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47 Commits

Author SHA1 Message Date
Michael Niedermayer
b88de7b31a Changelog: update for 2.3.2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-11 17:06:48 +02:00
Michael Niedermayer
11420649d0 avcodec/snow: fix null pointer dereference in cleanup after allocation failure
Fixes: snowf.avi
Found-by: Piotr Bandurski <ami_stuff@o2.pl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9a162146ca)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-11 15:45:42 +02:00
Michael Niedermayer
dbf5d7e5cd avcodec/huffyuvdec: fix overread checks
Fixes: ffvhuff_f.avi
Found-by: Piotr Bandurski <ami_stuff@o2.pl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit ba47d519e5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-11 15:32:00 +02:00
Michael Niedermayer
6badd558ce avcodec/get_bits: add BITS_LEFT() for finding the bits left with an opened reader
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 11512d70fa)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-11 15:31:59 +02:00
Michael Niedermayer
a1fe3b4150 avcodec/vc1dec: do not crash when flushing without an allocated frame
Fixes Ticket3837
Found-by: Piotr Bandurski <ami_stuff@o2.pl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6801eb0a09)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-11 02:26:29 +02:00
Michael Niedermayer
c5129da726 update for FFmpeg 2.3.2
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
fc57959fd5 avcodec/iff: Factorize av_pix_fmt_desc_get() out
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 9bcdb98839)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
83956309cc avcodec/iff: check pixfmt for rgb8 / rgbn
Fixes out of array access

Found-by: Piotr Bandurski <ami_stuff@o2.pl>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3539d6c63a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
ed15be7519 avformat/matroskaenc: fix MAX_CUEPOINT_SIZE calculation
Fixes assertion failure
Fixes Ticket3822

as a side-effect this makes some mkv files a few bytes smaller

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 64d029de41)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Felix Abecassis
6928193493 h264: fix interpretation of interleved stereo modes
Column and row frame packing arrangements were inverted.

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
(cherry picked from commit a7e541c992)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Anton Khirnov
a72b7286e6 cdgraphics: do not return 0 from the decode function
0 means no data consumed, so it can trigger an infinite loop in the
caller.

CC:libav-stable@libav.org
(cherry picked from commit c7d9b473e2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
901e275697 avutil/cpu: add aarch64 entries to 2nd table
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit efc4fe9d74)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
7a6b5d7a86 ffserver: initialize pbuffer in prepare_sdp_description()
also check pbuffer before use

Found-by: CSA
Reviewed-by: Stefano Sabatini <stefasab@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1d8d21b90a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
fdfe94f4b1 avcodec/wavpackenc: Fix log2sample() result value
Found-by: CSA
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e706fe7640)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
b63ec0cb0f avformat/tee: flip assigment direction
Found-by: CSA
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2e6fdcb7f3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-10 22:40:50 +02:00
Michael Niedermayer
2eb72d5bdc ffmpeg_opt: Use av_guess_codec() instead of AVOutputFormat->*codec
Fixes part of ticket2236

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 956f4087c6)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-02 03:41:46 +02:00
Michael Niedermayer
638c3aca64 avcodec/dvdsub_parser: print message if packet is smaller than the packet size field
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bcc898dd26)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-02 01:27:04 +02:00
Michael Niedermayer
76d59f1b34 avcodec/dvdsub_parser: Check buf_size before reading 32bit packet size
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 81c1657a59)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-02 01:27:03 +02:00
Michael Niedermayer
aa0a8ef50e avcodec/dvdsub_parser: never return 0 when the input isnt 0
Fixes a infinite loop
Fixes Ticket3804

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cfdb30d2f1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-02 01:27:02 +02:00
Michael Niedermayer
d6173ae341 avcodec/mpeg12dec: fix support for interlaced mpeg2 with missing last slice
Fixes Ticket3809

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit f95298c913)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-01 21:20:43 +02:00
Anshul Maheswhwari
3ed27832e7 v4l2enc: adding AVClass
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit fcb11ec291)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-31 19:06:12 +02:00
Michael Niedermayer
594b1fa961 libavcodec/h264_parser: Increase parse_history, fix huge resolutions
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0782fb6bcb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-31 16:50:32 +02:00
Benoit Fouet
7a1262fca3 h264_mp4toannexb_bsf: account for consecutive IDR pictures.
If there are consecutive IDR pictures, then SPS/PPS should be prepended
to all of them, not only the first one.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit bf428bb314)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-31 16:50:31 +02:00
Chris \"Koying\" Browet
6d2219e9f9 avcodec/h264_mp4toannexb_bsf: fix issue when sps/pps are already in the bistream
(cherry picked from commit ad91bf854b)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-31 16:50:26 +02:00
Michael Niedermayer
7c2d152f56 Changelog: highlighting for 2.3.1
Suggested-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-31 03:17:03 +02:00
Gerion Entrup
1f58590e1e doc: add very basic libcdio documentation
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit dc315cdb4e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-31 03:08:04 +02:00
Michael Niedermayer
64bbbcd7b0 update for FFmpeg 2.3.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-30 23:14:25 +02:00
Michael Niedermayer
de9d3f22f0 avdevice/pulse_audio_enc: use getter function for AVFrame.channels
This is required by the API/ABI for things outside libavutil

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1e51af13c7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-30 22:39:08 +02:00
Michael Niedermayer
ea5bb5613f MAINTAINERS: update list of releases i maintain
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5003b8b9c3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-30 22:39:08 +02:00
Michael Niedermayer
c61ac696e5 avcodec/avdct: Add avcodec_dct_get_class()
This should have been in the initial commit of AVDCT

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2f717be22a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-30 22:39:08 +02:00
Janne Grunau
6a250c858e fate: support testing of release branches
Adding 'branch=release/2.3' to the fate config file will check the
release/2.3 branch instead of master. If no branch is specified it will
use 'master' so that existing config are still valid.

(cherry picked from commit 42eb9154a8)

Conflicts:

	tests/fate.sh
2014-07-30 22:39:08 +02:00
Michael Niedermayer
5411040802 tests/fate.sh: If cat *.rep fails try it with a for loop.
Fixes fate on haiku, where cat dies due to too many arguments
xargs could be used too but we do not use xargs currently so it
would be an additional dependency.

Also the plain cat is left in place as it is faster than the loop

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d554d004a6)

Conflicts:

	tests/fate.sh
2014-07-30 22:39:08 +02:00
Vittorio Giovara
ab1ea597bd g2meet: allow size changes within original sizes
(cherry picked from commit 14b4e64eab)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-30 22:32:01 +02:00
Michael Niedermayer
ee606fd031 version.sh: Print versions based on the last git tag for release branches
release branches are detected by checking if "git" is not in RELEASE
This changes "N-64706-g2f71aeb" to "n2.3-8-g2f71aeb"
for git master theres no change

This should improve the readability of lists of versions which come from
more than 1 release branch or master + release. fate.ffmpeg.org is
one possible example

Reviewed-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-28 14:48:50 +02:00
Michael Niedermayer
2f71aeb301 remove VERSION file
it overrides what version.sh prints and thus makes its output
from release branches rather useless

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-28 02:22:11 +02:00
Christophe Gisquet
65259b4d68 x86: hevc_mc: replace one lea by add
Should have been in 036f11bdb5.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-27 17:32:01 +02:00
Michael Niedermayer
8f53d32dfb avfilter/vf_spp: use AVDCT
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e3fac20824)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-27 17:28:41 +02:00
Michael Niedermayer
fcc6568a10 avcodec: add avdct
This provides a public sustainable API/ABI for DCT functions.
Only externally used dct functions are included.
The structure is extensible without ABI issues compared to the
existing dct contexts.

See Mailing list and IRC log of 2014-07-26/27

Reviewed-by: ubitux
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 932ff70956)

Conflicts:

	doc/APIchanges
2014-07-27 17:28:32 +02:00
Timothy Gu
489d066d49 RELEASE_NOTES: fix WebVTT decoder typo (-> encoder)
Found-by: Clément Bœsch <u@pkh.me>
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 065fa66872)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-25 13:22:26 +02:00
Michael Niedermayer
9cb45f6ad2 doc/APIchanges: mention FF_INPUT_BUFFER_PADDING_SIZE chage
Suggested-by: Andreas Cadhalpun
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 94c4ad27dc)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-22 05:22:25 +02:00
Andreas Cadhalpun
0f04e2741e libavformat/img2dec.c: Add a long_name to the piped image demuxers
This fixes segfaults in gst-libav1.0 compiled against FFmpeg 2.3.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 135ea609ce)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-22 05:21:50 +02:00
Michael Niedermayer
84642ec879 avformat/dv: implement fallback in dv_extract_pack()
Fixes Ticket2340
Fixes Ticket2341

Based-on mail from Dave Rice <dave@dericed.com>
Tested-by: Dave Rice <dave@dericed.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 88f038ac97)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-17 16:00:21 +02:00
Timothy Gu
bef4d9bf87 RELEASE_NOTES: update
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:42:16 +02:00
Michael Niedermayer
bc4f6ae88e Changelog: change "version <next>" to "version 2.3"
Found-by: jamrial
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit faafd1e4f1)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:09:50 +02:00
Michael Niedermayer
2678b25099 doc/APIchanges: update
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 13a72d9b08)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:07:02 +02:00
Michael Niedermayer
e322496054 RELEASE_NOTES: update version numbers
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 02:01:14 +02:00
Michael Niedermayer
7fa72ff19c update for FFmpeg 2.3
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-07-16 01:45:34 +02:00
1824 changed files with 39459 additions and 71639 deletions

1
.gitattributes vendored
View File

@@ -1 +0,0 @@
*.pnm -diff -text

3
.gitignore vendored
View File

@@ -39,7 +39,6 @@
/doc/examples/avio_reading
/doc/examples/decoding_encoding
/doc/examples/demuxing_decoding
/doc/examples/extract_mvs
/doc/examples/filter_audio
/doc/examples/filtering_audio
/doc/examples/filtering_video
@@ -62,9 +61,7 @@
/tests/audiogen
/tests/base64
/tests/data/
/tests/pixfmts.mak
/tests/rotozoom
/tests/test_copy.ffmeta
/tests/tiny_psnr
/tests/tiny_ssim
/tests/videogen

545
Changelog
View File

@@ -1,533 +1,23 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version 2.5.10
- configure: bump copyright year to 2016
- avformat/hls: Even stricter URL checks
- avformat/hls: More strict url checks
- swscale/utils: Detect and skip unneeded sws_setColorspaceDetails() calls
- swscale/yuv2rgb: Increase YUV2RGB table headroom
- swscale/yuv2rgb: Factor YUVRGB_TABLE_LUMA_HEADROOM out
- avformat/hls: forbid all protocols except http(s) & file
- avformat/aviobuf: Fix end check in put_str16()
- avformat/asfenc: Check pts
- avcodec/mpeg4video: Check time_incr
- avcodec/wavpackenc: Check the number of channels
- avcodec/wavpackenc: Headers are per channel
- avcodec/dvdec: Fix "left shift of negative value -254"
- avcodec/mjpegdec: Fix negative shift
- avcodec/mss2: Check for repeat overflow
- avformat: Add integer fps from 31 to 60 to get_std_framerate()
- avcodec/mpegvideo_enc: Clip bits_per_raw_sample within valid range
- avfilter/vf_scale: set proper out frame color range
- avcodec/motion_est: Fix mv_penalty table size
- avcodec/h264_slice: Fix integer overflow in implicit weight computation
- swscale/utils: Use normal bilinear scaler if fast cannot be used due to tiny dimensions
- avcodec/put_bits: Always check buffer end before writing
- mjpegdec: extend check for incompatible values of s->rgb and s->ls
- swscale/utils: Fix intermediate format for cascaded alpha downscaling
- avcodec/h264_refs: Fix long_idx check
- avfilter/vf_mpdecimate: Add missing emms_c()
- avformat/mxfenc: Do not crash if there is no packet in the first stream
- avformat/utils: estimate_timings_from_pts - increase retry counter, fixes invalid duration for ts files with hevc codec
- avformat/matroskaenc: Check codecdelay before use
- avutil/mathematics: Fix division by 0
- x86/float_dsp: zero extend offset from ff_scalarproduct_float_sse
- avcodec/mpeg4videodec: also for empty partitioned slices
- nuv: sanitize negative fps rate
- rawdec: only exempt BIT0 with need_copy from buffer sanity check
- mlvdec: check that index_entries exist
- nutdec: reject negative value_len in read_sm_data
- xwddec: prevent overflow of lsize * avctx->height
- nutdec: only copy the header if it exists
- exr: fix out of bounds read in get_code
- on2avc: limit number of bits to 30 in get_egolomb
- sonic: make sure num_taps * channels is not larger than frame_size
- opus_silk: fix typo causing overflow in silk_stabilize_lsf
- ffm: reject invalid codec_id and codec_type
- aaccoder: prevent crash of anmr coder
- ffmdec: reject zero-sized chunks
- swscale/x86/rgb2rgb_template: Fallback to mmx in interleaveBytes() if the alignment is insufficient for SSE*
- swscale/x86/rgb2rgb_template: Do not crash on misaligend stride
version 2.5.9
- avcodec/hevc: Check max ctb addresses for WPP
- avcodec/vp3: ensure header is parsed successfully before tables
- avcodec/jpeg2000dec: Check bpno in decode_cblk()
- avcodec/pgssubdec: Fix left shift of 255 by 24 places cannot be represented in type int
- swscale/utils: Fix for runtime error: left shift of negative value -1
- avcodec/hevc: Fix integer overflow of entry_point_offset
- avcodec/dirac_parser: Check that there is a previous PU before accessing it
- avcodec/dirac_parser: Add basic validity checks for next_pu_offset and prev_pu_offset
- avcodec/dirac_parser: Fix potential overflows in pointer checks
- avcodec/wmaprodec: Check bits per sample to be within the range not causing integer overflows
- avcodec/wmaprodec: Fix overflow of cutoff
- avformat/smacker: fix integer overflow with pts_inc
- avcodec/vp3: Fix "runtime error: left shift of negative value"
- mpegencts: Fix overflow in cbr mode period calculations
- avutil/timecode: Fix fps check
- avutil/mathematics: return INT64_MIN (=AV_NOPTS_VALUE) from av_rescale_rnd() for overflows
- avcodec/apedec: Check length in long_filter_high_3800()
- avcodec/vp3: always set pix_fmt in theora_decode_header()
- avcodec/mpeg4videodec: Check available data before reading custom matrix
- avutil/mathematics: Do not treat INT64_MIN as positive in av_rescale_rnd
- avutil/integer: Fix av_mod_i() with negative dividend
- avformat/dump: Fix integer overflow in av_dump_format()
- avcodec/utils: Clear dimensions in ff_get_buffer() on failure
- avcodec/utils: Use 64bit for aspect ratio calculation in avcodec_string()
- avcodec/vp3: Clear context on reinitialization failure
- avcodec/hevc: allocate entries unconditionally
- avcodec/hevc_cabac: Fix multiple integer overflows
- avcodec/jpeg2000dwt: Check ndeclevels before calling dwt_encode*()
- avcodec/jpeg2000dwt: Check ndeclevels before calling dwt_decode*()
- avcodec/hevc: Check entry_point_offsets
- avcodec/cabac: Check initial cabac decoder state
- avcodec/cabac_functions: Fix "left shift of negative value -31767"
- avcodec/ffv1dec: Clear quant_table_count if its invalid
- avcodec/ffv1dec: Print an error if the quant table count is invalid
- doc/filters/drawtext: fix centering example
- avcodec/h264_slice: Limit max_contexts when slice_context_count is initialized
- avcodec/vp8: Do not use num_coeff_partitions in thread/buffer setup
- rtmpcrypt: Do the xtea decryption in little endian mode
- avformat/matroskadec: Check subtitle stream before dereferencing
- avformat/utils: Do not init parser if probing is unfinished
- avcodec/jpeg2000dec: Fix potential integer overflow with tile dimensions
- avcodec/jpeg2000dec: Check SIZ dimensions to be within the supported range
- avcodec/jpeg2000: Check comp coords to be within the supported size
- avcodec/jpeg2000: Use av_image_check_size() in ff_jpeg2000_init_component()
- avcodec/wmaprodec: Check for overread in decode_packet()
- avcodec/smacker: Check that the data size is a multiple of a sample vector
- avcodec/takdec: Skip last p2 sample (which is unused)
- avcodec/dxtory: Fix input size check in dxtory_decode_v1_410()
- avcodec/dxtory: Fix input size check in dxtory_decode_v1_420()
- avcodec/error_resilience: avoid accessing previous or next frames tables beyond height
- avcodec/dpx: Move need_align to act per line
- avcodec/flashsv: Check size before updating it
- avcodec/ivi: Check image dimensions
- avcodec/utils: Better check for channels in av_get_audio_frame_duration()
- avcodec/jpeg2000dec: Check for duplicate SIZ marker
- avcodec/jpeg2000dec: Clip all tile coordinates
- avcodec/microdvddec: Check for string end in 'P' case
- avcodec/dirac_parser: Fix undefined memcpy() use
- avformat/xmv: Discard remainder of packet on error
- avformat/xmv: factor return check out of if/else
- libavutil/channel_layout: Check strtol*() for failure
- avcodec/ffv1dec: Check for 0 quant tables
- avcodec/mjpegdec: Reinitialize IDCT on BPP changes
- avcodec/mjpegdec: Check index in ljpeg_decode_yuv_scan() before using it
- avutil/file_open: avoid file handle inheritance on Windows
- opusdec: Don't run vector_fmul_scalar on zero length arrays
- avcodec/ffv1: Initialize vlc_state on allocation
- avcodec/ffv1dec: update progress in case of broken pointer chains
- avcodec/ffv1dec: Clear slice coordinates if they are invalid or slice header decoding fails for other reasons
- avformat/httpauth: Add space after commas in HTTP/RTSP auth header
- avcodec/x86/sbrdsp: Fix using uninitialized upper 32bit of noise
- avcodec/ffv1dec: Fix off by 1 error in quant_table_count check
- avcodec/ffv1dec: Explicitly check read_quant_table() return value
- avcodec/rangecoder: Check e
- lavf/webvttenc: Require webvtt file to contain exactly one WebVTT stream.
- avcodec/mjpegdec: Fix decoding RGBA RCT LJPEG
- avfilter/af_asyncts: use llabs for int64_t
- avcodec/g2meet: Also clear tile dimensions on header_fail
- avcodec/g2meet: Fix potential overflow in tile dimensions check
- avcodec/svq1dec: Check init_get_bits8() for failure
- avcodec/tta: Check init_get_bits8() for failure
- swresample/swresample: Fix integer overflow in seed calculation
- avformat/mov: Fix integer overflow in FFABS
- avutil/common: Add FFNABS()
- avutil/common: Document FFABS() corner case
- avformat/dump: Fix integer overflow in aspect ratio calculation
- avcodec/truemotion1: Check for even width
- avcodec/mpeg12dec: Set dimensions in mpeg1_decode_sequence() only in absence of errors
- avcodec/libopusenc: Fix infinite loop on flushing after 0 input
- avformat/hevc: Check num_long_term_ref_pics_sps to avoid potentially long loops
- avformat/hevc: Fix parsing errors
- ffmpeg: Use correct codec_id for av_parser_change() check
- ffmpeg: Check av_parser_change() for failure
- ffmpeg: Check for RAWVIDEO and do not relay only on AVFMT_RAWPICTURE
- ffmpeg: check avpicture_fill() return value
- avformat/mux: Update sidedata in ff_write_chained()
- avcodec/flashsvenc: Correct max dimension in error message
- avcodec/svq1enc: Check dimensions
- avcodec/dcaenc: clear bitstream end
- libavcodec/aacdec_template: Use init_get_bits8() in aac_decode_frame()
- mxfdec: check edit_rate also for physical_track
- mpegvideo: clear overread in clear_context
- dvdsubdec: validate offset2 similar to offset1
- avcodec/takdec: Use memove, avoid undefined memcpy() use
- jvdec: avoid unsigned overflow in comparison
- avcodec/mpeg12dec: Do not call show_bits() with invalid bits
- riffdec: prevent negative bit rate
- Merge commit 'd80811c94e068085aab797f9ba35790529126f85'
- imc: use correct position for flcoeffs2 calculation
- wavpack: limit extra_bits to 32 and use get_bits_long
- wavpack: use get_bits_long to read up to 32 bits
- nutdec: check maxpos in read_sm_data before returning success
- s302m: fix arithmetic exception
- avcodec/s302m: Only set the sample rate when some data is output
- vp9: add support for resolution changes in inter frames.
- alsdec: limit avctx->bits_per_raw_sample to 32
- vp9: avoid infinite loop with broken files
- videodsp: don't overread edges in vfix3 emu_edge.
- avcodec/h264_mp4toannexb_bsf: Reorder operations in nal_size check
- avformat/oggenc: Check segments_count for headers too
- avformat/avidec: Workaround broken initial frame
- hevc: properly handle no_rasl_output_flag when removing pictures from the DPB
- hevc: fix wpp threading deadlock.
- avcodec/ffv1: separate slice_count from max_slice_count
- lavf/img2dec: Fix memory leak
- avcodec/mp3: fix skipping zeros
- avformat/srtdec: make sure we probe a number
- avformat/srtdec: more lenient first line probing
- doc: mention libavcodec can decode Opus natively
- MAINTAINERS: Remove myself as leader
version 2.5.8
- snow: remove an obsolete av_assert2
- huffyuvdec: validate image size
- vc1dec: use get_bits_long and limit the read bits to 32
- mpegaudiodec: copy AVFloatDSPContext from first context to all contexts
- libshine: fix support for shine 3.0
- avidec: check for valid bit_rate range
- avformat/nut: support WavPack
- avcodec/diracdec: Check slices malloc and propagate error code
- avcodec/vp8: Check buffer size in vp8_decode_frame_header()
- avcodec/vp8: Fix null pointer dereference in ff_vp8_decode_free()
- avcodec/diracdec: Check for hpel_base allocation failure
- avcodec/rv34: Clear pointers in ff_rv34_decode_init_thread_copy()
- avfilter/af_aresample: Check ff_all_* for allocation failures
- avcodec/pthread_frame: clear priv_data, avoid stale pointer in error case
- swscale/utils: Clear pix buffers
- avutil/fifo: Fix the case where func() returns less bytes than requested in av_fifo_generic_write()
- avformat/mov: Fix deallocation when MOVStreamContext failed to allocate
- ffmpeg: Fix crash with ost->last_frame allocation failure
- ffmpeg: Fix cleanup with ost = NULL
- avcodec/pthread_frame: check avctx on deallocation
- avcodec/sanm: Reset sizes in destroy_buffers()
- avcodec/alac: Clear pointers in allocate_buffers()
- bytestream2: set the reader to the end when reading more than available
- avcodec/utils: use a minimum 32pixel width in avcodec_align_dimensions2() for H.264
- avcodec/mpegvideo: Clear pointers in ff_mpv_common_init()
- oggparsedirac: check return value of init_get_bits
- wmalosslessdec: reset frame->nb_samples on packet loss
- wmalosslessdec: avoid reading 0 bits with get_bits
- avcodec/rawenc: Use ff_alloc_packet() instead of ff_alloc_packet2()
- avcodec/aacsbr: Assert that bs_num_env is positive
- avcodec/aacsbr: check that the element type matches before applying SBR
- avcodec/h264_slice: Use w/h from the AVFrame instead of mb_w/h
- vp9/update_prob: prevent out of bounds table read
- avfilter/vf_transpose: Fix rounding error
- avcodec/pngdec: Check values before updating context in decode_fctl_chunk()
- avcodec/pngdec: Require a IHDR chunk before fctl
- avcodec/pngdec: Only allow one IHDR chunk
- wmavoice: limit wmavoice_decode_packet return value to packet size
- swscale/swscale_unscaled: Fix rounding difference with RGBA output between little and big endian
- ffmpeg: Do not use the data/size of a bitstream filter after failure
- swscale/x86/rgb2rgb_template: fix signedness of v in shuffle_bytes_2103_{mmx,mmxext}
- swscale/x86/rgb2rgb_template: add missing xmm clobbers
- vda: unlock the pixel buffer base address.
- swscale/rgb2rgb_template: Fix signedness of v in shuffle_bytes_2103_c()
- swscale/rgb2rgb_template: Implement shuffle_bytes_0321_c and fix shuffle_bytes_2103_c on BE
- swscale/rgb2rgb_template: Disable shuffle_bytes_2103_c on big endian
- swr: Remember previously set int_sample_format from user
- matroskadec: check audio sample rate
- matroskadec: validate audio channels and bitdepth
- avcodec/dpxenc: implement write16/32 as functions
- postproc: fix unaligned access
- ffmpeg: Free last_frame instead of just unref
- avio: fix potential crashes when combining ffio_ensure_seekback + crc
- h264: er: Copy from the previous reference only if compatible
- sonic: set avctx->channels in sonic_decode_init
- vp8: change mv_{min,max}.{x,y} type to int
- vp9: change type of tile_size from unsigned to int64_t
- arm: only enable setend on ARMv6
- libopenjpegdec: check existence of image component data
- mov: abort on EOF in ff_mov_read_chan
- ffmpeg_opt: Check for localtime() failure
- avformat: Fix bug in parse_rps for HEVC.
- takdec: ensure chan2 is a valid channel index
- avcodec/h264_slice: Use AVFrame diemensions for grayscale handling
- avdevice/lavfi: do not rescale AV_NOPTS_VALUE in lavfi_read_packet()
- libavutil/channel_layout: Correctly return layout when channel specification ends with a trailing 'c'.
- avcodec/jpeg2000dec: Check that coords match before applying ICT
- avformat/ffmdec: Check ffio_set_buf_size() return value
- avcodec/adpcm: Check for overreads
- avcodec/alsdec: Check for overread
- avcodec/atrac3plusdec: consume only as many bytes as available
- libavutil/softfloat: Fix av_normalize1_sf bias.
- swresample/swresample: Cleanup on init failure.
- Revert "avformat/rtpenc: check av_packet_get_side_data() return, fix null ptr dereference"
- avformat/mxfenc: Accept MXF D-10 with 49.999840 Mbit/sec
- swresample/dither: check memory allocation
- libopenjpegenc: add NULL check for img before accessing it
- swresample: Check the return value of resampler->init()
- h264: Make sure reinit failures mark the context as not initialized
- ffmpeg_opt: Set the video VBV parameters only for the video stream from -target
- avcodec/bitstream: Assert that there is enough space left in avpriv_copy_bits()
- avcodec/put_bits: Assert that there is enough space left in skip_put_bytes()
- avcodec/mpegvideo_enc: Update the buffer size as more slices are merged
- avcodec/put_bits: Update size_in_bits in set_put_bits_buffer_size()
- avformat/wavdec: Increase dts packet threshold to fix more misdetections
- avformat/wavdec: Increase probe_packets limit
- avformat/swfdec: Do not error out on pixel format changes
- avfilter/x86/vf_hqdn3d: Fix register types
- avcodec/mjpegenc_common: Use ff_mpv_reallocate_putbitbuffer()
- avcodec/mpegvideo: Factor ff_mpv_reallocate_putbitbuffer() out
- avformat/mov: Mark avio context of decompressed atoms as seekable
- avcodec/hevc_ps: Only discard overread VPS if a previous is available
- avcodec/x86/h264_weight: handle weight1=128
- avcodec/exr: fix crash caused by merge
version 2.5.7
- avformat/nutdec: Fix recovery when immedeately after seeking a failure happens
- nutdec: fix memleaks on error in nut_read_header
- rtpenc_jpeg: handle case of picture dimensions not dividing by 8
- avformat/mov: Fix parsing short loci
- avcodec/shorten: Fix code depending on signed overflow behavior
- avcodec/proresdec2: Reset slice_count on deallocation
- ffmpeg_opt: Fix -timestamp parsing
- hevc: make avcodec_decode_video2() fail if get_format() fails
- avcodec/mpeg4audio: add some padding/alignment to MAX_PCE_SIZE
- swr: fix alignment issue caused by 8ch sse functions
- libswscale/x86/hscale_fast_bilinear_simd.c: Include BX in the clobber list on x86_64, because it isn't implicitly included when PIC is on.
- aacdec: don't return frames without data
- avformat/matroskadec: Cleanup error handling for bz2 & zlib
- avformat/nutdec: Fix use of uinitialized value
- tools/graph2dot: use larger data types than int for array/string sizes
- id3v2: catch avio_read errors in check_tag
- aacsbr: break infinite loop in sbr_hf_calc_npatches
- diracdec: avoid overflow of bytes*8 in decode_lowdelay
- diracdec: prevent overflow in data_unit_size check
- avidec: avoid infinite loop due to negative ast->sample_size
- pngdec: don't use AV_PIX_FMT_MONOBLACK for apng
- avcodec/wavpack: Check L/R values before use to avoid harmless integer overflow and undefined behavior in fate
- xcbgrab: Validate the capture area
- xcbgrab: Do not assume the non shm image data is always available
- avfilter/lavfutils: disable frame threads when decoding a single image
- nutdec: fix illegal count check in decode_main_header
- ffmpeg: remove incorrect network deinit
- OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c
- apedec: set s->samples only when init_frame_decoder succeeded
- swscale/ppc/swscale_altivec.c: POWER LE support in yuv2planeX_8() delete macro GET_VF()
- libvpxenc: only set noise reduction w/vp8
- tests/fate-run: do not attempt to parse tiny_psnrs output if it failed
- alac: reject rice_limit 0 if compression is used
- alsdec: only adapt order for positive max_order
- alsdec: check sample pointer range in revert_channel_correlation
- tests: drop bc dependency
- fate: Include branch information in the payload header
version 2.5.6
- avcodec/atrac3plusdsp: fix on stack alignment
- ac3: validate end in ff_ac3_bit_alloc_calc_mask
- aacpsy: avoid psy_band->threshold becoming NaN
- aasc: return correct buffer size from aasc_decode_frame
- msrledec: use signed pixel_ptr in msrle_decode_pal4
- swresample: Allow reinitialization without ever setting channel layouts (cherry picked from commit 80a28c7509a11114e1aea5b208d56c6646d69c07)
- swresample: Allow reinitialization without ever setting channel counts
- avcodec/h264: Do not fail with randomly truncated VUIs
- avcodec/h264_ps: Move truncation check from VUI to SPS
- avcodec/h264: Be more tolerant to changing pps id between slices
- avcodec/aacdec: Fix storing state before PCE decode
- avcodec/h264: reset the counts in the correct context
- avcodec/h264_slice: Do not reset mb_aff_frame per slice
- avcodec/h264: finish previous slices before switching to single thread mode
- avcodec/h264: Fix race between slices where one overwrites data from the next
- avcodec/h264_refs: Do not set reference to things which do not exist
- avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode
- h264: avoid unnecessary calls to get_format
- avcodec/msrledec: restructure msrle_decode_pal4() based on the line number instead of the pixel pointer
version 2.5.5:
- vp9: make above buffer pointer 32-byte aligned.
- avcodec/dnxhddec: Check that the frame is interlaced before using cur_field
- avformat/mov: Disallow ".." in dref unless use_absolute_path is set
- avformat/mov: Check for string truncation in mov_open_dref()
- avformat/mov: Use sizeof(filename) instead of a literal number
- eac3dec: fix scaling
- ac3_fixed: fix computation of spx_noise_blend
- ac3_fixed: fix out-of-bound read
- ac3dec_fixed: always use the USE_FIXED=1 variant of the AC3DecodeContext
- avcodec/012v: redesign main loop
- avcodec/012v: Check dimensions more completely
- asfenc: fix leaking asf->index_ptr on error
- avcodec/options_table: remove extradata_size from the AVOptions table
- ffmdec: limit the backward seek to the last resync position
- ffmdec: make sure the time base is valid
- ffmdec: fix infinite loop at EOF
- ffmdec: initialize f_cprv, f_stvi and f_stau
- avformat/rm: limit packet size
- avcodec/webp: validate the distance prefix code
- avcodec/rv10: check size of s->mb_width * s->mb_height
- eamad: check for out of bounds read
- mdec: check for out of bounds read
- arm: Suppress tags about used cpu arch and extensions
- aic: Fix decoding files with odd dimensions
- avcodec/tiff: move bpp check to after "end:"
- mxfdec: Fix the error handling for when strftime fails
- avcodec/opusdec: Fix delayed sample value
- avcodec/opusdec: Clear out pointers per packet
- avcodec/utils: Align YUV411 by as much as the other YUV variants
- vp9: fix segmentation map retention with threading enabled.
- webp: ensure that each transform is only used once
- doc/protocols/tcp: fix units of listen_timeout option value, from microseconds to milliseconds
- fix VP9 packet decoder returning 0 instead of the used data size
- avformat/flvenc: check that the codec_tag fits in the available bits
- avcodec/utils: use correct printf specifier in ff_set_sar
- avutil/imgutils: correctly check for negative SAR components
- swscale/utils: clear formatConvBuffer on allocation
- avformat/bit: only accept the g729 codec and 1 channel
- avformat/bit: check that pkt->size is 10 in write_packet
- avformat/adxdec: check avctx->channels for invalid values
- avformat/adxdec: set avctx->channels in adx_read_header
- Fix buffer_size argument to init_put_bits() in multiple encoders.
- mips/acelp_filters: fix incorrect register constraint
- avcodec/hevc_ps: Sanity checks for some log2_* values
- avcodec/zmbv: Check len before reading in decode_frame()
- avcodec/h264: Only reinit quant tables if a new PPS is allowed
- avcodec/snowdec: Fix ref value check
- swscale/utils: More carefully merge and clear coefficients outside the input
- avcodec/a64multienc: Assert that the Packet size does not grow
- avcodec/a64multienc: simplify frame handling code
- avcodec/a64multienc: fix use of uninitialized values in to_meta_with_crop
- avcodec/a64multienc: initialize mc_meta_charset to zero
- avcodec/a64multienc: don't set incorrect packet size
- avcodec/a64multienc: use av_frame_ref instead of copying the frame
- avcodec/x86/mlpdsp_init: Simplify mlp_filter_channel_x86()
- h264: initialize H264Context.avctx in init_thread_copy
- wtvdec: fix integer overflow resulting in errors with large files
- avcodec/gif: fix off by one in column offsetting finding
version 2.5.4:
- avcodec/arm/videodsp_armv5te: Fix linking failure with shared libs
- avcodec/mjpegdec: Skip blocks which are outside the visible area
- avcodec/h264_slice: ignore SAR changes in slices after the first
- avcodec/h264_slice: Check picture structure before setting the related fields
- avcodec/h264_slice: Do not change frame_num after the first slice
- avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT
- avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT
- avcodec/h264: Be more strict on rejecting pps/sps changes
- avcodec/h264: Be more strict on rejecting pps_id changes
- avcodec/h264_ps: More completely check the bit depths
- avformat/thp: Check av_get_packet() for failure not only for partial output
- swscale/utils: Limit filter shifting so as not to read from prior the array
- avcodec/mpegvideo_motion: Fix gmc chroma dimensions
- avcodec/mjpegdec: Check number of components for JPEG-LS
- avcodec/mjpegdec: Check escape sequence validity
- avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior
- avformat/mpc8: fix broken pointer math
- avformat/mpc8: fix hang with fuzzed file
- avformat/tta: fix crash with corrupted files
- avcodec/ppc/idctdsp.c: POWER LE support in idct_add_altivec()
- swscale/input: fix rgba64 alpha non native
- swscale/input: Fix alpha of YA16 input
- libavcodec/ppc/mpegvideoencdsp.c: fix stack smashing in pix_norm1_altivec() and pix_sum_altivec()
- avformat/rmdec: Check for overflow in ff_rm_read_mdpr_codecdata()
- avformat/mpeg: do not count PES packets inside PES packets during probing
- hevc: always clip luma_log2_weight_denom
- rtpdec_h263_rfc2190: Clear the stored bits if discarding buffered data
- aacenc: correctly check returned value
- swscale: check memory allocations
- opt: check memory allocation
- avformat/utils: check for malloc failure
- avcodec/flac_parser: fix handling EOF if no headers are found
- avfilter/vf_framepack: Check and update frame_rate
- vp8: improve memory allocation checks
- configure: enable vsx together with altivec for ppc64el
- avcodec/hevc: Fix handling of skipped_bytes() reallocation failures
- qpeg: avoid pointless invalid memcpy()
version 2.5.3:
- vp9: fix parser return values in error case
- ffmpeg: Clear error message array at init.
- avcodec/dvdsubdec: fix accessing dangling pointers
- avcodec/dvdsubdec: error on bitmaps with size 0
- cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file()
- mov: Fix negative size calculation in mov_read_default().
- avformat/mov: fix integer overflow in mov_read_udta_string()
- mov: Fix overflow and error handling in read_tfra().
- mov: Avoid overflow with mov_metadata_raw()
- avcodec/dvdsubdec: fix out of bounds accesses
- avfilter/vf_sab: fix filtering tiny images
- avformat/flvdec: Increase string array size
- avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream
- avformat/cdxl: Fix integer overflow of image_size
- libavformat: Build hevc.o when building the RTP muxer
version 2.5.2:
- avcodec/indeo3: ensure offsets are non negative
- avcodec/h264: Check *log2_weight_denom
- avcodec/hevc_ps: Check diff_cu_qp_delta_depth
- avcodec/h264: Clear delayed_pic on deallocation
- avcodec/hevc: clear filter_slice_edges() on allocation
- avcodec/dcadec: Check that the added xch channel isnt already there
- avcodec/indeo3: use signed variables to avoid underflow
- swscale: increase yuv2rgb table headroom
- avformat/mov: fix integer overflow of size
- avformat/mov: check atom nesting depth
- avcodec/utvideodec: Fix handling of slice_height=0
- avcodec/xface: correct the XFACE_MAX_* values
- avcodec/vmdvideo: Check len before using it in method 3
- configure: create the tests directory like the doc directory
- mmvideo: check frame dimensions
- jvdec: check frame dimensions
version 2.5.1:
- lavu/frame: fix malloc error path in av_frame_copy_props()
- avformat/aviobuf: Check that avio_seek() target is non negative
- swresample/soxr_resample: fix error handling
- avformat/flvdec: fix potential use of uninitialized variables
- avformat/crypto: fix key vs iv typo
- configure: use use_pkg_config() instead of check_pkg_config() for libsmbclient
- avcodec/ppc/vp3dsp_altivec: POWER LE support to vp3_idct_add_altivec()
- avformat/matroskadec: fix handling of recursive SeekHead elements
- doc/examples/filtering_video: fix frame rate
- avcodec/mpegaudiodec_template: only allocate fdsp when its used
- doc/examples/transcoding: check encoder before using it
- update MAINTAINERS file
- POWER LE support in put_vp8_epel_h_altivec_core() put_vp8_epel_v_altivec_core() put_vp8_pixels16_altivec()
- POWER LE support in vc1_inv_trans_8x4_altivec()
version 2.5:
- HEVC/H.265 RTP payload format (draft v6) packetizer
- SUP/PGS subtitle demuxer
- ffprobe -show_pixel_formats option
- CAST128 symmetric block cipher, ECB mode
- STL subtitle demuxer and decoder
- libutvideo YUV 4:2:2 10bit support
- XCB-based screen-grabber
- UDP-Lite support (RFC 3828)
- xBR scaling filter
- AVFoundation screen capturing support
- ffserver supports codec private options
- creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
- WebP muxer with animated WebP support
- zygoaudio decoding support
- APNG decoder and demuxer
- postproc visualization support
version 2.4:
- Icecast protocol
- ported lenscorrection filter from frei0r filter
- large optimizations in dctdnoiz to make it usable
- ICY metadata are now requested by default with the HTTP protocol
- support for using metadata in stream specifiers in fftools
- LZMA compression support in TIFF decoder
- support for H.261 RTP payload format (RFC 4587)
- HEVC/H.265 RTP payload format (draft v6) depacketizer
- added codecview filter to visualize information exported by some codecs
- Matroska 3D support thorugh side data
- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
- silenceremove filter
version 2.3.2:
- snow: fix null pointer dereference
- huffyucdec: fix overread
- vc1dec: fix crash
- iff: fix out of array access
- matroskaenc: fix assertion failure
- cdgraphics: fix infinite loop
- dvdsub_parser: fix infinite loop
- mpeg12dec: support decoding some broken files
- v4l2enc: fix crash
- h264_parser: fix handling huge resolutions
- h264_mp4toannexb_bsf: multiple bugfixes
version 2.3.1:
- public AVDCT API/ABI for DCT functions
- g2meet: allow size changes within original sizes
- dv: improved error resilience, fixing Ticket2340 and Ticket2341
version 2.3:
- AC3 fixed-point decoding
@@ -552,7 +42,7 @@ version 2.3:
- libbs2b-based stereo-to-binaural audio filter
- libx264 reference frames count limiting depending on level
- native Opus decoder
- display matrix export and rotation API
- display matrix export and rotation api
- WebVTT encoder
- showcqt multimedia filter
- zoompan filter
@@ -594,7 +84,6 @@ version 2.2:
- libx265 encoder
- dejudder filter
- Autodetect VDA like all other hardware accelerations
- aliases and defaults for Ogg subtypes (opus, spx)
version 2.1:

View File

@@ -15,7 +15,6 @@ Specifically, the GPL parts of FFmpeg are:
- libpostproc
- libmpcodecs
- optional x86 optimizations in the files
libavcodec/x86/flac_dsp_gpl.asm
libavcodec/x86/idct_mmx.c
- libutvideo encoding/decoding wrappers in
libavcodec/libutvideo*.cpp
@@ -34,7 +33,6 @@ Specifically, the GPL parts of FFmpeg are:
- vf_geq.c
- vf_histeq.c
- vf_hqdn3d.c
- vf_interlace.c
- vf_kerndeint.c
- vf_mcdeint.c
- vf_mp.c
@@ -64,7 +62,6 @@ There are a handful of files under other licensing terms, namely:
documentation accompanying your program if you only distribute executables.
You must also indicate any changes including additions and deletions to
those three files in the documentation.
tests/reference.pnm is under the expat license
external libraries

View File

@@ -14,6 +14,7 @@ patches and related discussions.
Project Leader
==============
Michael Niedermayer
final design decisions
@@ -53,7 +54,7 @@ release management Michael Niedermayer
Communication
=============
website Deby Barbara Lepage
website Robert Swain
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists Michael Niedermayer, Baptiste Coudurier, Lou Logan
@@ -308,7 +309,6 @@ libavdevice
avfoundation.m Thilo Borgmann
decklink* Deti Fliegl
dshow.c Roger Pack (CC rogerdpack@gmail.com)
fbdev_enc.c Lukasz Marek
gdigrab.c Roger Pack (CC rogerdpack@gmail.com)
@@ -319,7 +319,7 @@ libavdevice
pulse_audio_enc.c Lukasz Marek
qtkit.m Thilo Borgmann
sdl Stefano Sabatini
v4l2.c Giorgio Vazzana
v4l2.c Luca Abeni
vfwcap.c Ramiro Polla
xv.c Lukasz Marek
@@ -343,7 +343,6 @@ Filters:
af_compand.c Paul B Mahol
af_ladspa.c Paul B Mahol
af_pan.c Nicolas George
af_silenceremove.c Paul B Mahol
avf_avectorscope.c Paul B Mahol
avf_showcqt.c Muhammad Faiz
vf_blend.c Paul B Mahol
@@ -354,9 +353,7 @@ Filters:
vf_extractplanes.c Paul B Mahol
vf_histogram.c Paul B Mahol
vf_hqx.c Clément Bœsch
vf_idet.c Pascal Massimino
vf_il.c Paul B Mahol
vf_lenscorrection.c Daniel Oberhoff
vf_mergeplanes.c Paul B Mahol
vf_psnr.c Paul B Mahol
vf_scale.c Michael Niedermayer
@@ -385,7 +382,6 @@ Muxers/Demuxers:
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
ape.c Kostya Shishkov
apngdec.c Benoit Fouet
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
@@ -462,15 +458,12 @@ Muxers/Demuxers:
rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov
rtmp* Kostya Shishkov
rtp.c, rtpenc.c Martin Storsjo
rtpdec_h261.*, rtpenc_h261.* Thomas Volkert
rtpdec_hevc.*, rtpenc_hevc.* Thomas Volkert
rtpdec_asf.* Ronald S. Bultje
rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo
rtsp.c Luca Barbato
sbgdec.c Nicolas George
sdp.c Martin Storsjo
segafilm.c Mike Melanson
segment.c Stefano Sabatini
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
smjpeg* Paul B Mahol
@@ -497,7 +490,6 @@ Protocols:
libssh.c Lukasz Marek
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c Marvin Scholz
libswresample
@@ -536,8 +528,7 @@ x86 Michael Niedermayer
Releases
========
2.5 Michael Niedermayer
2.4 Michael Niedermayer
2.3 Michael Niedermayer
2.2 Michael Niedermayer
1.2 Michael Niedermayer

View File

@@ -32,7 +32,6 @@ OBJS-ffmpeg += ffmpeg_opt.o ffmpeg_filter.o
OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_vda.o
OBJS-ffserver += ffserver_config.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
@@ -64,7 +63,7 @@ FF_DEP_LIBS := $(DEP_LIBS)
all: $(AVPROGS)
$(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
@@ -93,7 +92,6 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
include $(SRC_PATH)/library.mak
endef
@@ -112,14 +110,14 @@ endef
$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
ffprobe.o cmdutils.o : libavutil/ffversion.h
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
$(CP) $< $@
$(STRIP) $@
%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
OBJDIRS += tools

View File

@@ -1,40 +1,19 @@
FFmpeg README
=============
FFmpeg is a collection of libraries and tools to process multimedia content
such as audio, video, subtitles and related metadata.
1) Documentation
----------------
## Libraries
* Read the documentation in the doc/ directory in git.
* `libavcodec` provides implementation of a wider range of codecs.
* `libavformat` implements streaming protocols, container formats and basic I/O access.
* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
* `libavdevice` provides an abstraction to access capture and playback devices.
* `libswresample` implements audio mixing and resampling routines.
* `libswscale` implements color conversion and scaling routines.
You can also view it online at http://ffmpeg.org/documentation.html
## Tools
2) Licensing
------------
* [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
manipulate, convert and stream multimedia content.
* [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
* [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analisys tool to inspect
multimedia content.
* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
* See the LICENSE file.
## Documentation
3) Build and Install
--------------------
The offline documentation is available in the **doc/** directory.
The online documentation is available in the main [website](http://ffmpeg.org)
and in the [wiki](http://trac.ffmpeg.org).
### Examples
Coding examples are available in the **doc/examples** directory.
## License
FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
GPL. Please refer to the LICENSE file for detailed information.
* See the INSTALL file.

