotoolbox.* @ePirat
libavutil/iamf.* @jamrial
libavutil/integer.* @michaelni
libavutil/lfg.* @michaelni
libavutil/lls.* @michaelni
libavutil/md5.* @michaelni
libavutil/mathematics.* @michaelni
libavutil/mem.* @michaelni
libavutil/qsort.* @michaelni
libavutil/random_seed.* @michaelni
libavutil/rational.* @michaelni
libavutil/sfc.* @michaelni
libavutil/softfloat.* @michaelni
libavutil/tree.* @michaelni
libavutil/tx.* @lynne
libavutil/aarch64/.* @lynne @mstorsjo
libavutil/arm/.* @mstorsjo
libavutil/ppc/.* @sean_mcg
libavutil/riscv/.* @Courmisch
libavutil/wasm/.* @quink
libavutil/x86/.* @lynne
# swresample
# =======
libswresample/aarch64/.* @mstorsjo
libswresample/arm/.* @mstorsjo
libswresample/.* @michaelni
# swscale
# =======
libswscale/aarch64/.* @mstorsjo
libswscale/arm/.* @mstorsjo
libswscale/ppc/.* @sean_mcg
libswscale/riscv/.* @Courmisch
libswscale/.* @haasn
# tools
# =====
fftools/ffplay_renderer.* @haasn
tools/target_dec_fuzzer.c @michaelni
# doc
# ===
doc/.* @GyanD
# Frameworks
# ==========
.*d3d12va.* @jianhuaw @tong1wu @younengxiao
.*vulkan.* @lynne @haasn
# tests
# =====
tests/checkasm/riscv/.* @Courmisch
libavutil/tests/ambient_viewing_environment.* @MarcosAsh
libavutil/tests/buffer.* @MarcosAsh
libavutil/tests/csp.* @MarcosAsh
libavutil/tests/dovi_meta.* @MarcosAsh
libavutil/tests/hdr_dynamic_vivid_metadata.* @MarcosAsh
libavutil/tests/mastering_display_metadata.* @MarcosAsh
libavutil/tests/tdrdi.* @MarcosAsh
libavutil/tests/timestamp.* @MarcosAsh
libavutil/tests/video_hint.* @MarcosAsh
tests/ref/.*drawvg.* @ayosec
tests/ref/fate/ambient_viewing_environment @MarcosAsh
tests/ref/fate/buffer @MarcosAsh
tests/ref/fate/csp @MarcosAsh
tests/ref/fate/dovi_meta @MarcosAsh
tests/ref/fate/hdr_dynamic_vivid_metadata @MarcosAsh
tests/ref/fate/mastering_display_metadata @MarcosAsh
tests/ref/fate/sub-mcc.* @programmerjake
tests/ref/fate/tdrdi @MarcosAsh
tests/ref/fate/timestamp @MarcosAsh
tests/ref/fate/video_hint @MarcosAsh
# Forgejo
# =======
.forgejo/CODEOWNERS @michaelni
# /
# =
MAINTAINERS @michaelni
-9
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@@ -1,9 +0,0 @@
# Summary of the bug
Briefly describe the issue you're experiencing. Include any error messages, unexpected behavior, or relevant observations.
# Steps to reproduce
List the steps required to trigger the bug.
Include the exact CLI command used, if any.
Provide sample input files, logs, or scripts if available.
-14
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@@ -1,14 +0,0 @@
# Summary of changes
Briefly describe what this PR does and why.
<!--
If this PR requires new FATE test samples, attach them to the PR and
list their target paths below (relative to the fate-suite root).
Attached filenames must match the sample's filename:
```fate-samples
# e.g. vorbis/new-sample.ogg
```
-->
-40
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@@ -1,40 +0,0 @@
name: Rebase PR onto target
inputs:
path:
description: Directory to run in
required: false
default: .
runs:
using: composite
steps:
- name: Rebase onto target branch
if: ${{ forge.event_name == 'pull_request' }}
working-directory: ${{ inputs.path }}
run: |
set -x
# Fetch all PR commits + merge-base
commits=$(curl -s -o /dev/null -w '%header{X-Total-Count}' '${{ forge.api_url }}/repos/${{ forge.repository }}/pulls/${{ forge.event.pull_request.number }}/commits?limit=1&verification=false&files=false')
git fetch -vv --deepen=$commits origin pull/${{ forge.event.pull_request.number }}/head
if [ "$(wc -l < .git/shallow)" -ne 1 ]; then
echo "Complex PR history detected, falling back to merge"
git fetch -vv --unshallow origin pull/${{ forge.event.pull_request.number }}/head
git fetch -vv origin ${{ forge.base_ref }}
git merge --no-edit origin/${{ forge.base_ref }} \
|| { echo "Does not merge cleanly with ${{ forge.base_ref }}"; exit 1; }
exit 0
fi
# Unfortunately, Forgejo does not update the merge-base when PR branch
# is updated, the `forge.event.pull_request.merge_base` is stale value,
# from the time when PR was created. We cannot rely on it, so try to
# infer parent commit of the PR from the shallow clone that we have.
base=$(cat .git/shallow)
# Fetch target branch
git fetch -vv --depth=1 origin ${{ forge.base_ref }}
# Rebase PR commits onto the current state of target branch
git rebase --onto origin/${{ forge.base_ref }} $base \
|| { echo "Does not rebase cleanly onto ${{ forge.base_ref }}"; exit 1; }
-174
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@@ -1,174 +0,0 @@
#!/usr/bin/env python3
# Copyright (c) 2026 Romain Beauxis <romain.beauxis@gmail.com>
#
# Redistribution and use in source and binary forms, with or without
# modification, are permitted provided that the following conditions are met:
#
# 1. Redistributions of source code must retain the above copyright notice,
# this list of conditions and the following disclaimer.
# 2. Redistributions in binary form must reproduce the above copyright notice,
# this list of conditions and the following disclaimer in the documentation
# and/or other materials provided with the distribution.
#
# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
# AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
# ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
# LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
# CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
# SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
# INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
# CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
# ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
# POSSIBILITY OF SUCH DAMAGE.
"""Inject PR attachment samples into the fate-suite directory.
Usage: inject-pr-samples.py <pr-number>
Reads SAMPLES from the environment (defaults to fate-suite). For each path
listed in a ```fate-samples``` block in the PR description, downloads the
matching PR attachment into $SAMPLES/<path>.
The PR description should contain a block like:
```fate-samples
vorbis/tos.ogg
mov/some-new-sample.mov
```
Each filename must match a file attached to the PR.
"""
import hashlib
import json
import os
import re
import sys
import tempfile
import urllib.request
from pathlib import Path, PurePosixPath
FORGEJO_API = "https://code.ffmpeg.org/api/v1/repos/ffmpeg/ffmpeg/issues"
ATTACHMENT_BASE = "https://code.ffmpeg.org/attachments/"
def fetch_json(url):
with urllib.request.urlopen(url) as r:
return json.load(r)
def parse_fate_samples(body):
paths = []
in_block = False
for line in body.splitlines():
if line == "```fate-samples":
in_block = True
elif line == "```" and in_block:
break
elif in_block:
parts = line.split()
if len(parts) == 1:
paths.append(parts[0])
return paths
MAX_PATH_DEPTH = 3
def validate_path(path):
p = PurePosixPath(path)
if p.is_absolute():
raise ValueError(f"path must be relative: {path!r}")
if ".." in p.parts:
raise ValueError(f"path must not contain '..': {path!r}")
if not p.parts:
raise ValueError(f"empty path")
if len(p.parts) > MAX_PATH_DEPTH:
raise ValueError(f"path too deep (max {MAX_PATH_DEPTH} components): {path!r}")
def validate_url(url):
if not url.startswith(ATTACHMENT_BASE):
raise ValueError(f"unexpected attachment URL: {url!r}")
def digest(path):
h = hashlib.sha256()
with open(path, "rb") as f:
while chunk := f.read(1 << 16):
h.update(chunk)
return h.digest()
def download(url, dst):
dst.parent.mkdir(parents=True, exist_ok=True)
with tempfile.NamedTemporaryFile(dir=dst.parent, delete=False) as tmp:
tmp_path = Path(tmp.name)
try:
with urllib.request.urlopen(url) as r:
while chunk := r.read(1 << 16):
tmp.write(chunk)
if dst.exists() and digest(dst) != digest(tmp_path):
raise ValueError(f"already exists with different content: {dst}")
tmp_path.rename(dst)
except:
tmp_path.unlink(missing_ok=True)
raise
def main():
if len(sys.argv) != 2 or not re.fullmatch(r"[0-9]+", sys.argv[1]):
print(f"Usage: {sys.argv[0]} <pr-number>", file=sys.stderr)
sys.exit(1)
pr_number = sys.argv[1]
samples_dir = Path(os.environ.get("SAMPLES", "fate-suite"))
pr = fetch_json(f"{FORGEJO_API}/{pr_number}")
assets = {a["name"]: a["browser_download_url"] for a in pr.get("assets", [])}
paths = parse_fate_samples(pr.get("body", ""))
if not paths:
sys.exit(0)
new_samples = False
for path in paths:
try:
validate_path(path)
except ValueError as e:
print(f"fate-samples: {e}", file=sys.stderr)
sys.exit(1)
name = PurePosixPath(path).name
url = assets.get(name)
if url is None:
print(f"fate-samples: no attachment named {name!r}", file=sys.stderr)
sys.exit(1)
try:
validate_url(url)
except ValueError as e:
print(f"fate-samples: {e}", file=sys.stderr)
sys.exit(1)
dst = samples_dir / path
is_new = not dst.exists()
try:
download(url, dst)
except ValueError as e:
print(f"fate-samples: {e}", file=sys.stderr)
sys.exit(1)
if is_new:
new_samples = True
print(f"Injected: {path}")
output_file = os.environ.get("FORGEJO_OUTPUT")
if output_file:
with open(output_file, "a") as f:
print(f"new_samples={'true' if new_samples else 'false'}", file=f)
if __name__ == "__main__":
main()
-74
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@@ -1,74 +0,0 @@
module.exports = async ({github, context}) => {
const title = (context.payload.pull_request?.title || context.payload.issue?.title || '').toLowerCase();
const labels = [];
const issueNumber = context.payload.pull_request?.number || context.payload.issue?.number;
const kwmap = {
'avcodec': 'avcodec',
'avdevice': 'avdevice',
'avfilter': 'avfilter',
'avformat': 'avformat',
'avutil': 'avutil',
'swresample': 'swresample',
'swscale': 'swscale',
'fftools': 'CLI',
'vulkan': 'vulkan'
};
async function isOrgMember(username) {
try {
const response = await github.rest.orgs.checkMembershipForUser({
org: context.repo.owner,
username: username
});
return response.status === 204;
} catch (error) {
return false;
}
}
if (context.payload.action === 'closed' ||
(context.payload.action !== 'opened' && (
context.payload.action === 'assigned' ||
context.payload.action === 'label_updated' ||
context.payload.action === 'labeled' ||
context.payload.comment) &&
await isOrgMember(context.payload.sender.login))
) {
try {
await github.rest.issues.removeLabel({
owner: context.repo.owner,
repo: context.repo.repo,
issue_number: issueNumber,
// this should say 'new', but forgejo deviates from GitHub API here and expects the ID
name: '41'
});
console.log('Removed "new" label');
} catch (error) {
if (error.status !== 404 && error.status !== 410) {
console.log('Could not remove "new" label');
}
}
} else if (context.payload.action === 'opened') {
labels.push('new');
console.log('Detected label: new');
}
if ((context.payload.action === 'opened' || context.payload.action === 'edited') && context.eventName !== 'issue_comment') {
for (const [kw, label] of Object.entries(kwmap)) {
if (title.includes(kw)) {
labels.push(label);
console.log('Detected label: ' + label);
}
}
}
if (labels.length > 0) {
await github.rest.issues.addLabels({
owner: context.repo.owner,
repo: context.repo.repo,
issue_number: issueNumber,
labels: labels,
});
}
}
-35
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@@ -1,35 +0,0 @@
avcodec:
- changed-files:
- any-glob-to-any-file: 'libavcodec/**'
avdevice:
- changed-files:
- any-glob-to-any-file: 'libavdevice/**'
avfilter:
- changed-files:
- any-glob-to-any-file: 'libavfilter/**'
avformat:
- changed-files:
- any-glob-to-any-file: 'libavformat/**'
avutil:
- changed-files:
- any-glob-to-any-file: 'libavutil/**'
swresample:
- changed-files:
- any-glob-to-any-file: 'libswresample/**'
swscale:
- changed-files:
- any-glob-to-any-file: 'libswscale/**'
CLI:
- changed-files:
- any-glob-to-any-file: 'fftools/**'
vulkan:
- changed-files:
- any-glob-to-any-file: '**/*vulkan*'
-51
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@@ -1,51 +0,0 @@
exclude: ^tests/ref/
# This repository keeps its pre-commit config in this nondefault path.
# Install hooks from here with:
# pre-commit install -c .forgejo/pre-commit/config.yaml
# The command above installs both the pre-commit and commit-msg hooks because
# default_install_hook_types includes them below.
default_install_hook_types:
- pre-commit
- commit-msg
repos:
- repo: https://github.com/pre-commit/pre-commit-hooks
rev: v5.0.0
hooks:
- id: check-case-conflict
- id: check-executables-have-shebangs
- id: check-illegal-windows-names
- id: check-shebang-scripts-are-executable
- id: check-yaml
- id: end-of-file-fixer
- id: file-contents-sorter
files:
.forgejo/pre-commit/ignored-words.txt
args:
- --ignore-case
- id: fix-byte-order-marker
- id: mixed-line-ending
- id: trailing-whitespace
- repo: local
hooks:
- id: check-commit-message
name: validate commit message format
language: script
entry: ./tools/check_commit_msg.sh
stages: [commit-msg]
always_run: true
- id: arm-asm-indent
name: fix arm/aarch64 assembly indentation
files: ^.*/(arm|aarch64)/.*\.S$
language: script
entry: ./tools/check_arm_indent.sh --apply
pass_filenames: false
- repo: https://github.com/codespell-project/codespell
rev: v2.4.1
hooks:
- id: codespell
args:
- --ignore-words=.forgejo/pre-commit/ignored-words.txt
- --ignore-multiline-regex=codespell:off.*?(codespell:on|\Z)
exclude: ^tools/(patcheck|clean-diff)|tests/checkasm/ext/.*$
-120
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@@ -1,120 +0,0 @@
abl
ACN
acount
addin
alis
alls
ALOG
ALS
als
ANC
anc
ANS
ans
anull
basf
bloc
brane
BREIF
BU
bu
bufer
CAF
caf
clen
clens
Collet
compre
dum
endin
erro
FIEL
fiel
filp
fils
FILTERD
filterd
fle
fo
FPR
fro
Hald
indx
ine
inh
inout
inouts
inport
ist
LAF
laf
lastr
LinS
mapp
mis
mot
nd
nIn
offsetp
orderd
ot
outout
padd
PAETH
paeth
PARM
parm
parms
pEvents
PixelX
Psot
quater
re-use
readd
recuse
redY
Reencode
reencode
remaind
renderD
rin
SAV
SEH
SER
ser
setts
shft
SIZ
siz
skipd
sme
som
sover
STAP
startd
statics
struc
suble
TE
tE
te
tha
tne
tolen
tpye
tre
TRUN
trun
truns
Tung
TYE
ue
UES
ues
vai
vas
vie
VILL
vor
wel
wih
-32
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@@ -1,32 +0,0 @@
name: Autolabel
on:
pull_request_target:
types: [opened, edited, synchronize, closed, assigned, labeled, unlabeled]
issues:
types: [opened, edited, closed, assigned, labeled, unlabeled]
issue_comment:
types: [created]
jobs:
pr_labeler:
name: Labeler
runs-on: utilities
if: ${{ github.event.sender.login != 'ffmpeg-devel' }}
steps:
- name: Checkout
uses: actions/checkout@v6
- name: Label by file-changes
uses: actions/labeler@v6
if: ${{ forge.event_name == 'pull_request_target' }}
with:
configuration-path: .forgejo/labeler/labeler.yml
repo-token: ${{ secrets.AUTOLABELER_TOKEN }}
sync-labels: true
- name: Label by title-match
uses: actions/github-script@v8
with:
script: |
const script = require('.forgejo/labeler/labeler.js')
await script({github, context})
github-token: ${{ secrets.AUTOLABELER_TOKEN }}
-34
View File
@@ -1,34 +0,0 @@
name: Lint
on:
push:
branches:
- master
pull_request:
concurrency:
cancel-in-progress: ${{ forge.event_name == 'pull_request' }}
jobs:
lint:
name: Pre-Commit
runs-on: utilities
steps:
- name: Checkout
uses: actions/checkout@v6
- name: Rebase onto target branch
uses: ./.forgejo/actions/rebase-pr
- name: Install pre-commit CI
id: install
run: |
python3 -m venv ~/pre-commit
~/pre-commit/bin/pip install --upgrade pip setuptools
~/pre-commit/bin/pip install pre-commit
echo "envhash=$({ python3 --version && cat .forgejo/pre-commit/config.yaml; } | sha256sum | cut -d' ' -f1)" >> $FORGEJO_OUTPUT
- name: Cache
uses: actions/cache@v5
with:
path: ~/.cache/pre-commit
key: pre-commit-${{ steps.install.outputs.envhash }}
- name: Run pre-commit CI
run: ~/pre-commit/bin/pre-commit run -c .forgejo/pre-commit/config.yaml --show-diff-on-failure --color=always --all-files
-145
View File
@@ -1,145 +0,0 @@
name: Test
on:
push:
branches:
- master
pull_request:
concurrency:
cancel-in-progress: ${{ forge.event_name == 'pull_request' }}
jobs:
run_fate:
name: Fate (${{ matrix.runner }}, ${{ matrix.shared }}, ${{ matrix.bits }} bit)
strategy:
fail-fast: false
matrix:
runner: [linux-aarch64]
shared: ['static']
bits: ['64']
include:
- runner: linux-amd64
shared: 'static'
bits: '32'
- runner: linux-amd64
shared: 'shared'
bits: '64'
runs-on: ${{ matrix.runner }}
steps:
- name: Checkout
uses: actions/checkout@v6
- name: Rebase onto target branch
uses: ./.forgejo/actions/rebase-pr
- name: Configure
run: |
./configure --enable-gpl --enable-nonfree --enable-memory-poisoning --assert-level=2 \
$([ "${{ matrix.bits }}" != "32" ] || echo --arch=x86_32 --extra-cflags=-m32 --extra-cxxflags=-m32 --extra-ldflags=-m32) \
$([ "${{ matrix.shared }}" != "shared" ] || echo --enable-shared --disable-static) \
|| CFGRES=$?
cat ffbuild/config.log
exit ${CFGRES:-0}
- name: Build
run: make -j$(nproc)
- name: Restore Cached Fate-Suite
id: cache
uses: actions/cache/restore@v5
with:
path: fate-suite
key: fate-suite
restore-keys: |
fate-suite-
- name: Sync Fate-Suite
id: fate
run: |
make fate-rsync SAMPLES=$PWD/fate-suite
echo "hash=$(find fate-suite -type f -printf "%P %s %T@\n" | sort | sha256sum | cut -d' ' -f1)" >> $FORGEJO_OUTPUT
- name: Cache Fate-Suite
uses: actions/cache/save@v5
if: ${{ format('fate-suite-{0}', steps.fate.outputs.hash) != steps.cache.outputs.cache-matched-key }}
with:
path: fate-suite
key: fate-suite-${{ steps.fate.outputs.hash }}
- name: Inject PR Samples
id: inject
if: ${{ forge.event_name == 'pull_request' }}
run: SAMPLES=$PWD/fate-suite .forgejo/inject-pr-samples.py ${{ forge.event.pull_request.number }}
- name: Run Fate
run: |
LD_LIBRARY_PATH="$(printf "%s:" "$PWD"/lib*)$PWD" make fate fate-build SAMPLES="$PWD/fate-suite" -j$(nproc) || FATERES=$?
find . -name "*.err" -exec printf '::group::%s\n' {} \; -exec cat {} \; -exec printf '::endgroup::\n' \;
exit ${FATERES:-0}
- name: Fail if new samples were injected
if: ${{ steps.inject.outputs.new_samples == 'true' }}
run: |
echo "New FATE samples were injected from PR attachments. Please add them to the official fate-suite before merging."
exit 1
run_fate_full:
name: Fate (Full, ${{ matrix.target_exec }})
strategy:
fail-fast: false
matrix:
image: ['ghcr.io/btbn/ffmpeg-builds/win64-gpl:latest']
target_exec: ['wine']
runs-on: linux-amd64
container: ${{ matrix.image }}
steps:
- name: Checkout
uses: actions/checkout@v6
with:
path: ffmpeg
- name: Rebase onto target branch
uses: ./ffmpeg/.forgejo/actions/rebase-pr
with:
path: ffmpeg
- name: Configure
run: |
mkdir build && cd build
../ffmpeg/configure --pkg-config-flags="--static" $FFBUILD_TARGET_FLAGS $FF_CONFIGURE \
--cc="$CC" --cxx="$CXX" --ar="$AR" --ranlib="$RANLIB" --nm="$NM" \
--extra-cflags="$FF_CFLAGS" --extra-cxxflags="$FF_CXXFLAGS" \
--extra-libs="$FF_LIBS" --extra-ldflags="$FF_LDFLAGS" --extra-ldexeflags="$FF_LDEXEFLAGS" \
--enable-memory-poisoning --assert-level=2 --target-exec="${{ matrix.target_exec }}" \
--enable-hardcoded-tables \
|| CFGRES=$?
cat ffbuild/config.log
exit ${CFGRES:-0}
- name: Build
run: make -C build -j$(nproc)
- name: Restore Cached Fate-Suite
id: cache
uses: actions/cache/restore@v5
with:
path: fate-suite
key: fate-suite
restore-keys: |
fate-suite-
- name: Sync Fate-Suite
id: fate
run: |
make -C build fate-rsync SAMPLES=$PWD/fate-suite
echo "hash=$(find fate-suite -type f -printf "%P %s %T@\n" | sort | sha256sum | cut -d' ' -f1)" >> $FORGEJO_OUTPUT
- name: Cache Fate-Suite
uses: actions/cache/save@v5
if: ${{ format('fate-suite-{0}', steps.fate.outputs.hash) != steps.cache.outputs.cache-matched-key }}
with:
path: fate-suite
key: fate-suite-${{ steps.fate.outputs.hash }}
- name: Inject PR Samples
id: inject
if: ${{ forge.event_name == 'pull_request' }}
run: SAMPLES=$PWD/fate-suite ffmpeg/.forgejo/inject-pr-samples.py ${{ forge.event.pull_request.number }}
- name: Run Fate
run: |
if [[ "${{ matrix.target_exec }}" == "wine" ]]; then
export WINEDEBUG=-all
export FREI0R_PATH="$(winepath -w "$FREI0R_PATH")"
fi
LD_LIBRARY_PATH="$(printf "%s:" "$PWD"/lib*)$PWD" make -C build fate fate-build SAMPLES="$PWD/fate-suite" -j$(nproc) || FATERES=$?
find . -name "*.err" -exec printf '::group::%s\n' {} \; -exec cat {} \; -exec printf '::endgroup::\n' \;
exit ${FATERES:-0}
- name: Fail if new samples were injected
if: ${{ steps.inject.outputs.new_samples == 'true' }}
run: |
echo "New FATE samples were injected from PR attachments. Please add them to the official fate-suite before merging."
exit 1
+1 -1
View File
@@ -1,2 +1,2 @@
*.pnm -diff -text
Changelog merge=union
tests/ref/fate/sub-scc eol=crlf
+1 -16
View File
@@ -1,6 +1,5 @@
*.a
*.o
*.objs
*.o.*
*.d
*.def
@@ -23,12 +22,6 @@
*.metal.air
*.metallib
*.metallib.c
*.spv
*.spv.c
*.spv.gz
*.gen.asm
*.gen.c
*.gen.S
*.ptx
*.ptx.c
*.ptx.gz
@@ -40,19 +33,11 @@
/ffmpeg
/ffplay
/ffprobe
/checkasm_config_generated.asm
/checkasm_config_generated.h
/checkasm_header_config_generated.h
/config.asm
/config.h
/config_components.asm
/config_components.h
/coverage.info
/avversion.h
/lcov/
/src
/mapfile
/tools/python/__pycache__/
/libavcodec/vulkan/*.c
/libavfilter/vulkan/*.c
/.*/
!/.forgejo/
+5 -10
View File
@@ -1,8 +1,9 @@
<james.darnley@gmail.com> <jdarnley@obe.tv>
<jeebjp@gmail.com> <jan.ekstrom@aminocom.com>
<sw@jkqxz.net> <mrt@jkqxz.net>
<u@pkh.me> <cboesch@gopro.com>
<quinkblack@foxmail.com> <wantlamy@gmail.com>
<quinkblack@foxmail.com> <zhilizhao@tencent.com>
<zhilizhao@tencent.com> <quinkblack@foxmail.com>
<zhilizhao@tencent.com> <wantlamy@gmail.com>
<modmaker@google.com> <modmaker-at-google.com@ffmpeg.org>
<stebbins@jetheaddev.com> <jstebbins@jetheaddev.com>
<barryjzhao@tencent.com> <mypopydev@gmail.com>
@@ -17,14 +18,8 @@
<atomnuker@gmail.com> <rpehlivanov@obe.tv>
<lizhong1008@gmail.com> <zhong.li@intel.com>
<lizhong1008@gmail.com> <zhongli_dev@126.com>
<andreas.rheinhardt@outlook.com> <andreas.rheinhardt@gmail.com>
<andreas.rheinhardt@outlook.com> <andreas.rheinhardt@googlemail.com>
<andreas.rheinhardt@gmail.com> <andreas.rheinhardt@googlemail.com>
rcombs <rcombs@rcombs.me> <rodger.combs@gmail.com>
<thilo.borgmann@mail.de> <thilo.borgmann@googlemail.com>
<lq@chinaffmpeg.org> <liuqi05@kuaishou.com>
<liuqi05@kuaishou.com> <lq@chinaffmpeg.org>
<ruiling.song83@gmail.com> <ruiling.song@intel.com>
Cosmin Stejerean <cosmin@cosmin.at> Cosmin Stejerean via ffmpeg-devel <ffmpeg-devel@ffmpeg.org>
<wutong1208@outlook.com> <tong1.wu-at-intel.com@ffmpeg.org>
<wutong1208@outlook.com> <tong1.wu@intel.com>
<toqsxw@outlook.com> <jianhua.wu-at-intel.com@ffmpeg.org>
<toqsxw@outlook.com> <jianhua.wu@intel.com>
+30
View File
@@ -0,0 +1,30 @@
language: c
sudo: false
os:
- linux
- osx
addons:
apt:
packages:
- nasm
- diffutils
compiler:
- clang
- gcc
matrix:
exclude:
- os: osx
compiler: gcc
cache:
directories:
- ffmpeg-samples
before_install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update; fi
install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
script:
- mkdir -p ffmpeg-samples
- ./configure --samples=ffmpeg-samples --cc=$CC
- make -j 8
- make fate-rsync
- make check -j 8
+1 -2
View File
@@ -1,5 +1,4 @@
# Note to Github users
Patches should be submitted to [Forgejo](https://code.ffmpeg.org/FFmpeg/FFmpeg/pulls) or the [ffmpeg-devel mailing list](https://ffmpeg.org/mailman/listinfo/ffmpeg-devel) using `git format-patch` or `git send-email`. Github pull requests should be avoided because they are not part of our review process and **will be ignored**.
Patches should be submitted to the [ffmpeg-devel mailing list](https://ffmpeg.org/mailman/listinfo/ffmpeg-devel) using `git format-patch` or `git send-email`. Github pull requests should be avoided because they are not part of our review process and **will be ignored**.
See [https://ffmpeg.org/developer.html#Contributing](https://ffmpeg.org/developer.html#Contributing) for more information.
+9 -9
View File
@@ -55,7 +55,7 @@ modified by someone else and passed on, the recipients should know
that what they have is not the original version, so that the original
author's reputation will not be affected by problems that might be
introduced by others.
Finally, software patents pose a constant threat to the existence of
any free program. We wish to make sure that a company cannot
effectively restrict the users of a free program by obtaining a
@@ -111,7 +111,7 @@ modification follow. Pay close attention to the difference between a
"work based on the library" and a "work that uses the library". The
former contains code derived from the library, whereas the latter must
be combined with the library in order to run.
GNU LESSER GENERAL PUBLIC LICENSE
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
@@ -158,7 +158,7 @@ Library.
You may charge a fee for the physical act of transferring a copy,
and you may at your option offer warranty protection in exchange for a
fee.
2. You may modify your copy or copies of the Library or any portion
of it, thus forming a work based on the Library, and copy and
distribute such modifications or work under the terms of Section 1
@@ -216,7 +216,7 @@ instead of to this License. (If a newer version than version 2 of the
ordinary GNU General Public License has appeared, then you can specify
that version instead if you wish.) Do not make any other change in
these notices.
Once this change is made in a given copy, it is irreversible for
that copy, so the ordinary GNU General Public License applies to all
subsequent copies and derivative works made from that copy.
@@ -267,7 +267,7 @@ Library will still fall under Section 6.)
distribute the object code for the work under the terms of Section 6.
Any executables containing that work also fall under Section 6,
whether or not they are linked directly with the Library itself.
6. As an exception to the Sections above, you may also combine or
link a "work that uses the Library" with the Library to produce a
work containing portions of the Library, and distribute that work
@@ -329,7 +329,7 @@ restrictions of other proprietary libraries that do not normally
accompany the operating system. Such a contradiction means you cannot
use both them and the Library together in an executable that you
distribute.
7. You may place library facilities that are a work based on the
Library side-by-side in a single library together with other library
facilities not covered by this License, and distribute such a combined
@@ -370,7 +370,7 @@ subject to these terms and conditions. You may not impose any further
restrictions on the recipients' exercise of the rights granted herein.
You are not responsible for enforcing compliance by third parties with
this License.
11. If, as a consequence of a court judgment or allegation of patent
infringement or for any other reason (not limited to patent issues),
conditions are imposed on you (whether by court order, agreement or
@@ -422,7 +422,7 @@ conditions either of that version or of any later version published by
the Free Software Foundation. If the Library does not specify a
license version number, you may choose any version ever published by
the Free Software Foundation.
14. If you wish to incorporate parts of the Library into other free
programs whose distribution conditions are incompatible with these,
write to the author to ask for permission. For software which is
@@ -456,7 +456,7 @@ SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
DAMAGES.
END OF TERMS AND CONDITIONS
How to Apply These Terms to Your New Libraries
If you develop a new library, and you want it to be of the greatest
+2 -2
View File
@@ -1,6 +1,6 @@
See the Git history of the project (https://git.ffmpeg.org/ffmpeg) to
See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
get the names of people who have contributed to FFmpeg.
To check the log, you can type the command "git log" in the FFmpeg
source directory, or browse the online repository at
https://git.ffmpeg.org/ffmpeg
http://source.ffmpeg.org.
-298
View File
@@ -1,304 +1,6 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>:
- extensively improved AAC encoder
- APV Vulkan encoder
version 9.0:
- Extend AMF Color Converter (vf_vpp_amf) HDR capabilities
- LCEVC track muxing support in MP4 muxer
- Playdate video encoder and muxer
- Add v360_vulkan filter
- HE-AAC 960 decoding (DAB+)
- transpose_cuda filter
- Add AMF Frame Rate Converter (vf_frc_amf) filter
- SMPTE 2094-50 metadata support and passthrough
- ProRes RAW VideoToolbox hwaccel
- APV Vulkan hwaccel
- Animated WebP decoder
- Animated WebP demuxer
- Remove CELT decoding support (doesn't affect Opus CELT)
- Remove ogg/celt parsing
- Bitstream filter to split Dolby Vision multi-layer HEVC
- Add AMF hardware memory mapping support.
