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44 Commits

Author SHA1 Message Date
Michael Niedermayer
b2c9cd36d3 Changelog: update for 2.6.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-16 20:09:11 +01:00
Michael Niedermayer
b3e6d3ee78 update for 2.6.1
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-16 19:45:24 +01:00
Michael Niedermayer
f6327e5fa6 avformat/mov: Disallow ".." in dref unless use_absolute_path is set
as this kind of allows to circumvent it to some extend.
We also could add a separate parameter or value to choose this

Found-by: ramiro
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1e4d0498df)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-16 04:01:58 +01:00
Clément Bœsch
b80c486fb3 avfilter/palettegen: make sure at least one frame was sent to the filter
Fix FPE.

(cherry picked from commit 84da9339c2)
2015-03-15 19:10:20 +01:00
Michael Niedermayer
37469af294 avformat/mov: Check for string truncation in mov_open_dref()
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 8003816e16)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-14 21:59:47 +01:00
Michael Niedermayer
350054e8e2 avformat/mov: Use sizeof(filename) instead of a literal number
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 21a53dd08d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-14 21:58:26 +01:00
Christophe Gisquet
7ff46a20d5 eac3dec: fix scaling
This is the remaining error, the output on the SPX samples,
respectively csi_miami_stereo_128_spx.eac3 and
csi_miami_5.1_256_spx.eac3, goes from:
stddev:    8.71 PSNR: 77.52 MAXDIFF:  235
stddev:24270.51 PSNR: 22.17 MAXDIFF:47166
to:
stddev:    0.12 PSNR:114.12 MAXDIFF:    1
stddev:    0.12 PSNR:114.73 MAXDIFF:    1

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0c3339f4bd)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-14 14:26:28 +01:00
Christophe Gisquet
7edd380668 ac3_fixed: fix computation of spx_noise_blend
It was set to 1 instead of sqrt(3)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit c4bf3833f4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-14 14:26:10 +01:00
Christophe Gisquet
26bed98d64 ac3_fixed: fix out-of-bound read
Should also improve decoding, but actually doesn't...

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b083440060)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-14 14:21:27 +01:00
Andreas Cadhalpun
480e18e6ff ac3dec_fixed: always use the USE_FIXED=1 variant of the AC3DecodeContext
The AC3DecodeContext has a float (USE_FIXED=0) and an integer
(USE_FIXED=1) variant, both of which can be present in the same binary.
This is not only very confusing, but it also breaks horribly, when one
variant is used by code expecting the other.

This currently happens, because eac3dec.c is only compiled for the float
variant, but also used from ac3dec_fixed.c, which uses the integer
variant.

The result is memory corruption, leading to crashes.

So compile eac3dec.c once for each variant and adapt it, so that it
works with the integer variant.

A loss of precission and scaling bug has been fixed by the committer
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 7b05b5093e)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-14 04:42:15 +01:00
James Almer
742bc7eea8 hevcdsp: fix compilation for arm and aarch64
Also add av_cold to ff_hevcdsp_init_arm.

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d5addf1555)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:32 +01:00
James Cowgill
c58edf52f9 mips/asmdefs: use _ABI64 as defined by gcc
Unfortunately android < api 21 (lollipop) doesn't have the sgidefs.h header,
the easiest way around this is to just use the preprocessor definitions from
gcc / clang.

Signed-off-by: James Cowgill <james410@cowgill.org.uk>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit a251aa1a35)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:32 +01:00
Rainer Hochecker
ea52c0614c hevc: delay ff_thread_finish_setup for hwaccel
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 31816eae32)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:32 +01:00
Michael Niedermayer
5e84ab838c avcodec/012v: redesign main loop
Fixes out of array accesses
Fixes: ffmpeg_012v_crash.ts

Found-by: Thomas Lindroth <thomas.lindroth@gmail.com>
Reviewed-by: Thomas Lindroth <thomas.lindroth@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 48df30d36c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Michael Niedermayer
3f7b89937d avcodec/012v: Check dimensions more completely
Fixes division by 0

Found-by: Thomas Lindroth <thomas.lindroth@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d3b25383da)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Andreas Cadhalpun
39c9b47bb6 asfenc: fix leaking asf->index_ptr on error
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2c8cff2be4)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Andreas Cadhalpun
016bf7cb81 roqvideoenc: set enc->avctx in roq_encode_init
So far it is only set in roq_encode_frame, but it is used in
roq_encode_end to free the coded_frame. This currently segfaults if
roq_encode_frame is not called between roq_encode_init and
roq_encode_end.

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit cf82c426fa)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Michael Niedermayer
bb00821f62 avcodec/options_table: remove extradata_size from the AVOptions table
allowing access to the size but not the extradata itself is not useful
and could lead to potential problems if writing happens through this field

Reviewed-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Reviewed-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1f4088b285)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Andreas Cadhalpun
d9dbd2362d ffmdec: limit the backward seek to the last resync position
If resyncing leads to the same position as previously, it will again
lead to a resync attempt, resulting in an infinite loop.

Thus don't seek back beyond the last syncpoint.

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6b8263b03a)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Arwa Arif
ec684aa58a Add dependencies to configure file for vf_fftfilt
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b4ec6afd3d)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Andreas Cadhalpun
a613dd627c ffmdec: make sure the time base is valid
A negative time base can trigger assertions.

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 4c91d81be2)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Andreas Cadhalpun
1903d6d2b0 ffmdec: fix infinite loop at EOF
If EOF is reached, while skipping bytes, avio_tell(pb) won't change
anymore, resulting in an infinite loop.

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6fa98822eb)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Andreas Cadhalpun
2319fddfd3 ffmdec: initialize f_cprv, f_stvi and f_stau
They are used in a switch statement, but it is not guaranteed that the
COMM case (where they are set to 0) is reached before the other cases.

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 66879ee125)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Martin Storsjö
5a6ef7d7cb configure: Move the .object_arch check to the right place
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit b77e335e44)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Martin Storsjö
47e8d17132 arm: Suppress tags about used cpu arch and extensions
When all the codepaths using manually set .arch/.fpu code is
behind runtime detection, the elf attributes should be suppressed.

This allows tools to know that the final built binary doesn't
strictly require these extensions.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit dcae2e32f7)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Michael Niedermayer
a156f86e91 avcodec/tiff: move bpp check to after "end:"
This ensures that all current and future code-pathes get bpp checked

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit d5e9fc7821)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Andreas Cadhalpun
3586314147 doc: avoid the incorrect phrase 'allow to'
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 10fd7ff814)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Martin Storsjö
28e17ba220 mxfdec: Fix the error handling for when strftime fails
The str variable is a char ** here.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 6448f15af0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Michael Niedermayer
998173ed94 avcodec/opusdec: Fix delayed sample value
Fixes out of array access
Fixes: ffmpeg_opus_crash1.ogg

This solution is likely not optimal in terms of error concealment but
its simple and fixes the out of array access.

Found-by: Thomas Lindroth <thomas.lindroth@gmail.com>
Tested-by: Thomas Lindroth <thomas.lindroth@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 6c583e9048)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Michael Niedermayer
bffed1d9d0 avcodec/opusdec: Clear out pointers per packet
This is safer than to assume that all error pathes cleared them and
nothing will use uncleared pointers.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 1ae092587f)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Michael Niedermayer
52bf144ad9 avcodec/utils: Align YUV411 by as much as the other YUV variants
Fixes out of array accesses
Fixes: ffmpeg_mjpeg_crash2.avi

Found-by: Thomas Lindroth <thomas.lindroth@gmail.com>
Tested-by: Thomas Lindroth <thomas.lindroth@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit e3201c38d5)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Xiangyu Liu
6fc8c72c7d ChangeLog: Remove the redundant VP9 RTP entry
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b1e9634c10)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-13 13:04:31 +01:00
Carl Eugen Hoyos
77621ca311 lavfi/boxblur: Fix colourspace list.
Fixes ticket #4363.
(cherry picked from commit 0637b59c2c)
2015-03-13 09:46:10 +01:00
Carl Eugen Hoyos
1a89aab3c0 lavf: Do not list mov-only codecs in riff.c.
Instead check for all mov code-points when demuxing avi
and print a warning if a video codec is found like this.
Fixes a regression similar to the one described in ticket #4307.
(cherry picked from commit 2e0b5f5c90)
2015-03-13 09:14:26 +01:00
Carl Eugen Hoyos
c43e5faf03 lavc/hevcdsp: Fix compilation for arm with --disable-neon.
(cherry picked from commit 1d523ea89a)
2015-03-10 15:00:08 +01:00
James Cowgill
876d2d8db8 mips/asmdefs: change include guard to read AVUTIL_ instead of AVCODEC_
Signed-off-by: James Cowgill <james410@cowgill.org.uk>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 3499a1c0a9)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-07 11:34:35 +01:00
Ronald S. Bultje
1caad74533 vp9: fix segmentation map retention with threading enabled.
Fixes ticket 4359.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit efff3854f0)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-07 10:38:07 +01:00
Michael Niedermayer
28950d1f8c Revert "avutil/opencl: is_compiled flag not being cleared in av_opencl_uninit"
Fixed build with opencl enabled
Found-by:  WJ Liu

This reverts commit 0f2359b869.
(cherry picked from commit ebd59d271c)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-07 10:38:07 +01:00
Reynaldo H. Verdejo Pinochet
791e085634 RELEASE_NOTES: small grammar fixes and improvements
Signed-off-by: Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
2015-03-06 22:15:47 -03:00
Clément Bœsch
3dc88e1fd6 Add release notes 2015-03-06 22:32:18 +01:00
Andreas Cadhalpun
4e2cab5a79 fix spelling errors
opttimizations -> optimizations
 grabing        -> grabbing
 many resource  -> many resources
 isnt           -> isn't
 silcense       -> silence

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 482c86f231)
2015-03-06 22:26:53 +01:00
Clément Bœsch
1dab67b647 avcodec/samidec: make sure to properly restore parsing context after a tag
(cherry picked from commit 70082a1e53)
2015-03-06 20:42:43 +01:00
Michael Niedermayer
0d4549c2d6 Add 2.6 to maintained releases
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 0bcb6ac150)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-06 17:49:22 +01:00
Michael Niedermayer
c741eb7d88 Update for 2.6
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-06 17:32:52 +01:00
1400 changed files with 18308 additions and 78813 deletions

1
.gitignore vendored
View File

@@ -36,7 +36,6 @@
/doc/avoptions_codec.texi
/doc/avoptions_format.texi
/doc/doxy/html/
/doc/examples/avio_list_dir
/doc/examples/avio_reading
/doc/examples/decoding_encoding
/doc/examples/demuxing_decoding

141
Changelog
View File

@@ -1,121 +1,30 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>:
version 2.7.2:
- imc: use correct position for flcoeffs2 calculation
- hevc: check slice address length
- snow: remove an obsolete av_assert2
- webp: fix infinite loop in webp_decode_frame
- wavpack: limit extra_bits to 32 and use get_bits_long
- ffmpeg: only count got_output/errors in decode_error_stat
- ffmpeg: exit_on_error if decoding a packet failed
- pthread_frame: forward error codes when flushing
- huffyuvdec: validate image size
- 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
- vc1dec: use get_bits_long and limit the read bits to 32
- mpegaudiodec: copy AVFloatDSPContext from first context to all contexts
- 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()
- ffmpeg: Fix cleanup after failed allocation of output_files
- 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
- Put a space between string literals and macros.
- avcodec/rawenc: Use ff_alloc_packet() instead of ff_alloc_packet2()
- 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/h264_refs: discard mismatching references
- avcodec/mjpegdec: Fix small picture upscale
- avcodec/pngdec: Check values before updating context in decode_fctl_chunk()
- avcodec/pngdec: Copy IHDR & plte state from last thread
- 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}
- 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
- swresample: soxr implementation for swr_get_out_samples()
- avformat/swfdec: Do not error out on pixel format changes
- ffmpeg_opt: Fix forcing fourccs
- configure: Check for x265_api_get
- swscale/x86/rgb2rgb_template: don't call emms on sse2/avx functions
- swscale/x86/rgb2rgb_template: add missing xmm clobbers
- library.mak: Workaround SDL redefining main and breaking fate tests on mingw
- vaapi_h264: fix RefPicList[] field flags.
version 2.7.1:
- postproc: fix unaligned access
- avformat: clarify what package needs to be compiled with SSL support
- avcodec/libx264: Avoid reconfig on equivalent aspect ratios
- avcodec/flacenc: Fix Invalid Rice order
- tls_gnutls: fix hang on disconnection
- avcodec/hevc_ps: Only discard overread VPS if a previous is available
- ffmpeg: Free last_frame instead of just unref
- avcodec/ffv1enc: fix bps for >8bit yuv when not explicitly set
- avio: fix potential crashes when combining ffio_ensure_seekback + crc
- examples/demuxing_decoding: use properties from frame instead of video_dec_ctx
- h264: er: Copy from the previous reference only if compatible
- doc: fix spelling errors
- configure: only disable VSX for !ppc64el
- ffmpeg_opt: Check for localtime() failure
- avformat/singlejpeg: fix standalone compilation
- configure: Disable VSX on unspecified / generic CPUs
- avformat: Fix bug in parse_rps for HEVC.
- takdec: ensure chan2 is a valid channel index
- avcodec/h264_slice: Use AVFrame dimensions for grayscale handling
version 2.7:
- FFT video filter
- TDSC decoder
- DTS lossless extension (XLL) decoding (not lossless, disabled by default)
- showwavespic filter
- DTS decoding through libdcadec
- Drop support for nvenc API before 5.0
- nvenc HEVC encoder
- Detelecine filter
- Intel QSV-accelerated H.264 encoding
- MMAL-accelerated H.264 decoding
- basic APNG encoder and muxer with default extension "apng"
- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
- WebM Live Chunk Muxer
- nvenc level and tier options
- chorus filter
- Canopus HQ/HQA decoder
- Automatically rotate videos based on metadata in ffmpeg
- improved Quickdraw compatibility
- VP9 high bit-depth and extended colorspaces decoding support
- WebPAnimEncoder API when available for encoding and muxing WebP
- Direct3D11-accelerated decoding
- Support Secure Transport
- Multipart JPEG demuxer
version 2.6.1:
- avformat/mov: Disallow ".." in dref unless use_absolute_path is set
- avfilter/palettegen: make sure at least one frame was sent to the filter
- avformat/mov: Check for string truncation in mov_open_dref()
- ac3_fixed: fix out-of-bound read
- mips/asmdefs: use _ABI64 as defined by gcc
- hevc: delay ff_thread_finish_setup for hwaccel
- avcodec/012v: Check dimensions more completely
- asfenc: fix leaking asf->index_ptr on error
- roqvideoenc: set enc->avctx in roq_encode_init
- avcodec/options_table: remove extradata_size from the AVOptions table
- ffmdec: limit the backward seek to the last resync position
- Add dependencies to configure file for vf_fftfilt
- ffmdec: make sure the time base is valid
- ffmdec: fix infinite loop at EOF
- ffmdec: initialize f_cprv, f_stvi and f_stau
- arm: Suppress tags about used cpu arch and extensions
- 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
- lavc/hevcdsp: Fix compilation for arm with --disable-neon.
- vp9: fix segmentation map retention with threading enabled.
- Revert "avutil/opencl: is_compiled flag not being cleared in av_opencl_uninit"
version 2.6:
- nvenc encoder
@@ -660,7 +569,7 @@ easier to use. The changes are:
all the stream in the first input file, except for the second audio
stream'.
* There is a new option -c (or -codec) for choosing the decoder/encoder to
use, which makes it possible to precisely specify target stream(s) consistently with
use, which allows to precisely specify target stream(s) consistently with
other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
libvorbis sets the codec for the first audio stream and -c copy copies all
the streams without reencoding. Old -vcodec/-acodec/-scodec options are now

View File

@@ -27,11 +27,9 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_blackframe.c`
- `vf_boxblur.c`
- `vf_colormatrix.c`
- `vf_cover_rect.c`
- `vf_cropdetect.c`
- `vf_delogo.c`
- `vf_eq.c`
- `vf_find_rect.c`
- `vf_fspp.c`
- `vf_geq.c`
- `vf_histeq.c`

View File

@@ -45,7 +45,7 @@ Miscellaneous Areas
documentation Stefano Sabatini, Mike Melanson, Timothy Gu
build system (configure, makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Lou Logan
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
@@ -59,7 +59,7 @@ 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
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan, Reynaldo H. Verdejo Pinochet
Twitter Lou Logan
Launchpad Timothy Gu
@@ -545,7 +545,6 @@ x86 Michael Niedermayer
Releases
========
2.7 Michael Niedermayer
2.6 Michael Niedermayer
2.5 Michael Niedermayer
2.4 Michael Niedermayer

View File

@@ -80,8 +80,8 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS \
HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS \
ALTIVEC-OBJS MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MSA-OBJS \
LOONGSON3-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS \
OBJS SLIBOBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=