View File

@@ -1 +1 @@
2.5.10
2.3.2

View File

@@ -1,101 +1,177 @@
┌────────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 2.5 "Bohr"
└────────────────────────────────────────┘
┌───────────────────────────────────────────
│ RELEASE NOTES for FFmpeg 2.3 "Mandelbrot"
└───────────────────────────────────────────
The FFmpeg Project proudly presents FFmpeg 2.5 "Bohr", 2.5 months after the
release of 2.4.
The FFmpeg Project proudly presents FFmpeg 2.3 "Mandelbrot", a major
release with all the great features committed during the three-month period
since the release of FFmpeg 2.2.
The most important new features are AVFoundation screen-grabbing support,
animated WebP decoding support, and Animated PNG support. In addition, many
exciting features for video streaming are also implemented, including MPEG-
DASH fragmenting muxer, HEVC RTP payload muxer, and UDP Lite support.
In this release, there are lots of internal overhauls that make FFmpeg a
more accessible project for new developers. Many important new
optimizations and features like QTKit and AVFoundation input devices are
committed. Contributions done by Libav such as a new native Opus decoder
are also merged.
As usual, if you have any question on this release or any FFmpeg related
topic, feel free to join us on the #ffmpeg IRC channel (on
irc.freenode.net).
Because of the increasing difficulty to maintain and lack of maintainers,
we are very sorry to say that we have removed all Blackfin and SPARC
architecture assembly optimizations with the cleanups done. If you are
interested in maintaining optimization for these two architecture, feel
free to contact us and we will restore the code!
Oh, and since this release, this modern-looking release note is provided in
addition to the old-style Changelog file, to make it easier for you to
focus on the most important features in this release.
Enjoy!
┌────────────────────────────┐
🔨 API Information │
* API Information │
└────────────────────────────┘
FFmpeg 2.5 includes the following library versions:
FFmpeg 2.3 is completely source-compatible to the FFmpeg 2.2 series. There
are however some API deprecations that you need to take care of. Use `git
diff n2.2 n2.3 doc/APIchanges` to show the list of added and deprecated
APIs. FFmpeg 2.3 includes the following library versions:
• libavutil 54.15.100
• libavcodec 56.13.100
• libavformat 56.15.102
• libavdevice 56. 3.100
• libavfilter 5. 2.103
• libswscale 3. 1.101
• libswresample 1. 1.100
• libpostproc 53. 3.100
• libavutil 52.92.100
• libavcodec 55.69.100
• libavformat 55.48.100
• libavdevice 55.13.102
• libavfilter 4.11.100
• libswscale 2. 6.100
• libswresample 0.19.100
• libpostproc 52. 3.100
Important API changes since 2.4:
Please refer to the doc/APIChanges file for more information.
• avpriv_dv_frame_profile2() has been deprecated
┌────────────────────────────┐
│ New Optimization │
└────────────────────────────┘
We are excited to announce that we have committed new x86 assembly
optimization for HEVC, and FFmpeg's audio resampler libswresample. ARM
users will get a boost in MLP/TrueHD decoding thanks to new optimization.
Decoding Huffyuv also got a major boost from optimization on the C code.
Please refer to the doc/APIchanges file for more information.
Of special interest for Microsoft Visual Studio users, we have also
converted some preexisting x86 assembly to NASM/Yasm format compatible
with MSVC setup, especially in the area of audio resampling.
Another major feature in this release is the introduction of AArch64
(ARMv8) assembly optimization. AArch64 is another name for the first
64-bit ARM architecture, used by Apple A7 SoC inside iPhone 5S. Some
32-bit ARM assembly has already been ported to AArch64, but more work is
underway.
┌────────────────────────────┐
│ Native Opus decoder │
└────────────────────────────┘
Opus is an open audio format jointly developed by Xiph.Org, Mozilla,
Skype/Microsoft, and Broadcom. It combines the features of the Skype Cilk
speech codec and the Xiph.Org CELT music codec into one low-latency
codec. Decoding Opus is already possible since FFmpeg 1.0 using the
libopus library, but the new Opus native decoder brings a higher level of
stability and speed.
┌────────────────────────────┐
│ QTKit and AVFoundation │
└────────────────────────────┘
For OS X users, the new QTKit and AVFoundation devices allow you to use
the integrated camera on Macs. AVFoundation is a newer API only available
on OS X 10.7 "Lion" or newer. For users with older OS X systems, the
QTKit device using the older OS X API is for you.
┌────────────────────────────┐
│ API Additions │
└────────────────────────────┘
In this release, stream side data are introduced as AVStream.side_data as
a way to store miscellaneous stream-wide information. The format is
similar to the previously anonymous structure AVPacket.side_data (now
named as AVPacketSideData). With this change, audio ReplayGain
information and video rotation matrix are now exported through this API,
if available in the demuxer.
We also have improved libswresample's Doxygen API documentation, so new
developers wishing to use FFmpeg's excellent libraries can get started
more easily and faster.
┌────────────────────────────┐
│ Last But Not Least │
└────────────────────────────┘
Other interesting new features including hqx video filter, a pixel art
scaling filter; a fixed-point AC-3 decoder contributed by Imagination
Technologies; an On2 TrueMotion VP7 video decoder; an HTML5 WebVTT
subtitle encoder that allows creation of WebVTT from any text-based
subtitles; and an 1-bit Direct Stream Digital audio decoder.
┌────────────────────────────┐
│ ★ List of New Features │
└────────────────────────────┘
┌────────────────────────────┐
ffprobe
Command line tools
└────────────────────────────┘
-show_pixel_formats option
┌────────────────────────────┐
│ ffserver │
└────────────────────────────┘
• codec private options support
Support for decoding through DXVA2 in ffmpeg
┌────────────────────────────┐
│ libavcodec │
└────────────────────────────┘
STL subtitle decoder
libutvideo YUV 4:2:2 10bit support
animated WebP decoding support
zygoaudio decoding support
APNG decoder
AC3 fixed-point decoding
VP7 video decoder
Alias PIX image encoder and decoder
Improvements to the BRender PIX image decoder
Improvements to the XBM decoder
• Improvements to OpenEXR image decoder
• Support decoding 16-bit RLE SGI images
• Direct Stream Digital (DSD) decoder
• On2 AVC (Audio for Video) decoder
• Native Opus decoder
• WebVTT encoder
┌────────────────────────────┐
│ libavdevice │
└────────────────────────────┘
XCB-based screen-grabber
AVFoundation screen capturing support
QTKit input device
GDI screen grabbing for Windows
• AVFoundation input device
┌────────────────────────────┐
│ libavformat │
└────────────────────────────┘
HEVC/H.265 RTP payload format (draft v6) packetizer
SUP/PGS subtitle demuxer
STL subtitle demuxer
UDP-Lite support (RFC 3828)
MPEG-DASH segmenting muxer, which allows creating DASH compatible
fragmented MP4
WebP muxer
APNG demuxer
subfile protocol
Phantom Cine demuxer
Alternative rendition support for HTTP Live Streaming
Magic Lantern Video (MLV) demuxer
Image format auto-detection
• LRC lyric file demuxer and muxer
Samba protocol (via libsmbclient)
WebM DASH Manifest muxer
┌────────────────────────────┐
│ libavfilter │
└────────────────────────────┘
xBR scaling filter
shuffleplanes filter
• libbs2b-based stereo-to-binaural audio filter
• showcqt multimedia filter
• zoompan filter
• signalstats filter
• hqx filter (hq2x, hq3x, hq4x)
• flanger filter
• libfribidi support in drawtext
┌────────────────────────────┐
│ libavutil
└────────────────────────────┘
┌────────────────────────────┐
│ ⚠ Behaviour changes
└────────────────────────────┘
CAST128 symmetric block cipher, ECB mode
┌────────────────────────────┐
│ libpostproc │
└────────────────────────────┘
• visualization support
libx264 reference frames count is now limited depending on level chosen
• Because of the new image format auto-detection feature, you don't need to
specify image format when decoding an image with no extension.

View File

@@ -166,7 +166,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
int first;
first = 1;
for (po = options; po->name; po++) {
for (po = options; po->name != NULL; po++) {
char buf[64];
if (((po->flags & req_flags) != req_flags) ||
@@ -205,7 +205,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
const char *p = strchr(name, ':');
int len = p ? p - name : strlen(name);
while (po->name) {
while (po->name != NULL) {
if (!strncmp(name, po->name, len) && strlen(po->name) == len)
break;
po++;
@@ -254,7 +254,7 @@ static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
win32_argv_utf8 = av_mallocz(sizeof(char *) * (win32_argc + 1) + buffsize);
argstr_flat = (char *)win32_argv_utf8 + sizeof(char *) * (win32_argc + 1);
if (!win32_argv_utf8) {
if (win32_argv_utf8 == NULL) {
LocalFree(argv_w);
return;
}
@@ -444,7 +444,7 @@ int locate_option(int argc, char **argv, const OptionDef *options,
(po->name && !strcmp(optname, po->name)))
return i;
if (!po->name || po->flags & HAS_ARG)
if (po->flags & HAS_ARG)
i++;
}
return 0;
@@ -959,10 +959,9 @@ static int init_report(const char *env)
report_file = fopen(filename.str, "w");
if (!report_file) {
int ret = AVERROR(errno);
av_log(NULL, AV_LOG_ERROR, "Failed to open report \"%s\": %s\n",
filename.str, strerror(errno));
return ret;
return AVERROR(errno);
}
av_log_set_callback(log_callback_report);
av_log(NULL, AV_LOG_INFO,
@@ -1243,7 +1242,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
is_dev = is_device(ofmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ofmt->name, name) < 0) &&
if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
strcmp(ofmt->name, last_name) > 0) {
name = ofmt->name;
long_name = ofmt->long_name;
@@ -1254,7 +1253,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
is_dev = is_device(ifmt->priv_class);
if (!is_dev && device_only)
continue;
if ((!name || strcmp(ifmt->name, name) < 0) &&
if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
strcmp(ifmt->name, last_name) > 0) {
name = ifmt->name;
long_name = ifmt->long_name;
@@ -1263,7 +1262,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
if (name && strcmp(ifmt->name, name) == 0)
decode = 1;
}
if (!name)
if (name == NULL)
break;
last_name = name;
@@ -1544,8 +1543,7 @@ int show_protocols(void *optctx, const char *opt, const char *arg)
int show_filters(void *optctx, const char *opt, const char *arg)
{
#if CONFIG_AVFILTER
const AVFilter *filter = NULL;
const AVFilter av_unused(*filter) = NULL;
char descr[64], *descr_cur;
int i, j;
const AVFilterPad *pad;
@@ -1558,6 +1556,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
" V = Video input/output\n"
" N = Dynamic number and/or type of input/output\n"
" | = Source or sink filter\n");
#if CONFIG_AVFILTER
while ((filter = avfilter_next(filter))) {
descr_cur = descr;
for (i = 0; i < 2; i++) {
@@ -1582,8 +1581,6 @@ int show_filters(void *optctx, const char *opt, const char *arg)
filter->process_command ? 'C' : '.',
filter->name, descr, filter->description);
}
#else
printf("No filters available: libavfilter disabled\n");
#endif
return 0;
}
@@ -1642,19 +1639,19 @@ int show_layouts(void *optctx, const char *opt, const char *arg)
const char *name, *descr;
printf("Individual channels:\n"
"NAME DESCRIPTION\n");
"NAME DESCRIPTION\n");
for (i = 0; i < 63; i++) {
name = av_get_channel_name((uint64_t)1 << i);
if (!name)
continue;
descr = av_get_channel_description((uint64_t)1 << i);
printf("%-14s %s\n", name, descr);
printf("%-12s%s\n", name, descr);
}
printf("\nStandard channel layouts:\n"
"NAME DECOMPOSITION\n");
"NAME DECOMPOSITION\n");
for (i = 0; !av_get_standard_channel_layout(i, &layout, &name); i++) {
if (name) {
printf("%-14s ", name);
printf("%-12s", name);
for (j = 1; j; j <<= 1)
if ((layout & j))
printf("%s%s", (layout & (j - 1)) ? "+" : "", av_get_channel_name(j));
@@ -1860,23 +1857,21 @@ int read_yesno(void)
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
{
int64_t ret;
int ret;
FILE *f = av_fopen_utf8(filename, "rb");
if (!f) {
ret = AVERROR(errno);
av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
strerror(errno));
return ret;
return AVERROR(errno);
}
fseek(f, 0, SEEK_END);
*size = ftell(f);
fseek(f, 0, SEEK_SET);
if (*size == (size_t)-1) {
ret = AVERROR(errno);
av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", strerror(errno));
fclose(f);
return ret;
return AVERROR(errno);
}
*bufptr = av_malloc(*size + 1);
if (!*bufptr) {
@@ -1888,9 +1883,9 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
if (ret < *size) {
av_free(*bufptr);
if (ferror(f)) {
ret = AVERROR(errno);
av_log(NULL, AV_LOG_ERROR, "Error while reading file '%s': %s\n",
filename, strerror(errno));
ret = AVERROR(errno);
} else
ret = AVERROR_EOF;
} else {
@@ -1997,7 +1992,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
switch (check_stream_specifier(s, st, p + 1)) {
case 1: *p = 0; break;
case 0: continue;
default: exit_program(1);
default: return NULL;
}
if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
@@ -2055,184 +2050,3 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
}
return array;
}
#if CONFIG_AVDEVICE
static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
{
int ret, i;
AVFormatContext *dev = NULL;
AVDeviceInfoList *device_list = NULL;
AVDictionary *tmp_opts = NULL;
if (!fmt || !fmt->priv_class || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
return AVERROR(EINVAL);
printf("Audo-detected sources for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sources. Not implemented.\n");
goto fail;
}
/* TODO: avformat_open_input calls read_header callback which is not necessary.
Function like avformat_alloc_output_context2 for input could be helpful here. */
av_dict_copy(&tmp_opts, opts, 0);
if ((ret = avformat_open_input(&dev, NULL, fmt, &tmp_opts)) < 0) {
printf("Cannot open device: %s.\n", fmt->name);
goto fail;
}
if ((ret = avdevice_list_devices(dev, &device_list)) < 0) {
printf("Cannot list sources.\n");
goto fail;
}
for (i = 0; i < device_list->nb_devices; i++) {
printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
device_list->devices[i]->device_name, device_list->devices[i]->device_description);
}
fail:
av_dict_free(&tmp_opts);
avdevice_free_list_devices(&device_list);
avformat_close_input(&dev);
return ret;
}
static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
{
int ret, i;
AVFormatContext *dev = NULL;
AVDeviceInfoList *device_list = NULL;
AVDictionary *tmp_opts = NULL;
if (!fmt || !fmt->priv_class || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
return AVERROR(EINVAL);
printf("Audo-detected sinks for %s:\n", fmt->name);
if (!fmt->get_device_list) {
ret = AVERROR(ENOSYS);
printf("Cannot list sinks. Not implemented.\n");
goto fail;
}
if ((ret = avformat_alloc_output_context2(&dev, fmt, NULL, NULL)) < 0) {
printf("Cannot open device: %s.\n", fmt->name);
goto fail;
}
av_dict_copy(&tmp_opts, opts, 0);
av_opt_set_dict2(dev, &tmp_opts, AV_OPT_SEARCH_CHILDREN);
if ((ret = avdevice_list_devices(dev, &device_list)) < 0) {
printf("Cannot list sinks.\n");
goto fail;
}
for (i = 0; i < device_list->nb_devices; i++) {
printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
device_list->devices[i]->device_name, device_list->devices[i]->device_description);
}
fail:
av_dict_free(&tmp_opts);
avdevice_free_list_devices(&device_list);
avformat_free_context(dev);
return ret;
}
static int show_sinks_sources_parse_arg(const char *arg, char **dev, AVDictionary **opts)
{
int ret;
if (arg) {
char *opts_str = NULL;
av_assert0(dev && opts);
*dev = av_strdup(arg);
if (!*dev)
return AVERROR(ENOMEM);
if ((opts_str = strchr(*dev, ','))) {
*(opts_str++) = '\0';
if (opts_str[0] && ((ret = av_dict_parse_string(opts, opts_str, "=", ":", 0)) < 0)) {
av_freep(dev);
return ret;
}
}
} else
printf("\nDevice name is not provided.\n"
"You can pass devicename[,opt1=val1[,opt2=val2...]] as an argument.\n\n");
return 0;
}
int show_sources(void *optctx, const char *opt, const char *arg)
{
AVInputFormat *fmt = NULL;
char *dev = NULL;
AVDictionary *opts = NULL;
int ret = 0;
int error_level = av_log_get_level();
av_log_set_level(AV_LOG_ERROR);
if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
goto fail;
do {
fmt = av_input_audio_device_next(fmt);
if (fmt) {
if (!strcmp(fmt->name, "lavfi"))
continue; //it's pointless to probe lavfi
if (dev && strcmp(fmt->name, dev))
continue;
print_device_sources(fmt, opts);
}
} while (fmt);
do {
fmt = av_input_video_device_next(fmt);
if (fmt) {
if (dev && strcmp(fmt->name, dev))
continue;
print_device_sources(fmt, opts);
}
} while (fmt);
fail:
av_dict_free(&opts);
av_free(dev);
av_log_set_level(error_level);
return ret;
}
int show_sinks(void *optctx, const char *opt, const char *arg)
{
AVOutputFormat *fmt = NULL;
char *dev = NULL;
AVDictionary *opts = NULL;
int ret = 0;
int error_level = av_log_get_level();
av_log_set_level(AV_LOG_ERROR);
if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
goto fail;
do {
fmt = av_output_audio_device_next(fmt);
if (fmt) {
if (dev && strcmp(fmt->name, dev))
continue;
print_device_sinks(fmt, opts);
}
} while (fmt);
do {
fmt = av_output_video_device_next(fmt);
if (fmt) {
if (dev && strcmp(fmt->name, dev))
continue;
print_device_sinks(fmt, opts);
}
} while (fmt);
fail:
av_dict_free(&opts);
av_free(dev);
av_log_set_level(error_level);
return ret;
}
#endif

View File

@@ -443,20 +443,6 @@ int show_formats(void *optctx, const char *opt, const char *arg);
*/
int show_devices(void *optctx, const char *opt, const char *arg);
#if CONFIG_AVDEVICE
/**
* Print a listing containing audodetected sinks of the output device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sinks(void *optctx, const char *opt, const char *arg);
/**
* Print a listing containing audodetected sources of the input device.
* Device name with options may be passed as an argument to limit results.
*/
int show_sources(void *optctx, const char *opt, const char *arg);
#endif
/**
* Print a listing containing all the codecs supported by the
* program.

View File

@@ -27,9 +27,3 @@
{ "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
{ "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, "set OpenCL environment options" },
#endif
#if CONFIG_AVDEVICE
{ "sources" , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
"list sources of the input device", "device" },
{ "sinks" , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
"list sinks of the output device", "device" },
#endif

View File

@@ -22,7 +22,6 @@
#include "libavutil/time.h"
#include "libavutil/log.h"
#include "libavutil/opencl.h"
#include "libavutil/avstring.h"
#include "cmdutils.h"
typedef struct {
@@ -239,8 +238,7 @@ int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
devices[count].platform_idx = i;
devices[count].device_idx = j;
devices[count].runtime = score;
av_strlcpy(devices[count].device_name, device_node->device_name,
sizeof(devices[count].device_name));
strcpy(devices[count].device_name, device_node->device_name);
count++;
}
}

View File

@@ -805,7 +805,7 @@ struct AVS_Library {
AVSC_INLINE AVS_Library * avs_load_library() {
AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
if (!library)
if (library == NULL)
return NULL;
library->handle = LoadLibrary("avisynth");
if (library->handle == NULL)
@@ -870,7 +870,7 @@ fail:
}
AVSC_INLINE void avs_free_library(AVS_Library *library) {
if (!library)
if (library == NULL)
return;
FreeLibrary(library->handle);
free(library);

View File

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

View File

@@ -39,7 +39,6 @@
#include <windows.h>
#include <process.h>
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/internal.h"
#include "libavutil/mem.h"
@@ -55,30 +54,36 @@ typedef struct pthread_t {
* not mutexes */
typedef CRITICAL_SECTION pthread_mutex_t;
/* This is the CONDITION_VARIABLE typedef for using Windows' native
* conditional variables on kernels 6.0+. */
#if HAVE_CONDITION_VARIABLE_PTR
typedef CONDITION_VARIABLE pthread_cond_t;
#else
/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
* conditional variables on kernels 6.0+.
* MinGW does not currently have this typedef. */
typedef struct pthread_cond_t {
void *Ptr;
void *ptr;
} pthread_cond_t;
/* function pointers to conditional variable API on windows 6.0+ kernels */
#if _WIN32_WINNT < 0x0600
static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
static void (WINAPI *cond_init)(pthread_cond_t *cond);
static void (WINAPI *cond_signal)(pthread_cond_t *cond);
static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
DWORD milliseconds);
#else
#define cond_init InitializeConditionVariable
#define cond_broadcast WakeAllConditionVariable
#define cond_signal WakeConditionVariable
#define cond_wait SleepConditionVariableCS
#endif
#if _WIN32_WINNT >= 0x0600
#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
#endif
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t *h = arg;
h->ret = h->func(h->arg);
return 0;
}
static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
static int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
{
thread->func = start_routine;
thread->arg = arg;
@@ -87,7 +92,7 @@ static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
return !thread->handle;
}
static av_unused void pthread_join(pthread_t thread, void **value_ptr)
static void pthread_join(pthread_t thread, void **value_ptr)
{
DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
if (ret != WAIT_OBJECT_0)
@@ -118,36 +123,6 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
return 0;
}
#if _WIN32_WINNT >= 0x0600
static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
InitializeConditionVariable(cond);
return 0;
}
/* native condition variables do not destroy */
static inline void pthread_cond_destroy(pthread_cond_t *cond)
{
return;
}
static inline void pthread_cond_broadcast(pthread_cond_t *cond)
{
WakeAllConditionVariable(cond);
}
static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
SleepConditionVariableCS(cond, mutex, INFINITE);
return 0;
}
static inline void pthread_cond_signal(pthread_cond_t *cond)
{
WakeConditionVariable(cond);
}
#else // _WIN32_WINNT < 0x0600
/* for pre-Windows 6.0 platforms we need to define and use our own condition
* variable and api */
typedef struct win32_cond_t {
@@ -159,14 +134,7 @@ typedef struct win32_cond_t {
volatile int is_broadcast;
} win32_cond_t;
/* function pointers to conditional variable API on windows 6.0+ kernels */
static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
static void (WINAPI *cond_init)(pthread_cond_t *cond);
static void (WINAPI *cond_signal)(pthread_cond_t *cond);
static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
DWORD milliseconds);
static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
static int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
win32_cond_t *win32_cond = NULL;
if (cond_init) {
@@ -178,7 +146,7 @@ static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_
win32_cond = av_mallocz(sizeof(win32_cond_t));
if (!win32_cond)
return ENOMEM;
cond->Ptr = win32_cond;
cond->ptr = win32_cond;
win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
if (!win32_cond->semaphore)
return ENOMEM;
@@ -191,9 +159,9 @@ static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_
return 0;
}
static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
static void pthread_cond_destroy(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
/* native condition variables do not destroy */
if (cond_init)
return;
@@ -204,12 +172,12 @@ static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
pthread_mutex_destroy(&win32_cond->mtx_broadcast);
av_freep(&win32_cond);
cond->Ptr = NULL;
cond->ptr = NULL;
}
static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
static void pthread_cond_broadcast(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int have_waiter;
if (cond_broadcast) {
@@ -238,9 +206,9 @@ static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
}
static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int last_waiter;
if (cond_wait) {
cond_wait(cond, mutex, INFINITE);
@@ -270,9 +238,9 @@ static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mu
return pthread_mutex_lock(mutex);
}
static av_unused void pthread_cond_signal(pthread_cond_t *cond)
static void pthread_cond_signal(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = cond->Ptr;
win32_cond_t *win32_cond = cond->ptr;
int have_waiter;
if (cond_signal) {
cond_signal(cond);
@@ -294,9 +262,8 @@ static av_unused void pthread_cond_signal(pthread_cond_t *cond)
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
}
#endif
static av_unused void w32thread_init(void)
static void w32thread_init(void)
{
#if _WIN32_WINNT < 0x0600
HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));

457
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -2,180 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
since the last major version increase or the API was added.
The last version increases were:
libavcodec: 2014-08-09
libavdevice: 2014-08-09
libavfilter: 2014-08-09
libavformat: 2014-08-09
libavresample: 2014-08-09
libpostproc: 2014-08-09
libswresample: 2014-08-09
libswscale: 2014-08-09
libavutil: 2014-08-09
libavcodec: 2013-03-xx
libavdevice: 2013-03-xx
libavfilter: 2013-12-xx
libavformat: 2013-03-xx
libavresample: 2012-10-05
libpostproc: 2011-04-18
libswresample: 2011-09-19
libswscale: 2011-06-20
libavutil: 2012-10-22
API changes, most recent first:
2014-11-21 - ab922f9 - lavu 54.15.100 - dict.h
Add av_dict_get_string().
2014-11-18 - a54a51c - lavu 54.14.100 - float_dsp.h
Add avpriv_float_dsp_alloc().
2014-11-16 - 6690d4c3 - lavf 56.13.100 - avformat.h
Add AVStream.recommended_encoder_configuration with accessors.
2014-11-16 - bee5844d - lavu 54.13.100 - opt.h
Add av_opt_serialize().
2014-11-16 - eec69332 - lavu 54.12.100 - opt.h
Add av_opt_is_set_to_default().
2014-11-06 - 44fa267 / 5e80fb7 - lavc 56.11.100 / 56.6.0 - vorbis_parser.h
Add a public API for parsing vorbis packets.
2014-10-15 - 17085a0 / 7ea1b34 - lavc 56.7.100 / 56.5.0 - avcodec.h
Replace AVCodecContext.time_base used for decoding
with AVCodecContext.framerate.
2014-10-15 - 51c810e / d565fef1 - lavc 56.6.100 / 56.4.0 - avcodec.h
Add AV_HWACCEL_FLAG_IGNORE_LEVEL flag to av_vdpau_bind_context().
2014-10-13 - da21895 / 2df0c32e - lavc 56.5.100 / 56.3.0 - avcodec.h
Add AVCodecContext.initial_padding. Deprecate the use of AVCodecContext.delay
for audio encoding.
2014-10-08 - bb44f7d / 5a419b2 - lavu 54.10.100 / 54.4.0 - pixdesc.h
Add API to return the name of frame and context color properties.
2014-10-06 - a61899a / e3e158e - lavc 56.3.100 / 56.2.0 - vdpau.h
Add av_vdpau_bind_context(). This function should now be used for creating
(or resetting) a AVVDPAUContext instead of av_vdpau_alloc_context().
2014-10-02 - cdd6f05 - lavc 56.2.100 - avcodec.h
2014-10-02 - cdd6f05 - lavu 54.9.100 - frame.h
Add AV_FRAME_DATA_SKIP_SAMPLES. Add lavc CODEC_FLAG2_SKIP_MANUAL and
AVOption "skip_manual", which makes lavc export skip information via
AV_FRAME_DATA_SKIP_SAMPLES AVFrame side data, instead of skipping and
discarding samples automatically.
2014-10-02 - 0d92b0d - lavu 54.8.100 - avstring.h
Add av_match_list()
2014-09-24 - ac68295 - libpostproc 53.1.100
Add visualization support
2014-09-19 - 6edd6a4 - lavc 56.1.101 - dv_profile.h
deprecate avpriv_dv_frame_profile2(), which was made public by accident.
-------- 8< --------- FFmpeg 2.4 was cut here -------- 8< ---------
2014-08-28 - f30a815 / 9301486 - lavc 56.1.100 / 56.1.0 - avcodec.h
Add AV_PKT_DATA_STEREO3D to export container-level stereo3d information.
2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h
Add AVFormatContext.max_ts_probe.
2014-08-23 - 8fc9bd0 - lavu 54.7.100 - dict.h
AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL arguments are now
freed even on error. This is consistent with the behaviour all users
of it we could find expect.
2014-08-21 - 980a5b0 - lavu 54.6.100 - frame.h motion_vector.h
Add AV_FRAME_DATA_MOTION_VECTORS side data and AVMotionVector structure
2014-08-16 - b7d5e01 - lswr 1.1.100 - swresample.h
Add AVFrame based API
2014-08-16 - c2829dc - lavu 54.4.100 - dict.h
Add av_dict_set_int helper function.
2014-08-13 - c8571c6 / 8ddc326 - lavu 54.3.100 / 54.3.0 - mem.h
Add av_strndup().
2014-08-13 - 2ba4577 / a8c104a - lavu 54.2.100 / 54.2.0 - opt.h
Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
dictionary types being set as options.
2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
the user when events happen in the file/stream.
2014-08-10 - 78eaaa8 / fb1ddcd - lavr 2.1.0 - avresample.h
Add avresample_convert_frame() and avresample_config().
2014-08-10 - 78eaaa8 / fb1ddcd - lavu 54.1.100 / 54.1.0 - error.h
Add AVERROR_INPUT_CHANGED and AVERROR_OUTPUT_CHANGED.
2014-08-08 - 3841f2a / d35b94f - lavc 55.73.102 / 55.57.4 - avcodec.h
Deprecate FF_IDCT_XVIDMMX define and xvidmmx idct option.
Replaced by FF_IDCT_XVID and xvid respectively.
2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
Add avio_feof() and deprecate url_feof().
2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
sws_getContext is not going to be removed in the future.
2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h
reordered_opaque is not going to be removed in the future.
2014-08-02 - 28a2107 - lavu 52.98.100 - pixelutils.h
Add pixelutils API with SAD functions
2014-08-04 - 6017c98 / e9abafc - lavu 52.97.100 / 53.22.0 - pixfmt.h
Add AV_PIX_FMT_YA16 pixel format for 16 bit packed gray with alpha.
2014-08-04 - 4c8bc6f / e96c3b8 - lavu 52.96.101 / 53.21.1 - avstring.h
Rename AV_PIX_FMT_Y400A to AV_PIX_FMT_YA8 to better identify the format.
An alias pixel format and color space name are provided for compatibility.
2014-08-04 - 073c074 / d2962e9 - lavu 52.96.100 / 53.21.0 - pixdesc.h
Support name aliases for pixel formats.
2014-08-03 - 71d008e / 1ef9e83 - lavc 55.72.101 / 55.57.2 - avcodec.h
2014-08-03 - 71d008e / 1ef9e83 - lavu 52.95.100 / 53.20.0 - frame.h
Deprecate AVCodecContext.dtg_active_format and use side-data instead.
2014-08-03 - e680c73 - lavc 55.72.100 - avcodec.h
Add get_pixels() to AVDCT
2014-08-03 - 9400603 / 9f17685 - lavc 55.71.101 / 55.57.1 - avcodec.h
Deprecate unused FF_IDCT_IPP define and ipp avcodec option.
Deprecate unused FF_DEBUG_PTS define and pts avcodec option.
Deprecate unused FF_CODER_TYPE_DEFLATE define and deflate avcodec option.
Deprecate unused FF_DCT_INT define and int avcodec option.
Deprecate unused avcodec option scenechange_factor.
2014-07-30 - ba3e331 - lavu 52.94.100 - frame.h
Add av_frame_side_data_name()
2014-07-29 - 80a3a66 / 3a19405 - lavf 56.01.100 / 56.01.0 - avformat.h
Add mime_type field to AVProbeData, which now MUST be initialized in
order to avoid uninitialized reads of the mime_type pointer, likely
leading to crashes.
Typically, this means you will do 'AVProbeData pd = { 0 };' instead of
'AVProbeData pd;'.
2014-07-29 - 31e0b5d / 69e7336 - lavu 52.92.100 / 53.19.0 - avstring.h
Make name matching function from lavf public as av_match_name().
2014-07-28 - 2e5c8b0 / c5fca01 - lavc 55.71.100 / 55.57.0 - avcodec.h
Add AV_CODEC_PROP_REORDER to mark codecs supporting frame reordering.
2014-07-27 - ff9a154 - lavf 55.50.100 - avformat.h
New field int64_t probesize2 instead of deprecated
field int probesize.
2014-07-27 - 932ff70 - lavc 55.70.100 - avdct.h
Add AVDCT / avcodec_dct_alloc() / avcodec_dct_init().
2014-07-23 - 8a4c086 - lavf 55.49.100 - avio.h
Add avio_read_to_bprint()
-------- 8< --------- FFmpeg 2.3 was cut here -------- 8< ---------
2014-07-14 - 62227a7 - lavf 55.47.100 - avformat.h
Add av_stream_get_parser()
@@ -200,7 +39,7 @@ API changes, most recent first:
Increase FF_INPUT_BUFFER_PADDING_SIZE to 32 due to some corner cases needing
it
2014-06-10 - 5482780 - lavf 55.43.100 - avformat.h
2014-06-10 - xxxxxxx - lavf 55.43.100 - avformat.h
New field int64_t max_analyze_duration2 instead of deprecated
int max_analyze_duration.
@@ -220,28 +59,28 @@ API changes, most recent first:
2014-05-29 - bdb2e80 / b2d4565 - lavr 1.3.0 - avresample.h
Add avresample_max_output_samples
2014-05-28 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
2014-05-24 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
Add strict_std_compliance and related AVOptions to support experimental
muxing.
2014-05-26 - 55cc60c - lavu 52.87.100 - threadmessage.h
2014-05-26 - xxxxxxx - lavu 52.87.100 - threadmessage.h
Add thread message queue API.
2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h
Add format_probesize to AVFormatContext.
2014-05-20 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
2014-05-19 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
Add av_stream_get_side_data() to access stream-level side data
in the same way as av_packet_get_side_data().
2014-05-20 - 7336e39 - lavu 52.86.100 - fifo.h
2014-05-xx - xxxxxxx - lavu 52.86.100 - fifo.h
Add av_fifo_alloc_array() function.
2014-05-19 - ef1d4ee / bddd8cb - lavu 52.85.100 / 53.15.0 - frame.h, display.h
Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level
spatial rendering on video frames for proper display.
2014-05-19 - ef1d4ee / bddd8cb - lavc 55.64.100 / 55.53.0 - avcodec.h
2014-05-xx - xxxxxxx - lavc 55.64.100 / 55.53.0 - avcodec.h
Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level
spatial rendering on video frames for proper display.
@@ -257,7 +96,7 @@ API changes, most recent first:
Add av_gettime_relative() av_gettime_relative_is_monotonic()
2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h
Add AVFMT_FLAG_BITEXACT flag. Muxers now use it instead of checking
Add AVMFT_FLAG_BITEXACT flag. Muxers now use it instead of checking
CODEC_FLAG_BITEXACT on the first stream.
2014-05-15 - 96cb4c8 - lswr 0.19.100 - swresample.h
@@ -266,10 +105,10 @@ API changes, most recent first:
2014-05-11 - 14aef38 / 66e6c8a - lavu 52.83.100 / 53.14.0 - pixfmt.h
Add AV_PIX_FMT_VDA for new-style VDA acceleration.
2014-05-07 - 351f611 - lavu 52.82.100 - fifo.h
2014-05-xx - xxxxxxx - lavu 52.82.0 - fifo.h
Add av_fifo_freep() function.
2014-05-02 - ba52fb11 - lavu 52.81.100 - opt.h
2014-05-02 - ba52fb11 - lavu 52.81.0 - opt.h
Add av_opt_set_dict2() function.
2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h
@@ -288,14 +127,10 @@ API changes, most recent first:
Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing
reference-counted frames to encoders.
2014-04-30 - 617e866 - lavu 52.81.100 - pixdesc.h
Add av_find_best_pix_fmt_of_2(), av_get_pix_fmt_loss()
Deprecate avcodec_get_pix_fmt_loss(), avcodec_find_best_pix_fmt_of_2()
2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h
Add AVCodecDescriptor.mime_types field.
2014-04-29 - b804eb4 - lavu 52.80.100 - hash.h
2014-04-29 - xxxxxxx - lavu 52.80.0 - hash.h
Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64().
2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h
@@ -307,7 +142,7 @@ API changes, most recent first:
2014-04-17 - a8d01a7 / 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h
Add AV_CRC_16_ANSI_LE crc variant.
2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
2014-04-XX - xxxxxxx - lavf xx.xx.1xx - avformat.h
Add av_format_inject_global_side_data()
2014-04-12 - 4f698be - lavu 52.76.100 - log.h
@@ -353,9 +188,6 @@ API changes, most recent first:
Give the name AVPacketSideData to the previously anonymous struct used for
AVPacket.side_data.
-------- 8< --------- FFmpeg 2.2 was cut here -------- 8< ---------
2014-03-18 - 37c07d4 - lsws 2.5.102
Make gray16 full-scale.
@@ -387,7 +219,7 @@ API changes, most recent first:
2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h
Deprecate unused AV_OPT_FLAG_METADATA.
2014-02-16 - 81c3f81 - lavd 55.10.100 - avdevice.h
2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
Add avdevice_list_devices() and avdevice_free_list_devices()
2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
@@ -428,7 +260,7 @@ API changes, most recent first:
2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h
Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
2014-01-19 - 3532dd5 - lavu 52.63.100 - rational.h
2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
Add av_make_q() function.
2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
@@ -498,9 +330,6 @@ API changes, most recent first:
2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
-------- 8< --------- FFmpeg 2.1 was cut here -------- 8< ---------
2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h
Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
@@ -573,9 +402,6 @@ API changes, most recent first:
Add avcodec_chroma_pos_to_enum()
Add avcodec_enum_to_chroma_pos()
-------- 8< --------- FFmpeg 2.0 was cut here -------- 8< ---------
2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
Deprecate avfilter_graph_parse() in favor of the equivalent
avfilter_graph_parse_ptr().
@@ -648,9 +474,6 @@ API changes, most recent first:
2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
-------- 8< --------- FFmpeg 1.2 was cut here -------- 8< ---------
2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
Add av_escape() and av_bprint_escape() API.
@@ -663,9 +486,6 @@ API changes, most recent first:
2013-01-01 - 2eb2e17 - lavfi 3.34.100
Add avfilter_get_audio_buffer_ref_from_arrays_channels.
-------- 8< --------- FFmpeg 1.1 was cut here -------- 8< ---------
2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
Add AVFilterLink.channels, avfilter_link_get_channels()
and avfilter_ref_get_channels().
@@ -711,9 +531,6 @@ API changes, most recent first:
Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD
and LIBSWRESAMPLE_IDENT symbols.
-------- 8< --------- FFmpeg 1.0 was cut here -------- 8< ---------
2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h
Add av_small_strptime() time parsing function.
@@ -918,9 +735,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
avresample_read() are now uint8_t** instead of void**.
Libavresample is now stable.
2012-09-26 - 3ba0dab7 / 1384df64 - lavf 54.29.101 / 56.06.3 - avformat.h
Add AVFormatContext.avoid_negative_ts.
2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h
Add avcodec_free_frame(). This function must now
be used for freeing an AVFrame.
@@ -1135,9 +949,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0
Add a new installed header -- libavfilter/version.h -- with version macros.
-------- 8< --------- FFmpeg 0.9 was cut here -------- 8< ---------
2011-12-08 - a502939 - lavfi 2.52.0
Add av_buffersink_poll_frame() to buffersink.h.
@@ -1347,13 +1158,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h
Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum.
-------- 8< --------- FFmpeg 0.7 was cut here -------- 8< ---------
-------- 8< --------- FFmpeg 0.8 was cut here -------- 8< ---------
2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h
Add layout negotiation fields and helper functions.
@@ -2031,9 +1835,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2010-06-02 - 7e566bb - lavc 52.73.0 - av_get_codec_tag_string()
Add av_get_codec_tag_string().
-------- 8< --------- FFmpeg 0.6 was cut here -------- 8< ---------
2010-06-01 - 2b99142 - lsws 0.11.0 - convertPalette API
Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24().
@@ -2051,6 +1852,10 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
2010-05-09 - b6bc205 - lavfi 1.20.0 - AVFilterPicRef
Add interlaced and top_field_first fields to AVFilterPicRef.
------------------------------8<-------------------------------------
0.6 branch was cut here
----------------------------->8--------------------------------------
2010-05-01 - 8e2ee18 - lavf 52.62.0 - probe function
Add av_probe_input_format2 to API, it allows ignoring probe
results below given score and returns the actual probe score.