- ONNX Runtime DNN backend with GPU execution provider support
- Remove deprecated NVENC options and support for pre-11.1 SDK versions
- iTerm2 inline image protocol muxer
version 8.1:
- ffprobe -codec option
- EXIF Metadata Parsing
- gfxcapture: Windows.Graphics.Capture based window/monitor capture
- hxvs demuxer for HXVS/HXVT IP camera format
- MPEG-H 3D Audio decoding via mpeghdec
- D3D12 H.264 encoder
- drawvg filter via libcairo
- ffmpeg CLI tiled HEIF support
- D3D12 AV1 encoder
- ProRes Vulkan hwaccel
- DPX Vulkan hwaccel
- Rockchip H.264/HEVC hardware encoder
- Add vf_scale_d3d12 filter
- JPEG-XS parser
- JPEG-XS decoder and encoder through libsvtjpegxs
- JPEG-XS raw bitstream muxer and demuxer
- IAMF Projection mode Ambisonic Audio Elements muxing and demuxing
- Add vf_mestimate_d3d12 filter
- xHE-AAC Mps212 decoding support (experimental)
- Remove the old HLS protocol handler
- Vulkan compute codec optimizations
- swscale Vulkan support
- LCEVC metadata bitstream filter
- Add vf_deinterlace_d3d12 filter
- ffprobe: only show refs field in stream section when reading frames
- ProRes Vulkan encoder
- LCEVC parser
- LCEVC enhancement layer exporting in MPEG-TS
version 8.0:
- Whisper filter
- Drop support for OpenSSL < 1.1.0
- Enable TLS peer certificate verification by default (on next major version bump)
- Drop support for OpenSSL < 1.1.1
- yasm support dropped, users need to use nasm
- VVC VAAPI decoder
- RealVideo 6.0 decoder
- OpenMAX encoders deprecated
- libx265 alpha layer encoding
- ADPCM IMA Xbox decoder
- Enhanced FLV v2: Multitrack audio/video, modern codec support
- Animated JPEG XL encoding (via libjxl)
- VVC in Matroska
- CENC AV1 support in MP4 muxer
- pngenc: set default prediction method to PAETH
- APV decoder and APV raw bitstream muxing and demuxing
- APV parser
- APV encoding support through a libopenapv wrapper
- VVC decoder supports all content of SCC (Screen Content Coding):
IBC (Inter Block Copy), Palette Mode and ACT (Adaptive Color Transform
- G.728 decoder
- pad_cuda filter
- Sanyo LD-ADPCM decoder
- APV in MP4/ISOBMFF muxing and demuxing
- OpenHarmony hardware decoder/encoder
- Colordetect filter
- Add vf_scale_d3d11 filter
- No longer disabling GCC autovectorization, on X86, ARM and AArch64
- VP9 Vulkan hwaccel
- AV1 Vulkan encoder
- ProRes RAW decoder
- ProRes RAW Vulkan hwaccel
- ffprobe -codec option
- HDR10+ metadata passthrough when decoding/encoding with libaom-av1
version 7.1:
- Raw Captions with Time (RCWT) closed caption demuxer
- LC3/LC3plus decoding/encoding using external library liblc3
- ffmpeg CLI filtergraph chaining
- LC3/LC3plus demuxer and muxer
- pad_vaapi, drawbox_vaapi filters
- vf_scale supports secondary ref input and framesync options
- vf_scale2ref deprecated
- qsv_params option added for QSV encoders
- VVC decoder compatible with DVB test content
- xHE-AAC decoder
- removed DEC Alpha DSP and support code
- VVC encoding support via libvvenc
- perlin video source
- D3D12VA HEVC encoder
- Cropping metadata parsing and writing in Matroska and MP4/MOV de/muxers
- Intel QSV-accelerated VVC decoding
- MediaCodec AAC/AMR-NB/AMR-WB/MP3 decoding
- YUV colorspace negotiation for codecs and filters, obsoleting the
YUVJ pixel format
- Vulkan H.264 encoder
- Vulkan H.265 encoder
- stream specifiers in fftools can now match by stream disposition
- LCEVC enhancement data exporting in H.26x and MP4/ISOBMFF
- LCEVC filter
- MV-HEVC decoding
- minor stream specifier syntax changes:
- when matching by metadata (:m:<key>:<val>), the colon character
in keys or values now has to be backslash-escaped
- in optional maps (-map ....?) with a metadata-matching stream specifier,
the value has to be separated from the question mark by a colon, i.e.
-map ....:m:<key>:<val>:? (otherwise it would be ambiguous whether the
question mark is a part of <val> or not)
- multiple stream types in a single specifier (e.g. :s:s:0) now cause an
error, as such a specifier makes no sense
- Mastering Display and Content Light Level metadata support in hevc_nvenc
and av1_nvenc encoders
- libswresample now accepts custom order channel layouts as input, with some
constrains
- FFV1 parser
version 7.0:
- DXV DXT1 encoder
- LEAD MCMP decoder
- EVC decoding using external library libxevd
- EVC encoding using external library libxeve
- QOA decoder and demuxer
- aap filter
- demuxing, decoding, filtering, encoding, and muxing in the
ffmpeg CLI now all run in parallel
- enable gdigrab device to grab a window using the hwnd=HANDLER syntax
- IAMF raw demuxer and muxer
- D3D12VA hardware accelerated H264, HEVC, VP9, AV1, MPEG-2 and VC1 decoding
- tiltandshift filter
- qrencode filter and qrencodesrc source
- quirc filter
- lavu/eval: introduce randomi() function in expressions
- VVC decoder (experimental)
- fsync filter
- Raw Captions with Time (RCWT) closed caption muxer
- ffmpeg CLI -bsf option may now be used for input as well as output
- ffmpeg CLI options may now be used as -/opt <path>, which is equivalent
to -opt <contents of file <path>>
- showinfo bitstream filter
- a C11-compliant compiler is now required; note that this requirement
will be bumped to C17 in the near future, so consider updating your
build environment if it lacks C17 support
- Change the default bitrate control method from VBR to CQP for QSV encoders.
- removed deprecated ffmpeg CLI options -psnr and -map_channel
- DVD-Video demuxer, powered by libdvdnav and libdvdread
- ffprobe -show_stream_groups option
- ffprobe (with -export_side_data film_grain) now prints film grain metadata
- AEA muxer
- ffmpeg CLI loopback decoders
- Support PacketTypeMetadata of PacketType in enhanced flv format
- ffplay with hwaccel decoding support (depends on vulkan renderer via libplacebo)
- dnn filter libtorch backend
- Android content URIs protocol
- AOMedia Film Grain Synthesis 1 (AFGS1)
- RISC-V optimizations for AAC, FLAC, JPEG-2000, LPC, RV4.0, SVQ, VC1, VP8, and more
- Loongarch optimizations for HEVC decoding
- Important AArch64 optimizations for HEVC
- IAMF support inside MP4/ISOBMFF
- Support for HEIF/AVIF still images and tiled still images
- Dolby Vision profile 10 support in AV1
- Support for Ambient Viewing Environment metadata in MP4/ISOBMFF
- HDR10 metadata passthrough when encoding with libx264, libx265, and libsvtav1
version 6.1:
- libaribcaption decoder
- Playdate video decoder and demuxer
- Extend VAAPI support for libva-win32 on Windows
- afireqsrc audio source filter
- arls filter
- ffmpeg CLI new option: -readrate_initial_burst
- zoneplate video source filter
- command support in the setpts and asetpts filters
- Vulkan decode hwaccel, supporting H264, HEVC and AV1
- color_vulkan filter
- bwdif_vulkan filter
- nlmeans_vulkan filter
- RivaTuner video decoder
- xfade_vulkan filter
- vMix video decoder
- Essential Video Coding parser, muxer and demuxer
- Essential Video Coding frame merge bsf
- bwdif_cuda filter
- Microsoft RLE video encoder
- Raw AC-4 muxer and demuxer
- Raw VVC bitstream parser, muxer and demuxer
- Bitstream filter for editing metadata in VVC streams
- Bitstream filter for converting VVC from MP4 to Annex B
- scale_vt filter for videotoolbox
- transpose_vt filter for videotoolbox
- support for the P_SKIP hinting to speed up libx264 encoding
- Support HEVC,VP9,AV1 codec in enhanced flv format
- apsnr and asisdr audio filters
- OSQ demuxer and decoder
- Support HEVC,VP9,AV1 codec fourcclist in enhanced rtmp protocol
- CRI USM demuxer
- ffmpeg CLI '-top' option deprecated in favor of the setfield filter
- VAAPI AV1 encoder
- ffprobe XML output schema changed to account for multiple
variable-fields elements within the same parent element
- ffprobe -output_format option added as an alias of -of
# codespell:off
version 6.0:
- Radiance HDR image support
- ddagrab (Desktop Duplication) video capture filter
- ffmpeg -shortest_buf_duration option
- ffmpeg now requires threading to be built
- ffmpeg now runs every muxer in a separate thread
- Add new mode to cropdetect filter to detect crop-area based on motion vectors and edges
- VAAPI decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
- WBMP (Wireless Application Protocol Bitmap) image format
- a3dscope filter
- bonk decoder and demuxer
- Micronas SC-4 audio decoder
- LAF demuxer
- APAC decoder and demuxer
- Media 100i decoders
- DTS to PTS reorder bsf
- ViewQuest VQC decoder
- backgroundkey filter
- nvenc AV1 encoding support
- MediaCodec decoder via NDKMediaCodec
- MediaCodec encoder
- oneVPL support for QSV
- QSV AV1 encoder
- QSV decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
- showcwt multimedia filter
- corr video filter
- adrc audio filter
- afdelaysrc audio filter
- WADY DPCM decoder and demuxer
- CBD2 DPCM decoder
- ssim360 video filter
- ffmpeg CLI new options: -stats_enc_pre[_fmt], -stats_enc_post[_fmt],
-stats_mux_pre[_fmt]
- hstack_vaapi, vstack_vaapi and xstack_vaapi filters
- XMD ADPCM decoder and demuxer
- media100 to mjpegb bsf
- ffmpeg CLI new option: -fix_sub_duration_heartbeat
- WavArc decoder and demuxer
- CrystalHD decoders deprecated
- SDNS demuxer
- RKA decoder and demuxer
- filtergraph syntax in ffmpeg CLI now supports passing file contents
as option values, by prefixing option name with '/'
- hstack_qsv, vstack_qsv and xstack_qsv filters
version 5.1:
- add ipfs/ipns gateway support
- dialogue enhance audio filter
- dropped obsolete XvMC hwaccel
- pcm-bluray encoder
- DFPWM audio encoder/decoder and raw muxer/demuxer
- SITI filter
- Vizrt Binary Image encoder/decoder
- avsynctest source filter
- feedback video filter
- pixelize video filter
- colormap video filter
- colorchart video source filter
- multiply video filter
- PGS subtitle frame merge bitstream filter
- blurdetect filter
- tiltshelf audio filter
- QOI image format support
- ffprobe -o option
- virtualbass audio filter
- VDPAU AV1 hwaccel
- PHM image format support
- remap_opencl filter
- added chromakey_cuda filter
- added bilateral_cuda filter
version 5.0:
- ADPCM IMA Westwood encoder
- Westwood AUD muxer
-7
View File
@@ -1,7 +0,0 @@
{
"drips": {
"ethereum": {
"ownedBy": "0x2f3900e7064eE63D30d749971265858612AA7139"
}
}
}
+4 -11
View File
@@ -1,7 +1,4 @@
# Installing FFmpeg
0. If you like to include source plugins, merge them before configure
for example run tools/merge-all-source-plugins
## Installing FFmpeg
1. Type `./configure` to create the configuration. A list of configure
options is printed by running `configure --help`.
@@ -14,11 +11,7 @@ path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
3. Type `make install` to install all binaries and libraries you built.
## NOTICE
NOTICE
------
- Non system dependencies (e.g. libx264, libvpx) are disabled by default.
## NOTICE for Package Maintainers
- It is recommended to build FFmpeg twice, first with minimal external dependencies so that 3rd party packages, which depend on FFmpegs libavutil/libavfilter/libavcodec/libavformat
can then be built. And last build FFmpeg with full dependencies (which may in turn depend on some of these 3rd party packages). This avoids circular dependencies during build.
- Non system dependencies (e.g. libx264, libvpx) are disabled by default.
+2
View File
@@ -12,6 +12,7 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are:
- libpostproc
- optional x86 optimization in the files
- `libavcodec/x86/flac_dsp_gpl.asm`
- `libavcodec/x86/idct_mmx.c`
@@ -44,6 +45,7 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_owdenoise.c`
- `vf_perspective.c`
- `vf_phase.c`
- `vf_pp.c`
- `vf_pp7.c`
- `vf_pullup.c`
- `vf_repeatfields.c`
+127 -127
View File
@@ -6,38 +6,28 @@ FFmpeg code.
Please try to keep entries where you are the maintainer up to date!
*Status*, one of the following:
[X] Old code. Something tagged obsolete generally means it has been replaced by a better system and you should be using that.
[0] No current maintainer [but maybe you could take the role as you write your new code].
[1] It has a maintainer but they don't have time to do much other than throw the odd patch in.
[2] Someone actually looks after it.
Names in () mean that the maintainer currently has no time to maintain the code.
A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
patches and related discussions.
(L <address>) *Mailing list* that is relevant to this area
(W <address>) *Web-page* with status/info
(B <address>) URI for where to file *bugs*. A web-page with detailed bug
filing info, a direct bug tracker link, or a mailto: URI.
(P <address>) *Subsystem Profile* document for more details submitting
patches to the given subsystem. This is either an in-tree file,
or a URI. See Documentation/maintainer/maintainer-entry-profile.rst
for details.
(T <address>) *SCM* tree type and location.
Type is one of: git, hg, quilt, stgit, topgit
Project Leader
==============
final design decisions
Applications
============
ffmpeg:
ffmpeg.c Michael Niedermayer, Anton Khirnov
ffmpeg.c Michael Niedermayer
ffplay:
ffplay.c [2] Marton Balint
ffplay.c Marton Balint
ffprobe:
ffprobe.c [2] Stefano Sabatini
ffprobe.c Stefano Sabatini
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
@@ -45,33 +35,30 @@ Commandline utility code:
QuickTime faststart:
tools/qt-faststart.c Baptiste Coudurier
Execution Graph Printing
fftools/graph, fftools/resources [2] softworkz
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
project server day to day operations (L: root@ffmpeg.org) Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov, Timo Rothenpieler
project server emergencies (L: root@ffmpeg.org) Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov, Timo Rothenpieler
presets [0]
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
API tests [0]
samples-request [2] Thilo Borgmann, James Almer, Ben Littler
API tests Ludmila Glinskih
Communication
=============
website (T: https://git.ffmpeg.org/ffmpeg-web) Deby Barbara Lepage
fate.ffmpeg.org (L: fate-admin@ffmpeg.org) (W: https://fate.ffmpeg.org) (P: https://ffmpeg.org/fate.html) (S: https://git.ffmpeg.org/fateserver) Timo Rothenpieler
Trac bug tracker [X] (W: https://trac.ffmpeg.org) Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
Forgejo issue tracker (W: https://code.ffmpeg.org/FFmpeg/FFmpeg/issues)
Patchwork [2] (W: https://patchwork.ffmpeg.org) Andriy Gelman
mailing lists (W: https://ffmpeg.org/contact.html#MailingLists) Baptiste Coudurier
website Deby Barbara Lepage
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
Patchwork Andriy Gelman
mailing lists Baptiste Coudurier
Twitter Reynaldo H. Verdejo Pinochet
Launchpad Timothy Gu
ffmpeg-security [2] (L: ffmpeg-security@ffmpeg.org) (W: https://ffmpeg.org/security.html) Michael Niedermayer, Reimar Doeffinger
ffmpeg-security Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, rcombs, wm4
libavutil
=========
@@ -85,27 +72,24 @@ Other:
aes_ctr.c, aes_ctr.h Eran Kornblau
bprint Nicolas George
bswap.h
csp.c, csp.h Leo Izen, Ronald S. Bultje
des Reimar Doeffinger
dynarray.h Nicolas George
eval.c, eval.h [2] Michael Niedermayer
eval.c, eval.h Michael Niedermayer
float_dsp Loren Merritt
hash Reimar Doeffinger
hwcontext_cuda* Timo Rothenpieler
hwcontext_d3d12va* Wu Jianhua
hwcontext_oh* Zhao Zhili
hwcontext_vulkan* [2] Lynne
hwcontext_vulkan* Lynne
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
mathematics.c, mathematics.h [2] Michael Niedermayer
mem.c, mem.h [2] Michael Niedermayer
mathematics.c, mathematics.h Michael Niedermayer
mem.c, mem.h Michael Niedermayer
opencl.c, opencl.h Wei Gao
opt.c, opt.h Michael Niedermayer
rational.c, rational.h [2] Michael Niedermayer
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
tx* [2] Lynne
tx* Lynne
libavcodec
@@ -127,18 +111,22 @@ Generic Parts:
DSP utilities:
dsputils.c, dsputils.h Michael Niedermayer
entropy coding:
rangecoder.c, rangecoder.h [2] Michael Niedermayer
rangecoder.c, rangecoder.h Michael Niedermayer
lzw.* Michael Niedermayer
floating point AAN DCT:
faandct.c, faandct.h [2] Michael Niedermayer
faandct.c, faandct.h Michael Niedermayer
Non-power-of-two MDCT:
mdct15.c, mdct15.h Rostislav Pehlivanov
Golomb coding:
golomb.c, golomb.h [2] Michael Niedermayer
golomb.c, golomb.h Michael Niedermayer
motion estimation:
motion* Michael Niedermayer
rate control:
ratecontrol.c [2] Michael Niedermayer
ratecontrol.c Michael Niedermayer
simple IDCT:
simple_idct.c, simple_idct.h [2] Michael Niedermayer
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
libpostproc/* Michael Niedermayer
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
fixed point FFT:
@@ -146,33 +134,33 @@ Generic Parts:
Text Subtitles Clément Bœsch
Codecs:
4xm.c [2] Michael Niedermayer
4xm.c Michael Niedermayer
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aacenc*, aaccoder.c Rostislav Pehlivanov
adpcm.c Zane van Iperen
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann, Umair Khan
amfdec*,amfenc* [2] Dmitrii Ovchinnikov, Araz Iusubov
aptx.c Aurelien Jacobs
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3plus* Maxim Poliakovski
audiotoolbox* rcombs
avs2* Huiwen Ren, Zhao Zhili
avs2* Huiwen Ren
bgmc.c, bgmc.h Thilo Borgmann
binkaudio.c Peter Ross
cavs* Stefan Gehrer
cdxl.c Paul B Mahol
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cinepakenc.c Rl / Aetey G.T. AB
ccaption_dec.c Anshul Maheshwari, Aman Gupta
cljr Alex Beregszaszi
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
cuviddec.c Timo Rothenpieler
dca* foo86
dfpwm* Jack Bruienne
dirac* Rostislav Pehlivanov
dnxhd* Baptiste Coudurier
dolby_e* foo86
@@ -180,10 +168,10 @@ Codecs:
dss_sp.c Oleksij Rempel
dv.c Roman Shaposhnik
dvbsubdec.c Anshul Maheshwari
dxv.*, dxvenc.* Emma Worley
eacmv*, eaidct*, eat* Peter Ross
evrc* Paul B Mahol
exif.c, exif.h Thilo Borgmann
ffv1* [2] Michael Niedermayer
ffv1* Michael Niedermayer
ffwavesynth.c Nicolas George
fifo.c Jan Sebechlebsky
flicvideo.c Mike Melanson
@@ -194,25 +182,23 @@ Codecs:
h263* Michael Niedermayer
h264* Loren Merritt, Michael Niedermayer
hap* Tom Butterworth
hevc/* Anton Khirnov
huffyuv* Michael Niedermayer
idcinvideo.c Mike Melanson
interplayvideo.c Mike Melanson
jni*, ffjni* Matthieu Bouron
jpeg2000* Nicolas Bertrand
jpegxl* Leo Izen
jvdec.c Peter Ross
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libcodec2.c Tomas Härdin
libdirac* David Conrad
libdavs2.c Huiwen Ren
libjxl*.c, libjxl.h Leo Izen
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenh264enc.c Martin Storsjo, Linjie Fu
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libtheoraenc.c [0]
libtheoraenc.c David Conrad
libvorbis.c David Conrad
libvpx* James Zern
libxavs.c Stefan Gehrer
@@ -229,12 +215,12 @@ Codecs:
mqc* Nicolas Bertrand
msmpeg4.c, msmpeg4data.h Michael Niedermayer
msrle.c Mike Melanson
msrleenc.c Tomas Härdin
msvideo1.c Mike Melanson
nuv.c Reimar Doeffinger
nvdec*, nvenc* Timo Rothenpieler
oh* Zhao Zhili
omx.c Martin Storsjo, Aman Gupta
opus* Rostislav Pehlivanov
paf.* Paul B Mahol
pcx.c Ivo van Poorten
pgssubdec.c Reimar Doeffinger
ptx.c Ivo van Poorten
@@ -244,25 +230,28 @@ Codecs:
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
rkmppenc* Zhao Zhili
rl2.c Sascha Sommer
rpza.c Roberto Togni
rtjpeg.c, rtjpeg.h Reimar Doeffinger
rv10.c Michael Niedermayer
sanm.c Manuel Lauss
s3tc* Ivo van Poorten
smc.c Mike Melanson
snow* Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
speedhq.c Steinar H. Gunderson
srt* Aurelien Jacobs
sunrast.c Ivo van Poorten
svq3.c Michael Niedermayer
tak* Paul B Mahol
truemotion1* Mike Melanson
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
v4l2_* Jorge Ramirez-Ortiz
vc2* Rostislav Pehlivanov
vcr1.c Michael Niedermayer
videotoolboxenc.c Rick Kern, Aman Gupta
vima.c Paul B Mahol
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
vp3* Mike Melanson
@@ -271,24 +260,24 @@ Codecs:
vp8 David Conrad, Ronald Bultje
vp9 Ronald Bultje
vqavideo.c Mike Melanson
vvc [2] Nuo Mi, Wu Jianhua, Frank Plowman
wmaprodec.c Sascha Sommer
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
xan.c Mike Melanson
xbm* Paul B Mahol
xface Stefano Sabatini
xvmc.c Ivan Kalvachev
xwd* Paul B Mahol
Hardware acceleration:
amf* [2] Dmitrii Ovchinnikov, Araz Iusubov
crystalhd.c Philip Langdale
dxva2* Hendrik Leppkes, Laurent Aimar, Steve Lhomme
d3d11va* Steve Lhomme
d3d12va* Wu Jianhua
d3d12va_encode* Tong Wu
mediacodec* Matthieu Bouron, Aman Gupta, Zhao Zhili
mediacodec* Matthieu Bouron, Aman Gupta
vaapi* Haihao Xiang
vaapi_encode* Mark Thompson, Haihao Xiang
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern, Aman Gupta, Zhao Zhili
videotoolbox* Rick Kern, Aman Gupta
libavdevice
@@ -318,49 +307,71 @@ libavfilter
===========
Generic parts:
Framework and orphaned filters Nicolas George
(except hardware acceleration)
graphdump.c Nicolas George
motion_estimation.c Davinder Singh
Filters:
f_drawgraph.c Paul B Mahol
af_adelay.c Paul B Mahol
af_aecho.c Paul B Mahol
af_afade.c Paul B Mahol
af_amerge.c Nicolas George
af_aphaser.c Paul B Mahol
af_aresample.c Michael Niedermayer
af_astats.c Paul B Mahol
af_atempo.c Pavel Koshevoy
af_biquads.c Paul B Mahol
af_chorus.c Paul B Mahol
af_compand.c Paul B Mahol
af_firequalizer.c Muhammad Faiz
af_hdcd.c Burt P.
af_ladspa.c Paul B Mahol
af_loudnorm.c Kyle Swanson
af_pan.c Nicolas George
af_sidechaincompress.c Paul B Mahol
af_silenceremove.c Paul B Mahol
avf_aphasemeter.c Paul B Mahol
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
vf_bwdif Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_chromakey.c Timo Rothenpieler
vf_colorchannelmixer.c Paul B Mahol
vf_colorconstancy.c Mina Sami (CC <minas.gorgy@gmail.com>)
vf_colorbalance.c Paul B Mahol
vf_colorkey.c Timo Rothenpieler
vf_colorlevels.c Paul B Mahol
vf_coreimage.m Thilo Borgmann
vf_deband.c Paul B Mahol
vf_dejudder.c Nicholas Robbins
vf_delogo.c Jean Delvare (CC <jdelvare@suse.com>)
vf_drawbox.c/drawgrid Andrey Utkin
vf_fsync.c Thilo Borgmann
vf_extractplanes.c Paul B Mahol
vf_histogram.c Paul B Mahol
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_(t)interlace Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_lenscorrection.c Daniel Oberhoff
vf_libplacebo.c Niklas Haas
vf_mergeplanes.c Paul B Mahol
vf_mestimate.c Davinder Singh
vf_minterpolate.c Davinder Singh
vf_neighbor.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_random.c Paul B Mahol
vf_readvitc.c Tobias Rapp (CC t.rapp at noa-archive dot com)
vf_scale.c [2] Michael Niedermayer
vf_scale_vt.c Zhao Zhili
vf_scale.c Michael Niedermayer
vf_separatefields.c Paul B Mahol
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_tonemap_opencl.c Ruiling Song
vf_transpose_vt.c Zhao Zhili
vf_yadif.c [2] Michael Niedermayer
vf_xfade_vulkan.c [2] Marvin Scholz (CC <epirat07@gmail.com>)
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c [2] Michael Niedermayer
vsrc_mandelbrot.c Michael Niedermayer
dnn Yejun Guo
@@ -378,7 +389,8 @@ Generic parts:
Muxers/Demuxers:
4xm.c Mike Melanson
aadec.c Vesselin Bontchev (vesselin.bontchev at yandex dot com)
adtsenc.c [0]
adtsenc.c Robert Swain
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
alp.c Zane van Iperen
@@ -389,53 +401,54 @@ Muxers/Demuxers:
argo_brp.c Zane van Iperen
argo_cvg.c Zane van Iperen
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
avi* Michael Niedermayer
avisynth.c Stephen Hutchinson
avr.c Paul B Mahol
bink.c Peter Ross
boadec.c Michael Niedermayer
brstm.c Paul B Mahol
caf* Peter Ross
cdxl.c Paul B Mahol
codec2.c Tomas Härdin
crc.c Michael Niedermayer
dashdec.c Steven Liu
dashenc.c Karthick Jeyapal
daud.c Reimar Doeffinger
dfpwmdec.c Jack Bruienne
dss.c Oleksij Rempel
dtsdec.c foo86
dtshddec.c Paul B Mahol
dv.c Roman Shaposhnik
dvdvideodec.c [2] Marth64
electronicarts.c Peter Ross
evc* Samsung (Dawid Kozinski)
epafdec.c Paul B Mahol
ffm* Baptiste Coudurier
flic.c Mike Melanson
flvdec.c Michael Niedermayer
flvenc.c Michael Niedermayer, Steven Liu
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hls.c Kacper Michajłow
hlsenc.c Christian Suloway, Steven Liu
hxvs.c Zhao Zhili
iamf* [2] James Almer
idcin.c Mike Melanson
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
imf* Pierre-Anthony Lemieux
img2*.c Michael Niedermayer
ipmovie.c Mike Melanson
ircam* Paul B Mahol
iss.c Stefan Gehrer
jpegxl* Leo Izen
jvdec.c Peter Ross
kvag.c Zane van Iperen
libmodplug.c Clément Bœsch
libopenmpt.c Josh de Kock
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
matroska.c Andreas Rheinhardt
matroskadec.c Andreas Rheinhardt
matroskaenc.c Andreas Rheinhardt
matroska.c Aurelien Jacobs, Andreas Rheinhardt
matroskadec.c Aurelien Jacobs, Andreas Rheinhardt
matroskaenc.c David Conrad, Andreas Rheinhardt
matroska subtitles (matroskaenc.c) John Peebles
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
microdvd* Aurelien Jacobs
mm.c Peter Ross
mov.c Baptiste Coudurier
@@ -448,20 +461,21 @@ Muxers/Demuxers:
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier, Tomas Härdin
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
nuv.c Reimar Doeffinger
oggdec.c, oggdec.h David Conrad, Romain Beauxis
oggenc.c Baptiste Coudurier, Romain Beauxis
oggparse*.c David Conrad, Romain Beauxis
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oma.c Maxim Poliakovski
paf.c Paul B Mahol
pp_bnk.c Zane van Iperen
psxstr.c Mike Melanson
pva.c Ivo van Poorten
pvfdec.c Paul B Mahol
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rcwtdec.c [2] Marth64
rcwtenc.c [2] Marth64
rdt.c Ronald S. Bultje
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje
@@ -480,10 +494,11 @@ Muxers/Demuxers:
sdp.c Martin Storsjo
segafilm.c Mike Melanson
segment.c Stefano Sabatini
smush.c Manuel Lauss
smjpeg* Paul B Mahol
spdif* Anssi Hannula
srtdec.c Aurelien Jacobs
swf.c Baptiste Coudurier
takdec.c Paul B Mahol
tta.c Alex Beregszaszi
txd.c Ivo van Poorten
voc.c Aurelien Jacobs
@@ -492,51 +507,47 @@ Muxers/Demuxers:
webm dash (matroskaenc.c) Vignesh Venkatasubramanian
webvtt* Matthew J Heaney
westwood.c Mike Melanson
whip.c [2] Jack Lau
wtv.c Peter Ross
wvenc.c Paul B Mahol
Protocols:
async.c Zhang Rui
bluray.c Petri Hintukainen
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libsrt.c Zhao Zhili
libssh.c Lukasz Marek
libzmq.c Andriy Gelman
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c [2] Marvin Scholz (CC <epirat07@gmail.com>)
icecast.c Marvin Scholz
libswresample
=============
Generic parts:
audioconvert.c [2] Michael Niedermayer
dither.c [2] Michael Niedermayer
rematrix*.c [2] Michael Niedermayer
swresample*.c [2] Michael Niedermayer
audioconvert.c Michael Niedermayer
dither.c Michael Niedermayer
rematrix*.c Michael Niedermayer
swresample*.c Michael Niedermayer
Resamplers:
resample*.c [2] Michael Niedermayer
resample*.c Michael Niedermayer
soxr_resample.c Rob Sykes
Operating systems / CPU architectures
=====================================
*BSD [2] Brad Smith
Alpha [0]
Alpha Falk Hueffner
MIPS Manojkumar Bhosale, Shiyou Yin
LoongArch [2] Shiyou Yin
Darwin (macOS, iOS) [2] Marvin Scholz
Mac OS X / PowerPC [0]
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC [2] Sean McGovern (CC <gseanmcg@gmail.com>), Lauri Kasanen
RISC-V [2] Rémi Denis-Courmont
Linux / PowerPC Lauri Kasanen
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Hendrik Leppkes
Windows MSVC Matthew Oliver, Hendrik Leppkes
Windows ICL Matthew Oliver
ADI/Blackfin DSP Marc Hoffman
Sparc Roman Shaposhnik
OS/2 KO Myung-Hun
@@ -552,7 +563,6 @@ Benjamin Larsson
Bobby Bingham
Daniel Verkamp
Derek Buitenhuis
Fei Wang
Ganesh Ajjanagadde
Henrik Gramner
Ivan Uskov
@@ -560,7 +570,6 @@ James Darnley
Jan Ekström
Joakim Plate
Jun Zhao
Kacper Michajłow
Kieran Kunhya
Kirill Gavrilov
Limin Wang
@@ -576,12 +585,10 @@ wm4
Releases
========
7.0 Michael Niedermayer
6.1 Michael Niedermayer
5.1 Michael Niedermayer
4.4 Michael Niedermayer
3.4 Michael Niedermayer
2.8 Michael Niedermayer
2.7 Michael Niedermayer
2.6 Michael Niedermayer
2.5 Michael Niedermayer
If you want to maintain an older release, please contact us
@@ -602,33 +609,26 @@ Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Clément Bœsch 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Frank Plowman 34E2 48D6 B7DF 4769 70C7 3304 03A8 4C6A 098F 2C6B
Ganesh Ajjanagadde C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Haihao Xiang (haihao) 1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
James Almer 7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Leo Izen (Traneptora) B6FD 3CFC 7ACF 83FC 9137 6945 5A71 C331 FD2F A19A
Leo Izen (Traneptora) 1D83 0A0B CE46 709E 203B 26FC 764E 48EA 4822 1833
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lynne FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Niklas Haas (haasn) 1DDB 8076 B14D 5B48 32FC 99D9 EB52 DA9C 02BA 6FB4
Nikolay Aleksandrov 8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
Philip Langdale 5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
Pierre-Anthony Lemieux (pal) F4B3 9492 E6F2 E4AF AEC8 46CB 698F A1F0 F8D4 EED4
Ramiro Polla 7859 C65B 751B 1179 792E DAE8 8E95 8B2F 9B6C 5700
Reimar Doeffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Sean McGovern (Sean_McG) 6D03 BC60 3A33 E615 6E2E 06AD 8C06 8175 6F59 8684
Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Steinar H. Gunderson C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
+19 -50
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@@ -11,8 +11,6 @@ vpath %.asm $(SRC_PATH)
vpath %.rc $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %.glsl $(SRC_PATH)
vpath %.spv $(SRC_PATH)
vpath %.cu $(SRC_PATH)
vpath %.ptx $(SRC_PATH)
vpath %.metal $(SRC_PATH)
@@ -21,20 +19,12 @@ vpath %/fate_config.sh.template $(SRC_PATH)
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
ALLFFLIBS = \
avcodec \
avdevice \
avfilter \
avformat \
avutil \
swscale \
swresample \
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
@@ -55,36 +45,26 @@ FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
$(TOOLS): %$(EXESUF): %.o
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(filter-out $(FF_DEP_LIBS), $^) $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
target_enc_%_fuzzer$(EXESUF): target_enc_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
target_dem_%_fuzzer$(EXESUF): target_dem_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_sws_fuzzer$(EXESUF): tools/target_sws_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
tools/target_swr_fuzzer$(EXESUF): tools/target_swr_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
tools/enc_recon_frame_test$(EXESUF): $(FF_DEP_LIBS)
tools/enc_recon_frame_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/scale_slice_test$(EXESUF): $(FF_DEP_LIBS)
tools/scale_slice_test$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
@@ -96,24 +76,22 @@ tools/target_dem_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
CONFIGURABLE_COMPONENTS = \
$(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c)) \
$(SRC_PATH)/libavcodec/bitstream_filters.c \
$(SRC_PATH)/libavcodec/hwaccels.h \
$(SRC_PATH)/libavcodec/parsers.c \
$(SRC_PATH)/libavformat/protocols.c \
config_components.h: ffbuild/.config
config.h: ffbuild/.config
ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
@-tput bold 2>/dev/null
@-printf '\nWARNING: $(?) newer than config_components.h, rerun configure\n\n'
@-printf '\nWARNING: $(?) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
SUBDIR_VARS := CLEANFILES FFLIBS DEVPROGS HOSTPROGS TESTPROGS TOOLS \
SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS \
HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS \
ALTIVEC-OBJS VSX-OBJS X86ASM-OBJS \
ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS \
MMI-OBJS LSX-OBJS LASX-OBJS RV-OBJS RVV-OBJS RVVB-OBJS \
OBJS SHLIBOBJS STLIBOBJS HOSTOBJS TESTOBJS SIMD128-OBJS \
SVE-OBJS SVE2-OBJS SME-OBJS SME2-OBJS
MMI-OBJS LSX-OBJS LASX-OBJS OBJS SLIBOBJS SHLIBOBJS \
STLIBOBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=
@@ -135,35 +113,27 @@ include $(SRC_PATH)/fftools/Makefile
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/doc/examples/Makefile
$(ALLFFLIBS:%=lib%/version.o): libavutil/ffversion.h
libavcodec/avcodec.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
ifeq ($(STRIPTYPE),direct)
$(STRIP) -o $@ $<
else
$(RM) $@
$(CP) $< $@
ifneq ($(STRIPTYPE),nostrip)
$(STRIP) $@
endif
endif
%$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
VERSION_SH = $(SRC_PATH)/ffbuild/version.sh
ifeq ($(VERSION_TRACKING),yes)
GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
endif
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) ffbuild/config.mak
.version: M=@
ifneq ($(VERSION_TRACKING),yes)
libavutil/ffversion.h .version: REVISION=unknown
endif
libavutil/ffversion.h .version:
$(M)revision=$(REVISION) $(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
$(Q)touch .version
# force version.sh to run whenever version might have changed
@@ -189,8 +159,7 @@ clean::
$(RM) -rf coverage.info coverage.info.in lcov
distclean:: clean
$(RM) .version config.asm config.h config_components.* mapfile \
checkasm_config_generated.* checkasm_header_config_generated.h \
$(RM) .version avversion.h config.asm config.h mapfile \
ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
version.h libavutil/ffversion.h libavcodec/codec_names.h \
libavcodec/bsf_list.c libavformat/protocol_list.c \
@@ -205,8 +174,8 @@ endif
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
build: all alltools devprogs examples testprogs
check: all alltools devprogs examples testprogs fate
build: all alltools examples testprogs
check: all alltools examples testprogs fate
include $(SRC_PATH)/tests/Makefile
+2 -1
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@@ -1,4 +1,5 @@
# FFmpeg README
FFmpeg README
=============
FFmpeg is a collection of libraries and tools to process multimedia content
such as audio, video, subtitles and related metadata.