View File

@@ -1 +1 @@
2.7.2
2.6.1

View File

@@ -1,12 +1,62 @@
┌─────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 2.7 "Nash" │
└─────────────────────────────────────┘
┌─────────────────────────────────────────────
│ RELEASE NOTES for FFmpeg 2.6 "Grothendieck" │
└─────────────────────────────────────────────
The FFmpeg Project proudly presents FFmpeg 2.7 "Nash", about 3
months after the release of FFmpeg 2.6.
The FFmpeg Project proudly presents FFmpeg 2.6 "Grothendieck", about 3
months after the release of FFmpeg 2.5.
A complete Changelog is available at the root of the project, and the
A lot of important work got in this time, so let's start talking about what
we like to brag the most about: features.
A lot of people will probably be happy to hear that we now have support for
NVENC — the Nvidia Video Encoder interface for H.264 encoding — thanks to
Timo Rothenpieler, with some little help from NVIDIA and Philip Langdale.
People in the broadcasting industry might also be interested in the first
steps of closed captions support with the introduction of a decoder by
Anshul Maheswhwari.
Regarding filters love, we improved and added many. We could talk about the
10-bit support in spp, but maybe it's more important to mention the addition
of colorlevels (yet another color handling filter), tblend (allowing you
to for example run a diff between successive frames of a video stream), or
the dcshift audio filter.
There are also two other important filters landing in libavfilter: palettegen
and paletteuse. Both submitted by the Stupeflix company. These filters will
be very useful in case you are looking for creating high quality GIFs, a
format that still bravely fights annihilation in 2015.
There are many other new features, but let's follow-up on one big cleanup
achievement: the libmpcodecs (MPlayer filters) wrapper is finally dead. The
last remaining filters (softpulldown/repeatfields, eq*, and various
postprocessing filters) were ported by Arwa Arif (OPW student) and Paul B
Mahol.
Concerning API changes, there are not many things to mention. Though, the
introduction of device inputs and outputs listing by Lukasz Marek is a
notable addition (try ffmpeg -sources or ffmpeg -sinks for an example of
the usage). As usual, see doc/APIchanges for more information.
Now let's talk about optimizations. Ronald S. Bultje made the VP9 decoder
usable on x86 32-bit systems and pre-ssse3 CPUs like Phenom (even dual core
Athlons can play 1080p 30fps VP9 content now), so we now secretly hope for
Google and Mozilla to use ffvp9 instead of libvpx. But VP9 is not the
center of attention anymore, and HEVC/H.265 is also getting many
improvements, which include C and x86 ASM optimizations, mainly from James
Almer, Christophe Gisquet and Pierre-Edouard Lepere.
Even though we had many x86 contributions, it is not the only architecture
getting some love, with Seppo Tomperi adding ARM NEON optimizations to the
HEVC stack, and James Cowgill adding MIPS64 assembly for all kind of audio
processing code in libavcodec.
And finally, Michael Niedermayer is still fixing many bugs, dealing with
most of the boring work such as making releases, applying tons of
contributors patches, and daily merging the changes from the Libav project.
A more complete Changelog is available at the root of the project, and the
complete Git history on http://source.ffmpeg.org.
We hope you will like this release as much as we enjoyed working on it, and

View File

@@ -7,11 +7,8 @@ OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes)
OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS) $(MIPSDSPR1-OBJS-yes)
OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS) $(MIPSDSPR2-OBJS-yes)
OBJS-$(HAVE_MSA) += $(MSA-OBJS) $(MSA-OBJS-yes)
OBJS-$(HAVE_LOONGSON3) += $(LOONGSON3-OBJS) $(LOONGSON3-OBJS-yes)
OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
OBJS-$(HAVE_VSX) += $(VSX-OBJS) $(VSX-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_YASM) += $(YASM-OBJS) $(YASM-OBJS-yes)

View File

@@ -41,10 +41,8 @@
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
#include "libavutil/display.h"
#include "libavutil/mathematics.h"
#include "libavutil/imgutils.h"
#include "libavutil/libm.h"
#include "libavutil/parseutils.h"
#include "libavutil/pixdesc.h"
#include "libavutil/eval.h"
@@ -482,22 +480,10 @@ static void dump_argument(const char *a)
fputc('"', report_file);
}
static void check_options(const OptionDef *po)
{
while (po->name) {
if (po->flags & OPT_PERFILE)
av_assert0(po->flags & (OPT_INPUT | OPT_OUTPUT));
po++;
}
}
void parse_loglevel(int argc, char **argv, const OptionDef *options)
{
int idx = locate_option(argc, argv, options, "loglevel");
const char *env;
check_options(options);
if (!idx)
idx = locate_option(argc, argv, options, "v");
if (idx && argv[idx + 1])
@@ -859,7 +845,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
{ "info" , AV_LOG_INFO },
{ "verbose", AV_LOG_VERBOSE },
{ "debug" , AV_LOG_DEBUG },
{ "trace" , AV_LOG_TRACE },
};
char *tail;
int level;
@@ -1550,10 +1535,10 @@ int show_protocols(void *optctx, const char *opt, const char *arg)
printf("Supported file protocols:\n"
"Input:\n");
while ((name = avio_enum_protocols(&opaque, 0)))
printf(" %s\n", name);
printf("%s\n", name);
printf("Output:\n");
while ((name = avio_enum_protocols(&opaque, 1)))
printf(" %s\n", name);
printf("%s\n", name);
return 0;
}
@@ -1568,7 +1553,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
printf("Filters:\n"
" T.. = Timeline support\n"
" .S. = Slice threading\n"
" ..C = Command support\n"
" ..C = Commmand support\n"
" A = Audio input/output\n"
" V = Video input/output\n"
" N = Dynamic number and/or type of input/output\n"
@@ -2087,30 +2072,6 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
return array;
}
double get_rotation(AVStream *st)
{
AVDictionaryEntry *rotate_tag = av_dict_get(st->metadata, "rotate", NULL, 0);
uint8_t* displaymatrix = av_stream_get_side_data(st,
AV_PKT_DATA_DISPLAYMATRIX, NULL);
double theta = 0;
if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
char *tail;
theta = av_strtod(rotate_tag->value, &tail);
if (*tail)
theta = 0;
}
if (displaymatrix && !theta)
theta = -av_display_rotation_get((int32_t*) displaymatrix);
theta -= 360*floor(theta/360 + 0.9/360);
if (fabs(theta - 90*round(theta/90)) > 2)
av_log_ask_for_sample(NULL, "Odd rotation angle\n");
return theta;
}
#if CONFIG_AVDEVICE
static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
{
@@ -2267,5 +2228,4 @@ int show_sinks(void *optctx, const char *opt, const char *arg)
av_log_set_level(error_level);
return ret;
}
#endif

View File

@@ -597,6 +597,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
char name[128];\
av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
double get_rotation(AVStream *st);
#endif /* CMDUTILS_H */

View File

@@ -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

@@ -7,7 +7,7 @@ all: all-yes
DEFAULT_YASMD=.dbg
ifeq ($(DBG),1)
ifeq (1, DBG)
YASMD=$(DEFAULT_YASMD)
else
YASMD=

View File

@@ -38,9 +38,40 @@
#ifndef __AVISYNTH_C__
#define __AVISYNTH_C__
#include "avs/config.h"
#include "avs/capi.h"
#include "avs/types.h"
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
typedef unsigned char BYTE;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
@@ -48,8 +79,8 @@
// Constants
//
#ifndef __AVISYNTH_6_H__
enum { AVISYNTH_INTERFACE_VERSION = 6 };
#ifndef __AVISYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 4 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
@@ -81,8 +112,8 @@ enum {AVS_CS_BGR = 1<<28,
AVS_CS_PLANAR = 1<<31,
AVS_CS_SHIFT_SUB_WIDTH = 0,
AVS_CS_SHIFT_SUB_HEIGHT = 8,
AVS_CS_SHIFT_SAMPLE_BITS = 16,
AVS_CS_SHIFT_SUB_HEIGHT = 1 << 3,
AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH,
AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
@@ -149,66 +180,15 @@ enum { //SUBTYPES
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
// New 2.6 explicitly defined cache hints.
AVS_CACHE_NOTHING=10, // Do not cache video.
AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
AVS_CACHE_GET_POLICY=30, // Get the current policy.
AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
AVS_CACHE_THREAD_SAFE=243, // Allow all threads in any instance.
AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#ifdef BUILDING_AVSCORE
struct AVS_ScriptEnvironment {
IScriptEnvironment * env;
const char * error;
AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
: env(e), error(0) {}
};
#endif
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
@@ -258,23 +238,29 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8 & AVS_CS_PLANAR_FILTER); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->image_type & property)==property ); }
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
@@ -288,18 +274,25 @@ AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
@@ -357,13 +350,11 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
p->fps_denominator = denominator/x;
}
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
#endif
/////////////////////////////////////////////////////////////////////
//
@@ -400,38 +391,89 @@ typedef struct AVS_VideoFrame {
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return avs_get_pitch_p(p, 0);}
#endif
return p->pitch;}
AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_sizeUV;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_sizeUV;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->heightUV;
return 0;
}
return p->height;}
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
return avs_get_read_ptr_p(p, 0);}
#endif
return p->vfb->data + p->offset;}
AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
return avs_get_write_ptr_p(p, 0);}
#endif
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
@@ -616,16 +658,12 @@ enum {
AVS_CPUF_SSSE3 = 0x200, // Core 2
AVS_CPUF_SSE4 = 0x400, // Penryn, Wolfdale, Yorkfield
AVS_CPUF_SSE4_1 = 0x400,
//AVS_CPUF_AVX = 0x800, // Sandy Bridge, Bulldozer
AVS_CPUF_SSE4_2 = 0x1000, // Nehalem
//AVS_CPUF_AVX2 = 0x2000, // Haswell
//AVS_CPUF_AVX512 = 0x4000, // Knights Landing
AVS_CPUF_SSE4_2 = 0x800, // Nehalem
};
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(int, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
@@ -662,12 +700,12 @@ AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
@@ -735,6 +773,7 @@ struct AVS_Library {
AVSC_DECLARE_FUNC(avs_function_exists);
AVSC_DECLARE_FUNC(avs_get_audio);
AVSC_DECLARE_FUNC(avs_get_cpu_flags);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_get_frame);
AVSC_DECLARE_FUNC(avs_get_parity);
AVSC_DECLARE_FUNC(avs_get_var);
@@ -759,27 +798,6 @@ struct AVS_Library {
AVSC_DECLARE_FUNC(avs_subframe_planar);
AVSC_DECLARE_FUNC(avs_take_clip);
AVSC_DECLARE_FUNC(avs_vsprintf);
AVSC_DECLARE_FUNC(avs_get_error);
AVSC_DECLARE_FUNC(avs_is_yv24);
AVSC_DECLARE_FUNC(avs_is_yv16);
AVSC_DECLARE_FUNC(avs_is_yv12);
AVSC_DECLARE_FUNC(avs_is_yv411);
AVSC_DECLARE_FUNC(avs_is_y8);
AVSC_DECLARE_FUNC(avs_is_color_space);
AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
AVSC_DECLARE_FUNC(avs_bits_per_pixel);
AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
AVSC_DECLARE_FUNC(avs_row_size);
AVSC_DECLARE_FUNC(avs_bmp_size);
AVSC_DECLARE_FUNC(avs_get_pitch_p);
AVSC_DECLARE_FUNC(avs_get_row_size_p);
AVSC_DECLARE_FUNC(avs_get_height_p);
AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
AVSC_DECLARE_FUNC(avs_is_writable);
AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
};
#undef AVSC_DECLARE_FUNC
@@ -787,7 +805,7 @@ struct AVS_Library {
AVSC_INLINE AVS_Library * avs_load_library() {
AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
if (library == NULL)
if (!library)
return NULL;
library->handle = LoadLibrary("avisynth");
if (library->handle == NULL)
@@ -814,6 +832,7 @@ AVSC_INLINE AVS_Library * avs_load_library() {
AVSC_LOAD_FUNC(avs_function_exists);
AVSC_LOAD_FUNC(avs_get_audio);
AVSC_LOAD_FUNC(avs_get_cpu_flags);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_get_frame);
AVSC_LOAD_FUNC(avs_get_parity);
AVSC_LOAD_FUNC(avs_get_var);
@@ -839,27 +858,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
AVSC_LOAD_FUNC(avs_take_clip);
AVSC_LOAD_FUNC(avs_vsprintf);
AVSC_LOAD_FUNC(avs_get_error);
AVSC_LOAD_FUNC(avs_is_yv24);
AVSC_LOAD_FUNC(avs_is_yv16);
AVSC_LOAD_FUNC(avs_is_yv12);
AVSC_LOAD_FUNC(avs_is_yv411);
AVSC_LOAD_FUNC(avs_is_y8);
AVSC_LOAD_FUNC(avs_is_color_space);
AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
AVSC_LOAD_FUNC(avs_bits_per_pixel);
AVSC_LOAD_FUNC(avs_bytes_from_pixels);
AVSC_LOAD_FUNC(avs_row_size);
AVSC_LOAD_FUNC(avs_bmp_size);
AVSC_LOAD_FUNC(avs_get_pitch_p);
AVSC_LOAD_FUNC(avs_get_row_size_p);
AVSC_LOAD_FUNC(avs_get_height_p);
AVSC_LOAD_FUNC(avs_get_read_ptr_p);
AVSC_LOAD_FUNC(avs_is_writable);
AVSC_LOAD_FUNC(avs_get_write_ptr_p);
#undef __AVSC_STRINGIFY
#undef AVSC_STRINGIFY
#undef AVSC_LOAD_FUNC
@@ -872,7 +870,7 @@ fail:
}
AVSC_INLINE void avs_free_library(AVS_Library *library) {
if (library == NULL)
if (!library)
return;
FreeLibrary(library->handle);
free(library);

View File

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

View File

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

View File

@@ -1,55 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_CONFIG_H
#define AVS_CONFIG_H
// Undefine this to get cdecl calling convention
#define AVSC_USE_STDCALL 1
// NOTE TO PLUGIN AUTHORS:
// Because FRAME_ALIGN can be substantially higher than the alignment
// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
// alignment. They should always request the exact alignment value they need.
// This is to make sure that plugins work over the widest range of AviSynth
// builds possible.
#define FRAME_ALIGN 32
#if defined(_M_AMD64) || defined(__x86_64)
# define X86_64
#elif defined(_M_IX86) || defined(__i386__)
# define X86_32
#else
# error Unsupported CPU architecture.
#endif
#endif //AVS_CONFIG_H

View File

@@ -1,51 +0,0 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_TYPES_H
#define AVS_TYPES_H
// Define all types necessary for interfacing with avisynth.dll
// Raster types used by VirtualDub & Avisynth
typedef unsigned int Pixel32;
typedef unsigned char BYTE;
// Audio Sample information
typedef float SFLOAT;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
#endif //AVS_TYPES_H

View File

@@ -66,8 +66,6 @@ static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr
struct thread_arg *thread_arg;
thread_arg = av_mallocz(sizeof(struct thread_arg));
if (!thread_arg)
return ENOMEM;
thread_arg->start_routine = start_routine;
thread_arg->arg = arg;

360
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -15,62 +15,6 @@ libavutil: 2014-08-09
API changes, most recent first:
-------- 8< --------- FFmpeg 2.7 was cut here -------- 8< ---------
2015-06-04 - cc17b43 - lswr 1.2.100
Add swr_get_out_samples()
2015-05-27 - c312bfa - lavu 54.26.100 - cpu.h
Add AV_CPU_FLAG_AVXSLOW.
2015-05-26 - 1fb9b2a - lavu 54.25.100 - rational.h
Add av_q2intfloat().
2015-05-13 - cc48409 / e7c5e17 - lavc 56.39.100 / 56.23.0
Add av_vda_default_init2.
2015-05-11 - 541d75f - lavf 56.33.100 - avformat.h
Add AVOpenCallback AVFormatContext.open_cb
2015-05-07 - a7dd933 - 56.38.100 - avcodec.h
Add av_packet_side_data_name().
2015-05-07 - 01e59d4 - 56.37.102 - avcodec.h
Add FF_PROFILE_VP9_2 and FF_PROFILE_VP9_3.
2015-05-04 - 079b7f6 - 56.37.100 - avcodec.h
Add FF_PROFILE_VP9_0 and FF_PROFILE_VP9_1.
2015-04-22 - 748d481 - lavf 56.31.100 - avformat.h
Add AVFMT_FLAG_FAST_SEEK flag. Some formats (initially mp3) use it to enable
fast, but inaccurate seeking.
2015-04-20 - 8e8219e / c253340 - lavu 54.23.100 / 54.12.0 - log.h
Add AV_LOG_TRACE for extremely verbose debugging.
2015-04-02 - 26e0e393 - lavf 56.29.100 - avio.h
Add AVIODirEntryType.AVIO_ENTRY_SERVER.
Add AVIODirEntryType.AVIO_ENTRY_SHARE.
Add AVIODirEntryType.AVIO_ENTRY_WORKGROUP.
2015-03-31 - 3188696 - lavu 54.22.100 - avstring.h
Add av_append_path_component()
2015-03-27 - 184084c - lavf 56.27.100 - avio.h url.h
New directory listing API.
Add AVIODirEntryType enum.
Add AVIODirEntry, AVIODirContext structures.
Add avio_open_dir(), avio_read_dir(), avio_close_dir(), avio_free_directory_entry().
Add ff_alloc_dir_entry().
Extend URLProtocol with url_open_dir(), url_read_dir(), url_close_dir().
2015-03-29 - 268ff17 / c484561 - lavu 54.21.100 / 54.10.0 - pixfmt.h
Add AV_PIX_FMT_MMAL for MMAL hardware acceleration.
2015-03-19 - 11fe56c - 56.29.100 / lavc 56.22.0
Add FF_PROFILE_DTS_EXPRESS.
-------- 8< --------- FFmpeg 2.6 was cut here -------- 8< ---------
2015-03-04 - cca4476 - lavf 56.25.100

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 = 2.7.2
PROJECT_NUMBER = 2.6.1
# 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

@@ -36,7 +36,6 @@ DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
DOC_EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec
DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding

View File

@@ -139,26 +139,6 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@section mp3_header_decompress
@section mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
DivX-style packed B-frames are not valid MPEG-4 and were only a
workaround for the broken Video for Windows subsystem.
They use more space, can cause minor AV sync issues, require more
CPU power to decode (unless the player has some decoded picture queue
to compensate the 2,0,2,0 frame per packet style) and cause
trouble if copied into a standard container like mp4 or mpeg-ps/ts,
because MPEG-4 decoders may not be able to decode them, since they are
not valid MPEG-4.
For example to fix an AVI file containing an MPEG-4 stream with
DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@end example
@section noise
Damages the contents of packets without damaging the container. Can be

View File

@@ -16,9 +16,6 @@ DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
GEN
Set to 1 to generate the missing or mismatched references.
Makefile targets:
all

View File

@@ -83,23 +83,6 @@ Loud sounds are fully compressed. Soft sounds are enhanced.
@end table
@section flac
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from Xiph.
@subsection FLAC Decoder options
@table @option
@item -use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc values
(like the default value). This option makes it possible to decode such streams
correctly by using lavc's old buggy lpc logic for decoding.
@end table
@section ffwavesynth
Internal wave synthetizer.