View File

@@ -31,7 +31,7 @@ PROJECT_NAME = FFmpeg
# This could be handy for archiving the generated documentation or
# if some version control system is used.
PROJECT_NUMBER = 2.5.10
PROJECT_NUMBER = 2.3.2
# With the PROJECT_LOGO tag one can specify a logo or icon that is included
# in the documentation. The maximum height of the logo should not exceed 55

View File

@@ -38,9 +38,7 @@ DOCS = $(DOCS-yes)
DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec
DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
@@ -82,25 +80,14 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%-all.html: TAG = HTML
ifdef HAVE_MAKEINFO_HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
else
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%-all.html: TAG = HTML
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
endif
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)

View File

@@ -13,16 +13,7 @@ bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
The ff* tools have a -bsf option applied per stream, taking a
comma-separated list of filters, whose parameters follow the filter
name after a '='.
@example
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
@end example
Below is a description of the currently available bitstream filters,
with their parameters, if any.
Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
@@ -144,16 +135,9 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
Parameters:
A numeral string, whose value is related to how often output bytes will
be modified. Therefore, values below or equal to 0 are forbidden, and
the lower the more frequent bytes will be modified, with 1 meaning
every byte is modified.
@example
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
ffmpeg -i INPUT -c copy -bsf noise output.mkv
@end example
applies the modification to every byte.
@section remove_extra

File diff suppressed because one or more lines are too long

View File

@@ -71,9 +71,7 @@ Force low delay.
@item global_header
Place global headers in extradata instead of every keyframe.
@item bitexact
Only write platform-, build- and time-independent data. (except (I)DCT).
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
Use only bitexact stuff (except (I)DCT).
@item aic
Apply H263 advanced intra coding / mpeg4 ac prediction.
@item cbp
@@ -500,8 +498,6 @@ threading operations
@item vismv @var{integer} (@emph{decoding,video})
Visualize motion vectors (MVs).
This option is deprecated, see the codecview filter instead.
Possible values:
@table @samp
@item pf
@@ -801,9 +797,6 @@ Frame data might be split into multiple chunks.
Show all frames before the first keyframe.
@item skiprd
Deprecated, use mpegvideo private options instead.
@item export_mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@end table
@item error @var{integer} (@emph{encoding,video})
@@ -1116,19 +1109,6 @@ Interlaced video, bottom coded first, top displayed first
Set to 1 to disable processing alpha (transparency). This works like the
@samp{gray} flag in the @option{flags} option which skips chroma information
instead of alpha. Default is 0.
@item codec_whitelist @var{list} (@emph{input})
"," separated List of allowed decoders. By default all are allowed.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@end table
@c man end CODEC OPTIONS

View File

@@ -163,9 +163,6 @@ Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
An FFmpeg native decoder for Opus exists, so users can decode Opus
without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@@ -190,15 +187,6 @@ The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@item ifo_palette
Specify the IFO file from which the global palette is obtained.
(experimental)
@item forced_subs_only
Only decode subtitle entries marked as forced. Some titles have forced
and non-forced subtitles in the same track. Setting this flag to @code{1}
will only keep the forced subtitles. Default value is @code{0}.
@end table
@section libzvbi-teletext

View File

@@ -29,26 +29,6 @@ the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section apng
Animated Portable Network Graphics demuxer.
This demuxer is used to demux APNG files.
All headers, but the PNG signature, up to (but not including) the first
fcTL chunk are transmitted as extradata.
Frames are then split as being all the chunks between two fcTL ones, or
between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
Ignore the loop variable in the file if set.
@item -max_fps @var{int}
Maximum framerate in frames per second (0 for no limit).
@item -default_fps @var{int}
Default framerate in frames per second when none is specified in the file
(0 meaning as fast as possible).
@end table
@section asf
Advanced Systems Format demuxer.
@@ -194,40 +174,6 @@ See @url{http://quvi.sourceforge.net/} for more information.
FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
enabled.
@section gif
Animated GIF demuxer.
It accepts the following options:
@table @option
@item min_delay
Set the minimum valid delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 2.
@item default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
@item ignore_loop
GIF files can contain information to loop a certain number of times (or
infinitely). If @option{ignore_loop} is set to 1, then the loop setting
from the input will be ignored and looping will not occur. If set to 0,
then looping will occur and will cycle the number of times according to
the GIF. Default value is 1.
@end table
For example, with the overlay filter, place an infinitely looping GIF
over another video:
@example
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
Note that in the above example the shortest option for overlay filter is
used to end the output video at the length of the shortest input file,
which in this case is @file{input.mp4} as the GIF in this example loops
infinitely.
@section image2
Image file demuxer.

View File

@@ -323,12 +323,9 @@ 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
@item
Make sure the author of the commit is set correctly. (see git commit --author)

View File

@@ -1032,7 +1032,7 @@ configuration. You need to explicitly configure the build with
@subsection Option Mapping
Most libopus options are modelled after the @command{opusenc} utility from
Most libopus options are modeled after the @command{opusenc} utility from
opus-tools. The following is an option mapping chart describing options
supported by the libopus wrapper, and their @command{opusenc}-equivalent
in parentheses.
@@ -1330,7 +1330,7 @@ ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
@section libvpx
VP8/VP9 format supported through libvpx.
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libvpx}.
@@ -1442,9 +1442,6 @@ g_lag_in_frames
@item vp8flags error_resilient
g_error_resilient
@item aq_mode
@code{VP9E_SET_AQ_MODE}
@end table
For more information about libvpx see:
@@ -1745,10 +1742,6 @@ Enable calculation and printing SSIM stats after the encoding.
Enable the use of Periodic Intra Refresh instead of IDR frames when set
to 1.
@item avcintra-class (@emph{class})
Configure the encoder to generate AVC-Intra.
Valid values are 50,100 and 200
@item bluray-compat (@emph{bluray-compat})
Configure the encoder to be compatible with the bluray standard.
It is a shorthand for setting "bluray-compat=1 force-cfr=1".
@@ -1886,34 +1879,6 @@ no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
Encoding ffpresets for common usages are provided so they can be used with the
general presets system (e.g. passing the @option{pre} option).
@section libx265
x265 H.265/HEVC encoder wrapper.
This encoder requires the presence of the libx265 headers and library
during configuration. You need to explicitly configure the build with
@option{--enable-libx265}.
@subsection Options
@table @option
@item preset
Set the x265 preset.
@item tune
Set the x265 tune parameter.
@item x265-params
Set x265 options using a list of @var{key}=@var{value} couples separated
by ":". See @command{x265 --help} for a list of options.
For example to specify libx265 encoding options with @option{-x265-params}:
@example
ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
@end example
@end table
@section libxvid
Xvid MPEG-4 Part 2 encoder wrapper.

View File

@@ -14,7 +14,6 @@ LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \
filtering_video \
filtering_audio \
metadata \
@@ -29,7 +28,6 @@ OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm

View File

@@ -288,7 +288,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
@@ -305,15 +304,15 @@ static void audio_decode_example(const char *outfilename, const char *filename)
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(c->sample_fmt);
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i=0; i<decoded_frame->nb_samples; i++)
for (ch=0; ch<c->channels; ch++)
fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
@@ -651,7 +650,7 @@ int main(int argc, char **argv)
video_encode_example("test.h264", AV_CODEC_ID_H264);
} else if (!strcmp(output_type, "mp2")) {
audio_encode_example("test.mp2");
audio_decode_example("test.pcm", "test.mp2");
audio_decode_example("test.sw", "test.mp2");
} else if (!strcmp(output_type, "mpg")) {
video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
video_decode_example("test%02d.pgm", "test.mpg");

View File

@@ -1,185 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
* Copyright (c) 2014 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavutil/motion_vector.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL;
static AVStream *video_stream = NULL;
static const char *src_filename = NULL;
static int video_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int decode_packet(int *got_frame, int cached)
{
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
int i;
AVFrameSideData *sd;
video_frame_count++;
sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
if (sd) {
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags);
}
}
}
}
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
exit(1);
}
src_filename = argv[1];
av_register_all();
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
}
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!video_stream) {
fprintf(stderr, "Could not find video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
end:
avcodec_close(video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
return ret < 0;
}

View File

@@ -45,7 +45,6 @@
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"

View File

@@ -90,7 +90,6 @@ static int init_filters(const char *filters_descr)
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();
@@ -103,7 +102,7 @@ static int init_filters(const char *filters_descr)
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",

View File

@@ -55,7 +55,6 @@ typedef struct OutputStream {
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
@@ -63,7 +62,6 @@ typedef struct OutputStream {
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
@@ -94,7 +92,6 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
@@ -118,24 +115,8 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
(*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->channels = 2;
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:
@@ -149,9 +130,8 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
@@ -178,47 +158,19 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
/**************************************************************/
/* 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;
int samples_count;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
struct SwrContext *swr_ctx = NULL;
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, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->st->codec;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
@@ -230,52 +182,84 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = av_frame_alloc();
if (!ost->frame)
exit(1);
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);
ost->frame->sample_rate = c->sample_rate;
ost->frame->format = AV_SAMPLE_FMT_S16;
ost->frame->channel_layout = c->channel_layout;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
ost->frame->nb_samples = 10000;
else
ost->frame->nb_samples = c->frame_size;
ost->tmp_frame = av_frame_alloc();
if (!ost->frame)
exit(1);
ost->tmp_frame->sample_rate = c->sample_rate;
ost->tmp_frame->format = c->sample_fmt;
ost->tmp_frame->channel_layout = c->channel_layout;
ost->tmp_frame->nb_samples = ost->frame->nb_samples;
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
ret = av_frame_get_buffer(ost->frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate an audio frame.\n");
exit(1);
}
ret = av_frame_get_buffer(ost->tmp_frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate an audio frame.\n");
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];
int j, i, v, ret;
int16_t *q = (int16_t*)ost->frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
for (j = 0; j < ost->frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->st->codec->channels; i++)
*q++ = v;
@@ -283,10 +267,10 @@ static AVFrame *get_audio_frame(OutputStream *ost)
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
ost->frame->pts = ost->next_pts;
ost->next_pts += ost->frame->nb_samples;
return frame;
return ost->frame;
}
/*
@@ -309,31 +293,27 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* 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,
ret = swr_convert(swr_ctx,
ost->tmp_frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame = ost->tmp_frame;
} else {
dst_nb_samples = frame->nb_samples;
}
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
@@ -380,17 +360,13 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
{
int ret;
AVCodecContext *c = ost->st->codec;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
@@ -547,7 +523,6 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
@@ -563,12 +538,11 @@ int main(int argc, char **argv)
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc < 2) {
if (argc != 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
@@ -580,9 +554,6 @@ int main(int argc, char **argv)
}
filename = argv[1];
if (argc > 3 && !strcmp(argv[2], "-flags")) {
av_dict_set(&opt, argv[2]+1, argv[3], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -611,10 +582,10 @@ int main(int argc, char **argv)
/* 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);
open_video(oc, video_codec, &video_st);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
open_audio(oc, audio_codec, &audio_st);
av_dump_format(oc, 0, filename, 1);
@@ -629,7 +600,7 @@ int main(int argc, char **argv)
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));

View File

@@ -199,7 +199,8 @@ int main(int argc, char **argv)
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (dst_file)
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);

View File

@@ -132,7 +132,8 @@ int main(int argc, char **argv)
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
if (dst_file)
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);

View File

@@ -52,7 +52,7 @@
* @param error Error code to be converted
* @return Corresponding error text (not thread-safe)
*/
static const char *get_error_text(const int error)
static char *const get_error_text(const int error)
{
static char error_buffer[255];
av_strerror(error, error_buffer, sizeof(error_buffer));
@@ -306,7 +306,7 @@ static int decode_audio_frame(AVFrame *frame,
/** Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
/** If we are the the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {

View File

@@ -116,10 +116,6 @@ static int open_output_file(const char *filename)
|| 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, "Neccessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
@@ -389,9 +385,17 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
enc_pkt.duration = av_rescale_q(enc_pkt.duration,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
@@ -505,9 +509,14 @@ int main(int argc, char **argv)
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base);
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
@@ -529,9 +538,14 @@ int main(int argc, char **argv)
}
} 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);
packet.dts = av_rescale_q_rnd(packet.dts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
packet.pts = av_rescale_q_rnd(packet.pts,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base,
AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)

View File

@@ -298,7 +298,7 @@ FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
video by merely concatenating the files containing them.
Hence you may concatenate your multimedia files by first transcoding them to

View File

@@ -339,7 +339,7 @@ ffmpeg -i in.avi -metadata title="my title" out.flv
To set the language of the first audio stream:
@example
ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
@end example
@item -target @var{type} (@emph{output})
@@ -360,7 +360,7 @@ ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dframes @var{number} (@emph{output})
Set the number of data frames to output. This is an alias for @code{-frames:d}.
Set the number of data frames to record. This is an alias for @code{-frames:d}.
@item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
Stop writing to the stream after @var{framecount} frames.
@@ -467,15 +467,12 @@ attachments.
@table @option
@item -vframes @var{number} (@emph{output})
Set the number of video frames to output. This is an alias for @code{-frames:v}.
Set the number of video frames to record. This is an alias for @code{-frames:v}.
@item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
Set frame rate (Hz value, fraction or abbreviation).
As an input option, ignore any timestamps stored in the file and instead
generate timestamps assuming constant frame rate @var{fps}.
This is not the same as the @option{-framerate} option used for some input formats
like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
If in doubt use @option{-framerate} instead of the input option @option{-r}.
As an output option, duplicate or drop input frames to achieve constant output
frame rate @var{fps}.
@@ -692,7 +689,7 @@ If this option is not specified, the default adapter is used.
@table @option
@item -aframes @var{number} (@emph{output})
Set the number of audio frames to output. This is an alias for @code{-frames:a}.
Set the number of audio frames to record. This is an alias for @code{-frames:a}.
@item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
Set the audio sampling frequency. For output streams it is set by
default to the frequency of the corresponding input stream. For input
@@ -824,11 +821,6 @@ To map all the streams except the second audio, use negative mappings
ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
@end example
To pick the English audio stream:
@example
ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
@end example
Note that using this option disables the default mappings for this output file.
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
@@ -1016,12 +1008,6 @@ processing (e.g. in case the format option @option{avoid_negative_ts}
is enabled) the output timestamps may mismatch with the input
timestamps even when this option is selected.
@item -start_at_zero
When used with @option{copyts}, shift input timestamps so they start at zero.
This means that using e.g. @code{-ss 50} will make output timestamps start at
50 seconds, regardless of what timestamp the input file started at.
@item -copytb @var{mode}
Specify how to set the encoder timebase when stream copying. @var{mode} is an
integer numeric value, and can assume one of the following values:
@@ -1464,11 +1450,11 @@ ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
You can put many streams of the same type in the output:
@example
ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
@end example
The resulting output file @file{test12.nut} will contain the first four streams
from the input files in reverse order.
The resulting output file @file{test12.avi} will contain first four streams from
the input file in reverse order.
@item
To force CBR video output:

View File

@@ -37,14 +37,10 @@ Force displayed height.
Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
@item -fs
Start in fullscreen mode.
@item -an
Disable audio.
@item -vn
Disable video.
@item -sn
Disable subtitles.
@item -ss @var{pos}
Seek to a given position in seconds.
@item -t @var{duration}
@@ -113,10 +109,15 @@ duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
@item -bug
Work around bugs.
@item -fast
Non-spec-compliant optimizations.
@item -genpts
Generate pts.
@item -rtp_tcp
Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
if you are streaming with the RTSP protocol.
@item -sync @var{type}
Set the master clock to audio (@code{type=audio}), video
(@code{type=video}) or external (@code{type=ext}). Default is audio. The
@@ -125,8 +126,7 @@ players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
@item -threads @var{count}
Set the thread count. By default, @command{ffplay} automatically detects a
suitable number of threads to use.
Set the thread count.
@item -ast @var{audio_stream_number}
Select the desired audio stream number, counting from 0. The number
refers to the list of all the input audio streams. If it is greater
@@ -164,20 +164,8 @@ Force a specific video decoder.
Force a specific subtitle decoder.
@item -autorotate
Automatically rotate the video according to presentation metadata. Enabled by
default, use @option{-noautorotate} to disable it.
@item -framedrop
Drop video frames if video is out of sync. Enabled by default if the master
clock is not set to video. Use this option to enable frame dropping for all
master clock sources, use @option{-noframedrop} to disable it.
@item -infbuf
Do not limit the input buffer size, read as much data as possible from the
input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use @option{-noinfbuf} to disable it.
Automatically rotate the video according to presentation metadata. Set by
default, use -noautorotate to disable.
@end table
@section While playing

View File

@@ -119,10 +119,6 @@ Show payload data, as a hexadecimal and ASCII dump. Coupled with
The dump is printed as the "data" field. It may contain newlines.
@item -show_data_hash @var{algorithm}
Show a hash of payload data, for packets with @option{-show_packets} and for
codec extradata with @option{-show_streams}.
@item -show_error
Show information about the error found when trying to probe the input.
@@ -184,7 +180,7 @@ format : stream=codec_type
To show all the tags in the stream and format sections:
@example
stream_tags : format_tags
format_tags : format_tags
@end example
To show only the @code{title} tag (if available) in the stream
@@ -321,12 +317,6 @@ Show information related to program and library versions. This is the
equivalent of setting both @option{-show_program_version} and
@option{-show_library_versions} options.
@item -show_pixel_formats
Show information about all pixel formats supported by FFmpeg.
Pixel format information for each format is printed within a section
with name "PIXEL_FORMAT".
@item -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.

View File

@@ -10,10 +10,8 @@
<xsd:sequence>
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="pixel_formats" type="ffprobe:pixelFormatsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
@@ -37,16 +35,6 @@
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsAndFramesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
@@ -62,13 +50,11 @@
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
@@ -105,16 +91,6 @@
<xsd:attribute name="repeat_pict" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
@@ -166,7 +142,6 @@
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
@@ -178,9 +153,6 @@
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<!-- audio attributes -->
@@ -200,7 +172,6 @@
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="max_bit_rate" type="xsd:int"/>
<xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
@@ -292,45 +263,4 @@
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatFlagsType">
<xsd:attribute name="big_endian" type="xsd:int" use="required"/>
<xsd:attribute name="palette" type="xsd:int" use="required"/>
<xsd:attribute name="bitstream" type="xsd:int" use="required"/>
<xsd:attribute name="hwaccel" type="xsd:int" use="required"/>
<xsd:attribute name="planar" type="xsd:int" use="required"/>
<xsd:attribute name="rgb" type="xsd:int" use="required"/>
<xsd:attribute name="pseudopal" type="xsd:int" use="required"/>
<xsd:attribute name="alpha" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentType">
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="bit_depth" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentsType">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatType">
<xsd:sequence>
<xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="nb_components" type="xsd:int" use="required"/>
<xsd:attribute name="log2_chroma_w" type="xsd:int"/>
<xsd:attribute name="log2_chroma_h" type="xsd:int"/>
<xsd:attribute name="bits_per_pixel" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatsType">
<xsd:sequence>
<xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

View File

@@ -1,11 +1,11 @@
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
HTTPPort 8090
Port 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
HTTPBindAddress 0.0.0.0
BindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the

View File

@@ -66,12 +66,12 @@ http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
where @var{ffserver_ip_address} is the IP address of the machine where
@command{ffserver} is installed, @var{http_port} is the port number of
the HTTP server (configured through the @option{HTTPPort} option), and
the HTTP server (configured through the @option{Port} option), and
@var{feed_name} is the name of the corresponding feed defined in the
configuration file.
Each feed is associated to a file which is stored on disk. This stored
file is used to send pre-recorded data to a player as fast as
file is used to allow to send pre-recorded data to a player as fast as
possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by
@@ -101,7 +101,7 @@ http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}
the configuration file. @var{options} is a list of options specified
after the URL which affects how the stream is served by
@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
and RTSP ports configured with the options @var{HTTPPort} and
and RTSP ports configured with the options @var{Port} and
@var{RTSPPort} respectively.
In case the stream is associated to a feed, the encoding parameters
@@ -203,9 +203,11 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
transfer the entire file before starting to play.
The same is true of AVI files.
You should edit the @file{ffserver.conf} file to suit your needs (in
terms of frame rates etc). Then install @command{ffserver} and
@command{ffmpeg}, write a script to start them up, and off you go.
@section What happens next?
You should edit the ffserver.conf file to suit your needs (in terms of
frame rates etc). Then install ffserver and ffmpeg, write a script to start
them up, and off you go.
@section What else can it do?
@@ -352,29 +354,20 @@ allow everybody else.
@section Global options
@table @option
@item HTTPPort @var{port_number}
@item Port @var{port_number}
@item RTSPPort @var{port_number}
@var{HTTPPort} sets the HTTP server listening TCP port number,
@var{RTSPPort} sets the RTSP server listening TCP port number.
@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
You must select a different port from your standard HTTP web server if
it is running on the same computer.
Set TCP port number on which the HTTP/RTSP server is listening. You
must select a different port from your standard HTTP web server if it
is running on the same computer.
If not specified, no corresponding server will be created.
@item HTTPBindAddress @var{ip_address}
@item BindAddress @var{ip_address}
@item RTSPBindAddress @var{ip_address}
Set address on which the HTTP/RTSP server is bound. Only useful if you
have several network interfaces.
@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
deprecated.
@item MaxHTTPConnections @var{n}
Set number of simultaneous HTTP connections that can be handled. It
has to be defined @emph{before} the @option{MaxClients} parameter,
@@ -408,12 +401,6 @@ ignored, and the log is written to standard output.
Set no-daemon mode. This option is currently ignored since now
@command{ffserver} will always work in no-daemon mode, and is
deprecated.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the all streams or not.
Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
The lastest occurrence overrides previous if multiple definitions.
@end table
@section Feed section
@@ -577,11 +564,6 @@ deprecated in favor of @option{Metadata}.
@item Metadata @var{key} @var{value}
Set metadata value on the output stream.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the stream or not.
Default is @var{UseDefaults} unless disabled globally.
@item NoAudio
@item NoVideo
Suppress audio/video.
@@ -600,9 +582,8 @@ Set sampling frequency for audio. When using low bitrates, you should
lower this frequency to 22050 or 11025. The supported frequencies
depend on the selected audio codec.
@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
Set generic or private option for audio stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
Set generic option for audio stream.
@item AVPresetAudio @var{preset} (@emph{encoding,audio})
Set preset for audio stream.
@@ -679,9 +660,8 @@ Set video @option{qdiff} encoding option.
@item DarkMask @var{float} (@emph{encoding,video})
Set @option{lumi_mask}/@option{dark_mask} encoding options.
@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
Set generic or private option for video stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
Set generic option for video stream.
@item AVPresetVideo @var{preset} (@emph{encoding,video})
Set preset for video stream.

View File

@@ -3,7 +3,7 @@ representing a number as input, which may be followed by one of the SI
unit prefixes, for example: 'K', 'M', or 'G'.
If 'i' is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiples, which are based on
interpreted as a unit prefix for binary multiplies, which are based on
powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
prefix multiplies the value by 8. This allows using, for example:
'KB', 'MiB', 'G' and 'B' as number suffixes.
@@ -46,13 +46,6 @@ in the program with the id @var{program_id}. Otherwise, it matches all streams i
program.
@item #@var{stream_id} or i:@var{stream_id}
Match the stream by stream id (e.g. PID in MPEG-TS container).
@item m:@var{key}[:@var{value}]
Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value.
Note that in @command{ffmpeg}, matching by metadata will only work properly for
input files.
@end table
@section Generic options
@@ -103,10 +96,7 @@ Print detailed information about the filter name @var{filter_name}. Use the
Show version.
@item -formats
Show available formats (including devices).
@item -devices
Show available devices.
Show available formats.
@item -codecs
Show all codecs known to libavcodec.
@@ -141,22 +131,6 @@ Show channel names and standard channel layouts.
@item -colors
Show recognized color names.
@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
Show autodetected sources of the intput device.
Some devices may provide system-dependent source names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
@example
ffmpeg -sources pulse,server=192.168.0.4
@end example
@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
Show autodetected sinks of the output device.
Some devices may provide system-dependent sink names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
@example
ffmpeg -sinks pulse,server=192.168.0.4
@end example
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
Set the logging level used by the library.
Adding "repeat+" indicates that repeated log output should not be compressed
@@ -253,14 +227,10 @@ Possible flags for this option are:
@item sse4.1
@item sse4.2
@item avx
@item avx2
@item xop
@item fma3
@item fma4
@item 3dnow
@item 3dnowext
@item bmi1
@item bmi2
@item cmov
@end table
@item ARM
@@ -271,13 +241,6 @@ Possible flags for this option are:
@item vfp
@item vfpv3
@item neon
@item setend
@end table
@item AArch64
@table @samp
@item armv8
@item vfp
@item neon
@end table
@item PowerPC
@table @samp