+1 -1
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@@ -1 +1 @@
8.0.git
5.0
+15
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@@ -0,0 +1,15 @@
┌────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 5.0 "Lorentz" │
└────────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 5.0 "Lorentz", about 9
months after the release of FFmpeg 4.4.
A complete Changelog is available at the root of the project, and the
complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
We hope you will like this release as much as we enjoyed working on it, and
as usual, if you have any questions about it, or any FFmpeg related topic,
feel free to join us on the #ffmpeg IRC channel (on irc.libera.chat) or ask
on the mailing-lists.
-114
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@@ -1,114 +0,0 @@
/*
* Android Binder handler
*
* Copyright (c) 2025 Dmitrii Okunev
*
* 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
*/
#if defined(__ANDROID__)
#include <dlfcn.h>
#include <stdint.h>
#include <stdlib.h>
#include "libavutil/log.h"
#include "binder.h"
#define THREAD_POOL_SIZE 1
static void *dlopen_libbinder_ndk(void)
{
/*
* libbinder_ndk.so often does not contain the functions we need, so making
* this dependency optional, thus using dlopen/dlsym instead of linking.
*
* See also: https://source.android.com/docs/core/architecture/aidl/aidl-backends
*/
void *h = dlopen("libbinder_ndk.so", RTLD_NOW | RTLD_LOCAL);
if (h != NULL)
return h;
av_log(NULL, AV_LOG_WARNING,
"android/binder: unable to load libbinder_ndk.so: '%s'; skipping binder threadpool init (MediaCodec likely won't work)\n",
dlerror());
return NULL;
}
static void android_binder_threadpool_init(void)
{
typedef int (*set_thread_pool_max_fn)(uint32_t);
typedef void (*start_thread_pool_fn)(void);
set_thread_pool_max_fn set_thread_pool_max = NULL;
start_thread_pool_fn start_thread_pool = NULL;
void *h = dlopen_libbinder_ndk();
if (h == NULL)
return;
unsigned thead_pool_size = THREAD_POOL_SIZE;
set_thread_pool_max =
(set_thread_pool_max_fn) dlsym(h,
"ABinderProcess_setThreadPoolMaxThreadCount");
start_thread_pool =
(start_thread_pool_fn) dlsym(h, "ABinderProcess_startThreadPool");
if (start_thread_pool == NULL) {
av_log(NULL, AV_LOG_WARNING,
"android/binder: ABinderProcess_startThreadPool not found; skipping threadpool init (MediaCodec likely won't work)\n");
return;
}
if (set_thread_pool_max != NULL) {
int ok = set_thread_pool_max(thead_pool_size);
av_log(NULL, AV_LOG_DEBUG,
"android/binder: ABinderProcess_setThreadPoolMaxThreadCount(%u) => %s\n",
thead_pool_size, ok ? "ok" : "fail");
} else {
av_log(NULL, AV_LOG_DEBUG,
"android/binder: ABinderProcess_setThreadPoolMaxThreadCount is unavailable; using the library default\n");
}
start_thread_pool();
av_log(NULL, AV_LOG_DEBUG,
"android/binder: ABinderProcess_startThreadPool() called\n");
}
void android_binder_threadpool_init_if_required(void)
{
#if __ANDROID_API__ >= 24
if (android_get_device_api_level() < 35) {
// the issue with the thread pool was introduced in Android 15 (API 35)
av_log(NULL, AV_LOG_DEBUG,
"android/binder: API<35, thus no need to initialize a thread pool\n");
return;
}
android_binder_threadpool_init();
#else
// android_get_device_api_level was introduced in API 24, so we cannot use it
// to detect the API level in API<24. For simplicity we just assume
// libbinder_ndk.so on the system running this code would have API level < 35;
av_log(NULL, AV_LOG_DEBUG,
"android/binder: is built with API<24, assuming this is not Android 15+\n");
#endif
}
#endif /* __ANDROID__ */
-31
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@@ -1,31 +0,0 @@
/*
* Android Binder handler
*
* Copyright (c) 2025 Dmitrii Okunev
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_ANDROID_BINDER_H
#define COMPAT_ANDROID_BINDER_H
/**
* Initialize Android Binder thread pool.
*/
void android_binder_threadpool_init_if_required(void);
#endif // COMPAT_ANDROID_BINDER_H
+173
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@@ -0,0 +1,173 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
#ifndef COMPAT_ATOMICS_GCC_STDATOMIC_H
#define COMPAT_ATOMICS_GCC_STDATOMIC_H
#include <stddef.h>
#include <stdint.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_thread_fence(order) \
__sync_synchronize()
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef _Bool atomic_flag;
typedef _Bool atomic_bool;
typedef char atomic_char;
typedef signed char atomic_schar;
typedef unsigned char atomic_uchar;
typedef short atomic_short;
typedef unsigned short atomic_ushort;
typedef int atomic_int;
typedef unsigned int atomic_uint;
typedef long atomic_long;
typedef unsigned long atomic_ulong;
typedef long long atomic_llong;
typedef unsigned long long atomic_ullong;
typedef wchar_t atomic_wchar_t;
typedef int_least8_t atomic_int_least8_t;
typedef uint_least8_t atomic_uint_least8_t;
typedef int_least16_t atomic_int_least16_t;
typedef uint_least16_t atomic_uint_least16_t;
typedef int_least32_t atomic_int_least32_t;
typedef uint_least32_t atomic_uint_least32_t;
typedef int_least64_t atomic_int_least64_t;
typedef uint_least64_t atomic_uint_least64_t;
typedef int_fast8_t atomic_int_fast8_t;
typedef uint_fast8_t atomic_uint_fast8_t;
typedef int_fast16_t atomic_int_fast16_t;
typedef uint_fast16_t atomic_uint_fast16_t;
typedef int_fast32_t atomic_int_fast32_t;
typedef uint_fast32_t atomic_uint_fast32_t;
typedef int_fast64_t atomic_int_fast64_t;
typedef uint_fast64_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef uintptr_t atomic_uintptr_t;
typedef size_t atomic_size_t;
typedef ptrdiff_t atomic_ptrdiff_t;
typedef intmax_t atomic_intmax_t;
typedef uintmax_t atomic_uintmax_t;
#define atomic_store(object, desired) \
do { \
*(object) = (desired); \
__sync_synchronize(); \
} while (0)
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
#define atomic_load(object) \
(__sync_synchronize(), *(object))
#define atomic_load_explicit(object, order) \
atomic_load(object)
#define atomic_exchange(object, desired) \
({ \
__typeof__(object) _obj = (object); \
__typeof__(*object) _old; \
do \
_old = atomic_load(_obj); \
while (!__sync_bool_compare_and_swap(_obj, _old, (desired))); \
_old; \
})
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
#define atomic_compare_exchange_strong(object, expected, desired) \
({ \
__typeof__(object) _exp = (expected); \
__typeof__(*object) _old = *_exp; \
*_exp = __sync_val_compare_and_swap((object), _old, (desired)); \
*_exp == _old; \
})
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
#define atomic_fetch_add(object, operand) \
__sync_fetch_and_add(object, operand)
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub(object, operand) \
__sync_fetch_and_sub(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or(object, operand) \
__sync_fetch_and_or(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor(object, operand) \
__sync_fetch_and_xor(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and(object, operand) \
__sync_fetch_and_and(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_GCC_STDATOMIC_H */
+39
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@@ -0,0 +1,39 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
#include <pthread.h>
#include <stdint.h>
#include "stdatomic.h"
static pthread_mutex_t atomic_lock = PTHREAD_MUTEX_INITIALIZER;
void avpriv_atomic_lock(void)
{
pthread_mutex_lock(&atomic_lock);
}
void avpriv_atomic_unlock(void)
{
pthread_mutex_unlock(&atomic_lock);
}
+197
View File
@@ -0,0 +1,197 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* based on vlc_atomic.h from VLC
* Copyright (C) 2010 Rémi Denis-Courmont
*/
#ifndef COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
#define COMPAT_ATOMICS_PTHREAD_STDATOMIC_H
#include <stdint.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef intptr_t atomic_flag;
typedef intptr_t atomic_bool;
typedef intptr_t atomic_char;
typedef intptr_t atomic_schar;
typedef intptr_t atomic_uchar;
typedef intptr_t atomic_short;
typedef intptr_t atomic_ushort;
typedef intptr_t atomic_int;
typedef intptr_t atomic_uint;
typedef intptr_t atomic_long;
typedef intptr_t atomic_ulong;
typedef intptr_t atomic_llong;
typedef intptr_t atomic_ullong;
typedef intptr_t atomic_wchar_t;
typedef intptr_t atomic_int_least8_t;
typedef intptr_t atomic_uint_least8_t;
typedef intptr_t atomic_int_least16_t;
typedef intptr_t atomic_uint_least16_t;
typedef intptr_t atomic_int_least32_t;
typedef intptr_t atomic_uint_least32_t;
typedef intptr_t atomic_int_least64_t;
typedef intptr_t atomic_uint_least64_t;
typedef intptr_t atomic_int_fast8_t;
typedef intptr_t atomic_uint_fast8_t;
typedef intptr_t atomic_int_fast16_t;
typedef intptr_t atomic_uint_fast16_t;
typedef intptr_t atomic_int_fast32_t;
typedef intptr_t atomic_uint_fast32_t;
typedef intptr_t atomic_int_fast64_t;
typedef intptr_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef intptr_t atomic_uintptr_t;
typedef intptr_t atomic_size_t;
typedef intptr_t atomic_ptrdiff_t;
typedef intptr_t atomic_intmax_t;
typedef intptr_t atomic_uintmax_t;
void avpriv_atomic_lock(void);
void avpriv_atomic_unlock(void);
static inline void atomic_thread_fence(int order)
{
avpriv_atomic_lock();
avpriv_atomic_unlock();
}
static inline void atomic_store(intptr_t *object, intptr_t desired)
{
avpriv_atomic_lock();
*object = desired;
avpriv_atomic_unlock();
}
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
static inline intptr_t atomic_load(intptr_t *object)
{
intptr_t ret;
avpriv_atomic_lock();
ret = *object;
avpriv_atomic_unlock();
return ret;
}
#define atomic_load_explicit(object, order) \
atomic_load(object)
static inline intptr_t atomic_exchange(intptr_t *object, intptr_t desired)
{
intptr_t ret;
avpriv_atomic_lock();
ret = *object;
*object = desired;
avpriv_atomic_unlock();
return ret;
}
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
intptr_t desired)
{
int ret;
avpriv_atomic_lock();
if (*object == *expected) {
ret = 1;
*object = desired;
} else {
ret = 0;
*expected = *object;
}
avpriv_atomic_unlock();
return ret;
}
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
#define FETCH_MODIFY(opname, op) \
static inline intptr_t atomic_fetch_ ## opname(intptr_t *object, intptr_t operand) \
{ \
intptr_t ret; \
avpriv_atomic_lock(); \
ret = *object; \
*object = *object op operand; \
avpriv_atomic_unlock(); \
return ret; \
}
FETCH_MODIFY(add, +)
FETCH_MODIFY(sub, -)
FETCH_MODIFY(or, |)
FETCH_MODIFY(xor, ^)
FETCH_MODIFY(and, &)
#undef FETCH_MODIFY
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_PTHREAD_STDATOMIC_H */
+186
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@@ -0,0 +1,186 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_ATOMICS_SUNCC_STDATOMIC_H
#define COMPAT_ATOMICS_SUNCC_STDATOMIC_H
#include <atomic.h>
#include <mbarrier.h>
#include <stddef.h>
#include <stdint.h>
#define ATOMIC_FLAG_INIT 0
#define ATOMIC_VAR_INIT(value) (value)
#define atomic_init(obj, value) \
do { \
*(obj) = (value); \
} while(0)
#define kill_dependency(y) ((void)0)
#define atomic_thread_fence(order) \
__machine_rw_barrier();
#define atomic_signal_fence(order) \
((void)0)
#define atomic_is_lock_free(obj) 0
typedef intptr_t atomic_flag;
typedef intptr_t atomic_bool;
typedef intptr_t atomic_char;
typedef intptr_t atomic_schar;
typedef intptr_t atomic_uchar;
typedef intptr_t atomic_short;
typedef intptr_t atomic_ushort;
typedef intptr_t atomic_int;
typedef intptr_t atomic_uint;
typedef intptr_t atomic_long;
typedef intptr_t atomic_ulong;
typedef intptr_t atomic_llong;
typedef intptr_t atomic_ullong;
typedef intptr_t atomic_wchar_t;
typedef intptr_t atomic_int_least8_t;
typedef intptr_t atomic_uint_least8_t;
typedef intptr_t atomic_int_least16_t;
typedef intptr_t atomic_uint_least16_t;
typedef intptr_t atomic_int_least32_t;
typedef intptr_t atomic_uint_least32_t;
typedef intptr_t atomic_int_least64_t;
typedef intptr_t atomic_uint_least64_t;
typedef intptr_t atomic_int_fast8_t;
typedef intptr_t atomic_uint_fast8_t;
typedef intptr_t atomic_int_fast16_t;
typedef intptr_t atomic_uint_fast16_t;
typedef intptr_t atomic_int_fast32_t;
typedef intptr_t atomic_uint_fast32_t;
typedef intptr_t atomic_int_fast64_t;
typedef intptr_t atomic_uint_fast64_t;
typedef intptr_t atomic_intptr_t;
typedef intptr_t atomic_uintptr_t;
typedef intptr_t atomic_size_t;
typedef intptr_t atomic_ptrdiff_t;
typedef intptr_t atomic_intmax_t;
typedef intptr_t atomic_uintmax_t;
static inline void atomic_store(intptr_t *object, intptr_t desired)
{
*object = desired;
__machine_rw_barrier();
}
#define atomic_store_explicit(object, desired, order) \
atomic_store(object, desired)
static inline intptr_t atomic_load(intptr_t *object)
{
__machine_rw_barrier();
return *object;
}
#define atomic_load_explicit(object, order) \
atomic_load(object)
#define atomic_exchange(object, desired) \
atomic_swap_ptr(object, desired)
#define atomic_exchange_explicit(object, desired, order) \
atomic_exchange(object, desired)
static inline int atomic_compare_exchange_strong(intptr_t *object, intptr_t *expected,
intptr_t desired)
{
intptr_t old = *expected;
*expected = (intptr_t)atomic_cas_ptr(object, (void *)old, (void *)desired);
return *expected == old;
}
#define atomic_compare_exchange_strong_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak(object, expected, desired) \
atomic_compare_exchange_strong(object, expected, desired)
#define atomic_compare_exchange_weak_explicit(object, expected, desired, success, failure) \
atomic_compare_exchange_weak(object, expected, desired)
static inline intptr_t atomic_fetch_add(intptr_t *object, intptr_t operand)
{
return atomic_add_ptr_nv(object, operand) - operand;
}
#define atomic_fetch_sub(object, operand) \
atomic_fetch_add(object, -(operand))
static inline intptr_t atomic_fetch_or(intptr_t *object, intptr_t operand)
{
intptr_t old;
do {
old = atomic_load(object);
} while (!atomic_compare_exchange_strong(object, old, old | operand));
return old;
}
static inline intptr_t atomic_fetch_xor(intptr_t *object, intptr_t operand)
{
intptr_t old;
do {
old = atomic_load(object);
} while (!atomic_compare_exchange_strong(object, old, old ^ operand));
return old;
}
static inline intptr_t atomic_fetch_and(intptr_t *object, intptr_t operand)
{
intptr_t old;
do {
old = atomic_load(object);
} while (!atomic_compare_exchange_strong(object, old, old & operand));
return old;
}
#define atomic_fetch_add_explicit(object, operand, order) \
atomic_fetch_add(object, operand)
#define atomic_fetch_sub_explicit(object, operand, order) \
atomic_fetch_sub(object, operand)
#define atomic_fetch_or_explicit(object, operand, order) \
atomic_fetch_or(object, operand)
#define atomic_fetch_xor_explicit(object, operand, order) \
atomic_fetch_xor(object, operand)
#define atomic_fetch_and_explicit(object, operand, order) \
atomic_fetch_and(object, operand)
#define atomic_flag_test_and_set(object) \
atomic_exchange(object, 1)
#define atomic_flag_test_and_set_explicit(object, order) \
atomic_flag_test_and_set(object)
#define atomic_flag_clear(object) \
atomic_store(object, 0)
#define atomic_flag_clear_explicit(object, order) \
atomic_flag_clear(object)
#endif /* COMPAT_ATOMICS_SUNCC_STDATOMIC_H */
+1
View File
@@ -19,6 +19,7 @@
#ifndef COMPAT_ATOMICS_WIN32_STDATOMIC_H
#define COMPAT_ATOMICS_WIN32_STDATOMIC_H
#define WIN32_LEAN_AND_MEAN
#include <stddef.h>
#include <stdint.h>
#include <windows.h>
-7
View File
@@ -181,16 +181,9 @@ static inline __device__ double trunc(double a) { return __builtin_trunc(a); }
static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
static inline __device__ float sqrtf(float a) { return __builtin_sqrtf(a); }
static inline __device__ float rintf(float a) { return __builtin_rintf(a); }
static inline __device__ float __saturatef(float a) { return __nvvm_saturate_f(a); }
static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }
static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a); }
static inline __device__ float __expf(float a) { return __nvvm_ex2_approx_f(a * (float)__builtin_log2(__builtin_exp(1))); }
static inline __device__ float __powf(float a, float b) { return __nvvm_ex2_approx_f(__nvvm_lg2_approx_f(a) * b); }
// Misc helper functions
extern "C" __device__ int printf(const char*, ...);
#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */
+1 -1
View File
@@ -38,7 +38,7 @@ static int optind = 1;
static int optopt;
static char *optarg;
static int getopt(int argc, char *argv[], const char *opts)
static int getopt(int argc, char *argv[], char *opts)
{
static int sp = 1;
int c;
+1 -1
View File
@@ -218,7 +218,7 @@ while (<F>) {
# Lines of the form '} SOME_VERSION_NAME_1.0;'
if (/^[ \t]*\}[ \tA-Z0-9_.a-z]+;[ \t]*$/) {
$glob = 'glob';
# We tried to match symbols against this version, but none matched.
# We tried to match symbols agains this version, but none matched.
# Emit dummy hidden symbol to avoid marking this version WEAK.
if ($matches_attempted && $matched_symbols == 0) {
print " hidden:\n";
-599
View File
@@ -1,599 +0,0 @@
/*
* Copyright (C) 2023 Rémi Denis-Courmont
*
* This program 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.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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.
*/
#ifndef __STDC_VERSION_STDBIT_H__
#define __STDC_VERSION_STDBIT_H__ 202311L
#include <stdbool.h>
#include <limits.h> /* CHAR_BIT */
#define __STDC_ENDIAN_LITTLE__ 1234
#define __STDC_ENDIAN_BIG__ 4321
#ifdef __BYTE_ORDER__
# if (__BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
# define __STDC_ENDIAN_NATIVE__ __STDC_ENDIAN_LITTLE__
# elif (__BYTE_ORDER__ == __ORDER_BIG_ENDIAN__)
# define __STDC_ENDIAN_NATIVE__ __STDC_ENDIAN_BIG__
# else
# define __STDC_ENDIAN_NATIVE__ 3412
# endif
#elif defined(_MSC_VER)
# define __STDC_ENDIAN_NATIVE__ __STDC_ENDIAN_LITTLE__
#else
# error Not implemented.
#endif
#define __stdbit_generic_type_func(func, value) \
_Generic (value, \
unsigned long long: stdc_##func##_ull((unsigned long long)(value)), \
unsigned long: stdc_##func##_ul((unsigned long)(value)), \
unsigned int: stdc_##func##_ui((unsigned int)(value)), \
unsigned short: stdc_##func##_us((unsigned short)(value)), \
unsigned char: stdc_##func##_uc((unsigned char)(value)))
#if defined (__GNUC__) || defined (__clang__)
static inline unsigned int stdc_leading_zeros_ull(unsigned long long value)
{
return value ? __builtin_clzll(value) : (CHAR_BIT * sizeof (value));
}
static inline unsigned int stdc_leading_zeros_ul(unsigned long value)
{
return value ? __builtin_clzl(value) : (CHAR_BIT * sizeof (value));
}
static inline unsigned int stdc_leading_zeros_ui(unsigned int value)
{
return value ? __builtin_clz(value) : (CHAR_BIT * sizeof (value));
}
static inline unsigned int stdc_leading_zeros_us(unsigned short value)
{
return stdc_leading_zeros_ui(value)
- CHAR_BIT * (sizeof (int) - sizeof (value));
}
static inline unsigned int stdc_leading_zeros_uc(unsigned char value)
{
return stdc_leading_zeros_ui(value) - (CHAR_BIT * (sizeof (int) - 1));
}
#else
static inline unsigned int __stdc_leading_zeros(unsigned long long value,
unsigned int size)
{
unsigned int zeros = size * CHAR_BIT;
while (value != 0) {
value >>= 1;
zeros--;
}
return zeros;
}
static inline unsigned int stdc_leading_zeros_ull(unsigned long long value)
{
return __stdc_leading_zeros(value, sizeof (value));
}
static inline unsigned int stdc_leading_zeros_ul(unsigned long value)
{
return __stdc_leading_zeros(value, sizeof (value));
}
static inline unsigned int stdc_leading_zeros_ui(unsigned int value)
{
return __stdc_leading_zeros(value, sizeof (value));
}
static inline unsigned int stdc_leading_zeros_us(unsigned short value)
{
return __stdc_leading_zeros(value, sizeof (value));
}
static inline unsigned int stdc_leading_zeros_uc(unsigned char value)
{
return __stdc_leading_zeros(value, sizeof (value));
}
#endif
#define stdc_leading_zeros(value) \
__stdbit_generic_type_func(leading_zeros, value)
static inline unsigned int stdc_leading_ones_ull(unsigned long long value)
{
return stdc_leading_zeros_ull(~value);
}
static inline unsigned int stdc_leading_ones_ul(unsigned long value)
{
return stdc_leading_zeros_ul(~value);
}
static inline unsigned int stdc_leading_ones_ui(unsigned int value)
{
return stdc_leading_zeros_ui(~value);
}
static inline unsigned int stdc_leading_ones_us(unsigned short value)
{
return stdc_leading_zeros_us(~value);
}
static inline unsigned int stdc_leading_ones_uc(unsigned char value)
{
return stdc_leading_zeros_uc(~value);
}
#define stdc_leading_ones(value) \
__stdbit_generic_type_func(leading_ones, value)
#if defined (__GNUC__) || defined (__clang__)
static inline unsigned int stdc_trailing_zeros_ull(unsigned long long value)
{
return value ? (unsigned int)__builtin_ctzll(value)
: (CHAR_BIT * sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_ul(unsigned long value)
{
return value ? (unsigned int)__builtin_ctzl(value)
: (CHAR_BIT * sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_ui(unsigned int value)
{
return value ? (unsigned int)__builtin_ctz(value)
: (CHAR_BIT * sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_us(unsigned short value)
{
return value ? (unsigned int)__builtin_ctz(value)
: (CHAR_BIT * sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_uc(unsigned char value)
{
return value ? (unsigned int)__builtin_ctz(value)
: (CHAR_BIT * sizeof (value));
}
#else
static inline unsigned int __stdc_trailing_zeros(unsigned long long value,
unsigned int size)
{
unsigned int zeros = 0;
if (!value)
return size * CHAR_BIT;
while ((value & 1) == 0) {
value >>= 1;
zeros++;
}
return zeros;
}
static inline unsigned int stdc_trailing_zeros_ull(unsigned long long value)
{
return __stdc_trailing_zeros(value, sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_ul(unsigned long value)
{
return __stdc_trailing_zeros(value, sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_ui(unsigned int value)
{
return __stdc_trailing_zeros(value, sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_us(unsigned short value)
{
return __stdc_trailing_zeros(value, sizeof (value));
}
static inline unsigned int stdc_trailing_zeros_uc(unsigned char value)
{
return __stdc_trailing_zeros(value, sizeof (value));
}
#endif
#define stdc_trailing_zeros(value) \
__stdbit_generic_type_func(trailing_zeros, value)
static inline unsigned int stdc_trailing_ones_ull(unsigned long long value)
{
return stdc_trailing_zeros_ull(~value);
}
static inline unsigned int stdc_trailing_ones_ul(unsigned long value)
{
return stdc_trailing_zeros_ul(~value);
}
static inline unsigned int stdc_trailing_ones_ui(unsigned int value)
{
return stdc_trailing_zeros_ui(~value);
}
static inline unsigned int stdc_trailing_ones_us(unsigned short value)
{
return stdc_trailing_zeros_us(~value);
}
static inline unsigned int stdc_trailing_ones_uc(unsigned char value)
{
return stdc_trailing_zeros_uc(~value);
}
#define stdc_trailing_ones(value) \
__stdbit_generic_type_func(trailing_ones, value)
static inline unsigned int stdc_first_leading_one_ull(unsigned long long value)
{
return value ? (stdc_leading_zeros_ull(value) + 1) : 0;
}
static inline unsigned int stdc_first_leading_one_ul(unsigned long value)
{
return value ? (stdc_leading_zeros_ul(value) + 1) : 0;
}
static inline unsigned int stdc_first_leading_one_ui(unsigned int value)
{
return value ? (stdc_leading_zeros_ui(value) + 1) : 0;
}
static inline unsigned int stdc_first_leading_one_us(unsigned short value)
{
return value ? (stdc_leading_zeros_us(value) + 1) : 0;
}
static inline unsigned int stdc_first_leading_one_uc(unsigned char value)
{
return value ? (stdc_leading_zeros_uc(value) + 1) : 0;
}
#define stdc_first_leading_one(value) \
__stdbit_generic_type_func(first_leading_one, value)
static inline unsigned int stdc_first_leading_zero_ull(unsigned long long value)
{
return stdc_leading_ones_ull(~value);
}
static inline unsigned int stdc_first_leading_zero_ul(unsigned long value)
{
return stdc_leading_ones_ul(~value);
}
static inline unsigned int stdc_first_leading_zero_ui(unsigned int value)
{
return stdc_leading_ones_ui(~value);
}
static inline unsigned int stdc_first_leading_zero_us(unsigned short value)
{
return stdc_leading_ones_us(~value);
}
static inline unsigned int stdc_first_leading_zero_uc(unsigned char value)
{
return stdc_leading_ones_uc(~value);
}
#define stdc_first_leading_zero(value) \
__stdbit_generic_type_func(first_leading_zero, value)
#if defined (__GNUC__) || defined (__clang__)
static inline unsigned int stdc_first_trailing_one_ull(unsigned long long value)
{
return __builtin_ffsll(value);
}
static inline unsigned int stdc_first_trailing_one_ul(unsigned long value)
{
return __builtin_ffsl(value);
}
static inline unsigned int stdc_first_trailing_one_ui(unsigned int value)
{
return __builtin_ffs(value);
}
static inline unsigned int stdc_first_trailing_one_us(unsigned short value)
{
return __builtin_ffs(value);
}
static inline unsigned int stdc_first_trailing_one_uc(unsigned char value)
{
return __builtin_ffs(value);
}
#else
static inline unsigned int stdc_first_trailing_one_ull(unsigned long long value)
{
return value ? (1 + stdc_trailing_zeros_ull(value)) : 0;
}
static inline unsigned int stdc_first_trailing_one_ul(unsigned long value)
{
return value ? (1 + stdc_trailing_zeros_ul(value)) : 0;
}
static inline unsigned int stdc_first_trailing_one_ui(unsigned int value)
{
return value ? (1 + stdc_trailing_zeros_ui(value)) : 0;
}
static inline unsigned int stdc_first_trailing_one_us(unsigned short value)
{
return value ? (1 + stdc_trailing_zeros_us(value)) : 0;
}
static inline unsigned int stdc_first_trailing_one_uc(unsigned char value)
{
return value ? (1 + stdc_trailing_zeros_uc(value)) : 0;
}
#endif
#define stdc_first_trailing_one(value) \
__stdbit_generic_type_func(first_trailing_one, value)
static inline unsigned int stdc_first_trailing_zero_ull(unsigned long long value)
{
return stdc_first_trailing_one_ull(~value);
}
static inline unsigned int stdc_first_trailing_zero_ul(unsigned long value)
{
return stdc_first_trailing_one_ul(~value);
}
static inline unsigned int stdc_first_trailing_zero_ui(unsigned int value)
{
return stdc_first_trailing_one_ui(~value);
}
static inline unsigned int stdc_first_trailing_zero_us(unsigned short value)
{
return stdc_first_trailing_one_us(~value);
}
static inline unsigned int stdc_first_trailing_zero_uc(unsigned char value)
{
return stdc_first_trailing_one_uc(~value);
}
#define stdc_first_trailing_zero(value) \
__stdbit_generic_type_func(first_trailing_zero, value)
#if defined (__GNUC__) || defined (__clang__)
static inline unsigned int stdc_count_ones_ull(unsigned long long value)
{
return __builtin_popcountll(value);
}
static inline unsigned int stdc_count_ones_ul(unsigned long value)
{
return __builtin_popcountl(value);
}
static inline unsigned int stdc_count_ones_ui(unsigned int value)
{
return __builtin_popcount(value);
}
static inline unsigned int stdc_count_ones_us(unsigned short value)
{
return __builtin_popcount(value);
}
static inline unsigned int stdc_count_ones_uc(unsigned char value)
{
return __builtin_popcount(value);
}
#else
static inline unsigned int __stdc_count_ones(unsigned long long value,
unsigned int size)
{
unsigned int ones = 0;
for (unsigned int c = 0; c < (size * CHAR_BIT); c++) {
ones += value & 1;
value >>= 1;
}
return ones;
}
static inline unsigned int stdc_count_ones_ull(unsigned long long value)
{
return __stdc_count_ones(value, sizeof (value));
}
static inline unsigned int stdc_count_ones_ul(unsigned long value)
{
return __stdc_count_ones(value, sizeof (value));
}
static inline unsigned int stdc_count_ones_ui(unsigned int value)
{
return __stdc_count_ones(value, sizeof (value));
}
static inline unsigned int stdc_count_ones_us(unsigned short value)
{
return __stdc_count_ones(value, sizeof (value));
}
static inline unsigned int stdc_count_ones_uc(unsigned char value)
{
return __stdc_count_ones(value, sizeof (value));
}
#endif
#define stdc_count_ones(value) \
__stdbit_generic_type_func(count_ones, value)
static inline unsigned int stdc_count_zeros_ull(unsigned long long value)
{
return stdc_count_ones_ull(~value);
}
static inline unsigned int stdc_count_zeros_ul(unsigned long value)
{
return stdc_count_ones_ul(~value);
}
static inline unsigned int stdc_count_zeros_ui(unsigned int value)
{
return stdc_count_ones_ui(~value);
}
static inline unsigned int stdc_count_zeros_us(unsigned short value)
{
return stdc_count_ones_us(~value);
}
static inline unsigned int stdc_count_zeros_uc(unsigned char value)
{
return stdc_count_ones_uc(~value);
}
#define stdc_count_zeros(value) \
__stdbit_generic_type_func(count_zeros, value)
static inline bool stdc_has_single_bit_ull(unsigned long long value)
{
return value && (value & (value - 1)) == 0;
}
static inline bool stdc_has_single_bit_ul(unsigned long value)
{
return value && (value & (value - 1)) == 0;
}
static inline bool stdc_has_single_bit_ui(unsigned int value)
{
return value && (value & (value - 1)) == 0;
}
static inline bool stdc_has_single_bit_us(unsigned short value)
{
return value && (value & (value - 1)) == 0;
}
static inline bool stdc_has_single_bit_uc(unsigned char value)
{
return value && (value & (value - 1)) == 0;
}
#define stdc_has_single_bit(value) \
__stdbit_generic_type_func(has_single_bit, value)
static inline unsigned int stdc_bit_width_ull(unsigned long long value)
{
return (CHAR_BIT * sizeof (value)) - stdc_leading_zeros_ull(value);
}
static inline unsigned int stdc_bit_width_ul(unsigned long value)
{
return (CHAR_BIT * sizeof (value)) - stdc_leading_zeros_ul(value);
}
static inline unsigned int stdc_bit_width_ui(unsigned int value)
{
return (CHAR_BIT * sizeof (value)) - stdc_leading_zeros_ui(value);
}
static inline unsigned int stdc_bit_width_us(unsigned short value)
{
return (CHAR_BIT * sizeof (value)) - stdc_leading_zeros_us(value);
}
static inline unsigned int stdc_bit_width_uc(unsigned char value)
{
return (CHAR_BIT * sizeof (value)) - stdc_leading_zeros_uc(value);
}
#define stdc_bit_width(value) \
__stdbit_generic_type_func(bit_width, value)
static inline unsigned long long stdc_bit_floor_ull(unsigned long long value)
{
return value ? (1ULL << (stdc_bit_width_ull(value) - 1)) : 0ULL;
}
static inline unsigned long stdc_bit_floor_ul(unsigned long value)
{
return value ? (1UL << (stdc_bit_width_ul(value) - 1)) : 0UL;
}
static inline unsigned int stdc_bit_floor_ui(unsigned int value)
{
return value ? (1U << (stdc_bit_width_ui(value) - 1)) : 0U;
}
static inline unsigned short stdc_bit_floor_us(unsigned short value)
{
return value ? (1U << (stdc_bit_width_us(value) - 1)) : 0U;
}
static inline unsigned int stdc_bit_floor_uc(unsigned char value)
{
return value ? (1U << (stdc_bit_width_uc(value) - 1)) : 0U;
}
#define stdc_bit_floor(value) \
__stdbit_generic_type_func(bit_floor, value)
/* NOTE: Bit ceiling undefines overflow. */
static inline unsigned long long stdc_bit_ceil_ull(unsigned long long value)
{
return 1ULL << (value ? stdc_bit_width_ull(value - 1) : 0);
}
static inline unsigned long stdc_bit_ceil_ul(unsigned long value)
{
return 1UL << (value ? stdc_bit_width_ul(value - 1) : 0);
}
static inline unsigned int stdc_bit_ceil_ui(unsigned int value)
{
return 1U << (value ? stdc_bit_width_ui(value - 1) : 0);
}
static inline unsigned short stdc_bit_ceil_us(unsigned short value)
{
return 1U << (value ? stdc_bit_width_us(value - 1) : 0);
}
static inline unsigned int stdc_bit_ceil_uc(unsigned char value)
{
return 1U << (value ? stdc_bit_width_uc(value - 1) : 0);
}
#define stdc_bit_ceil(value) \
__stdbit_generic_type_func(bit_ceil, value)
#endif /* __STDC_VERSION_STDBIT_H__ */
+23 -80
View File
@@ -20,41 +20,11 @@
#define COMPAT_W32DLFCN_H
#ifdef _WIN32
#include <stdint.h>
#include <windows.h>
#include "config.h"
#include "libavutil/macros.h"
#include "libavutil/mem.h"
#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
#include "libavutil/wchar_filename.h"
static inline wchar_t *get_module_filename(HMODULE module)
{
wchar_t *path = NULL, *new_path;
DWORD path_size = 0, path_len;
do {
path_size = path_size ? FFMIN(2 * path_size, INT16_MAX + 1) : MAX_PATH;
new_path = av_realloc_array(path, path_size, sizeof *path);
if (!new_path) {
av_free(path);
return NULL;
}
path = new_path;
// Returns path_size in case of insufficient buffer.