View File

@@ -151,7 +151,6 @@ probed and 0 otherwise.
@item auto_convert
If set to 1, try to perform automatic conversions on packet data to make the
streams concatenable.
The default is 1.
Currently, the only conversion is adding the h264_mp4toannexb bitstream
filter to H.264 streams in MP4 format. This is necessary in particular if
@@ -206,11 +205,6 @@ It accepts the following options:
Set the minimum valid delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 2.
@item max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds.
Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
the maximum value allowed by the specification.
@item default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
@@ -259,10 +253,6 @@ Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item none
Disable pattern matching, therefore the video will only contain the specified
image. You should use this option if you do not want to create sequences from
multiple images and your filenames may contain special pattern characters.
@item sequence
Select a sequence pattern type, used to specify a sequence of files
indexed by sequential numbers.

View File

@@ -228,7 +228,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
@example
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
@@ -239,7 +239,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
)
)
(setq c-default-style "ffmpeg")
@end lisp
@end example
@section Development Policy

View File

@@ -494,85 +494,6 @@ Selected by Encoder (default)
@end table
@anchor{flac}
@section flac
FLAC (Free Lossless Audio Codec) Encoder
@subsection Options
The following options are supported by FFmpeg's flac encoder.
@table @option
@item compression_level
Sets the compression level, which chooses defaults for many other options
if they are not set explicitly.
@item frame_size
Sets the size of the frames in samples per channel.
@item lpc_coeff_precision
Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the
default.
@item lpc_type
Sets the first stage LPC algorithm
@table @samp
@item none
LPC is not used
@item fixed
fixed LPC coefficients
@item levinson
@item cholesky
@end table
@item lpc_passes
Number of passes to use for Cholesky factorization during LPC analysis
@item min_partition_order
The minimum partition order
@item max_partition_order
The maximum partition order
@item prediction_order_method
@table @samp
@item estimation
@item 2level
@item 4level
@item 8level
@item search
Bruteforce search
@item log
@end table
@item ch_mode
Channel mode
@table @samp
@item auto
The mode is chosen automatically for each frame
@item indep
Chanels are independently coded
@item left_side
@item right_side
@item mid_side
@end table
@item exact_rice_parameters
Chooses if rice parameters are calculated exactly or approximately.
if set to 1 then they are chosen exactly, which slows the code down slightly and
improves compression slightly.
@item multi_dim_quant
Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is
applied after the first stage to finetune the coefficients. This is quite slow
and slightly improves compression.
@end table
@anchor{libfaac}
@section libfaac

View File

@@ -11,8 +11,7 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_list_dir \
avio_reading \
EXAMPLES= avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \

View File

@@ -1,120 +0,0 @@
/*
* Copyright (c) 2014 Lukasz Marek
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
static const char *type_string(int type)
{
switch (type) {
case AVIO_ENTRY_DIRECTORY:
return "<DIR>";
case AVIO_ENTRY_FILE:
return "<FILE>";
case AVIO_ENTRY_BLOCK_DEVICE:
return "<BLOCK DEVICE>";
case AVIO_ENTRY_CHARACTER_DEVICE:
return "<CHARACTER DEVICE>";
case AVIO_ENTRY_NAMED_PIPE:
return "<PIPE>";
case AVIO_ENTRY_SYMBOLIC_LINK:
return "<LINK>";
case AVIO_ENTRY_SOCKET:
return "<SOCKET>";
case AVIO_ENTRY_SERVER:
return "<SERVER>";
case AVIO_ENTRY_SHARE:
return "<SHARE>";
case AVIO_ENTRY_WORKGROUP:
return "<WORKGROUP>";
case AVIO_ENTRY_UNKNOWN:
default:
break;
}
return "<UNKNOWN>";
}
int main(int argc, char *argv[])
{
const char *input_dir = NULL;
AVIODirEntry *entry = NULL;
AVIODirContext *ctx = NULL;
int cnt, ret;
char filemode[4], uid_and_gid[20];
av_log_set_level(AV_LOG_DEBUG);
if (argc != 2) {
fprintf(stderr, "usage: %s input_dir\n"
"API example program to show how to list files in directory "
"accessed through AVIOContext.\n", argv[0]);
return 1;
}
input_dir = argv[1];
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
avformat_network_init();
if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
goto fail;
}
cnt = 0;
for (;;) {
if ((ret = avio_read_dir(ctx, &entry)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
goto fail;
}
if (!entry)
break;
if (entry->filemode == -1) {
snprintf(filemode, 4, "???");
} else {
snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
}
snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
if (cnt == 0)
av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
"TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
"ACCESSED", "STATUS_CHANGED");
av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
type_string(entry->type),
entry->size,
entry->name,
uid_and_gid,
filemode,
entry->modification_timestamp,
entry->access_timestamp,
entry->status_change_timestamp);
avio_free_directory_entry(&entry);
cnt++;
};
fail:
avio_close_dir(&ctx);
avformat_network_deinit();
return ret < 0 ? 1 : 0;
}

View File

@@ -81,24 +81,22 @@ static int decode_packet(int *got_frame, int cached)
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (video_dec_ctx->width != width || video_dec_ctx->height != height ||
video_dec_ctx->pix_fmt != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
video_dec_ctx->width, video_dec_ctx->height,
av_get_pix_fmt_name(video_dec_ctx->pix_fmt));
return -1;
}
if (*got_frame) {
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame%s n:%d coded_n:%d pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number,

View File

@@ -117,7 +117,7 @@ static int open_output_file(const char *filename)
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
return AVERROR_INVALIDDATA;
}

View File

@@ -13,36 +13,36 @@
@chapter Introduction
FATE is an extended regression suite on the client-side and a means
FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
The first part of this document explains how you can use FATE from
The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg's
FATE server.
In any way you can have a look at the publicly viewable FATE results
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@url{http://fate.ffmpeg.org/}
@url{http://fate.ffmpeg.org/}
This is especially recommended for all people contributing source
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
the developers could not test on.
The second part of this document describes how you can run FATE to
The second part of this document describes how you can run FATE to
submit your results to FFmpeg's FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
In the third part you can find a comprehensive listing of FATE makefile
In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine you need to have the samples
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@@ -51,11 +51,11 @@ make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
The above commands set the samples location by passing a makefile
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
@option{--samples=<path to the samples directory>}. Afterwards you can
invoke the makefile targets without setting the @var{SAMPLES} makefile
`--samples=<path to the samples directory>'. Afterwards you can
invoke the makefile targets without setting the SAMPLES makefile
variable. This is illustrated by the following commands:
@example
@@ -64,7 +64,7 @@ make fate-rsync
make fate
@end example
Yet another way to tell FATE about the location of the sample
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
@@ -85,7 +85,7 @@ To use a custom wrapper to run the test, pass @option{--target-exec} to
@chapter Submitting the results to the FFmpeg result aggregation server
To submit your results to the server you should run fate through the
To submit your results to the server you should run fate through the
shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
@@ -93,23 +93,23 @@ to be invoked with a configuration file as its first argument.
tests/fate.sh /path/to/fate_config
@end example
A configuration file template with comments describing the individual
A configuration file template with comments describing the individual
configuration variables can be found at @file{doc/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
template. The @env{slot} configuration variable can be any string that is not
Create a configuration that suits your needs, based on the configuration
template. The `slot' configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
configuration file itself will be sourced in a shell script, therefore all
shell features may be used. This enables you to setup the environment as you
need it for your build.
pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
itself will be sourced in a shell script, therefore all shell features may
be used. This enables you to setup the environment as you need it for your
build.
For your first test runs the @env{fate_recv} variable should be empty or
For your first test runs the `fate_recv' variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
@@ -122,29 +122,29 @@ present in $workdir as specified in the configuration file:
@item version
@end itemize
When you have everything working properly you can create an SSH key pair
When you have everything working properly you can create an SSH key pair
and send the public key to the FATE server administrator who can be contacted
at the email address @email{fate-admin@@ffmpeg.org}.
Configure your SSH client to use public key authentication with that key
Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server's fingerprint is:
@table @samp
@table @option
@item RSA
d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table
If you have problems connecting to the FATE server, it may help to try out
If you have problems connecting to the FATE server, it may help to try out
the @command{ssh} command with one or more @option{-v} options. You should
get detailed output concerning your SSH configuration and the authentication
process.
The only thing left is to automate the execution of the fate.sh script and
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@@ -165,7 +165,7 @@ Run the FATE test suite (requires the fate-suite dataset).
@section Makefile variables
@table @env
@table @option
@item V
Verbosity level, can be set to 0, 1 or 2.
@itemize
@@ -183,20 +183,20 @@ Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@item THREAD_TYPE
Specify which threading strategy test, either @samp{slice} or @samp{frame},
by default @samp{slice+frame}
Specify which threading strategy test, either @var{slice} or @var{frame},
by default @var{slice+frame}
@item CPUFLAGS
Specify CPU flags.
@item TARGET_EXEC
Specify or override the wrapper used to run the tests.
The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
through @command{ssh}.
@item GEN
Set to @samp{1} to generate the missing or mismatched references.
Set to @var{1} to generate the missing or mismatched references.
@end table
@section Examples

View File

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

View File

@@ -80,7 +80,7 @@ The format option may be needed for raw input files.
The transcoding process in @command{ffmpeg} for each output can be described by
the following diagram:
@verbatim
@example
_______ ______________
| | | |
| input | demuxer | encoded data | decoder
@@ -99,7 +99,7 @@ the following diagram:
|________| |______________|
@end verbatim
@end example
@command{ffmpeg} calls the libavformat library (containing demuxers) to read
input files and get packets containing encoded data from them. When there are
@@ -124,7 +124,7 @@ Simple filtergraphs are those that have exactly one input and output, both of
the same type. In the above diagram they can be represented by simply inserting
an additional step between decoding and encoding:
@verbatim
@example
_________ ______________
| | | |
| decoded | | encoded data |
@@ -136,19 +136,19 @@ an additional step between decoding and encoding:
| frames |
|__________|
@end verbatim
@end example
Simple filtergraphs are configured with the per-stream @option{-filter} option
(with @option{-vf} and @option{-af} aliases for video and audio respectively).
A simple filtergraph for video can look for example like this:
@verbatim
@example
_______ _____________ _______ ________
| | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | output |
|_______| |_____________| |_______| |________|
@end verbatim
@end example
Note that some filters change frame properties but not frame contents. E.g. the
@code{fps} filter in the example above changes number of frames, but does not
@@ -161,7 +161,7 @@ processing chain applied to one stream. This is the case, for example, when the
more than one input and/or output, or when output stream type is different from
input. They can be represented with the following diagram:
@verbatim
@example
_________
| |
| input 0 |\ __________
@@ -179,7 +179,7 @@ input. They can be represented with the following diagram:
| input 2 |/
|_________|
@end verbatim
@end example
Complex filtergraphs are configured with the @option{-filter_complex} option.
Note that this option is global, since a complex filtergraph, by its nature,
@@ -198,14 +198,14 @@ step for the specified stream, so it does only demuxing and muxing. It is useful
for changing the container format or modifying container-level metadata. The
diagram above will, in this case, simplify to this:
@verbatim
@example
_______ ______________ ________
| | | | | |
| input | demuxer | encoded data | muxer | output |
| file | ---------> | packets | -------> | file |
|_______| |______________| |________|
@end verbatim
@end example
Since there is no decoding or encoding, it is very fast and there is no quality
loss. However, it might not work in some cases because of many factors. Applying
@@ -462,9 +462,6 @@ Technical note -- attachments are implemented as codec extradata, so this
option can actually be used to extract extradata from any stream, not just
attachments.
@item -noautorotate
Disable automatically rotating video based on file metadata.
@end table
@section Video Options
@@ -835,14 +832,6 @@ ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
Note that using this option disables the default mappings for this output file.
@item -ignore_unknown
Ignore input streams with unknown type instead of failing if copying
such streams is attempted.
@item -copy_unknown
Allow input streams with unknown type to be copied instead of failing if copying
such streams is attempted.
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
Map an audio channel from a given input to an output. If
@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
@@ -1169,14 +1158,7 @@ This option enables or disables accurate seeking in input files with the
transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
e.g. when copying some streams and transcoding the others.
@item -seek_timestamp (@emph{input})
This option enables or disables seeking by timestamp in input files with the
@option{-ss} option. It is disabled by default. If enabled, the argument
to the @option{-ss} option is considered an actual timestamp, and is not
offset by the start time of the file. This matters only for files which do
not start from timestamp 0, such as transport streams.
@item -thread_queue_size @var{size} (@emph{input})
@item -thread_message_queue @var{size} (@emph{input})
This option sets the maximum number of queued packets when reading from the
file or device. With low latency / high rate live streams, packets may be
discarded if they are not read in a timely manner; raising this value can
@@ -1221,9 +1203,6 @@ Discard all frames excepts keyframes.
Discard all frames.
@end table
@item -xerror (@emph{global})
Stop and exit on error
@end table
As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1490,7 +1469,7 @@ combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
@example
ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example
The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
@@ -1505,7 +1484,7 @@ image2-specific @code{-pattern_type glob} option.
For example, for creating a video from filenames matching the glob pattern
@code{foo-*.jpeg}:
@example
ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
@end example
@item

View File

@@ -161,7 +161,7 @@ Force a specific video decoder.
Force a specific subtitle decoder.
@item -autorotate
Automatically rotate the video according to file metadata. Enabled by
Automatically rotate the video according to presentation metadata. Enabled by
default, use @option{-noautorotate} to disable it.
@item -framedrop

View File

@@ -447,17 +447,17 @@ writer).
It can assume one of the following values:
@table @option
@item c
Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
character (@samp{\}) or the item separator character @var{SEP} are escaped
using C-like fashioned escaping, so that a newline is converted to the
sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
the separator @var{SEP} is converted to @samp{\@var{SEP}}.
Perform C-like escaping. Strings containing a newline ('\n'), carriage
return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
escaping, so that a newline is converted to the sequence "\n", a
carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
converted to "\@var{SEP}".
@item csv
Perform CSV-like escaping, as described in RFC4180. Strings
containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
(@samp{"}), or @var{SEP} are enclosed in double-quotes.
containing a newline ('\n'), a carriage return ('\r'), a double quote
('"'), or @var{SEP} are enclosed in double-quotes.
@item none
Perform no escaping.
@@ -485,7 +485,7 @@ The description of the accepted options follows.
Separator character used to separate the chapter, the section name, IDs and
potential tags in the printed field key.
Default value is @samp{.}.
Default value is '.'.
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
@@ -507,22 +507,21 @@ The following conventions are adopted:
@item
all key and values are UTF-8
@item
@samp{.} is the subgroup separator
'.' is the subgroup separator
@item
newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
escaped
newline, '\t', '\f', '\b' and the following characters are escaped
@item
@samp{\} is the escape character
'\' is the escape character
@item
@samp{#} is the comment indicator
'#' is the comment indicator
@item
@samp{=} is the key/value separator
'=' is the key/value separator
@item
@samp{:} is not used but usually parsed as key/value separator
':' is not used but usually parsed as key/value separator
@end itemize
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by @samp{:}.
separated by ":".
The description of the accepted options follows.