View File

@@ -282,10 +282,6 @@ sequential number of the input frame, starting from 0
@item pos
the position in the file of the input frame, NAN if unknown
@item w
@item h
width and height of the input frame if video
@end table
Additionally, these filters support an @option{enable} command that can be used
@@ -313,6 +309,41 @@ build.
Below is a description of the currently available audio filters.
@section aconvert
Convert the input audio format to the specified formats.
@emph{This filter is deprecated. Use @ref{aformat} instead.}
The filter accepts a string of the form:
"@var{sample_format}:@var{channel_layout}".
@var{sample_format} specifies the sample format, and can be a string or the
corresponding numeric value defined in @file{libavutil/samplefmt.h}. Use 'p'
suffix for a planar sample format.
@var{channel_layout} specifies the channel layout, and can be a string
or the corresponding number value defined in @file{libavutil/channel_layout.h}.
The special parameter "auto", signifies that the filter will
automatically select the output format depending on the output filter.
@subsection Examples
@itemize
@item
Convert input to float, planar, stereo:
@example
aconvert=fltp:stereo
@end example
@item
Convert input to unsigned 8-bit, automatically select out channel layout:
@example
aconvert=u8:auto
@end example
@end itemize
@section adelay
Delay one or more audio channels.
@@ -460,7 +491,7 @@ aeval=val(ch)/2:c=same
@item
Invert phase of the second channel:
@example
aeval=val(0)|-val(1)
eval=val(0)|-val(1)
@end example
@end itemize
@@ -486,21 +517,27 @@ volume as the input audio, at the end of the fade-out transition
the output audio will be silence. Default is 44100.
@item start_time, st
Specify the start time of the fade effect. Default is 0.
The value must be specified as a time duration; see
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
If set this option is used instead of @var{start_sample}.
Specify time for starting to apply the fade effect. Default is 0.
The accepted syntax is:
@example
[-]HH[:MM[:SS[.m...]]]
[-]S+[.m...]
@end example
See also the function @code{av_parse_time()}.
If set this option is used instead of @var{start_sample} one.
@item duration, d
Specify the duration of the fade effect. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
Specify the duration for which the fade effect has to last. Default is 0.
The accepted syntax is:
@example
[-]HH[:MM[:SS[.m...]]]
[-]S+[.m...]
@end example
See also the function @code{av_parse_time()}.
At the end of the fade-in effect the output audio will have the same
volume as the input audio, at the end of the fade-out transition
the output audio will be silence.
By default the duration is determined by @var{nb_samples}.
If set this option is used instead of @var{nb_samples}.
If set this option is used instead of @var{nb_samples} one.
@item curve
Set curve for fade transition.
@@ -705,58 +742,8 @@ Pass the audio source unchanged to the output.
@section apad
Pad the end of an audio stream with silence.
This can be used together with @command{ffmpeg} @option{-shortest} to
extend audio streams to the same length as the video stream.
A description of the accepted options follows.
@table @option
@item packet_size
Set silence packet size. Default value is 4096.
@item pad_len
Set the number of samples of silence to add to the end. After the
value is reached, the stream is terminated. This option is mutually
exclusive with @option{whole_len}.
@item whole_len
Set the minimum total number of samples in the output audio stream. If
the value is longer than the input audio length, silence is added to
the end, until the value is reached. This option is mutually exclusive
with @option{pad_len}.
@end table
If neither the @option{pad_len} nor the @option{whole_len} option is
set, the filter will add silence to the end of the input stream
indefinitely.
@subsection Examples
@itemize
@item
Add 1024 samples of silence to the end of the input:
@example
apad=pad_len=1024
@end example
@item
Make sure the audio output will contain at least 10000 samples, pad
the input with silence if required:
@example
apad=whole_len=10000
@end example
@item
Use @command{ffmpeg} to pad the audio input with silence, so that the
video stream will always result the shortest and will be converted
until the end in the output file when using the @option{shortest}
option:
@example
ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
@end example
@end itemize
Pad the end of a audio stream with silence, this can be used together with
-shortest to extend audio streams to the same length as the video stream.
@section aphaser
Add a phasing effect to the input audio.
@@ -875,7 +862,7 @@ The input audio is not modified.
The shown line contains a sequence of key/value pairs of the form
@var{key}:@var{value}.
The following values are shown in the output:
It accepts the following parameters:
@table @option
@item n
@@ -1083,9 +1070,9 @@ The number of the first sample that should be output.
The number of the first sample that should be dropped.
@end table
@option{start}, @option{end}, and @option{duration} are expressed as time
duration specifications; see
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@option{start}, @option{end}, @option{duration} are expressed as time
duration specifications, check the "Time duration" section in the
ffmpeg-utils manual.
Note that the first two sets of the start/end options and the @option{duration}
option look at the frame timestamp, while the _sample options simply count the
@@ -1600,9 +1587,9 @@ Set the number of samples per channel per each output frame, default
is 1024. Only used if plugin have zero inputs.
@item duration, d
Set the minimum duration of the sourced audio. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
Set the minimum duration of the sourced audio. See the function
@code{av_parse_time()} for the accepted format, also check the "Time duration"
section in the ffmpeg-utils manual.
Note that the resulting duration may be greater than the specified duration,
as the generated audio is always cut at the end of a complete frame.
If not specified, or the expressed duration is negative, the audio is
@@ -1710,11 +1697,11 @@ The default is 0.707q and gives a Butterworth response.
Mix channels with specific gain levels. The filter accepts the output
channel layout followed by a set of channels definitions.
This filter is also designed to efficiently remap the channels of an audio
This filter is also designed to remap efficiently the channels of an audio
stream.
The filter accepts parameters of the form:
"@var{l}|@var{outdef}|@var{outdef}|..."
"@var{l}:@var{outdef}:@var{outdef}:..."
@table @option
@item l
@@ -1745,13 +1732,13 @@ avoiding clipping noise.
For example, if you want to down-mix from stereo to mono, but with a bigger
factor for the left channel:
@example
pan=1c|c0=0.9*c0+0.1*c1
pan=1:c0=0.9*c0+0.1*c1
@end example
A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
7-channels surround:
@example
pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
@end example
Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
@@ -1774,25 +1761,25 @@ remapping.
For example, if you have a 5.1 source and want a stereo audio stream by
dropping the extra channels:
@example
pan="stereo| c0=FL | c1=FR"
pan="stereo: c0=FL : c1=FR"
@end example
Given the same source, you can also switch front left and front right channels
and keep the input channel layout:
@example
pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"
pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5"
@end example
If the input is a stereo audio stream, you can mute the front left channel (and
still keep the stereo channel layout) with:
@example
pan="stereo|c1=c1"
pan="stereo:c1=c1"
@end example
Still with a stereo audio stream input, you can copy the right channel in both
front left and right:
@example
pan="stereo| c0=FR | c1=FR"
pan="stereo: c0=FR : c1=FR"
@end example
@section replaygain
@@ -1844,75 +1831,6 @@ ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
@end example
@end itemize
@section silenceremove
Remove silence from the beginning, middle or end of the audio.
The filter accepts the following options:
@table @option
@item start_periods
This value is used to indicate if audio should be trimmed at beginning of
the audio. A value of zero indicates no silence should be trimmed from the
beginning. When specifying a non-zero value, it trims audio up until it
finds non-silence. Normally, when trimming silence from beginning of audio
the @var{start_periods} will be @code{1} but it can be increased to higher
values to trim all audio up to specific count of non-silence periods.
Default value is @code{0}.
@item start_duration
Specify the amount of time that non-silence must be detected before it stops
trimming audio. By increasing the duration, bursts of noises can be treated
as silence and trimmed off. Default value is @code{0}.
@item start_threshold
This indicates what sample value should be treated as silence. For digital
audio, a value of @code{0} may be fine but for audio recorded from analog,
you may wish to increase the value to account for background noise.
Can be specified in dB (in case "dB" is appended to the specified value)
or amplitude ratio. Default value is @code{0}.
@item stop_periods
Set the count for trimming silence from the end of audio.
To remove silence from the middle of a file, specify a @var{stop_periods}
that is negative. This value is then threated as a positive value and is
used to indicate the effect should restart processing as specified by
@var{start_periods}, making it suitable for removing periods of silence
in the middle of the audio.
Default value is @code{0}.
@item stop_duration
Specify a duration of silence that must exist before audio is not copied any
more. By specifying a higher duration, silence that is wanted can be left in
the audio.
Default value is @code{0}.
@item stop_threshold
This is the same as @option{start_threshold} but for trimming silence from
the end of audio.
Can be specified in dB (in case "dB" is appended to the specified value)
or amplitude ratio. Default value is @code{0}.
@item leave_silence
This indicate that @var{stop_duration} length of audio should be left intact
at the beginning of each period of silence.
For example, if you want to remove long pauses between words but do not want
to remove the pauses completely. Default value is @code{0}.
@end table
@subsection Examples
@itemize
@item
The following example shows how this filter can be used to start a recording
that does not contain the delay at the start which usually occurs between
pressing the record button and the start of the performance:
@example
silenceremove=1:5:0.02
@end example
@end itemize
@section treble
Boost or cut treble (upper) frequencies of the audio using a two-pole
@@ -2221,9 +2139,8 @@ Set the channel layout. The number of channels in the specified layout
must be equal to the number of specified expressions.
@item duration, d
Set the minimum duration of the sourced audio. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
Set the minimum duration of the sourced audio. See the function
@code{av_parse_time()} for the accepted format.
Note that the resulting duration may be greater than the specified
duration, as the generated audio is always cut at the end of a
complete frame.
@@ -2521,26 +2438,6 @@ Same as the @ref{subtitles} filter, except that it doesn't require libavcodec
and libavformat to work. On the other hand, it is limited to ASS (Advanced
Substation Alpha) subtitles files.
This filter accepts the following option in addition to the common options from
the @ref{subtitles} filter:
@table @option
@item shaping
Set the shaping engine
Available values are:
@table @samp
@item auto
The default libass shaping engine, which is the best available.
@item simple
Fast, font-agnostic shaper that can do only substitutions
@item complex
Slower shaper using OpenType for substitutions and positioning
@end table
The default is @code{auto}.
@end table
@section bbox
Compute the bounding box for the non-black pixels in the input frame
@@ -2858,42 +2755,6 @@ boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chrom
@end example
@end itemize
@section codecview
Visualize information exported by some codecs.
Some codecs can export information through frames using side-data or other
means. For example, some MPEG based codecs export motion vectors through the
@var{export_mvs} flag in the codec @option{flags2} option.
The filter accepts the following option:
@table @option
@item mv
Set motion vectors to visualize.
Available flags for @var{mv} are:
@table @samp
@item pf
forward predicted MVs of P-frames
@item bf
forward predicted MVs of B-frames
@item bb
backward predicted MVs of B-frames
@end table
@end table
@subsection Examples
@itemize
@item
Visualizes multi-directionals MVs from P and B-Frames using @command{ffplay}:
@example
ffplay -flags2 +export_mvs input.mpg -vf codecview=mv=pf+bf+bb
@end example
@end itemize
@section colorbalance
Modify intensity of primary colors (red, green and blue) of input frames.
@@ -3358,7 +3219,7 @@ curves=psfile='MyCurvesPresets/purple.asv':green='0.45/0.53'
Denoise frames using 2D DCT (frequency domain filtering).
This filter is not designed for real time.
This filter is not designed for real time and can be extremely slow.
The filter accepts the following options:
@@ -3374,14 +3235,14 @@ If you need a more advanced filtering, see @option{expr}.
Default is @code{0}.
@item overlap
Set number overlapping pixels for each block. Since the filter can be slow, you
may want to reduce this value, at the cost of a less effective filter and the
risk of various artefacts.
Set number overlapping pixels for each block. Each block is of size
@code{16x16}. Since the filter can be slow, you may want to reduce this value,
at the cost of a less effective filter and the risk of various artefacts.
If the overlapping value doesn't permit processing the whole input width or
If the overlapping value doesn't allow to process the whole input width or
height, a warning will be displayed and according borders won't be denoised.
Default value is @var{blocksize}-1, which is the best possible setting.
Default value is @code{15}.
@item expr, e
Set the coefficient factor expression.
@@ -3393,15 +3254,6 @@ If this is option is set, the @option{sigma} option will be ignored.
The absolute value of the coefficient can be accessed through the @var{c}
variable.
@item n
Set the @var{blocksize} using the number of bits. @code{1<<@var{n}} defines the
@var{blocksize}, which is the width and height of the processed blocks.
The default value is @var{3} (8x8) and can be raised to @var{4} for a
@var{blocksize} of 16x16. Note that changing this setting has huge consequences
on the speed processing. Also, a larger block size does not necessarily means a
better de-noising.
@end table
@subsection Examples
@@ -3416,11 +3268,6 @@ The same operation can be achieved using the expression system:
dctdnoiz=e='gte(c, 4.5*3)'
@end example
Violent denoise using a block size of @code{16x16}:
@example
dctdnoiz=15:n=4
@end example
@anchor{decimate}
@section decimate
@@ -3851,11 +3698,6 @@ the "Color" section in the ffmpeg-utils manual.
The default value of @var{fontcolor} is "black".
@item fontcolor_expr
String which is expanded the same way as @var{text} to obtain dynamic
@var{fontcolor} value. By default this option has empty value and is not
processed. When this option is set, it overrides @var{fontcolor} option.
@item font
The font family to be used for drawing text. By default Sans.
@@ -4069,15 +3911,6 @@ example the text size is not known when evaluating the expression, so
the constants @var{text_w} and @var{text_h} will have an undefined
value.
@item expr_int_format, eif
Evaluate the expression's value and output as formatted integer.
The first argument is the expression to be evaluated, just as for the @var{expr} function.
The second argument specifies the output format. Allowed values are 'x', 'X', 'd' and
'u'. They are treated exactly as in the printf function.
The third parameter is optional and sets the number of positions taken by the output.
It can be used to add padding with zeros from the left.
@item gmtime
The time at which the filter is running, expressed in UTC.
It can accept an argument: a strftime() format string.
@@ -4135,7 +3968,7 @@ within the parameter list.
@item
Show the text at the center of the video frame:
@example
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h-line_h)/2"
@end example
@item
@@ -4174,18 +4007,7 @@ drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'
@item
Print the date of a real-time encoding (see strftime(3)):
@example
drawtext='fontfile=FreeSans.ttf:text=%@{localtime\:%a %b %d %Y@}'
@end example
@item
Show text fading in and out (appearing/disappearing):
@example
#!/bin/sh
DS=1.0 # display start
DE=10.0 # display end
FID=1.5 # fade in duration
FOD=5 # fade out duration
ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%@{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 @}"
drawtext='fontfile=FreeSans.ttf:text=%@{localtime:%a %b %d %Y@}'
@end example
@end itemize
@@ -4447,10 +4269,6 @@ and VIVTC/VFM (VapourSynth project). The later is a light clone of TFM from
which @code{fieldmatch} is based on. While the semantic and usage are very
close, some behaviour and options names can differ.
The @ref{decimate} filter currently only works for constant frame rate input.
Do not use @code{fieldmatch} and @ref{decimate} if your input has mixed
telecined and progressive content with changing framerate.
The filter accepts the following options:
@table @option
@@ -5117,22 +4935,6 @@ Modify RGB components depending on pixel position:
@example
geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'
@end example
@item
Create a radial gradient that is the same size as the input (also see
the @ref{vignette} filter):
@example
geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray
@end example
@item
Create a linear gradient to use as a mask for another filter, then
compose with @ref{overlay}. In this example the video will gradually
become more blurry from the top to the bottom of the y-axis as defined
by the linear gradient:
@example
ffmpeg -i input.mp4 -filter_complex "geq=lum=255*(Y/H),format=gray[grad];[0:v]boxblur=4[blur];[blur][grad]alphamerge[alpha];[0:v][alpha]overlay" output.mp4
@end example
@end itemize
@section gradfun
@@ -5574,62 +5376,8 @@ value.
Detect video interlacing type.
This filter tries to detect if the input frames as interlaced, progressive,
top or bottom field first. It will also try and detect fields that are
repeated between adjacent frames (a sign of telecine).
Single frame detection considers only immediately adjacent frames when classifying each frame.
Multiple frame detection incorporates the classification history of previous frames.
The filter will log these metadata values:
@table @option
@item single.current_frame
Detected type of current frame using single-frame detection. One of:
``tff'' (top field first), ``bff'' (bottom field first),
``progressive'', or ``undetermined''
@item single.tff
Cumulative number of frames detected as top field first using single-frame detection.
@item multiple.tff
Cumulative number of frames detected as top field first using multiple-frame detection.
@item single.bff
Cumulative number of frames detected as bottom field first using single-frame detection.
@item multiple.current_frame
Detected type of current frame using multiple-frame detection. One of:
``tff'' (top field first), ``bff'' (bottom field first),
``progressive'', or ``undetermined''
@item multiple.bff
Cumulative number of frames detected as bottom field first using multiple-frame detection.
@item single.progressive
Cumulative number of frames detected as progressive using single-frame detection.
@item multiple.progressive
Cumulative number of frames detected as progressive using multiple-frame detection.
@item single.undetermined
Cumulative number of frames that could not be classified using single-frame detection.
@item multiple.undetermined
Cumulative number of frames that could not be classified using multiple-frame detection.
@item repeated.current_frame
Which field in the current frame is repeated from the last. One of ``neither'', ``top'', or ``bottom''.
@item repeated.neither
Cumulative number of frames with no repeated field.
@item repeated.top
Cumulative number of frames with the top field repeated from the previous frame's top field.
@item repeated.bottom
Cumulative number of frames with the bottom field repeated from the previous frame's bottom field.
@end table
This filter tries to detect if the input is interlaced or progressive,
top or bottom field first.
The filter accepts the following options:
@@ -5638,13 +5386,6 @@ The filter accepts the following options:
Set interlacing threshold.
@item prog_thres
Set progressive threshold.
@item repeat_thres
Threshold for repeated field detection.
@item half_life
Number of frames after which a given frame's contribution to the
statistics is halved (i.e., it contributes only 0.5 to it's
classification). The default of 0 means that all frames seen are given
full weight of 1.0 forever.
@end table
@section il
@@ -5688,7 +5429,8 @@ Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is @code
Simple interlacing filter from progressive contents. This interleaves upper (or
lower) lines from odd frames with lower (or upper) lines from even frames,
halving the frame rate and preserving image height.
halving the frame rate and preserving image height. A vertical lowpass filter
is always applied in order to avoid twitter effects and reduce moiré patterns.
@example
Original Original New Frame
@@ -5708,10 +5450,6 @@ It accepts the following optional parameters:
@item scan
This determines whether the interlaced frame is taken from the even
(tff - default) or odd (bff) lines of the progressive frame.
@item lowpass
Enable (default) or disable the vertical lowpass filter to avoid twitter
interlacing and reduce moire patterns.
@end table
@section kerndeint
@@ -5766,51 +5504,6 @@ kerndeint=map=1
@end example
@end itemize
@section lenscorrection
Correct radial lens distortion
This filter can be used to correct for radial distortion as can result from the use
of wide angle lenses, and thereby re-rectify the image. To find the right parameters
one can use tools available for example as part of opencv or simply trial-and-error.
To use opencv use the calibration sample (under samples/cpp) from the opencv sources
and extract the k1 and k2 coefficients from the resulting matrix.
Note that effectively the same filter is available in the open-source tools Krita and
Digikam from the KDE project.
In contrast to the @ref{vignette} filter, which can also be used to compensate lens errors,
this filter corrects the distortion of the image, whereas @ref{vignette} corrects the
brightness distribution, so you may want to use both filters together in certain
cases, though you will have to take care of ordering, i.e. whether vignetting should
be applied before or after lens correction.
@subsection Options
The filter accepts the following options:
@table @option
@item cx
Relative x-coordinate of the focal point of the image, and thereby the center of the
distrortion. This value has a range [0,1] and is expressed as fractions of the image
width.
@item cy
Relative y-coordinate of the focal point of the image, and thereby the center of the
distrortion. This value has a range [0,1] and is expressed as fractions of the image
height.
@item k1
Coefficient of the quadratic correction term. 0.5 means no correction.
@item k2
Coefficient of the double quadratic correction term. 0.5 means no correction.
@end table
The formula that generates the correction is:
@var{r_src} = @var{r_tgt} * (1 + @var{k1} * (@var{r_tgt} / @var{r_0})^2 + @var{k2} * (@var{r_tgt} / @var{r_0})^4)
where @var{r_0} is halve of the image diagonal and @var{r_src} and @var{r_tgt} are the
distances from the focal point in the source and target images, respectively.
@anchor{lut3d}
@section lut3d
@@ -6291,7 +5984,7 @@ values are assumed.
Refer to the official libopencv documentation for more precise
information:
@url{http://docs.opencv.org/master/modules/imgproc/doc/filtering.html}
@url{http://opencv.willowgarage.com/documentation/c/image_filtering.html}
Several libopencv filters are supported; see the following subsections.
@@ -6783,9 +6476,6 @@ A description of the accepted parameters follows.
@item y3
Set coordinates expression for top left, top right, bottom left and bottom right corners.
Default values are @code{0:0:W:0:0:H:W:H} with which perspective will remain unchanged.
If the @code{sense} option is set to @code{source}, then the specified points will be sent
to the corners of the destination. If the @code{sense} option is set to @code{destination},
then the corners of the source will be sent to the specified coordinates.
The expressions can use the following variables:
@@ -6805,24 +6495,6 @@ It accepts the following values:
@end table
Default value is @samp{linear}.
@item sense
Set interpretation of coordinate options.
It accepts the following values:
@table @samp
@item 0, source
Send point in the source specified by the given coordinates to
the corners of the destination.
@item 1, destination
Send the corners of the source to the point in the destination specified
by the given coordinates.
Default value is @samp{source}.
@end table
@end table
@section phase
@@ -7841,7 +7513,7 @@ The input video is not modified.
The shown line contains a sequence of key/value pairs of the form
@var{key}:@var{value}.
The following values are shown in the output:
It accepts the following parameters:
@table @option
@item n
@@ -8701,10 +8373,9 @@ The number of the first frame that should be passed to the output.
The number of the first frame that should be dropped.
@end table
@option{start}, @option{end}, and @option{duration} are expressed as time
duration specifications; see
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
@option{start}, @option{end}, @option{duration} are expressed as time
duration specifications, check the "Time duration" section in the
ffmpeg-utils manual.
Note that the first two sets of the start/end options and the @option{duration}
option look at the frame timestamp, while the _frame variants simply count the
@@ -8737,7 +8408,6 @@ ffmpeg -i INPUT -vf trim=duration=1
@end itemize
@anchor{unsharp}
@section unsharp
Sharpen or blur the input video.
@@ -8899,7 +8569,7 @@ Read a file with transform information for each frame and
apply/compensate them. Together with the @ref{vidstabdetect}
filter this can be used to deshake videos. See also
@url{http://public.hronopik.de/vid.stab}. It is important to also use
the @ref{unsharp} filter, see below.
the unsharp filter, see below.
To enable compilation of this filter you need to configure FFmpeg with
@code{--enable-libvidstab}.
@@ -8909,7 +8579,7 @@ To enable compilation of this filter you need to configure FFmpeg with
@table @option
@item input
Set path to the file used to read the transforms. Default value is
@file{transforms.trf}.
@file{transforms.trf}).
@item smoothing
Set the number of frames (value*2 + 1) used for lowpass filtering the
@@ -8917,9 +8587,9 @@ camera movements. Default value is 10.
For example a number of 10 means that 21 frames are used (10 in the
past and 10 in the future) to smoothen the motion in the video. A
larger value leads to a smoother video, but limits the acceleration of
the camera (pan/tilt movements). 0 is a special case where a static
camera is simulated.
larger values leads to a smoother video, but limits the acceleration
of the camera (pan/tilt movements). 0 is a special case where a
static camera is simulated.
@item optalgo
Set the camera path optimization algorithm.
@@ -8956,7 +8626,7 @@ fill the border black
Invert transforms if set to 1. Default value is 0.
@item relative
Consider transforms as relative to previous frame if set to 1,
Consider transforms as relative to previsou frame if set to 1,
absolute if set to 0. Default value is 0.
@item zoom
@@ -9022,7 +8692,7 @@ Use @command{ffmpeg} for a typical stabilization with default values:
ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg
@end example
Note the use of the @ref{unsharp} filter which is always recommended.
Note the use of the unsharp filter which is always recommended.
@item
Zoom in a bit more and load transform data from a given file:
@@ -9046,7 +8716,6 @@ For example, to vertically flip a video with @command{ffmpeg}:
ffmpeg -i in.avi -vf "vflip" out.avi
@end example
@anchor{vignette}
@section vignette
Make or reverse a natural vignetting effect.
@@ -9196,20 +8865,6 @@ Only deinterlace frames marked as interlaced.
Default value is @samp{all}.
@end table
@section xbr
Apply the xBR high-quality magnification filter which is designed for pixel
art. It follows a set of edge-detection rules, see
@url{http://www.libretro.com/forums/viewtopic.php?f=6&t=134}.
It accepts the following option:
@table @option
@item n
Set the scaling dimension: @code{2} for @code{2xBR}, @code{3} for
@code{3xBR} and @code{4} for @code{4xBR}.
Default is @code{3}.
@end table
@anchor{yadif}
@section yadif
@@ -9630,9 +9285,12 @@ number or a valid video frame rate abbreviation. The default value is
"25".
@item duration, d
Set the duration of the sourced video. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
Set the video duration of the sourced video. The accepted syntax is:
@example
[-]HH:MM:SS[.m...]
[-]S+[.m...]
@end example
See also the function @code{av_parse_time()}.
If not specified, or the expressed duration is negative, the video is
supposed to be generated forever.
@@ -9660,7 +9318,7 @@ Default value is "all", which will cycle through the list of all tests.
Some examples:
@example
mptestsrc=t=dc_luma
testsrc=t=dc_luma
@end example
will generate a "dc_luma" test pattern.
@@ -9894,9 +9552,12 @@ number or a valid video frame rate abbreviation. The default value is
Set the sample aspect ratio of the sourced video.
@item duration, d
Set the duration of the sourced video. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
Set the video duration of the sourced video. The accepted syntax is:
@example
[-]HH[:MM[:SS[.m...]]]
[-]S+[.m...]
@end example
Also see the the @code{av_parse_time()} function.
If not specified, or the expressed duration is negative, the video is
supposed to be generated forever.
@@ -10812,34 +10473,8 @@ The filter accepts the following options:
@table @option
@item volume
Specify transform volume (multiplier) expression. The expression can contain
variables:
@table @option
@item frequency, freq, f
the frequency where transform is evaluated
@item timeclamp, tc
value of timeclamp option
@end table
and functions:
@table @option
@item a_weighting(f)
A-weighting of equal loudness
@item b_weighting(f)
B-weighting of equal loudness
@item c_weighting(f)
C-weighting of equal loudness
@end table
Default value is @code{16}.
@item tlength
Specify transform length expression. The expression can contain variables:
@table @option
@item frequency, freq, f
the frequency where transform is evaluated
@item timeclamp, tc
value of timeclamp option
@end table
Default value is @code{384/f*tc/(384/f+tc)}.
Specify the transform volume (multiplier). Acceptable value is [1.0, 100.0].
Default value is @code{16.0}.
@item timeclamp
Specify the transform timeclamp. At low frequency, there is trade-off between
@@ -10861,26 +10496,6 @@ Default value is @code{3.0}.
@item fontfile
Specify font file for use with freetype. If not specified, use embedded font.
@item fontcolor
Specify font color expression. This is arithmetic expression that should return
integer value 0xRRGGBB. The expression can contain variables:
@table @option
@item frequency, freq, f
the frequency where transform is evaluated
@item timeclamp, tc
value of timeclamp option
@end table
and functions:
@table @option
@item midi(f)
midi number of frequency f, some midi numbers: E0(16), C1(24), C2(36), A4(69)
@item r(x), g(x), b(x)
red, green, and blue value of intensity x
@end table
Default value is @code{st(0, (midi(f)-59.5)/12);
st(1, if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)), 0));
r(1-ld(1)) + b(ld(1))}
@item fullhd
If set to 1 (the default), the video size is 1920x1080 (full HD),
if set to 0, the video size is 960x540. Use this option to make CPU usage lower.
@@ -10930,24 +10545,6 @@ ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*
asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'
@end example
@item
B-weighting of equal loudness
@example
volume=16*b_weighting(f)
@end example
@item
Lower Q factor
@example
tlength=100/f*tc/(100/f+tc)
@end example
@item
Custom fontcolor, C-note is colored green, others are colored blue
@example
fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))'
@end example
@end itemize
@section showspectrum
@@ -10964,19 +10561,8 @@ the "Video size" section in the ffmpeg-utils manual. Default value is
@code{640x512}.
@item slide
Specify how the spectrum should slide along the window.
It accepts the following values:
@table @samp
@item replace
the samples start again on the left when they reach the right
@item scroll
the samples scroll from right to left
@item fullframe
frames are only produced when the samples reach the right
@end table
Default value is @code{replace}.
Specify if the spectrum should slide along the window. Default value is
@code{0}.
@item mode
Specify display mode.
@@ -11087,12 +10673,6 @@ Draw a point for each sample.
@item line
Draw a vertical line for each sample.
@item p2p
Draw a point for each sample and a line between them.
@item cline
Draw a centered vertical line for each sample.
@end table
Default value is @code{point}.
@@ -11107,9 +10687,6 @@ is not explicitly specified.
Set the (approximate) output frame rate. This is done by setting the
option @var{n}. Default value is "25".
@item split_channels
Set if channels should be drawn separately or overlap. Default value is 0.
@end table
@subsection Examples

View File

@@ -23,7 +23,7 @@ Reduce buffering.
@item probesize @var{integer} (@emph{input})
Set probing size in bytes, i.e. the size of the data to analyze to get
stream information. A higher value will enable detecting more
stream information. A higher value will allow to detect more
information in case it is dispersed into the stream, but will increase
latency. Must be an integer not lesser than 32. It is 5000000 by default.
@@ -55,10 +55,6 @@ Do not merge side data.
Enable RTP MP4A-LATM payload.
@item nobuffer
Reduce the latency introduced by optional buffering
@item bitexact
Only write platform-, build- and time-independent data.
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@end table
@item seek2any @var{integer} (@emph{input})
@@ -67,7 +63,7 @@ Default is 0.
@item analyzeduration @var{integer} (@emph{input})
Specify how many microseconds are analyzed to probe the input. A
higher value will enable detecting more accurate information, but will
higher value will allow to detect more accurate information, but will
increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
@item cryptokey @var{hexadecimal string} (@emph{input})
@@ -172,18 +168,6 @@ The offset is added by the muxer to the output timestamps.
Specifying a positive offset means that the corresponding streams are
delayed bt the time duration specified in @var{offset}. Default value
is @code{0} (meaning that no offset is applied).
@item format_whitelist @var{list} (@emph{input})
"," separated List of allowed demuxers. By default all are allowed.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@end table
@c man end FORMAT OPTIONS

View File

@@ -130,7 +130,7 @@ Go to @url{http://x265.org/developers.html} and follow the instructions
for installing the library. Then pass @code{--enable-libx265} to configure
to enable it.
@float NOTE
@float note
x265 is under the GNU Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
details), you must upgrade FFmpeg's license to GPL in order to use it.
@@ -454,7 +454,6 @@ library:
@item Sony Wave64 (W64) @tab X @tab X
@item SoX native format @tab X @tab X
@item SUN AU format @tab X @tab X
@item SUP raw PGS subtitles @tab @tab X
@item Text files @tab @tab X
@item THP @tab @tab X
@tab Used on the Nintendo GameCube.
@@ -936,8 +935,8 @@ following image formats are supported:
@item Musepack SV8 @tab @tab X
@item Nellymoser Asao @tab X @tab X
@item On2 AVC (Audio for Video Codec) @tab @tab X
@item Opus @tab E @tab X
@tab encoding supported through external library libopus
@item Opus @tab E @tab E
@tab supported through external library libopus
@item PCM A-law @tab X @tab X
@item PCM mu-law @tab X @tab X
@item PCM signed 8-bit planar @tab X @tab X
@@ -1031,7 +1030,6 @@ performance on systems without hardware floating point support).
@item PJS (Phoenix) @tab @tab X @tab @tab X
@item RealText @tab @tab X @tab @tab X
@item SAMI @tab @tab X @tab @tab X
@item Spruce format (STL) @tab @tab X @tab @tab X
@item SSA/ASS @tab X @tab X @tab X @tab X
@item SubRip (SRT) @tab X @tab X @tab X @tab X
@item SubViewer v1 @tab @tab X @tab @tab X
@@ -1057,7 +1055,6 @@ performance on systems without hardware floating point support).
@item HLS @tab X
@item HTTP @tab X
@item HTTPS @tab X
@item Icecast @tab X
@item MMSH @tab X
@item MMST @tab X
@item pipe @tab X

View File

@@ -1,7 +1,7 @@
@chapter Input Devices
@c man begin INPUT DEVICES
Input devices are configured elements in FFmpeg which enable accessing
Input devices are configured elements in FFmpeg which allow to access
the data coming from a multimedia device attached to your system.
When you configure your FFmpeg build, all the supported input devices
@@ -58,94 +58,34 @@ AVFoundation input device.
AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
The older QTKit framework has been marked deprecated since OSX version 10.7.
The input filename has to be given in the following syntax:
@example
-i "[[VIDEO]:[AUDIO]]"
@end example
The first entry selects the video input while the latter selects the audio input.
The stream has to be specified by the device name or the device index as shown by the device list.
Alternatively, the video and/or audio input device can be chosen by index using the
@option{
-video_device_index <INDEX>
}
and/or
@option{
-audio_device_index <INDEX>
}
, overriding any
device name or index given in the input filename.
All available devices can be enumerated by using @option{-list_devices true}, listing
all device names and corresponding indices.
There are two device name aliases:
@table @code
@item default
Select the AVFoundation default device of the corresponding type.
@item none
Do not record the corresponding media type.
This is equivalent to specifying an empty device name or index.
@end table
@subsection Options
AVFoundation supports the following options:
@table @option
@item -list_devices <TRUE|FALSE>
If set to true, a list of all available input devices is given showing all
device names and indices.
@item -video_device_index <INDEX>
Specify the video device by its index. Overrides anything given in the input filename.
@item -audio_device_index <INDEX>
Specify the audio device by its index. Overrides anything given in the input filename.
@item -pixel_format <FORMAT>
Request the video device to use a specific pixel format.
If the specified format is not supported, a list of available formats is given
und the first one in this list is used instead. Available pixel formats are:
@code{monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
The filename passed as input is parsed to contain either a device name or index.
The device index can also be given by using -video_device_index.
A given device index will override any given device name.
If the desired device consists of numbers only, use -video_device_index to identify it.
The default device will be chosen if an empty string or the device name "default" is given.
The available devices can be enumerated by using -list_devices.
The pixel format can be set using -pixel_format.
Available formats:
monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
yuv420p, nv12, yuyv422, gray}
yuv420p, nv12, yuyv422, gray
@end table
@subsection Examples
@itemize
@item
Print the list of AVFoundation supported devices and exit:
@example
$ ffmpeg -f avfoundation -list_devices true -i ""
ffmpeg -f avfoundation -i "0" out.mpg
@end example
@item
Record video from video device 0 and audio from audio device 0 into out.avi:
@example
$ ffmpeg -f avfoundation -i "0:0" out.avi
ffmpeg -f avfoundation -video_device_index 0 -i "" out.mpg
@end example
@item
Record video from video device 2 and audio from audio device 1 into out.avi:
@example
$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
ffmpeg -f avfoundation -pixel_format bgr0 -i "default" out.mpg
@end example
@item
Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi:
@example
$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
ffmpeg -f avfoundation -list_devices true -i ""
@end example
@end itemize
@section bktr
BSD video input device.
@@ -548,8 +488,7 @@ ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
Audio-CD input device based on cdio.
To enable this input device during configuration you need libcdio
installed on your system. Requires the configure option
@code{--enable-libcdio}.
installed on your system.
This device allows playing and grabbing from an Audio-CD.
@@ -563,8 +502,6 @@ ffmpeg -f libcdio -i /dev/sr0 cd.wav
IIDC1394 input device, based on libdc1394 and libraw1394.
Requires the configure option @code{--enable-libdc1394}.
@section openal
The OpenAL input device provides audio capture on all systems with a
@@ -893,9 +830,6 @@ other filename will be interpreted as device number 0.
X11 video input device.
Depends on X11, Xext, and Xfixes. Requires the configure option
@code{--enable-x11grab}.
This device allows one to capture a region of an X11 display.
The filename passed as input has the syntax:
@@ -976,64 +910,6 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
@item video_size
Set the video frame size. Default value is @code{vga}.
@item use_shm
Use the MIT-SHM extension for shared memory. Default value is @code{1}.
It may be necessary to disable it for remote displays.
@end table
@section decklink
The decklink input device provides capture capabilities for Blackmagic
DeckLink devices.
To enable this input device, you need the Blackmagic DeckLink SDK and you
need to configure with the appropriate @code{--extra-cflags}
and @code{--extra-ldflags}.
On Windows, you need to run the IDL files through @command{widl}.
DeckLink is very picky about the formats it supports. Pixel format is always
uyvy422, framerate and video size must be determined for your device with
@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
of channels currently is limited to 2 (stereo).
@subsection Options
@table @option
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
Defaults to @option{false}.
@end table
@subsection Examples
@itemize
@item
List input devices:
@example
ffmpeg -f decklink -list_devices 1 -i dummy
@end example
@item
List supported formats:
@example
ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
@end example
@item
Capture video clip at 1080i50 (format 11):
@example
ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
@end example
@end itemize
@c man end INPUT DEVICES

View File

@@ -22,7 +22,7 @@ a mail for every change to every issue.
(the above does all work already after light testing)
The subscription URL for the ffmpeg-trac list is:
http(s)://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
The URL of the webinterface of the tracker is:
http(s)://trac.ffmpeg.org

View File

@@ -194,19 +194,15 @@ can not be smaller than one centi second.
Apple HTTP Live Streaming muxer that segments MPEG-TS according to
the HTTP Live Streaming (HLS) specification.
It creates a playlist file, and one or more segment files. The output filename
specifies the playlist filename.
By default, the muxer creates a file for each segment produced. These files
have the same name as the playlist, followed by a sequential number and a
.ts extension.
It creates a playlist file and numbered segment files. The output
filename specifies the playlist filename; the segment filenames
receive the same basename as the playlist, a sequential number and
a .ts extension.
For example, to convert an input file with @command{ffmpeg}:
@example
ffmpeg -i in.nut out.m3u8
@end example
This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{out0.ts}, @file{out1.ts}, @file{out2.ts}, etc.
See also the @ref{segment} muxer, which provides a more generic and
flexible implementation of a segmenter, and can be used to perform HLS
@@ -224,11 +220,6 @@ Set the segment length in seconds. Default value is 2.
Set the maximum number of playlist entries. If set to 0 the list file
will contain all the segments. Default value is 5.
@item hls_ts_options @var{options_list}
Set output format options using a :-separated list of key=value
parameters. Values containing @code{:} special characters must be
escaped.
@item hls_wrap @var{wrap}
Set the number after which the segment filename number (the number
specified in each segment file) wraps. If set to 0 the number will be
@@ -242,9 +233,6 @@ to @var{wrap}.
Start the playlist sequence number from @var{number}. Default value is
0.
@item hls_allow_cache @var{allowcache}
Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments.
@item hls_base_url @var{baseurl}
Append @var{baseurl} to every entry in the playlist.
Useful to generate playlists with absolute paths.
@@ -253,17 +241,6 @@ Note that the playlist sequence number must be unique for each segment
and it is not to be confused with the segment filename sequence number
which can be cyclic, for example if the @option{wrap} option is
specified.
@item hls_flags single_file
If this flag is set, the muxer will store all segments in a single MPEG-TS
file, and will use byte ranges in the playlist. HLS playlists generated with
this way will have the version number 4.
For example:
@example
ffmpeg -i in.nut -hls_flags single_file out.m3u8
@end example
Will produce the playlist, @file{out.m3u8}, and a single segment file,
@file{out.ts}.
@end table
@anchor{ico}
@@ -559,6 +536,7 @@ a short portion of the file. With this option set, there is no initial
mdat atom, and the moov atom only describes the tracks but has
a zero duration.
Files written with this option set do not work in QuickTime.
This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -movflags separate_moof
Write a separate moof (movie fragment) atom for each track. Normally,
@@ -573,22 +551,6 @@ This operation can take a while, and will not work in various situations such
as fragmented output, thus it is not enabled by default.
@item -movflags rtphint
Add RTP hinting tracks to the output file.
@item -movflags disable_chpl
Disable Nero chapter markers (chpl atom). Normally, both Nero chapters
and a QuickTime chapter track are written to the file. With this option
set, only the QuickTime chapter track will be written. Nero chapters can
cause failures when the file is reprocessed with certain tagging programs, like
mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well.
@item -movflags omit_tfhd_offset
Do not write any absolute base_data_offset in tfhd atoms. This avoids
tying fragments to absolute byte positions in the file/streams.
@item -movflags default_base_moof
Similarly to the omit_tfhd_offset, this flag avoids writing the
absolute base_data_offset field in tfhd atoms, but does so by using
the new default-base-is-moof flag instead. This flag is new from
14496-12:2012. This may make the fragments easier to parse in certain
circumstances (avoiding basing track fragment location calculations
on the implicit end of the previous track fragment).
@end table
@subsection Example
@@ -601,38 +563,29 @@ ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe
@section mp3
The MP3 muxer writes a raw MP3 stream with the following optional features:
@itemize @bullet
@item
An ID3v2 metadata header at the beginning (enabled by default). Versions 2.3 and
2.4 are supported, the @code{id3v2_version} private option controls which one is
used (3 or 4). Setting @code{id3v2_version} to 0 disables the ID3v2 header
completely.
The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
@code{id3v2_version} option controls which one is used. Setting
@code{id3v2_version} to 0 will disable the ID3v2 header completely. The legacy
ID3v1 tag is not written by default, but may be enabled with the
@code{write_id3v1} option.
The muxer supports writing attached pictures (APIC frames) to the ID3v2 header.
The pictures are supplied to the muxer in form of a video stream with a single
packet. There can be any number of those streams, each will correspond to a
single APIC frame. The stream metadata tags @var{title} and @var{comment} map
to APIC @var{description} and @var{picture type} respectively. See
The muxer may also write a Xing frame at the beginning, which contains the
number of frames in the file. It is useful for computing duration of VBR files.
The Xing frame is written if the output stream is seekable and if the
@code{write_xing} option is set to 1 (the default).
The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
are supplied to the muxer in form of a video stream with a single packet. There
can be any number of those streams, each will correspond to a single APIC frame.
The stream metadata tags @var{title} and @var{comment} map to APIC
@var{description} and @var{picture type} respectively. See
@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
Note that the APIC frames must be written at the beginning, so the muxer will
buffer the audio frames until it gets all the pictures. It is therefore advised
to provide the pictures as soon as possible to avoid excessive buffering.
@item
A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by
default, but will be written only if the output is seekable. The
@code{write_xing} private option can be used to disable it. The frame contains
various information that may be useful to the decoder, like the audio duration
or encoder delay.
@item
A legacy ID3v1 tag at the end of the file (disabled by default). It may be
enabled with the @code{write_id3v1} private option, but as its capabilities are
very limited, its usage is not recommended.
@end itemize
Examples:
Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
@@ -845,11 +798,6 @@ reference stream. The default value is @code{auto}.
Override the inner container format, by default it is guessed by the filename
extension.
@item segment_format_options @var{options_list}
Set output format options using a :-separated list of key=value
parameters. Values containing the @code{:} special character must be
escaped.
@item segment_list @var{name}
Generate also a listfile named @var{name}. If not specified no
listfile is generated.
@@ -874,7 +822,7 @@ Select the listing format.
@end table
@item segment_list_size @var{size}
Update the list file so that it contains at most @var{size}
Update the list file so that it contains at most the last @var{size}
segments. If 0 the list file will contain all the segments. Default
value is 0.
@@ -997,7 +945,7 @@ argument must be a time duration specification, and defaults to 0.
@itemize
@item
Remux the content of file @file{in.mkv} to a list of segments
To remux the content of file @file{in.mkv} to a list of segments
@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
generated segments to @file{out.list}:
@example
@@ -1005,20 +953,14 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nu
@end example
@item
Segment input and set output format options for the output segments:
@example
ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
@end example
@item
Segment the input file according to the split points specified by the
@var{segment_times} option:
As the example above, but segment the input file according to the split
points specified by the @var{segment_times} option:
@example
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
@end example
@item
Use the @command{ffmpeg} @option{force_key_frames}
As the example above, but use the @command{ffmpeg} @option{force_key_frames}
option to force key frames in the input at the specified location, together
with the segment option @option{segment_time_delta} to account for
possible roundings operated when setting key frame times.
@@ -1037,7 +979,7 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_fr
@end example
@item
Convert the @file{in.mkv} to TS segments using the @code{libx264}
To convert the @file{in.mkv} to TS segments using the @code{libx264}
and @code{libfaac} encoders:
@example
ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts

View File

@@ -35,9 +35,6 @@ to your project LDFLAGS:
-Wl,-Bsymbolic
@end example
If your target platform requires position independent binaries, you should
pass the correct linking flag (e.g. @code{-pie}) to @code{--extra-ldexeflags}.
@section BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
@@ -135,6 +132,8 @@ You will need the following prerequisites:
(if using MSVC 2012 or earlier)
@item @uref{http://www.mingw.org/, MSYS}
@item @uref{http://yasm.tortall.net/, YASM}
@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
you want to run @uref{fate.html, FATE}.
@end itemize
To set up a proper environment in MSYS, you need to run @code{msys.bat} from
@@ -276,12 +275,12 @@ llrint() in its C library.
Install your Cygwin with all the "Base" packages, plus the
following "Devel" ones:
@example
binutils, gcc4-core, make, git, mingw-runtime, texinfo
binutils, gcc4-core, make, git, mingw-runtime, texi2html
@end example
In order to run FATE you will also need the following "Utils" packages:
@example
diffutils
bc, diffutils
@end example
If you want to build FFmpeg with additional libraries, download Cygwin

View File

@@ -26,10 +26,6 @@
#include <string.h>
#include <float.h>
// print_options is build for the host, os_support.h isn't needed and is setup
// for the target. without this build breaks on mingw
#define AVFORMAT_OS_SUPPORT_H
#include "libavformat/avformat.h"
#include "libavformat/options_table.h"
#include "libavcodec/avcodec.h"

View File

@@ -255,7 +255,7 @@ Export the MIME type.
If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
supports this, the metadata has to be retrieved by the application by reading
the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
The default is 1.
The default is 0.
@item icy_metadata_headers
If the server supports ICY metadata, this contains the ICY-specific HTTP reply
@@ -293,50 +293,6 @@ The required syntax to play a stream specifying a cookie is:
ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
@end example
@section Icecast
Icecast protocol (stream to Icecast servers)
This protocol accepts the following options:
@table @option
@item ice_genre
Set the stream genre.
@item ice_name
Set the stream name.
@item ice_description
Set the stream description.
@item ice_url
Set the stream website URL.
@item ice_public
Set if the stream should be public.
The default is 0 (not public).
@item user_agent
Override the User-Agent header. If not specified a string of the form
"Lavf/<version>" will be used.
@item password
Set the Icecast mountpoint password.
@item content_type
Set the stream content type. This must be set if it is different from
audio/mpeg.
@item legacy_icecast
This enables support for Icecast versions < 2.4.0, that do not support the
HTTP PUT method but the SOURCE method.
@end table
@example
icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
@end example
@section mmst
MMS (Microsoft Media Server) protocol over TCP.
@@ -583,7 +539,7 @@ firewalls.
@section libsmbclient
libsmbclient permits one to manipulate CIFS/SMB network resources.
libsmbclient permits to manipulate CIFS/SMB network resources.
Following syntax is required.
@@ -750,7 +706,7 @@ port will be used for the local RTP and RTCP ports.
@item
If @option{localrtcpport} (the local RTCP port) is not set it will be
set to the local RTP port value plus 1.
set to the the local RTP port value plus 1.
@end enumerate
@section rtsp
@@ -1081,8 +1037,8 @@ Set raise error timeout, expressed in microseconds.
This option is only relevant in read mode: if no data arrived in more
than this time interval, raise error.
@item listen_timeout=@var{milliseconds}
Set listen timeout, expressed in milliseconds.
@item listen_timeout=@var{microseconds}
Set listen timeout, expressed in microseconds.
@end table
The following example shows how to setup a listening TCP connection

View File

@@ -618,6 +618,7 @@ flip wavelet?
try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients
try the MV length as context for coding the residual coefficients
use extradata for stuff which is in the keyframes now?
the MV median predictor is patented IIRC
implement per picture halfpel interpolation
try different range coder state transition tables for different contexts

23
doc/style.min.css vendored

File diff suppressed because one or more lines are too long

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@@ -1,35 +1,26 @@
# Init file for texi2html.
# This is deprecated, and the makeinfo/texi2any version is doc/t2h.pm
# no horiz rules between sections
$end_section = \&FFmpeg_end_section;
sub FFmpeg_end_section($$)
{
}
my $TEMPLATE_HEADER1 = $ENV{"FFMPEG_HEADER1"} || <<EOT;
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="utf-8" />
<meta http-equiv="X-UA-Compatible" content="IE=edge" />
<title>FFmpeg documentation</title>
<link rel="stylesheet" href="bootstrap.min.css" />
<link rel="stylesheet" href="style.min.css" />
$EXTRA_HEAD =
'<link rel="icon" href="favicon.png" type="image/png" />
';
$CSS_LINES = $ENV{"FFMPEG_CSS"} || <<EOT;
<link rel="stylesheet" type="text/css" href="default.css" />
EOT
my $TEMPLATE_HEADER2 = $ENV{"FFMPEG_HEADER2"} || <<EOT;
</head>
<body>
<div style="width: 95%; margin: auto">
my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
<link rel="icon" href="favicon.png" type="image/png" />
</head>
<body>
<div id="container">
<div id="body">
EOT
my $TEMPLATE_FOOTER = $ENV{"FFMPEG_FOOTER"} || <<EOT;
</div>
</body>
</html>
EOT
$PRE_BODY_CLOSE = '</div></div>';
$SMALL_RULE = '';
$BODYTEXT = '';
@@ -91,25 +82,21 @@ sub FFmpeg_print_page_head($$)
$longtitle = "FFmpeg documentation : " . $longtitle;
print $fh <<EOT;
$TEMPLATE_HEADER1
$description
<meta name="keywords" content="$longtitle">
<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
<!DOCTYPE html>
<html>
$Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} -->
<!--
$Texi2HTML::THISDOC{program_authors}
-->
$encoding
$TEMPLATE_HEADER2
EOT
}
<head>
<title>$longtitle</title>
$print_page_foot = \&FFmpeg_print_page_foot;
sub FFmpeg_print_page_foot($$)
{
my $fh = shift;
print $fh <<EOT;
$TEMPLATE_FOOTER
$description
<meta name="keywords" content="$longtitle">
<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
$encoding
$CSS_LINES
$TEMPLATE_HEADER
EOT
}

View File

@@ -1,221 +0,0 @@
# makeinfo HTML output init file
#
# Copyright (c) 2011, 2012 Free Software Foundation, Inc.
# Copyright (c) 2014 Andreas Cadhalpun
# Copyright (c) 2014 Tiancheng "Timothy" Gu
#
# This file is part of FFmpeg.
#
# FFmpeg is free software; you can redistribute it and/or modify
# it under the terms of the GNU General Public License as published by
# the Free Software Foundation; either version 3 of the License, or
# (at your option) any later version.
#
# 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
# General Public License for more details.
#
# You should have received a copy of the GNU 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
# no navigation elements
set_from_init_file('HEADERS', 0);
# TOC and Chapter headings link
set_from_init_file('TOC_LINKS', 1);
# print the TOC where @contents is used
set_from_init_file('INLINE_CONTENTS', 1);
# make chapters <h2>
set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
# Do not add <hr>
set_from_init_file('DEFAULT_RULE', '');
set_from_init_file('BIG_RULE', '');
# Customized file beginning
sub ffmpeg_begin_file($$$)
{
my $self = shift;
my $filename = shift;
my $element = shift;
my $command;
if ($element and $self->get_conf('SPLIT')) {
$command = $self->element_command($element);
}
my ($title, $description, $encoding, $date, $css_lines,
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_informations($command);
my $links = $self->_get_links ($filename, $element);
my $head1 = $ENV{"FFMPEG_HEADER1"} || <<EOT;
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by $program_and_version, $program_homepage -->
<head>
<meta charset="utf-8">
<title>
EOT
my $head_title = <<EOT;
$title
EOT
my $head2 = $ENV{"FFMPEG_HEADER2"} || <<EOT;
</title>
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
EOT
my $head3 = $ENV{"FFMPEG_HEADER3"} || <<EOT;
</h1>
EOT
return $head1 . $head_title . $head2 . $head_title . $head3;
}
texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
# Customized file ending
sub ffmpeg_end_file($)
{
my $self = shift;
my $program_string = &{$self->{'format_program_string'}}($self);
my $program_text = <<EOT;
<p style="font-size: small;">
$program_string
</p>
EOT
my $footer = $ENV{FFMPEG_FOOTER} || <<EOT;
</div>
</body>
</html>
EOT
return $program_text . $footer;
}
texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
# Dummy title command
# Ignore title. Title is handled through ffmpeg_begin_file().
set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
sub ffmpeg_title($$$$)
{
return '';
}
texinfo_register_command_formatting('titlefont',
\&ffmpeg_title);
# Customized float command. Part of code borrowed from GNU Texinfo.
sub ffmpeg_float($$$$$)
{
my $self = shift;
my $cmdname = shift;
my $command = shift;
my $args = shift;
my $content = shift;
my ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
$command);
my $caption_text = '';
my $prepended_text;
my $prepended_save = '';
if ($self->in_string()) {
if ($prepended) {
$prepended_text = $self->convert_tree_new_formatting_context(
$prepended, 'float prepended');
} else {
$prepended_text = '';
}
if ($caption) {
$caption_text = $self->convert_tree_new_formatting_context(
{'contents' => $caption->{'args'}->[0]->{'contents'}},
'float caption');
}
return $prepended.$content.$caption_text;
}
my $id = $self->command_id($command);
my $label;
if (defined($id) and $id ne '') {
$label = "<a name=\"$id\"></a>";
} else {
$label = '';
}
if ($prepended) {
if ($caption) {
# prepend the prepended tree to the first paragraph
my @caption_original_contents = @{$caption->{'args'}->[0]->{'contents'}};
my @caption_contents;
my $new_paragraph;
while (@caption_original_contents) {
my $content = shift @caption_original_contents;
if ($content->{'type'} and $content->{'type'} eq 'paragraph') {
%{$new_paragraph} = %{$content};
$new_paragraph->{'contents'} = [@{$content->{'contents'}}];
unshift (@{$new_paragraph->{'contents'}}, {'cmdname' => 'strong',
'args' => [{'type' => 'brace_command_arg',
'contents' => [$prepended]}]});
push @caption_contents, $new_paragraph;
last;
} else {
push @caption_contents, $content;
}
}
push @caption_contents, @caption_original_contents;
if ($new_paragraph) {
$caption_text = $self->convert_tree_new_formatting_context(
{'contents' => \@caption_contents}, 'float caption');
$prepended_text = '';
}
}
if ($caption_text eq '') {
$prepended_text = $self->convert_tree_new_formatting_context(
$prepended, 'float prepended');
if ($prepended_text ne '') {
$prepended_save = $prepended_text;
$prepended_text = '<p><strong>'.$prepended_text.'</strong></p>';
}
}
} else {
$prepended_text = '';
}
if ($caption and $caption_text eq '') {
$caption_text = $self->convert_tree_new_formatting_context(
$caption->{'args'}->[0], 'float caption');
}
if ($prepended_text.$caption_text ne '') {
$prepended_text = $self->_attribute_class('div','float-caption'). '>'
. $prepended_text;
$caption_text .= '</div>';
}
my $html_class = '';
if ($prepended_save =~ /NOTE/) {
$html_class = 'info';
$prepended_text = '';
$caption_text = '';
} elsif ($prepended_save =~ /IMPORTANT/) {
$html_class = 'warning';
$prepended_text = '';
$caption_text = '';
}
return $self->_attribute_class('div', $html_class). '>' . "\n" .
$prepended_text . $caption_text . $content . '</div>';
}
texinfo_register_command_formatting('float',
\&ffmpeg_float);
1;

View File

@@ -332,12 +332,12 @@ $inf = pop @instack;
die "No filename or title\n" unless defined $fn && defined $tl;
# always use utf8
print "=encoding utf8\n\n";
$chapters{NAME} = "$fn \- $tl\n";
$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};
# always use utf8
print "=encoding utf8\n\n";
unshift @chapters_sequence, "NAME";
for $chapter (@chapters_sequence) {
if (exists $chapters{$chapter}) {

View File

@@ -782,9 +782,6 @@ large numbers (usually 2^53 and larger).
Round the value of expression @var{expr} upwards to the nearest
integer. For example, "ceil(1.5)" is "2.0".
@item clip(x, min, max)
Return the value of @var{x} clipped between @var{min} and @var{max}.
@item cos(x)
Compute cosine of @var{x}.
@@ -844,7 +841,7 @@ Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
Return 1.0 if @var{x} is NAN, 0.0 otherwise.
@item ld(var)
Load the value of the internal variable with number
Allow to load the value of the internal variable with number
@var{var}, which was previously stored with st(@var{var}, @var{expr}).
The function returns the loaded value.
@@ -912,7 +909,7 @@ Compute the square root of @var{expr}. This is equivalent to
Compute expression @code{1/(1 + exp(4*x))}.
@item st(var, expr)
Store the value of the expression @var{expr} in an internal
Allow to store the value of the expression @var{expr} in an internal
variable. @var{var} specifies the number of the variable where to
store the value, and it is a value ranging from 0 to 9. The function
returns the value stored in the internal variable.