// Whether the error is set or not and whether the output
// is null-terminated or not depends on the version of Windows.
path_len = GetModuleFileNameW(module, path, path_size);
} while (path_len && path_size <= INT16_MAX && path_size <= path_len);
if (!path_len) {
av_free(path);
return NULL;
}
return path;
}
#endif
/**
* Safe function used to open dynamic libs. This attempts to improve program security
* by removing the current directory from the dll search path. Only dll's found in the
@@ -64,53 +34,29 @@ static inline wchar_t *get_module_filename(HMODULE module)
*/
static inline HMODULE win32_dlopen(const char *name)
{
wchar_t *name_w;
HMODULE module = NULL;
if (utf8towchar(name, &name_w))
name_w = NULL;
#if _WIN32_WINNT < 0x0602
// On Win7 and earlier we check if KB2533623 is available
// Need to check if KB2533623 is available
if (!GetProcAddress(GetModuleHandleW(L"kernel32.dll"), "SetDefaultDllDirectories")) {
wchar_t *path = NULL, *new_path;
DWORD pathlen, pathsize, namelen;
if (!name_w)
HMODULE module = NULL;
wchar_t *path = NULL, *name_w = NULL;
DWORD pathlen;
if (utf8towchar(name, &name_w))
goto exit;
namelen = wcslen(name_w);
path = (wchar_t *)av_calloc(MAX_PATH, sizeof(wchar_t));
// Try local directory first
path = get_module_filename(NULL);
if (!path)
pathlen = GetModuleFileNameW(NULL, path, MAX_PATH);
pathlen = wcsrchr(path, '\\') - path;
if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
goto exit;
new_path = wcsrchr(path, '\\');
if (!new_path)
goto exit;
pathlen = new_path - path;
pathsize = pathlen + namelen + 2;
new_path = av_realloc_array(path, pathsize, sizeof *path);
if (!new_path)
goto exit;
path = new_path;
path[pathlen] = '\\';
wcscpy(path + pathlen + 1, name_w);
module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
if (module == NULL) {
// Next try System32 directory
pathlen = GetSystemDirectoryW(path, pathsize);
if (!pathlen)
pathlen = GetSystemDirectoryW(path, MAX_PATH);
if (pathlen == 0 || pathlen + wcslen(name_w) + 2 > MAX_PATH)
goto exit;
// Buffer is not enough in two cases:
// 1. system directory + \ + module name
// 2. system directory even without the module name.
if (pathlen + namelen + 2 > pathsize) {
pathsize = pathlen + namelen + 2;
new_path = av_realloc_array(path, pathsize, sizeof *path);
if (!new_path)
goto exit;
path = new_path;
// Query again to handle the case #2.
pathlen = GetSystemDirectoryW(path, pathsize);
if (!pathlen)
goto exit;
}
path[pathlen] = L'\\';
path[pathlen] = '\\';
wcscpy(path + pathlen + 1, name_w);
module = LoadLibraryExW(path, NULL, LOAD_WITH_ALTERED_SEARCH_PATH);
}
@@ -127,19 +73,16 @@ exit:
# define LOAD_LIBRARY_SEARCH_SYSTEM32 0x00000800
#endif
#if HAVE_WINRT
if (!name_w)
wchar_t *name_w = NULL;
int ret;
if (utf8towchar(name, &name_w))
return NULL;
module = LoadPackagedLibrary(name_w, 0);
#else
#define LOAD_FLAGS (LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32)
/* filename may be be in CP_ACP */
if (!name_w)
return LoadLibraryExA(name, NULL, LOAD_FLAGS);
module = LoadLibraryExW(name_w, NULL, LOAD_FLAGS);
#undef LOAD_FLAGS
#endif
ret = LoadPackagedLibrary(name_w, 0);
av_free(name_w);
return module;
return ret;
#else
return LoadLibraryExA(name, NULL, LOAD_LIBRARY_SEARCH_APPLICATION_DIR | LOAD_LIBRARY_SEARCH_SYSTEM32);
#endif
}
#define dlopen(name, flags) win32_dlopen(name)
#define dlclose FreeLibrary
+18 -74
View File
@@ -35,6 +35,7 @@
* As most functions here are used without checking return values,
* only implement return values as necessary. */
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <process.h>
#include <time.h>
@@ -44,14 +45,13 @@
#include "libavutil/internal.h"
#include "libavutil/mem.h"
#include "libavutil/time.h"
#include "libavutil/wchar_filename.h"
typedef struct w32pthread_t {
typedef struct pthread_t {
void *handle;
void *(*func)(void* arg);
void *arg;
void *ret;
} *pthread_t;
} pthread_t;
/* use light weight mutex/condition variable API for Windows Vista and later */
typedef SRWLOCK pthread_mutex_t;
@@ -66,52 +66,31 @@ typedef CONDITION_VARIABLE pthread_cond_t;
#define PTHREAD_CANCEL_ENABLE 1
#define PTHREAD_CANCEL_DISABLE 0
#if HAVE_WINRT
#define THREADFUNC_RETTYPE DWORD
#else
#define THREADFUNC_RETTYPE unsigned
#endif
av_unused static THREADFUNC_RETTYPE
__stdcall attribute_align_arg win32thread_worker(void *arg)
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t h = (pthread_t)arg;
pthread_t *h = (pthread_t*)arg;
h->ret = h->func(h->arg);
return 0;
}
av_unused static int pthread_create(pthread_t *thread, const void *unused_attr,
static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
{
pthread_t ret;
ret = (pthread_t)av_mallocz(sizeof(*ret));
if (!ret)
return EAGAIN;
ret->func = start_routine;
ret->arg = arg;
thread->func = start_routine;
thread->arg = arg;
#if HAVE_WINRT
ret->handle = (void*)CreateThread(NULL, 0, win32thread_worker, ret,
0, NULL);
thread->handle = (void*)CreateThread(NULL, 0, win32thread_worker, thread,
0, NULL);
#else
ret->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, ret,
0, NULL);
thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
0, NULL);
#endif
if (!ret->handle) {
av_free(ret);
return EAGAIN;
}
*thread = ret;
return 0;
return !thread->handle;
}
av_unused static int pthread_join(pthread_t thread, void **value_ptr)
static av_unused int pthread_join(pthread_t thread, void **value_ptr)
{
DWORD ret = WaitForSingleObject(thread->handle, INFINITE);
DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
if (ret != WAIT_OBJECT_0) {
if (ret == WAIT_ABANDONED)
return EINVAL;
@@ -119,9 +98,8 @@ av_unused static int pthread_join(pthread_t thread, void **value_ptr)
return EDEADLK;
}
if (value_ptr)
*value_ptr = thread->ret;
CloseHandle(thread->handle);
av_free(thread);
*value_ptr = thread.ret;
CloseHandle(thread.handle);
return 0;
}
@@ -149,7 +127,7 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
av_unused static int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
BOOL pending = FALSE;
InitOnceBeginInitialize(once_control, 0, &pending, NULL);
@@ -210,38 +188,4 @@ static inline int pthread_setcancelstate(int state, int *oldstate)
return 0;
}
static inline int win32_thread_setname(const char *name)
{
#if !HAVE_UWP
typedef HRESULT (WINAPI *SetThreadDescriptionFn)(HANDLE, PCWSTR);
// Although SetThreadDescription lives in kernel32.dll, on Windows Server 2016,
// Windows 10 LTSB 2016 and Windows 10 version 1607, it was only available in
// kernelbase.dll. So, load it from there for maximum coverage.
HMODULE kernelbase = GetModuleHandleW(L"kernelbase.dll");
if (!kernelbase)
return AVERROR(ENOSYS);
SetThreadDescriptionFn pSetThreadDescription =
(SetThreadDescriptionFn)GetProcAddress(kernelbase, "SetThreadDescription");
if (!pSetThreadDescription)
return AVERROR(ENOSYS);
wchar_t *wname;
if (utf8towchar(name, &wname) < 0)
return AVERROR(ENOMEM);
HRESULT hr = pSetThreadDescription(GetCurrentThread(), wname);
av_free(wname);
return SUCCEEDED(hr) ? 0 : AVERROR(EINVAL);
#else
// UWP is not supported because we cannot use LoadLibrary/GetProcAddress to
// detect the availability of the SetThreadDescription API. There is a small
// gap in Windows builds 1507-1607 where it was not available. UWP allows
// querying the availability of APIs with QueryOptionalDelayLoadedAPI, but it
// requires /DELAYLOAD:kernel32.dll during linking, and we cannot enforce that.
return AVERROR(ENOSYS);
#endif
}
#endif /* COMPAT_W32PTHREADS_H */
+5 -25
View File
@@ -33,11 +33,6 @@ if [ ! -f "$vscript" ]; then
fi
for object in "$@"; do
case "$object" in
@*)
object="${object#@}"
;;
esac
if [ ! -f "$object" ]; then
echo "Object does not exist: ${object}" >&2
exit 1
@@ -53,13 +48,7 @@ trap 'rm -f -- $libname' EXIT
if [ -n "$AR" ]; then
$AR rcs ${libname} $@ >/dev/null
else
machine_flag=""
case "$LDFLAGS" in
*"machine:arm64ec"*)
machine_flag="-machine:arm64ec"
;;
esac
lib.exe ${machine_flag} -out:${libname} $@ >/dev/null
lib.exe -out:${libname} $@ >/dev/null
fi
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
@@ -117,21 +106,12 @@ if [ -n "$NM" ]; then
grep -v : |
grep -v ^$ |
cut -d' ' -f3 |
sed -e "s/^${prefix}//" -e "s/^#//" |
grep -v '\$entry_thunk' |
grep -v '\$exit_thunk')
sed -e "s/^${prefix}//")
else
member=1
case "$LDFLAGS" in
*"machine:arm64ec"*)
member=32
;;
esac
dump=$(dumpbin.exe -linkermember:${member} ${libname} |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e 's/^[[:space:]]*[0-9A-Fa-f]\{1,\}[[:space:]]\{1,\}//' -e "s/^${prefix}//" -e 's/^#//' |
dump=$(dumpbin.exe -linkermember:1 ${libname} |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
grep -v '\$entry_thunk' |
grep -v '\$exit_thunk')
cut -d' ' -f3)
fi
rm ${libname}
-32
View File
@@ -1,32 +0,0 @@
#!/bin/sh
if [ "$1" = "--version" ]; then
rc.exe -?
exit $?
fi
if [ $# -lt 2 ]; then
echo "Usage: mswindres [-I/include/path ...] [-DSOME_DEFINE ...] [-o output.o] input.rc [output.o]" >&2
exit 0
fi
EXTRA_OPTS="-nologo"
while [ $# -gt 2 ]; do
case $1 in
-D*) EXTRA_OPTS="$EXTRA_OPTS -d$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-I*) EXTRA_OPTS="$EXTRA_OPTS -i$(echo $1 | sed -e "s/^..//" -e "s/ /\\\\ /g")" ;;
-o) OPT_OUT="$2"; shift ;;
esac
shift
done
IN="$1"
if [ -z "$OPT_OUT" ]; then
OUT="$2"
else
OUT="$OPT_OUT"
fi
eval set -- $EXTRA_OPTS
rc.exe "$@" -fo "$OUT" "$IN"
Vendored
+862 -1989
View File
File diff suppressed because it is too large Load Diff
+14 -1098
View File
File diff suppressed because it is too large Load Diff
-40
View File
@@ -1,40 +0,0 @@
This document is work in progress
*What is CVSS*
The Common Vulnerability Scoring System (CVSS) is an open, industry-standard framework used to measure and communicate the severity of software vulnerabilities, ranging from 0.0 to 10.0.
*Why we need this Document*
It is important that FFmpeg CVEs have consistent and correct CVSS, not only for the obvious reason that one can recognize the severity of an issue at first glance.
But also as these numbers form the basis of rewards paid in bug bounty systems. Inconsistent CVSS could lead to unfair payouts.
*What is this Document*
Prior 2026, FFmpeg had no guideline about CVSS.
This document describes how to select the CVSS for a FFmpeg related CVE. It currently only covers the Base Score.
*What is the CVSS Base Score*
AV Attack Vector (Network, Adjacent, Local, Physical)
AC Attack Complexity (Low, High)
PR Privileges Required (None, Low, High)
UI User Interaction (None, Required)
S Scope (Unchanged, Changed)
C Confidentiality (None, Low, High)
I Integrity (None, Low, High)
A Availability (None, Low, High)
*Things people have set incorrectly*
Below are general guidelines and in specific cases other things may apply.
Attack Vector.
Quote from https://www.first.org/cvss/v3.1/user-guide
"Specifically, analysts should only score for Network or Adjacent when a vulnerability is bound to the network stack.
Vulnerabilities which require user interaction to download or receive malicious content (which could also be delivered locally, e.g., via USB drives) should be scored as Local."
Availability.
FFmpeg Crashes -> AVAILABILITY IMPACT: Low
FFmpeg is frequently used as a short-lived, single-run process instead of a continuously running service that handles ongoing streams of user input. In that usage model, a crash usually causes only limited disruption.
User Interaction
Please consider if an attacker can actually set the parameters required for an attack.
In general arbitrary filter parameters cannot be set by an attacker and require the user/account owner/admin to set them
+4 -5
View File
@@ -38,7 +38,7 @@ PROJECT_NAME = FFmpeg
# could be handy for archiving the generated documentation or if some version
# control system is used.
PROJECT_NUMBER =
PROJECT_NUMBER = 5.0
# Using the PROJECT_BRIEF tag one can provide an optional one line description
# for a project that appears at the top of each page and should give viewer a
@@ -295,7 +295,7 @@ MARKDOWN_SUPPORT = YES
# When enabled doxygen tries to link words that correspond to documented
# classes, or namespaces to their corresponding documentation. Such a link can
# be prevented in individual cases by putting a % sign in front of the word
# be prevented in individual cases by by putting a % sign in front of the word
# or globally by setting AUTOLINK_SUPPORT to NO.
# The default value is: YES.
@@ -1093,7 +1093,7 @@ HTML_STYLESHEET =
# cascading style sheets that are included after the standard style sheets
# created by doxygen. Using this option one can overrule certain style aspects.
# This is preferred over using HTML_STYLESHEET since it does not replace the
# standard style sheet and is therefore more robust against future updates.
# standard style sheet and is therefor more robust against future updates.
# Doxygen will copy the style sheet files to the output directory.
# Note: The order of the extra stylesheet files is of importance (e.g. the last
# stylesheet in the list overrules the setting of the previous ones in the
@@ -1636,7 +1636,7 @@ EXTRA_PACKAGES =
# Note: Only use a user-defined header if you know what you are doing! The
# following commands have a special meaning inside the header: $title,
# $datetime, $date, $doxygenversion, $projectname, $projectnumber,
# $projectbrief, $projectlogo. Doxygen will replace $title with the empty string,
# $projectbrief, $projectlogo. Doxygen will replace $title with the empy string,
# for the replacement values of the other commands the user is referred to
# HTML_HEADER.
# This tag requires that the tag GENERATE_LATEX is set to YES.
@@ -1980,7 +1980,6 @@ PREDEFINED = __attribute__(x)= \
av_alloc_size(...)= \
AV_GCC_VERSION_AT_LEAST(x,y)=1 \
AV_GCC_VERSION_AT_MOST(x,y)=0 \
"FF_PAD_STRUCTURE(name,size,...)=typedef struct name { __VA_ARGS__ } name;" \
__GNUC__
# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then this
+4 -9
View File
@@ -19,7 +19,6 @@ 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) \
doc/community.html \
doc/developer.html \
doc/faq.html \
doc/fate.html \
@@ -28,10 +27,6 @@ HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
doc/mailing-list-faq.html \
doc/nut.html \
doc/platform.html \
doc/drawvg-reference.html \
$(SRC_PATH)/doc/bootstrap.min.css \
$(SRC_PATH)/doc/style.min.css \
$(SRC_PATH)/doc/default.css \
TXTPAGES = doc/fate.txt \
@@ -54,14 +49,14 @@ TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
$(Q)$(TEXIDEP)
$(M)$(MAKEINFO) --force --no-headers -o $@ $< 2>/dev/null
$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
GENTEXI = format codec
GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
$(GENTEXI): TAG = GENTEXI
$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options$(HOSTEXESUF) $* > $@
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%-all.html: TAG = HTML
@@ -69,11 +64,11 @@ doc/%-all.html: TAG = HTML
ifdef HAVE_MAKEINFO_HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)$(MAKEINFO) --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)$(MAKEINFO) --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
else
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
+2 -2
View File
@@ -3,9 +3,9 @@
The FFmpeg developers.
For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
@command{git log} in the FFmpeg source directory, or browsing the
online repository at @url{https://git.ffmpeg.org/ffmpeg}.
online repository at @url{http://source.ffmpeg.org}.
Maintainers for the specific components are listed in the file
@file{MAINTAINERS} in the source code tree.
+12 -253
View File
@@ -101,56 +101,6 @@ Remove zero padding at the end of a packet.
Extract the core from a DCA/DTS stream, dropping extensions such as
DTS-HD.
@section dovi_rpu
Manipulate Dolby Vision metadata in a HEVC/AV1 bitstream, optionally enabling
metadata compression.
@table @option
@item strip
If enabled, strip all Dolby Vision metadata (configuration record + RPU data
blocks) from the stream.
@item compression
Which compression level to enable.
@table @samp
@item none
No metadata compression.
@item limited
Limited metadata compression scheme. Should be compatible with most devices.
This is the default.
@item extended
Extended metadata compression. Devices are not required to support this. Note
that this level currently behaves the same as @samp{limited} in libavcodec.
@end table
@end table
@section dovi_split
Split a Dolby Vision Profile 7 multi-layer HEVC bitstream. Profile 7 carries the
enhancement-layer HEVC bitstream interleaved inside the base-layer access units,
wrapped in user-unspecified NAL units of type 63 (UNSPEC63), and the RPU metadata
as a sibling user-unspecified NAL of type 62 (UNSPEC62).
@table @option
@item mode
Which Dolby Vision components to keep in the output bitstream.
@table @samp
@item bl
Base layer only: drop every UNSPEC63 (EL) and every UNSPEC62 (RPU).
The output is a plain HEVC stream with no Dolby Vision markers. This is the
default.
@item bl_rpu
Base layer with the RPU NAL kept.
@item el
Enhancement layer only: for every UNSPEC63 NAL, strip the two-byte outer NAL
header and emit the inner payload. The result is a standalone HEVC bitstream.
UNSPEC62 (RPU) is dropped.
@item el_rpu
Enhancement layer with the RPU NAL kept verbatim. Same as @samp{el}, but the
UNSPEC62 RPU NALs are also emitted alongside the unwrapped EL NALs.
@end table
@end table
@section dump_extra
Add extradata to the beginning of the filtered packets except when
@@ -182,86 +132,10 @@ the header stored in extradata to the key packets:
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section dv_error_marker
Blocks in DV which are marked as damaged are replaced by blocks of the specified color.
@table @option
@item color
The color to replace damaged blocks by
@item sta
A 16 bit mask which specifies which of the 16 possible error status values are
to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0
error status values.
@table @samp
@item ok
No error, no concealment
@item err
Error, No concealment
@item res
Reserved
@item notok
Error or concealment
@item notres
Not reserved
@item Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
The specific error status code
@end table
see page 44-46 or section 5.5 of
@url{http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf}
@end table
@section eac3_core
Extract the core from a E-AC-3 stream, dropping extra channels.
@section eia608_to_smpte436m
Convert from a @code{EIA_608} stream to a @code{SMPTE_436M_ANC} data stream, wrapping the closed captions in CTA-708 CDP VANC packets.
@table @option
@item line_number
Choose which line number the generated VANC packets should go on. You generally want either line 9 (the default) or 11.
@item wrapping_type
Choose the SMPTE 436M wrapping type, defaults to @samp{vanc_frame}.
It accepts the values:
@table @samp
@item vanc_frame
VANC frame (interlaced or segmented progressive frame)
@item vanc_field_1
@item vanc_field_2
@item vanc_progressive_frame
@end table
@item sample_coding
Choose the SMPTE 436M sample coding, defaults to @samp{8bit_luma}.
It accepts the values:
@table @samp
@item 8bit_luma
8-bit component luma samples
@item 8bit_color_diff
8-bit component color difference samples
@item 8bit_luma_and_color_diff
8-bit component luma and color difference samples
@item 10bit_luma
10-bit component luma samples
@item 10bit_color_diff
10-bit component color difference samples
@item 10bit_luma_and_color_diff
10-bit component luma and color difference samples
@item 8bit_luma_parity_error
8-bit component luma samples with parity error
@item 8bit_color_diff_parity_error
8-bit component color difference samples with parity error
@item 8bit_luma_and_color_diff_parity_error
8-bit component luma and color difference samples with parity error
@end table
@item initial_cdp_sequence_cntr
The initial value of the CDP's 16-bit unsigned integer @code{cdp_hdr_sequence_cntr} and @code{cdp_ftr_sequence_cntr} fields. Defaults to 0.
@item cdp_frame_rate
Set the CDP's @code{cdp_frame_rate} field. This doesn't actually change the timing of the data stream, it just changes the values inserted in that field in the generated CDP packets. Defaults to @samp{30000/1001}.
@end table
@section extract_extradata
Extract the in-band extradata.
@@ -295,13 +169,6 @@ Identical to @option{pass_types}, except the units in the given set
removed and all others passed through.
@end table
The types used by pass_types and remove_types correspond to NAL unit types
(nal_unit_type) in H.264, HEVC and H.266 (see Table 7-1 in the H.264
and HEVC specifications or Table 5 in the H.266 specification), to
marker values for JPEG (without 0xFF prefix) and to start codes without
start code prefix (i.e. the byte following the 0x000001) for MPEG-2.
For VP8 and VP9, every unit has type zero.
Extradata is unchanged by this transformation, but note that if the stream
contains inline parameter sets then the output may be unusable if they are
removed.
@@ -316,21 +183,6 @@ To remove all AUDs, SEI and filler from an H.265 stream:
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
@end example
To remove all user data from a MPEG-2 stream, including Closed Captions:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=178' OUTPUT
@end example
To remove all SEI from a H264 stream, including Closed Captions:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=6' OUTPUT
@end example
To remove all prefix and suffix SEI from a HEVC stream, including Closed Captions and dynamic HDR:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=39|40' OUTPUT
@end example
@section hapqa_extract
Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
@@ -365,16 +217,12 @@ Modify metadata embedded in an H.264 stream.
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item pass
@item insert
@item remove
@end table
Default is pass.
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
See H.264 table E-1.
@item overscan_appropriate_flag
Set whether the stream is suitable for display using overscan
@@ -433,37 +281,6 @@ insert the string ``hello'' associated with the given UUID.
@item delete_filler
Deletes both filler NAL units and filler SEI messages.
@item display_orientation
Insert, extract or remove Display orientation SEI messages.
See H.264 section D.1.27 and D.2.27 for syntax and semantics.
@table @samp
@item pass
@item insert
@item remove
@item extract
@end table
Default is pass.
Insert mode works in conjunction with @code{rotate} and @code{flip} options.
Any pre-existing Display orientation messages will be removed in insert or remove mode.
Extract mode attaches the display matrix to the packet as side data.
@item rotate
Set rotation in display orientation SEI (anticlockwise angle in degrees).
Range is -360 to +360. Default is NaN.
@item flip
Set flip in display orientation SEI.
@table @samp
@item horizontal
@item vertical
@end table
Default is unset.
@item level
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1
to A-5.
@@ -496,21 +313,12 @@ Please note that this filter is auto-inserted for MPEG-TS (muxer
@section h264_redundant_pps
This applies a specific fixup to some Blu-ray BDMV H264 streams
which contain redundant PPSs. The PPSs modify irrelevant parameters
of the stream, confusing other transformations which require
the correct extradata.
This applies a specific fixup to some Blu-ray streams which contain
redundant PPSs modifying irrelevant parameters of the stream which
confuse other transformations which require correct extradata.
The encoder used on these impacted streams adds extra PPSs throughout
the stream, varying the initial QP and whether weighted prediction
was enabled. This causes issues after copying the stream into
a global header container, as the starting PPS is not suitable
for the rest of the stream. One side effect, for example,
is seeking will return garbled output until a new PPS appears.
This BSF removes the extra PPSs and rewrites the slice headers
such that the stream uses a single leading PPS in the global header,
which resolves the issue.
A new single global PPS is created, and all of the redundant PPSs
within the stream are removed.
@section hevc_metadata
@@ -564,10 +372,6 @@ will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
@item width
@item height
Set width and height after crop.
@item level
Set the level in the VPS and SPS. See H.265 section A.4 and tables
A.6 and A.7.
@@ -662,6 +466,10 @@ metadata header from each subtitle packet.
See also the @ref{text2movsub} filter.
@section mp3decomp
Decompress non-standard compressed MP3 audio headers.
@section mpeg2_metadata
Modify metadata embedded in an MPEG-2 stream.
@@ -781,12 +589,12 @@ ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv
Drop every video packet not marked as a keyframe after timestamp 30s but do not
modify any of the remaining packets.
@example
ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(pts*tb\,30)*not(key)' output.mkv
ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
@end example
Drop one second of audio every 10 seconds and add some random noise to the rest.
@example
ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(pts*tb\,10)\,9\,10)' output.mkv
ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
@end example
@section null
@@ -822,14 +630,6 @@ for NTSC frame rate using the @option{frame_rate} option.
ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example
@section pgs_frame_merge
Merge a sequence of PGS Subtitle segments ending with an "end of display set"
segment into a single packet.
This is required by some containers that support PGS subtitles
(muxer @code{matroska}).
@section prores_metadata
Modify color property metadata embedded in prores stream.
@@ -936,12 +736,6 @@ It accepts the following parameters:
@item pts
@item dts
Set expressions for PTS, DTS or both.
@item duration
Set expression for duration.
@item time_base
Set output time base.
@item prescale
Set whether to convert time fields to user-set output time base before evaluation of expressions. Defaults to 0.
@end table
The expressions are evaluated through the eval API and can contain the following
@@ -965,9 +759,6 @@ The demux timestamp in input.
@item PTS
The presentation timestamp in input.
@item DURATION
The duration in input.
@item STARTDTS
The DTS of the first packet.
@@ -980,32 +771,14 @@ The previous input DTS.
@item PREV_INPTS
The previous input PTS.
@item PREV_INDURATION
The previous input duration.
@item PREV_OUTDTS
The previous output DTS.
@item PREV_OUTPTS
The previous output PTS.
@item PREV_OUTDURATION
The previous output duration.
@item NEXT_DTS
The next input DTS.
@item NEXT_PTS
The next input PTS.
@item NEXT_DURATION
The next input duration.
@item TB
The timebase in which time fields are denominated. The user-set output timebase if prescale is enabled, else the input timebase.
@item TB_OUT
The output timebase.
The timebase of stream packet belongs.
@item SR
The sample rate of stream packet belongs.
@@ -1014,20 +787,6 @@ The sample rate of stream packet belongs.
The AV_NOPTS_VALUE constant.
@end table
For example, to set PTS equal to DTS (not recommended if B-frames are involved):
@example
ffmpeg -i INPUT -c:a copy -bsf:a setts=pts=DTS out.mkv
@end example
@section showinfo
Log basic packet information. Mainly useful for testing, debugging,
and development.
@section smpte436m_to_eia608
Convert from a @code{SMPTE_436M_ANC} data stream to a @code{EIA_608} stream,
extracting the closed captions from CTA-708 CDP VANC packets, and ignoring all other data.
@anchor{text2movsub}
@section text2movsub
+1 -1
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File diff suppressed because one or more lines are too long
+6 -7
View File
@@ -7,6 +7,11 @@ V
Disable the default terse mode, the full command issued by make and its
output will be shown on the screen.
DBG
Preprocess x86 external assembler files to a .dbg.asm file in the object
directory, which then gets compiled. Helps in developing those assembler
files.
DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
@@ -25,13 +30,6 @@ fate
fate-list
List all fate/regression test targets.
fate-list-failing
List the fate tests that failed the last time they were executed.
fate-clear-reports
Remove the test reports from previous test executions (getting rid of
potentially stale results from fate-list-failing).
install
Install headers, libraries and programs.
@@ -65,3 +63,4 @@ make -j<num>
make -k
Continue build in case of errors, this is useful for the regression tests
sometimes but note that it will still not run all reg tests.
+11 -23
View File
@@ -3,7 +3,7 @@
@c man begin CODEC OPTIONS
libavcodec provides some generic global options, which can be set on
all the encoders and decoders. In addition, each codec may support
all the encoders and decoders. In addition each codec may support
so-called private options, which are specific for a given codec.
Sometimes, a global option may only affect a specific kind of codec,
@@ -644,10 +644,6 @@ for codecs that support it. See also @file{doc/examples/export_mvs.c}.
Do not skip samples and export skip information as frame side data.
@item ass_ro_flush_noop
Do not reset ASS ReadOrder field on flush.
@item icc_profiles
Generate/parse embedded ICC profiles from/to colorimetry tags.
@item fixed_frame_size
Force audio encoders to use a fixed frame size.
@end table
@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
@@ -666,8 +662,6 @@ for codecs that support it. At present, those are H.264 and VP9.
@item film_grain
Export film grain parameters through frame side data (see @code{AV_FRAME_DATA_FILM_GRAIN_PARAMS}).
Supported at present by AV1 decoders.
@item enhancements
Export picture enhancement metadata through frame side data, e.g. LCEVC (see @code{AV_FRAME_DATA_LCEVC}).
@end table
@item threads @var{integer} (@emph{decoding/encoding,video})
@@ -701,13 +695,10 @@ profiles are documented in the relevant encoder documentation.