View File

@@ -48,11 +48,6 @@
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
@@ -70,16 +65,6 @@
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="packetSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -87,7 +72,6 @@
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="stream_index" type="xsd:int" />
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
@@ -171,7 +155,6 @@
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
@@ -188,8 +171,6 @@
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="coded_width" type="xsd:int"/>
<xsd:attribute name="coded_height" type="xsd:int"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -273,8 +254,6 @@
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string"/>
<xsd:attribute name="build_time" type="xsd:string"/>
<xsd:attribute name="compiler_ident" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>

View File

@@ -82,7 +82,6 @@ Feed feed1.ffm
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# mjpeg : Generate a M-JPEG stream.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)

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@@ -118,8 +118,7 @@ Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following
graph:
@verbatim
@example
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
@@ -144,8 +143,7 @@ ffmpeg 2 -----| feed 3 |-----| stream 4 |
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
@end verbatim
@end example
@anchor{FFM}
@section FFM, FFM2 formats

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@@ -50,9 +50,6 @@ Match the stream by stream id (e.g. PID in MPEG-TS container).
Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value.
@item u
Matches streams with usable configuration, the codec must be defined and the
essential information such as video dimension or audio sample rate must be present.
Note that in @command{ffmpeg}, matching by metadata will only work properly for
input files.
@@ -190,7 +187,6 @@ warnings and errors. This is the default value.
Same as @code{info}, except more verbose.
@item debug, 48
Show everything, including debugging information.
@item trace, 56
@end table
By default the program logs to stderr, if coloring is supported by the
@@ -265,14 +261,10 @@ 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
@@ -283,13 +275,6 @@ 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

@@ -8,13 +8,13 @@ outputs.
To illustrate the sorts of things that are possible, we consider the
following filtergraph.
@verbatim
@example
[main]
input --> split ---------------------> overlay --> output
| ^
|[tmp] [flip]|
+-----> crop --> vflip -------+
@end verbatim
@end example
This filtergraph splits the input stream in two streams, then sends one
stream through the crop filter and the vflip filter, before merging it
@@ -163,13 +163,13 @@ follow the same constraints order of the previous point. The following
If the option value itself is a list of items (e.g. the @code{format} filter
takes a list of pixel formats), the items in the list are usually separated by
@samp{|}.
'|'.
The list of arguments can be quoted using the character @samp{'} as initial
and ending mark, and the character @samp{\} for escaping the characters
The list of arguments can be quoted using the character "'" as initial
and ending mark, and the character '\' for escaping the characters
within the quoted text; otherwise the argument string is considered
terminated when the next special character (belonging to the set
@samp{[]=;,}) is encountered.
"[]=;,") is encountered.
The name and arguments of the filter are optionally preceded and
followed by a list of link labels.
@@ -1320,61 +1320,6 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
side_right.wav
@end example
@section chorus
Add a chorus effect to the audio.
Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
constant, with chorus, it is varied using using sinusoidal or triangular modulation.
The modulation depth defines the range the modulated delay is played before or after
the delay. Hence the delayed sound will sound slower or faster, that is the delayed
sound tuned around the original one, like in a chorus where some vocals are slightly
off key.
It accepts the following parameters:
@table @option
@item in_gain
Set input gain. Default is 0.4.
@item out_gain
Set output gain. Default is 0.4.
@item delays
Set delays. A typical delay is around 40ms to 60ms.
@item decays
Set decays.
@item speeds
Set speeds.
@item depths
Set depths.
@end table
@subsection Examples
@itemize
@item
A single delay:
@example
chorus=0.7:0.9:55:0.4:0.25:2
@end example
@item
Two delays:
@example
chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
@end example
@item
Fuller sounding chorus with three delays:
@example
chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3
@end example
@end itemize
@section compand
Compress or expand the audio's dynamic range.
@@ -2748,11 +2693,8 @@ Available values for component modes are:
@item divide
@item dodge
@item exclusion
@item glow
@item hardlight
@item hardmix
@item lighten
@item linearlight
@item multiply
@item negation
@item normal
@@ -3786,33 +3728,6 @@ FFmpeg was configured with @code{--enable-opencl}. Default value is 0.
@end table
@section detelecine
Apply an exact inverse of the telecine operation. It requires a predefined
pattern specified using the pattern option which must be the same as that passed
to the telecine filter.
This filter accepts the following options:
@table @option
@item first_field
@table @samp
@item top, t
top field first
@item bottom, b
bottom field first
The default value is @code{top}.
@end table
@item pattern
A string of numbers representing the pulldown pattern you wish to apply.
The default value is @code{23}.
@item start_frame
A number representing position of the first frame with respect to the telecine
pattern. This is to be used if the stream is cut. The default value is @code{0}.
@end table
@section drawbox
Draw a colored box on the input image.
@@ -4013,10 +3928,6 @@ Used to draw a box around text using the background color.
The value must be either 1 (enable) or 0 (disable).
The default value of @var{box} is 0.
@item boxborderw
Set the width of the border to be drawn around the box using @var{boxcolor}.
The default value of @var{boxborderw} is 0.
@item boxcolor
The color to be used for drawing box around text. For the syntax of this
option, check the "Color" section in the ffmpeg-utils manual.
@@ -4060,16 +3971,6 @@ The font family to be used for drawing text. By default Sans.
The font file to be used for drawing text. The path must be included.
This parameter is mandatory if the fontconfig support is disabled.
@item draw
This option does not exist, please see the timeline system
@item alpha
Draw the text applying alpha blending. The value can
be either a number between 0.0 and 1.0
The expression accepts the same variables @var{x, y} do.
The default value is 1.
Please see fontcolor_expr
@item fontsize
The font size to be used for drawing text.
The default value of @var{fontsize} is 16.
@@ -4248,7 +4149,7 @@ If @option{expansion} is set to @code{none}, the text is printed verbatim.
If @option{expansion} is set to @code{normal} (which is the default),
the following expansion mechanism is used.
The backslash character @samp{\}, followed by any character, always expands to
The backslash character '\', followed by any character, always expands to
the second character.
Sequence of the form @code{%@{...@}} are expanded. The text between the
@@ -4280,9 +4181,8 @@ value.
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 @samp{x},
@samp{X}, @samp{d} and @samp{u}. They are treated exactly as in the
@code{printf} 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.
@@ -4466,72 +4366,40 @@ The filter accepts the following options:
@table @option
@item contrast
Set the contrast expression. The value must be a float value in range
@code{-2.0} to @code{2.0}. The default value is "0".
Set the contrast value. It accepts a float value in range @code{-2.0} to
@code{2.0}. The default value is @code{0.0}.
@item brightness
Set the brightness expression. The value must be a float value in
range @code{-1.0} to @code{1.0}. The default value is "0".
Set the brightness value. It accepts a float value in range @code{-1.0} to
@code{1.0}. The default value is @code{0.0}.
@item saturation
Set the saturation expression. The value must be a float in
range @code{0.0} to @code{3.0}. The default value is "1".
Set the saturation value. It accepts a float value in range @code{0.0} to
@code{3.0}. The default value is @code{1.0}.
@item gamma
Set the gamma expression. The value must be a float in range
@code{0.1} to @code{10.0}. The default value is "1".
Set the gamma value. It accepts a float value in range @code{0.1} to @code{10.0}.
The default value is @code{1.0}.
@item gamma_r
Set the gamma expression for red. The value must be a float in
range @code{0.1} to @code{10.0}. The default value is "1".
Set the gamma value for red. It accepts a float value in range
@code{0.1} to @code{10.0}. The default value is @code{1.0}.
@item gamma_g
Set the gamma expression for green. The value must be a float in range
@code{0.1} to @code{10.0}. The default value is "1".
Set the gamma value for green. It accepts a float value in range
@code{0.1} to @code{10.0}. The default value is @code{1.0}.
@item gamma_b
Set the gamma expression for blue. The value must be a float in range
@code{0.1} to @code{10.0}. The default value is "1".
Set the gamma value for blue. It accepts a float value in range
@code{0.1} to @code{10.0}. The default value is @code{1.0}.
@item gamma_weight
Set the gamma weight expression. It can be used to reduce the effect
of a high gamma value on bright image areas, e.g. keep them from
getting overamplified and just plain white. The value must be a float
in range @code{0.0} to @code{1.0}. A value of @code{0.0} turns the
gamma correction all the way down while @code{1.0} leaves it at its
full strength. Default is "1".
Can be used to reduce the effect of a high gamma value on bright image areas,
e.g. keep them from getting overamplified and just plain white. It accepts a
float value in range @code{0.0} to @code{1.0}.A value of @code{0.0} turns the
gamma correction all the way down while @code{1.0} leaves it at its full strength.
Default is @code{1.0}.
@item eval
Set when the expressions for brightness, contrast, saturation and
gamma expressions are evaluated.
It accepts the following values:
@table @samp
@item init
only evaluate expressions once during the filter initialization or
when a command is processed
@item frame
evaluate expressions for each incoming frame
@end table
Default value is @samp{init}.
@end table
The expressions accept the following parameters:
@table @option
@item n
frame count of the input frame starting from 0
@item pos
byte position of the corresponding packet in the input file, NAN if
unspecified
@item r
frame rate of the input video, NAN if the input frame rate is unknown
@item t
timestamp expressed in seconds, NAN if the input timestamp is unknown
@end table
@subsection Commands
@@ -4670,8 +4538,7 @@ The number of seconds for which the fade effect has to last. At the end of the
fade-in effect the output video will have the same intensity as the input video,
at the end of the fade-out transition the output video will be filled with the
selected @option{color}.
If both duration and nb_frames are specified, duration is used. Default is 0
(nb_frames is used by default).
If both duration and nb_frames are specified, duration is used. Default is 0.
@item color, c
Specify the color of the fade. Default is "black".
@@ -4724,67 +4591,6 @@ fade=t=in:st=5.5:d=0.5
@end itemize
@section fftfilt
Apply arbitrary expressions to samples in frequency domain
@table @option
@item dc_Y
Adjust the dc value (gain) of the luma plane of the image. The filter
accepts an integer value in range @code{0} to @code{1000}. The default
value is set to @code{0}.
@item dc_U
Adjust the dc value (gain) of the 1st chroma plane of the image. The
filter accepts an integer value in range @code{0} to @code{1000}. The
default value is set to @code{0}.
@item dc_V
Adjust the dc value (gain) of the 2nd chroma plane of the image. The
filter accepts an integer value in range @code{0} to @code{1000}. The
default value is set to @code{0}.
@item weight_Y
Set the frequency domain weight expression for the luma plane.
@item weight_U
Set the frequency domain weight expression for the 1st chroma plane.
@item weight_V
Set the frequency domain weight expression for the 2nd chroma plane.
The filter accepts the following variables:
@item X
@item Y
The coordinates of the current sample.
@item W
@item H
The width and height of the image.
@end table
@subsection Examples
@itemize
@item
High-pass:
@example
fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'
@end example
@item
Low-pass:
@example
fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'
@end example
@item
Sharpen:
@example
fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'
@end example
@end itemize
@section field
Extract a single field from an interlaced image using stride
@@ -5166,70 +4972,6 @@ framework.
It does not take parameters.
@section find_rect
Find a rectangular object
It accepts the following options:
@table @option
@item object
Filepath of the object image, needs to be in gray8.
@item threshold
Detection threshold, default is 0.5.
@item mipmaps
Number of mipmaps, default is 3.
@item xmin, ymin, xmax, ymax
Specifies the rectangle in which to search.
@end table
@subsection Examples
@itemize
@item
Generate a representative palette of a given video using @command{ffmpeg}:
@example
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
@end example
@end itemize
@section cover_rect
Cover a rectangular object
It accepts the following options:
@table @option
@item cover
Filepath of the optional cover image, needs to be in yuv420.
@item mode
Set covering mode.
It accepts the following values:
@table @samp
@item cover
cover it by the supplied image
@item blur
cover it by interpolating the surrounding pixels
@end table
Default value is @var{blur}.
@end table
@subsection Examples
@itemize
@item
Generate a representative palette of a given video using @command{ffmpeg}:
@example
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
@end example
@end itemize
@anchor{format}
@section format
@@ -6124,7 +5866,7 @@ full weight of 1.0 forever.
When this is not 0 then idet will use the specified number of frames to determine
if the interlaced flag is accurate, it will not count undetermined frames.
If the flag is found to be accurate it will be used without any further
computations, if it is found to be inaccurate it will be cleared without any
computations, if it is found to be inaccuarte it will be cleared without any
further computations. This allows inserting the idet filter as a low computational
method to clean up the interlaced flag
@end table
@@ -11937,33 +11679,6 @@ aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=
@end example
@end itemize
@section showwavespic
Convert input audio to a single video frame, representing the samples waves.
The filter accepts the following options:
@table @option
@item size, s
Specify the video size for the output. For the syntax of this option, check the
@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}.
Default value is @code{600x240}.
@item split_channels
Set if channels should be drawn separately or overlap. Default value is 0.
@end table
@subsection Examples
@itemize
@item
Extract a channel split representation of the wave form of a whole audio track
in a 1024x800 picture using @command{ffmpeg}:
@example
ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png
@end example
@end itemize
@section split, asplit
Split input into several identical outputs.

View File

@@ -37,8 +37,6 @@ Possible values:
@table @samp
@item ignidx
Ignore index.
@item fastseek
Enable fast, but inaccurate seeks for some formats.
@item genpts
Generate PTS.
@item nofillin
@@ -127,25 +125,6 @@ Consider all spec non compliancies as errors.
Consider things that a sane encoder should not do as an error.
@end table
@item max_interleave_delta @var{integer} (@emph{output})
Set maximum buffering duration for interleaving. The duration is
expressed in microseconds, and defaults to 1000000 (1 second).
To ensure all the streams are interleaved correctly, libavformat will
wait until it has at least one packet for each stream before actually
writing any packets to the output file. When some streams are
"sparse" (i.e. there are large gaps between successive packets), this
can result in excessive buffering.
This field specifies the maximum difference between the timestamps of the
first and the last packet in the muxing queue, above which libavformat
will output a packet regardless of whether it has queued a packet for all
the streams.
If set to 0, libavformat will continue buffering packets until it has
a packet for each stream, regardless of the maximum timestamp
difference between the buffered packets.
@item use_wallclock_as_timestamps @var{integer} (@emph{input})
Use wallclock as timestamps.

View File

@@ -179,8 +179,8 @@ included in compat/avisynth/, which allows the user to enable support
without needing to search for these headers themselves.
For Windows, supported AviSynth variants are
@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
For Linux and OS X, the supported AviSynth variant is
@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
@@ -222,7 +222,6 @@ library:
@tab Audio format used on the Nintendo Gamecube.
@item AFC @tab @tab X
@tab Audio format used on the Nintendo Gamecube.
@item APNG @tab X @tab X
@item ASF @tab X @tab X
@item AST @tab X @tab X
@tab Audio format used on the Nintendo Wii.
@@ -254,8 +253,6 @@ library:
@item Delphine Software International CIN @tab @tab X
@tab Multimedia format used by Delphine Software games.
@item Digital Speech Standard (DSS) @tab @tab X
@item Canopus HQ @tab @tab X
@item Canopus HQA @tab @tab X
@item Canopus HQX @tab @tab X
@item CD+G @tab @tab X
@tab Video format used by CD+G karaoke disks
@@ -469,7 +466,6 @@ library:
@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 TDSC @tab @tab X
@item Text files @tab @tab X
@item THP @tab @tab X
@tab Used on the Nintendo GameCube.
@@ -511,7 +507,6 @@ following image formats are supported:
@item Alias PIX @tab X @tab X
@tab Alias/Wavefront PIX image format
@item animated GIF @tab X @tab X
@item APNG @tab X @tab X
@item BMP @tab X @tab X
@tab Microsoft BMP image
@item BRender PIX @tab @tab X
@@ -910,7 +905,6 @@ following image formats are supported:
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
@item DCA (DTS Coherent Acoustics) @tab X @tab X
@tab supported extensions: XCh, XLL (partially)
@item DPCM id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2 and other computer games.
@item DPCM Interplay @tab @tab X

View File

@@ -75,7 +75,6 @@ git config --global core.autocrlf false
@end example
@anchor{Updating the source tree to the latest revision}
@section Updating the source tree to the latest revision
@example
@@ -331,7 +330,7 @@ git push
Will push the changes to the default remote (@var{origin}).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to @ref{Updating the source tree to the latest revision}.
out of sync. Refer to and to sync the local tree.
@example
git remote add <name> <url>

View File

@@ -150,81 +150,6 @@ $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
BSD video input device.
@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
uyvy422 or v210, 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 can be 2, 8 or 16.
@subsection Options
@table @option
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
Defaults to @option{false}.
@item bm_v210
If set to @samp{1}, video is captured in 10 bit v210 instead
of uyvy422. Not all Blackmagic devices support this option.
@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
@item
Capture video clip at 1080i50 10 bit:
@example
ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example
@item
Capture video clip at 720p50 with 32bit audio:
@example
ffmpeg -bm_audiodepth 32 -f decklink -i 'UltraStudio Mini Recorder@@14' -acodec copy -vcodec copy output.avi
@end example
@item
Capture video clip at 576i50 with 8 audio channels:
@example
ffmpeg -bm_channels 8 -f decklink -i 'UltraStudio Mini Recorder@@3' -acodec copy -vcodec copy output.avi
@end example
@end itemize
@section dshow
Windows DirectShow input device.
@@ -350,30 +275,6 @@ If set to @option{true}, before capture starts, popup a display
dialog to the end user, allowing them to manually
modify TV audio (like mono vs. stereo, Language A,B or C).
@item audio_device_load
Load an audio capture filter device from file instead of searching
it by name. It may load additional parameters too, if the filter
supports the serialization of its properties to.
To use this an audio capture source has to be specified, but it can
be anything even fake one.
@item audio_device_save
Save the currently used audio capture filter device and its
parameters (if the filter supports it) to a file.
If a file with the same name exists it will be overwritten.
@item video_device_load
Load a video capture filter device from file instead of searching
it by name. It may load additional parameters too, if the filter
supports the serialization of its properties to.
To use this a video capture source has to be specified, but it can
be anything even fake one.
@item video_device_save
Save the currently used video capture filter device and its
parameters (if the filter supports it) to a file.
If a file with the same name exists it will be overwritten.
@end table
@subsection Examples
@@ -1003,12 +904,6 @@ conversion into the real time clock.
Some usage examples of the video4linux2 device with @command{ffmpeg}
and @command{ffplay}:
@itemize
@item
List supported formats for a video4linux2 device:
@example
ffplay -f video4linux2 -list_formats all /dev/video0
@end example
@item
Grab and show the input of a video4linux2 device:
@example
@@ -1217,5 +1112,86 @@ The syntax is:
Set the grabbing region coordinates. They are expressed as offset from the top left
corner of the X11 window. The default value is 0.
@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
uyvy422 or v210, 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 can be 2, 8 or 16.
@subsection Options
@table @option
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
Defaults to @option{false}.
@item bm_v210
If set to @samp{1}, video is captured in 10 bit v210 instead
of uyvy422. Not all Blackmagic devices support this option.
@item bm_channels <CHANNELS>
Number of audio channels, can be 2, 8 or 16
@item bm_audiodepth <BITDEPTH>
Audio bit depth, can be 16 or 32.
@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
@item
Capture video clip at 1080i50 10 bit:
@example
ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example
@item
Capture video clip at 720p50 with 32bit audio:
@example
ffmpeg -bm_audiodepth 32 -f decklink -i 'UltraStudio Mini Recorder@@14' -acodec copy -vcodec copy output.avi
@end example
@item
Capture video clip at 576i50 with 8 audio channels:
@example
ffmpeg -bm_channels 8 -f decklink -i 'UltraStudio Mini Recorder@@3' -acodec copy -vcodec copy output.avi
@end example
@end itemize
@c man end INPUT DEVICES