View File

@@ -16,15 +16,16 @@ outputs the modified frame. The most simple way of doing this is to take a
similar filter. We'll pick edgedetect, but any other should do. You can look
for others using the `./ffmpeg -v 0 -filters|grep ' V->V '` command.
- sed 's/edgedetect/foobar/g;s/EdgeDetect/Foobar/g' libavfilter/vf_edgedetect.c > libavfilter/vf_foobar.c
- cp libavfilter/vf_{edgedetect,foobar}.c
- sed -i s/edgedetect/foobar/g -i libavfilter/vf_foobar.c
- sed -i s/EdgeDetect/Foobar/g -i libavfilter/vf_foobar.c
- edit libavfilter/Makefile, and add an entry for "foobar" following the
pattern of the other filters.
- edit libavfilter/allfilters.c, and add an entry for "foobar" following the
pattern of the other filters.
- ./configure ...
- make -j<whatever> ffmpeg
- ./ffmpeg -i http://samples.ffmpeg.org/image-samples/lena.pnm -vf foobar foobar.png
Note here: you can obviously use a random local image instead of a remote URL.
- ./ffmpeg -i tests/lena.pnm -vf foobar foobar.png
If everything went right, you should get a foobar.png with Lena edge-detected.

292
ffmpeg.c
View File

@@ -352,6 +352,7 @@ void term_init(void)
signal(SIGQUIT, sigterm_handler); /* Quit (POSIX). */
}
#endif
avformat_network_deinit();
signal(SIGINT , sigterm_handler); /* Interrupt (ANSI). */
signal(SIGTERM, sigterm_handler); /* Termination (ANSI). */
@@ -466,12 +467,7 @@ static void ffmpeg_cleanup(int ret)
}
for (i = 0; i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
AVBitStreamFilterContext *bsfc;
if (!ost)
continue;
bsfc = ost->bitstream_filters;
AVBitStreamFilterContext *bsfc = ost->bitstream_filters;
while (bsfc) {
AVBitStreamFilterContext *next = bsfc->next;
av_bitstream_filter_close(bsfc);
@@ -479,7 +475,6 @@ static void ffmpeg_cleanup(int ret)
}
ost->bitstream_filters = NULL;
av_frame_free(&ost->filtered_frame);
av_frame_free(&ost->last_frame);
av_parser_close(ost->parser);
@@ -488,9 +483,6 @@ static void ffmpeg_cleanup(int ret)
av_freep(&ost->avfilter);
av_freep(&ost->logfile_prefix);
av_freep(&ost->audio_channels_map);
ost->audio_channels_mapped = 0;
avcodec_free_context(&ost->enc_ctx);
av_freep(&output_streams[i]);
@@ -540,15 +532,6 @@ static void ffmpeg_cleanup(int ret)
term_exit();
}
void remove_avoptions(AVDictionary **a, AVDictionary *b)
{
AVDictionaryEntry *t = NULL;
while ((t = av_dict_get(b, "", t, AV_DICT_IGNORE_SUFFIX))) {
av_dict_set(a, t->key, NULL, AV_DICT_MATCH_CASE);
}
}
void assert_avoptions(AVDictionary *m)
{
AVDictionaryEntry *t;
@@ -595,14 +578,6 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
AVCodecContext *avctx = ost->st->codec;
int ret;
if (!ost->st->codec->extradata_size && ost->enc_ctx->extradata_size) {
ost->st->codec->extradata = av_mallocz(ost->enc_ctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (ost->st->codec->extradata) {
memcpy(ost->st->codec->extradata, ost->enc_ctx->extradata, ost->enc_ctx->extradata_size);
ost->st->codec->extradata_size = ost->enc_ctx->extradata_size;
}
}
if ((avctx->codec_type == AVMEDIA_TYPE_VIDEO && video_sync_method == VSYNC_DROP) ||
(avctx->codec_type == AVMEDIA_TYPE_AUDIO && audio_sync_method < 0))
pkt->pts = pkt->dts = AV_NOPTS_VALUE;
@@ -627,11 +602,7 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
while (bsfc) {
AVPacket new_pkt = *pkt;
AVDictionaryEntry *bsf_arg = av_dict_get(ost->bsf_args,
bsfc->filter->name,
NULL, 0);
int a = av_bitstream_filter_filter(bsfc, avctx,
bsf_arg ? bsf_arg->value : NULL,
int a = av_bitstream_filter_filter(bsfc, avctx, NULL,
&new_pkt.data, &new_pkt.size,
pkt->data, pkt->size,
pkt->flags & AV_PKT_FLAG_KEY);
@@ -647,15 +618,12 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
a = AVERROR(ENOMEM);
}
if (a > 0) {
pkt->side_data = NULL;
pkt->side_data_elems = 0;
av_free_packet(pkt);
new_pkt.buf = av_buffer_create(new_pkt.data, new_pkt.size,
av_buffer_default_free, NULL, 0);
if (!new_pkt.buf)
exit_program(1);
} else if (a < 0) {
new_pkt = *pkt;
av_log(NULL, AV_LOG_ERROR, "Failed to open bitstream filter %s for stream %d with codec %s",
bsfc->filter->name, pkt->stream_index,
avctx->codec ? avctx->codec->name : "copy");
@@ -669,17 +637,6 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
}
if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS)) {
if (pkt->dts != AV_NOPTS_VALUE &&
pkt->pts != AV_NOPTS_VALUE &&
pkt->dts > pkt->pts) {
av_log(s, AV_LOG_WARNING, "Invalid DTS: %"PRId64" PTS: %"PRId64" in output stream %d:%d, replacing by guess\n",
pkt->dts, pkt->pts,
ost->file_index, ost->st->index);
pkt->pts =
pkt->dts = pkt->pts + pkt->dts + ost->last_mux_dts + 1
- FFMIN3(pkt->pts, pkt->dts, ost->last_mux_dts + 1)
- FFMAX3(pkt->pts, pkt->dts, ost->last_mux_dts + 1);
}
if(
(avctx->codec_type == AVMEDIA_TYPE_AUDIO || avctx->codec_type == AVMEDIA_TYPE_VIDEO) &&
pkt->dts != AV_NOPTS_VALUE &&
@@ -702,6 +659,15 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
pkt->dts = max;
}
}
if (pkt->dts != AV_NOPTS_VALUE &&
pkt->pts != AV_NOPTS_VALUE &&
pkt->dts > pkt->pts) {
av_log(s, AV_LOG_WARNING, "Invalid DTS: %"PRId64" PTS: %"PRId64" in output stream %d:%d\n",
pkt->dts, pkt->pts,
ost->file_index, ost->st->index);
pkt->pts = AV_NOPTS_VALUE;
pkt->dts = AV_NOPTS_VALUE;
}
}
ost->last_mux_dts = pkt->dts;
@@ -884,14 +850,14 @@ static void do_subtitle_out(AVFormatContext *s,
static void do_video_out(AVFormatContext *s,
OutputStream *ost,
AVFrame *next_picture)
AVFrame *in_picture)
{
int ret, format_video_sync;
AVPacket pkt;
AVCodecContext *enc = ost->enc_ctx;
AVCodecContext *mux_enc = ost->st->codec;
int nb_frames, nb0_frames, i;
double sync_ipts, delta, delta0;
int nb_frames, i;
double sync_ipts, delta;
double duration = 0;
int frame_size = 0;
InputStream *ist = NULL;
@@ -902,20 +868,10 @@ static void do_video_out(AVFormatContext *s,
if(ist && ist->st->start_time != AV_NOPTS_VALUE && ist->st->first_dts != AV_NOPTS_VALUE && ost->frame_rate.num)
duration = 1/(av_q2d(ost->frame_rate) * av_q2d(enc->time_base));
if (!ost->filters_script &&
!ost->filters &&
next_picture &&
ist &&
lrintf(av_frame_get_pkt_duration(next_picture) * av_q2d(ist->st->time_base) / av_q2d(enc->time_base)) > 0) {
duration = lrintf(av_frame_get_pkt_duration(next_picture) * av_q2d(ist->st->time_base) / av_q2d(enc->time_base));
}
sync_ipts = next_picture->pts;
delta0 = sync_ipts - ost->sync_opts;
delta = delta0 + duration;
sync_ipts = in_picture->pts;
delta = sync_ipts - ost->sync_opts + duration;
/* by default, we output a single frame */
nb0_frames = 0;
nb_frames = 1;
format_video_sync = video_sync_method;
@@ -935,34 +891,19 @@ static void do_video_out(AVFormatContext *s,
}
}
if (delta0 < 0 &&
delta > 0 &&
format_video_sync != VSYNC_PASSTHROUGH &&
format_video_sync != VSYNC_DROP) {
double cor = FFMIN(-delta0, duration);
av_log(NULL, AV_LOG_WARNING, "Past duration %f too large\n", -delta0);
sync_ipts += cor;
duration -= cor;
delta0 += cor;
}
switch (format_video_sync) {
case VSYNC_VSCFR:
if (ost->frame_number == 0 && delta - duration >= 0.5) {
av_log(NULL, AV_LOG_DEBUG, "Not duplicating %d initial frames\n", (int)lrintf(delta - duration));
delta = duration;
delta0 = 0;
ost->sync_opts = lrint(sync_ipts);
}
case VSYNC_CFR:
// FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
if (delta < -1.1)
nb_frames = 0;
else if (delta > 1.1) {
else if (delta > 1.1)
nb_frames = lrintf(delta);
if (delta0 > 1.1)
nb0_frames = lrintf(delta0 - 0.6);
}
break;
case VSYNC_VFR:
if (delta <= -0.6)
@@ -979,36 +920,28 @@ static void do_video_out(AVFormatContext *s,
}
nb_frames = FFMIN(nb_frames, ost->max_frames - ost->frame_number);
nb0_frames = FFMIN(nb0_frames, nb_frames);
if (nb0_frames == 0 && ost->last_droped) {
if (nb_frames == 0) {
nb_frames_drop++;
av_log(NULL, AV_LOG_VERBOSE,
"*** dropping frame %d from stream %d at ts %"PRId64"\n",
ost->frame_number, ost->st->index, ost->last_frame->pts);
}
if (nb_frames > (nb0_frames && ost->last_droped) + (nb_frames > nb0_frames)) {
ost->frame_number, ost->st->index, in_picture->pts);
return;
} else if (nb_frames > 1) {
if (nb_frames > dts_error_threshold * 30) {
av_log(NULL, AV_LOG_ERROR, "%d frame duplication too large, skipping\n", nb_frames - 1);
nb_frames_drop++;
return;
}
nb_frames_dup += nb_frames - (nb0_frames && ost->last_droped) - (nb_frames > nb0_frames);
nb_frames_dup += nb_frames - 1;
av_log(NULL, AV_LOG_VERBOSE, "*** %d dup!\n", nb_frames - 1);
}
ost->last_droped = nb_frames == nb0_frames;
/* duplicates frame if needed */
for (i = 0; i < nb_frames; i++) {
AVFrame *in_picture;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (i < nb0_frames && ost->last_frame) {
in_picture = ost->last_frame;
} else
in_picture = next_picture;
in_picture->pts = ost->sync_opts;
#if 1
@@ -1023,8 +956,10 @@ static void do_video_out(AVFormatContext *s,
/* raw pictures are written as AVPicture structure to
avoid any copies. We support temporarily the older
method. */
if (in_picture->interlaced_frame)
mux_enc->field_order = in_picture->top_field_first ? AV_FIELD_TB:AV_FIELD_BT;
mux_enc->coded_frame->interlaced_frame = in_picture->interlaced_frame;
mux_enc->coded_frame->top_field_first = in_picture->top_field_first;
if (mux_enc->coded_frame->interlaced_frame)
mux_enc->field_order = mux_enc->coded_frame->top_field_first ? AV_FIELD_TB:AV_FIELD_BT;
else
mux_enc->field_order = AV_FIELD_PROGRESSIVE;
pkt.data = (uint8_t *)in_picture;
@@ -1050,7 +985,8 @@ static void do_video_out(AVFormatContext *s,
mux_enc->field_order = AV_FIELD_PROGRESSIVE;
in_picture->quality = enc->global_quality;
in_picture->pict_type = 0;
if (!enc->me_threshold)
in_picture->pict_type = 0;
pts_time = in_picture->pts != AV_NOPTS_VALUE ?
in_picture->pts * av_q2d(enc->time_base) : NAN;
@@ -1144,14 +1080,6 @@ static void do_video_out(AVFormatContext *s,
if (vstats_filename && frame_size)
do_video_stats(ost, frame_size);
}
if (!ost->last_frame)
ost->last_frame = av_frame_alloc();
av_frame_unref(ost->last_frame);
if (next_picture && ost->last_frame)
av_frame_ref(ost->last_frame, next_picture);
else
av_frame_free(&ost->last_frame);
}
static double psnr(double d)
@@ -1320,6 +1248,7 @@ static void print_final_stats(int64_t total_size)
if (data_size && total_size>0 && total_size >= data_size)
percent = 100.0 * (total_size - data_size) / data_size;
av_log(NULL, AV_LOG_INFO, "\n");
av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB subtitle:%1.0fkB other streams:%1.0fkB global headers:%1.0fkB muxing overhead: ",
video_size / 1024.0,
audio_size / 1024.0,
@@ -1510,8 +1439,6 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
if (av_stream_get_end_pts(ost->st) != AV_NOPTS_VALUE)
pts = FFMAX(pts, av_rescale_q(av_stream_get_end_pts(ost->st),
ost->st->time_base, AV_TIME_BASE_Q));
if (is_last_report)
nb_frames_drop += ost->last_droped;
}
secs = pts / AV_TIME_BASE;
@@ -1547,11 +1474,10 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
av_bprintf(&buf_script, "drop_frames=%d\n", nb_frames_drop);
if (print_stats || is_last_report) {
const char end = is_last_report ? '\n' : '\r';
if (print_stats==1 && AV_LOG_INFO > av_log_get_level()) {
fprintf(stderr, "%s %c", buf, end);
fprintf(stderr, "%s \r", buf);
} else
av_log(NULL, AV_LOG_INFO, "%s %c", buf, end);
av_log(NULL, AV_LOG_INFO, "%s \r", buf);
fflush(stderr);
}
@@ -1736,21 +1662,17 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->st->time_base);
opkt.flags = pkt->flags;
// FIXME remove the following 2 lines they shall be replaced by the bitstream filters
if ( ost->st->codec->codec_id != AV_CODEC_ID_H264
&& ost->st->codec->codec_id != AV_CODEC_ID_MPEG1VIDEO
&& ost->st->codec->codec_id != AV_CODEC_ID_MPEG2VIDEO
&& ost->st->codec->codec_id != AV_CODEC_ID_VC1
if ( ost->enc_ctx->codec_id != AV_CODEC_ID_H264
&& ost->enc_ctx->codec_id != AV_CODEC_ID_MPEG1VIDEO
&& ost->enc_ctx->codec_id != AV_CODEC_ID_MPEG2VIDEO
&& ost->enc_ctx->codec_id != AV_CODEC_ID_VC1
) {
int ret = av_parser_change(ost->parser, ost->st->codec,
if (av_parser_change(ost->parser, ost->st->codec,
&opkt.data, &opkt.size,
pkt->data, pkt->size,
pkt->flags & AV_PKT_FLAG_KEY);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "av_parser_change failed\n");
exit_program(1);
}
if (ret) {
pkt->flags & AV_PKT_FLAG_KEY)) {
opkt.buf = av_buffer_create(opkt.data, opkt.size, av_buffer_default_free, NULL, 0);
if (!opkt.buf)
exit_program(1);
@@ -1761,15 +1683,9 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
}
av_copy_packet_side_data(&opkt, pkt);
if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO &&
ost->st->codec->codec_id == AV_CODEC_ID_RAWVIDEO &&
(of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
/* store AVPicture in AVPacket, as expected by the output format */
int ret = avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "avpicture_fill failed\n");
exit_program(1);
}
avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height);
opkt.data = (uint8_t *)&pict;
opkt.size = sizeof(AVPicture);
opkt.flags |= AV_PKT_FLAG_KEY;
@@ -1883,10 +1799,18 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
for (i = 0; i < nb_filtergraphs; i++)
if (ist_in_filtergraph(filtergraphs[i], ist)) {
FilterGraph *fg = filtergraphs[i];
int j;
if (configure_filtergraph(fg) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error reinitializing filters!\n");
exit_program(1);
}
for (j = 0; j < fg->nb_outputs; j++) {
OutputStream *ost = fg->outputs[j]->ost;
if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
!(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
av_buffersink_set_frame_size(ost->filter->filter,
ost->enc_ctx->frame_size);
}
}
}
@@ -1968,20 +1892,6 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
if (*got_output || ret<0 || pkt->size)
decode_error_stat[ret<0] ++;
if (*got_output && ret >= 0) {
if (ist->dec_ctx->width != decoded_frame->width ||
ist->dec_ctx->height != decoded_frame->height ||
ist->dec_ctx->pix_fmt != decoded_frame->format) {
av_log(NULL, AV_LOG_DEBUG, "Frame parameters mismatch context %d,%d,%d != %d,%d,%d\n",
decoded_frame->width,
decoded_frame->height,
decoded_frame->format,
ist->dec_ctx->width,
ist->dec_ctx->height,
ist->dec_ctx->pix_fmt);
}
}
if (!*got_output || ret < 0) {
if (!pkt->size) {
for (i = 0; i < ist->nb_filters; i++)
@@ -2138,7 +2048,7 @@ out:
}
/* pkt = NULL means EOF (needed to flush decoder buffers) */
static int process_input_packet(InputStream *ist, const AVPacket *pkt)
static int output_packet(InputStream *ist, const AVPacket *pkt)
{
int ret = 0, i;
int got_output = 0;
@@ -2147,7 +2057,7 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
if (!ist->saw_first_ts) {
ist->dts = ist->st->avg_frame_rate.num ? - ist->dec_ctx->has_b_frames * AV_TIME_BASE / av_q2d(ist->st->avg_frame_rate) : 0;
ist->pts = 0;
if (pkt && pkt->pts != AV_NOPTS_VALUE && !ist->decoding_needed) {
if (pkt != NULL && pkt->pts != AV_NOPTS_VALUE && !ist->decoding_needed) {
ist->dts += av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q);
ist->pts = ist->dts; //unused but better to set it to a value thats not totally wrong
}
@@ -2159,7 +2069,7 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
if (ist->next_pts == AV_NOPTS_VALUE)
ist->next_pts = ist->pts;
if (!pkt) {
if (pkt == NULL) {
/* EOF handling */
av_init_packet(&avpkt);
avpkt.data = NULL;
@@ -2198,11 +2108,11 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
ret = decode_video (ist, &avpkt, &got_output);
if (avpkt.duration) {
duration = av_rescale_q(avpkt.duration, ist->st->time_base, AV_TIME_BASE_Q);
} else if(ist->dec_ctx->framerate.num != 0 && ist->dec_ctx->framerate.den != 0) {
} else if(ist->dec_ctx->time_base.num != 0 && ist->dec_ctx->time_base.den != 0) {
int ticks= av_stream_get_parser(ist->st) ? av_stream_get_parser(ist->st)->repeat_pict+1 : ist->dec_ctx->ticks_per_frame;
duration = ((int64_t)AV_TIME_BASE *
ist->dec_ctx->framerate.den * ticks) /
ist->dec_ctx->framerate.num / ist->dec_ctx->ticks_per_frame;
ist->dec_ctx->time_base.num * ticks) /
ist->dec_ctx->time_base.den;
} else
duration = 0;
@@ -2257,11 +2167,11 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
ist->next_dts = av_rescale_q(next_dts + 1, av_inv_q(ist->framerate), time_base_q);
} else if (pkt->duration) {
ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q);
} else if(ist->dec_ctx->framerate.num != 0) {
} else if(ist->dec_ctx->time_base.num != 0) {
int ticks= av_stream_get_parser(ist->st) ? av_stream_get_parser(ist->st)->repeat_pict + 1 : ist->dec_ctx->ticks_per_frame;
ist->next_dts += ((int64_t)AV_TIME_BASE *
ist->dec_ctx->framerate.den * ticks) /
ist->dec_ctx->framerate.num / ist->dec_ctx->ticks_per_frame;
ist->dec_ctx->time_base.num * ticks) /
ist->dec_ctx->time_base.den;
}
break;
}
@@ -2373,12 +2283,8 @@ static int init_input_stream(int ist_index, char *error, int error_len)
ist->dec_ctx->thread_safe_callbacks = 1;
av_opt_set_int(ist->dec_ctx, "refcounted_frames", 1, 0);
if (ist->dec_ctx->codec_id == AV_CODEC_ID_DVB_SUBTITLE &&
(ist->decoding_needed & DECODING_FOR_OST)) {
av_dict_set(&ist->decoder_opts, "compute_edt", "1", AV_DICT_DONT_OVERWRITE);
if (ist->decoding_needed & DECODING_FOR_FILTER)
av_log(NULL, AV_LOG_WARNING, "Warning using DVB subtitles for filtering and output at the same time is not fully supported, also see -compute_edt [0|1]\n");
}
if(ist->dec_ctx->codec_id == AV_CODEC_ID_DVB_SUBTITLE)
av_dict_set(&ist->decoder_opts, "compute_edt", "1", 0);
if (!av_dict_get(ist->decoder_opts, "threads", NULL, 0))
av_dict_set(&ist->decoder_opts, "threads", "auto", 0);
@@ -2539,7 +2445,7 @@ static int transcode_init(void)
AVFormatContext *oc;
OutputStream *ost;
InputStream *ist;
char error[1024] = {0};
char error[1024];
int want_sdp = 1;
for (i = 0; i < nb_filtergraphs; i++) {
@@ -2638,13 +2544,11 @@ static int transcode_init(void)
enc_ctx->rc_max_rate = dec_ctx->rc_max_rate;
enc_ctx->rc_buffer_size = dec_ctx->rc_buffer_size;
enc_ctx->field_order = dec_ctx->field_order;
if (dec_ctx->extradata_size) {
enc_ctx->extradata = av_mallocz(extra_size);
if (!enc_ctx->extradata) {
return AVERROR(ENOMEM);
}
memcpy(enc_ctx->extradata, dec_ctx->extradata, dec_ctx->extradata_size);
enc_ctx->extradata = av_mallocz(extra_size);
if (!enc_ctx->extradata) {
return AVERROR(ENOMEM);
}
memcpy(enc_ctx->extradata, dec_ctx->extradata, dec_ctx->extradata_size);
enc_ctx->extradata_size= dec_ctx->extradata_size;
enc_ctx->bits_per_coded_sample = dec_ctx->bits_per_coded_sample;
@@ -2699,26 +2603,6 @@ static int transcode_init(void)
av_reduce(&enc_ctx->time_base.num, &enc_ctx->time_base.den,
enc_ctx->time_base.num, enc_ctx->time_base.den, INT_MAX);
if (ist->st->nb_side_data) {
ost->st->side_data = av_realloc_array(NULL, ist->st->nb_side_data,
sizeof(*ist->st->side_data));
if (!ost->st->side_data)
return AVERROR(ENOMEM);
for (j = 0; j < ist->st->nb_side_data; j++) {
const AVPacketSideData *sd_src = &ist->st->side_data[j];
AVPacketSideData *sd_dst = &ost->st->side_data[j];
sd_dst->data = av_malloc(sd_src->size);
if (!sd_dst->data)
return AVERROR(ENOMEM);
memcpy(sd_dst->data, sd_src->data, sd_src->size);
sd_dst->size = sd_src->size;
sd_dst->type = sd_src->type;
ost->st->nb_side_data++;
}
}
ost->parser = av_parser_init(enc_ctx->codec_id);
switch (enc_ctx->codec_type) {
@@ -2733,10 +2617,7 @@ static int transcode_init(void)
enc_ctx->frame_size = dec_ctx->frame_size;
enc_ctx->audio_service_type = dec_ctx->audio_service_type;
enc_ctx->block_align = dec_ctx->block_align;
enc_ctx->initial_padding = dec_ctx->delay;
#if FF_API_AUDIOENC_DELAY
enc_ctx->delay = dec_ctx->delay;
#endif
if((enc_ctx->block_align == 1 || enc_ctx->block_align == 1152 || enc_ctx->block_align == 576) && enc_ctx->codec_id == AV_CODEC_ID_MP3)
enc_ctx->block_align= 0;
if(enc_ctx->codec_id == AV_CODEC_ID_AC3)
@@ -2783,7 +2664,7 @@ static int transcode_init(void)
}
if (ist)
ist->decoding_needed |= DECODING_FOR_OST;
ist->decoding_needed++;
ost->encoding_needed = 1;
set_encoder_id(output_files[ost->file_index], ost);
@@ -2983,11 +2864,10 @@ static int transcode_init(void)
av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
" It takes bits/s as argument, not kbits/s\n");
} else {
ret = av_opt_set_dict(ost->enc_ctx, &ost->encoder_opts);
if (ret < 0) {
if (av_opt_set_dict(ost->enc_ctx, &ost->encoder_opts) < 0) {
av_log(NULL, AV_LOG_FATAL,
"Error setting up codec context options.\n");
return ret;
exit_program(1);
}
}
@@ -3187,9 +3067,9 @@ static OutputStream *choose_output(void)
OutputStream *ost = output_streams[i];
int64_t opts = av_rescale_q(ost->st->cur_dts, ost->st->time_base,
AV_TIME_BASE_Q);
if (!ost->finished && opts < opts_min) {
if (!ost->unavailable && !ost->finished && opts < opts_min) {
opts_min = opts;
ost_min = ost->unavailable ? NULL : ost;
ost_min = ost;
}
}
return ost_min;
@@ -3451,7 +3331,7 @@ static int process_input(int file_index)
for (i = 0; i < ifile->nb_streams; i++) {
ist = input_streams[ifile->ist_index + i];
if (ist->decoding_needed) {
ret = process_input_packet(ist, NULL);
ret = output_packet(ist, NULL);
if (ret>0)
return 0;
}
@@ -3539,7 +3419,7 @@ static int process_input(int file_index)
}
/* add the stream-global side data to the first packet */
if (ist->nb_packets == 1) {
if (ist->nb_packets == 1)
if (ist->st->nb_side_data)
av_packet_split_side_data(&pkt);
for (i = 0; i < ist->st->nb_side_data; i++) {
@@ -3555,7 +3435,6 @@ static int process_input(int file_index)
memcpy(dst_data, src_sd->data, src_sd->size);
}
}
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts += av_rescale_q(ifile->ts_offset, AV_TIME_BASE_Q, ist->st->time_base);
@@ -3567,14 +3446,13 @@ static int process_input(int file_index)
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts *= ist->ts_scale;
if ((ist->dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO ||
ist->dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) &&
pkt.dts != AV_NOPTS_VALUE && ist->next_dts == AV_NOPTS_VALUE && !copy_ts
if (pkt.dts != AV_NOPTS_VALUE && ist->next_dts == AV_NOPTS_VALUE && !copy_ts
&& (is->iformat->flags & AVFMT_TS_DISCONT) && ifile->last_ts != AV_NOPTS_VALUE) {
int64_t pkt_dts = av_rescale_q(pkt.dts, ist->st->time_base, AV_TIME_BASE_Q);
int64_t delta = pkt_dts - ifile->last_ts;
if (delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
delta > 1LL*dts_delta_threshold*AV_TIME_BASE){
if(delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
(delta > 1LL*dts_delta_threshold*AV_TIME_BASE &&
ist->dec_ctx->codec_type != AVMEDIA_TYPE_SUBTITLE)){
ifile->ts_offset -= delta;
av_log(NULL, AV_LOG_DEBUG,
"Inter stream timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
@@ -3585,15 +3463,14 @@ static int process_input(int file_index)
}
}
if ((ist->dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO ||
ist->dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) &&
pkt.dts != AV_NOPTS_VALUE && ist->next_dts != AV_NOPTS_VALUE &&
if (pkt.dts != AV_NOPTS_VALUE && ist->next_dts != AV_NOPTS_VALUE &&
!copy_ts) {
int64_t pkt_dts = av_rescale_q(pkt.dts, ist->st->time_base, AV_TIME_BASE_Q);
int64_t delta = pkt_dts - ist->next_dts;
if (is->iformat->flags & AVFMT_TS_DISCONT) {
if (delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
delta > 1LL*dts_delta_threshold*AV_TIME_BASE ||
(delta > 1LL*dts_delta_threshold*AV_TIME_BASE &&
ist->dec_ctx->codec_type != AVMEDIA_TYPE_SUBTITLE) ||
pkt_dts + AV_TIME_BASE/10 < FFMAX(ist->pts, ist->dts)) {
ifile->ts_offset -= delta;
av_log(NULL, AV_LOG_DEBUG,
@@ -3605,7 +3482,7 @@ static int process_input(int file_index)
}
} else {
if ( delta < -1LL*dts_error_threshold*AV_TIME_BASE ||
delta > 1LL*dts_error_threshold*AV_TIME_BASE) {
(delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->dec_ctx->codec_type != AVMEDIA_TYPE_SUBTITLE)) {
av_log(NULL, AV_LOG_WARNING, "DTS %"PRId64", next:%"PRId64" st:%d invalid dropping\n", pkt.dts, ist->next_dts, pkt.stream_index);
pkt.dts = AV_NOPTS_VALUE;
}
@@ -3613,7 +3490,7 @@ static int process_input(int file_index)
int64_t pkt_pts = av_rescale_q(pkt.pts, ist->st->time_base, AV_TIME_BASE_Q);
delta = pkt_pts - ist->next_dts;
if ( delta < -1LL*dts_error_threshold*AV_TIME_BASE ||
delta > 1LL*dts_error_threshold*AV_TIME_BASE) {
(delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->dec_ctx->codec_type != AVMEDIA_TYPE_SUBTITLE)) {
av_log(NULL, AV_LOG_WARNING, "PTS %"PRId64", next:%"PRId64" invalid dropping st:%d\n", pkt.pts, ist->next_dts, pkt.stream_index);
pkt.pts = AV_NOPTS_VALUE;
}
@@ -3635,7 +3512,7 @@ static int process_input(int file_index)
sub2video_heartbeat(ist, pkt.pts);
ret = process_input_packet(ist, &pkt);
ret = output_packet(ist, &pkt);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d: %s\n",
ist->file_index, ist->st->index, av_err2str(ret));
@@ -3801,7 +3678,7 @@ static int transcode(void)
for (i = 0; i < nb_input_streams; i++) {
ist = input_streams[i];
if (!input_files[ist->file_index]->eof_reached && ist->decoding_needed) {
process_input_packet(ist, NULL);
output_packet(ist, NULL);
}
}
flush_encoders();
@@ -3856,7 +3733,6 @@ static int transcode(void)
av_dict_free(&ost->encoder_opts);
av_dict_free(&ost->swr_opts);
av_dict_free(&ost->resample_opts);
av_dict_free(&ost->bsf_args);
}
}
}
@@ -3878,7 +3754,7 @@ static int64_t getutime(void)
GetProcessTimes(proc, &c, &e, &k, &u);
return ((int64_t) u.dwHighDateTime << 32 | u.dwLowDateTime) / 10;
#else
return av_gettime_relative();
return av_gettime();
#endif
}

View File

@@ -243,10 +243,7 @@ typedef struct InputStream {
AVStream *st;
int discard; /* true if stream data should be discarded */
int user_set_discard;
int decoding_needed; /* non zero if the packets must be decoded in 'raw_fifo', see DECODING_FOR_* */
#define DECODING_FOR_OST 1
#define DECODING_FOR_FILTER 2
int decoding_needed; /* true if the packets must be decoded in 'raw_fifo' */
AVCodecContext *dec_ctx;
AVCodec *dec;
AVFrame *decoded_frame;
@@ -388,8 +385,6 @@ typedef struct OutputStream {
AVCodec *enc;
int64_t max_frames;
AVFrame *filtered_frame;
AVFrame *last_frame;
int last_droped;
/* video only */
AVRational frame_rate;
@@ -407,7 +402,7 @@ typedef struct OutputStream {
double forced_keyframes_expr_const_values[FKF_NB];
/* audio only */
int *audio_channels_map; /* list of the channels id to pick from the source stream */
int audio_channels_map[SWR_CH_MAX]; /* list of the channels id to pick from the source stream */
int audio_channels_mapped; /* number of channels in audio_channels_map */
char *logfile_prefix;
@@ -422,7 +417,6 @@ typedef struct OutputStream {
AVDictionary *encoder_opts;
AVDictionary *swr_opts;
AVDictionary *resample_opts;
AVDictionary *bsf_args;
char *apad;
OSTFinished finished; /* no more packets should be written for this stream */
int unavailable; /* true if the steram is unavailable (possibly temporarily) */
@@ -484,7 +478,6 @@ extern int do_deinterlace;
extern int do_hex_dump;
extern int do_pkt_dump;
extern int copy_ts;
extern int start_at_zero;
extern int copy_tb;
extern int debug_ts;
extern int exit_on_error;
@@ -494,7 +487,6 @@ extern int stdin_interaction;
extern int frame_bits_per_raw_sample;
extern AVIOContext *progress_avio;
extern float max_error_rate;
extern int vdpau_api_ver;
extern const AVIOInterruptCB int_cb;
@@ -510,7 +502,6 @@ void show_usage(void);
void opt_output_file(void *optctx, const char *filename);
void remove_avoptions(AVDictionary **a, AVDictionary *b);
void assert_avoptions(AVDictionary *m);
int guess_input_channel_layout(InputStream *ist);

View File

@@ -275,7 +275,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
av_assert0(ist);
ist->discard = 0;
ist->decoding_needed |= DECODING_FOR_FILTER;
ist->decoding_needed++;
ist->st->discard = AVDISCARD_NONE;
GROW_ARRAY(fg->inputs, fg->nb_inputs);
@@ -383,8 +383,9 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
snprintf(name, sizeof(name), "pixel format for output stream %d:%d",
ost->file_index, ost->index);
ret = avfilter_graph_create_filter(&filter,
avfilter_get_by_name("format"),
"format", pix_fmts, NULL, fg->graph);
avfilter_get_by_name("format"),
"format", pix_fmts, NULL,
fg->graph);
av_freep(&pix_fmts);
if (ret < 0)
return ret;
@@ -619,7 +620,6 @@ static int sub2video_prepare(InputStream *ist)
ist->sub2video.frame = av_frame_alloc();
if (!ist->sub2video.frame)
return AVERROR(ENOMEM);
ist->sub2video.last_pts = INT64_MIN;
return 0;
}
@@ -637,7 +637,6 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
AVBPrint args;
char name[255];
int ret, pad_idx = 0;
int64_t tsoffset = 0;
if (ist->dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
av_log(NULL, AV_LOG_ERROR, "Cannot connect video filter to audio input\n");
@@ -712,14 +711,8 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
snprintf(name, sizeof(name), "trim for input stream %d:%d",
ist->file_index, ist->st->index);
if (copy_ts) {
tsoffset = f->start_time == AV_NOPTS_VALUE ? 0 : f->start_time;
if (!start_at_zero && f->ctx->start_time != AV_NOPTS_VALUE)
tsoffset += f->ctx->start_time;
}
ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
AV_NOPTS_VALUE : tsoffset, f->recording_time,
&last_filter, &pad_idx, name);
AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
if (ret < 0)
return ret;
@@ -738,7 +731,6 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
AVBPrint args;
char name[255];
int ret, pad_idx = 0;
int64_t tsoffset = 0;
if (ist->dec_ctx->codec_type != AVMEDIA_TYPE_AUDIO) {
av_log(NULL, AV_LOG_ERROR, "Cannot connect audio filter to non audio input\n");
@@ -821,14 +813,8 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
snprintf(name, sizeof(name), "trim for input stream %d:%d",
ist->file_index, ist->st->index);
if (copy_ts) {
tsoffset = f->start_time == AV_NOPTS_VALUE ? 0 : f->start_time;
if (!start_at_zero && f->ctx->start_time != AV_NOPTS_VALUE)
tsoffset += f->ctx->start_time;
}
ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
AV_NOPTS_VALUE : tsoffset, f->recording_time,
&last_filter, &pad_idx, name);
AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
if (ret < 0)
return ret;
@@ -844,12 +830,6 @@ static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
av_freep(&ifilter->name);
DESCRIBE_FILTER_LINK(ifilter, in, 1);
if (!ifilter->ist->dec) {
av_log(NULL, AV_LOG_ERROR,
"No decoder for stream #%d:%d, filtering impossible\n",
ifilter->ist->file_index, ifilter->ist->st->index);
return AVERROR_DECODER_NOT_FOUND;
}
switch (avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx)) {
case AVMEDIA_TYPE_VIDEO: return configure_input_video_filter(fg, ifilter, in);
case AVMEDIA_TYPE_AUDIO: return configure_input_audio_filter(fg, ifilter, in);
@@ -912,11 +892,8 @@ int configure_filtergraph(FilterGraph *fg)
init_input_filter(fg, cur);
for (cur = inputs, i = 0; cur; cur = cur->next, i++)
if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0) {
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0)
return ret;
}
avfilter_inout_free(&inputs);
if (!init || simple) {
@@ -942,16 +919,6 @@ int configure_filtergraph(FilterGraph *fg)
}
fg->reconfiguration = 1;
for (i = 0; i < fg->nb_outputs; i++) {
OutputStream *ost = fg->outputs[i]->ost;
if (ost &&
ost->enc->type == AVMEDIA_TYPE_AUDIO &&
!(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
av_buffersink_set_frame_size(ost->filter->filter,
ost->enc_ctx->frame_size);
}
return 0;
}