@item level @var{integer} (@emph{encoding,audio,video})
Set the encoder level. This level depends on the specific codec, and
might correspond to the profile level. It is set by default to
@samp{unknown}.
Possible values:
@table @samp
@item unknown
@end table
@item lowres @var{integer} (@emph{decoding,audio,video})
@@ -771,11 +762,18 @@ Set rate-distortion optimal quantization.
@item compression_level @var{integer} (@emph{encoding,audio,video})
@item bits_per_raw_sample @var{integer}
@item channel_layout @var{integer} (@emph{decoding/encoding,audio})
See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the required syntax.
Possible values:
@table @samp
@end table
@item request_channel_layout @var{integer} (@emph{decoding,audio})
Possible values:
@table @samp
@end table
@item rc_max_vbv_use @var{float} (@emph{encoding,video})
@item rc_min_vbv_use @var{float} (@emph{encoding,video})
@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video})
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
Possible values:
@@ -890,11 +888,9 @@ Possible values:
@table @samp
@item tv
@item mpeg
@item limited
MPEG (219*2^(n-8))
@item pc
@item jpeg
@item full
JPEG (2^n-1)
@end table
@@ -915,14 +911,6 @@ Possible values:
@end table
@item alpha_mode @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item premultiplied
@item straight
@end table
@item log_level_offset @var{integer}
Set the log level offset.
-182
View File
@@ -1,182 +0,0 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Community
@titlepage
@center @titlefont{Community}
@end titlepage
@top
@contents
@anchor{Organisation}
@chapter Organisation
The FFmpeg project is organized through a community working on global consensus.
Decisions are taken by the ensemble of active members, through voting and are aided by two committees.
@anchor{General Assembly}
@chapter General Assembly
The ensemble of active members is called the General Assembly (GA).
The General Assembly is sovereign and legitimate for all its decisions regarding the FFmpeg project.
The General Assembly is made up of active contributors.
Contributors are considered "active contributors" if they have authored more than 20 patches in the last 36 months in the main FFmpeg repository, or if they have been voted in by the GA.
The list of active contributors is updated twice each year, on 1st January and 1st July, 0:00 UTC.
Additional members are added to the General Assembly through a vote after proposal by a member of the General Assembly. They are part of the GA for two years, after which they need a confirmation by the GA.
A script to generate the current members of the general assembly (minus members voted in) can be found in `tools/general_assembly.pl`.
@anchor{Voting}
@chapter Voting
Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
Majority vote means more than 50% of the expressed ballots.
@anchor{Technical Committee}
@chapter Technical Committee
The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project. They will consider the merits of all the positions, judge them and make a decision.
The TC resolves technical conflicts but is not a technical steering committee.
Decisions by the TC are binding for all the contributors.
Decisions made by the TC can be re-opened after 1 year or by a majority vote of the General Assembly, requested by one of the member of the GA.
The TC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the TC.
The members of the TC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
The conflict resolution process is detailed in the resolution process document.
The TC can be contacted at <tc@@ffmpeg>.
@anchor{Resolution Process}
@section Resolution Process
The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project.
The TC main role is to resolve technical conflicts. It is therefore not a technical steering committee, but it is understood that some decisions might impact the future of the project.
@subsection Seizing
The TC can take possession of any technical matter that it sees fit.
To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
As members of TC are developers, they also can email tc@ to raise an issue.
@subsection Announcement
The TC, once seized, must announce itself on the main mailing list, with a [TC] tag.
The TC has 2 modes of operation: a RFC one and an internal one.
If the TC thinks it needs the input from the larger community, the TC can call for a RFC. Else, it can decide by itself.
The decision to use a RFC process or an internal discussion is a discretionary decision of the TC.
The TC can also reject a seizure for a few reasons such as: the matter was not discussed enough previously; it lacks expertise to reach a beneficial decision on the matter; or the matter is too trivial.
@subsection RFC call
In the RFC mode, one person from the TC posts on the mailing list the technical question and will request input from the community.
The mail will have the following specification:
a precise title
a specific tag [TC RFC]
a top-level email
contain a precise question that does not exceed 100 words and that is answerable by developers
may have an extra description, or a link to a previous discussion, if deemed necessary,
contain a precise end date for the answers.
The answers from the community must be on the main mailing list and must have the following specification:
keep the tag and the title unchanged
limited to 400 words
a first-level, answering directly to the main email
answering to the question.
Further replies to answers are permitted, as long as they conform to the community standards of politeness, they are limited to 100 words, and are not nested more than once. (max-depth=2)
After the end-date, mails on the thread will be ignored.
Violations of those rules will be escalated through the Community Committee.
After all the emails are in, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay, that will be notified on the mailing list.
@subsection Within TC
In the internal case, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay.
@subsection Decisions
The decisions from the TC will be sent on the mailing list, with the [TC] tag.
Internally, the TC should take decisions with a majority, or using ranked-choice voting.
Each TC member must vote on such decision according to what is, in their view, best for the project.
If a TC member feels they are affected by a conflict of interest with regards to the case, they should announce it and recuse themselves from the TC
discussion and vote.
A conflict of interest is presumed to occur when a TC member has a personal interest (e.g. financial) in a specific outcome of the case.
The decision from the TC should be published with a summary of the reasons that lead to this decision.
The decisions from the TC are final, until the matters are reopened after no less than one year.
@anchor{Community Committee}
@chapter Community Committee
The Community Committee (CC) is here to arbitrage and make decisions when inter-personal conflicts occur in the project. It will decide quickly and take actions, for the sake of the project.
The CC can remove privileges of offending members, including removal of commit access and temporary ban from the community.
Decisions made by the CC can be re-opened after 1 year or by a majority vote of the General Assembly. Indefinite bans from the community must be confirmed by the General Assembly, in a majority vote.
The CC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the CC.
The members of the CC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
The CC is governed by and responsible for enforcing the Code of Conduct.
The CC can be contacted at <cc@@ffmpeg>.
@anchor{Code of Conduct}
@chapter Code of Conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it's rarely good to start with that as initial assumption.
Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
Try to help other team members and cooperate if you can.
The goal of software development is to create technical excellence, not for any
individual to be better and "win" against the others. Large software projects
are only possible and successful through teamwork.
If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@bye
+19 -303
View File
@@ -38,51 +38,6 @@ Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.
@end table
@section hevc
HEVC (AKA ITU-T H.265 or ISO/IEC 23008-2) decoder.
The decoder supports MV-HEVC multiview streams with at most two views. Views to
be output are selected by supplying a list of view IDs to the decoder (the
@option{view_ids} option). This option may be set either statically before
decoder init, or from the @code{get_format()} callback - useful for the case
when the view count or IDs change dynamically during decoding.
Only the base layer is decoded by default.
Note that if you are using the @code{ffmpeg} CLI tool, you should be using view
specifiers as documented in its manual, rather than the options documented here.
@subsection Options
@table @option
@item view_ids (MV-HEVC)
Specify a list of view IDs that should be output. This option can also be set to
a single '-1', which will cause all views defined in the VPS to be decoded and
output.
@item view_ids_available (MV-HEVC)
This option may be read by the caller to retrieve an array of view IDs available
in the active VPS. The array is empty for single-layer video.
The value of this option is guaranteed to be accurate when read from the
@code{get_format()} callback. It may also be set at other times (e.g. after
opening the decoder), but the value is informational only and may be incorrect
(e.g. when the stream contains multiple distinct VPS NALUs).
@item view_pos_available (MV-HEVC)
This option may be read by the caller to retrieve an array of view positions
(left, right, or unspecified) available in the active VPS, as
@code{AVStereo3DView} values. When the array is available, its elements apply to
the corresponding elements of @option{view_ids_available}, i.e.
@code{view_pos_available[i]} contains the position of view with ID
@code{view_ids_available[i]}.
Same validity restrictions as for @option{view_ids_available} apply to
this option.
@end table
@section rawvideo
Raw video decoder.
@@ -119,9 +74,15 @@ The following options are supported by the libdav1d wrapper.
@table @option
@item max_frame_delay
Set max amount of frames the decoder may buffer internally. The default value is 0
(autodetect).
@item framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
global option @code{threads} instead.
@item tilethreads
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
global option @code{threads} instead.
@item filmgrain
Apply film grain to the decoded video if present in the bitstream. Defaults to the
@@ -165,87 +126,6 @@ Set amount of frame threads to use during decoding. The default value is 0 (auto
@end table
@section libxevd
eXtra-fast Essential Video Decoder (XEVD) MPEG-5 EVC decoder wrapper.
This decoder requires the presence of the libxevd headers and library
during configuration. You need to explicitly configure the build with
@option{--enable-libxevd}.
The xevd project website is at @url{https://github.com/mpeg5/xevd}.
@subsection Options
The following options are supported by the libxevd wrapper.
The xevd-equivalent options or values are listed in parentheses for easy migration.
To get a more accurate and extensive documentation of the libxevd options,
invoke the command @code{xevd_app --help} or consult the libxevd documentation.
@table @option
@item threads (@emph{threads})
Force to use a specific number of threads
@end table
@section QSV Decoders
The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC,
JPEG/MJPEG, VP8, VP9, AV1, VVC).
@subsection Common Options
The following options are supported by all qsv decoders.
@table @option
@item @var{async_depth}
Internal parallelization depth, the higher the value the higher the latency.
@item @var{gpu_copy}
A GPU-accelerated copy between video and system memory
@table @samp
@item default
@item on
@item off
@end table
@end table
@subsection HEVC Options
Extra options for hevc_qsv.
@table @option
@item @var{load_plugin}
A user plugin to load in an internal session
@table @samp
@item none
@item hevc_sw
@item hevc_hw
@end table
@item @var{load_plugins}
A :-separate list of hexadecimal plugin UIDs to load in an internal session
@end table
@section v210
Uncompressed 4:2:2 10-bit decoder.
@subsection Options
@table @option
@item custom_stride
Set the line size of the v210 data in bytes. The default value is 0
(autodetect). You can use the special -1 value for a strideless v210 as seen in
BOXX files.
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@@ -307,6 +187,14 @@ This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
documented.
@section libcelt
libcelt decoder wrapper.
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
Requires the presence of the libcelt headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libcelt}.
@section libgsm
libgsm decoder wrapper.
@@ -338,15 +226,6 @@ value is 0 (disabled).
@end table
@section libmpeghdec
libmpeghdec decoder wrapper.
libmpeghdec allows libmpeghdec to decode the MPEG-H 3D audio codec.
Requires the presence of the libmpeghdec headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libmpeghdec --enable-nonfree}.
@section libopencore-amrnb
libopencore-amrnb decoder wrapper.
@@ -386,7 +265,7 @@ without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTITLES DECODERS
@c man begin SUBTILES DECODERS
@section libaribb24
@@ -413,169 +292,6 @@ Enabled by default.
@end table
@section libaribcaption
Yet another ARIB STD-B24 caption decoder using external @dfn{libaribcaption}
library.
Implements profiles A and C of the Japanese ARIB STD-B24 standard,
Brazilian ABNT NBR 15606-1, and Philippines version of ISDB-T.
Requires the presence of the libaribcaption headers and library
(@url{https://github.com/xqq/libaribcaption}) during configuration.
You need to explicitly configure the build with @code{--enable-libaribcaption}.
If both @dfn{libaribb24} and @dfn{libaribcaption} are enabled, @dfn{libaribcaption}
decoder precedes.
@subsection libaribcaption Decoder Options
@table @option
@item -sub_type @var{subtitle_type}
Specifies the format of the decoded subtitles.
@table @samp
@item bitmap
Graphical image.
@item ass
ASS formatted text.
@item text
Simple text based output without formatting.
@end table
The default is @dfn{ass} as same as @dfn{libaribb24} decoder.
Some present players (e.g., @dfn{mpv}) expect ASS format for ARIB caption.
@item -caption_encoding @var{encoding_scheme}
Specifies the encoding scheme of input subtitle text.
@table @samp
@item auto
Automatically detect text encoding (default).
@item jis
8bit-char JIS encoding defined in ARIB STD B24.
This encoding used in Japan for ISDB captions.
@item utf8
UTF-8 encoding defined in ARIB STD B24.
This encoding is used in Philippines for ISDB-T captions.
@item latin
Latin character encoding defined in ABNT NBR 15606-1.
This encoding is used in South America for SBTVD / ISDB-Tb captions.
@end table
@item -font @var{font_name[,font_name2,...]}
Specify comma-separated list of font family names to be used for @dfn{bitmap}
or @dfn{ass} type subtitle rendering.
Only first font name is used for @dfn{ass} type subtitle.
If not specified, use internally defined default font family.
@item -ass_single_rect @var{boolean}
ARIB STD-B24 specifies that some captions may be displayed at different
positions at a time (multi-rectangle subtitle).
Since some players (e.g., old @dfn{mpv}) can't handle multiple ASS rectangles
in a single AVSubtitle, or multiple ASS rectangles of indeterminate duration
with the same start timestamp, this option can change the behavior so that
all the texts are displayed in a single ASS rectangle.
The default is @var{false}.
If your player cannot handle AVSubtitles with multiple ASS rectangles properly,
set this option to @var{true} or define @env{ASS_SINGLE_RECT=1} to change
default behavior at compilation.
@item -force_outline_text @var{boolean}
Specify whether always render outline text for all characters regardless of
the indication by character style.
The default is @var{false}.
@item -outline_width @var{number} (0.0 - 3.0)
Specify width for outline text, in dots (relative).
The default is @var{1.5}.
@item -ignore_background @var{boolean}
Specify whether to ignore background color rendering.
The default is @var{false}.
@item -ignore_ruby @var{boolean}
Specify whether to ignore rendering for ruby-like (furigana) characters.
The default is @var{false}.
@item -replace_drcs @var{boolean}
Specify whether to render replaced DRCS characters as Unicode characters.
The default is @var{true}.
@item -replace_msz_ascii @var{boolean}
Specify whether to replace MSZ (Middle Size; half width) fullwidth
alphanumerics with halfwidth alphanumerics.
The default is @var{true}.
@item -replace_msz_japanese @var{boolean}
Specify whether to replace some MSZ (Middle Size; half width) fullwidth
japanese special characters with halfwidth ones.
The default is @var{true}.
@item -replace_msz_glyph @var{boolean}
Specify whether to replace MSZ (Middle Size; half width) characters
with halfwidth glyphs if the fonts supports it.
This option works under FreeType or DirectWrite renderer
with Adobe-Japan1 compliant fonts.
e.g., IBM Plex Sans JP, Morisawa BIZ UDGothic, Morisawa BIZ UDMincho,
Yu Gothic, Yu Mincho, and Meiryo.
The default is @var{true}.
@item -canvas_size @var{image_size}
Specify the resolution of the canvas to render subtitles to; usually, this
should be frame size of input video.
This only applies when @code{-subtitle_type} is set to @var{bitmap}.
The libaribcaption decoder assumes input frame size for bitmap rendering as below:
@enumerate
@item
PROFILE_A : 1440 x 1080 with SAR (PAR) 4:3
@item
PROFILE_C : 320 x 180 with SAR (PAR) 1:1
@end enumerate
If actual frame size of input video does not match above assumption,
the rendered captions may be distorted.
To make the captions undistorted, add @code{-canvas_size} option to specify
actual input video size.
Note that the @code{-canvas_size} option is not required for video with
different size but same aspect ratio.
In such cases, the caption will be stretched or shrunk to actual video size
if @code{-canvas_size} option is not specified.
If @code{-canvas_size} option is specified with different size,
the caption will be stretched or shrunk as specified size with calculated SAR.
@end table
@subsection libaribcaption decoder usage examples
Display MPEG-TS file with ARIB subtitle by @code{ffplay} tool:
@example
ffplay -sub_type bitmap MPEG.TS
@end example
Display MPEG-TS file with input frame size 1920x1080 by @code{ffplay} tool:
@example
ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS
@end example
Embed ARIB subtitle in transcoded video:
@example
ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h264 dest.mp4
@end example
@section dvbsub
@subsection Options
@@ -687,4 +403,4 @@ box and an end box, typically subtitles. Default value is 0 if
@end table
@c man end SUBTITLES DECODERS
@c man end SUBTILES DECODERS
+26 -366
View File
@@ -274,211 +274,11 @@ which streams to actually receive.
Each stream mirrors the @code{id} and @code{bandwidth} properties from the
@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
@subsection Options
This demuxer accepts the following option:
@table @option
@item cenc_decryption_key
Default 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item cenc_decryption_keys
Dictionary of 16-byte key ID => 16-byte key, both in hex, to decrypt files encrypted using ISO Common Encryption
(CENC/AES-128 CTR; ISO/IEC 23001-7).
@end table
@section dvdvideo
DVD-Video demuxer, powered by libdvdnav and libdvdread.
Can directly ingest DVD titles, specifically sequential PGCs, into
a conversion pipeline. Menu assets, such as background video or audio,
can also be demuxed given the menu's coordinates (at best effort).
Block devices (DVD drives), ISO files, and directory structures are accepted.
Activate with @code{-f dvdvideo} in front of one of these inputs.
This demuxer does NOT have decryption code of any kind. You are on your own
working with encrypted DVDs, and should not expect support on the matter.
Underlying playback is handled by libdvdnav, and structure parsing by libdvdread.
FFmpeg must be built with GPL library support available as well as the
configure switches @code{--enable-libdvdnav} and @code{--enable-libdvdread}.
You will need to provide either the desired "title number" or exact PGC/PG coordinates.
Many open-source DVD players and tools can aid in providing this information.
If not specified, the demuxer will default to title 1 which works for many discs.
However, due to the flexibility of the format, it is recommended to check manually.
There are many discs that are authored strangely or with invalid headers.
If the input is a real DVD drive, please note that there are some drives which may
silently fail on reading bad sectors from the disc, returning random bits instead
which is effectively corrupt data. This is especially prominent on aging or rotting discs.
A second pass and integrity checks would be needed to detect the corruption.
This is not an FFmpeg issue.
@subsection Background
DVD-Video is not a directly accessible, linear container format in the
traditional sense. Instead, it allows for complex and programmatic playback of
carefully muxed MPEG-PS streams that are stored in headerless VOB files.
To the end-user, these streams are known simply as "titles", but the actual
logical playback sequence is defined by one or more "PGCs", or Program Group Chains,
within the title. The PGC is in turn comprised of multiple "PGs", or Programs",
which are the actual video segments (and for a typical video feature, sequentially
ordered). The PGC structure, along with stream layout and metadata, are stored in
IFO files that need to be parsed. PGCs can be thought of as playlists in easier terms.
An actual DVD player relies on user GUI interaction via menus and an internal VM
to drive the direction of demuxing. Generally, the user would either navigate (via menus)
or automatically be redirected to the PGC of their choice. During this process and
the subsequent playback, the DVD player's internal VM also maintains a state and
executes instructions that can create jumps to different sectors during playback.
This is why libdvdnav is involved, as a linear read of the MPEG-PS blobs on the
disc (VOBs) is not enough to produce the right sequence in many cases.
There are many other DVD structures (a long subject) that will not be discussed here.
NAV packets, in particular, are handled by this demuxer to build accurate timing
but not emitted as a stream. For a good high-level understanding, refer to:
@url{https://code.videolan.org/videolan/libdvdnav/-/blob/master/doc/dvd_structures}
@subsection Options
This demuxer accepts the following options:
@table @option
@item title @var{int}
The title number to play. Must be set if @option{pgc} and @option{pg} are not set.
Not applicable to menus.
Default is 0 (auto), which currently only selects the first available title (title 1)
and notifies the user about the implications.
@item chapter_start @var{int}
The chapter, or PTT (part-of-title), number to start at. Not applicable to menus.
Default is 1.
@item chapter_end @var{int}
The chapter, or PTT (part-of-title), number to end at. Not applicable to menus.
Default is 0, which is a special value to signal end at the last possible chapter.
@item angle @var{int}
The video angle number, referring to what is essentially an additional
video stream that is composed from alternate frames interleaved in the VOBs.
Not applicable to menus.
Default is 1.
@item region @var{int}
The region code to use for playback. Some discs may use this to default playback
at a particular angle in different regions. This option will not affect the region code
of a real DVD drive, if used as an input. Not applicable to menus.
Default is 0, "world".
@item menu @var{bool}
Demux menu assets instead of navigating a title. Requires exact coordinates
of the menu (@option{menu_lu}, @option{menu_vts}, @option{pgc}, @option{pg}).
Default is false.
@item menu_lu @var{int}
The menu language to demux. In DVD, menus are grouped by language.
Default is 1, the first language unit.
@item menu_vts @var{int}
The VTS where the menu lives, or 0 if it is a VMG menu (root-level).
Default is 1, menu of the first VTS.
@item pgc @var{int}
The entry PGC to start playback, in conjunction with @option{pg}.
Alternative to setting @option{title}.
Chapter markers are not supported at this time.
Must be explicitly set for menus.
Default is 0, automatically resolve from value of @option{title}.
@item pg @var{int}
The entry PG to start playback, in conjunction with @option{pgc}.
Alternative to setting @option{title}.
Chapter markers are not supported at this time.
Default is 1, the first PG of the PGC.
@item preindex @var{bool}
Enable this to have accurate chapter (PTT) markers and duration measurement,
which requires a slow second pass read in order to index the chapter marker
timestamps from NAV packets. This is non-ideal extra work for real optical drives.
It is recommended and faster to use this option with a backup of the DVD structure
stored on a hard drive. Not compatible with @option{pgc} and @option{pg}.
Default is 0, false.
@item trim @var{bool}
Skip padding cells (i.e. cells shorter than 1 second) from the beginning.
There exist many discs with filler segments at the beginning of the PGC,
often with junk data intended for controlling a real DVD player's
buffering speed and with no other material data value.
Not applicable to menus.
Default is 1, true.
@end table
@subsection Examples
@itemize
@item
Open title 3 from a given DVD structure:
@example
ffmpeg -f dvdvideo -title 3 -i <path to DVD> ...
@end example
@item
Open chapters 3-6 from title 1 from a given DVD structure:
@example
ffmpeg -f dvdvideo -chapter_start 3 -chapter_end 6 -title 1 -i <path to DVD> ...
@end example
@item
Open only chapter 5 from title 1 from a given DVD structure:
@example
ffmpeg -f dvdvideo -chapter_start 5 -chapter_end 5 -title 1 -i <path to DVD> ...
@end example
@item
Demux menu with language 1 from VTS 1, PGC 1, starting at PG 1:
@example
ffmpeg -f dvdvideo -menu 1 -menu_lu 1 -menu_vts 1 -pgc 1 -pg 1 -i <path to DVD> ...
@end example
@end itemize
@section ea
Electronic Arts Multimedia format demuxer.
This format is used by various Electronic Arts games.
@subsection Options
@table @option
@item merge_alpha @var{bool}
Normally the VP6 alpha channel (if exists) is returned as a secondary video
stream, by setting this option you can make the demuxer return a single video
stream which contains the alpha channel in addition to the ordinary video.
@end table
@section imf
Interoperable Master Format demuxer.
This demuxer presents audio and video streams found in an IMF Composition, as
specified in @url{https://doi.org/10.5594/SMPTE.ST2067-2.2020, SMPTE ST 2067-2}.
@example
ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...
@end example
If @code{-assetmaps} is not specified, the demuxer looks for a file called
@file{ASSETMAP.xml} in the same directory as the CPL.
This demuxer presents audio and video streams found in an IMF Composition.
@section flv, live_flv, kux
@@ -562,19 +362,9 @@ It accepts the following options:
@item live_start_index
segment index to start live streams at (negative values are from the end).
@item prefer_x_start
prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
@item allowed_extensions
',' separated list of file extensions that hls is allowed to access.
@item extension_picky
This blocks disallowed extensions from probing
It also requires all available segments to have matching extensions to the format
except mpegts, which is always allowed.
It is recommended to set the whitelists correctly instead of depending on extensions
Enabled by default.
@item max_reload
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@@ -597,10 +387,6 @@ Use HTTP partial requests for downloading HTTP segments.
@item seg_format_options
Set options for the demuxer of media segments using a list of key=value pairs separated by @code{:}.
@item seg_max_retry
Maximum number of times to reload a segment on error, useful when segment skip on network error is not desired.
Default value is 0.
@end table
@section image2
@@ -668,9 +454,30 @@ Select a glob wildcard pattern type.
The pattern is interpreted like a @code{glob()} pattern. This is only
selectable if libavformat was compiled with globbing support.
@item glob_sequence @emph{(deprecated, will be removed)}
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and
the provided pattern contains at least one glob meta character among
@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
interpreted like a @code{glob()} pattern, otherwise it is interpreted
like a sequence pattern.
All glob special characters @code{%*?[]@{@}} must be prefixed
with "%". To escape a literal "%" you shall use "%%".
For example the pattern @code{foo-%*.jpeg} will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating
with ".jpeg".
This pattern type is deprecated in favor of @var{glob} and
@var{sequence}.
@end table
Default value is @var{sequence}.
Default value is @var{glob_sequence}.
@item pixel_format
Set the pixel format of the images to read. If not specified the pixel
format is guessed from the first image file in the sequence.
@@ -838,32 +645,6 @@ Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@anchor{mccdec}
@section mcc
Demuxer for MacCaption MCC files, it supports MCC versions 1.0 and 2.0.
MCC files store VANC data, which can include closed captions (EIA-608 and CEA-708), ancillary time code, pan-scan data, etc.
By default, for backward compatibility, the MCC demuxer extracts just the EIA-608 and CEA-708 closed captions and returns a @code{EIA_608} stream, ignoring all other VANC data.
You can change it to return all VANC data in a @code{SMPTE_436M_ANC} data stream by setting @option{-eia608_extract 0}
@subsection Examples
@itemize
@item
Convert a MCC file to Scenarist (SCC) format:
@example
ffmpeg -i CC.mcc -c:s copy CC.scc
@end example
Note that the SCC format only supports EIA-608, so this will discard all other data such as CEA-708 extensions.
@item
Merge a MCC file into a MXF file:
@example
ffmpeg -i video_and_audio.mxf -eia608_extract 0 -i CC.mcc -c copy -map 0 -map 1 out.mxf
@end example
This retains all VANC data and inserts it into the output MXF file as a @code{SMPTE_436M_ANC} data stream.
@end itemize
@section mov/mp4/3gp
Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
@@ -944,11 +725,7 @@ Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so shoul
specify.
@item decryption_key
Default 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item decryption_keys
Dictionary of 16-byte key ID => 16-byte key, both in hex, to decrypt files encrypted using ISO Common Encryption
(CENC/AES-128 CTR; ISO/IEC 23001-7).
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item max_stts_delta
Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in
@@ -956,15 +733,8 @@ error or used to store a negative value for dts correction when treated as signe
the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when
cast to int32 are used to adjust onward dts.
Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows up to
Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows upto
a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
@item interleaved_read
Interleave packets from multiple tracks at demuxer level. For badly interleaved files, this prevents playback issues
caused by large gaps between packets in different tracks, as MOV/MP4 do not have packet placement requirements.
However, this can cause excessive seeking on very badly interleaved files, due to seeking between tracks, so disabling
it may prevent I/O issues, at the expense of playback.
@end table
@subsection Audible AAX
@@ -1003,12 +773,8 @@ to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
@item merge_pmt_versions
Reuse existing streams when a PMT's version is updated and elementary
Re-use existing streams when a PMT's version is updated and elementary
streams move to different PIDs. Default value is 0.
@item max_packet_size
Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size
are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.
@end table
@section mpjpeg
@@ -1043,11 +809,6 @@ Set input video frame rate. Default value is 25.
@item pixel_format
Set the input video pixel format. Default value is @code{yuv420p}.
@item stride
Set frame line size in bytes, only required if there are extra padding.
For multiplane formats, @option{stride} is a list of line size for each
plane.
@item video_size
Set the input video size. This value must be specified explicitly.
@end table
@@ -1060,43 +821,6 @@ the command:
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
@end example
Read a rawvideo with @command{ffplay}, assuming a pixel format of @code{yuv420p},
a video size of @code{1080x1920}, has 8 bytes of padding in each line of Y plane,
and 4 bytes of padding with UV plane,
@example
ffplay -f rawvideo -pixel_format yuv420p -video_size 1080x1920 -stride 1088,544,544 input.raw
@end example
@anchor{rcwtdec}
@section rcwt
RCWT (Raw Captions With Time) is a format native to ccextractor, a commonly
used open source tool for processing 608/708 Closed Captions (CC) sources.
For more information on the format, see @ref{rcwtenc,,,ffmpeg-formats}.
This demuxer implements the specification as of March 2024, which has
been stable and unchanged since April 2014.
@subsection Examples
@itemize
@item
Render CC to ASS using the built-in decoder:
@example
ffmpeg -i CC.rcwt.bin CC.ass
@end example
Note that if your output appears to be empty, you may have to manually
set the decoder's @option{data_field} option to pick the desired CC substream.
@item
Convert an RCWT backup to Scenarist (SCC) format:
@example
ffmpeg -i CC.rcwt.bin -c:s copy CC.scc
@end example
Note that the SCC format does not support all of the possible CC extensions
that can be stored in RCWT (such as EIA-708).
@end itemize
@section sbg
SBaGen script demuxer.
@@ -1164,68 +888,4 @@ which in turn, acts as a ceiling for the size of scripts that can be read.
Default is 1 MiB.
@end table
@section w64
Sony Wave64 Audio demuxer.
This demuxer accepts the following options:
@table @option
@item max_size
See the same option for the @ref{wav} demuxer.
@end table
@anchor{wav}
@section wav
RIFF Wave Audio demuxer.
This demuxer accepts the following options:
@table @option
@item ignore_length @var{bool}
Ignore the size of the @code{data} chunk and keep reading until the end of the file if set.
May be useful to read broken or partial files where the header was not properly updated,
but will misinterpret files with non-audio chunks after the @code{data} chunk.
Default is disabled.
@item max_size
Specify the maximum packet size in bytes for the demuxed packets. By default
this is set to 0, which means that a sensible value is chosen based on the
input format.
@end table
@section webp
Animated WebP demuxer.
It accepts the following options:
@table @option
@item -min_delay @var{int}
Set the minimum valid delay between frames in milliseconds.
Range is 0 to 60000. Default value is 10.
@item -max_webp_delay @var{int}
Set the maximum valid delay between frames in milliseconds.
Range is 0 to 16777215. Default value is 16777215 (over four hours),
the maximum value allowed by the specification.
@item -default_delay @var{int}
Set the default delay between frames in milliseconds.
Range is 0 to 60000. Default value is 100.
@item -ignore_loop @var{bool}
WebP files can contain information to loop a certain number of times
(or infinitely). If @option{ignore_loop} is set to true, then the loop
setting from the input will be ignored and looping will not occur.
If set to false, then looping will occur and will cycle the number
of times according to the WebP. Default value is true.
@item usebgcolor @var{bool}
WebP files contain a background color hint in the ANIM chunk, but the
WebP specification says that viewer applications are not required to
use it. If @option{usebgcolor} is set to true, then the background
color hint will be used, otherwise transparent black will be used for
the background. Default value is false.
@end table
@c man end DEMUXERS
+79
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@@ -0,0 +1,79 @@
# FFmpeg project
## Organisation
The FFmpeg project is organized through a community working on global consensus.
Decisions are taken by the ensemble of active members, through voting and
are aided by two committees.
## General Assembly
The ensemble of active members is called the General Assembly (GA).
The General Assembly is sovereign and legitimate for all its decisions
regarding the FFmpeg project.
The General Assembly is made up of active contributors.
Contributors are considered "active contributors" if they have pushed more
than 20 patches in the last 36 months in the main FFmpeg repository, or
if they have been voted in by the GA.
Additional members are added to the General Assembly through a vote after
proposal by a member of the General Assembly.
They are part of the GA for two years, after which they need a confirmation by
the GA.
## Voting
Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
Majority vote means more than 50% of the expressed ballots.
## Technical Committee
The Technical Committee (TC) is here to arbitrate and make decisions when
technical conflicts occur in the project.
They will consider the merits of all the positions, judge them and make a
decision.
The TC resolves technical conflicts but is not a technical steering committee.
Decisions by the TC are binding for all the contributors.
Decisions made by the TC can be re-opened after 1 year or by a majority vote
of the General Assembly, requested by one of the member of the GA.
The TC is elected by the General Assembly for a duration of 1 year, and
is composed of 5 members.
Members can be re-elected if they wish. A majority vote in the General Assembly
can trigger a new election of the TC.
The members of the TC can be elected from outside of the GA.
Candidates for election can either be suggested or self-nominated.
The conflict resolution process is detailed in the [resolution process](resolution_process.md) document.
## Community committee
The Community Committee (CC) is here to arbitrage and make decisions when
inter-personal conflicts occur in the project. It will decide quickly and
take actions, for the sake of the project.
The CC can remove privileges of offending members, including removal of
commit access and temporary ban from the community.
Decisions made by the CC can be re-opened after 1 year or by a majority vote
of the General Assembly. Indefinite bans from the community must be confirmed
by the General Assembly, in a majority vote.
The CC is elected by the General Assembly for a duration of 1 year, and is
composed of 5 members.
Members can be re-elected if they wish. A majority vote in the General Assembly
can trigger a new election of the CC.
The members of the CC can be elected from outside of the GA.
Candidates for election can either be suggested or self-nominated.
The CC is governed by and responsible for enforcing the Code of Conduct.
+91
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@@ -0,0 +1,91 @@
# Technical Committee
_This document only makes sense with the rules from [the community document](community)_.
The Technical Committee (**TC**) is here to arbitrate and make decisions when
technical conflicts occur in the project.
The TC main role is to resolve technical conflicts.
It is therefore not a technical steering committee, but it is understood that
some decisions might impact the future of the project.
# Process
## Seizing
The TC can take possession of any technical matter that it sees fit.