View File

@@ -12,10 +12,10 @@ A file consists of a header and a number of metadata tags divided into sections,
each on its own line.
@item
The header is a @samp{;FFMETADATA} string, followed by a version number (now 1).
The header is a ';FFMETADATA' string, followed by a version number (now 1).
@item
Metadata tags are of the form @samp{key=value}
Metadata tags are of the form 'key=value'
@item
Immediately after header follows global metadata
@@ -26,30 +26,26 @@ metadata.
@item
A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
brackets (@samp{[}, @samp{]}) and ends with next section or end of file.
brackets ('[', ']') and ends with next section or end of file.
@item
At the beginning of a chapter section there may be an optional timebase to be
used for start/end values. It must be in form
@samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are
integers. If the timebase is missing then start/end times are assumed to
used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
den are integers. If the timebase is missing then start/end times are assumed to
be in milliseconds.
Next a chapter section must contain chapter start and end times in form
@samp{START=@var{num}}, @samp{END=@var{num}}, where @var{num} is a positive
integer.
'START=num', 'END=num', where num is a positive integer.
@item
Empty lines and lines starting with @samp{;} or @samp{#} are ignored.
Empty lines and lines starting with ';' or '#' are ignored.
@item
Metadata keys or values containing special characters (@samp{=}, @samp{;},
@samp{#}, @samp{\} and a newline) must be escaped with a backslash @samp{\}.
Metadata keys or values containing special characters ('=', ';', '#', '\' and a
newline) must be escaped with a backslash '\'.
@item
Note that whitespace in metadata (e.g. @samp{foo = bar}) is considered to be
a part of the tag (in the example above key is @samp{foo }, value is
@samp{ bar}).
Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
the tag (in the example above key is 'foo ', value is ' bar').
@end enumerate
A ffmetadata file might look like this:

View File

@@ -1030,12 +1030,6 @@ segments to write. If this is selected, the output segment name must
contain a @code{strftime} function template. Default value is
@code{0}.
@item break_non_keyframes @var{1|0}
If enabled, allow segments to start on frames other than keyframes. This
improves behavior on some players when the time between keyframes is
inconsistent, but may make things worse on others, and can cause some oddities
during seeking. Defaults to @code{0}.
@item reset_timestamps @var{1|0}
Reset timestamps at the begin of each segment, so that each segment
will start with near-zero timestamps. It is meant to ease the playback
@@ -1216,17 +1210,7 @@ is the @option{global_header} flag.
WebM DASH Manifest muxer.
This muxer implements the WebM DASH Manifest specification to generate the DASH
manifest XML. It also supports manifest generation for DASH live streams.
For more information see:
@itemize @bullet
@item
WebM DASH Specification: @url{https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification}
@item
ISO DASH Specification: @url{http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip}
@end itemize
This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
@subsection Options
@@ -1237,32 +1221,6 @@ This muxer supports the following options:
This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
audio and video streams. Any number of adaptation sets can be added using this option.
@item live
Set this to 1 to create a live stream DASH Manifest. Default: 0.
@item chunk_start_index
Start index of the first chunk. This will go in the @samp{startNumber} attribute
of the @samp{SegmentTemplate} element in the manifest. Default: 0.
@item chunk_duration_ms
Duration of each chunk in milliseconds. This will go in the @samp{duration}
attribute of the @samp{SegmentTemplate} element in the manifest. Default: 1000.
@item utc_timing_url
URL of the page that will return the UTC timestamp in ISO format. This will go
in the @samp{value} attribute of the @samp{UTCTiming} element in the manifest.
Default: None.
@item time_shift_buffer_depth
Smallest time (in seconds) shifting buffer for which any Representation is
guaranteed to be available. This will go in the @samp{timeShiftBufferDepth}
attribute of the @samp{MPD} element. Default: 60.
@item minimum_update_period
Minimum update period (in seconds) of the manifest. This will go in the
@samp{minimumUpdatePeriod} attribute of the @samp{MPD} element. Default: 0.
@end table
@subsection Example
@@ -1278,47 +1236,4 @@ ffmpeg -f webm_dash_manifest -i video1.webm \
manifest.xml
@end example
@section webm_chunk
WebM Live Chunk Muxer.
This muxer writes out WebM headers and chunks as separate files which can be
consumed by clients that support WebM Live streams via DASH.
@subsection Options
This muxer supports the following options:
@table @option
@item chunk_start_index
Index of the first chunk (defaults to 0).
@item header
Filename of the header where the initialization data will be written.
@item audio_chunk_duration
Duration of each audio chunk in milliseconds (defaults to 5000).
@end table
@subsection Example
@example
ffmpeg -f v4l2 -i /dev/video0 \
-f alsa -i hw:0 \
-map 0:0 \
-c:v libvpx-vp9 \
-s 640x360 -keyint_min 30 -g 30 \
-f webm_chunk \
-header webm_live_video_360.hdr \
-chunk_start_index 1 \
webm_live_video_360_%d.chk \
-map 1:0 \
-c:a libvorbis \
-b:a 128k \
-f webm_chunk \
-header webm_live_audio_128.hdr \
-chunk_start_index 1 \
-audio_chunk_duration 1000 \
webm_live_audio_128_%d.chk
@end example
@c man end MUXERS

View File

@@ -277,41 +277,6 @@ Set initial byte offset.
@item end_offset
Try to limit the request to bytes preceding this offset.
@item method
When used as a client option it sets the HTTP method for the request.
When used as a server option it sets the HTTP method that is going to be
expected from the client(s).
If the expected and the received HTTP method do not match the client will
be given a Bad Request response.
When unset the HTTP method is not checked for now. This will be replaced by
autodetection in the future.
@item listen
If set to 1 enables experimental HTTP server. This can be used to send data when
used as an output option, or read data from a client with HTTP POST when used as
an input option.
@example
# Server side (sending):
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
# Client side (receiving):
ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
# Client can also be done with wget:
wget http://@var{server}:@var{port} -O somefile.ogg
# Server side (receiving):
ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
# Client side (sending):
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
# Client can also be done with wget:
wget --post-file=somefile.ogg http://@var{server}:@var{port}
@end example
@end table
@subsection HTTP Cookies
@@ -1090,9 +1055,7 @@ subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
@end example
Play an AVI file directly from a TAR archive:
@example
subfile,,start,183241728,end,366490624,,:archive.tar
@end example
@section tcp

View File

@@ -166,7 +166,7 @@ INF: while(<$inf>) {
if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) {
$skipping = pop @skstack;
next;
} elsif ($ended =~ /^(?:example|smallexample|verbatim|display)$/) {
} elsif ($ended =~ /^(?:example|smallexample|display)$/) {
$shift = "";
$_ = ""; # need a paragraph break
} elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
@@ -290,7 +290,7 @@ INF: while(<$inf>) {
$_ = "\n=over 4\n";
};
/^\@((?:small)?example|verbatim|display)/ and do {
/^\@((?:small)?example|display)/ and do {
push @endwstack, $endw;
$endw = $1;
$shift = "\t";

View File

@@ -12,17 +12,17 @@ explicitly specified. The following rules are applied:
@itemize
@item
@samp{'} and @samp{\} are special characters (respectively used for
@code{'} and @code{\} are special characters (respectively used for
quoting and escaping). In addition to them, there might be other
special characters depending on the specific syntax where the escaping
and quoting are employed.
@item
A special character is escaped by prefixing it with a @samp{\}.
A special character is escaped by prefixing it with a '\'.
@item
All characters enclosed between @samp{''} are included literally in the
parsed string. The quote character @samp{'} itself cannot be quoted,
All characters enclosed between '' are included literally in the
parsed string. The quote character @code{'} itself cannot be quoted,
so you may need to close the quote and escape it.
@item
@@ -71,7 +71,7 @@ Escaping and quoting can be mixed together:
@end example
@item
To include a literal @samp{\} you can use either escaping or quoting:
To include a literal @code{\} you can use either escaping or quoting:
@example
'c:\foo' can be written as c:\\foo
@end example

285
ffmpeg.c
View File

@@ -60,7 +60,6 @@
#include "libavutil/bprint.h"
#include "libavutil/time.h"
#include "libavutil/threadmessage.h"
#include "libavcodec/mathops.h"
#include "libavformat/os_support.h"
# include "libavfilter/avcodec.h"
@@ -352,6 +351,7 @@ 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). */
@@ -456,10 +456,7 @@ static void ffmpeg_cleanup(int ret)
/* close files */
for (i = 0; i < nb_output_files; i++) {
OutputFile *of = output_files[i];
AVFormatContext *s;
if (!of)
continue;
s = of->ctx;
AVFormatContext *s = of->ctx;
if (s && s->oformat && !(s->oformat->flags & AVFMT_NOFILE))
avio_closep(&s->pb);
avformat_free_context(s);
@@ -469,12 +466,7 @@ 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);
@@ -595,7 +587,7 @@ static void close_all_output_streams(OutputStream *ost, OSTFinished this_stream,
static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
{
AVBitStreamFilterContext *bsfc = ost->bitstream_filters;
AVCodecContext *avctx = ost->encoding_needed ? ost->enc_ctx : ost->st->codec;
AVCodecContext *avctx = ost->st->codec;
int ret;
if (!ost->st->codec->extradata_size && ost->enc_ctx->extradata_size) {
@@ -658,7 +650,6 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
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");
@@ -923,93 +914,80 @@ static void do_video_out(AVFormatContext *s,
duration = lrintf(av_frame_get_pkt_duration(next_picture) * av_q2d(ist->st->time_base) / av_q2d(enc->time_base));
}
if (!next_picture) {
//end, flushing
nb0_frames = nb_frames = mid_pred(ost->last_nb0_frames[0],
ost->last_nb0_frames[1],
ost->last_nb0_frames[2]);
} else {
delta0 = sync_ipts - ost->sync_opts;
delta = delta0 + duration;
delta0 = sync_ipts - ost->sync_opts;
delta = delta0 + duration;
/* by default, we output a single frame */
nb0_frames = 0;
nb_frames = 1;
/* by default, we output a single frame */
nb0_frames = 0;
nb_frames = 1;
format_video_sync = video_sync_method;
if (format_video_sync == VSYNC_AUTO) {
if(!strcmp(s->oformat->name, "avi")) {
format_video_sync = VSYNC_VFR;
} else
format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR;
if ( ist
&& format_video_sync == VSYNC_CFR
&& input_files[ist->file_index]->ctx->nb_streams == 1
&& input_files[ist->file_index]->input_ts_offset == 0) {
format_video_sync = VSYNC_VSCFR;
}
if (format_video_sync == VSYNC_CFR && copy_ts) {
format_video_sync = VSYNC_VSCFR;
}
format_video_sync = video_sync_method;
if (format_video_sync == VSYNC_AUTO) {
if(!strcmp(s->oformat->name, "avi")) {
format_video_sync = VSYNC_VFR;
} else
format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR;
if ( ist
&& format_video_sync == VSYNC_CFR
&& input_files[ist->file_index]->ctx->nb_streams == 1
&& input_files[ist->file_index]->input_ts_offset == 0) {
format_video_sync = VSYNC_VSCFR;
}
if (delta0 < 0 &&
delta > 0 &&
format_video_sync != VSYNC_PASSTHROUGH &&
format_video_sync != VSYNC_DROP) {
double cor = FFMIN(-delta0, duration);
if (delta0 < -0.6) {
av_log(NULL, AV_LOG_WARNING, "Past duration %f too large\n", -delta0);
} else
av_log(NULL, AV_LOG_DEBUG, "Cliping frame in rate conversion by %f\n", -delta0);
sync_ipts += cor;
duration -= cor;
delta0 += cor;
if (format_video_sync == VSYNC_CFR && copy_ts) {
format_video_sync = VSYNC_VSCFR;
}
}
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 (frame_drop_threshold && delta < frame_drop_threshold && ost->frame_number) {
nb_frames = 0;
} else if (delta < -1.1)
nb_frames = 0;
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)
nb_frames = 0;
else if (delta > 0.6)
ost->sync_opts = lrint(sync_ipts);
break;
case VSYNC_DROP:
case VSYNC_PASSTHROUGH:
if (delta0 < 0 &&
delta > 0 &&
format_video_sync != VSYNC_PASSTHROUGH &&
format_video_sync != VSYNC_DROP) {
double cor = FFMIN(-delta0, duration);
if (delta0 < -0.6) {
av_log(NULL, AV_LOG_WARNING, "Past duration %f too large\n", -delta0);
} else
av_log(NULL, AV_LOG_DEBUG, "Cliping frame in rate conversion by %f\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);
break;
default:
av_assert0(0);
}
case VSYNC_CFR:
// FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
if (frame_drop_threshold && delta < frame_drop_threshold && ost->frame_number) {
nb_frames = 0;
} else if (delta < -1.1)
nb_frames = 0;
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)
nb_frames = 0;
else if (delta > 0.6)
ost->sync_opts = lrint(sync_ipts);
break;
case VSYNC_DROP:
case VSYNC_PASSTHROUGH:
ost->sync_opts = lrint(sync_ipts);
break;
default:
av_assert0(0);
}
nb_frames = FFMIN(nb_frames, ost->max_frames - ost->frame_number);
nb0_frames = FFMIN(nb0_frames, nb_frames);
memmove(ost->last_nb0_frames + 1,
ost->last_nb0_frames,
sizeof(ost->last_nb0_frames[0]) * (FF_ARRAY_ELEMS(ost->last_nb0_frames) - 1));
ost->last_nb0_frames[0] = nb0_frames;
if (nb0_frames == 0 && ost->last_droped) {
nb_frames_drop++;
av_log(NULL, AV_LOG_VERBOSE,
@@ -1025,7 +1003,7 @@ static void do_video_out(AVFormatContext *s,
nb_frames_dup += nb_frames - (nb0_frames && ost->last_droped) - (nb_frames > nb0_frames);
av_log(NULL, AV_LOG_VERBOSE, "*** %d dup!\n", nb_frames - 1);
}
ost->last_droped = nb_frames == nb0_frames && next_picture;
ost->last_droped = nb_frames == nb0_frames;
/* duplicates frame if needed */
for (i = 0; i < nb_frames; i++) {
@@ -1039,9 +1017,6 @@ static void do_video_out(AVFormatContext *s,
} else
in_picture = next_picture;
if (!in_picture)
return;
in_picture->pts = ost->sync_opts;
#if 1
@@ -1113,10 +1088,6 @@ static void do_video_out(AVFormatContext *s,
}
ost->forced_keyframes_expr_const_values[FKF_N] += 1;
} else if ( ost->forced_keyframes
&& !strncmp(ost->forced_keyframes, "source", 6)
&& in_picture->key_frame==1) {
forced_keyframe = 1;
}
if (forced_keyframe) {
@@ -1185,10 +1156,7 @@ static void do_video_out(AVFormatContext *s,
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);
av_frame_ref(ost->last_frame, next_picture);
}
static double psnr(double d)
@@ -1251,7 +1219,7 @@ static void finish_output_stream(OutputStream *ost)
*
* @return 0 for success, <0 for severe errors
*/
static int reap_filters(int flush)
static int reap_filters(void)
{
AVFrame *filtered_frame = NULL;
int i;
@@ -1281,9 +1249,6 @@ static int reap_filters(int flush)
if (ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
av_log(NULL, AV_LOG_WARNING,
"Error in av_buffersink_get_frame_flags(): %s\n", av_err2str(ret));
} else if (flush && ret == AVERROR_EOF) {
if (filter->inputs[0]->type == AVMEDIA_TYPE_VIDEO)
do_video_out(of->ctx, ost, NULL, AV_NOPTS_VALUE);
}
break;
}
@@ -1353,7 +1318,6 @@ static void print_final_stats(int64_t total_size)
uint64_t data_size = 0;
float percent = -1.0;
int i, j;
int pass1_used = 1;
for (i = 0; i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
@@ -1365,9 +1329,6 @@ static void print_final_stats(int64_t total_size)
}
extra_size += ost->enc_ctx->extradata_size;
data_size += ost->data_size;
if ( (ost->enc_ctx->flags & (CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2))
!= CODEC_FLAG_PASS1)
pass1_used = 0;
}
if (data_size && total_size>0 && total_size >= data_size)
@@ -1454,12 +1415,7 @@ static void print_final_stats(int64_t total_size)
total_packets, total_size);
}
if(video_size + data_size + audio_size + subtitle_size + extra_size == 0){
av_log(NULL, AV_LOG_WARNING, "Output file is empty, nothing was encoded ");
if (pass1_used) {
av_log(NULL, AV_LOG_WARNING, "\n");
} else {
av_log(NULL, AV_LOG_WARNING, "(check -ss / -t / -frames parameters if used)\n");
}
av_log(NULL, AV_LOG_WARNING, "Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)\n");
}
}
@@ -1874,12 +1830,9 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
ret = AVERROR_INVALIDDATA;
}
if (*got_output || ret<0)
if (*got_output || ret<0 || pkt->size)
decode_error_stat[ret<0] ++;
if (ret < 0 && exit_on_error)
exit_program(1);
if (!*got_output || ret < 0) {
if (!pkt->size) {
for (i = 0; i < ist->nb_filters; i++)
@@ -2022,12 +1975,9 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
);
}
if (*got_output || ret<0)
if (*got_output || ret<0 || pkt->size)
decode_error_stat[ret<0] ++;
if (ret < 0 && exit_on_error)
exit_program(1);
if (*got_output && ret >= 0) {
if (ist->dec_ctx->width != decoded_frame->width ||
ist->dec_ctx->height != decoded_frame->height ||
@@ -2143,12 +2093,9 @@ static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output)
int i, ret = avcodec_decode_subtitle2(ist->dec_ctx,
&subtitle, got_output, pkt);
if (*got_output || ret<0)
if (*got_output || ret<0 || pkt->size)
decode_error_stat[ret<0] ++;
if (ret < 0 && exit_on_error)
exit_program(1);
if (ret < 0 || !*got_output) {
if (!pkt->size)
sub2video_flush(ist);
@@ -2413,7 +2360,7 @@ static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat
"%s hwaccel requested for input stream #%d:%d, "
"but cannot be initialized.\n", hwaccel->name,
ist->file_index, ist->st->index);
return AV_PIX_FMT_NONE;
exit_program(1);
}
continue;
}
@@ -2672,7 +2619,7 @@ static int transcode_init(void)
if (ost->attachment_filename)
continue;
enc_ctx = ost->stream_copy ? ost->st->codec : ost->enc_ctx;
enc_ctx = ost->enc_ctx;
if (ist) {
dec_ctx = ist->dec_ctx;
@@ -2719,13 +2666,11 @@ 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;
@@ -2786,13 +2731,9 @@ static int transcode_init(void)
if (!ost->st->side_data)
return AVERROR(ENOMEM);
ost->st->nb_side_data = 0;
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[ost->st->nb_side_data];
if (ost->rotate_overridden && sd_src->type == AV_PKT_DATA_DISPLAYMATRIX)
continue;
AVPacketSideData *sd_dst = &ost->st->side_data[j];
sd_dst->data = av_malloc(sd_src->size);
if (!sd_dst->data)
@@ -2851,7 +2792,6 @@ static int transcode_init(void)
enc_ctx->width = dec_ctx->width;
enc_ctx->height = dec_ctx->height;
break;
case AVMEDIA_TYPE_UNKNOWN:
case AVMEDIA_TYPE_DATA:
case AVMEDIA_TYPE_ATTACHMENT:
break;
@@ -2887,7 +2827,7 @@ static int transcode_init(void)
}
if (enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
if (!ost->frame_rate.num)
if (ost->filter && !ost->frame_rate.num)
ost->frame_rate = av_buffersink_get_frame_rate(ost->filter->filter);
if (ist && !ost->frame_rate.num)
ost->frame_rate = ist->framerate;
@@ -2924,7 +2864,7 @@ static int transcode_init(void)
break;
case AVMEDIA_TYPE_VIDEO:
enc_ctx->time_base = av_inv_q(ost->frame_rate);
if (!(enc_ctx->time_base.num && enc_ctx->time_base.den))
if (ost->filter && !(enc_ctx->time_base.num && enc_ctx->time_base.den))
enc_ctx->time_base = ost->filter->filter->inputs[0]->time_base;
if ( av_q2d(enc_ctx->time_base) < 0.001 && video_sync_method != VSYNC_PASSTHROUGH
&& (video_sync_method == VSYNC_CFR || video_sync_method == VSYNC_VSCFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){
@@ -2980,10 +2920,7 @@ static int transcode_init(void)
ost->forced_keyframes_expr_const_values[FKF_N_FORCED] = 0;
ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N] = NAN;
ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T] = NAN;
// Don't parse the 'forced_keyframes' in case of 'keep-source-keyframes',
// parse it only for static kf timings
} else if(strncmp(ost->forced_keyframes, "source", 6)) {
} else {
parse_forced_key_frames(ost->forced_keyframes, ost, ost->enc_ctx);
}
}
@@ -3106,17 +3043,6 @@ static int transcode_init(void)
if (ost->enc_ctx->bit_rate && ost->enc_ctx->bit_rate < 1000)
av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
" It takes bits/s as argument, not kbits/s\n");
ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL,
"Error initializing the output stream codec context.\n");
exit_program(1);
}
// copy timebase while removing common factors
ost->st->time_base = av_add_q(ost->enc_ctx->time_base, (AVRational){0, 1});
ost->st->codec->codec= ost->enc_ctx->codec;
} else {
ret = av_opt_set_dict(ost->enc_ctx, &ost->encoder_opts);
if (ret < 0) {
@@ -3124,9 +3050,18 @@ static int transcode_init(void)
"Error setting up codec context options.\n");
return ret;
}
// copy timebase while removing common factors
ost->st->time_base = av_add_q(ost->st->codec->time_base, (AVRational){0, 1});
}
ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL,
"Error initializing the output stream codec context.\n");
exit_program(1);
}
ost->st->codec->codec= ost->enc_ctx->codec;
// copy timebase while removing common factors
ost->st->time_base = av_add_q(ost->enc_ctx->time_base, (AVRational){0, 1});
}
/* init input streams */
@@ -3236,22 +3171,17 @@ static int transcode_init(void)
const char *in_codec_name = "?";
const char *encoder_name = "?";
const char *out_codec_name = "?";
const AVCodecDescriptor *desc;
if (in_codec) {
decoder_name = in_codec->name;
desc = avcodec_descriptor_get(in_codec->id);
if (desc)
in_codec_name = desc->name;
in_codec_name = avcodec_descriptor_get(in_codec->id)->name;
if (!strcmp(decoder_name, in_codec_name))
decoder_name = "native";
}
if (out_codec) {
encoder_name = out_codec->name;
desc = avcodec_descriptor_get(out_codec->id);
if (desc)
out_codec_name = desc->name;
out_codec_name = avcodec_descriptor_get(out_codec->id)->name;
if (!strcmp(encoder_name, out_codec_name))
encoder_name = "native";
}
@@ -3378,8 +3308,6 @@ static int check_keyboard_interaction(int64_t cur_time)
ret = AVERROR_PATCHWELCOME;
} else {
ret = avfilter_graph_queue_command(fg->graph, target, command, arg, 0, time);
if (ret < 0)
fprintf(stderr, "Queing command failed with error %s\n", av_err2str(ret));
}
}
}
@@ -3690,8 +3618,6 @@ static int process_input(int file_index)
if (av_packet_get_side_data(&pkt, src_sd->type, NULL))
continue;
if (ist->autorotate && src_sd->type == AV_PKT_DATA_DISPLAYMATRIX)
continue;
dst_data = av_packet_new_side_data(&pkt, src_sd->type, src_sd->size);
if (!dst_data)
@@ -3810,10 +3736,10 @@ static int transcode_from_filter(FilterGraph *graph, InputStream **best_ist)
*best_ist = NULL;
ret = avfilter_graph_request_oldest(graph->graph);
if (ret >= 0)
return reap_filters(0);
return reap_filters();
if (ret == AVERROR_EOF) {
ret = reap_filters(1);
ret = reap_filters();
for (i = 0; i < graph->nb_outputs; i++)
close_output_stream(graph->outputs[i]->ost);
return ret;
@@ -3879,11 +3805,10 @@ static int transcode_step(void)
ost->unavailable = 1;
return 0;
}
if (ret < 0)
return ret == AVERROR_EOF ? 0 : ret;
return reap_filters(0);
return reap_filters();
}
/*
@@ -3928,15 +3853,11 @@ static int transcode(void)
ret = transcode_step();
if (ret < 0) {
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)) {
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
continue;
} else {
char errbuf[128];
av_strerror(ret, errbuf, sizeof(errbuf));
av_log(NULL, AV_LOG_ERROR, "Error while filtering: %s\n", errbuf);
break;
}
av_log(NULL, AV_LOG_ERROR, "Error while filtering.\n");
break;
}
/* dump report by using the output first video and audio streams */