View File

@@ -91,7 +91,6 @@ int do_benchmark_all = 0;
int do_hex_dump = 0;
int do_pkt_dump = 0;
int copy_ts = 0;
int start_at_zero = 0;
int copy_tb = -1;
int debug_ts = 0;
int exit_on_error = 0;
@@ -509,8 +508,7 @@ static int opt_recording_timestamp(void *optctx, const char *opt, const char *ar
char buf[128];
int64_t recording_timestamp = parse_time_or_die(opt, arg, 0) / 1E6;
struct tm time = *gmtime((time_t*)&recording_timestamp);
if (!strftime(buf, sizeof(buf), "creation_time=%FT%T%z", &time))
return -1;
strftime(buf, sizeof(buf), "creation_time=%FT%T%z", &time);
parse_option(o, "metadata", buf, options);
av_log(NULL, AV_LOG_WARNING, "%s is deprecated, set the 'creation_time' metadata "
@@ -704,7 +702,7 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
MATCH_PER_STREAM_OPT(fix_sub_duration, i, ist->fix_sub_duration, ic, st);
MATCH_PER_STREAM_OPT(canvas_sizes, str, canvas_size, ic, st);
if (canvas_size &&
av_parse_video_size(&ist->dec_ctx->width, &ist->dec_ctx->height, canvas_size) < 0) {
av_parse_video_size(&dec->width, &dec->height, canvas_size) < 0) {
av_log(NULL, AV_LOG_FATAL, "Invalid canvas size: %s.\n", canvas_size);
exit_program(1);
}
@@ -787,6 +785,7 @@ static int open_input_file(OptionsContext *o, const char *filename)
AVInputFormat *file_iformat = NULL;
int err, i, ret;
int64_t timestamp;
uint8_t buf[128];
AVDictionary **opts;
AVDictionary *unused_opts = NULL;
AVDictionaryEntry *e = NULL;
@@ -794,7 +793,6 @@ static int open_input_file(OptionsContext *o, const char *filename)
char * video_codec_name = NULL;
char * audio_codec_name = NULL;
char *subtitle_codec_name = NULL;
int scan_all_pmts_set = 0;
if (o->format) {
if (!(file_iformat = av_find_input_format(o->format))) {
@@ -816,7 +814,8 @@ static int open_input_file(OptionsContext *o, const char *filename)
exit_program(1);
}
if (o->nb_audio_sample_rate) {
av_dict_set_int(&o->g->format_opts, "sample_rate", o->audio_sample_rate[o->nb_audio_sample_rate - 1].u.i, 0);
snprintf(buf, sizeof(buf), "%d", o->audio_sample_rate[o->nb_audio_sample_rate - 1].u.i);
av_dict_set(&o->g->format_opts, "sample_rate", buf, 0);
}
if (o->nb_audio_channels) {
/* because we set audio_channels based on both the "ac" and
@@ -825,7 +824,9 @@ static int open_input_file(OptionsContext *o, const char *filename)
if (file_iformat && file_iformat->priv_class &&
av_opt_find(&file_iformat->priv_class, "channels", NULL, 0,
AV_OPT_SEARCH_FAKE_OBJ)) {
av_dict_set_int(&o->g->format_opts, "channels", o->audio_channels[o->nb_audio_channels - 1].u.i, 0);
snprintf(buf, sizeof(buf), "%d",
o->audio_channels[o->nb_audio_channels - 1].u.i);
av_dict_set(&o->g->format_opts, "channels", buf, 0);
}
}
if (o->nb_frame_rates) {
@@ -865,19 +866,12 @@ static int open_input_file(OptionsContext *o, const char *filename)
ic->flags |= AVFMT_FLAG_NONBLOCK;
ic->interrupt_callback = int_cb;
if (!av_dict_get(o->g->format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE)) {
av_dict_set(&o->g->format_opts, "scan_all_pmts", "1", AV_DICT_DONT_OVERWRITE);
scan_all_pmts_set = 1;
}
/* open the input file with generic avformat function */
err = avformat_open_input(&ic, filename, file_iformat, &o->g->format_opts);
if (err < 0) {
print_error(filename, err);
exit_program(1);
}
if (scan_all_pmts_set)
av_dict_set(&o->g->format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE);
remove_avoptions(&o->g->format_opts, o->g->codec_opts);
assert_avoptions(o->g->format_opts);
/* apply forced codec ids */
@@ -930,7 +924,7 @@ static int open_input_file(OptionsContext *o, const char *filename)
f->start_time = o->start_time;
f->recording_time = o->recording_time;
f->input_ts_offset = o->input_ts_offset;
f->ts_offset = o->input_ts_offset - (copy_ts ? (start_at_zero && ic->start_time != AV_NOPTS_VALUE ? ic->start_time : 0) : timestamp);
f->ts_offset = o->input_ts_offset - (copy_ts ? 0 : timestamp);
f->nb_streams = ic->nb_streams;
f->rate_emu = o->rate_emu;
f->accurate_seek = o->accurate_seek;
@@ -1138,11 +1132,8 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
MATCH_PER_STREAM_OPT(bitstream_filters, str, bsf, oc, st);
while (bsf) {
char *arg = NULL;
if (next = strchr(bsf, ','))
*next++ = 0;
if (arg = strchr(bsf, '='))
*arg++ = 0;
if (!(bsfc = av_bitstream_filter_init(bsf))) {
av_log(NULL, AV_LOG_FATAL, "Unknown bitstream filter %s\n", bsf);
exit_program(1);
@@ -1151,7 +1142,6 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
bsfc_prev->next = bsfc;
else
ost->bitstream_filters = bsfc;
av_dict_set(&ost->bsf_args, bsfc->filter->name, arg, 0);
bsfc_prev = bsfc;
bsf = next;
@@ -1289,8 +1279,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
av_log(NULL, AV_LOG_FATAL, "Invalid framerate value: %s\n", frame_rate);
exit_program(1);
}
if (frame_rate && video_sync_method == VSYNC_PASSTHROUGH)
av_log(NULL, AV_LOG_ERROR, "Using -vsync 0 and -r can produce invalid output files\n");
MATCH_PER_STREAM_OPT(frame_aspect_ratios, str, frame_aspect_ratio, oc, st);
if (frame_aspect_ratio) {
@@ -1489,13 +1477,11 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in
}
if (!ist || (ist->file_index == map->file_idx && ist->st->index == map->stream_idx)) {
if (av_reallocp_array(&ost->audio_channels_map,
ost->audio_channels_mapped + 1,
sizeof(*ost->audio_channels_map)
) < 0 )
exit_program(1);
ost->audio_channels_map[ost->audio_channels_mapped++] = map->channel_idx;
if (ost->audio_channels_mapped < FF_ARRAY_ELEMS(ost->audio_channels_map))
ost->audio_channels_map[ost->audio_channels_mapped++] = map->channel_idx;
else
av_log(NULL, AV_LOG_FATAL, "Max channel mapping for output %d.%d reached\n",
ost->file_index, ost->st->index);
}
}
}
@@ -1635,36 +1621,27 @@ static int read_ffserver_streams(OptionsContext *o, AVFormatContext *s, const ch
AVStream *st;
OutputStream *ost;
AVCodec *codec;
const char *enc_config;
AVCodecContext *avctx;
codec = avcodec_find_encoder(ic->streams[i]->codec->codec_id);
if (!codec) {
av_log(s, AV_LOG_ERROR, "no encoder found for codec id %i\n", ic->streams[i]->codec->codec_id);
return AVERROR(EINVAL);
}
if (codec->type == AVMEDIA_TYPE_AUDIO)
opt_audio_codec(o, "c:a", codec->name);
else if (codec->type == AVMEDIA_TYPE_VIDEO)
opt_video_codec(o, "c:v", codec->name);
ost = new_output_stream(o, s, codec->type, -1);
st = ost->st;
avctx = st->codec;
ost->enc = codec;
avcodec_get_context_defaults3(st->codec, codec);
enc_config = av_stream_get_recommended_encoder_configuration(ic->streams[i]);
if (enc_config) {
AVDictionary *opts = NULL;
av_dict_parse_string(&opts, enc_config, "=", ",", 0);
av_opt_set_dict2(st->codec, &opts, AV_OPT_SEARCH_CHILDREN);
av_dict_free(&opts);
}
// FIXME: a more elegant solution is needed
memcpy(st, ic->streams[i], sizeof(AVStream));
st->cur_dts = 0;
st->info = av_malloc(sizeof(*st->info));
memcpy(st->info, ic->streams[i]->info, sizeof(*st->info));
st->codec= avctx;
avcodec_copy_context(st->codec, ic->streams[i]->codec);
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && !ost->stream_copy)
choose_sample_fmt(st, codec);
else if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO && !ost->stream_copy)
choose_pixel_fmt(st, st->codec, codec, st->codec->pix_fmt);
avcodec_copy_context(ost->enc_ctx, st->codec);
if (enc_config)
av_dict_parse_string(&ost->encoder_opts, enc_config, "=", ",", 0);
}
avformat_close_input(&ic);
@@ -1752,8 +1729,8 @@ static int open_output_file(OptionsContext *o, const char *filename)
if (o->stop_time != INT64_MAX && o->recording_time == INT64_MAX) {
int64_t start_time = o->start_time == AV_NOPTS_VALUE ? 0 : o->start_time;
if (o->stop_time <= start_time) {
av_log(NULL, AV_LOG_ERROR, "-to value smaller than -ss; aborting.\n");
exit_program(1);
av_log(NULL, AV_LOG_WARNING, "-to value smaller than -ss; ignoring -to.\n");
o->stop_time = INT64_MAX;
} else {
o->recording_time = o->stop_time - start_time;
}
@@ -1892,27 +1869,8 @@ static int open_output_file(OptionsContext *o, const char *filename)
if (!o->subtitle_disable && (avcodec_find_encoder(oc->oformat->subtitle_codec) || subtitle_codec_name)) {
for (i = 0; i < nb_input_streams; i++)
if (input_streams[i]->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
AVCodecDescriptor const *input_descriptor =
avcodec_descriptor_get(input_streams[i]->st->codec->codec_id);
AVCodecDescriptor const *output_descriptor = NULL;
AVCodec const *output_codec =
avcodec_find_encoder(oc->oformat->subtitle_codec);
int input_props = 0, output_props = 0;
if (output_codec)
output_descriptor = avcodec_descriptor_get(output_codec->id);
if (input_descriptor)
input_props = input_descriptor->props & (AV_CODEC_PROP_TEXT_SUB | AV_CODEC_PROP_BITMAP_SUB);
if (output_descriptor)
output_props = output_descriptor->props & (AV_CODEC_PROP_TEXT_SUB | AV_CODEC_PROP_BITMAP_SUB);
if (subtitle_codec_name ||
input_props & output_props ||
// Map dvb teletext which has neither property to any output subtitle encoder
input_descriptor && output_descriptor &&
(!input_descriptor->props ||
!output_descriptor->props)) {
new_subtitle_stream(o, oc, i);
break;
}
new_subtitle_stream(o, oc, i);
break;
}
}
/* do something with data? */
@@ -2034,13 +1992,8 @@ loop_end:
const AVClass *class = avcodec_get_class();
const AVOption *option = av_opt_find(&class, e->key, NULL, 0,
AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
const AVClass *fclass = avformat_get_class();
const AVOption *foption = av_opt_find(&fclass, e->key, NULL, 0,
AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
if (!option || foption)
if (!option)
continue;
if (!(option->flags & AV_OPT_FLAG_ENCODING_PARAM)) {
av_log(NULL, AV_LOG_ERROR, "Codec AVOption %s (%s) specified for "
"output file #%d (%s) is not an encoding option.\n", e->key,
@@ -2085,7 +2038,9 @@ loop_end:
assert_file_overwrite(filename);
if (o->mux_preload) {
av_dict_set_int(&of->opts, "preload", o->mux_preload*AV_TIME_BASE, 0);
uint8_t buf[64];
snprintf(buf, sizeof(buf), "%d", (int)(o->mux_preload*AV_TIME_BASE));
av_dict_set(&of->opts, "preload", buf, 0);
}
oc->max_delay = (int)(o->mux_max_delay * AV_TIME_BASE);
@@ -2211,8 +2166,7 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
for (j = 0; j < nb_input_files; j++) {
for (i = 0; i < input_files[j]->nb_streams; i++) {
AVCodecContext *c = input_files[j]->ctx->streams[i]->codec;
if (c->codec_type != AVMEDIA_TYPE_VIDEO ||
!c->time_base.num)
if (c->codec_type != AVMEDIA_TYPE_VIDEO)
continue;
fr = c->time_base.den * 1000 / c->time_base.num;
if (fr == 25000) {
@@ -2245,19 +2199,19 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
parse_option(o, "s", norm == PAL ? "352x288" : "352x240", options);
parse_option(o, "r", frame_rates[norm], options);
opt_default(NULL, "g", norm == PAL ? "15" : "18");
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
opt_default(NULL, "b:v", "1150000");
opt_default(NULL, "maxrate:v", "1150000");
opt_default(NULL, "minrate:v", "1150000");
opt_default(NULL, "bufsize:v", "327680"); // 40*1024*8;
av_dict_set(&o->g->codec_opts, "b:v", "1150000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "maxrate", "1150000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "minrate", "1150000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "bufsize", "327680", AV_DICT_DONT_OVERWRITE); // 40*1024*8;
opt_default(NULL, "b:a", "224000");
av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE);
parse_option(o, "ar", "44100", options);
parse_option(o, "ac", "2", options);
opt_default(NULL, "packetsize", "2324");
opt_default(NULL, "muxrate", "1411200"); // 2352 * 75 * 8;
av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->format_opts, "muxrate", "1411200", AV_DICT_DONT_OVERWRITE); // 2352 * 75 * 8;
/* We have to offset the PTS, so that it is consistent with the SCR.
SCR starts at 36000, but the first two packs contain only padding
@@ -2274,18 +2228,18 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
parse_option(o, "s", norm == PAL ? "480x576" : "480x480", options);
parse_option(o, "r", frame_rates[norm], options);
parse_option(o, "pix_fmt", "yuv420p", options);
opt_default(NULL, "g", norm == PAL ? "15" : "18");
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
opt_default(NULL, "b:v", "2040000");
opt_default(NULL, "maxrate:v", "2516000");
opt_default(NULL, "minrate:v", "0"); // 1145000;
opt_default(NULL, "bufsize:v", "1835008"); // 224*1024*8;
opt_default(NULL, "scan_offset", "1");
av_dict_set(&o->g->codec_opts, "b:v", "2040000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "maxrate", "2516000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1145000;
av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8;
av_dict_set(&o->g->codec_opts, "scan_offset", "1", AV_DICT_DONT_OVERWRITE);
opt_default(NULL, "b:a", "224000");
av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE);
parse_option(o, "ar", "44100", options);
opt_default(NULL, "packetsize", "2324");
av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE);
} else if (!strcmp(arg, "dvd")) {
@@ -2296,17 +2250,17 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
parse_option(o, "s", norm == PAL ? "720x576" : "720x480", options);
parse_option(o, "r", frame_rates[norm], options);
parse_option(o, "pix_fmt", "yuv420p", options);
opt_default(NULL, "g", norm == PAL ? "15" : "18");
av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
opt_default(NULL, "b:v", "6000000");
opt_default(NULL, "maxrate:v", "9000000");
opt_default(NULL, "minrate:v", "0"); // 1500000;
opt_default(NULL, "bufsize:v", "1835008"); // 224*1024*8;
av_dict_set(&o->g->codec_opts, "b:v", "6000000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "maxrate", "9000000", AV_DICT_DONT_OVERWRITE);
av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1500000;
av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8;
opt_default(NULL, "packetsize", "2048"); // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
opt_default(NULL, "muxrate", "10080000"); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
av_dict_set(&o->g->format_opts, "packetsize", "2048", AV_DICT_DONT_OVERWRITE); // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
av_dict_set(&o->g->format_opts, "muxrate", "10080000", AV_DICT_DONT_OVERWRITE); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
opt_default(NULL, "b:a", "448000");
av_dict_set(&o->g->codec_opts, "b:a", "448000", AV_DICT_DONT_OVERWRITE);
parse_option(o, "ar", "48000", options);
} else if (!strncmp(arg, "dv", 2)) {
@@ -2325,10 +2279,6 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
av_log(NULL, AV_LOG_ERROR, "Unknown target: %s\n", arg);
return AVERROR(EINVAL);
}
av_dict_copy(&o->g->codec_opts, codec_opts, AV_DICT_DONT_OVERWRITE);
av_dict_copy(&o->g->format_opts, format_opts, AV_DICT_DONT_OVERWRITE);
return 0;
}
@@ -2345,9 +2295,6 @@ static int opt_vstats(void *optctx, const char *opt, const char *arg)
time_t today2 = time(NULL);
struct tm *today = localtime(&today2);
if (!today)
return AVERROR(errno);
snprintf(filename, sizeof(filename), "vstats_%02d%02d%02d.log", today->tm_hour, today->tm_min,
today->tm_sec);
return opt_vstats_file(NULL, opt, filename);
@@ -2828,13 +2775,13 @@ const OptionDef options[] = {
{ "itsscale", HAS_ARG | OPT_DOUBLE | OPT_SPEC |
OPT_EXPERT | OPT_INPUT, { .off = OFFSET(ts_scale) },
"set the input ts scale", "scale" },
{ "timestamp", HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_recording_timestamp },
{ "timestamp", HAS_ARG | OPT_PERFILE, { .func_arg = opt_recording_timestamp },
"set the recording timestamp ('now' to set the current time)", "time" },
{ "metadata", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(metadata) },
"add metadata", "string=string" },
{ "dframes", HAS_ARG | OPT_PERFILE | OPT_EXPERT |
OPT_OUTPUT, { .func_arg = opt_data_frames },
"set the number of data frames to output", "number" },
"set the number of data frames to record", "number" },
{ "benchmark", OPT_BOOL | OPT_EXPERT, { &do_benchmark },
"add timings for benchmarking" },
{ "benchmark_all", OPT_BOOL | OPT_EXPERT, { &do_benchmark_all },
@@ -2863,8 +2810,6 @@ const OptionDef options[] = {
"audio drift threshold", "threshold" },
{ "copyts", OPT_BOOL | OPT_EXPERT, { &copy_ts },
"copy timestamps" },
{ "start_at_zero", OPT_BOOL | OPT_EXPERT, { &start_at_zero },
"shift input timestamps to start at 0 when using copyts" },
{ "copytb", HAS_ARG | OPT_INT | OPT_EXPERT, { &copy_tb },
"copy input stream time base when stream copying", "mode" },
{ "shortest", OPT_BOOL | OPT_EXPERT | OPT_OFFSET |
@@ -2885,7 +2830,7 @@ const OptionDef options[] = {
{ "copypriorss", OPT_INT | HAS_ARG | OPT_EXPERT | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(copy_prior_start) },
"copy or discard frames before start time" },
{ "frames", OPT_INT64 | HAS_ARG | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(max_frames) },
"set the number of frames to output", "number" },
"set the number of frames to record", "number" },
{ "tag", OPT_STRING | HAS_ARG | OPT_SPEC |
OPT_EXPERT | OPT_OUTPUT | OPT_INPUT, { .off = OFFSET(codec_tags) },
"force codec tag/fourcc", "fourcc/tag" },
@@ -2927,7 +2872,7 @@ const OptionDef options[] = {
/* video options */
{ "vframes", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_video_frames },
"set the number of video frames to output", "number" },
"set the number of video frames to record", "number" },
{ "r", OPT_VIDEO | HAS_ARG | OPT_STRING | OPT_SPEC |
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(frame_rates) },
"set frame rate (Hz value, fraction or abbreviation)", "rate" },
@@ -3009,13 +2954,10 @@ const OptionDef options[] = {
{ "hwaccel_device", OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
OPT_SPEC | OPT_INPUT, { .off = OFFSET(hwaccel_devices) },
"select a device for HW acceleration" "devicename" },
#if HAVE_VDPAU_X11
{ "vdpau_api_ver", HAS_ARG | OPT_INT | OPT_EXPERT, { &vdpau_api_ver }, "" },
#endif
/* audio options */
{ "aframes", OPT_AUDIO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_audio_frames },
"set the number of audio frames to output", "number" },
"set the number of audio frames to record", "number" },
{ "aq", OPT_AUDIO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_audio_qscale },
"set audio quality (codec-specific)", "quality", },
{ "ar", OPT_AUDIO | HAS_ARG | OPT_INT | OPT_SPEC |

View File

@@ -77,8 +77,6 @@ static int vda_retrieve_data(AVCodecContext *s, AVFrame *frame)
frame->width, frame->height);
ret = av_frame_copy_props(vda->tmp_frame, frame);
CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
if (ret < 0)
return ret;

View File

@@ -42,11 +42,9 @@ typedef struct VDPAUContext {
VdpGetErrorString *get_error_string;
VdpGetInformationString *get_information_string;
VdpDeviceDestroy *device_destroy;
#if 1 // for ffmpegs older vdpau API, not the oldest though
VdpDecoderCreate *decoder_create;
VdpDecoderDestroy *decoder_destroy;
VdpDecoderRender *decoder_render;
#endif
VdpVideoSurfaceCreate *video_surface_create;
VdpVideoSurfaceDestroy *video_surface_destroy;
VdpVideoSurfaceGetBitsYCbCr *video_surface_get_bits;
@@ -59,8 +57,6 @@ typedef struct VDPAUContext {
VdpYCbCrFormat vdpau_format;
} VDPAUContext;
int vdpau_api_ver = 2;
static void vdpau_uninit(AVCodecContext *s)
{
InputStream *ist = s->opaque;
@@ -243,11 +239,9 @@ do {
GET_CALLBACK(VDP_FUNC_ID_GET_ERROR_STRING, get_error_string);
GET_CALLBACK(VDP_FUNC_ID_GET_INFORMATION_STRING, get_information_string);
GET_CALLBACK(VDP_FUNC_ID_DEVICE_DESTROY, device_destroy);
if (vdpau_api_ver == 1) {
GET_CALLBACK(VDP_FUNC_ID_DECODER_CREATE, decoder_create);
GET_CALLBACK(VDP_FUNC_ID_DECODER_DESTROY, decoder_destroy);
GET_CALLBACK(VDP_FUNC_ID_DECODER_RENDER, decoder_render);
}
GET_CALLBACK(VDP_FUNC_ID_DECODER_CREATE, decoder_create);
GET_CALLBACK(VDP_FUNC_ID_DECODER_DESTROY, decoder_destroy);
GET_CALLBACK(VDP_FUNC_ID_DECODER_RENDER, decoder_render);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_CREATE, video_surface_create);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_DESTROY, video_surface_destroy);
GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_GET_BITS_Y_CB_CR, video_surface_get_bits);
@@ -276,16 +270,12 @@ do {
ctx->vdpau_format = vdpau_formats[i][0];
ctx->pix_fmt = vdpau_formats[i][1];
if (vdpau_api_ver == 1) {
vdpau_ctx = av_vdpau_alloc_context();
if (!vdpau_ctx)
goto fail;
vdpau_ctx->render = ctx->decoder_render;
s->hwaccel_context = vdpau_ctx;
} else
if (av_vdpau_bind_context(s, ctx->device, ctx->get_proc_address, 0))
vdpau_ctx = av_vdpau_alloc_context();
if (!vdpau_ctx)
goto fail;
vdpau_ctx->render = ctx->decoder_render;
s->hwaccel_context = vdpau_ctx;
ctx->get_information_string(&vendor);
av_log(NULL, AV_LOG_VERBOSE, "Using VDPAU -- %s -- on X11 display %s, "
@@ -301,7 +291,7 @@ fail:
return AVERROR(EINVAL);
}
static int vdpau_old_init(AVCodecContext *s)
int vdpau_init(AVCodecContext *s)
{
InputStream *ist = s->opaque;
int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
@@ -343,22 +333,3 @@ static int vdpau_old_init(AVCodecContext *s)
return 0;
}
int vdpau_init(AVCodecContext *s)
{
InputStream *ist = s->opaque;
if (vdpau_api_ver == 1)
return vdpau_old_init(s);
if (!ist->hwaccel_ctx) {
int ret = vdpau_alloc(s);
if (ret < 0)
return ret;
}
ist->hwaccel_get_buffer = vdpau_get_buffer;
ist->hwaccel_retrieve_data = vdpau_retrieve_data;
return 0;
}

1115
ffplay.c

File diff suppressed because it is too large Load Diff

164
ffprobe.c
View File

@@ -33,7 +33,6 @@
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
#include "libavutil/hash.h"
#include "libavutil/opt.h"
#include "libavutil/pixdesc.h"
#include "libavutil/dict.h"
@@ -66,9 +65,6 @@ static int do_show_stream_disposition = 0;
static int do_show_data = 0;
static int do_show_program_version = 0;
static int do_show_library_versions = 0;
static int do_show_pixel_formats = 0;
static int do_show_pixel_format_flags = 0;
static int do_show_pixel_format_components = 0;
static int do_show_chapter_tags = 0;
static int do_show_format_tags = 0;
@@ -84,7 +80,6 @@ static int show_private_data = 1;
static char *print_format;
static char *stream_specifier;
static char *show_data_hash;
typedef struct {
int id; ///< identifier
@@ -128,18 +123,11 @@ typedef enum {
SECTION_ID_FRAME,
SECTION_ID_FRAMES,
SECTION_ID_FRAME_TAGS,
SECTION_ID_FRAME_SIDE_DATA_LIST,
SECTION_ID_FRAME_SIDE_DATA,
SECTION_ID_LIBRARY_VERSION,
SECTION_ID_LIBRARY_VERSIONS,
SECTION_ID_PACKET,
SECTION_ID_PACKETS,
SECTION_ID_PACKETS_AND_FRAMES,
SECTION_ID_PIXEL_FORMAT,
SECTION_ID_PIXEL_FORMAT_FLAGS,
SECTION_ID_PIXEL_FORMAT_COMPONENT,
SECTION_ID_PIXEL_FORMAT_COMPONENTS,
SECTION_ID_PIXEL_FORMATS,
SECTION_ID_PROGRAM_STREAM_DISPOSITION,
SECTION_ID_PROGRAM_STREAM_TAGS,
SECTION_ID_PROGRAM,
@@ -164,20 +152,13 @@ static struct section sections[] = {
[SECTION_ID_FORMAT] = { SECTION_ID_FORMAT, "format", 0, { SECTION_ID_FORMAT_TAGS, -1 } },
[SECTION_ID_FORMAT_TAGS] = { SECTION_ID_FORMAT_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "format_tags" },
[SECTION_ID_FRAMES] = { SECTION_ID_FRAMES, "frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME, SECTION_ID_SUBTITLE, -1 } },
[SECTION_ID_FRAME] = { SECTION_ID_FRAME, "frame", 0, { SECTION_ID_FRAME_TAGS, SECTION_ID_FRAME_SIDE_DATA_LIST, -1 } },
[SECTION_ID_FRAME] = { SECTION_ID_FRAME, "frame", 0, { SECTION_ID_FRAME_TAGS, -1 } },
[SECTION_ID_FRAME_TAGS] = { SECTION_ID_FRAME_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "frame_tags" },
[SECTION_ID_FRAME_SIDE_DATA_LIST] ={ SECTION_ID_FRAME_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME_SIDE_DATA, -1 } },
[SECTION_ID_FRAME_SIDE_DATA] = { SECTION_ID_FRAME_SIDE_DATA, "side_data", 0, { -1 } },
[SECTION_ID_LIBRARY_VERSIONS] = { SECTION_ID_LIBRARY_VERSIONS, "library_versions", SECTION_FLAG_IS_ARRAY, { SECTION_ID_LIBRARY_VERSION, -1 } },
[SECTION_ID_LIBRARY_VERSION] = { SECTION_ID_LIBRARY_VERSION, "library_version", 0, { -1 } },
[SECTION_ID_PACKETS] = { SECTION_ID_PACKETS, "packets", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
[SECTION_ID_PACKETS_AND_FRAMES] = { SECTION_ID_PACKETS_AND_FRAMES, "packets_and_frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
[SECTION_ID_PACKET] = { SECTION_ID_PACKET, "packet", 0, { -1 } },
[SECTION_ID_PIXEL_FORMATS] = { SECTION_ID_PIXEL_FORMATS, "pixel_formats", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PIXEL_FORMAT, -1 } },
[SECTION_ID_PIXEL_FORMAT] = { SECTION_ID_PIXEL_FORMAT, "pixel_format", 0, { SECTION_ID_PIXEL_FORMAT_FLAGS, SECTION_ID_PIXEL_FORMAT_COMPONENTS, -1 } },
[SECTION_ID_PIXEL_FORMAT_FLAGS] = { SECTION_ID_PIXEL_FORMAT_FLAGS, "flags", 0, { -1 }, .unique_name = "pixel_format_flags" },
[SECTION_ID_PIXEL_FORMAT_COMPONENTS] = { SECTION_ID_PIXEL_FORMAT_COMPONENTS, "components", SECTION_FLAG_IS_ARRAY, {SECTION_ID_PIXEL_FORMAT_COMPONENT, -1 }, .unique_name = "pixel_format_components" },
[SECTION_ID_PIXEL_FORMAT_COMPONENT] = { SECTION_ID_PIXEL_FORMAT_COMPONENT, "component", 0, { -1 } },
[SECTION_ID_PROGRAM_STREAM_DISPOSITION] = { SECTION_ID_PROGRAM_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "program_stream_disposition" },
[SECTION_ID_PROGRAM_STREAM_TAGS] = { SECTION_ID_PROGRAM_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "program_stream_tags" },
[SECTION_ID_PROGRAM] = { SECTION_ID_PROGRAM, "program", 0, { SECTION_ID_PROGRAM_TAGS, SECTION_ID_PROGRAM_STREAMS, -1 } },
@@ -188,8 +169,7 @@ static struct section sections[] = {
[SECTION_ID_PROGRAMS] = { SECTION_ID_PROGRAMS, "programs", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PROGRAM, -1 } },
[SECTION_ID_ROOT] = { SECTION_ID_ROOT, "root", SECTION_FLAG_IS_WRAPPER,
{ SECTION_ID_CHAPTERS, SECTION_ID_FORMAT, SECTION_ID_FRAMES, SECTION_ID_PROGRAMS, SECTION_ID_STREAMS,
SECTION_ID_PACKETS, SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS,
SECTION_ID_PIXEL_FORMATS, -1} },
SECTION_ID_PACKETS, SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS, -1} },
[SECTION_ID_STREAMS] = { SECTION_ID_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM, -1 } },
[SECTION_ID_STREAM] = { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
[SECTION_ID_STREAM_DISPOSITION] = { SECTION_ID_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_disposition" },
@@ -203,8 +183,6 @@ static const OptionDef *options;
static const char *input_filename;
static AVInputFormat *iformat = NULL;
static struct AVHashContext *hash;
static const char *const binary_unit_prefixes [] = { "", "Ki", "Mi", "Gi", "Ti", "Pi" };
static const char *const decimal_unit_prefixes[] = { "", "K" , "M" , "G" , "T" , "P" };
@@ -703,21 +681,6 @@ static void writer_print_data(WriterContext *wctx, const char *name,
av_bprint_finalize(&bp, NULL);
}
static void writer_print_data_hash(WriterContext *wctx, const char *name,
uint8_t *data, int size)
{
char *p, buf[AV_HASH_MAX_SIZE * 2 + 64] = { 0 };
if (!hash)
return;
av_hash_init(hash);
av_hash_update(hash, data, size);
snprintf(buf, sizeof(buf), "%s:", av_hash_get_name(hash));
p = buf + strlen(buf);
av_hash_final_hex(hash, p, buf + sizeof(buf) - p);
writer_print_string(wctx, name, buf, 0);
}
#define MAX_REGISTERED_WRITERS_NB 64
static const Writer *registered_writers[MAX_REGISTERED_WRITERS_NB + 1];
@@ -1725,7 +1688,6 @@ static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pk
print_fmt("flags", "%c", pkt->flags & AV_PKT_FLAG_KEY ? 'K' : '_');
if (do_show_data)
writer_print_data(w, "data", pkt->data, pkt->size);
writer_print_data_hash(w, "data_hash", pkt->data, pkt->size);
writer_print_section_footer(w);
av_bprint_finalize(&pbuf, NULL);
@@ -1760,7 +1722,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
{
AVBPrint pbuf;
const char *s;
int i;
av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);
@@ -1823,20 +1784,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
}
if (do_show_frame_tags)
show_tags(w, av_frame_get_metadata(frame), SECTION_ID_FRAME_TAGS);
if (frame->nb_side_data) {
writer_print_section_header(w, SECTION_ID_FRAME_SIDE_DATA_LIST);
for (i = 0; i < frame->nb_side_data; i++) {
AVFrameSideData *sd = frame->side_data[i];
const char *name;
writer_print_section_header(w, SECTION_ID_FRAME_SIDE_DATA);
name = av_frame_side_data_name(sd->type);
print_str("side_data_type", name ? name : "unknown");
print_int("side_data_size", sd->size);
writer_print_section_footer(w);
}
writer_print_section_footer(w);
}
writer_print_section_footer(w);
@@ -2112,28 +2059,12 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
else print_str_opt("pix_fmt", "unknown");
print_int("level", dec_ctx->level);
if (dec_ctx->color_range != AVCOL_RANGE_UNSPECIFIED)
print_str ("color_range", av_color_range_name(dec_ctx->color_range));
print_str ("color_range", dec_ctx->color_range == AVCOL_RANGE_MPEG ? "tv": "pc");
else
print_str_opt("color_range", "N/A");
s = av_get_colorspace_name(dec_ctx->colorspace);
if (s) print_str ("color_space", s);
else print_str_opt("color_space", "unknown");
if (dec_ctx->color_trc != AVCOL_TRC_UNSPECIFIED)
print_str("color_transfer", av_color_transfer_name(dec_ctx->color_trc));
else
print_str_opt("color_transfer", av_color_transfer_name(dec_ctx->color_trc));
if (dec_ctx->color_primaries != AVCOL_PRI_UNSPECIFIED)
print_str("color_primaries", av_color_primaries_name(dec_ctx->color_primaries));
else
print_str_opt("color_primaries", av_color_primaries_name(dec_ctx->color_primaries));
if (dec_ctx->chroma_sample_location != AVCHROMA_LOC_UNSPECIFIED)
print_str("chroma_location", av_chroma_location_name(dec_ctx->chroma_sample_location));
else
print_str_opt("chroma_location", av_chroma_location_name(dec_ctx->chroma_sample_location));
if (dec_ctx->timecode_frame_start >= 0) {
char tcbuf[AV_TIMECODE_STR_SIZE];
av_timecode_make_mpeg_tc_string(tcbuf, dec_ctx->timecode_frame_start);
@@ -2200,8 +2131,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
else print_str_opt("bit_rate", "N/A");
if (dec_ctx->rc_max_rate > 0) print_val ("max_bit_rate", dec_ctx->rc_max_rate, unit_bit_per_second_str);
else print_str_opt("max_bit_rate", "N/A");
if (dec_ctx->bits_per_raw_sample > 0) print_fmt("bits_per_raw_sample", "%d", dec_ctx->bits_per_raw_sample);
else print_str_opt("bits_per_raw_sample", "N/A");
if (stream->nb_frames) print_fmt ("nb_frames", "%"PRId64, stream->nb_frames);
else print_str_opt("nb_frames", "N/A");
if (nb_streams_frames[stream_idx]) print_fmt ("nb_read_frames", "%"PRIu64, nb_streams_frames[stream_idx]);
@@ -2211,8 +2140,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
if (do_show_data)
writer_print_data(w, "extradata", dec_ctx->extradata,
dec_ctx->extradata_size);
writer_print_data_hash(w, "extradata_hash", dec_ctx->extradata,
dec_ctx->extradata_size);
/* Print disposition information */
#define PRINT_DISPOSITION(flagname, name) do { \
@@ -2386,19 +2313,12 @@ static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *t;
AVDictionary **opts;
int scan_all_pmts_set = 0;
if (!av_dict_get(format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE)) {
av_dict_set(&format_opts, "scan_all_pmts", "1", AV_DICT_DONT_OVERWRITE);
scan_all_pmts_set = 1;
}
if ((err = avformat_open_input(&fmt_ctx, filename,
iformat, &format_opts)) < 0) {
print_error(filename, err);
return err;
}
if (scan_all_pmts_set)
av_dict_set(&format_opts, "scan_all_pmts", NULL, AV_DICT_MATCH_CASE);
if ((t = av_dict_get(format_opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
av_log(NULL, AV_LOG_ERROR, "Option %s not found.\n", t->key);
return AVERROR_OPTION_NOT_FOUND;
@@ -2594,58 +2514,6 @@ static void ffprobe_show_library_versions(WriterContext *w)
writer_print_section_footer(w);
}
#define PRINT_PIX_FMT_FLAG(flagname, name) \
do { \
print_int(name, !!(pixdesc->flags & AV_PIX_FMT_FLAG_##flagname)); \
} while (0)
static void ffprobe_show_pixel_formats(WriterContext *w)
{
const AVPixFmtDescriptor *pixdesc = NULL;
int i, n;
writer_print_section_header(w, SECTION_ID_PIXEL_FORMATS);
while (pixdesc = av_pix_fmt_desc_next(pixdesc)) {
writer_print_section_header(w, SECTION_ID_PIXEL_FORMAT);
print_str("name", pixdesc->name);
print_int("nb_components", pixdesc->nb_components);
if ((pixdesc->nb_components >= 3) && !(pixdesc->flags & AV_PIX_FMT_FLAG_RGB)) {
print_int ("log2_chroma_w", pixdesc->log2_chroma_w);
print_int ("log2_chroma_h", pixdesc->log2_chroma_h);
} else {
print_str_opt("log2_chroma_w", "N/A");
print_str_opt("log2_chroma_h", "N/A");
}
n = av_get_bits_per_pixel(pixdesc);
if (n) print_int ("bits_per_pixel", n);
else print_str_opt("bits_per_pixel", "N/A");
if (do_show_pixel_format_flags) {
writer_print_section_header(w, SECTION_ID_PIXEL_FORMAT_FLAGS);
PRINT_PIX_FMT_FLAG(BE, "big_endian");
PRINT_PIX_FMT_FLAG(PAL, "palette");
PRINT_PIX_FMT_FLAG(BITSTREAM, "bitstream");
PRINT_PIX_FMT_FLAG(HWACCEL, "hwaccel");
PRINT_PIX_FMT_FLAG(PLANAR, "planar");
PRINT_PIX_FMT_FLAG(RGB, "rgb");
PRINT_PIX_FMT_FLAG(PSEUDOPAL, "pseudopal");
PRINT_PIX_FMT_FLAG(ALPHA, "alpha");
writer_print_section_footer(w);
}
if (do_show_pixel_format_components && (pixdesc->nb_components > 0)) {
writer_print_section_header(w, SECTION_ID_PIXEL_FORMAT_COMPONENTS);
for (i = 0; i < pixdesc->nb_components; i++) {
writer_print_section_header(w, SECTION_ID_PIXEL_FORMAT_COMPONENT);
print_int("index", i + 1);
print_int("bit_depth", pixdesc->comp[i].depth_minus1 + 1);
writer_print_section_footer(w);
}
writer_print_section_footer(w);
}
writer_print_section_footer(w);
}
writer_print_section_footer(w);
}
static int opt_format(void *optctx, const char *opt, const char *arg)
{
iformat = av_find_input_format(arg);
@@ -2979,7 +2847,6 @@ DEFINE_OPT_SHOW_SECTION(format, FORMAT);
DEFINE_OPT_SHOW_SECTION(frames, FRAMES);
DEFINE_OPT_SHOW_SECTION(library_versions, LIBRARY_VERSIONS);
DEFINE_OPT_SHOW_SECTION(packets, PACKETS);
DEFINE_OPT_SHOW_SECTION(pixel_formats, PIXEL_FORMATS);
DEFINE_OPT_SHOW_SECTION(program_version, PROGRAM_VERSION);
DEFINE_OPT_SHOW_SECTION(streams, STREAMS);
DEFINE_OPT_SHOW_SECTION(programs, PROGRAMS);
@@ -3001,7 +2868,6 @@ static const OptionDef real_options[] = {
{ "select_streams", OPT_STRING | HAS_ARG, {(void*)&stream_specifier}, "select the specified streams", "stream_specifier" },
{ "sections", OPT_EXIT, {.func_arg = opt_sections}, "print sections structure and section information, and exit" },
{ "show_data", OPT_BOOL, {(void*)&do_show_data}, "show packets data" },
{ "show_data_hash", OPT_STRING | HAS_ARG, {(void*)&show_data_hash}, "show packets data hash" },
{ "show_error", 0, {(void*)&opt_show_error}, "show probing error" },
{ "show_format", 0, {(void*)&opt_show_format}, "show format/container info" },
{ "show_frames", 0, {(void*)&opt_show_frames}, "show frames info" },
@@ -3018,7 +2884,6 @@ static const OptionDef real_options[] = {
{ "show_program_version", 0, {(void*)&opt_show_program_version}, "show ffprobe version" },
{ "show_library_versions", 0, {(void*)&opt_show_library_versions}, "show library versions" },
{ "show_versions", 0, {(void*)&opt_show_versions}, "show program and library versions" },
{ "show_pixel_formats", 0, {(void*)&opt_show_pixel_formats}, "show pixel format descriptions" },
{ "show_private_data", OPT_BOOL, {(void*)&show_private_data}, "show private data" },
{ "private", OPT_BOOL, {(void*)&show_private_data}, "same as show_private_data" },
{ "bitexact", OPT_BOOL, {&do_bitexact}, "force bitexact output" },
@@ -3075,9 +2940,6 @@ int main(int argc, char **argv)
SET_DO_SHOW(FRAMES, frames);
SET_DO_SHOW(LIBRARY_VERSIONS, library_versions);
SET_DO_SHOW(PACKETS, packets);
SET_DO_SHOW(PIXEL_FORMATS, pixel_formats);
SET_DO_SHOW(PIXEL_FORMAT_FLAGS, pixel_format_flags);
SET_DO_SHOW(PIXEL_FORMAT_COMPONENTS, pixel_format_components);
SET_DO_SHOW(PROGRAM_VERSION, program_version);
SET_DO_SHOW(PROGRAMS, programs);
SET_DO_SHOW(STREAMS, streams);
@@ -3109,21 +2971,6 @@ int main(int argc, char **argv)
w_name = av_strtok(print_format, "=", &buf);
w_args = buf;
if (show_data_hash) {
if ((ret = av_hash_alloc(&hash, show_data_hash)) < 0) {
if (ret == AVERROR(EINVAL)) {
const char *n;
av_log(NULL, AV_LOG_ERROR,
"Unknown hash algorithm '%s'\nKnown algorithms:",
show_data_hash);
for (i = 0; (n = av_hash_names(i)); i++)
av_log(NULL, AV_LOG_ERROR, " %s", n);
av_log(NULL, AV_LOG_ERROR, "\n");
}
goto end;
}
}
w = writer_get_by_name(w_name);
if (!w) {
av_log(NULL, AV_LOG_ERROR, "Unknown output format with name '%s'\n", w_name);
@@ -3142,12 +2989,10 @@ int main(int argc, char **argv)
ffprobe_show_program_version(wctx);
if (do_show_library_versions)
ffprobe_show_library_versions(wctx);
if (do_show_pixel_formats)
ffprobe_show_pixel_formats(wctx);
if (!input_filename &&
((do_show_format || do_show_programs || do_show_streams || do_show_chapters || do_show_packets || do_show_error) ||
(!do_show_program_version && !do_show_library_versions && !do_show_pixel_formats))) {
(!do_show_program_version && !do_show_library_versions))) {
show_usage();
av_log(NULL, AV_LOG_ERROR, "You have to specify one input file.\n");
av_log(NULL, AV_LOG_ERROR, "Use -h to get full help or, even better, run 'man %s'.\n", program_name);
@@ -3165,7 +3010,6 @@ int main(int argc, char **argv)
end:
av_freep(&print_format);
av_freep(&read_intervals);
av_hash_freep(&hash);
uninit_opts();
for (i = 0; i < FF_ARRAY_ELEMS(sections); i++)