To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
As members of TC are developers, they also can email tc@ to raise an issue.
## Announcement
The TC, once seized, must announce itself on the main mailing list, with a _[TC]_ tag.
The TC has 2 modes of operation: a RFC one and an internal one.
If the TC thinks it needs the input from the larger community, the TC can call
for a RFC. Else, it can decide by itself.
If the disagreement involves a member of the TC, that member should recuse
themselves from the decision.
The decision to use a RFC process or an internal discussion is a discretionary
decision of the TC.
The TC can also reject a seizure for a few reasons such as:
the matter was not discussed enough previously; it lacks expertise to reach a
beneficial decision on the matter; or the matter is too trivial.
### RFC call
In the RFC mode, one person from the TC posts on the mailing list the
technical question and will request input from the community.
The mail will have the following specification:
* a precise title
* a specific tag [TC RFC]
* a top-level email
* contain a precise question that does not exceed 100 words and that is answerable by developers
* may have an extra description, or a link to a previous discussion, if deemed necessary,
* contain a precise end date for the answers.
The answers from the community must be on the main mailing list and must have
the following specification:
* keep the tag and the title unchanged
* limited to 400 words
* a first-level, answering directly to the main email
* answering to the question.
Further replies to answers are permitted, as long as they conform to the
community standards of politeness, they are limited to 100 words, and are not
nested more than once. (max-depth=2)
After the end-date, mails on the thread will be ignored.
Violations of those rules will be escalated through the Community Committee.
After all the emails are in, the TC has 96 hours to give its final decision.
Exceptionally, the TC can request an extra delay, that will be notified on the
mailing list.
### Within TC
In the internal case, the TC has 96 hours to give its final decision.
Exceptionally, the TC can request an extra delay.
## Decisions
The decisions from the TC will be sent on the mailing list, with the _[TC]_ tag.
Internally, the TC should take decisions with a majority, or using
ranked-choice voting.
The decision from the TC should be published with a summary of the reasons that
lead to this decision.
The decisions from the TC are final, until the matters are reopened after
no less than one year.
+221 -518
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@@ -10,110 +10,44 @@
@contents
@chapter Introduction
@chapter Notes for external developers
This text is concerned with the development @emph{of} FFmpeg itself. Information
on using the FFmpeg libraries in other programs can be found elsewhere, e.g. in:
@itemize @bullet
@item
the installed header files
@item
@url{http://ffmpeg.org/doxygen/trunk/index.html, the Doxygen documentation}
generated from the headers
@item
the examples under @file{doc/examples}
@end itemize
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in @file{doc/examples} and in the source code to
see how the public API is employed.
You can use the FFmpeg libraries in your commercial program, but you
are encouraged to @emph{publish any patch you make}. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
If you modify FFmpeg code for your own use case, you are highly encouraged to
@emph{submit your changes back to us}, using this document as a guide. There are
both pragmatic and ideological reasons to do so:
@chapter Contributing
There are 2 ways by which code gets into FFmpeg:
@itemize @bullet
@item
Maintaining external changes to keep up with upstream development is
time-consuming and error-prone. With your code in the main tree, it will be
maintained by FFmpeg developers.
@item
FFmpeg developers include leading experts in the field who can find bugs or
design flaws in your code.
@item
By supporting the project you find useful you ensure it continues to be
maintained and developed.
@item Submitting patches to the ffmpeg-devel mailing list.
See @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@end itemize
All proposed code changes should be submitted for review to
@url{https://code.ffmpeg.org/FFmpeg/FFmpeg/pulls, Forgejo} or
@url{mailto:ffmpeg-devel@@ffmpeg.org, the development mailing list}, as
described in more detail in the @ref{Submitting patches} chapter. The code
should comply with the @ref{Development Policy} and follow the @ref{Coding Rules}.
Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the @ref{Coding Rules}.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@chapter Coding Rules
@section Language
FFmpeg is mainly programmed in the ISO C11 language, except for the public
headers which must stay C99 compatible.
Compiler-specific extensions may be used with good reason, but must not be
depended on, i.e. the code must still compile and work with compilers lacking
the extension.
The following C99 features must not be used anywhere in the codebase:
@itemize @bullet
@item
variable-length arrays;
@item
complex numbers;
@end itemize
@subsection SIMD/DSP
@anchor{SIMD/DSP}
As modern compilers are unable to generate efficient SIMD or other
performance-critical DSP code from plain C, handwritten assembly is used.
Usually such code is isolated in a separate function. Then the standard approach
is writing multiple versions of this function a plain C one that works
everywhere and may also be useful for debugging, and potentially multiple
architecture-specific optimized implementations. Initialization code then
chooses the best available version at runtime and loads it into a function
pointer; the function in question is then always called through this pointer.
The specific syntax used for writing assembly is:
@itemize @bullet
@item
NASM on x86;
@item
GAS on ARM and RISC-V.
@end itemize
A unit testing framework for assembly called @code{checkasm} lives under
@file{tests/checkasm}. All new assembly should come with @code{checkasm} tests;
adding tests for existing assembly that lacks them is also strongly encouraged.
@subsection Other languages
Other languages than C may be used in special cases:
@itemize @bullet
@item
Compiler intrinsics or inline assembly when the code in question cannot be
written in the standard way described in the @ref{SIMD/DSP} section. This
typically applies to code that needs to be inlined.
@item
Objective-C where required for interacting with macOS-specific interfaces.
@end itemize
@section Code formatting conventions
There are the following guidelines regarding the code style in files:
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
@@ -133,156 +67,8 @@ K&R coding style is used.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
@subsection Examples
Some notable examples to illustrate common code style in FFmpeg:
@itemize @bullet
@item
Space around assignments and after
@code{if}/@code{do}/@code{while}/@code{for} keywords:
@example c, good
// Good
if (condition)
av_foo();
@end example
@example c, good
// Good
for (size_t i = 0; i < len; i++)
av_bar(i);
@end example
@example c, good
// Good
size_t size = 0;
@end example
However no spaces between the parentheses and condition, unless it helps
readability of complex conditions, so the following should not be done:
@example c, bad
// Bad style
if ( condition )
av_foo();
@end example
@item
No unnecessary parentheses, unless it helps readability:
@example c, good
// Good
int fields = ilace ? 2 : 1;
@end example
@item
Don't wrap single-line blocks in braces. Use braces only if there is an accompanying else statement. This keeps future code changes easier to keep track of.
@example c, good
// Good
if (bits_pixel == 24) @{
avctx->pix_fmt = AV_PIX_FMT_BGR24;
@} else if (bits_pixel == 8) @{
avctx->pix_fmt = AV_PIX_FMT_GRAY8;
@} else
return AVERROR_INVALIDDATA;
@end example
@item
Avoid assignments in conditions where it makes sense:
@example c, good
// Good
video_enc->chroma_intra_matrix = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64)
if (!video_enc->chroma_intra_matrix)
return AVERROR(ENOMEM);
@end example
@example c, bad
// Bad style
if (!(video_enc->chroma_intra_matrix = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64)))
return AVERROR(ENOMEM);
@end example
@example c, good
// Ok
while ((entry = av_dict_iterate(options, entry)))
av_log(ctx, AV_LOG_INFO, "Item '%s': '%s'\n", entry->key, entry->value);
@end example
@item
When declaring a pointer variable, the @code{*} goes with the variable not the type:
@example c, good
// Good
AVStream *stream;
@end example
@example c, bad
// Bad style
AVStream* stream;
@end example
@item
When sensible, prefer a narrow variable scope, especially in for loops:
@example c, good
// Good
for (unsigned i = 0; i < submix->nb_elements; i++) @{
// Do something...
@}
@end example
@example c, bad
// Bad style
unsigned i;
//...
for (i = 0; i < submix->nb_elements; i++) @{
// Do something...
@}
@end example
@end itemize
If you work on a file that does not follow these guidelines consistently,
change the parts that you are editing to follow these guidelines but do
not make unrelated changes in the file to make it conform to these.
@subsection Vim configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
" Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
@subsection Emacs configuration
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
@end lisp
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@section Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
@@ -324,52 +110,92 @@ int myfunc(int my_parameter)
...
@end example
@anchor{Naming conventions}
@section Naming conventions
@section C language features
Names of functions, variables, and struct members must be lowercase, using
underscores (_) to separate words. For example, @samp{avfilter_get_video_buffer}
is an acceptable function name and @samp{AVFilterGetVideo} is not.
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
Struct, union, enum, and typedeffed type names must use CamelCase. All structs
and unions should be typedeffed to the same name as the struct/union tag, e.g.
@code{typedef struct AVFoo @{ ... @} AVFoo;}. Enums are typically not
typedeffed.
Enumeration constants and macros must be UPPERCASE, except for macros
masquerading as functions, which should use the function naming convention.
All identifiers in the libraries should be namespaced as follows:
@itemize @bullet
@item
No namespacing for identifiers with file and lower scope (e.g. local variables,
static functions), and struct and union members,
the @samp{inline} keyword;
@item
The @code{ff_} prefix must be used for variables and functions visible outside
of file scope, but only used internally within a single library, e.g.
@samp{ff_w64_demuxer}. This prevents name collisions when FFmpeg is statically
linked.
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
@item
for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
@item
Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
@item
Implementation defined behavior for signed integers is assumed to match the
expected behavior for two's complement. Non representable values in integer
casts are binary truncated. Shift right of signed values uses sign extension.
@end itemize
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@section Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in CamelCase.
There are the following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For file-scope variables and functions declared as @code{static}, no prefix
is required.
@item
For variables and functions visible outside of file scope, but only used
internally by a library, an @code{ff_} prefix should be used,
e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_report_missing_feature}.
@item
All other internal identifiers, like private type or macro names, should be
namespaced only to avoid possible internal conflicts. E.g. @code{H264_NAL_SPS}
vs. @code{HEVC_NAL_SPS}.
@item
Each library has its own prefix for public symbols, in addition to the
commonly used @code{av_} (@code{avformat_} for libavformat,
@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
Check the existing code and choose names accordingly.
@item
Other public identifiers (struct, union, enum, macro, type names) must use their
library's public prefix (@code{AV}, @code{Sws}, or @code{Swr}).
Note that some symbols without these prefixes are also exported for
retro-compatibility reasons. These exceptions are declared in the
@code{lib<name>/lib<name>.v} files.
@end itemize
Furthermore, name space reserved for the system should not be invaded.
@@ -383,53 +209,50 @@ symbols. If in doubt, just avoid names starting with @code{_} altogether.
@section Miscellaneous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@item
Where applicable, SI units shall be used. For example timeouts should use seconds as the fundamental unit not microseconds.
That means a bare value like @samp{1.0} must mean 1 second, @samp{50m} means 50 milliseconds. For weight, gram shall be used.
@end itemize
@anchor{Development Policy}
@section Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
" Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
@end lisp
@chapter Development Policy
@section Code behaviour
@subheading Correctness
The code must be valid. It must not crash, abort, access invalid pointers, leak
memory, cause data races or signed integer overflow, or otherwise cause
undefined behaviour. Error codes should be checked and, when applicable,
forwarded to the caller.
@subheading Thread- and library-safety
Our libraries may be called by multiple independent callers in the same process.
These calls may happen from any number of threads and the different call sites
may not be aware of each other - e.g. a user program may be calling our
libraries directly, and use one or more libraries that also call our libraries.
The code must behave correctly under such conditions.
@subheading Robustness
The code must treat as untrusted any bytestream received from a caller or read
from a file, network, etc. It must not misbehave when arbitrary data is sent to
it - typically it should print an error message and return
@code{AVERROR_INVALIDDATA} on encountering invalid input data.
@subheading Memory allocation
The code must use the @code{av_malloc()} family of functions from
@file{libavutil/mem.h} to perform all memory allocation, except in special cases
(e.g. when interacting with an external library that requires a specific
allocator to be used).
All allocations should be checked and @code{AVERROR(ENOMEM)} returned on
failure. A common mistake is that error paths leak memory - make sure that does
not happen.
@subheading stdio
Our libraries must not access the stdio streams stdin/stdout/stderr directly
(e.g. via @code{printf()} family of functions), as that is not library-safe. For
logging, use @code{av_log()}.
@section Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
Contributions should be licensed under the
@@ -452,24 +275,13 @@ missing samples or an implementation with a small subset of features.
Always check the mailing list for any reviewers with issues and test
FATE before you push.
@subheading Commit messages
Commit messages are highly important tools for informing other developers on
what a given change does and why. Every commit must always have a properly
filled out commit message with the following format:
@example
area changed: short 1 line description
details describing what and why and giving references.
@end example
If the commit addresses a known bug on our bug tracker or other external issue
(e.g. CVE), the commit message should include the relevant bug ID(s) or other
external identifiers. Note that this should be done in addition to a proper
explanation and not instead of it. Comments such as "fixed!" or "Changed it."
are not acceptable.
When applying patches that have been discussed at length on the mailing list,
reference the thread in the commit message.
@subheading Keep the main commit message short with an extended description below.
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
@subheading Testing must be adequate but not excessive.
If it works for you, others, and passes FATE then it should be OK to commit
@@ -488,39 +300,50 @@ later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@subheading Bug fixes intended for backporting should stay focused.
Keep bug fixes as small as possible while fully fixing the reported issue.
Changes not needed for correctness or robustness should be submitted separately,
especially if they increase complexity or affect backporting.
@subheading Ask before you change the build system (configure, etc).
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@subheading Cosmetic changes should be kept in separate patches.
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. However,
indentation changes that can be ignored by @code{git diff --ignore-all-space}
(e.g. changes in whitespace amount, leading/trailing spaces) may be mixed with
functional changes, since reviewers can use @code{git diff -w} or
@code{git log -p --ignore-all-space} to review only the functional parts of
the change. Forgejo's pull request interface also provides a
``Hide whitespace changes'' option for this purpose. Every
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
non-whitespace cosmetic changes, separate them strictly from real changes.
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@subheading Commit messages should always be filled out properly.
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
@example
area changed: Short 1 line description
details describing what and why and giving references.
@end example
@subheading Credit the author of the patch.
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@subheading Credit any researchers
If a commit/patch fixes an issues found by some researcher, always credit the
researcher in the commit message for finding/reporting the issue.
@subheading Complex patches should refer to discussion surrounding them.
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@subheading Always wait long enough before pushing changes
Do NOT commit to code actively maintained by others without permission.
@@ -530,6 +353,22 @@ time-frame (12h for build failures and security fixes, 3 days small changes,
Also note, the maintainer can simply ask for more time to review!
@section Code
@subheading API/ABI changes should be discussed before they are made.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove widely used functionality or features (redundant code can be removed).
@subheading Remember to check if you need to bump versions for libav*.
Depending on the change, you may need to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@subheading Warnings for correct code may be disabled if there is no other option.
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
@@ -539,150 +378,10 @@ If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@section Library public interfaces
Every library in FFmpeg provides a set of public APIs in its installed headers,
which are those listed in the variable @code{HEADERS} in that library's
@file{Makefile}. All identifiers defined in those headers (except for those
explicitly documented otherwise), and corresponding symbols exported from
compiled shared or static libraries are considered public interfaces and must
comply with the API and ABI compatibility rules described in this section.
Public APIs must be backward compatible within a given major version. I.e. any
valid user code that compiles and works with a given library version must still
compile and work with any later version, as long as the major version number is
unchanged. "Valid user code" here means code that is calling our APIs in a
documented and/or intended manner and is not relying on any undefined behavior.
Incrementing the major version may break backward compatibility, but only to the
extent described in @ref{Major version bumps}.
We also guarantee backward ABI compatibility for shared and static libraries.
I.e. it should be possible to replace a shared or static build of our library
with a build of any later version (re-linking the user binary in the static
case) without breaking any valid user binaries, as long as the major version
number remains unchanged.
@subsection Adding new interfaces
Any new public identifiers in installed headers are considered new API - this
includes new functions, structs, macros, enum values, typedefs, new fields in
existing structs, new installed headers, etc. Consider the following
guidelines when adding new APIs.
@subsubheading Motivation
While new APIs can be added relatively easily, changing or removing them is much
harder due to abovementioned compatibility requirements. You should then
consider carefully whether the functionality you are adding really needs to be
exposed to our callers as new public API.
Your new API should have at least one well-established use case outside of the
library that cannot be easily achieved with existing APIs. Every library in
FFmpeg also has a defined scope - your new API must fit within it.
@subsubheading Replacing existing APIs
If your new API is replacing an existing one, it should be strictly superior to
it, so that the advantages of using the new API outweigh the cost to the
callers of changing their code. After adding the new API you should then
deprecate the old one and schedule it for removal, as described in
@ref{Removing interfaces}.
If you deem an existing API deficient and want to fix it, the preferred approach
in most cases is to add a differently-named replacement and deprecate the
existing API rather than modify it. It is important to make the changes visible
to our callers (e.g. through compile- or run-time deprecation warnings) and make
it clear how to transition to the new API (e.g. in the Doxygen documentation or
on the wiki).
@subsubheading API design
The FFmpeg libraries are used by a variety of callers to perform a wide range of
multimedia-related processing tasks. You should therefore - within reason - try
to design your new API for the broadest feasible set of use cases and avoid
unnecessarily limiting it to a specific type of callers (e.g. just media
playback or just transcoding).
@subsubheading Consistency
Check whether similar APIs already exist in FFmpeg. If they do, try to model
your new addition on them to achieve better overall consistency.
The naming of your new identifiers should follow the @ref{Naming conventions}
and be aligned with other similar APIs, if applicable.
@subsubheading Extensibility
You should also consider how your API might be extended in the future in a
backward-compatible way. If you are adding a new struct @code{AVFoo}, the
standard approach is requiring the caller to always allocate it through a
constructor function, typically named @code{av_foo_alloc()}. This way new fields
may be added to the end of the struct without breaking ABI compatibility.
Typically you will also want a destructor - @code{av_foo_free(AVFoo**)} that
frees the indirectly supplied object (and its contents, if applicable) and
writes @code{NULL} to the supplied pointer, thus eliminating the potential
dangling pointer in the caller's memory.
If you are adding new functions, consider whether it might be desirable to tweak
their behavior in the future - you may want to add a flags argument, even though
it would be unused initially.
@subsubheading Documentation
All new APIs must be documented as Doxygen-formatted comments above the
identifiers you add to the public headers. You should also briefly mention the
change in @file{doc/APIchanges}.
@subsubheading Bump the version
Backward-incompatible API or ABI changes require incrementing (bumping) the
major version number, as described in @ref{Major version bumps}. Major
bumps are significant events that happen on a schedule - so if your change
strictly requires one you should add it under @code{#if} preprocessor guards that
disable it until the next major bump happens.
New APIs that can be added without breaking API or ABI compatibility require
bumping the minor version number.
Incrementing the third (micro) version component means a noteworthy binary
compatible change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@anchor{Removing interfaces}
@subsection Removing interfaces
Due to abovementioned compatibility guarantees, removing APIs is an involved
process that should only be undertaken with good reason. Typically a deficient,
restrictive, or otherwise inadequate API is replaced by a superior one, though
it does at times happen that we remove an API without any replacement (e.g. when
the feature it provides is deemed not worth the maintenance effort, out of scope
of the project, fundamentally flawed, etc.).
The removal has two steps - first the API is deprecated and scheduled for
removal, but remains present and functional. The second step is actually
removing the API - this is described in @ref{Major version bumps}.
To deprecate an API you should signal to our users that they should stop using
it. E.g. if you intend to remove struct members or functions, you should mark
them with @code{attribute_deprecated}. When this cannot be done, it may be
possible to detect the use of the deprecated API at runtime and print a warning
(though take care not to print it too often). You should also document the
deprecation (and the replacement, if applicable) in the relevant Doxygen
documentation block.
Finally, you should define a deprecation guard along the lines of
@code{#define FF_API_<FOO> (LIBAVBAR_VERSION_MAJOR < XX)} (where XX is the major
version in which the API will be removed) in @file{libavbar/version_major.h}
(@file{version.h} in case of @code{libavutil}). Then wrap all uses of the
deprecated API in @code{#if FF_API_<FOO> .... #endif}, so that the code will
automatically get disabled once the major version reaches XX. You can also use
@code{FF_DISABLE_DEPRECATION_WARNINGS} and @code{FF_ENABLE_DEPRECATION_WARNINGS}
to suppress compiler deprecation warnings inside these guards. You should test
that the code compiles and works with the guard macro evaluating to both true
and false.
@anchor{Major version bumps}
@subsection Major version bumps
A major version bump signifies an API and/or ABI compatibility break. To reduce
the negative effects on our callers, who are required to adapt their code,
backward-incompatible changes during a major bump should be limited to:
@itemize @bullet
@item
Removing previously deprecated APIs.
@item
Performing ABI- but not API-breaking changes, like reordering struct contents.
@end itemize
@subheading Check untrusted input properly.
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
@@ -726,6 +425,35 @@ finding a new maintainer and also don't forget to update the @file{MAINTAINERS}
We think our rules are not too hard. If you have comments, contact us.
@chapter Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it's rarely good to start with that as initial assumption.
Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
Try to help other team members and cooperate if you can.
The goal of software development is to create technical excellence, not for any
individual to be better and "win" against the others. Large software projects
are only possible and successful through teamwork.
If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@chapter Submitting patches
@@ -755,8 +483,7 @@ and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Patches should be posted to
@uref{https://code.ffmpeg.org/FFmpeg/FFmpeg/pulls, Forgejo} or the
Patches should be posted to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
@@ -767,11 +494,6 @@ patch is inline or attached per mail.
You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
likely was wrong.
@subheading How to setup git send-email?
Please see @url{https://git-send-email.io/}.
For gmail additionally see @url{https://shallowsky.com/blog/tech/email/gmail-app-passwds.html}.
@subheading Sending patches from email clients
Using @code{git send-email} might not be desirable for everyone. The
following trick allows to send patches via email clients in a safe
@@ -816,7 +538,7 @@ number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the AVCodecID to @file{codec_id.h}?
Did you add the AVCodecID to @file{avcodec.h}?
When adding new codec IDs, also add an entry to the codec descriptor
list in @file{libavcodec/codec_desc.c}.
@@ -831,7 +553,7 @@ already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general_contents.texi}?
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
@@ -948,13 +670,15 @@ Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider adding a regression test for your code. All new modules
should be covered by tests. That includes demuxers, muxers, decoders, encoders
filters, bitstream filters, parsers. If its not possible to do that, add
an explanation why to your patchset, its ok to not test if there's a reason.
Consider adding a regression test for your code.
@item
If you added NASM code please check that things still work with --disable-x86asm.
If you added YASM code please check that things still work with --disable-yasm.
@item
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@item
Test your code with valgrind and or Address Sanitizer to ensure it's free
@@ -1005,8 +729,6 @@ accordingly].
@section Adding files to the fate-suite dataset
If you need a sample uploaded send a mail to samples-request.
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
First please make sure that the sample file is as small as possible to test the
@@ -1056,25 +778,6 @@ In case you need finer control over how valgrind is invoked, use the
@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
your configure line instead.
@anchor{Maintenance}
@chapter Maintenance process
@anchor{MAINTAINERS}
@section MAINTAINERS
The developers maintaining each part of the codebase are listed in @file{MAINTAINERS}.
Being listed in @file{MAINTAINERS}, gives one the right to have git write access to
the specific repository.
@anchor{Becoming a maintainer}
@section Becoming a maintainer
People add themselves to @file{MAINTAINERS} by sending a patch like any other code
change. These get reviewed by the community like any other patch. It is expected
that, if someone has an objection to a new maintainer, she is willing to object
in public with her full name and is willing to take over maintainership for the area.
@anchor{Release process}
@chapter Release process
File diff suppressed because it is too large Load Diff
+112 -1378
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File diff suppressed because it is too large Load Diff
-1
View File
@@ -22,4 +22,3 @@
/transcoding
/vaapi_encode
/vaapi_transcode
/qsv_transcode
+12 -13
View File
@@ -1,27 +1,26 @@
EXAMPLES-$(CONFIG_AVIO_HTTP_SERVE_FILES) += avio_http_serve_files
EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
EXAMPLES-$(CONFIG_AVIO_READ_CALLBACK_EXAMPLE) += avio_read_callback
EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
EXAMPLES-$(CONFIG_DECODE_FILTER_AUDIO_EXAMPLE) += decode_filter_audio
EXAMPLES-$(CONFIG_DECODE_FILTER_VIDEO_EXAMPLE) += decode_filter_video
EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
EXAMPLES-$(CONFIG_DEMUX_DECODE_EXAMPLE) += demux_decode
EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video
EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE) += hw_decode
EXAMPLES-$(CONFIG_MUX_EXAMPLE) += mux
EXAMPLES-$(CONFIG_QSV_DECODE_EXAMPLE) += qsv_decode
EXAMPLES-$(CONFIG_REMUX_EXAMPLE) += remux
EXAMPLES-$(CONFIG_RESAMPLE_AUDIO_EXAMPLE) += resample_audio
EXAMPLES-$(CONFIG_SCALE_VIDEO_EXAMPLE) += scale_video
EXAMPLES-$(CONFIG_SHOW_METADATA_EXAMPLE) += show_metadata
EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
EXAMPLES-$(CONFIG_TRANSCODE_EXAMPLE) += transcode
EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE) += vaapi_transcode
EXAMPLES-$(CONFIG_QSV_TRANSCODE_EXAMPLE) += qsv_transcode
EXAMPLES := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
EXAMPLES_G := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+14 -21
View File
@@ -11,40 +11,33 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
# missing the following targets, since they need special options in the FFmpeg build:
# qsv_decode
# qsv_transcode
# vaapi_encode
# vaapi_transcode
EXAMPLES=\
avio_http_serve_files \
avio_list_dir \
avio_read_callback \
EXAMPLES= avio_list_dir \
avio_reading \
decode_audio \
decode_filter_audio \
decode_filter_video \
decode_video \
demux_decode \
demuxing_decoding \
encode_audio \
encode_video \
extract_mvs \
filtering_video \
filtering_audio \
http_multiclient \
hw_decode \
mux \
remux \
resample_audio \
scale_video \
show_metadata \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcode
transcoding \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
encode_audio: LDLIBS += -lm
mux: LDLIBS += -lm
resample_audio: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
.phony: all clean-test clean
+3 -5
View File
@@ -7,10 +7,8 @@ that you have them installed and working on your system.
Method 1: build the installed examples in a generic read/write user directory
Copy to a read/write user directory and run:
make -f Makefile.example
It will link to the libraries on your system, assuming the PKG_CONFIG_PATH is
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
@@ -22,4 +20,4 @@ examples using "make examplesclean"
If you want to try the dedicated Makefile examples (to emulate the first
method), go into doc/examples and run a command such as
PKG_CONFIG_PATH=pc-uninstalled make -f Makefile.example
PKG_CONFIG_PATH=pc-uninstalled make.
-155
View File
@@ -1,155 +0,0 @@
/*
* Copyright (c) 2015 Stephan Holljes
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file libavformat multi-client network API usage example
* @example avio_http_serve_files.c
*
* Serve a file without decoding or demuxing it over the HTTP protocol. Multiple
* clients can connect and will receive the same file.
*/
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <unistd.h>
static void process_client(AVIOContext *client, const char *in_uri)
{
AVIOContext *input = NULL;
uint8_t buf[1024];
int ret, n, reply_code;
uint8_t *resource = NULL;
while ((ret = avio_handshake(client)) > 0) {
av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
// check for strlen(resource) is necessary, because av_opt_get()
// may return empty string.
if (resource && strlen(resource))
break;
av_freep(&resource);
}
if (ret < 0)
goto end;
av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
reply_code = 200;
} else {
reply_code = AVERROR_HTTP_NOT_FOUND;
}
if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
goto end;
}
av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
while ((ret = avio_handshake(client)) > 0);
if (ret < 0)
goto end;
fprintf(stderr, "Handshake performed.\n");
if (reply_code != 200)
goto end;
fprintf(stderr, "Opening input file.\n");
if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
av_err2str(ret));
goto end;
}
for(;;) {
n = avio_read(input, buf, sizeof(buf));
if (n < 0) {
if (n == AVERROR_EOF)
break;
av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
av_err2str(n));
break;
}
avio_write(client, buf, n);
avio_flush(client);
}
end:
fprintf(stderr, "Flushing client\n");
avio_flush(client);
fprintf(stderr, "Closing client\n");
avio_close(client);
fprintf(stderr, "Closing input\n");
avio_close(input);
av_freep(&resource);
}
int main(int argc, char **argv)
{
AVDictionary *options = NULL;
AVIOContext *client = NULL, *server = NULL;
const char *in_uri, *out_uri;
int ret, pid;
av_log_set_level(AV_LOG_TRACE);
if (argc < 3) {
printf("usage: %s input http://hostname[:port]\n"
"API example program to serve http to multiple clients.\n"
"\n", argv[0]);
return 1;
}
in_uri = argv[1];
out_uri = argv[2];
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
return ret;
}
if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
return ret;
}
fprintf(stderr, "Entering main loop.\n");
for(;;) {
if ((ret = avio_accept(server, &client)) < 0)
goto end;
fprintf(stderr, "Accepted client, forking process.\n");
// XXX: Since we don't reap our children and don't ignore signals
// this produces zombie processes.
pid = fork();
if (pid < 0) {
perror("Fork failed");
ret = AVERROR(errno);
goto end;
}
if (pid == 0) {
fprintf(stderr, "In child.\n");
process_client(client, in_uri);
avio_close(server);
exit(0);
}
if (pid > 0)
avio_close(client);
}
end:
avio_close(server);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}
-7
View File
@@ -20,13 +20,6 @@
* THE SOFTWARE.
*/
/**
* @file libavformat AVIOContext list directory API usage example
* @example avio_list_dir.c
*
* Show how to list directories through the libavformat AVIOContext API.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
-135
View File
@@ -1,135 +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 read callback API usage example
* @example avio_read_callback.c
*
* Make libavformat demuxer access media content through a custom
* AVIOContext read callback.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
#include <libavutil/mem.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);
if (!buf_size)
return AVERROR_EOF;
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];
/* 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) {
av_freep(&avio_ctx_buffer);
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);
avio_context_free(&avio_ctx);
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}
+134
View File
@@ -0,0 +1,134 @@
/*
* 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);
if (!buf_size)
return AVERROR_EOF;
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];
/* 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);
avio_context_free(&avio_ctx);
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}
+6 -11
View File
@@ -21,11 +21,10 @@
*/
/**
* @file libavcodec audio decoding API usage example
* @example decode_audio.c
* @file
* audio decoding with libavcodec API example
*
* Decode data from an MP2 input file and generate a raw audio file to
* be played with ffplay.
* @example decode_audio.c
*/
#include <stdio.h>
@@ -98,7 +97,7 @@ static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
exit(1);
}
for (i = 0; i < frame->nb_samples; i++)
for (ch = 0; ch < dec_ctx->ch_layout.nb_channels; ch++)
for (ch = 0; ch < dec_ctx->channels; ch++)
fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
}
}
@@ -128,10 +127,6 @@ int main(int argc, char **argv)
outfilename = argv[2];
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
exit(1); /* or proper cleanup and returning */
}
/* find the MPEG audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
@@ -165,7 +160,7 @@ int main(int argc, char **argv)
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
fprintf(stderr, "Could not open %s\n", outfilename);
av_free(c);
exit(1);
}
@@ -220,7 +215,7 @@ int main(int argc, char **argv)
sfmt = av_get_packed_sample_fmt(sfmt);
}
n_channels = c->ch_layout.nb_channels;
n_channels = c->channels;
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
-319
View File
@@ -1,319 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file audio decoding and filtering usage example
* @example decode_filter_audio.c
*
* Demux, decode and filter audio input file, generate a raw audio
* file to be played with ffplay.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/channel_layout.h>
#include <libavutil/mem.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
const AVCodec *dec;
int ret;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const int out_sample_rate = 8000;
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
ret = snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt));
av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "out");
if (!buffersink_ctx) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
ret = AVERROR(ENOMEM);
goto end;
}
ret = av_opt_set(buffersink_ctx, "sample_formats", "s16",
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(buffersink_ctx, "channel_layouts", "mono",
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_array(buffersink_ctx, "samplerates", AV_OPT_SEARCH_CHILDREN,
0, 1, AV_OPT_TYPE_INT, &out_sample_rate);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize audio buffer sink\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * frame->ch_layout.nb_channels;
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket *packet = av_packet_alloc();
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (!packet || !frame || !filt_frame) {
fprintf(stderr, "Could not allocate frame or packet\n");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
break;
if (packet->stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
if (ret >= 0) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
}
av_packet_unref(packet);
}
if (ret == AVERROR_EOF) {
/* signal EOF to the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, NULL, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while closing the filtergraph\n");
goto end;
}
/* pull remaining frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_packet_free(&packet);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}
-317
View File
@@ -1,317 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 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
* API example for decoding and filtering
* @example decode_filter_video.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/mem.h>
#include <libavutil/opt.h>
#include <libavutil/time.h>
const char *filter_descr = "scale=78:24,transpose=cclock";
/* other way:
scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
*/
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
const AVCodec *dec;
int ret;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->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;
}
/* buffer video sink: to terminate the filter chain. */
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, buffersink, "out");
if (!buffersink_ctx) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
ret = AVERROR(ENOMEM);
goto end;
}
ret = av_opt_set(buffersink_ctx, "pixel_formats", "gray8",
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize buffer sink\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
av_usleep(delay);
}
last_pts = frame->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket *packet;
AVFrame *frame;
AVFrame *filt_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
packet = av_packet_alloc();
if (!frame || !filt_frame || !packet) {
fprintf(stderr, "Could not allocate frame or packet\n");
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
break;
if (packet->stream_index == video_stream_index) {
ret = avcodec_send_packet(dec_ctx, packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
frame->pts = frame->best_effort_timestamp;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_packet_unref(packet);
}
if (ret == AVERROR_EOF) {
/* signal EOF to the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, NULL, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while closing the filtergraph\n");
goto end;
}
/* pull remaining frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
av_packet_free(&packet);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}
+10 -19
View File
@@ -21,11 +21,10 @@
*/
/**
* @file libavcodec video decoding API usage example
* @example decode_video.c *
* @file
* video decoding with libavcodec API example
*
* Read from an MPEG1 video file, decode frames, and generate PGM images as
* output.