View File

@@ -92,7 +92,6 @@ typedef struct OptionsContext {
/* input/output options */
int64_t start_time;
int seek_timestamp;
const char *format;
SpecifierOpt *codec_names;
@@ -122,8 +121,6 @@ typedef struct OptionsContext {
int nb_hwaccels;
SpecifierOpt *hwaccel_devices;
int nb_hwaccel_devices;
SpecifierOpt *autorotate;
int nb_autorotate;
/* output options */
StreamMap *stream_maps;
@@ -278,7 +275,6 @@ typedef struct InputStream {
int top_field_first;
int guess_layout_max;
int autorotate;
int resample_height;
int resample_width;
int resample_pix_fmt;
@@ -343,7 +339,6 @@ typedef struct InputFile {
int64_t ts_offset;
int64_t last_ts;
int64_t start_time; /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
int seek_timestamp;
int64_t recording_time;
int nb_streams; /* number of stream that ffmpeg is aware of; may be different
from ctx.nb_streams if new streams appear during av_read_frame() */
@@ -399,13 +394,11 @@ typedef struct OutputStream {
AVFrame *filtered_frame;
AVFrame *last_frame;
int last_droped;
int last_nb0_frames[3];
/* video only */
AVRational frame_rate;
int force_fps;
int top_field_first;
int rotate_overridden;
AVRational frame_aspect_ratio;

View File

@@ -364,7 +364,7 @@ static int dxva2_alloc(AVCodecContext *s)
d3dpp.SwapEffect = D3DSWAPEFFECT_DISCARD;
d3dpp.Flags = D3DPRESENTFLAG_VIDEO;
hr = IDirect3D9_CreateDevice(ctx->d3d9, adapter, D3DDEVTYPE_HAL, GetDesktopWindow(),
hr = IDirect3D9_CreateDevice(ctx->d3d9, adapter, D3DDEVTYPE_HAL, GetShellWindow(),
D3DCREATE_SOFTWARE_VERTEXPROCESSING | D3DCREATE_MULTITHREADED | D3DCREATE_FPU_PRESERVE,
&d3dpp, &ctx->d3d9device);
if (FAILED(hr)) {

View File

@@ -31,7 +31,6 @@
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
#include "libavutil/channel_layout.h"
#include "libavutil/display.h"
#include "libavutil/opt.h"
#include "libavutil/pixdesc.h"
#include "libavutil/pixfmt.h"
@@ -340,28 +339,6 @@ static int insert_trim(int64_t start_time, int64_t duration,
return 0;
}
static int insert_filter(AVFilterContext **last_filter, int *pad_idx,
const char *filter_name, const char *args)
{
AVFilterGraph *graph = (*last_filter)->graph;
AVFilterContext *ctx;
int ret;
ret = avfilter_graph_create_filter(&ctx,
avfilter_get_by_name(filter_name),
filter_name, args, NULL, graph);
if (ret < 0)
return ret;
ret = avfilter_link(*last_filter, *pad_idx, ctx, 0);
if (ret < 0)
return ret;
*last_filter = ctx;
*pad_idx = 0;
return 0;
}
static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
{
char *pix_fmts;
@@ -699,27 +676,6 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
return ret;
last_filter = ifilter->filter;
if (ist->autorotate) {
double theta = get_rotation(ist->st);
if (fabs(theta - 90) < 1.0) {
ret = insert_filter(&last_filter, &pad_idx, "transpose", "clock");
} else if (fabs(theta - 180) < 1.0) {
ret = insert_filter(&last_filter, &pad_idx, "hflip", NULL);
if (ret < 0)
return ret;
ret = insert_filter(&last_filter, &pad_idx, "vflip", NULL);
} else if (fabs(theta - 270) < 1.0) {
ret = insert_filter(&last_filter, &pad_idx, "transpose", "cclock");
} else if (fabs(theta) > 1.0) {
char rotate_buf[64];
snprintf(rotate_buf, sizeof(rotate_buf), "%f*PI/180", theta);
ret = insert_filter(&last_filter, &pad_idx, "rotate", rotate_buf);
}
if (ret < 0)
return ret;
}
if (ist->framerate.num) {
AVFilterContext *setpts;

View File

@@ -110,9 +110,6 @@ static int no_file_overwrite = 0;
static int do_psnr = 0;
static int input_sync;
static int override_ffserver = 0;
static int input_stream_potentially_available = 0;
static int ignore_unknown_streams = 0;
static int copy_unknown_streams = 0;
static void uninit_options(OptionsContext *o)
{
@@ -604,9 +601,6 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
ist->ts_scale = 1.0;
MATCH_PER_STREAM_OPT(ts_scale, dbl, ist->ts_scale, ic, st);
ist->autorotate = 1;
MATCH_PER_STREAM_OPT(autorotate, i, ist->autorotate, ic, st);
MATCH_PER_STREAM_OPT(codec_tags, str, codec_tag, ic, st);
if (codec_tag) {
uint32_t tag = strtol(codec_tag, &next, 0);
@@ -924,25 +918,12 @@ static int open_input_file(OptionsContext *o, const char *filename)
timestamp = (o->start_time == AV_NOPTS_VALUE) ? 0 : o->start_time;
/* add the stream start time */
if (!o->seek_timestamp && ic->start_time != AV_NOPTS_VALUE)
if (ic->start_time != AV_NOPTS_VALUE)
timestamp += ic->start_time;
/* if seeking requested, we execute it */
if (o->start_time != AV_NOPTS_VALUE) {
int64_t seek_timestamp = timestamp;
if (!(ic->iformat->flags & AVFMT_SEEK_TO_PTS)) {
int dts_heuristic = 0;
for (i=0; i<ic->nb_streams; i++) {
AVCodecContext *avctx = ic->streams[i]->codec;
if (avctx->has_b_frames)
dts_heuristic = 1;
}
if (dts_heuristic) {
seek_timestamp -= 3*AV_TIME_BASE / 23;
}
}
ret = avformat_seek_file(ic, -1, INT64_MIN, seek_timestamp, seek_timestamp, 0);
ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, timestamp, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_WARNING, "%s: could not seek to position %0.3f\n",
filename, (double)timestamp / AV_TIME_BASE);
@@ -1027,8 +1008,6 @@ static int open_input_file(OptionsContext *o, const char *filename)
av_dict_free(&opts[i]);
av_freep(&opts);
input_stream_potentially_available = 1;
return 0;
}
@@ -1203,7 +1182,6 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
uint32_t tag = strtol(codec_tag, &next, 0);
if (*next)
tag = AV_RL32(codec_tag);
ost->st->codec->codec_tag =
ost->enc_ctx->codec_tag = tag;
}
@@ -1568,19 +1546,6 @@ static OutputStream *new_data_stream(OptionsContext *o, AVFormatContext *oc, int
return ost;
}
static OutputStream *new_unknown_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
{
OutputStream *ost;
ost = new_output_stream(o, oc, AVMEDIA_TYPE_UNKNOWN, source_index);
if (!ost->stream_copy) {
av_log(NULL, AV_LOG_FATAL, "Unknown stream encoding not supported yet (only streamcopy)\n");
exit_program(1);
}
return ost;
}
static OutputStream *new_attachment_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
{
OutputStream *ost = new_output_stream(o, oc, AVMEDIA_TYPE_ATTACHMENT, source_index);
@@ -2033,22 +1998,10 @@ loop_end:
case AVMEDIA_TYPE_SUBTITLE: ost = new_subtitle_stream (o, oc, src_idx); break;
case AVMEDIA_TYPE_DATA: ost = new_data_stream (o, oc, src_idx); break;
case AVMEDIA_TYPE_ATTACHMENT: ost = new_attachment_stream(o, oc, src_idx); break;
case AVMEDIA_TYPE_UNKNOWN:
if (copy_unknown_streams) {
ost = new_unknown_stream (o, oc, src_idx);
break;
}
default:
av_log(NULL, ignore_unknown_streams ? AV_LOG_WARNING : AV_LOG_FATAL,
"Cannot map stream #%d:%d - unsupported type.\n",
av_log(NULL, AV_LOG_FATAL, "Cannot map stream #%d:%d - unsupported type.\n",
map->file_index, map->stream_index);
if (!ignore_unknown_streams) {
av_log(NULL, AV_LOG_FATAL,
"If you want unsupported types ignored instead "
"of failing, please use the -ignore_unknown option\n"
"If you want them copied, please use -copy_unknown\n");
exit_program(1);
}
exit_program(1);
}
}
}
@@ -2082,8 +2035,8 @@ loop_end:
ost->stream_copy = 0;
ost->attachment_filename = o->attachments[i];
ost->finished = 1;
ost->st->codec->extradata = attachment;
ost->st->codec->extradata_size = len;
ost->enc_ctx->extradata = attachment;
ost->enc_ctx->extradata_size = len;
p = strrchr(o->attachments[i], '/');
av_dict_set(&ost->st->metadata, "filename", (p && *p) ? p + 1 : o->attachments[i], AV_DICT_DONT_OVERWRITE);
@@ -2151,12 +2104,6 @@ loop_end:
}
}
if (!(oc->oformat->flags & AVFMT_NOSTREAMS) && !input_stream_potentially_available) {
av_log(NULL, AV_LOG_ERROR,
"No input streams but output needs an input stream\n");
exit_program(1);
}
if (!(oc->oformat->flags & AVFMT_NOFILE)) {
/* test if it already exists to avoid losing precious files */
assert_file_overwrite(filename);
@@ -2225,11 +2172,8 @@ loop_end:
continue;
ist = input_streams[output_streams[i]->source_index];
av_dict_copy(&output_streams[i]->st->metadata, ist->st->metadata, AV_DICT_DONT_OVERWRITE);
if (!output_streams[i]->stream_copy) {
if (!output_streams[i]->stream_copy)
av_dict_set(&output_streams[i]->st->metadata, "encoder", NULL, 0);
if (ist->autorotate)
av_dict_set(&output_streams[i]->st->metadata, "rotate", NULL, 0);
}
}
/* process manually set metadata */
@@ -2250,12 +2194,8 @@ loop_end:
parse_meta_type(o->metadata[i].specifier, &type, &index, &stream_spec);
if (type == 's') {
for (j = 0; j < oc->nb_streams; j++) {
ost = output_streams[nb_output_streams - oc->nb_streams + j];
if ((ret = check_stream_specifier(oc, oc->streams[j], stream_spec)) > 0) {
av_dict_set(&oc->streams[j]->metadata, o->metadata[i].u.str, *val ? val : NULL, 0);
if (!strcmp(o->metadata[i].u.str, "rotate")) {
ost->rotate_overridden = 1;
}
} else if (ret < 0)
exit_program(1);
}
@@ -2342,9 +2282,9 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
opt_default(NULL, "g", norm == PAL ? "15" : "18");
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;
opt_default(NULL, "maxrate", "1150000");
opt_default(NULL, "minrate", "1150000");
opt_default(NULL, "bufsize", "327680"); // 40*1024*8;
opt_default(NULL, "b:a", "224000");
parse_option(o, "ar", "44100", options);
@@ -2371,9 +2311,9 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
opt_default(NULL, "g", norm == PAL ? "15" : "18");
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, "maxrate", "2516000");
opt_default(NULL, "minrate", "0"); // 1145000;
opt_default(NULL, "bufsize", "1835008"); // 224*1024*8;
opt_default(NULL, "scan_offset", "1");
opt_default(NULL, "b:a", "224000");
@@ -2393,9 +2333,9 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
opt_default(NULL, "g", norm == PAL ? "15" : "18");
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;
opt_default(NULL, "maxrate", "9000000");
opt_default(NULL, "minrate", "0"); // 1500000;
opt_default(NULL, "bufsize", "1835008"); // 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
@@ -2439,9 +2379,6 @@ 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);
@@ -2671,9 +2608,6 @@ static int opt_filter_complex(void *optctx, const char *opt, const char *arg)
filtergraphs[nb_filtergraphs - 1]->graph_desc = av_strdup(arg);
if (!filtergraphs[nb_filtergraphs - 1]->graph_desc)
return AVERROR(ENOMEM);
input_stream_potentially_available = 1;
return 0;
}
@@ -2688,9 +2622,6 @@ static int opt_filter_complex_script(void *optctx, const char *opt, const char *
return AVERROR(ENOMEM);
filtergraphs[nb_filtergraphs - 1]->index = nb_filtergraphs - 1;
filtergraphs[nb_filtergraphs - 1]->graph_desc = graph_desc;
input_stream_potentially_available = 1;
return 0;
}
@@ -2886,10 +2817,6 @@ const OptionDef options[] = {
"overwrite output files" },
{ "n", OPT_BOOL, { &no_file_overwrite },
"never overwrite output files" },
{ "ignore_unknown", OPT_BOOL, { &ignore_unknown_streams },
"Ignore unknown stream types" },
{ "copy_unknown", OPT_BOOL | OPT_EXPERT, { &copy_unknown_streams },
"Copy unknown stream types" },
{ "c", HAS_ARG | OPT_STRING | OPT_SPEC |
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(codec_names) },
"codec name", "codec" },
@@ -2923,9 +2850,6 @@ const OptionDef options[] = {
{ "ss", HAS_ARG | OPT_TIME | OPT_OFFSET |
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(start_time) },
"set the start time offset", "time_off" },
{ "seek_timestamp", HAS_ARG | OPT_INT | OPT_OFFSET |
OPT_INPUT, { .off = OFFSET(seek_timestamp) },
"enable/disable seeking by timestamp with -ss" },
{ "accurate_seek", OPT_BOOL | OPT_OFFSET | OPT_EXPERT |
OPT_INPUT, { .off = OFFSET(accurate_seek) },
"enable/disable accurate seeking with -ss" },
@@ -2935,7 +2859,7 @@ 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" },
@@ -3101,7 +3025,7 @@ const OptionDef options[] = {
OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(top_field_first) },
"top=1/bottom=0/auto=-1 field first", "" },
{ "vtag", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_PERFILE |
OPT_INPUT | OPT_OUTPUT, { .func_arg = opt_old2new },
OPT_OUTPUT, { .func_arg = opt_old2new },
"force video tag/fourcc", "fourcc/tag" },
{ "qphist", OPT_VIDEO | OPT_BOOL | OPT_EXPERT , { &qp_hist },
"show QP histogram" },
@@ -3127,9 +3051,6 @@ const OptionDef options[] = {
#if HAVE_VDPAU_X11
{ "vdpau_api_ver", HAS_ARG | OPT_INT | OPT_EXPERT, { &vdpau_api_ver }, "" },
#endif
{ "autorotate", HAS_ARG | OPT_BOOL | OPT_SPEC |
OPT_EXPERT | OPT_INPUT, { .off = OFFSET(autorotate) },
"automatically insert correct rotate filters" },
/* audio options */
{ "aframes", OPT_AUDIO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_audio_frames },