1295
ffserver.c

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

View File

@@ -1,133 +0,0 @@
/*
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* 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 FFSERVER_CONFIG_H
#define FFSERVER_CONFIG_H
#include "libavutil/dict.h"
#include "libavformat/avformat.h"
#include "libavformat/network.h"
#define FFSERVER_MAX_STREAMS 20
/* each generated stream is described here */
enum FFServerStreamType {
STREAM_TYPE_LIVE,
STREAM_TYPE_STATUS,
STREAM_TYPE_REDIRECT,
};
enum FFServerIPAddressAction {
IP_ALLOW = 1,
IP_DENY,
};
typedef struct FFServerIPAddressACL {
struct FFServerIPAddressACL *next;
enum FFServerIPAddressAction action;
/* These are in host order */
struct in_addr first;
struct in_addr last;
} FFServerIPAddressACL;
/* description of each stream of the ffserver.conf file */
typedef struct FFServerStream {
enum FFServerStreamType stream_type;
char filename[1024]; /* stream filename */
struct FFServerStream *feed; /* feed we are using (can be null if coming from file) */
AVDictionary *in_opts; /* input parameters */
AVDictionary *metadata; /* metadata to set on the stream */
AVInputFormat *ifmt; /* if non NULL, force input format */
AVOutputFormat *fmt;
FFServerIPAddressACL *acl;
char dynamic_acl[1024];
int nb_streams;
int prebuffer; /* Number of milliseconds early to start */
int64_t max_time; /* Number of milliseconds to run */
int send_on_key;
AVStream *streams[FFSERVER_MAX_STREAMS];
int feed_streams[FFSERVER_MAX_STREAMS]; /* index of streams in the feed */
char feed_filename[1024]; /* file name of the feed storage, or
input file name for a stream */
pid_t pid; /* Of ffmpeg process */
time_t pid_start; /* Of ffmpeg process */
char **child_argv;
struct FFServerStream *next;
unsigned bandwidth; /* bandwidth, in kbits/s */
/* RTSP options */
char *rtsp_option;
/* multicast specific */
int is_multicast;
struct in_addr multicast_ip;
int multicast_port; /* first port used for multicast */
int multicast_ttl;
int loop; /* if true, send the stream in loops (only meaningful if file) */
/* feed specific */
int feed_opened; /* true if someone is writing to the feed */
int is_feed; /* true if it is a feed */
int readonly; /* True if writing is prohibited to the file */
int truncate; /* True if feeder connection truncate the feed file */
int conns_served;
int64_t bytes_served;
int64_t feed_max_size; /* maximum storage size, zero means unlimited */
int64_t feed_write_index; /* current write position in feed (it wraps around) */
int64_t feed_size; /* current size of feed */
struct FFServerStream *next_feed;
} FFServerStream;
typedef struct FFServerConfig {
char *filename;
FFServerStream *first_feed; /* contains only feeds */
FFServerStream *first_stream; /* contains all streams, including feeds */
unsigned int nb_max_http_connections;
unsigned int nb_max_connections;
uint64_t max_bandwidth;
int debug;
char logfilename[1024];
struct sockaddr_in http_addr;
struct sockaddr_in rtsp_addr;
int errors;
int warnings;
int use_defaults;
// Following variables MUST NOT be used outside configuration parsing code.
enum AVCodecID guessed_audio_codec_id;
enum AVCodecID guessed_video_codec_id;
AVDictionary *video_opts; /* AVOptions for video encoder */
AVDictionary *audio_opts; /* AVOptions for audio encoder */
AVCodecContext *dummy_actx; /* Used internally to test audio AVOptions. */
AVCodecContext *dummy_vctx; /* Used internally to test video AVOptions. */
int no_audio;
int no_video;
int line_num;
int stream_use_defaults;
} FFServerConfig;
void ffserver_get_arg(char *buf, int buf_size, const char **pp);
void ffserver_parse_acl_row(FFServerStream *stream, FFServerStream* feed,
FFServerIPAddressACL *ext_acl,
const char *p, const char *filename, int line_num);
int ffserver_parse_ffconfig(const char *filename, FFServerConfig *config);
void ffserver_free_child_args(void *argsp);
#endif /* FFSERVER_CONFIG_H */

View File

@@ -38,15 +38,15 @@ static av_cold int zero12v_decode_init(AVCodecContext *avctx)
static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame, AVPacket *avpkt)
{
int line, ret;
int line = 0, ret;
const int width = avctx->width;
AVFrame *pic = data;
uint16_t *y, *u, *v;
const uint8_t *line_end, *src = avpkt->data;
int stride = avctx->width * 8 / 3;
if (width <= 1 || avctx->height <= 0) {
av_log(avctx, AV_LOG_ERROR, "Dimensions %dx%d not supported.\n", width, avctx->height);
if (width == 1) {
av_log(avctx, AV_LOG_ERROR, "Width 1 not supported.\n");
return AVERROR_INVALIDDATA;
}
@@ -67,45 +67,45 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
pic->pict_type = AV_PICTURE_TYPE_I;
pic->key_frame = 1;
y = (uint16_t *)pic->data[0];
u = (uint16_t *)pic->data[1];
v = (uint16_t *)pic->data[2];
line_end = avpkt->data + stride;
for (line = 0; line < avctx->height; line++) {
uint16_t y_temp[6] = {0x8000, 0x8000, 0x8000, 0x8000, 0x8000, 0x8000};
uint16_t u_temp[3] = {0x8000, 0x8000, 0x8000};
uint16_t v_temp[3] = {0x8000, 0x8000, 0x8000};
int x;
y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
for (x = 0; x < width; x += 6) {
uint32_t t;
if (width - x < 6 || line_end - src < 16) {
y = y_temp;
u = u_temp;
v = v_temp;
}
if (line_end - src < 4)
break;
t = AV_RL32(src);
while (line++ < avctx->height) {
while (1) {
uint32_t t = AV_RL32(src);
src += 4;
*u++ = t << 6 & 0xFFC0;
*y++ = t >> 4 & 0xFFC0;
*v++ = t >> 14 & 0xFFC0;
if (line_end - src < 4)
if (src >= line_end - 1) {
*y = 0x80;
src++;
line_end += stride;
y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
break;
}
t = AV_RL32(src);
src += 4;
*y++ = t << 6 & 0xFFC0;
*u++ = t >> 4 & 0xFFC0;
*y++ = t >> 14 & 0xFFC0;
if (line_end - src < 4)
if (src >= line_end - 2) {
if (!(width & 1)) {
*y = 0x80;
src += 2;
}
line_end += stride;
y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
break;
}
t = AV_RL32(src);
src += 4;
@@ -113,8 +113,15 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
*y++ = t >> 4 & 0xFFC0;
*u++ = t >> 14 & 0xFFC0;
if (line_end - src < 4)
if (src >= line_end - 1) {
*y = 0x80;
src++;
line_end += stride;
y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
break;
}
t = AV_RL32(src);
src += 4;
@@ -122,21 +129,18 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
*v++ = t >> 4 & 0xFFC0;
*y++ = t >> 14 & 0xFFC0;
if (width - x < 6)
if (src >= line_end - 2) {
if (width & 1) {
*y = 0x80;
src += 2;
}
line_end += stride;
y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
break;
}
}
if (x < width) {
y = x + (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
u = x/2 + (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
v = x/2 + (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
memcpy(y, y_temp, sizeof(*y) * (width - x));
memcpy(u, u_temp, sizeof(*u) * (width - x + 1) / 2);
memcpy(v, v_temp, sizeof(*v) * (width - x + 1) / 2);
}
line_end += stride;
src = line_end - stride;
}
*got_frame = 1;

View File

@@ -339,25 +339,21 @@ static inline void mcdc(uint16_t *dst, const uint16_t *src, int log2w,
}
}
static int decode_p_block(FourXContext *f, uint16_t *dst, const uint16_t *src,
static int decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
int log2w, int log2h, int stride)
{
int index, h, code, ret, scale = 1;
uint16_t *start, *end;
const int index = size2index[log2h][log2w];
const int h = 1 << log2h;
int code = get_vlc2(&f->gb,
block_type_vlc[1 - (f->version > 1)][index].table,
BLOCK_TYPE_VLC_BITS, 1);
uint16_t *start = f->last_frame_buffer;
uint16_t *end = start + stride * (f->avctx->height - h + 1) - (1 << log2w);
int ret;
int scale = 1;
unsigned dc = 0;
av_assert0(log2w >= 0 && log2h >= 0);
index = size2index[log2h][log2w];
av_assert0(index >= 0);
h = 1 << log2h;
code = get_vlc2(&f->gb, block_type_vlc[1 - (f->version > 1)][index].table,
BLOCK_TYPE_VLC_BITS, 1);
av_assert0(code >= 0 && code <= 6);
start = f->last_frame_buffer;
end = start + stride * (f->avctx->height - h + 1) - (1 << log2w);
av_assert0(code >= 0 && code <= 6 && log2w >= 0);
if (code == 1) {
log2h--;

View File

@@ -101,7 +101,7 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
}
if (avpkt->size < (hdr_size + 1) * avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
return AVERROR_INVALIDDATA;
return AVERROR(EINVAL);
}
esc->fib_acc[0] = avpkt->data[1] + 128;
@@ -124,7 +124,7 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
}
if (!esc->data[0]) {
av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
return AVERROR_INVALIDDATA;
return AVERROR(EINVAL);
}
/* decode next piece of data from the buffer */

View File

@@ -11,7 +11,6 @@ HEADERS = avcodec.h \
vda.h \
vdpau.h \
version.h \
vorbis_parser.h \
xvmc.h \
OBJS = allcodecs.o \
@@ -32,10 +31,8 @@ OBJS = allcodecs.o \
resample.o \
resample2.o \
utils.o \
vorbis_parser.o \
xiph.o \
# subsystems
# parts needed for many different codecs
OBJS-$(CONFIG_AANDCTTABLES) += aandcttab.o
OBJS-$(CONFIG_AC3DSP) += ac3dsp.o
OBJS-$(CONFIG_AUDIO_FRAME_QUEUE) += audio_frame_queue.o
@@ -45,11 +42,12 @@ OBJS-$(CONFIG_BSWAPDSP) += bswapdsp.o
OBJS-$(CONFIG_CABAC) += cabac.o
OBJS-$(CONFIG_CRYSTALHD) += crystalhd.o
OBJS-$(CONFIG_DCT) += dct.o dct32_fixed.o dct32_float.o
OBJS-$(CONFIG_DSPUTIL) += dsputil.o
OBJS-$(CONFIG_DXVA2) += dxva2.o
OBJS-$(CONFIG_ERROR_RESILIENCE) += error_resilience.o
OBJS-$(CONFIG_EXIF) += exif.o tiff_common.o
OBJS-$(CONFIG_FAANDCT) += faandct.o
OBJS-$(CONFIG_FAANIDCT) += faanidct.o
OBJS-$(CONFIG_FDCTDSP) += fdctdsp.o jfdctfst.o jfdctint.o
OBJS-$(CONFIG_FDCTDSP) += fdctdsp.o faandct.o \
jfdctfst.o jfdctint.o
FFT-OBJS-$(CONFIG_HARDCODED_TABLES) += cos_tables.o cos_fixed_tables.o
OBJS-$(CONFIG_FFT) += avfft.o fft_fixed.o fft_float.o \
fft_fixed_32.o fft_init_table.o \
@@ -57,15 +55,15 @@ OBJS-$(CONFIG_FFT) += avfft.o fft_fixed.o fft_float.o \
OBJS-$(CONFIG_GOLOMB) += golomb.o
OBJS-$(CONFIG_H263DSP) += h263dsp.o
OBJS-$(CONFIG_H264CHROMA) += h264chroma.o
OBJS-$(CONFIG_H264DSP) += h264dsp.o h264idct.o
OBJS-$(CONFIG_H264DSP) += h264dsp.o h264idct.o startcode.o
OBJS-$(CONFIG_H264PRED) += h264pred.o
OBJS-$(CONFIG_H264QPEL) += h264qpel.o
OBJS-$(CONFIG_HPELDSP) += hpeldsp.o
OBJS-$(CONFIG_HUFFMAN) += huffman.o
OBJS-$(CONFIG_HUFFYUVDSP) += huffyuvdsp.o
OBJS-$(CONFIG_HUFFYUVENCDSP) += huffyuvencdsp.o
OBJS-$(CONFIG_IDCTDSP) += idctdsp.o simple_idct.o jrevdct.o
OBJS-$(CONFIG_IIRFILTER) += iirfilter.o
OBJS-$(CONFIG_IDCTDSP) += idctdsp.o faanidct.o \
simple_idct.o jrevdct.o
OBJS-$(CONFIG_INTRAX8) += intrax8.o intrax8dsp.o
OBJS-$(CONFIG_LIBXVID) += libxvid_rc.o
OBJS-$(CONFIG_LLAUDDSP) += lossless_audiodsp.o
@@ -73,7 +71,6 @@ OBJS-$(CONFIG_LLVIDDSP) += lossless_videodsp.o
OBJS-$(CONFIG_LPC) += lpc.o
OBJS-$(CONFIG_LSP) += lsp.o
OBJS-$(CONFIG_MDCT) += mdct_fixed.o mdct_float.o mdct_fixed_32.o
OBJS-$(CONFIG_ME_CMP) += me_cmp.o
OBJS-$(CONFIG_MPEG_ER) += mpeg_er.o
OBJS-$(CONFIG_MPEGAUDIO) += mpegaudio.o mpegaudiodata.o \
mpegaudiodecheader.o
@@ -93,11 +90,12 @@ RDFT-OBJS-$(CONFIG_HARDCODED_TABLES) += sin_tables.o
OBJS-$(CONFIG_RDFT) += rdft.o $(RDFT-OBJS-yes)
OBJS-$(CONFIG_SHARED) += log2_tab.o
OBJS-$(CONFIG_SINEWIN) += sinewin.o
OBJS-$(CONFIG_STARTCODE) += startcode.o
OBJS-$(CONFIG_TPELDSP) += tpeldsp.o
OBJS-$(CONFIG_VAAPI) += vaapi.o
OBJS-$(CONFIG_VDA) += vda.o
OBJS-$(CONFIG_VDPAU) += vdpau.o
OBJS-$(CONFIG_VIDEODSP) += videodsp.o
OBJS-$(CONFIG_VP3DSP) += vp3dsp.o
OBJS-$(CONFIG_WMA_FREQS) += wma_freqs.o
# decoders/encoders
OBJS-$(CONFIG_ZERO12V_DECODER) += 012v.o
@@ -108,7 +106,8 @@ OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
sbrdsp.o aacpsdsp.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o mpeg4audio.o kbdwin.o
psymodel.o iirfilter.o \
mpeg4audio.o kbdwin.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_DECODER) += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o
@@ -136,7 +135,6 @@ OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpeg.o mjpegenc_common.o \
OBJS-$(CONFIG_ANM_DECODER) += anm.o
OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_APNG_DECODER) += png.o pngdec.o pngdsp.o
OBJS-$(CONFIG_SSA_DECODER) += assdec.o ass.o ass_split.o
OBJS-$(CONFIG_SSA_ENCODER) += assenc.o ass.o
OBJS-$(CONFIG_ASS_DECODER) += assdec.o ass.o ass_split.o
@@ -162,8 +160,8 @@ OBJS-$(CONFIG_AYUV_ENCODER) += v408enc.o
OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o
OBJS-$(CONFIG_BFI_DECODER) += bfi.o
OBJS-$(CONFIG_BINK_DECODER) += bink.o binkdsp.o
OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER) += binkaudio.o
OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER) += binkaudio.o
OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER) += binkaudio.o wma.o wma_common.o
OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER) += binkaudio.o wma.o wma_common.o
OBJS-$(CONFIG_BINTEXT_DECODER) += bintext.o cga_data.o
OBJS-$(CONFIG_BMP_DECODER) += bmp.o msrledec.o
OBJS-$(CONFIG_BMP_ENCODER) += bmpenc.o
@@ -210,7 +208,7 @@ OBJS-$(CONFIG_DVVIDEO_DECODER) += dvdec.o dv.o dvdata.o
OBJS-$(CONFIG_DVVIDEO_ENCODER) += dvenc.o dv.o dvdata.o
OBJS-$(CONFIG_DXA_DECODER) += dxa.o
OBJS-$(CONFIG_DXTORY_DECODER) += dxtory.o
OBJS-$(CONFIG_EAC3_DECODER) += eac3_data.o
OBJS-$(CONFIG_EAC3_DECODER) += eac3dec.o eac3_data.o
OBJS-$(CONFIG_EAC3_ENCODER) += eac3enc.o eac3_data.o
OBJS-$(CONFIG_EACMV_DECODER) += eacmv.o
OBJS-$(CONFIG_EAMAD_DECODER) += eamad.o eaidct.o mpeg12.o \
@@ -280,7 +278,7 @@ OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
OBJS-$(CONFIG_JACOSUB_DECODER) += jacosubdec.o ass.o
OBJS-$(CONFIG_JPEG2000_ENCODER) += j2kenc.o mqcenc.o mqc.o jpeg2000.o \
jpeg2000dwt.o
OBJS-$(CONFIG_JPEG2000_DECODER) += jpeg2000dec.o jpeg2000.o jpeg2000dsp.o \
OBJS-$(CONFIG_JPEG2000_DECODER) += jpeg2000dec.o jpeg2000.o \
jpeg2000dwt.o mqcdec.o mqc.o
OBJS-$(CONFIG_JPEGLS_DECODER) += jpeglsdec.o jpegls.o
OBJS-$(CONFIG_JPEGLS_ENCODER) += jpeglsenc.o jpegls.o
@@ -321,12 +319,12 @@ OBJS-$(CONFIG_MP3ON4_DECODER) += mpegaudiodec_fixed.o mpeg4audio.o
OBJS-$(CONFIG_MP3ON4FLOAT_DECODER) += mpegaudiodec_float.o mpeg4audio.o
OBJS-$(CONFIG_MPC7_DECODER) += mpc7.o mpc.o
OBJS-$(CONFIG_MPC8_DECODER) += mpc8.o mpc.o
OBJS-$(CONFIG_MPEGVIDEO_DECODER) += mpeg12dec.o mpeg12.o mpeg12data.o
OBJS-$(CONFIG_MPEGVIDEO_DECODER) += mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG1VIDEO_DECODER) += mpeg12dec.o mpeg12.o mpeg12data.o
OBJS-$(CONFIG_MPEG1VIDEO_ENCODER) += mpeg12enc.o mpeg12.o
OBJS-$(CONFIG_MPEG2VIDEO_DECODER) += mpeg12dec.o mpeg12.o mpeg12data.o
OBJS-$(CONFIG_MPEG2VIDEO_ENCODER) += mpeg12enc.o mpeg12.o
OBJS-$(CONFIG_MPEG4_DECODER) += xvididct.o
OBJS-$(CONFIG_MPL2_DECODER) += mpl2dec.o ass.o
OBJS-$(CONFIG_MSMPEG4V1_DECODER) += msmpeg4dec.o msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_MSMPEG4V2_DECODER) += msmpeg4dec.o msmpeg4.o msmpeg4data.o
@@ -433,7 +431,6 @@ OBJS-$(CONFIG_SONIC_LS_ENCODER) += sonic.o
OBJS-$(CONFIG_SP5X_DECODER) += sp5xdec.o
OBJS-$(CONFIG_SRT_DECODER) += srtdec.o ass.o
OBJS-$(CONFIG_SRT_ENCODER) += srtenc.o ass_split.o
OBJS-$(CONFIG_STL_DECODER) += textdec.o ass.o
OBJS-$(CONFIG_SUBRIP_DECODER) += srtdec.o ass.o
OBJS-$(CONFIG_SUBRIP_ENCODER) += srtenc.o ass_split.o
OBJS-$(CONFIG_SUBVIEWER1_DECODER) += textdec.o ass.o
@@ -449,6 +446,7 @@ OBJS-$(CONFIG_TAK_DECODER) += takdec.o tak.o
OBJS-$(CONFIG_TARGA_DECODER) += targa.o
OBJS-$(CONFIG_TARGA_ENCODER) += targaenc.o rle.o
OBJS-$(CONFIG_TARGA_Y216_DECODER) += targa_y216dec.o
OBJS-$(CONFIG_THEORA_DECODER) += xiph.o
OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
OBJS-$(CONFIG_TIFF_DECODER) += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
OBJS-$(CONFIG_TIFF_ENCODER) += tiffenc.o rle.o lzwenc.o tiff_data.o
@@ -477,17 +475,15 @@ OBJS-$(CONFIG_V410_ENCODER) += v410enc.o
OBJS-$(CONFIG_V210X_DECODER) += v210x.o
OBJS-$(CONFIG_VB_DECODER) += vb.o
OBJS-$(CONFIG_VBLE_DECODER) += vble.o
OBJS-$(CONFIG_VC1_DECODER) += vc1dec.o vc1_block.o vc1_loopfilter.o \
vc1_mc.o vc1_pred.o vc1.o vc1data.o \
vc1dsp.o \
OBJS-$(CONFIG_VC1_DECODER) += vc1dec.o vc1.o vc1data.o vc1dsp.o \
msmpeg4dec.o msmpeg4.o msmpeg4data.o \
wmv2dsp.o
wmv2dsp.o startcode.o
OBJS-$(CONFIG_VCR1_DECODER) += vcr1.o
OBJS-$(CONFIG_VMDAUDIO_DECODER) += vmdaudio.o
OBJS-$(CONFIG_VMDVIDEO_DECODER) += vmdvideo.o
OBJS-$(CONFIG_VMNC_DECODER) += vmnc.o
OBJS-$(CONFIG_VORBIS_DECODER) += vorbisdec.o vorbisdsp.o vorbis.o \
vorbis_data.o
vorbis_data.o xiph.o
OBJS-$(CONFIG_VORBIS_ENCODER) += vorbisenc.o vorbis.o \
vorbis_data.o
OBJS-$(CONFIG_VP3_DECODER) += vp3.o
@@ -504,7 +500,7 @@ OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o
OBJS-$(CONFIG_WAVPACK_ENCODER) += wavpackenc.o
OBJS-$(CONFIG_WEBP_DECODER) += vp8.o vp8dsp.o vp56rac.o
OBJS-$(CONFIG_WEBP_DECODER) += webp.o exif.o tiff_common.o
OBJS-$(CONFIG_WEBVTT_DECODER) += webvttdec.o ass.o
OBJS-$(CONFIG_WEBVTT_DECODER) += webvttdec.o
OBJS-$(CONFIG_WEBVTT_ENCODER) += webvttenc.o ass_split.o
OBJS-$(CONFIG_WMALOSSLESS_DECODER) += wmalosslessdec.o wma_common.o
OBJS-$(CONFIG_WMAPRO_DECODER) += wmaprodec.o wma.o wma_common.o
@@ -652,11 +648,6 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_VIMA_DECODER) += vima.o adpcm_data.o
# hardware accelerators
OBJS-$(CONFIG_DXVA2) += dxva2.o
OBJS-$(CONFIG_VAAPI) += vaapi.o
OBJS-$(CONFIG_VDA) += vda.o
OBJS-$(CONFIG_VDPAU) += vdpau.o
OBJS-$(CONFIG_H263_VAAPI_HWACCEL) += vaapi_mpeg4.o vaapi_mpeg.o
OBJS-$(CONFIG_H263_VDPAU_HWACCEL) += vdpau_mpeg4.o
OBJS-$(CONFIG_H264_DXVA2_HWACCEL) += dxva2_h264.o
@@ -679,18 +670,20 @@ OBJS-$(CONFIG_VC1_VDPAU_HWACCEL) += vdpau_vc1.o
OBJS-$(CONFIG_ADTS_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_CAF_DEMUXER) += mpeg4audio.o mpegaudiodata.o \
ac3tab.o
OBJS-$(CONFIG_FLAC_DEMUXER) += flac.o flacdata.o vorbis_data.o
OBJS-$(CONFIG_FLAC_DEMUXER) += flac.o flacdata.o vorbis_data.o \
vorbis_parser.o xiph.o
OBJS-$(CONFIG_FLAC_MUXER) += flac.o flacdata.o vorbis_data.o
OBJS-$(CONFIG_FLV_DEMUXER) += mpeg4audio.o
OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
OBJS-$(CONFIG_IFF_DEMUXER) += iff.o
OBJS-$(CONFIG_ISMV_MUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_LATM_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += mpeg4audio.o vorbis_data.o \
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o vorbis_data.o \
flac.o flacdata.o
OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o mpegaudiodata.o \
vorbis_parser.o xiph.o
OBJS-$(CONFIG_MATROSKA_MUXER) += mpeg4audio.o mpegaudiodata.o \
flac.o flacdata.o vorbis_data.o
flac.o flacdata.o vorbis_data.o xiph.o
OBJS-$(CONFIG_MP2_MUXER) += mpegaudiodata.o mpegaudiodecheader.o
OBJS-$(CONFIG_MP3_MUXER) += mpegaudiodata.o mpegaudiodecheader.o
OBJS-$(CONFIG_MOV_DEMUXER) += mpeg4audio.o mpegaudiodata.o ac3tab.o
@@ -699,18 +692,19 @@ OBJS-$(CONFIG_MPEGTS_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_MPEGTS_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MXF_MUXER) += dnxhddata.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
OBJS-$(CONFIG_OGA_MUXER) += flac.o flacdata.o
OBJS-$(CONFIG_OGG_DEMUXER) += mpeg12data.o \
OBJS-$(CONFIG_OGA_MUXER) += xiph.o flac.o flacdata.o
OBJS-$(CONFIG_OGG_DEMUXER) += xiph.o flac.o flacdata.o \
mpeg12data.o vorbis_parser.o \
dirac.o vorbis_data.o
OBJS-$(CONFIG_OGG_MUXER) += flac.o flacdata.o \
OBJS-$(CONFIG_OGG_MUXER) += xiph.o flac.o flacdata.o \
vorbis_data.o
OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o xiph.o
OBJS-$(CONFIG_RTPDEC) += mjpeg.o
OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_SPDIF_MUXER) += dca.o
OBJS-$(CONFIG_TAK_DEMUXER) += tak.o
OBJS-$(CONFIG_WEBM_MUXER) += mpeg4audio.o mpegaudiodata.o \
flac.o flacdata.o \
xiph.o flac.o flacdata.o \
vorbis_data.o
OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o
@@ -723,10 +717,10 @@ OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
OBJS-$(CONFIG_LIBFDK_AAC_DECODER) += libfdk-aacdec.o
OBJS-$(CONFIG_LIBFDK_AAC_ENCODER) += libfdk-aacenc.o
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsmdec.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsmenc.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsmdec.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsmenc.o
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBILBC_DECODER) += libilbc.o
OBJS-$(CONFIG_LIBILBC_ENCODER) += libilbc.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o mpegaudiodecheader.o
@@ -755,7 +749,7 @@ OBJS-$(CONFIG_LIBVO_AACENC_ENCODER) += libvo-aacenc.o mpeg4audio.o
OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER) += libvo-amrwbenc.o
OBJS-$(CONFIG_LIBVORBIS_DECODER) += libvorbisdec.o
OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbisenc.o \
vorbis_data.o
vorbis_data.o vorbis_parser.o xiph.o
OBJS-$(CONFIG_LIBVPX_VP8_DECODER) += libvpxdec.o
OBJS-$(CONFIG_LIBVPX_VP8_ENCODER) += libvpxenc.o
OBJS-$(CONFIG_LIBVPX_VP9_DECODER) += libvpxdec.o libvpx.o
@@ -810,7 +804,8 @@ OBJS-$(CONFIG_RV40_PARSER) += rv34_parser.o
OBJS-$(CONFIG_TAK_PARSER) += tak_parser.o tak.o
OBJS-$(CONFIG_VC1_PARSER) += vc1_parser.o vc1.o vc1data.o vc1dsp.o \
msmpeg4.o msmpeg4data.o mpeg4video.o \
h263.o
h263.o startcode.o
OBJS-$(CONFIG_VORBIS_PARSER) += vorbis_parser.o xiph.o
OBJS-$(CONFIG_VP3_PARSER) += vp3_parser.o
OBJS-$(CONFIG_VP8_PARSER) += vp8_parser.o
OBJS-$(CONFIG_VP9_PARSER) += vp9_parser.o
@@ -857,25 +852,24 @@ SKIPHEADERS-$(CONFIG_VAAPI) += vaapi_internal.h
SKIPHEADERS-$(CONFIG_VDA) += vda.h vda_internal.h
SKIPHEADERS-$(CONFIG_VDPAU) += vdpau.h vdpau_internal.h
TESTPROGS = imgconvert \
TESTPROGS-$(CONFIG_CABAC) += cabac
TESTPROGS-$(CONFIG_FFT) += fft fft-fixed fft-fixed32
TESTPROGS-$(CONFIG_IDCTDSP) += dct
TESTPROGS-$(CONFIG_IIRFILTER) += iirfilter
TESTPROGS-$(HAVE_MMX) += motion
TESTPROGS-$(CONFIG_GOLOMB) += golomb
TESTPROGS-$(CONFIG_RANGECODER) += rangecoder
TESTPROGS-$(CONFIG_SNOW_ENCODER) += snowenc
TESTPROGS = cabac \
fft \
fft-fixed \
fft-fixed32 \
golomb \
iirfilter \
imgconvert \
rangecoder \
snowenc \
TESTPROGS-$(CONFIG_DCT) += dct
TESTPROGS-$(HAVE_MMX) += motion
TESTOBJS = dctref.o
TOOLS = fourcc2pixfmt
HOSTPROGS = aac_tablegen \
aacps_tablegen \
aacsbr_tablegen \
cabac_tablegen \
cbrt_tablegen \
cos_tablegen \
dsd_tablegen \
@@ -903,7 +897,7 @@ else
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
endif
GEN_HEADERS = cabac_tables.h cbrt_tables.h aacps_tables.h aacsbr_tables.h aac_tables.h dsd_tables.h dv_tables.h \
GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dsd_tables.h dv_tables.h \
sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
pcm_tables.h qdm2_tables.h
GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))
@@ -914,9 +908,7 @@ $(GEN_HEADERS): $(SUBDIR)%_tables.h: $(SUBDIR)%_tablegen$(HOSTEXESUF)
ifdef CONFIG_HARDCODED_TABLES
$(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
$(SUBDIR)aacsbr.o: $(SUBDIR)aacsbr_tables.h
$(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
$(SUBDIR)cabac.o: $(SUBDIR)cabac_tables.h
$(SUBDIR)dsddec.o: $(SUBDIR)dsd_tables.h
$(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
$(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h

View File

@@ -28,7 +28,6 @@
#include "a64tables.h"
#include "elbg.h"
#include "internal.h"
#include "libavutil/avassert.h"
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
@@ -66,7 +65,7 @@ static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
//static const int mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
//static const int mc_colors[5]={0x0,0x9,0x8,0xa,0x3};
static void to_meta_with_crop(AVCodecContext *avctx, const AVFrame *p, int *dest)
static void to_meta_with_crop(AVCodecContext *avctx, AVFrame *p, int *dest)
{
int blockx, blocky, x, y;
int luma = 0;
@@ -79,13 +78,9 @@ static void to_meta_with_crop(AVCodecContext *avctx, const AVFrame *p, int *dest
for (y = blocky; y < blocky + 8 && y < C64YRES; y++) {
for (x = blockx; x < blockx + 8 && x < C64XRES; x += 2) {
if(x < width && y < height) {
if (x + 1 < width) {
/* build average over 2 pixels */
luma = (src[(x + 0 + y * p->linesize[0])] +
src[(x + 1 + y * p->linesize[0])]) / 2;
} else {
luma = src[(x + y * p->linesize[0])];
}
/* build average over 2 pixels */
luma = (src[(x + 0 + y * p->linesize[0])] +
src[(x + 1 + y * p->linesize[0])]) / 2;
/* write blocks as linear data now so they are suitable for elbg */
dest[0] = luma;
}
@@ -191,11 +186,12 @@ static void render_charset(AVCodecContext *avctx, uint8_t *charset,
static av_cold int a64multi_close_encoder(AVCodecContext *avctx)
{
A64Context *c = avctx->priv_data;
av_freep(&c->mc_meta_charset);
av_freep(&c->mc_best_cb);
av_freep(&c->mc_charset);
av_freep(&c->mc_charmap);
av_freep(&c->mc_colram);
av_frame_free(&avctx->coded_frame);
av_free(c->mc_meta_charset);
av_free(c->mc_best_cb);
av_free(c->mc_charset);
av_free(c->mc_charmap);
av_free(c->mc_colram);
return 0;
}
@@ -224,7 +220,7 @@ static av_cold int a64multi_encode_init(AVCodecContext *avctx)
a64_palette[mc_colors[a]][2] * 0.11;
}
if (!(c->mc_meta_charset = av_mallocz_array(c->mc_lifetime, 32000 * sizeof(int))) ||
if (!(c->mc_meta_charset = av_malloc_array(c->mc_lifetime, 32000 * sizeof(int))) ||
!(c->mc_best_cb = av_malloc(CHARSET_CHARS * 32 * sizeof(int))) ||
!(c->mc_charmap = av_mallocz_array(c->mc_lifetime, 1000 * sizeof(int))) ||
!(c->mc_colram = av_mallocz(CHARSET_CHARS * sizeof(uint8_t))) ||
@@ -242,6 +238,14 @@ static av_cold int a64multi_encode_init(AVCodecContext *avctx)
AV_WB32(avctx->extradata, c->mc_lifetime);
AV_WB32(avctx->extradata + 16, INTERLACED);
avctx->coded_frame = av_frame_alloc();
if (!avctx->coded_frame) {
a64multi_close_encoder(avctx);
return AVERROR(ENOMEM);
}
avctx->coded_frame->pict_type = AV_PICTURE_TYPE_I;
avctx->coded_frame->key_frame = 1;
if (!avctx->codec_tag)
avctx->codec_tag = AV_RL32("a64m");
@@ -266,9 +270,10 @@ static void a64_compress_colram(unsigned char *buf, int *charmap, uint8_t *colra
}
static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
const AVFrame *p, int *got_packet)
const AVFrame *pict, int *got_packet)
{
A64Context *c = avctx->priv_data;
AVFrame *const p = avctx->coded_frame;
int frame;
int x, y;
@@ -299,7 +304,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
}
/* no data, means end encoding asap */
if (!p) {
if (!pict) {
/* all done, end encoding */
if (!c->mc_lifetime) return 0;
/* no more frames in queue, prepare to flush remaining frames */
@@ -312,10 +317,13 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
} else {
/* fill up mc_meta_charset with data until lifetime exceeds */
if (c->mc_frame_counter < c->mc_lifetime) {
*p = *pict;
p->pict_type = AV_PICTURE_TYPE_I;
p->key_frame = 1;
to_meta_with_crop(avctx, p, meta + 32000 * c->mc_frame_counter);
c->mc_frame_counter++;
if (c->next_pts == AV_NOPTS_VALUE)
c->next_pts = p->pts;
c->next_pts = pict->pts;
/* lifetime is not reached so wait for next frame first */
return 0;
}
@@ -326,8 +334,8 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
req_size = 0;
/* any frames to encode? */
if (c->mc_lifetime) {
int alloc_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
if ((ret = ff_alloc_packet2(avctx, pkt, alloc_size)) < 0)
req_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
if ((ret = ff_alloc_packet2(avctx, pkt, req_size)) < 0)
return ret;
buf = pkt->data;
@@ -343,7 +351,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
/* advance pointers */
buf += charset_size;
req_size += charset_size;
charset += charset_size;
}
/* write x frames to buf */
@@ -380,7 +388,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
pkt->pts = pkt->dts = c->next_pts;
c->next_pts = AV_NOPTS_VALUE;
av_assert0(pkt->size >= req_size);
pkt->size = req_size;
pkt->flags |= AV_PKT_FLAG_KEY;
*got_packet = !!req_size;

View File

@@ -245,7 +245,6 @@ typedef struct SingleChannelElement {
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct ChannelElement {
int present;
// CPE specific
int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
@@ -275,7 +274,6 @@ struct AACContext {
ChannelElement *che[4][MAX_ELEM_ID];
ChannelElement *tag_che_map[4][MAX_ELEM_ID];
int tags_mapped;
int warned_remapping_once;
/** @} */
/**
@@ -295,7 +293,7 @@ struct AACContext {
FFTContext mdct_ld;
FFTContext mdct_ltp;
FmtConvertContext fmt_conv;
AVFloatDSPContext *fdsp;
AVFloatDSPContext fdsp;
int random_state;
/** @} */

View File

@@ -57,7 +57,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
if (avpriv_aac_parse_header(&gb, &hdr) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing ADTS frame header!\n");
return AVERROR_INVALIDDATA;
return -1;
}
if (!hdr.crc_absent && hdr.num_aac_frames > 1) {

View File

@@ -34,7 +34,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
int size;
union {
uint64_t u64;
uint8_t u8[8 + FF_INPUT_BUFFER_PADDING_SIZE];
uint8_t u8[8];
} tmp;
tmp.u64 = av_be2ne64(state);

View File

@@ -31,7 +31,7 @@
#include "libavutil/mathematics.h"
float ff_aac_pow2sf_tab[428];
av_cold void ff_aac_tableinit(void)
void ff_aac_tableinit(void)
{
int i;
for (i = 0; i < 428; i++)

View File

@@ -53,7 +53,7 @@ static const uint8_t run_value_bits_short[16] = {
3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
};
static const uint8_t * const run_value_bits[2] = {
static const uint8_t *run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
@@ -691,7 +691,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
}
while (idx) {
sce->sf_idx[bandaddr[idx]] = minq + q0;
minq = FFMAX(paths[idx][minq].prev, 0);
minq = paths[idx][minq].prev;
idx--;
}
//set the same quantizers inside window groups
@@ -776,6 +776,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
do {
int prev = -1;
tbits = 0;
fflag = 0;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = w*128;
for (g = 0; g < sce->ics.num_swb; g++) {
@@ -952,6 +953,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
}
sce->zeroes[w*16+g] = 0;
scf = prev_scf = av_clip(SCALE_ONE_POS - SCALE_DIV_512 - log2f(1/maxq[w*16+g])*16/3, 60, 218);
step = 16;
for (;;) {
float dist = 0.0f;
int quant_max;