* @example decode_video.c
*/
#include <stdio.h>
@@ -42,11 +41,7 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
FILE *f;
int i;
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
return;
}
f = fopen(filename,"wb");
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);
@@ -74,12 +69,12 @@ static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
exit(1);
}
printf("saving frame %3"PRId64"\n", dec_ctx->frame_num);
printf("saving frame %3d\n", dec_ctx->frame_number);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), "%s-%"PRId64, filename, dec_ctx->frame_num);
snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
pgm_save(frame->data[0], frame->linesize[0],
frame->width, frame->height, buf);
}
@@ -97,7 +92,6 @@ int main(int argc, char **argv)
uint8_t *data;
size_t data_size;
int ret;
int eof;
AVPacket *pkt;
if (argc <= 2) {
@@ -156,16 +150,15 @@ int main(int argc, char **argv)
exit(1);
}
do {
while (!feof(f)) {
/* read raw data from the input file */
data_size = fread(inbuf, 1, INBUF_SIZE, f);
if (ferror(f))
if (!data_size)
break;
eof = !data_size;
/* use the parser to split the data into frames */
data = inbuf;
while (data_size > 0 || eof) {
while (data_size > 0) {
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
@@ -177,10 +170,8 @@ int main(int argc, char **argv)
if (pkt->size)
decode(c, frame, pkt, outfilename);
else if (eof)
break;
}
} while (!eof);
}
/* flush the decoder */
decode(c, frame, NULL, outfilename);
-385
View File
@@ -1,385 +0,0 @@
/*
* Copyright (c) 2012 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 and libavcodec demuxing and decoding API usage example
* @example demux_decode.c
*
* Show how to use the libavformat and libavcodec API to demux and decode audio
* and video data. Write the output as raw audio and input files to be played by
* ffplay.
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static int width, height;
static enum AVPixelFormat pix_fmt;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket *pkt = NULL;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int output_video_frame(AVFrame *frame)
{
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame n:%d\n",
video_frame_count++);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy2(video_dst_data, video_dst_linesize,
frame->data, frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
return 0;
}
static int output_audio_frame(AVFrame *frame)
{
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame n:%d nb_samples:%d pts:%s\n",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
return 0;
}
static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
{
int ret = 0;
// submit the packet to the decoder
ret = avcodec_send_packet(dec, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
return ret;
}
// get all the available frames from the decoder
while (ret >= 0) {
ret = avcodec_receive_frame(dec, frame);
if (ret < 0) {
// those two return values are special and mean there is no output
// frame available, but there were no errors during decoding
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
return 0;
fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
return ret;
}
// write the frame data to output file
if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
ret = output_video_frame(frame);
else
ret = output_audio_frame(frame);
av_frame_unref(frame);
}
return ret;
}
static int open_codec_context(int *stream_idx,
AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret, stream_index;
AVStream *st;
const AVCodec *dec = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
stream_index = ret;
st = fmt_ctx->streams[stream_index];
/* find decoder for the stream */
dec = avcodec_find_decoder(st->codecpar->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Allocate a codec context for the decoder */
*dec_ctx = avcodec_alloc_context3(dec);
if (!*dec_ctx) {
fprintf(stderr, "Failed to allocate the %s codec context\n",
av_get_media_type_string(type));
return AVERROR(ENOMEM);
}
/* Copy codec parameters from input stream to output codec context */
if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
av_get_media_type_string(type));
return ret;
}
/* Init the decoders */
if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
*stream_idx = stream_index;
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0;
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",
argv[0]);
exit(1);
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
width = video_dec_ctx->width;
height = video_dec_ctx->height;
pix_fmt = video_dec_ctx->pix_fmt;
ret = av_image_alloc(video_dst_data, video_dst_linesize,
width, height, pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate packet\n");
ret = AVERROR(ENOMEM);
goto end;
}
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, pkt) >= 0) {
// check if the packet belongs to a stream we are interested in, otherwise
// skip it
if (pkt->stream_index == video_stream_idx)
ret = decode_packet(video_dec_ctx, pkt);
else if (pkt->stream_index == audio_stream_idx)
ret = decode_packet(audio_dec_ctx, pkt);
av_packet_unref(pkt);
if (ret < 0)
break;
}
/* flush the decoders */
if (video_dec_ctx)
decode_packet(video_dec_ctx, NULL);
if (audio_dec_ctx)
decode_packet(audio_dec_ctx, NULL);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pixel_format %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(pix_fmt), width, height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
AVChannelLayout mono = AV_CHANNEL_LAYOUT_MONO;
AVChannelLayout *ch_layout = &audio_dec_ctx->ch_layout;
const char *fmt;
char buf[64];
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
ch_layout = &mono;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
if ((ret = av_channel_layout_describe(ch_layout, buf, sizeof(buf))) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ch_layout %s -sample_rate %d %s\n",
fmt, buf, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_free_context(&video_dec_ctx);
avcodec_free_context(&audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_packet_free(&pkt);
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;
}
+382
View File
@@ -0,0 +1,382 @@
/*
* Copyright (c) 2012 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
* Demuxing and decoding example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example demuxing_decoding.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static int width, height;
static enum AVPixelFormat pix_fmt;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket *pkt = NULL;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int output_video_frame(AVFrame *frame)
{
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame n:%d coded_n:%d\n",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
return 0;
}
static int output_audio_frame(AVFrame *frame)
{
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame n:%d nb_samples:%d pts:%s\n",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
return 0;
}
static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
{
int ret = 0;
// submit the packet to the decoder
ret = avcodec_send_packet(dec, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
return ret;
}
// get all the available frames from the decoder
while (ret >= 0) {
ret = avcodec_receive_frame(dec, frame);
if (ret < 0) {
// those two return values are special and mean there is no output
// frame available, but there were no errors during decoding
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
return 0;
fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
return ret;
}
// write the frame data to output file
if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
ret = output_video_frame(frame);
else
ret = output_audio_frame(frame);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
return 0;
}
static int open_codec_context(int *stream_idx,
AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret, stream_index;
AVStream *st;
const AVCodec *dec = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
stream_index = ret;
st = fmt_ctx->streams[stream_index];
/* find decoder for the stream */
dec = avcodec_find_decoder(st->codecpar->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Allocate a codec context for the decoder */
*dec_ctx = avcodec_alloc_context3(dec);
if (!*dec_ctx) {
fprintf(stderr, "Failed to allocate the %s codec context\n",
av_get_media_type_string(type));
return AVERROR(ENOMEM);
}
/* Copy codec parameters from input stream to output codec context */
if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
av_get_media_type_string(type));
return ret;
}
/* Init the decoders */
if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
*stream_idx = stream_index;
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0;
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",
argv[0]);
exit(1);
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
width = video_dec_ctx->width;
height = video_dec_ctx->height;
pix_fmt = video_dec_ctx->pix_fmt;
ret = av_image_alloc(video_dst_data, video_dst_linesize,
width, height, pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate packet\n");
ret = AVERROR(ENOMEM);
goto end;
}
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, pkt) >= 0) {
// check if the packet belongs to a stream we are interested in, otherwise
// skip it
if (pkt->stream_index == video_stream_idx)
ret = decode_packet(video_dec_ctx, pkt);
else if (pkt->stream_index == audio_stream_idx)
ret = decode_packet(audio_dec_ctx, pkt);
av_packet_unref(pkt);
if (ret < 0)
break;
}
/* flush the decoders */
if (video_dec_ctx)
decode_packet(video_dec_ctx, NULL);
if (audio_dec_ctx)
decode_packet(audio_dec_ctx, NULL);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(pix_fmt), width, height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_free_context(&video_dec_ctx);
avcodec_free_context(&audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_packet_free(&pkt);
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;
}
+23 -47
View File
@@ -21,10 +21,10 @@
*/
/**
* @file libavcodec encoding audio API usage examples
* @example encode_audio.c
* @file
* audio encoding with libavcodec API example.
*
* Generate a synthetic audio signal and encode it to an output MP2 file.
* @example encode_audio.c
*/
#include <stdint.h>
@@ -41,17 +41,7 @@
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const void *out_config;
int ret = avcodec_get_supported_config(NULL, codec, AV_CODEC_CONFIG_SAMPLE_FORMAT,
0, &out_config, NULL);
if (ret < 0) {
fprintf(stderr, "Error getting supported sample formats\n");
exit(1);
}
const enum AVSampleFormat *p = out_config;
if (!p)
return 1;
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
@@ -64,19 +54,13 @@ static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt
/* just pick the highest supported samplerate */
static int select_sample_rate(const AVCodec *codec)
{
const void *out_config;
int ret = avcodec_get_supported_config(NULL, codec, AV_CODEC_CONFIG_SAMPLE_RATE,
0, &out_config, NULL);
if (ret < 0) {
fprintf(stderr, "Error getting supported sample rates\n");
exit(1);
}
const int *p = out_config;
const int *p;
int best_samplerate = 0;
if (!p)
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
best_samplerate = *p;
@@ -86,31 +70,26 @@ static int select_sample_rate(const AVCodec *codec)
}
/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
static int select_channel_layout(const AVCodec *codec)
{
const void *out_config;
int ret = avcodec_get_supported_config(NULL, codec, AV_CODEC_CONFIG_CHANNEL_LAYOUT,
0, &out_config, NULL);
if (ret < 0) {
fprintf(stderr, "Error getting supported channel layouts\n");
exit(1);
}
const AVChannelLayout *p = out_config, *best_ch_layout;
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!p)
return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
while (p->nb_channels) {
int nb_channels = p->nb_channels;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = p;
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return av_channel_layout_copy(dst, best_ch_layout);
return best_ch_layout;
}
static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
@@ -185,9 +164,8 @@ int main(int argc, char **argv)
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
ret = select_channel_layout(codec, &c->ch_layout);
if (ret < 0)
exit(1);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
@@ -217,9 +195,7 @@ int main(int argc, char **argv)
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
if (ret < 0)
exit(1);
frame->channel_layout = c->channel_layout;
/* allocate the data buffers */
ret = av_frame_get_buffer(frame, 0);
@@ -240,10 +216,10 @@ int main(int argc, char **argv)
samples = (uint16_t*)frame->data[0];
for (j = 0; j < c->frame_size; j++) {
samples[c->ch_layout.nb_channels*j] = (int)(sin(t) * 10000);
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->ch_layout.nb_channels; k++)
samples[c->ch_layout.nb_channels*j + k] = samples[c->ch_layout.nb_channels*j];
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
encode(c, frame, pkt, f);
+4 -4
View File
@@ -21,10 +21,10 @@
*/
/**
* @file libavcodec encoding video API usage example
* @example encode_video.c
* @file
* video encoding with libavcodec API example
*
* Generate synthetic video data and encode it to an output file.
* @example encode_video.c
*/
#include <stdio.h>
@@ -202,7 +202,7 @@ int main(int argc, char **argv)
It makes only sense because this tiny examples writes packets
directly. This is called "elementary stream" and only works for some
codecs. To create a valid file, you usually need to write packets
into a proper file format or protocol; see mux.c.
into a proper file format or protocol; see muxing.c.
*/
if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
fwrite(endcode, 1, sizeof(endcode), f);
+3 -13
View File
@@ -21,14 +21,6 @@
* THE SOFTWARE.
*/
/**
* @file libavcodec motion vectors extraction API usage example
* @example extract_mvs.c
*
* Read from input file, decode video stream and print a motion vectors
* representation to stdout.
*/
#include <libavutil/motion_vector.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
@@ -69,11 +61,10 @@ static int decode_packet(const AVPacket *pkt)
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64",%4d,%4d,%4d\n",
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags,
mv->motion_x, mv->motion_y, mv->motion_scale);
mv->dst_x, mv->dst_y, mv->flags);
}
}
av_frame_unref(frame);
@@ -109,7 +100,6 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
ret = avcodec_parameters_to_context(dec_ctx, st->codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters to codec context\n");
avcodec_free_context(&dec_ctx);
return ret;
}
@@ -176,7 +166,7 @@ int main(int argc, char **argv)
goto end;
}
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags,motion_x,motion_y,motion_scale\n");
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
/* read frames from the file */
while (av_read_frame(fmt_ctx, pkt) >= 0) {
+25 -27
View File
@@ -19,11 +19,13 @@
*/
/**
* @file libavfilter audio filtering API usage example
* @example filter_audio.c
* @file
* libavfilter API usage example.
*
* 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.
* @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)
@@ -41,19 +43,19 @@
#include <stdio.h>
#include <stdlib.h>
#include <libavutil/channel_layout.h>
#include <libavutil/md5.h>
#include <libavutil/mem.h>
#include <libavutil/opt.h>
#include <libavutil/samplefmt.h>
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include <libavfilter/avfilter.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#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 (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
@@ -98,7 +100,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
}
/* Set the filter options through the AVOptions API. */
av_channel_layout_describe(&INPUT_CHANNEL_LAYOUT, ch_layout, sizeof(ch_layout));
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
@@ -152,8 +154,9 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=stereo",
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100);
"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");
@@ -212,7 +215,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
static int process_output(struct AVMD5 *md5, AVFrame *frame)
{
int planar = av_sample_fmt_is_planar(frame->format);
int channels = frame->ch_layout.nb_channels;
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
@@ -245,7 +248,7 @@ static int get_input(AVFrame *frame, int frame_num)
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
av_channel_layout_copy(&frame->ch_layout, &INPUT_CHANNEL_LAYOUT);
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;
@@ -270,6 +273,7 @@ int main(int argc, char *argv[])
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
@@ -294,7 +298,6 @@ int main(int argc, char *argv[])
md5 = av_md5_alloc();
if (!md5) {
av_frame_free(&frame);
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
@@ -302,10 +305,8 @@ int main(int argc, char *argv[])
/* Set up the filtergraph. */
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
av_frame_free(&frame);
av_freep(&md5);
fprintf(stderr, "Unable to init filter graph:");
return 1;
goto fail;
}
/* the main filtering loop */
@@ -356,10 +357,7 @@ int main(int argc, char *argv[])
return 0;
fail:
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
fprintf(stderr, "%s\n", av_err2str(err));
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}
+294
View File
@@ -0,0 +1,294 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
const AVCodec *dec;
int ret;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
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)
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,
time_base.num, 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, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "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_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;
}
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;
}
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;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket *packet = av_packet_alloc();
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (!packet || !frame || !filt_frame) {
fprintf(stderr, "Could not allocate frame or packet\n");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
break;
if (packet->stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
if (ret >= 0) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
}
av_packet_unref(packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_packet_free(&packet);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}
+293
View File
@@ -0,0 +1,293 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 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
* API example for decoding and filtering
* @example filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24,transpose=cclock";
/* other way:
scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
*/
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
const AVCodec *dec;
int ret;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->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;
}
/* buffer video sink: to terminate the filter chain. */
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_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = frame->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket *packet;
AVFrame *frame;
AVFrame *filt_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
packet = av_packet_alloc();
if (!frame || !filt_frame || !packet) {
fprintf(stderr, "Could not allocate frame or packet\n");
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, packet)) < 0)
break;
if (packet->stream_index == video_stream_index) {
ret = avcodec_send_packet(dec_ctx, packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
frame->pts = frame->best_effort_timestamp;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_packet_unref(packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
av_packet_free(&packet);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}
+156
View File
@@ -0,0 +1,156 @@
/*
* Copyright (c) 2015 Stephan Holljes
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat multi-client network API usage example.
*
* @example http_multiclient.c
* This example will serve a file without decoding or demuxing it over http.
* Multiple clients can connect and will receive the same file.
*/
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <unistd.h>
static void process_client(AVIOContext *client, const char *in_uri)
{
AVIOContext *input = NULL;
uint8_t buf[1024];
int ret, n, reply_code;
uint8_t *resource = NULL;
while ((ret = avio_handshake(client)) > 0) {
av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
// check for strlen(resource) is necessary, because av_opt_get()
// may return empty string.
if (resource && strlen(resource))
break;
av_freep(&resource);
}
if (ret < 0)
goto end;
av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
reply_code = 200;
} else {
reply_code = AVERROR_HTTP_NOT_FOUND;
}
if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
goto end;
}
av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
while ((ret = avio_handshake(client)) > 0);
if (ret < 0)
goto end;
fprintf(stderr, "Handshake performed.\n");
if (reply_code != 200)
goto end;
fprintf(stderr, "Opening input file.\n");
if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
av_err2str(ret));
goto end;
}
for(;;) {
n = avio_read(input, buf, sizeof(buf));
if (n < 0) {
if (n == AVERROR_EOF)
break;
av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
av_err2str(n));
break;
}
avio_write(client, buf, n);
avio_flush(client);
}
end:
fprintf(stderr, "Flushing client\n");
avio_flush(client);
fprintf(stderr, "Closing client\n");
avio_close(client);
fprintf(stderr, "Closing input\n");
avio_close(input);
av_freep(&resource);
}
int main(int argc, char **argv)
{
AVDictionary *options = NULL;
AVIOContext *client = NULL, *server = NULL;
const char *in_uri, *out_uri;
int ret, pid;
av_log_set_level(AV_LOG_TRACE);
if (argc < 3) {
printf("usage: %s input http://hostname[:port]\n"
"API example program to serve http to multiple clients.\n"
"\n", argv[0]);
return 1;
}
in_uri = argv[1];
out_uri = argv[2];
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
return ret;
}
if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
return ret;
}
fprintf(stderr, "Entering main loop.\n");
for(;;) {
if ((ret = avio_accept(server, &client)) < 0)
goto end;
fprintf(stderr, "Accepted client, forking process.\n");
// XXX: Since we don't reap our children and don't ignore signals
// this produces zombie processes.
pid = fork();
if (pid < 0) {
perror("Fork failed");
ret = AVERROR(errno);
goto end;
}
if (pid == 0) {
fprintf(stderr, "In child.\n");
process_client(client, in_uri);
avio_close(server);
exit(0);
}
if (pid > 0)
avio_close(client);
}
end:
avio_close(server);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}
+7 -14
View File
@@ -24,18 +24,18 @@
*/
/**
* @file HW-accelerated decoding API usage.example
* @example hw_decode.c
* @file
* HW-Accelerated decoding example.
*
* Perform HW-accelerated decoding with output frames from HW video
* surfaces.
* @example hw_decode.c
* This example shows how to do HW-accelerated decoding with output
* frames from the HW video surfaces.
*/
#include <stdio.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/mem.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
#include <libavutil/opt.h>
@@ -132,9 +132,8 @@ static int decode_write(AVCodecContext *avctx, AVPacket *packet)
goto fail;
}
if (fwrite(buffer, 1, size, output_file) != size) {
if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
fprintf(stderr, "Failed to dump raw data.\n");
ret = -1;
goto fail;
}
@@ -216,10 +215,8 @@ int main(int argc, char *argv[])
return AVERROR(ENOMEM);
video = input_ctx->streams[video_stream];
if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0) {
avcodec_free_context(&decoder_ctx);
if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
return -1;
}
decoder_ctx->get_format = get_hw_format;
@@ -233,10 +230,6 @@ int main(int argc, char *argv[])
/* open the file to dump raw data */
output_file = fopen(argv[3], "w+b");
if (!output_file) {
fprintf(stderr, "Cannot open output file '%s'\n", argv[3]);
return -1;
}
/* actual decoding and dump the raw data */
while (ret >= 0) {
+60
View File
@@ -0,0 +1,60 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* 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
* Shows how the metadata API can be used in application programs.
* @example metadata.c
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
const AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
return 0;
}
-659
View File
@@ -1,659 +0,0 @@
/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file libavformat muxing API usage example
* @example mux.c
*
* Generate a synthetic audio and video signal and mux them to a media file in
* any supported libavformat format. The default codecs are used.
*/
#include <stdlib.h>
#include <stdio.h>
#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 <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#define STREAM_DURATION 10.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
AVCodecContext *enc;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
AVPacket *tmp_pkt;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
AVStream *st, AVFrame *frame, AVPacket *pkt)
{
int ret;
// send the frame to the encoder
ret = avcodec_send_frame(c, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame to the encoder: %s\n",
av_err2str(ret));
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_packet(c, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
exit(1);
}
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, c->time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
ret = av_interleaved_write_frame(fmt_ctx, pkt);
/* pkt is now blank (av_interleaved_write_frame() takes ownership of
* its contents and resets pkt), so that no unreferencing is necessary.
* This would be different if one used av_write_frame(). */
if (ret < 0) {
fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
exit(1);
}
}
return ret == AVERROR_EOF ? 1 : 0;
}
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
const AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int ret;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
ost->tmp_pkt = av_packet_alloc();
if (!ost->tmp_pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = avcodec_alloc_context3(*codec);
if (!c) {
fprintf(stderr, "Could not alloc an encoding context\n");
exit(1);
}
ost->enc = c;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO: {
const void *codec_config;
c->bit_rate = 64000;
ret = avcodec_get_supported_config(c, NULL, AV_CODEC_CONFIG_SAMPLE_FORMAT,
0, &codec_config, NULL);
if (ret < 0) {
fprintf(stderr, "Failed to get supported sample formats\n");
exit(1);
}
c->sample_fmt = codec_config ? *(const enum AVSampleFormat*)codec_config
: AV_SAMPLE_FMT_FLTP;
ret = avcodec_get_supported_config(c, NULL, AV_CODEC_CONFIG_SAMPLE_RATE,
0, &codec_config, NULL);
if (ret < 0) {
fprintf(stderr, "Failed to get supported sample rates\n");
exit(1);
}
if (codec_config) {
const int *supported_samplerates = codec_config;
c->sample_rate = supported_samplerates[0];
for (; *supported_samplerates; supported_samplerates++) {
if (*supported_samplerates == 44100)
c->sample_rate = 44100;
}
} else {
c->sample_rate = 44100;
}
av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
}
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
const AVChannelLayout *channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
av_channel_layout_copy(&frame->ch_layout, channel_layout);
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
if (av_frame_get_buffer(frame, 0) < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, const AVCodec *codec,
OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, &c->ch_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, &c->ch_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_chlayout (ost->swr_ctx, "in_chlayout", &c->ch_layout, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_chlayout (ost->swr_ctx, "out_chlayout", &c->ch_layout, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->enc->ch_layout.nb_channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVFrame *frame;
int ret;
int dst_nb_samples;
c = ost->enc;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples;
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
}
/**************************************************************/
/* video output */
static AVFrame *alloc_frame(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *frame;
int ret;
frame = av_frame_alloc();
if (!frame)
return NULL;
frame->format = pix_fmt;
frame->width = width;
frame->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return frame;
}
static void open_video(AVFormatContext *oc, const AVCodec *codec,
OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a reusable frame */
ost->frame = alloc_frame(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
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_frame(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary video frame\n");
exit(1);
}
}
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i;
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->enc;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
return NULL;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally; make sure we do not overwrite it here */
if (av_frame_make_writable(ost->frame) < 0)
exit(1);
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
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)
{
return write_frame(oc, ost->enc, ost->st, get_video_frame(ost), ost->tmp_pkt);
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
av_packet_free(&ost->tmp_pkt);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const AVOutputFormat *fmt;
const char *filename;
AVFormatContext *oc;
const AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
for (i = 2; i+1 < argc; i+=2) {
if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
return 1;
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
}
}
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}
+661
View File
@@ -0,0 +1,661 @@
/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format. The default
* codecs are used.
* @example muxing.c
*/
#include <stdlib.h>
#include <stdio.h>
#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 <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#define STREAM_DURATION 10.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
AVCodecContext *enc;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
AVPacket *tmp_pkt;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
AVStream *st, AVFrame *frame, AVPacket *pkt)
{
int ret;
// send the frame to the encoder
ret = avcodec_send_frame(c, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame to the encoder: %s\n",
av_err2str(ret));
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_packet(c, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
exit(1);
}
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, c->time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
ret = av_interleaved_write_frame(fmt_ctx, pkt);
/* pkt is now blank (av_interleaved_write_frame() takes ownership of
* its contents and resets pkt), so that no unreferencing is necessary.
* This would be different if one used av_write_frame(). */
if (ret < 0) {
fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
exit(1);
}
}
return ret == AVERROR_EOF ? 1 : 0;
}
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
const AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
ost->tmp_pkt = av_packet_alloc();
if (!ost->tmp_pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = avcodec_alloc_context3(*codec);
if (!c) {
fprintf(stderr, "Could not alloc an encoding context\n");
exit(1);
}
ost->enc = c;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, const AVCodec *codec,
OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->enc->channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVFrame *frame;
int ret;
int dst_nb_samples;
c = ost->enc;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
}
/**************************************************************/
/* video output */
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
int ret;
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, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
static void open_video(AVFormatContext *oc, const AVCodec *codec,
OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
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");
exit(1);
}
}
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i;
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->enc;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
return NULL;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally; make sure we do not overwrite it here */
if (av_frame_make_writable(ost->frame) < 0)
exit(1);
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
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)
{
return write_frame(oc, ost->enc, ost->st, get_video_frame(ost), ost->tmp_pkt);
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
av_packet_free(&ost->tmp_pkt);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const AVOutputFormat *fmt;
const char *filename;
AVFormatContext *oc;
const AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
for (i = 2; i+1 < argc; i+=2) {
if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
return 1;
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}
-246
View File
@@ -1,246 +0,0 @@
/*
* Copyright (c) 2015 Anton Khirnov
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file Intel QSV-accelerated H.264 decoding API usage example
* @example qsv_decode.c
*
* Perform QSV-accelerated H.264 decoding with output frames in the
* GPU video surfaces, write the decoded frames to an output file.
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavcodec/avcodec.h>
#include <libavutil/buffer.h>
#include <libavutil/error.h>
#include <libavutil/hwcontext.h>
#include <libavutil/hwcontext_qsv.h>
#include <libavutil/imgutils.h>
#include <libavutil/mem.h>
static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
{
while (*pix_fmts != AV_PIX_FMT_NONE) {
if (*pix_fmts == AV_PIX_FMT_QSV) {
return AV_PIX_FMT_QSV;
}
pix_fmts++;
}
fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
return AV_PIX_FMT_NONE;
}
static int decode_packet(AVCodecContext *decoder_ctx,
AVFrame *frame, AVFrame *sw_frame,
AVPacket *pkt, AVIOContext *output_ctx)
{
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
while (ret >= 0) {
int i, j;
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
/* A real program would do something useful with the decoded frame here.
* We just retrieve the raw data and write it to a file, which is rather
* useless but pedagogic. */
ret = av_hwframe_transfer_data(sw_frame, frame, 0);
if (ret < 0) {
fprintf(stderr, "Error transferring the data to system memory\n");
goto fail;
}
for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++) {
int h = sw_frame->height >> (i > 0);
int linesize = av_image_get_linesize(sw_frame->format, sw_frame->width, i);
if (linesize < 0) {
ret = linesize;
goto fail;
}
for (j = 0; j < h; j++)
avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], linesize);
}
fail:
av_frame_unref(sw_frame);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
AVFormatContext *input_ctx = NULL;
AVStream *video_st = NULL;
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder;
AVPacket *pkt = NULL;
AVFrame *frame = NULL, *sw_frame = NULL;
AVIOContext *output_ctx = NULL;
int ret, i;
AVBufferRef *device_ref = NULL;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
/* open the input file */
ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
goto finish;
}
/* find the first H.264 video stream */
for (i = 0; i < input_ctx->nb_streams; i++) {
AVStream *st = input_ctx->streams[i];
if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
video_st = st;
else
st->discard = AVDISCARD_ALL;
}
if (!video_st) {
fprintf(stderr, "No H.264 video stream in the input file\n");
goto finish;
}
/* open the hardware device */
ret = av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV,
"auto", NULL, 0);
if (ret < 0) {
fprintf(stderr, "Cannot open the hardware device\n");
goto finish;
}
/* initialize the decoder */
decoder = avcodec_find_decoder_by_name("h264_qsv");
if (!decoder) {
fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
goto finish;
}
decoder_ctx = avcodec_alloc_context3(decoder);
if (!decoder_ctx) {
ret = AVERROR(ENOMEM);
goto finish;
}
decoder_ctx->codec_id = AV_CODEC_ID_H264;
if (video_st->codecpar->extradata_size) {
decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!decoder_ctx->extradata) {
ret = AVERROR(ENOMEM);
goto finish;
}
memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
video_st->codecpar->extradata_size);
decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
}
decoder_ctx->hw_device_ctx = av_buffer_ref(device_ref);
decoder_ctx->get_format = get_format;
ret = avcodec_open2(decoder_ctx, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Error opening the decoder: ");
goto finish;
}
/* open the output stream */
ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Error opening the output context: ");
goto finish;
}
frame = av_frame_alloc();
sw_frame = av_frame_alloc();
pkt = av_packet_alloc();
if (!frame || !sw_frame || !pkt) {
ret = AVERROR(ENOMEM);
goto finish;
}
/* actual decoding */
while (ret >= 0) {
ret = av_read_frame(input_ctx, pkt);
if (ret < 0)
break;
if (pkt->stream_index == video_st->index)
ret = decode_packet(decoder_ctx, frame, sw_frame, pkt, output_ctx);
av_packet_unref(pkt);
}
/* flush the decoder */
ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
finish:
if (ret < 0)
fprintf(stderr, "%s\n", av_err2str(ret));
avformat_close_input(&input_ctx);
av_frame_free(&frame);
av_frame_free(&sw_frame);
av_packet_free(&pkt);
avcodec_free_context(&decoder_ctx);
av_buffer_unref(&device_ref);
avio_close(output_ctx);
return ret;
}
-442
View File
@@ -1,442 +0,0 @@
/*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file Intel QSV-accelerated video transcoding API usage example
* @example qsv_transcode.c
*
* Perform QSV-accelerated transcoding and show to dynamically change
* encoder's options.
*
* Usage: qsv_transcode input_stream codec output_stream initial option
* { frame_number new_option }
* e.g: - qsv_transcode input.mp4 h264_qsv output_h264.mp4 "g 60"
* - qsv_transcode input.mp4 hevc_qsv output_hevc.mp4 "g 60 async_depth 1"
* 100 "g 120"
* (initialize codec with gop_size 60 and change it to 120 after 100
* frames)
*/
#include <stdio.h>
#include <errno.h>
#include <libavutil/hwcontext.h>
#include <libavutil/mem.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
static AVBufferRef *hw_device_ctx = NULL;
static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
static int video_stream = -1;
typedef struct DynamicSetting {
int frame_number;
char* optstr;
} DynamicSetting;
static DynamicSetting *dynamic_setting;
static int setting_number;
static int current_setting_number;
static int str_to_dict(char* optstr, AVDictionary **opt)
{
char *key, *value;
if (strlen(optstr) == 0)
return 0;
key = strtok(optstr, " ");
if (key == NULL)
return AVERROR(EINVAL);
value = strtok(NULL, " ");
if (value == NULL)
return AVERROR(EINVAL);
av_dict_set(opt, key, value, 0);
do {
key = strtok(NULL, " ");
if (key == NULL)
return 0;
value = strtok(NULL, " ");
if (value == NULL)
return AVERROR(EINVAL);
av_dict_set(opt, key, value, 0);
} while(1);
}
static int dynamic_set_parameter(AVCodecContext *avctx)
{
AVDictionary *opts = NULL;
int ret = 0;
static int frame_number = 0;
frame_number++;
if (current_setting_number < setting_number &&
frame_number == dynamic_setting[current_setting_number].frame_number) {
AVDictionaryEntry *e = NULL;
ret = str_to_dict(dynamic_setting[current_setting_number++].optstr, &opts);
if (ret < 0) {
fprintf(stderr, "The dynamic parameter is wrong\n");
goto fail;
}
/* Set common option. The dictionary will be freed and replaced
* by a new one containing all options not found in common option list.