View File

@@ -73,12 +73,10 @@ static int vda_retrieve_data(AVCodecContext *s, AVFrame *frame)
}
av_image_copy(vda->tmp_frame->data, vda->tmp_frame->linesize,
(const uint8_t **)data, linesize, vda->tmp_frame->format,
data, linesize, vda->tmp_frame->format,
frame->width, frame->height);
ret = av_frame_copy_props(vda->tmp_frame, frame);
CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
if (ret < 0)
return ret;

View File

@@ -32,7 +32,6 @@
#include "libavutil/avstring.h"
#include "libavutil/colorspace.h"
#include "libavutil/eval.h"
#include "libavutil/mathematics.h"
#include "libavutil/pixdesc.h"
#include "libavutil/imgutils.h"
@@ -1532,7 +1531,7 @@ static void step_to_next_frame(VideoState *is)
static double compute_target_delay(double delay, VideoState *is)
{
double sync_threshold, diff = 0;
double sync_threshold, diff;
/* update delay to follow master synchronisation source */
if (get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER) {
@@ -1554,7 +1553,7 @@ static double compute_target_delay(double delay, VideoState *is)
}
}
av_log(NULL, AV_LOG_TRACE, "video: delay=%0.3f A-V=%f\n",
av_dlog(NULL, "video: delay=%0.3f A-V=%f\n",
delay, -diff);
return delay;
@@ -1996,20 +1995,20 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
/* Note: this macro adds a filter before the lastly added filter, so the
* processing order of the filters is in reverse */
#define INSERT_FILT(name, arg) do { \
AVFilterContext *filt_ctx; \
\
ret = avfilter_graph_create_filter(&filt_ctx, \
avfilter_get_by_name(name), \
"ffplay_" name, arg, NULL, graph); \
if (ret < 0) \
goto fail; \
\
ret = avfilter_link(filt_ctx, 0, last_filter, 0); \
if (ret < 0) \
goto fail; \
\
last_filter = filt_ctx; \
#define INSERT_FILT(name, arg) do { \
AVFilterContext *filt_ctx; \
\
ret = avfilter_graph_create_filter(&filt_ctx, \
avfilter_get_by_name(name), \
"ffplay_" name, arg, NULL, graph); \
if (ret < 0) \
goto fail; \
\
ret = avfilter_link(filt_ctx, 0, last_filter, 0); \
if (ret < 0) \
goto fail; \
\
last_filter = filt_ctx; \
} while (0)
/* SDL YUV code is not handling odd width/height for some driver
@@ -2017,19 +2016,20 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
INSERT_FILT("crop", "floor(in_w/2)*2:floor(in_h/2)*2");
if (autorotate) {
double theta = get_rotation(is->video_st);
if (fabs(theta - 90) < 1.0) {
INSERT_FILT("transpose", "clock");
} else if (fabs(theta - 180) < 1.0) {
INSERT_FILT("hflip", NULL);
INSERT_FILT("vflip", NULL);
} else if (fabs(theta - 270) < 1.0) {
INSERT_FILT("transpose", "cclock");
} else if (fabs(theta) > 1.0) {
char rotate_buf[64];
snprintf(rotate_buf, sizeof(rotate_buf), "%f*PI/180", theta);
INSERT_FILT("rotate", rotate_buf);
AVDictionaryEntry *rotate_tag = av_dict_get(is->video_st->metadata, "rotate", NULL, 0);
if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
if (!strcmp(rotate_tag->value, "90")) {
INSERT_FILT("transpose", "clock");
} else if (!strcmp(rotate_tag->value, "180")) {
INSERT_FILT("hflip", NULL);
INSERT_FILT("vflip", NULL);
} else if (!strcmp(rotate_tag->value, "270")) {
INSERT_FILT("transpose", "cclock");
} else {
char rotate_buf[64];
snprintf(rotate_buf, sizeof(rotate_buf), "%s*PI/180", rotate_tag->value);
INSERT_FILT("rotate", rotate_buf);
}
}
}
@@ -2231,19 +2231,10 @@ static int video_thread(void *arg)
enum AVPixelFormat last_format = -2;
int last_serial = -1;
int last_vfilter_idx = 0;
if (!graph) {
av_frame_free(&frame);
return AVERROR(ENOMEM);
}
#endif
if (!frame) {
#if CONFIG_AVFILTER
avfilter_graph_free(&graph);
#endif
if (!frame)
return AVERROR(ENOMEM);
}
for (;;) {
ret = get_video_frame(is, frame);
@@ -2412,9 +2403,9 @@ static int synchronize_audio(VideoState *is, int nb_samples)
wanted_nb_samples = nb_samples + (int)(diff * is->audio_src.freq);
min_nb_samples = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100));
max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100));
wanted_nb_samples = av_clip(wanted_nb_samples, min_nb_samples, max_nb_samples);
wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples);
}
av_log(NULL, AV_LOG_TRACE, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
diff, avg_diff, wanted_nb_samples - nb_samples,
is->audio_clock, is->audio_diff_threshold);
}

View File

@@ -33,12 +33,10 @@
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
#include "libavutil/display.h"
#include "libavutil/hash.h"
#include "libavutil/opt.h"
#include "libavutil/pixdesc.h"
#include "libavutil/dict.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/libm.h"
#include "libavutil/parseutils.h"
#include "libavutil/timecode.h"
@@ -137,8 +135,6 @@ typedef enum {
SECTION_ID_PACKET,
SECTION_ID_PACKETS,
SECTION_ID_PACKETS_AND_FRAMES,
SECTION_ID_PACKET_SIDE_DATA_LIST,
SECTION_ID_PACKET_SIDE_DATA,
SECTION_ID_PIXEL_FORMAT,
SECTION_ID_PIXEL_FORMAT_FLAGS,
SECTION_ID_PIXEL_FORMAT_COMPONENT,
@@ -157,8 +153,6 @@ typedef enum {
SECTION_ID_STREAM_DISPOSITION,
SECTION_ID_STREAMS,
SECTION_ID_STREAM_TAGS,
SECTION_ID_STREAM_SIDE_DATA_LIST,
SECTION_ID_STREAM_SIDE_DATA,
SECTION_ID_SUBTITLE,
} SectionID;
@@ -178,9 +172,7 @@ static struct section sections[] = {
[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, { SECTION_ID_PACKET_SIDE_DATA_LIST, -1 } },
[SECTION_ID_PACKET_SIDE_DATA_LIST] ={ SECTION_ID_PACKET_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET_SIDE_DATA, -1 } },
[SECTION_ID_PACKET_SIDE_DATA] = { SECTION_ID_PACKET_SIDE_DATA, "side_data", 0, { -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" },
@@ -199,11 +191,9 @@ static struct section sections[] = {
SECTION_ID_PACKETS, SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS,
SECTION_ID_PIXEL_FORMATS, -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, SECTION_ID_STREAM_SIDE_DATA_LIST, -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" },
[SECTION_ID_STREAM_TAGS] = { SECTION_ID_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "stream_tags" },
[SECTION_ID_STREAM_SIDE_DATA_LIST] ={ SECTION_ID_STREAM_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_SIDE_DATA, -1 } },
[SECTION_ID_STREAM_SIDE_DATA] = { SECTION_ID_STREAM_SIDE_DATA, "side_data", 0, { -1 } },
[SECTION_ID_SUBTITLE] = { SECTION_ID_SUBTITLE, "subtitle", 0, { -1 } },
};
@@ -728,32 +718,6 @@ static void writer_print_data_hash(WriterContext *wctx, const char *name,
writer_print_string(wctx, name, buf, 0);
}
static void writer_print_integers(WriterContext *wctx, const char *name,
uint8_t *data, int size, const char *format,
int columns, int bytes, int offset_add)
{
AVBPrint bp;
int offset = 0, l, i;
av_bprint_init(&bp, 0, AV_BPRINT_SIZE_UNLIMITED);
av_bprintf(&bp, "\n");
while (size) {
av_bprintf(&bp, "%08x: ", offset);
l = FFMIN(size, columns);
for (i = 0; i < l; i++) {
if (bytes == 1) av_bprintf(&bp, format, *data);
else if (bytes == 2) av_bprintf(&bp, format, AV_RN16(data));
else if (bytes == 4) av_bprintf(&bp, format, AV_RN32(data));
data += bytes;
size --;
}
av_bprintf(&bp, "\n");
offset += offset_add;
}
writer_print_string(wctx, name, bp.str, 0);
av_bprint_finalize(&bp, NULL);
}
#define MAX_REGISTERED_WRITERS_NB 64
static const Writer *registered_writers[MAX_REGISTERED_WRITERS_NB + 1];
@@ -1759,25 +1723,6 @@ static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pk
if (pkt->pos != -1) print_fmt ("pos", "%"PRId64, pkt->pos);
else print_str_opt("pos", "N/A");
print_fmt("flags", "%c", pkt->flags & AV_PKT_FLAG_KEY ? 'K' : '_');
if (pkt->side_data_elems) {
int i;
writer_print_section_header(w, SECTION_ID_PACKET_SIDE_DATA_LIST);
for (i = 0; i < pkt->side_data_elems; i++) {
AVPacketSideData *sd = &pkt->side_data[i];
const char *name = av_packet_side_data_name(sd->type);
writer_print_section_header(w, SECTION_ID_PACKET_SIDE_DATA);
print_str("side_data_type", name ? name : "unknown");
print_int("side_data_size", sd->size);
if (sd->type == AV_PKT_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
}
writer_print_section_footer(w);
}
writer_print_section_footer(w);
}
if (do_show_data)
writer_print_data(w, "data", pkt->data, pkt->size);
writer_print_data_hash(w, "data_hash", pkt->data, pkt->size);
@@ -1824,7 +1769,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
s = av_get_media_type_string(stream->codec->codec_type);
if (s) print_str ("media_type", s);
else print_str_opt("media_type", "unknown");
print_int("stream_index", stream->index);
print_int("key_frame", frame->key_frame);
print_ts ("pkt_pts", frame->pkt_pts);
print_time("pkt_pts_time", frame->pkt_pts, &stream->time_base);
@@ -1889,11 +1833,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
name = av_frame_side_data_name(sd->type);
print_str("side_data_type", name ? name : "unknown");
print_int("side_data_size", sd->size);
if (sd->type == AV_FRAME_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
abort();
writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
}
writer_print_section_footer(w);
}
writer_print_section_footer(w);
@@ -2155,8 +2094,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
case AVMEDIA_TYPE_VIDEO:
print_int("width", dec_ctx->width);
print_int("height", dec_ctx->height);
print_int("coded_width", dec_ctx->coded_width);
print_int("coded_height", dec_ctx->coded_height);
print_int("has_b_frames", dec_ctx->has_b_frames);
sar = av_guess_sample_aspect_ratio(fmt_ctx, stream, NULL);
if (sar.den) {
@@ -2302,25 +2239,6 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
if (do_show_stream_tags)
ret = show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
if (stream->nb_side_data) {
int i;
writer_print_section_header(w, SECTION_ID_STREAM_SIDE_DATA_LIST);
for (i = 0; i < stream->nb_side_data; i++) {
AVPacketSideData *sd = &stream->side_data[i];
const char *name = av_packet_side_data_name(sd->type);
writer_print_section_header(w, SECTION_ID_STREAM_SIDE_DATA);
print_str("side_data_type", name ? name : "unknown");
print_int("side_data_size", sd->size);
if (sd->type == AV_PKT_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
}
writer_print_section_footer(w);
}
writer_print_section_footer(w);
}
writer_print_section_footer(w);
av_bprint_finalize(&pbuf, NULL);
fflush(stdout);