View File

@@ -425,7 +425,7 @@ static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
* Save current output configuration if and only if it has been locked.
*/
static void push_output_configuration(AACContext *ac) {
if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
if (ac->oc[1].status == OC_LOCKED) {
ac->oc[0] = ac->oc[1];
}
ac->oc[1].status = OC_NONE;
@@ -621,12 +621,6 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
* If we seem to have encountered such a stream, transfer
* the LFE[0] element to the SCE[1]'s mapping */
if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
av_log(ac->avctx, AV_LOG_WARNING,
"This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
type == TYPE_SCE ? "SCE" : "LFE", elem_id);
ac->warned_remapping_once++;
}
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
}
@@ -636,22 +630,6 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
}
case 4:
/* Some streams incorrectly code 4.0 audio as
* SCE[0] CPE[0] LFE[0]
* instead of
* SCE[0] CPE[0] SCE[1].
* If we seem to have encountered such a stream, transfer
* the SCE[1] element to the LFE[0]'s mapping */
if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
av_log(ac->avctx, AV_LOG_WARNING,
"This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
type == TYPE_SCE ? "SCE" : "LFE", elem_id);
ac->warned_remapping_once++;
}
ac->tags_mapped++;
return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
}
if (ac->tags_mapped == 2 &&
ac->oc[1].m4ac.chan_config == 4 &&
type == TYPE_SCE) {
@@ -703,7 +681,6 @@ static void decode_channel_map(uint8_t layout_map[][3],
syn_ele = TYPE_LFE;
break;
default:
// AAC_CHANNEL_OFF has no channel map
av_assert0(0);
}
layout_map[0][0] = syn_ele;
@@ -904,7 +881,7 @@ static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
if (len == 15 + 255)
len += get_bits(gb, 16);
if (get_bits_left(gb) < len * 8 + 4) {
av_log(avctx, AV_LOG_ERROR, overread_err);
av_log(ac->avctx, AV_LOG_ERROR, overread_err);
return AVERROR_INVALIDDATA;
}
skip_bits_long(gb, 8 * len);
@@ -1137,10 +1114,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_sbr_init();
ff_fmt_convert_init(&ac->fmt_conv, avctx);
ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
if (!ac->fdsp) {
return AVERROR(ENOMEM);
}
avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ac->random_state = 0x1f2e3d4c;
@@ -1516,12 +1490,13 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
int ms_present)
{
int idx;
int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
if (ms_present == 1) {
for (idx = 0; idx < max_idx; idx++)
for (idx = 0;
idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
idx++)
cpe->ms_mask[idx] = get_bits1(gb);
} else if (ms_present == 2) {
memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);
}
}
@@ -1643,9 +1618,9 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
cfo[k] = ac->random_state;
}
band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
scale = sf[idx] / sqrtf(band_energy);
ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
}
} else {
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
@@ -1791,7 +1766,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
}
} while (len -= 2);
ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
ac->fdsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
}
}
@@ -2004,7 +1979,7 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
cpe->ch[0].band_type[idx] < NOISE_BT &&
cpe->ch[1].band_type[idx] < NOISE_BT) {
for (group = 0; group < ics->group_len[g]; group++) {
ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
}
@@ -2043,7 +2018,7 @@ static void apply_intensity_stereo(AACContext *ac,
c *= 1 - 2 * cpe->ms_mask[idx];
scale = c * sce1->sf[idx];
for (group = 0; group < ics->group_len[g]; group++)
ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
ac->fdsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
offsets[i + 1] - offsets[i]);
@@ -2293,12 +2268,7 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
{
int crc_flag = 0;
int res = cnt;
int type = get_bits(gb, 4);
if (ac->avctx->debug & FF_DEBUG_STARTCODE)
av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
switch (type) { // extension type
switch (get_bits(gb, 4)) { // extension type
case EXT_SBR_DATA_CRC:
crc_flag++;
case EXT_SBR_DATA:
@@ -2411,15 +2381,15 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,
const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
} else {
memset(in, 0, 448 * sizeof(float));
ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
}
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
} else {
ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
memset(in + 1024 + 576, 0, 448 * sizeof(float));
}
ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
@@ -2472,17 +2442,17 @@ static void update_ltp(AACContext *ac, SingleChannelElement *sce)
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy(saved_ltp, saved, 512 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
for (i = 0; i < 64; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
for (i = 0; i < 64; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
} else { // LONG_STOP or ONLY_LONG
ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
ac->fdsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
for (i = 0; i < 512; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
}
@@ -2523,19 +2493,19 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
*/
if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
ac->fdsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512);
} else {
memcpy( out, saved, 448 * sizeof(float));
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
ac->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
ac->fdsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
} else {
ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
ac->fdsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
memcpy( out + 576, buf + 64, 448 * sizeof(float));
}
}
@@ -2543,9 +2513,9 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy( saved, temp + 64, 64 * sizeof(float));
ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
ac->fdsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy( saved, buf + 512, 448 * sizeof(float));
@@ -2570,10 +2540,10 @@ static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
if (ics->use_kb_window[1]) {
// AAC LD uses a low overlap sine window instead of a KBD window
memcpy(out, saved, 192 * sizeof(float));
ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
ac->fdsp.vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
memcpy( out + 320, buf + 64, 192 * sizeof(float));
} else {
ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256);
ac->fdsp.vector_fmul_window(out, saved, buf, ff_sine_512, 256);
}
// buffer update
@@ -2746,7 +2716,7 @@ static void spectral_to_sample(AACContext *ac)
for (type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
if (che && che->present) {
if (che) {
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
@@ -2778,9 +2748,6 @@ static void spectral_to_sample(AACContext *ac)
}
if (type <= TYPE_CCE)
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
che->present = 0;
} else if (che) {
av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
}
}
}
@@ -2885,7 +2852,6 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
elem_type, elem_id);
return AVERROR_INVALIDDATA;
}
che->present = 1;
if (aot != AOT_ER_AAC_ELD)
skip_bits(gb, 4);
switch (elem_type) {
@@ -2949,9 +2915,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
elem_id = get_bits(gb, 4);
if (avctx->debug & FF_DEBUG_STARTCODE)
av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
if (elem_type < TYPE_DSE) {
if (!(che=get_che(ac, elem_type, elem_id))) {
av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
@@ -2960,7 +2923,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
goto fail;
}
samples = 1024;
che->present = 1;
}
switch (elem_type) {
@@ -3059,12 +3021,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
AV_WL32(side, 2*AV_RL32(side));
}
if (!ac->frame->data[0] && samples) {
av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
err = AVERROR_INVALIDDATA;
goto fail;
}
*got_frame_ptr = !!samples;
if (samples) {
ac->frame->nb_samples = samples;
@@ -3134,7 +3090,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
if (INT_MAX / 8 <= buf_size)
return AVERROR_INVALIDDATA;
if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
return err;
switch (ac->oc[1].m4ac.object_type) {
@@ -3175,7 +3131,6 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
ff_mdct_end(&ac->mdct_small);
ff_mdct_end(&ac->mdct_ld);
ff_mdct_end(&ac->mdct_ltp);
av_freep(&ac->fdsp);
return 0;
}
@@ -3485,18 +3440,6 @@ static const AVClass aac_decoder_class = {
.version = LIBAVUTIL_VERSION_INT,
};
static const AVProfile profiles[] = {
{ FF_PROFILE_AAC_MAIN, "Main" },
{ FF_PROFILE_AAC_LOW, "LC" },
{ FF_PROFILE_AAC_SSR, "SSR" },
{ FF_PROFILE_AAC_LTP, "LTP" },
{ FF_PROFILE_AAC_HE, "HE-AAC" },
{ FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
{ FF_PROFILE_AAC_LD, "LD" },
{ FF_PROFILE_AAC_ELD, "ELD" },
{ FF_PROFILE_UNKNOWN },
};
AVCodec ff_aac_decoder = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
@@ -3513,7 +3456,6 @@ AVCodec ff_aac_decoder = {
.channel_layouts = aac_channel_layout,
.flush = flush,
.priv_class = &aac_decoder_class,
.profiles = profiles,
};
/*
@@ -3536,5 +3478,4 @@ AVCodec ff_aac_latm_decoder = {
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
.flush = flush,
.profiles = profiles,
};

View File

@@ -165,7 +165,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
PutBitContext pb;
AACEncContext *s = avctx->priv_data;
init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
put_bits(&pb, 5, 2); //object type - AAC-LC
put_bits(&pb, 4, s->samplerate_index); //sample rate index
put_bits(&pb, 4, s->channels);
@@ -252,7 +252,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
int i;
float *output = sce->ret_buf;
apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
@@ -567,10 +567,6 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
ics->group_len[w] = wi[ch].grouping[w];
apply_window_and_mdct(s, &cpe->ch[ch], overlap);
if (isnan(cpe->ch->coeffs[0])) {
av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
return AVERROR(EINVAL);
}
}
start_ch += chans;
}
@@ -682,7 +678,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
ff_psy_preprocess_end(s->psypp);
av_freep(&s->buffer.samples);
av_freep(&s->cpe);
av_freep(&s->fdsp);
ff_af_queue_close(&s->afq);
return 0;
}
@@ -691,9 +686,7 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
{
int ret = 0;
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
if (!s->fdsp)
return AVERROR(ENOMEM);
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
@@ -712,8 +705,8 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
{
int ch;
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
for(ch = 0; ch < s->channels; ch++)
@@ -753,10 +746,10 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
s->chan_map = aac_chan_configs[s->channels-1];
if ((ret = dsp_init(avctx, s)) < 0)
if (ret = dsp_init(avctx, s))
goto fail;
if ((ret = alloc_buffers(avctx, s)) < 0)
if (ret = alloc_buffers(avctx, s))
goto fail;
avctx->extradata_size = 5;
@@ -768,8 +761,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
lengths[1] = ff_aac_num_swb_128[i];
for (i = 0; i < s->chan_map[0]; i++)
grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
s->chan_map[0], grouping)) < 0)
if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
goto fail;
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[s->options.aac_coder];
@@ -784,7 +776,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
for (i = 0; i < 428; i++)
ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
avctx->initial_padding = 1024;
avctx->delay = 1024;
ff_af_queue_init(avctx, &s->afq);
return 0;

View File

@@ -67,7 +67,7 @@ typedef struct AACEncContext {
PutBitContext pb;
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
AVFloatDSPContext *fdsp;
AVFloatDSPContext fdsp;
float *planar_samples[6]; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index

View File

@@ -82,7 +82,7 @@ int main(void)
write_float_3d_array(f34_2_4, 4, 8, 2);
printf("};\n");
printf("static const DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
printf("static TABLE_CONST DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
printf("};\n");
printf("static const DECLARE_ALIGNED(16, float, phi_fract)[2][50][2] = {\n");

View File

@@ -70,7 +70,7 @@ static const float g2_Q4[] = {
0.16486303567403f, 0.23279856662996f, 0.25f
};
static av_cold void make_filters_from_proto(float (*filter)[8][2], const float *proto, int bands)
static void make_filters_from_proto(float (*filter)[8][2], const float *proto, int bands)
{
int q, n;
for (q = 0; q < bands; q++) {
@@ -82,7 +82,7 @@ static av_cold void make_filters_from_proto(float (*filter)[8][2], const float *
}
}
static av_cold void ps_tableinit(void)
static void ps_tableinit(void)
{
static const float ipdopd_sin[] = { 0, M_SQRT1_2, 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2 };
static const float ipdopd_cos[] = { 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2, 0, M_SQRT1_2 };

View File

@@ -727,10 +727,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
if (active_lines > 0.0f)
band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction);
pe += calc_pe_3gpp(band);
if (band->thr > 0.0f)
band->norm_fac = band->active_lines / band->thr;
else
band->norm_fac = 0.0f;
band->norm_fac = band->active_lines / band->thr;
norm_fac += band->norm_fac;
}
}

View File

@@ -30,7 +30,6 @@
#include "sbr.h"
#include "aacsbr.h"
#include "aacsbrdata.h"
#include "aacsbr_tablegen.h"
#include "fft.h"
#include "aacps.h"
#include "sbrdsp.h"
@@ -96,6 +95,7 @@ static void aacsbr_func_ptr_init(AACSBRContext *c);
av_cold void ff_aac_sbr_init(void)
{
int n;
static const struct {
const void *sbr_codes, *sbr_bits;
const unsigned int table_size, elem_size;
@@ -124,7 +124,13 @@ av_cold void ff_aac_sbr_init(void)
SBR_INIT_VLC_STATIC(8, 592);
SBR_INIT_VLC_STATIC(9, 512);
aacsbr_tableinit();
for (n = 1; n < 320; n++)
sbr_qmf_window_us[320 + n] = sbr_qmf_window_us[320 - n];
sbr_qmf_window_us[384] = -sbr_qmf_window_us[384];
sbr_qmf_window_us[512] = -sbr_qmf_window_us[512];
for (n = 0; n < 320; n++)
sbr_qmf_window_ds[n] = sbr_qmf_window_us[2*n];
ff_ps_init();
}
@@ -326,7 +332,7 @@ static int check_n_master(AVCodecContext *avctx, int n_master, int bs_xover_band
static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
SpectrumParameters *spectrum)
{
unsigned int temp, max_qmf_subbands = 0;
unsigned int temp, max_qmf_subbands;
unsigned int start_min, stop_min;
int k;
const int8_t *sbr_offset_ptr;
@@ -514,7 +520,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
/// High Frequency Generation - Patch Construction (14496-3 sp04 p216 fig. 4.46)
static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
{
int i, k, last_k = -1, last_msb = -1, sb = 0;
int i, k, sb = 0;
int msb = sbr->k[0];
int usb = sbr->kx[1];
int goal_sb = ((1000 << 11) + (sbr->sample_rate >> 1)) / sbr->sample_rate;
@@ -528,12 +534,6 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
do {
int odd = 0;
if (k == last_k && msb == last_msb) {
av_log(ac->avctx, AV_LOG_ERROR, "patch construction failed\n");
return AVERROR_INVALIDDATA;
}
last_k = k;
last_msb = msb;
for (i = k; i == k || sb > (sbr->k[0] - 1 + msb - odd); i--) {
sb = sbr->f_master[i];
odd = (sb + sbr->k[0]) & 1;
@@ -562,8 +562,7 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
k = sbr->n_master;
} while (sb != sbr->kx[1] + sbr->m[1]);
if (sbr->num_patches > 1 &&
sbr->patch_num_subbands[sbr->num_patches - 1] < 3)
if (sbr->num_patches > 1 && sbr->patch_num_subbands[sbr->num_patches-1] < 3)
sbr->num_patches--;
return 0;
@@ -1018,8 +1017,6 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
{
unsigned int cnt = get_bits_count(gb);
sbr->id_aac = id_aac;
if (id_aac == TYPE_SCE || id_aac == TYPE_CCE) {
if (read_sbr_single_channel_element(ac, sbr, gb)) {
sbr_turnoff(sbr);
@@ -1620,14 +1617,8 @@ static void sbr_hf_assemble(float Y1[38][64][2],
memcpy(q_temp[i + 2*ch_data->t_env[0]], sbr->q_m[0], m_max * sizeof(sbr->q_m[0][0]));
}
} else if (h_SL) {
for (i = 0; i < 4; i++) {
memcpy(g_temp[i + 2 * ch_data->t_env[0]],
g_temp[i + 2 * ch_data->t_env_num_env_old],
sizeof(g_temp[0]));
memcpy(q_temp[i + 2 * ch_data->t_env[0]],
q_temp[i + 2 * ch_data->t_env_num_env_old],
sizeof(q_temp[0]));
}
memcpy(g_temp[2*ch_data->t_env[0]], g_temp[2*ch_data->t_env_num_env_old], 4*sizeof(g_temp[0]));
memcpy(q_temp[2*ch_data->t_env[0]], q_temp[2*ch_data->t_env_num_env_old], 4*sizeof(q_temp[0]));
}
for (e = 0; e < ch_data->bs_num_env; e++) {
@@ -1696,12 +1687,6 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
int nch = (id_aac == TYPE_CPE) ? 2 : 1;
int err;
if (id_aac != sbr->id_aac) {
av_log(ac->avctx, AV_LOG_ERROR,
"element type mismatch %d != %d\n", id_aac, sbr->id_aac);
sbr_turnoff(sbr);
}
if (!sbr->kx_and_m_pushed) {
sbr->kx[0] = sbr->kx[1];
sbr->m[0] = sbr->m[1];
@@ -1714,7 +1699,7 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
}
for (ch = 0; ch < nch; ch++) {
/* decode channel */
sbr_qmf_analysis(ac->fdsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
sbr_qmf_analysis(&ac->fdsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
(float*)sbr->qmf_filter_scratch,
sbr->data[ch].W, sbr->data[ch].Ypos);
sbr->c.sbr_lf_gen(ac, sbr, sbr->X_low,
@@ -1725,7 +1710,6 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
sbr->c.sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1,
(const float (*)[40][2]) sbr->X_low, sbr->k[0]);
sbr_chirp(sbr, &sbr->data[ch]);
av_assert0(sbr->data[ch].bs_num_env > 0);
sbr_hf_gen(ac, sbr, sbr->X_high,
(const float (*)[40][2]) sbr->X_low,
(const float (*)[2]) sbr->alpha0,
@@ -1761,13 +1745,13 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
nch = 2;
}
sbr_qmf_synthesis(&sbr->mdct, &sbr->dsp, ac->fdsp,
sbr_qmf_synthesis(&sbr->mdct, &sbr->dsp, &ac->fdsp,
L, sbr->X[0], sbr->qmf_filter_scratch,
sbr->data[0].synthesis_filterbank_samples,
&sbr->data[0].synthesis_filterbank_samples_offset,
downsampled);
if (nch == 2)
sbr_qmf_synthesis(&sbr->mdct, &sbr->dsp, ac->fdsp,
sbr_qmf_synthesis(&sbr->mdct, &sbr->dsp, &ac->fdsp,
R, sbr->X[1], sbr->qmf_filter_scratch,
sbr->data[1].synthesis_filterbank_samples,
&sbr->data[1].synthesis_filterbank_samples_offset,

View File

@@ -1,39 +0,0 @@
/*
* Header file for hardcoded AAC SBR windows
*
* Copyright (c) 2014 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "libavutil/common.h"
#include "aacsbr_tablegen.h"
#include "tableprint.h"
int main(void)
{
aacsbr_tableinit();
write_fileheader();
WRITE_ARRAY_ALIGNED("static const", 32, float, sbr_qmf_window_ds);
WRITE_ARRAY_ALIGNED("static const", 32, float, sbr_qmf_window_us);
return 0;
}

View File

@@ -1,129 +0,0 @@
/*
* Header file for hardcoded AAC SBR windows
*
* Copyright (c) 2014 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
*
* 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 AVCODEC_AACSBR_TABLEGEN_H
#define AVCODEC_AACSBR_TABLEGEN_H
#if CONFIG_HARDCODED_TABLES
#define aacsbr_tableinit()
#include "libavcodec/aacsbr_tables.h"
#else
///< window coefficients for analysis/synthesis QMF banks
static DECLARE_ALIGNED(32, float, sbr_qmf_window_ds)[320];
static DECLARE_ALIGNED(32, float, sbr_qmf_window_us)[640] = {
0.0000000000, -0.0005525286, -0.0005617692, -0.0004947518,
-0.0004875227, -0.0004893791, -0.0005040714, -0.0005226564,
-0.0005466565, -0.0005677802, -0.0005870930, -0.0006132747,
-0.0006312493, -0.0006540333, -0.0006777690, -0.0006941614,
-0.0007157736, -0.0007255043, -0.0007440941, -0.0007490598,
-0.0007681371, -0.0007724848, -0.0007834332, -0.0007779869,
-0.0007803664, -0.0007801449, -0.0007757977, -0.0007630793,
-0.0007530001, -0.0007319357, -0.0007215391, -0.0006917937,
-0.0006650415, -0.0006341594, -0.0005946118, -0.0005564576,
-0.0005145572, -0.0004606325, -0.0004095121, -0.0003501175,
-0.0002896981, -0.0002098337, -0.0001446380, -0.0000617334,
0.0000134949, 0.0001094383, 0.0002043017, 0.0002949531,
0.0004026540, 0.0005107388, 0.0006239376, 0.0007458025,
0.0008608443, 0.0009885988, 0.0011250155, 0.0012577884,
0.0013902494, 0.0015443219, 0.0016868083, 0.0018348265,
0.0019841140, 0.0021461583, 0.0023017254, 0.0024625616,
0.0026201758, 0.0027870464, 0.0029469447, 0.0031125420,
0.0032739613, 0.0034418874, 0.0036008268, 0.0037603922,
0.0039207432, 0.0040819753, 0.0042264269, 0.0043730719,
0.0045209852, 0.0046606460, 0.0047932560, 0.0049137603,
0.0050393022, 0.0051407353, 0.0052461166, 0.0053471681,
0.0054196775, 0.0054876040, 0.0055475714, 0.0055938023,
0.0056220643, 0.0056455196, 0.0056389199, 0.0056266114,
0.0055917128, 0.0055404363, 0.0054753783, 0.0053838975,
0.0052715758, 0.0051382275, 0.0049839687, 0.0048109469,
0.0046039530, 0.0043801861, 0.0041251642, 0.0038456408,
0.0035401246, 0.0032091885, 0.0028446757, 0.0024508540,
0.0020274176, 0.0015784682, 0.0010902329, 0.0005832264,
0.0000276045, -0.0005464280, -0.0011568135, -0.0018039472,
-0.0024826723, -0.0031933778, -0.0039401124, -0.0047222596,
-0.0055337211, -0.0063792293, -0.0072615816, -0.0081798233,
-0.0091325329, -0.0101150215, -0.0111315548, -0.0121849995,
0.0132718220, 0.0143904666, 0.0155405553, 0.0167324712,
0.0179433381, 0.0191872431, 0.0204531793, 0.0217467550,
0.0230680169, 0.0244160992, 0.0257875847, 0.0271859429,
0.0286072173, 0.0300502657, 0.0315017608, 0.0329754081,
0.0344620948, 0.0359697560, 0.0374812850, 0.0390053679,
0.0405349170, 0.0420649094, 0.0436097542, 0.0451488405,
0.0466843027, 0.0482165720, 0.0497385755, 0.0512556155,
0.0527630746, 0.0542452768, 0.0557173648, 0.0571616450,
0.0585915683, 0.0599837480, 0.0613455171, 0.0626857808,
0.0639715898, 0.0652247106, 0.0664367512, 0.0676075985,
0.0687043828, 0.0697630244, 0.0707628710, 0.0717002673,
0.0725682583, 0.0733620255, 0.0741003642, 0.0747452558,
0.0753137336, 0.0758008358, 0.0761992479, 0.0764992170,
0.0767093490, 0.0768173975, 0.0768230011, 0.0767204924,
0.0765050718, 0.0761748321, 0.0757305756, 0.0751576255,
0.0744664394, 0.0736406005, 0.0726774642, 0.0715826364,
0.0703533073, 0.0689664013, 0.0674525021, 0.0657690668,
0.0639444805, 0.0619602779, 0.0598166570, 0.0575152691,
0.0550460034, 0.0524093821, 0.0495978676, 0.0466303305,
0.0434768782, 0.0401458278, 0.0366418116, 0.0329583930,
0.0290824006, 0.0250307561, 0.0207997072, 0.0163701258,
0.0117623832, 0.0069636862, 0.0019765601, -0.0032086896,
-0.0085711749, -0.0141288827, -0.0198834129, -0.0258227288,
-0.0319531274, -0.0382776572, -0.0447806821, -0.0514804176,
-0.0583705326, -0.0654409853, -0.0726943300, -0.0801372934,
-0.0877547536, -0.0955533352, -0.1035329531, -0.1116826931,
-0.1200077984, -0.1285002850, -0.1371551761, -0.1459766491,
-0.1549607071, -0.1640958855, -0.1733808172, -0.1828172548,
-0.1923966745, -0.2021250176, -0.2119735853, -0.2219652696,
-0.2320690870, -0.2423016884, -0.2526480309, -0.2631053299,
-0.2736634040, -0.2843214189, -0.2950716717, -0.3059098575,
-0.3168278913, -0.3278113727, -0.3388722693, -0.3499914122,
0.3611589903, 0.3723795546, 0.3836350013, 0.3949211761,
0.4062317676, 0.4175696896, 0.4289119920, 0.4402553754,
0.4515996535, 0.4629308085, 0.4742453214, 0.4855253091,
0.4967708254, 0.5079817500, 0.5191234970, 0.5302240895,
0.5412553448, 0.5522051258, 0.5630789140, 0.5738524131,
0.5845403235, 0.5951123086, 0.6055783538, 0.6159109932,
0.6261242695, 0.6361980107, 0.6461269695, 0.6559016302,
0.6655139880, 0.6749663190, 0.6842353293, 0.6933282376,
0.7022388719, 0.7109410426, 0.7194462634, 0.7277448900,
0.7358211758, 0.7436827863, 0.7513137456, 0.7587080760,
0.7658674865, 0.7727780881, 0.7794287519, 0.7858353120,
0.7919735841, 0.7978466413, 0.8034485751, 0.8087695004,
0.8138191270, 0.8185776004, 0.8230419890, 0.8272275347,
0.8311038457, 0.8346937361, 0.8379717337, 0.8409541392,
0.8436238281, 0.8459818469, 0.8480315777, 0.8497805198,
0.8511971524, 0.8523047035, 0.8531020949, 0.8535720573,
0.8537385600,
};
static av_cold void aacsbr_tableinit(void)
{
int n;
for (n = 1; n < 320; n++)
sbr_qmf_window_us[320 + n] = sbr_qmf_window_us[320 - n];
sbr_qmf_window_us[384] = -sbr_qmf_window_us[384];
sbr_qmf_window_us[512] = -sbr_qmf_window_us[512];
for (n = 0; n < 320; n++)
sbr_qmf_window_ds[n] = sbr_qmf_window_us[2*n];
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AVCODEC_AACSBR_TABLEGEN_H */

View File

@@ -266,6 +266,92 @@ static const int8_t sbr_offset[6][16] = {
{-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24}, // 64000 Hz < fs_sbr
};
///< window coefficients for analysis/synthesis QMF banks
static DECLARE_ALIGNED(32, float, sbr_qmf_window_ds)[320];
static DECLARE_ALIGNED(32, float, sbr_qmf_window_us)[640] = {
0.0000000000, -0.0005525286, -0.0005617692, -0.0004947518,
-0.0004875227, -0.0004893791, -0.0005040714, -0.0005226564,
-0.0005466565, -0.0005677802, -0.0005870930, -0.0006132747,
-0.0006312493, -0.0006540333, -0.0006777690, -0.0006941614,
-0.0007157736, -0.0007255043, -0.0007440941, -0.0007490598,
-0.0007681371, -0.0007724848, -0.0007834332, -0.0007779869,
-0.0007803664, -0.0007801449, -0.0007757977, -0.0007630793,
-0.0007530001, -0.0007319357, -0.0007215391, -0.0006917937,
-0.0006650415, -0.0006341594, -0.0005946118, -0.0005564576,
-0.0005145572, -0.0004606325, -0.0004095121, -0.0003501175,
-0.0002896981, -0.0002098337, -0.0001446380, -0.0000617334,
0.0000134949, 0.0001094383, 0.0002043017, 0.0002949531,
0.0004026540, 0.0005107388, 0.0006239376, 0.0007458025,
0.0008608443, 0.0009885988, 0.0011250155, 0.0012577884,
0.0013902494, 0.0015443219, 0.0016868083, 0.0018348265,
0.0019841140, 0.0021461583, 0.0023017254, 0.0024625616,
0.0026201758, 0.0027870464, 0.0029469447, 0.0031125420,
0.0032739613, 0.0034418874, 0.0036008268, 0.0037603922,
0.0039207432, 0.0040819753, 0.0042264269, 0.0043730719,
0.0045209852, 0.0046606460, 0.0047932560, 0.0049137603,
0.0050393022, 0.0051407353, 0.0052461166, 0.0053471681,
0.0054196775, 0.0054876040, 0.0055475714, 0.0055938023,
0.0056220643, 0.0056455196, 0.0056389199, 0.0056266114,
0.0055917128, 0.0055404363, 0.0054753783, 0.0053838975,
0.0052715758, 0.0051382275, 0.0049839687, 0.0048109469,
0.0046039530, 0.0043801861, 0.0041251642, 0.0038456408,
0.0035401246, 0.0032091885, 0.0028446757, 0.0024508540,
0.0020274176, 0.0015784682, 0.0010902329, 0.0005832264,
0.0000276045, -0.0005464280, -0.0011568135, -0.0018039472,
-0.0024826723, -0.0031933778, -0.0039401124, -0.0047222596,
-0.0055337211, -0.0063792293, -0.0072615816, -0.0081798233,
-0.0091325329, -0.0101150215, -0.0111315548, -0.0121849995,
0.0132718220, 0.0143904666, 0.0155405553, 0.0167324712,
0.0179433381, 0.0191872431, 0.0204531793, 0.0217467550,
0.0230680169, 0.0244160992, 0.0257875847, 0.0271859429,
0.0286072173, 0.0300502657, 0.0315017608, 0.0329754081,
0.0344620948, 0.0359697560, 0.0374812850, 0.0390053679,
0.0405349170, 0.0420649094, 0.0436097542, 0.0451488405,
0.0466843027, 0.0482165720, 0.0497385755, 0.0512556155,
0.0527630746, 0.0542452768, 0.0557173648, 0.0571616450,
0.0585915683, 0.0599837480, 0.0613455171, 0.0626857808,
0.0639715898, 0.0652247106, 0.0664367512, 0.0676075985,
0.0687043828, 0.0697630244, 0.0707628710, 0.0717002673,
0.0725682583, 0.0733620255, 0.0741003642, 0.0747452558,
0.0753137336, 0.0758008358, 0.0761992479, 0.0764992170,
0.0767093490, 0.0768173975, 0.0768230011, 0.0767204924,
0.0765050718, 0.0761748321, 0.0757305756, 0.0751576255,
0.0744664394, 0.0736406005, 0.0726774642, 0.0715826364,
0.0703533073, 0.0689664013, 0.0674525021, 0.0657690668,
0.0639444805, 0.0619602779, 0.0598166570, 0.0575152691,
0.0550460034, 0.0524093821, 0.0495978676, 0.0466303305,
0.0434768782, 0.0401458278, 0.0366418116, 0.0329583930,
0.0290824006, 0.0250307561, 0.0207997072, 0.0163701258,
0.0117623832, 0.0069636862, 0.0019765601, -0.0032086896,
-0.0085711749, -0.0141288827, -0.0198834129, -0.0258227288,
-0.0319531274, -0.0382776572, -0.0447806821, -0.0514804176,
-0.0583705326, -0.0654409853, -0.0726943300, -0.0801372934,
-0.0877547536, -0.0955533352, -0.1035329531, -0.1116826931,
-0.1200077984, -0.1285002850, -0.1371551761, -0.1459766491,
-0.1549607071, -0.1640958855, -0.1733808172, -0.1828172548,
-0.1923966745, -0.2021250176, -0.2119735853, -0.2219652696,
-0.2320690870, -0.2423016884, -0.2526480309, -0.2631053299,
-0.2736634040, -0.2843214189, -0.2950716717, -0.3059098575,
-0.3168278913, -0.3278113727, -0.3388722693, -0.3499914122,
0.3611589903, 0.3723795546, 0.3836350013, 0.3949211761,
0.4062317676, 0.4175696896, 0.4289119920, 0.4402553754,
0.4515996535, 0.4629308085, 0.4742453214, 0.4855253091,
0.4967708254, 0.5079817500, 0.5191234970, 0.5302240895,
0.5412553448, 0.5522051258, 0.5630789140, 0.5738524131,
0.5845403235, 0.5951123086, 0.6055783538, 0.6159109932,
0.6261242695, 0.6361980107, 0.6461269695, 0.6559016302,
0.6655139880, 0.6749663190, 0.6842353293, 0.6933282376,
0.7022388719, 0.7109410426, 0.7194462634, 0.7277448900,
0.7358211758, 0.7436827863, 0.7513137456, 0.7587080760,
0.7658674865, 0.7727780881, 0.7794287519, 0.7858353120,
0.7919735841, 0.7978466413, 0.8034485751, 0.8087695004,
0.8138191270, 0.8185776004, 0.8230419890, 0.8272275347,
0.8311038457, 0.8346937361, 0.8379717337, 0.8409541392,
0.8436238281, 0.8459818469, 0.8480315777, 0.8497805198,
0.8511971524, 0.8523047035, 0.8531020949, 0.8535720573,
0.8537385600,
};
/* First eight entries repeated at end to simplify SIMD implementations. */
const DECLARE_ALIGNED(16, float, ff_sbr_noise_table)[][2] = {
{-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},

View File

@@ -336,7 +336,7 @@ function fft_pass_neon
endfunc
.macro def_fft n, n2, n4
function fft\n\()_neon, align=6
function fft\n\()_neon align=6
sub sp, sp, #16
stp x28, x30, [sp]
add x28, x0, #\n4*2*8
@@ -376,8 +376,7 @@ function ff_fft_calc_neon, export=1
ld1 {v30.16b}, [x10]
mov x7, #-8
movrel x12, pmmp
ldr x4, [x3, x2, lsl #3]
add x3, x3, x4
ldr x3, [x3, x2, lsl #3]
movrel x13, mppm
movrel x14, X(ff_cos_16)
ld1 {v31.16b}, [x11]
@@ -417,21 +416,21 @@ function ff_fft_permute_neon, export=1
endfunc
const fft_tab_neon
.quad fft4_neon - fft_tab_neon
.quad fft8_neon - fft_tab_neon
.quad fft16_neon - fft_tab_neon
.quad fft32_neon - fft_tab_neon
.quad fft64_neon - fft_tab_neon
.quad fft128_neon - fft_tab_neon
.quad fft256_neon - fft_tab_neon
.quad fft512_neon - fft_tab_neon
.quad fft1024_neon - fft_tab_neon
.quad fft2048_neon - fft_tab_neon
.quad fft4096_neon - fft_tab_neon
.quad fft8192_neon - fft_tab_neon
.quad fft16384_neon - fft_tab_neon
.quad fft32768_neon - fft_tab_neon
.quad fft65536_neon - fft_tab_neon
.quad fft4_neon
.quad fft8_neon
.quad fft16_neon
.quad fft32_neon
.quad fft64_neon
.quad fft128_neon
.quad fft256_neon
.quad fft512_neon
.quad fft1024_neon
.quad fft2048_neon
.quad fft4096_neon
.quad fft8192_neon
.quad fft16384_neon
.quad fft32768_neon
.quad fft65536_neon
endconst
const pmmp, align=4

View File

@@ -78,7 +78,6 @@ av_cold void ff_h264dsp_init_aarch64(H264DSPContext *c, const int bit_depth,
c->h264_v_loop_filter_luma = ff_h264_v_loop_filter_luma_neon;
c->h264_h_loop_filter_luma = ff_h264_h_loop_filter_luma_neon;
c->h264_v_loop_filter_chroma = ff_h264_v_loop_filter_chroma_neon;
if (chroma_format_idc <= 1)
c->h264_h_loop_filter_chroma = ff_h264_h_loop_filter_chroma_neon;
c->weight_h264_pixels_tab[0] = ff_weight_h264_pixels_16_neon;

View File

@@ -27,73 +27,73 @@
#include "libavutil/aarch64/cpu.h"
#include "libavcodec/h264qpel.h"
void ff_put_h264_qpel16_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc00_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc10_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc20_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc30_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc01_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc11_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc21_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc31_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc02_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc12_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc22_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc32_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc03_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc13_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc23_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel16_mc33_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc00_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc10_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc20_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc30_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc01_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc11_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc21_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc31_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc02_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc12_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc22_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc32_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc03_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc13_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc23_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_put_h264_qpel8_mc33_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc00_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc10_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc20_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc30_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc01_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc11_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc21_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc31_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc02_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc12_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc22_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc32_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc03_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc13_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc23_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel16_mc33_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc00_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc10_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc20_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc30_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc01_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc11_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc21_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc31_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc02_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc12_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc22_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc32_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc03_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc13_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc23_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc33_neon(uint8_t *dst, const uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc00_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc10_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc20_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc30_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc01_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc11_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc21_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc31_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc02_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc12_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc22_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc32_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc03_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc13_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc23_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
void ff_avg_h264_qpel8_mc33_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride);
av_cold void ff_h264qpel_init_aarch64(H264QpelContext *c, int bit_depth)
{

View File

@@ -24,7 +24,7 @@
#define WFRAC_BITS 16 // fractional bits for window
#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
const tbl_rev128.s, align=4
const tbl_rev128.s align=4
.byte 12, 13, 14, 15
.byte 8, 9, 10, 11
.byte 4, 5, 6, 7

View File

@@ -23,7 +23,7 @@
#include "asm-offsets.h"
.macro shuffle a, b, c, d
const shuffle_\a\b\c\d, align=4
const shuffle_\a\b\c\d align=4
.byte (\a * 4), (\a * 4 + 1), (\a * 4 + 2), (\a * 4 + 3)
.byte (\b * 4), (\b * 4 + 1), (\b * 4 + 2), (\b * 4 + 3)
.byte (\c * 4), (\c * 4 + 1), (\c * 4 + 2), (\c * 4 + 3)
@@ -344,7 +344,7 @@ function fft15_pass
ret
endfunc
function fft30_neon, align=6
function fft30_neon align=6
sub sp, sp, #0x20
stp x20, x21, [sp]
stp x22, x30, [sp, #0x10]
@@ -372,7 +372,7 @@ function fft30_neon, align=6
endfunc
.macro def_fft n, n2
function fft\n\()_neon, align=6
function fft\n\()_neon align=6
sub sp, sp, #0x30
stp x20, x21, [sp]
stp x22, x30, [sp, #0x10]
@@ -438,8 +438,8 @@ function fft_b15_calc_neon
uzp1 v12.4s, v4.4s, v5.4s // exp[11 - 14].re
uzp2 v13.4s, v4.4s, v5.4s // exp[11 - 14].im
zip1 v14.4s, v6.4s, v7.4s // exp[5,10].re/exp[5,10].im
ldr x6, [x5, x3, lsl #3]
add x5, x5, x6
add x5, x5, x3, lsl #3
ldr x5, [x5]
mov x10, x0
blr x5
ldp x20, x30, [sp]
@@ -452,13 +452,13 @@ function fft_b15_calc_neon
endfunc
const fft_tab_neon
.quad fft15_neon - fft_tab_neon
.quad fft30_neon - fft_tab_neon
.quad fft60_neon - fft_tab_neon
.quad fft120_neon - fft_tab_neon
.quad fft240_neon - fft_tab_neon
.quad fft480_neon - fft_tab_neon
.quad fft960_neon - fft_tab_neon
.quad fft15_neon
.quad fft30_neon
.quad fft60_neon
.quad fft120_neon
.quad fft240_neon
.quad fft480_neon
.quad fft960_neon
endconst
function ff_celt_imdct_half_neon, export=1
@@ -641,7 +641,7 @@ function ff_celt_imdct_half_neon, export=1
endfunc
// [0] = exp(2 * i * pi / 5), [1] = exp(2 * i * pi * 2 / 5)
const fact5, align=4
const fact5 align=4
.float 0.30901699437494745, 0.95105651629515353
.float -0.80901699437494734, 0.58778525229247325
endconst

View File

@@ -1,6 +1,6 @@
/*
* Autodesk RLE Decoder
* Copyright (c) 2005 The FFmpeg Project
* Copyright (C) 2005 the ffmpeg project
*
* This file is part of FFmpeg.
*
@@ -137,7 +137,7 @@ static int aasc_decode_frame(AVCodecContext *avctx,
return ret;
/* report that the buffer was completely consumed */
return avpkt->size;
return buf_size;
}
static av_cold int aasc_decode_end(AVCodecContext *avctx)

View File

@@ -131,9 +131,6 @@ int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
int band_start, band_end, begin, end1;
int lowcomp, fastleak, slowleak;
if (end <= 0)
return AVERROR_INVALIDDATA;
/* excitation function */
band_start = ff_ac3_bin_to_band_tab[start];
band_end = ff_ac3_bin_to_band_tab[end-1] + 1;

View File

@@ -67,8 +67,7 @@
#define AC3_RENAME(x) x ## _fixed
#define AC3_NORM(norm) (1<<24)/(norm)
#define AC3_MUL(a,b) ((((int64_t) (a)) * (b))>>12)
#define AC3_RANGE(x) (x|((x&128)<<1))
#define AC3_HEAVY_RANGE(x) (x<<1)
#define AC3_RANGE(x) (x)
#define AC3_DYNAMIC_RANGE(x) (x)
#define AC3_SPX_BLEND(x) (x)
#define AC3_DYNAMIC_RANGE1 0
@@ -87,7 +86,6 @@
#define AC3_NORM(norm) (1.0f/(norm))
#define AC3_MUL(a,b) ((a) * (b))
#define AC3_RANGE(x) (dynamic_range_tab[(x)])
#define AC3_HEAVY_RANGE(x) (heavy_dynamic_range_tab[(x)])
#define AC3_DYNAMIC_RANGE(x) (powf(x, s->drc_scale))
#define AC3_SPX_BLEND(x) (x)* (1.0f/32)
#define AC3_DYNAMIC_RANGE1 1.0f

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