* Then this new dictionary is used to set private option. */
if ((ret = av_opt_set_dict(avctx, &opts)) < 0)
goto fail;
/* Set codec specific option */
if ((ret = av_opt_set_dict(avctx->priv_data, &opts)) < 0)
goto fail;
/* There is no "framerate" option in common option list. Use "-r" to set
* framerate, which is compatible with ffmpeg commandline. The video is
* assumed to be average frame rate, so set time_base to 1/framerate. */
e = av_dict_get(opts, "r", NULL, 0);
if (e) {
avctx->framerate = av_d2q(atof(e->value), INT_MAX);
encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
}
}
fail:
av_dict_free(&opts);
return ret;
}
static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
{
while (*pix_fmts != AV_PIX_FMT_NONE) {
if (*pix_fmts == AV_PIX_FMT_QSV) {
return AV_PIX_FMT_QSV;
}
pix_fmts++;
}
fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
return AV_PIX_FMT_NONE;
}
static int open_input_file(char *filename)
{
int ret;
const AVCodec *decoder = NULL;
AVStream *video = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
filename, av_err2str(ret));
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
av_err2str(ret));
return ret;
}
ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
video_stream = ret;
video = ifmt_ctx->streams[video_stream];
switch(video->codecpar->codec_id) {
case AV_CODEC_ID_H264:
decoder = avcodec_find_decoder_by_name("h264_qsv");
break;
case AV_CODEC_ID_HEVC:
decoder = avcodec_find_decoder_by_name("hevc_qsv");
break;
case AV_CODEC_ID_VP9:
decoder = avcodec_find_decoder_by_name("vp9_qsv");
break;
case AV_CODEC_ID_VP8:
decoder = avcodec_find_decoder_by_name("vp8_qsv");
break;
case AV_CODEC_ID_AV1:
decoder = avcodec_find_decoder_by_name("av1_qsv");
break;
case AV_CODEC_ID_MPEG2VIDEO:
decoder = avcodec_find_decoder_by_name("mpeg2_qsv");
break;
case AV_CODEC_ID_MJPEG:
decoder = avcodec_find_decoder_by_name("mjpeg_qsv");
break;
default:
fprintf(stderr, "Codec is not supported by qsv\n");
return AVERROR(EINVAL);
}
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
av_err2str(ret));
return ret;
}
decoder_ctx->framerate = av_guess_frame_rate(ifmt_ctx, video, NULL);
decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
if (!decoder_ctx->hw_device_ctx) {
fprintf(stderr, "A hardware device reference create failed.\n");
return AVERROR(ENOMEM);
}
decoder_ctx->get_format = get_format;
decoder_ctx->pkt_timebase = video->time_base;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
av_err2str(ret));
return ret;
}
static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
{
int ret = 0;
av_packet_unref(enc_pkt);
if((ret = dynamic_set_parameter(encoder_ctx)) < 0) {
fprintf(stderr, "Failed to set dynamic parameter. Error code: %s\n",
av_err2str(ret));
goto end;
}
if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
if (ret = avcodec_receive_packet(encoder_ctx, enc_pkt))
break;
enc_pkt->stream_index = 0;
av_packet_rescale_ts(enc_pkt, encoder_ctx->time_base,
ofmt_ctx->streams[0]->time_base);
if ((ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt)) < 0) {
fprintf(stderr, "Error during writing data to output file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
}
end:
if (ret == AVERROR_EOF)
return 0;
ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
return ret;
}
static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec, char *optstr)
{
AVFrame *frame;
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
return ret;
}
while (ret >= 0) {
if (!(frame = av_frame_alloc()))
return AVERROR(ENOMEM);
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
goto fail;
}
if (!encoder_ctx->hw_frames_ctx) {
AVDictionaryEntry *e = NULL;
AVDictionary *opts = NULL;
AVStream *ost;
/* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
Only after we get a decoded frame, can we obtain its hw_frames_ctx */
encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
if (!encoder_ctx->hw_frames_ctx) {
ret = AVERROR(ENOMEM);
goto fail;
}
/* set AVCodecContext Parameters for encoder, here we keep them stay
* the same as decoder.
*/
encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
encoder_ctx->pix_fmt = AV_PIX_FMT_QSV;
encoder_ctx->width = decoder_ctx->width;
encoder_ctx->height = decoder_ctx->height;
if ((ret = str_to_dict(optstr, &opts)) < 0) {
fprintf(stderr, "Failed to set encoding parameter.\n");
goto fail;
}
/* There is no "framerate" option in common option list. Use "-r" to
* set framerate, which is compatible with ffmpeg commandline. The
* video is assumed to be average frame rate, so set time_base to
* 1/framerate. */
e = av_dict_get(opts, "r", NULL, 0);
if (e) {
encoder_ctx->framerate = av_d2q(atof(e->value), INT_MAX);
encoder_ctx->time_base = av_inv_q(encoder_ctx->framerate);
}
if ((ret = avcodec_open2(encoder_ctx, enc_codec, &opts)) < 0) {
fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
av_err2str(ret));
av_dict_free(&opts);
goto fail;
}
av_dict_free(&opts);
if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
fprintf(stderr, "Failed to allocate stream for output format.\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ost->time_base = encoder_ctx->time_base;
ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
if (ret < 0) {
fprintf(stderr, "Failed to copy the stream parameters. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
/* write the stream header */
if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
fprintf(stderr, "Error while writing stream header. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
}
frame->pts = av_rescale_q(frame->pts, decoder_ctx->pkt_timebase,
encoder_ctx->time_base);
if ((ret = encode_write(pkt, frame)) < 0)
fprintf(stderr, "Error during encoding and writing.\n");
fail:
av_frame_free(&frame);
}
return ret;
}
int main(int argc, char **argv)
{
const AVCodec *enc_codec;
int ret = 0;
AVPacket *dec_pkt = NULL;
if (argc < 5 || (argc - 5) % 2) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <encoder> <output file>"
" <\"encoding option set 0\"> [<frame_number> <\"encoding options set 1\">]...\n", argv[0]);
return 1;
}
setting_number = (argc - 5) / 2;
dynamic_setting = av_malloc(setting_number * sizeof(*dynamic_setting));
if (!dynamic_setting) {
ret = AVERROR(ENOMEM);
goto end;
}
current_setting_number = 0;
for (int i = 0; i < setting_number; i++) {
dynamic_setting[i].frame_number = atoi(argv[i*2 + 5]);
dynamic_setting[i].optstr = argv[i*2 + 6];
}
ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_QSV, NULL, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Failed to create a QSV device. Error code: %s\n", av_err2str(ret));
goto end;
}
dec_pkt = av_packet_alloc();
if (!dec_pkt) {
fprintf(stderr, "Failed to allocate decode packet\n");
goto end;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
ret = -1;
goto end;
}
if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
"%s\n", av_err2str(ret));
goto end;
}
if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
ret = AVERROR(ENOMEM);
goto end;
}
ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Cannot open output file. "
"Error code: %s\n", av_err2str(ret));
goto end;
}
/* read all packets and only transcoding video */
while (ret >= 0) {
if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 0)
break;
if (video_stream == dec_pkt->stream_index)
ret = dec_enc(dec_pkt, enc_codec, argv[4]);
av_packet_unref(dec_pkt);
}
/* flush decoder */
av_packet_unref(dec_pkt);
if ((ret = dec_enc(dec_pkt, enc_codec, argv[4])) < 0) {
fprintf(stderr, "Failed to flush decoder %s\n", av_err2str(ret));
goto end;
}
/* flush encoder */
if ((ret = encode_write(dec_pkt, NULL)) < 0) {
fprintf(stderr, "Failed to flush encoder %s\n", av_err2str(ret));
goto end;
}
/* write the trailer for output stream */
if ((ret = av_write_trailer(ofmt_ctx)) < 0)
fprintf(stderr, "Failed to write trailer %s\n", av_err2str(ret));
end:
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
avcodec_free_context(&decoder_ctx);
avcodec_free_context(&encoder_ctx);
av_buffer_unref(&hw_device_ctx);
av_packet_free(&dec_pkt);
av_freep(&dynamic_setting);
return ret;
}
+244
View File
@@ -0,0 +1,244 @@
/*
* Copyright (c) 2015 Anton Khirnov
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel QSV-accelerated H.264 decoding example.
*
* @example qsvdec.c
* This example shows how to do QSV-accelerated H.264 decoding with output
* frames in the GPU video surfaces.
*/
#include "config.h"
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/buffer.h"
#include "libavutil/error.h"
#include "libavutil/hwcontext.h"
#include "libavutil/hwcontext_qsv.h"
#include "libavutil/mem.h"
static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
{
while (*pix_fmts != AV_PIX_FMT_NONE) {
if (*pix_fmts == AV_PIX_FMT_QSV) {
return AV_PIX_FMT_QSV;
}
pix_fmts++;
}
fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
return AV_PIX_FMT_NONE;
}
static int decode_packet(AVCodecContext *decoder_ctx,
AVFrame *frame, AVFrame *sw_frame,
AVPacket *pkt, AVIOContext *output_ctx)
{
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
while (ret >= 0) {
int i, j;
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
/* A real program would do something useful with the decoded frame here.
* We just retrieve the raw data and write it to a file, which is rather
* useless but pedagogic. */
ret = av_hwframe_transfer_data(sw_frame, frame, 0);
if (ret < 0) {
fprintf(stderr, "Error transferring the data to system memory\n");
goto fail;
}
for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++)
for (j = 0; j < (sw_frame->height >> (i > 0)); j++)
avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], sw_frame->width);
fail:
av_frame_unref(sw_frame);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
AVFormatContext *input_ctx = NULL;
AVStream *video_st = NULL;
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder;
AVPacket *pkt = NULL;
AVFrame *frame = NULL, *sw_frame = NULL;
AVIOContext *output_ctx = NULL;
int ret, i;
AVBufferRef *device_ref = NULL;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
/* open the input file */
ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
goto finish;
}
/* find the first H.264 video stream */
for (i = 0; i < input_ctx->nb_streams; i++) {
AVStream *st = input_ctx->streams[i];
if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
video_st = st;
else
st->discard = AVDISCARD_ALL;
}
if (!video_st) {
fprintf(stderr, "No H.264 video stream in the input file\n");
goto finish;
}
/* open the hardware device */
ret = av_hwdevice_ctx_create(&device_ref, AV_HWDEVICE_TYPE_QSV,
"auto", NULL, 0);
if (ret < 0) {
fprintf(stderr, "Cannot open the hardware device\n");
goto finish;
}
/* initialize the decoder */
decoder = avcodec_find_decoder_by_name("h264_qsv");
if (!decoder) {
fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
goto finish;
}
decoder_ctx = avcodec_alloc_context3(decoder);
if (!decoder_ctx) {
ret = AVERROR(ENOMEM);
goto finish;
}
decoder_ctx->codec_id = AV_CODEC_ID_H264;
if (video_st->codecpar->extradata_size) {
decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!decoder_ctx->extradata) {
ret = AVERROR(ENOMEM);
goto finish;
}
memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
video_st->codecpar->extradata_size);
decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
}
decoder_ctx->hw_device_ctx = av_buffer_ref(device_ref);
decoder_ctx->get_format = get_format;
ret = avcodec_open2(decoder_ctx, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Error opening the decoder: ");
goto finish;
}
/* open the output stream */
ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Error opening the output context: ");
goto finish;
}
frame = av_frame_alloc();
sw_frame = av_frame_alloc();
pkt = av_packet_alloc();
if (!frame || !sw_frame || !pkt) {
ret = AVERROR(ENOMEM);
goto finish;
}
/* actual decoding */
while (ret >= 0) {
ret = av_read_frame(input_ctx, pkt);
if (ret < 0)
break;
if (pkt->stream_index == video_st->index)
ret = decode_packet(decoder_ctx, frame, sw_frame, pkt, output_ctx);
av_packet_unref(pkt);
}
/* flush the decoder */
ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
finish:
if (ret < 0) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "%s\n", buf);
}
avformat_close_input(&input_ctx);
av_frame_free(&frame);
av_frame_free(&sw_frame);
av_packet_free(&pkt);
avcodec_free_context(&decoder_ctx);
av_buffer_unref(&device_ref);
avio_close(output_ctx);
return ret;
}
-199
View File
@@ -1,199 +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 usage example
* @example remux.c
*
* Remux streams from one container format to another. Data is copied from the
* input to the output without transcoding.
*/
#include <libavutil/mem.h>
#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)
{
const AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket *pkt = NULL;
const char *in_filename, *out_filename;
int ret, i;
int stream_index = 0;
int *stream_mapping = NULL;
int stream_mapping_size = 0;
if (argc < 3) {
printf("usage: %s input output\n"
"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];
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
return 1;
}
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
}
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;
}
stream_mapping_size = ifmt_ctx->nb_streams;
stream_mapping = av_calloc(stream_mapping_size, sizeof(*stream_mapping));
if (!stream_mapping) {
ret = AVERROR(ENOMEM);
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *out_stream;
AVStream *in_stream = ifmt_ctx->streams[i];
AVCodecParameters *in_codecpar = in_stream->codecpar;
if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO &&
in_codecpar->codec_type != AVMEDIA_TYPE_SUBTITLE) {
stream_mapping[i] = -1;
continue;
}
stream_mapping[i] = stream_index++;
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters\n");
goto end;
}
out_stream->codecpar->codec_tag = 0;
}
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];
if (pkt->stream_index >= stream_mapping_size ||
stream_mapping[pkt->stream_index] < 0) {
av_packet_unref(pkt);
continue;
}
pkt->stream_index = stream_mapping[pkt->stream_index];
out_stream = ofmt_ctx->streams[pkt->stream_index];
log_packet(ifmt_ctx, pkt, "in");
/* copy packet */
av_packet_rescale_ts(pkt, 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);
/* pkt is now blank (av_interleaved_write_frame() takes ownership of
* its contents and resets pkt), so that no unreferencing is necessary.
* This would be different if one used av_write_frame(). */
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
}
av_write_trailer(ofmt_ctx);
end:
av_packet_free(&pkt);
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
av_freep(&stream_mapping);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}
+198
View File
@@ -0,0 +1,198 @@
/*
* 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)
{
const AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket *pkt = NULL;
const char *in_filename, *out_filename;
int ret, i;
int stream_index = 0;
int *stream_mapping = NULL;
int stream_mapping_size = 0;
if (argc < 3) {
printf("usage: %s input output\n"
"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];
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
return 1;
}
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
}
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;
}
stream_mapping_size = ifmt_ctx->nb_streams;
stream_mapping = av_calloc(stream_mapping_size, sizeof(*stream_mapping));
if (!stream_mapping) {
ret = AVERROR(ENOMEM);
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *out_stream;
AVStream *in_stream = ifmt_ctx->streams[i];
AVCodecParameters *in_codecpar = in_stream->codecpar;
if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO &&
in_codecpar->codec_type != AVMEDIA_TYPE_SUBTITLE) {
stream_mapping[i] = -1;
continue;
}
stream_mapping[i] = stream_index++;
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters\n");
goto end;
}
out_stream->codecpar->codec_tag = 0;
}
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];
if (pkt->stream_index >= stream_mapping_size ||
stream_mapping[pkt->stream_index] < 0) {
av_packet_unref(pkt);
continue;
}
pkt->stream_index = stream_mapping[pkt->stream_index];
out_stream = ofmt_ctx->streams[pkt->stream_index];
log_packet(ifmt_ctx, pkt, "in");
/* copy packet */
av_packet_rescale_ts(pkt, 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);
/* pkt is now blank (av_interleaved_write_frame() takes ownership of
* its contents and resets pkt), so that no unreferencing is necessary.
* This would be different if one used av_write_frame(). */
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
}
av_write_trailer(ofmt_ctx);
end:
av_packet_free(&pkt);
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
av_freep(&stream_mapping);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}
-220
View File
@@ -1,220 +0,0 @@
/*
* Copyright (c) 2012 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 audio resampling API usage example
* @example resample_audio.c
*
* Generate a synthetic audio signal, and Use libswresample API to perform audio
* resampling. The output is written to a raw audio file to be played with
* ffplay.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* 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)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
char buf[64];
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_chlayout(swr_ctx, "in_chlayout", &src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = src_ch_layout.nb_channels;
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* 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 = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = dst_ch_layout.nb_channels;
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
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]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
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);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
fmt, buf, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}
+214
View File
@@ -0,0 +1,214 @@
/*
* Copyright (c) 2012 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.
*/
/**
* @example resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* 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)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* 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 = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
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]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
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);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}
-141
View File
@@ -1,141 +0,0 @@
/*
* Copyright (c) 2012 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 libswscale API usage example
* @example scale_video.c
*
* Generate a synthetic video signal and use libswscale to perform rescaling.
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignment */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}
+140
View File
@@ -0,0 +1,140 @@
/*
* Copyright (c) 2012 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
* libswscale API use example.
* @example scaling_video.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignment */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}
-61
View File
@@ -1,61 +0,0 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* 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 metadata extraction API usage example
* @example show_metadata.c
*
* Show metadata from an input file.
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
const AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
while ((tag = av_dict_iterate(fmt_ctx->metadata, tag)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
return 0;
}
-691
View File
@@ -1,691 +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 demuxing, decoding, filtering, encoding and muxing API usage example
* @example transcode.c
*
* Convert input to output file, applying some hard-coded filter-graph on both
* audio and video streams.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/channel_layout.h>
#include <libavutil/mem.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;
AVPacket *enc_pkt;
AVFrame *filtered_frame;
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
AVFrame *dec_frame;
} StreamContext;
static StreamContext *stream_ctx;
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;
}
stream_ctx = av_calloc(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
if (!stream_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream = ifmt_ctx->streams[i];
const AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
AVCodecContext *codec_ctx;
if (!dec) {
av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
return AVERROR_DECODER_NOT_FOUND;
}
codec_ctx = avcodec_alloc_context3(dec);
if (!codec_ctx) {
av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
return AVERROR(ENOMEM);
}
ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
"for stream #%u\n", i);
return ret;
}
/* Inform the decoder about the timebase for the packet timestamps.
* This is highly recommended, but not mandatory. */
codec_ctx->pkt_timebase = stream->time_base;
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
/* Open decoder */
ret = avcodec_open2(codec_ctx, dec, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
stream_ctx[i].dec_ctx = codec_ctx;
stream_ctx[i].dec_frame = av_frame_alloc();
if (!stream_ctx[i].dec_frame)
return AVERROR(ENOMEM);
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
const 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 = stream_ctx[i].dec_ctx;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
enc_ctx = avcodec_alloc_context3(encoder);
if (!enc_ctx) {
av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
return AVERROR(ENOMEM);
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
const enum AVPixelFormat *pix_fmts = NULL;
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
ret = avcodec_get_supported_config(enc_ctx, NULL,
AV_CODEC_CONFIG_PIX_FORMAT, 0,
(const void**)&pix_fmts, NULL);
/* take first format from list of supported formats */
enc_ctx->pix_fmt = (ret >= 0 && pix_fmts) ?
pix_fmts[0] : dec_ctx->pix_fmt;
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
} else {
const enum AVSampleFormat *sample_fmts = NULL;
enc_ctx->sample_rate = dec_ctx->sample_rate;
ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
if (ret < 0)
return ret;
ret = avcodec_get_supported_config(enc_ctx, NULL,
AV_CODEC_CONFIG_SAMPLE_FORMAT, 0,
(const void**)&sample_fmts, NULL);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = (ret >= 0 && sample_fmts) ?
sample_fmts[0] : dec_ctx->sample_fmt;
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open %s encoder for stream #%u\n", encoder->name, i);
return ret;
}
ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
return ret;
}
out_stream->time_base = enc_ctx->time_base;
stream_ctx[i].enc_ctx = enc_ctx;
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
return ret;
}
out_stream->time_base = in_stream->time_base;
}
}
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;
const AVFilter *buffersrc = NULL;
const 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->pkt_timebase.num, dec_ctx->pkt_timebase.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;
}
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, buffersink, "out");
if (!buffersink_ctx) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
ret = AVERROR(ENOMEM);
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;
}
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize buffer sink\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
char buf[64];
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
av_channel_layout_describe(&dec_ctx->ch_layout, buf, sizeof(buf));
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
dec_ctx->pkt_timebase.num, dec_ctx->pkt_timebase.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
buf);
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;
}
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, buffersink, "out");
if (!buffersink_ctx) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
ret = AVERROR(ENOMEM);
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;
}
av_channel_layout_describe(&enc_ctx->ch_layout, buf, sizeof(buf));
ret = av_opt_set(buffersink_ctx, "ch_layouts",
buf, 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;
}
if (enc_ctx->frame_size > 0)
av_buffersink_set_frame_size(buffersink_ctx, enc_ctx->frame_size);
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize audio buffer sink\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]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
stream_ctx[i].enc_ctx, filter_spec);
if (ret)
return ret;
filter_ctx[i].enc_pkt = av_packet_alloc();
if (!filter_ctx[i].enc_pkt)
return AVERROR(ENOMEM);
filter_ctx[i].filtered_frame = av_frame_alloc();
if (!filter_ctx[i].filtered_frame)
return AVERROR(ENOMEM);
}
return 0;
}
static int encode_write_frame(unsigned int stream_index, int flush)
{
StreamContext *stream = &stream_ctx[stream_index];
FilteringContext *filter = &filter_ctx[stream_index];
AVFrame *filt_frame = flush ? NULL : filter->filtered_frame;
AVPacket *enc_pkt = filter->enc_pkt;
int ret;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
av_packet_unref(enc_pkt);
if (filt_frame && filt_frame->pts != AV_NOPTS_VALUE)
filt_frame->pts = av_rescale_q(filt_frame->pts, filt_frame->time_base,
stream->enc_ctx->time_base);
ret = avcodec_send_frame(stream->enc_ctx, filt_frame);
if (ret < 0)
return ret;
while (ret >= 0) {
ret = avcodec_receive_packet(stream->enc_ctx, enc_pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return 0;
/* prepare packet for muxing */
enc_pkt->stream_index = stream_index;
av_packet_rescale_ts(enc_pkt,
stream->enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt);
}
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
FilteringContext *filter = &filter_ctx[stream_index];
int ret;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter->buffersrc_ctx,
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) {
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter->buffersink_ctx,
filter->filtered_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;
break;
}
filter->filtered_frame->time_base = av_buffersink_get_time_base(filter->buffersink_ctx);;
filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(stream_index, 0);
av_frame_unref(filter->filtered_frame);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
return encode_write_frame(stream_index, 1);
}
int main(int argc, char **argv)
{
int ret;
AVPacket *packet = NULL;
unsigned int stream_index;
unsigned int i;
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
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;
if (!(packet = av_packet_alloc()))
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, packet)) < 0)
break;
stream_index = packet->stream_index;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
StreamContext *stream = &stream_ctx[stream_index];
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
ret = avcodec_send_packet(stream->dec_ctx, packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
break;
else if (ret < 0)
goto end;
stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
ret = filter_encode_write_frame(stream->dec_frame, stream_index);
if (ret < 0)
goto end;
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, packet);
if (ret < 0)
goto end;
}
av_packet_unref(packet);
}
/* flush decoders, filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
StreamContext *stream;
if (!filter_ctx[i].filter_graph)
continue;
stream = &stream_ctx[i];
av_log(NULL, AV_LOG_INFO, "Flushing stream %u decoder\n", i);
/* flush decoder */
ret = avcodec_send_packet(stream->dec_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing decoding failed\n");
goto end;
}
while (ret >= 0) {
ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
if (ret == AVERROR_EOF)
break;
else if (ret < 0)
goto end;
stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
ret = filter_encode_write_frame(stream->dec_frame, i);
if (ret < 0)
goto end;
}
/* flush filter */
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_packet_free(&packet);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_free_context(&stream_ctx[i].dec_ctx);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
avcodec_free_context(&stream_ctx[i].enc_ctx);
if (filter_ctx && filter_ctx[i].filter_graph) {
avfilter_graph_free(&filter_ctx[i].filter_graph);
av_packet_free(&filter_ctx[i].enc_pkt);
av_frame_free(&filter_ctx[i].filtered_frame);
}
av_frame_free(&stream_ctx[i].dec_frame);
}
av_free(filter_ctx);
av_free(stream_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}
+60 -54
View File
@@ -1,5 +1,5 @@
/*
* Copyright (c) 2013-2022 Andreas Unterweger
* Copyright (c) 2013-2018 Andreas Unterweger
*
* This file is part of FFmpeg.
*
@@ -19,30 +19,30 @@
*/
/**
* @file audio transcoding to MPEG/AAC API usage example
* @example transcode_aac.c
* @file
* Simple audio converter
*
* Convert an input audio file to AAC in an MP4 container. Formats other than
* MP4 are supported based on the output file extension.
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
#include <stdio.h>
#include <libavutil/mem.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include <libavcodec/avcodec.h>
#include "libavcodec/avcodec.h"
#include <libavutil/audio_fifo.h>
#include <libavutil/avassert.h>
#include <libavutil/avstring.h>
#include <libavutil/channel_layout.h>
#include <libavutil/frame.h>
#include <libavutil/opt.h>
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include <libswresample/swresample.h>
#include "libswresample/swresample.h"
/* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
@@ -62,7 +62,6 @@ static int open_input_file(const char *filename,
{
AVCodecContext *avctx;
const AVCodec *input_codec;
const AVStream *stream;
int error;
/* Open the input file to read from it. */
@@ -90,10 +89,8 @@ static int open_input_file(const char *filename,
return AVERROR_EXIT;
}
stream = (*input_format_context)->streams[0];
/* Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
@@ -108,7 +105,7 @@ static int open_input_file(const char *filename,
}
/* Initialize the stream parameters with demuxer information. */
error = avcodec_parameters_to_context(avctx, stream->codecpar);
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
avcodec_free_context(&avctx);
@@ -124,9 +121,6 @@ static int open_input_file(const char *filename,
return error;
}
/* Set the packet timebase for the decoder. */
avctx->pkt_timebase = stream->time_base;
/* Save the decoder context for easier access later. */
*input_codec_context = avctx;
@@ -152,7 +146,6 @@ static int open_output_file(const char *filename,
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
const AVCodec *output_codec = NULL;
const void *sample_fmts;
int error;
/* Open the output file to write to it. */
@@ -207,15 +200,15 @@ static int open_output_file(const char *filename,
/* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion. */
av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
error = avcodec_get_supported_config(avctx, NULL, AV_CODEC_CONFIG_SAMPLE_FORMAT,
0, &sample_fmts, NULL);
av_assert0(error >= 0);
avctx->sample_fmt = sample_fmts ? *(const enum AVSampleFormat*)sample_fmts
: input_codec_context->sample_fmt;
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
/* Allow the use of the experimental AAC encoder. */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
@@ -297,18 +290,21 @@ static int init_resampler(AVCodecContext *input_codec_context,
/*
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
error = swr_alloc_set_opts2(resample_context,
&output_codec_context->ch_layout,
*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,
&input_codec_context->ch_layout,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (error < 0) {
if (!*resample_context) {
fprintf(stderr, "Could not allocate resample context\n");
return error;
return AVERROR(ENOMEM);
}
/*
* Perform a sanity check so that the number of converted samples is
@@ -336,7 +332,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
/* Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->ch_layout.nb_channels, 1))) {
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
@@ -385,8 +381,6 @@ static int decode_audio_frame(AVFrame *frame,
if (error < 0)
return error;
*data_present = 0;
*finished = 0;
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
/* If we are at the end of the file, flush the decoder below. */
@@ -453,17 +447,26 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
int error;
/* Allocate as many pointers as there are audio channels.
* Each pointer will point to the audio samples of the corresponding
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
* Allocate memory for the samples of all channels in one consecutive
*/
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_array_and_samples(converted_input_samples, NULL,
output_codec_context->ch_layout.nb_channels,
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
av_err2str(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
@@ -556,7 +559,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFrame *input_frame = NULL;
/* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int data_present = 0;
int ret = AVERROR_EXIT;
/* Initialize temporary storage for one input frame. */
@@ -595,9 +598,10 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
ret = 0;
cleanup:
if (converted_input_samples)
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
av_freep(&converted_input_samples);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
@@ -629,7 +633,7 @@ static int init_output_frame(AVFrame **frame,
* Default channel layouts based on the number of channels
* are assumed for simplicity. */
(*frame)->nb_samples = frame_size;
av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
@@ -676,16 +680,17 @@ static int encode_audio_frame(AVFrame *frame,
pts += frame->nb_samples;
}
*data_present = 0;
/* Send the audio frame stored in the temporary packet to the encoder.
* The output audio stream encoder is used to do this. */
error = avcodec_send_frame(output_codec_context, frame);
/* Check for errors, but proceed with fetching encoded samples if the
* encoder signals that it has nothing more to encode. */
if (error < 0 && error != AVERROR_EOF) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
goto cleanup;
/* The encoder signals that it has nothing more to encode. */
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
goto cleanup;
}
/* Receive one encoded frame from the encoder. */
@@ -856,6 +861,7 @@ int main(int argc, char **argv)
int data_written;
/* Flush the encoder as it may have delayed frames. */
do {
data_written = 0;
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
+622
View File
@@ -0,0 +1,622 @@
/*
* 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/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/channel_layout.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;
AVPacket *enc_pkt;
AVFrame *filtered_frame;
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
AVFrame *dec_frame;
} StreamContext;
static StreamContext *stream_ctx;
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;
}
stream_ctx = av_calloc(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
if (!stream_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream = ifmt_ctx->streams[i];
const AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
AVCodecContext *codec_ctx;
if (!dec) {
av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
return AVERROR_DECODER_NOT_FOUND;
}
codec_ctx = avcodec_alloc_context3(dec);
if (!codec_ctx) {
av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
return AVERROR(ENOMEM);
}
ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
"for stream #%u\n", i);
return ret;
}
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
/* Open decoder */
ret = avcodec_open2(codec_ctx, dec, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
stream_ctx[i].dec_ctx = codec_ctx;
stream_ctx[i].dec_frame = av_frame_alloc();
if (!stream_ctx[i].dec_frame)
return AVERROR(ENOMEM);
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
const 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 = stream_ctx[i].dec_ctx;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
enc_ctx = avcodec_alloc_context3(encoder);
if (!enc_ctx) {
av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
return AVERROR(ENOMEM);
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
if (encoder->pix_fmts)
enc_ctx->pix_fmt = encoder->pix_fmts[0];
else
enc_ctx->pix_fmt = dec_ctx->pix_fmt;
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
} 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};
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
return ret;
}
out_stream->time_base = enc_ctx->time_base;
stream_ctx[i].enc_ctx = enc_ctx;
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
return ret;
}
out_stream->time_base = in_stream->time_base;
}
}
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;
const AVFilter *buffersrc = NULL;
const 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]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
stream_ctx[i].enc_ctx, filter_spec);
if (ret)
return ret;
filter_ctx[i].enc_pkt = av_packet_alloc();
if (!filter_ctx[i].enc_pkt)
return AVERROR(ENOMEM);
filter_ctx[i].filtered_frame = av_frame_alloc();
if (!filter_ctx[i].filtered_frame)
return AVERROR(ENOMEM);
}
return 0;
}
static int encode_write_frame(unsigned int stream_index, int flush)
{
StreamContext *stream = &stream_ctx[stream_index];
FilteringContext *filter = &filter_ctx[stream_index];
AVFrame *filt_frame = flush ? NULL : filter->filtered_frame;
AVPacket *enc_pkt = filter->enc_pkt;
int ret;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
av_packet_unref(enc_pkt);
ret = avcodec_send_frame(stream->enc_ctx, filt_frame);
if (ret < 0)
return ret;
while (ret >= 0) {
ret = avcodec_receive_packet(stream->enc_ctx, enc_pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return 0;
/* prepare packet for muxing */
enc_pkt->stream_index = stream_index;
av_packet_rescale_ts(enc_pkt,
stream->enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt);
}
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
FilteringContext *filter = &filter_ctx[stream_index];
int ret;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter->buffersrc_ctx,
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) {
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter->buffersink_ctx,
filter->filtered_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;
break;
}
filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(stream_index, 0);
av_frame_unref(filter->filtered_frame);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
return encode_write_frame(stream_index, 1);
}
int main(int argc, char **argv)
{
int ret;
AVPacket *packet = NULL;
unsigned int stream_index;
unsigned int i;
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
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;
if (!(packet = av_packet_alloc()))
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, packet)) < 0)
break;
stream_index = packet->stream_index;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
StreamContext *stream = &stream_ctx[stream_index];
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
av_packet_rescale_ts(packet,
ifmt_ctx->streams[stream_index]->time_base,
stream->dec_ctx->time_base);
ret = avcodec_send_packet(stream->dec_ctx, packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
break;
else if (ret < 0)
goto end;
stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
ret = filter_encode_write_frame(stream->dec_frame, stream_index);
if (ret < 0)
goto end;
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, packet);
if (ret < 0)
goto end;
}
av_packet_unref(packet);
}
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_packet_free(&packet);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_free_context(&stream_ctx[i].dec_ctx);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
avcodec_free_context(&stream_ctx[i].enc_ctx);
if (filter_ctx && filter_ctx[i].filter_graph) {
avfilter_graph_free(&filter_ctx[i].filter_graph);
av_packet_free(&filter_ctx[i].enc_pkt);
av_frame_free(&filter_ctx[i].filtered_frame);
}
av_frame_free(&stream_ctx[i].dec_frame);
}
av_free(filter_ctx);
av_free(stream_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}
+11 -12
View File
@@ -1,4 +1,6 @@
/*
* Video Acceleration API (video encoding) encode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -19,12 +21,13 @@
*/
/**
* @file Intel VAAPI-accelerated encoding API usage example
* @example vaapi_encode.c
* @file
* Intel VAAPI-accelerated encoding example.
*
* @example vaapi_encode.c
* This example shows how to do VAAPI-accelerated encoding. now only support NV12
* raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
*
* Perform VAAPI-accelerated encoding. Read input from an NV12 raw
* file, and write the H.264 encoded data to an output raw file.
* Usage: vaapi_encode 1920 1080 input.yuv output.h264
*/
#include <stdio.h>
@@ -88,14 +91,11 @@ static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
enc_pkt->stream_index = 0;
ret = fwrite(enc_pkt->data, enc_pkt->size, 1, fout);
av_packet_unref(enc_pkt);
if (!ret) {
ret = AVERROR(errno);
break;
}
}
end:
av_packet_free(&enc_pkt);
ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
return ret;
}
@@ -117,7 +117,7 @@ int main(int argc, char *argv[])
height = atoi(argv[2]);
size = width * height;
if (!(fin = fopen(argv[3], "rb"))) {
if (!(fin = fopen(argv[3], "r"))) {
fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
return -1;
}
@@ -197,8 +197,7 @@ int main(int argc, char *argv[])
goto close;
}
err = encode_write(avctx, hw_frame, fout);
if (err != AVERROR(EAGAIN) && err < 0) {
if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
fprintf(stderr, "Failed to encode.\n");
goto close;
}

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