View File

@@ -967,10 +967,6 @@ static int handle_connection(HTTPContext *c)
/* close connection if trailer sent */
if (c->state == HTTPSTATE_SEND_DATA_TRAILER)
return -1;
/* Check if it is a single jpeg frame 123 */
if (c->stream->single_frame && c->data_count > c->cur_frame_bytes && c->cur_frame_bytes > 0) {
close_connection(c);
}
break;
case HTTPSTATE_RECEIVE_DATA:
/* no need to read if no events */
@@ -1208,10 +1204,6 @@ static FFServerIPAddressACL* parse_dynamic_acl(FFServerStream *stream,
}
acl = av_mallocz(sizeof(FFServerIPAddressACL));
if (!acl) {
fclose(f);
return NULL;
}
/* Build ACL */
while (fgets(line, sizeof(line), f)) {
@@ -2063,19 +2055,12 @@ static int open_input_stream(HTTPContext *c, const char *info)
ret = avformat_open_input(&s, input_filename, c->stream->ifmt,
&c->stream->in_opts);
if (ret < 0) {
http_log("Could not open input '%s': %s\n",
input_filename, av_err2str(ret));
http_log("Could not open input '%s': %s\n", input_filename, av_err2str(ret));
return ret;
}
/* set buffer size */
if (buf_size > 0) {
ret = ffio_set_buf_size(s->pb, buf_size);
if (ret < 0) {
http_log("Failed to set buffer size\n");
return ret;
}
}
if (buf_size > 0) ffio_set_buf_size(s->pb, buf_size);
s->flags |= AVFMT_FLAG_GENPTS;
c->fmt_in = s;
@@ -2137,20 +2122,14 @@ static int http_prepare_data(HTTPContext *c)
switch(c->state) {
case HTTPSTATE_SEND_DATA_HEADER:
ctx = avformat_alloc_context();
if (!ctx)
return AVERROR(ENOMEM);
c->fmt_ctx = *ctx;
av_freep(&ctx);
av_dict_copy(&(c->fmt_ctx.metadata), c->stream->metadata, 0);
c->fmt_ctx.streams = av_mallocz_array(c->stream->nb_streams,
sizeof(AVStream *));
if (!c->fmt_ctx.streams)
return AVERROR(ENOMEM);
c->fmt_ctx.streams = av_mallocz_array(c->stream->nb_streams, sizeof(AVStream *));
for(i=0;i<c->stream->nb_streams;i++) {
AVStream *src;
c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream));
/* if file or feed, then just take streams from FFServerStream struct */
if (!c->stream->feed ||
c->stream->feed == c->stream)
@@ -2301,8 +2280,7 @@ static int http_prepare_data(HTTPContext *c)
max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
else
max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
ret = ffio_open_dyn_packet_buf(&ctx->pb,
max_packet_size);
ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
} else {
ret = avio_open_dyn_buf(&ctx->pb);
}
@@ -2314,13 +2292,10 @@ static int http_prepare_data(HTTPContext *c)
ctx->pb->seekable = 0;
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, ist->time_base,
ost->time_base);
pkt.dts = av_rescale_q(pkt.dts, ist->time_base, ost->time_base);
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, ist->time_base,
ost->time_base);
pkt.duration = av_rescale_q(pkt.duration, ist->time_base,
ost->time_base);
pkt.pts = av_rescale_q(pkt.pts, ist->time_base, ost->time_base);
pkt.duration = av_rescale_q(pkt.duration, ist->time_base, ost->time_base);
if ((ret = av_write_frame(ctx, &pkt)) < 0) {
http_log("Error writing frame to output for stream '%s': %s\n",
c->stream->filename, av_err2str(ret));
@@ -2462,8 +2437,7 @@ static int http_send_data(HTTPContext *c)
ffurl_write(c->rtp_handles[c->packet_stream_index],
c->buffer_ptr, len);
c->buffer_ptr += len;
/* here we continue as we can send several packets
* per 10 ms slot */
/* here we continue as we can send several packets per 10 ms slot */
}
} else {
/* TCP data output */
@@ -2668,9 +2642,6 @@ static int http_receive_data(HTTPContext *c)
pb = avio_alloc_context(c->buffer, c->buffer_end - c->buffer,
0, NULL, NULL, NULL, NULL);
if (!pb)
goto fail;
pb->seekable = 0;
s->pb = pb;
@@ -2859,12 +2830,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
snprintf(avc->filename, 1024, "rtp://0.0.0.0");
}
avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
if (!avc->streams)
if (!(avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams))))
goto sdp_done;
avs = av_malloc_array(avc->nb_streams, sizeof(*avs));
if (!avs)
if (!(avs = av_malloc_array(avc->nb_streams, sizeof(*avs))))
goto sdp_done;
for(i = 0; i < stream->nb_streams; i++) {
@@ -2872,8 +2840,6 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
avc->streams[i]->codec = stream->streams[i]->codec;
}
*pbuffer = av_mallocz(2048);
if (!*pbuffer)
goto sdp_done;
av_sdp_create(&avc, 1, *pbuffer, 2048);
sdp_done:
@@ -2928,8 +2894,7 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url)
/* get the host IP */
len = sizeof(my_addr);
getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
content_length = prepare_sdp_description(stream, &content,
my_addr.sin_addr);
content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
if (content_length < 0) {
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return;
@@ -3390,10 +3355,6 @@ static AVStream *add_av_stream1(FFServerStream *stream,
return NULL;
if (copy) {
fst->codec = avcodec_alloc_context3(codec->codec);
if (!fst->codec) {
av_free(fst);
return NULL;
}
avcodec_copy_context(fst->codec, codec);
} else {
/* live streams must use the actual feed's codec since it may be
@@ -3502,6 +3463,7 @@ static void extract_mpeg4_header(AVFormatContext *infile)
if (p[0] == 0x00 && p[1] == 0x00 &&
p[2] == 0x01 && p[3] == 0xb6) {
size = p - pkt.data;
// av_hex_dump_log(infile, AV_LOG_DEBUG, pkt.data, size);
st->codec->extradata = av_mallocz(size + FF_INPUT_BUFFER_PADDING_SIZE);
st->codec->extradata_size = size;
memcpy(st->codec->extradata, pkt.data, size);
@@ -3588,8 +3550,7 @@ static void build_feed_streams(void)
} else {
/* we handle a stream coming from a feed */
for(i=0;i<stream->nb_streams;i++)
stream->feed_streams[i] = add_av_stream(feed,
stream->streams[i]);
stream->feed_streams[i] = add_av_stream(feed, stream->streams[i]);
}
}
}
@@ -3605,12 +3566,7 @@ static void build_feed_streams(void)
if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) >= 0) {
/* set buffer size */
int ret = ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
if (ret < 0) {
http_log("Failed to set buffer size\n");
exit(1);
}
ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
/* Now see if it matches */
if (s->nb_streams == feed->nb_streams) {
matches = 1;
@@ -3681,11 +3637,6 @@ static void build_feed_streams(void)
if (avio_check(feed->feed_filename, AVIO_FLAG_WRITE) <= 0) {
AVFormatContext *s = avformat_alloc_context();
if (!s) {
http_log("Failed to allocate context\n");
exit(1);
}
if (feed->readonly) {
http_log("Unable to create feed file '%s' as it is marked readonly\n",
feed->feed_filename);

View File

@@ -862,10 +862,8 @@ static int ffserver_parse_config_stream(FFServerConfig *config, const char *cmd,
} else {
stream->stream_type = STREAM_TYPE_LIVE;
/* JPEG cannot be used here, so use single frame MJPEG */
if (!strcmp(arg, "jpeg")) {
strcpy(arg, "singlejpeg");
stream->single_frame=1;
}
if (!strcmp(arg, "jpeg"))
strcpy(arg, "mjpeg");
stream->fmt = ffserver_guess_format(arg, NULL, NULL);
if (!stream->fmt)
ERROR("Unknown Format: '%s'\n", arg);

View File

@@ -79,7 +79,6 @@ typedef struct FFServerStream {
int multicast_port; /* first port used for multicast */
int multicast_ttl;
int loop; /* if true, send the stream in loops (only meaningful if file) */
char single_frame; /* only single frame */
/* feed specific */
int feed_opened; /* true if someone is writing to the feed */

View File

@@ -5,7 +5,6 @@ NAME = avcodec
HEADERS = avcodec.h \
avfft.h \
dv_profile.h \
d3d11va.h \
dxva2.h \
old_codec_ids.h \
qsv.h \
@@ -71,7 +70,6 @@ OBJS-$(CONFIG_IDCTDSP) += idctdsp.o simple_idct.o jrevdct.o
OBJS-$(CONFIG_IIRFILTER) += iirfilter.o
OBJS-$(CONFIG_IMDCT15) += imdct15.o
OBJS-$(CONFIG_INTRAX8) += intrax8.o intrax8dsp.o
OBJS-$(CONFIG_JPEGTABLES) += jpegtables.o
OBJS-$(CONFIG_LIBXVID) += libxvid_rc.o
OBJS-$(CONFIG_LLAUDDSP) += lossless_audiodsp.o
OBJS-$(CONFIG_LLVIDDSP) += lossless_videodsp.o
@@ -86,18 +84,14 @@ OBJS-$(CONFIG_MPEGAUDIODSP) += mpegaudiodsp.o \
mpegaudiodsp_data.o \
mpegaudiodsp_fixed.o \
mpegaudiodsp_float.o
OBJS-$(CONFIG_MPEGVIDEO) += mpegvideo.o mpegvideodsp.o rl.o \
mpegvideo_motion.o mpegutils.o \
mpegvideodata.o
OBJS-$(CONFIG_MPEGVIDEO) += mpegvideo.o mpegvideodsp.o \
mpegvideo_motion.o mpegutils.o
OBJS-$(CONFIG_MPEGVIDEOENC) += mpegvideo_enc.o mpeg12data.o \
motion_est.o ratecontrol.o \
mpegvideoencdsp.o
OBJS-$(CONFIG_NVENC) += nvenc.o
OBJS-$(CONFIG_PIXBLOCKDSP) += pixblockdsp.o
OBJS-$(CONFIG_QPELDSP) += qpeldsp.o
OBJS-$(CONFIG_QSV) += qsv.o
OBJS-$(CONFIG_QSVDEC) += qsvdec.o
OBJS-$(CONFIG_QSVENC) += qsvenc.o
OBJS-$(CONFIG_RANGECODER) += rangecoder.o
RDFT-OBJS-$(CONFIG_HARDCODED_TABLES) += sin_tables.o
OBJS-$(CONFIG_RDFT) += rdft.o $(RDFT-OBJS-yes)
@@ -139,7 +133,7 @@ OBJS-$(CONFIG_AMRWB_DECODER) += amrwbdec.o celp_filters.o \
celp_math.o acelp_filters.o \
acelp_vectors.o \
acelp_pitch_delay.o
OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o \
OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpeg.o mjpegenc_common.o \
mpegvideo_enc.o motion_est.o \
ratecontrol.o mpeg12data.o \
mpegvideo.o
@@ -147,7 +141,6 @@ 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-$(CONFIG_APNG_ENCODER) += png.o pngenc.o
OBJS-$(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
@@ -162,7 +155,7 @@ OBJS-$(CONFIG_ATRAC3P_DECODER) += atrac3plusdec.o atrac3plus.o \
atrac3plusdsp.o atrac.o
OBJS-$(CONFIG_AURA_DECODER) += cyuv.o
OBJS-$(CONFIG_AURA2_DECODER) += aura.o
OBJS-$(CONFIG_AVRN_DECODER) += avrndec.o mjpegdec.o
OBJS-$(CONFIG_AVRN_DECODER) += avrndec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_AVRP_DECODER) += r210dec.o
OBJS-$(CONFIG_AVRP_ENCODER) += r210enc.o
OBJS-$(CONFIG_AVS_DECODER) += avs.o
@@ -191,7 +184,7 @@ OBJS-$(CONFIG_CINEPAK_DECODER) += cinepak.o
OBJS-$(CONFIG_CINEPAK_ENCODER) += cinepakenc.o elbg.o
OBJS-$(CONFIG_CLJR_DECODER) += cljrdec.o
OBJS-$(CONFIG_CLJR_ENCODER) += cljrenc.o
OBJS-$(CONFIG_CLLC_DECODER) += cllc.o canopus.o
OBJS-$(CONFIG_CLLC_DECODER) += cllc.o
OBJS-$(CONFIG_COOK_DECODER) += cook.o
OBJS-$(CONFIG_COMFORTNOISE_DECODER) += cngdec.o celp_filters.o
OBJS-$(CONFIG_COMFORTNOISE_ENCODER) += cngenc.o
@@ -200,7 +193,7 @@ OBJS-$(CONFIG_CSCD_DECODER) += cscd.o
OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadsp.o \
dcadata.o dca_exss.o \
dca_xll.o synth_filter.o
synth_filter.o
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o
OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o \
dirac_arith.o mpeg12data.o dirac_dwt.o
@@ -253,7 +246,7 @@ OBJS-$(CONFIG_FLIC_DECODER) += flicvideo.o
OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o
OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o
OBJS-$(CONFIG_FRWU_DECODER) += frwu.o
OBJS-$(CONFIG_G2M_DECODER) += g2meet.o
OBJS-$(CONFIG_G2M_DECODER) += g2meet.o mjpeg.o
OBJS-$(CONFIG_G723_1_DECODER) += g723_1.o acelp_vectors.o \
celp_filters.o celp_math.o
OBJS-$(CONFIG_G723_1_ENCODER) += g723_1.o acelp_vectors.o celp_math.o
@@ -273,17 +266,13 @@ OBJS-$(CONFIG_H264_DECODER) += h264.o h264_cabac.o h264_cavlc.o \
h264_direct.o h264_loopfilter.o \
h264_mb.o h264_picture.o h264_ps.o \
h264_refs.o h264_sei.o h264_slice.o
OBJS-$(CONFIG_H264_MMAL_DECODER) += mmaldec.o
OBJS-$(CONFIG_H264_VDA_DECODER) += vda_h264_dec.o
OBJS-$(CONFIG_H264_QSV_DECODER) += qsvdec_h264.o
OBJS-$(CONFIG_H264_QSV_ENCODER) += qsvenc_h264.o
OBJS-$(CONFIG_H264_QSV_DECODER) += qsv_h264.o
OBJS-$(CONFIG_HEVC_DECODER) += hevc.o hevc_mvs.o hevc_ps.o hevc_sei.o \
hevc_cabac.o hevc_refs.o hevcpred.o \
hevcdsp.o hevc_filter.o
OBJS-$(CONFIG_HNM4_VIDEO_DECODER) += hnm4video.o
OBJS-$(CONFIG_HQ_HQA_DECODER) += hq_hqa.o hq_hqadata.o hq_hqadsp.o \
canopus.o
OBJS-$(CONFIG_HQX_DECODER) += hqx.o hqxvlc.o hqxdsp.o canopus.o
OBJS-$(CONFIG_HQX_DECODER) += hqx.o hqxvlc.o
OBJS-$(CONFIG_HUFFYUV_DECODER) += huffyuv.o huffyuvdec.o
OBJS-$(CONFIG_HUFFYUV_ENCODER) += huffyuv.o huffyuvenc.o
OBJS-$(CONFIG_IDCIN_DECODER) += idcinvideo.o
@@ -293,8 +282,8 @@ OBJS-$(CONFIG_IFF_ILBM_DECODER) += iff.o
OBJS-$(CONFIG_IMC_DECODER) += imc.o
OBJS-$(CONFIG_INDEO2_DECODER) += indeo2.o
OBJS-$(CONFIG_INDEO3_DECODER) += indeo3.o
OBJS-$(CONFIG_INDEO4_DECODER) += indeo4.o ivi.o ivi_dsp.o
OBJS-$(CONFIG_INDEO5_DECODER) += indeo5.o ivi.o ivi_dsp.o
OBJS-$(CONFIG_INDEO4_DECODER) += indeo4.o ivi_common.o ivi_dsp.o
OBJS-$(CONFIG_INDEO5_DECODER) += indeo5.o ivi_common.o ivi_dsp.o
OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER) += dpcm.o
OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
OBJS-$(CONFIG_JACOSUB_DECODER) += jacosubdec.o ass.o
@@ -308,7 +297,7 @@ OBJS-$(CONFIG_JV_DECODER) += jvdec.o
OBJS-$(CONFIG_KGV1_DECODER) += kgv1dec.o
OBJS-$(CONFIG_KMVC_DECODER) += kmvc.o
OBJS-$(CONFIG_LAGARITH_DECODER) += lagarith.o lagarithrac.o
OBJS-$(CONFIG_LJPEG_ENCODER) += ljpegenc.o mjpegenc_common.o
OBJS-$(CONFIG_LJPEG_ENCODER) += ljpegenc.o mjpeg.o mjpegenc_common.o
OBJS-$(CONFIG_LOCO_DECODER) += loco.o
OBJS-$(CONFIG_MACE3_DECODER) += mace.o
OBJS-$(CONFIG_MACE6_DECODER) += mace.o
@@ -317,8 +306,8 @@ OBJS-$(CONFIG_METASOUND_DECODER) += metasound.o metasound_data.o \
twinvq.o
OBJS-$(CONFIG_MICRODVD_DECODER) += microdvddec.o ass.o
OBJS-$(CONFIG_MIMIC_DECODER) += mimic.o
OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpegenc_common.o
OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o mjpeg.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o mjpegenc_common.o
OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o
OBJS-$(CONFIG_MLP_DECODER) += mlpdec.o mlpdsp.o
OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o
@@ -367,6 +356,7 @@ OBJS-$(CONFIG_MXPEG_DECODER) += mxpegdec.o
OBJS-$(CONFIG_NELLYMOSER_DECODER) += nellymoserdec.o nellymoser.o
OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o
OBJS-$(CONFIG_NUV_DECODER) += nuv.o rtjpeg.o
OBJS-$(CONFIG_NVENC_ENCODER) += nvenc.o
OBJS-$(CONFIG_ON2AVC_DECODER) += on2avc.o on2avcdata.o
OBJS-$(CONFIG_OPUS_DECODER) += opusdec.o opus.o opus_celt.o \
opus_silk.o vorbis_data.o
@@ -468,7 +458,6 @@ OBJS-$(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_TDSC_DECODER) += tdsc.o
OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
OBJS-$(CONFIG_TIFF_DECODER) += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
OBJS-$(CONFIG_TIFF_ENCODER) += tiffenc.o rle.o lzwenc.o tiff_data.o
@@ -517,8 +506,7 @@ OBJS-$(CONFIG_VP6_DECODER) += vp6.o vp56.o vp56data.o vp56dsp.o \
vp6dsp.o vp56rac.o
OBJS-$(CONFIG_VP7_DECODER) += vp8.o vp8dsp.o vp56rac.o
OBJS-$(CONFIG_VP8_DECODER) += vp8.o vp8dsp.o vp56rac.o
OBJS-$(CONFIG_VP9_DECODER) += vp9.o vp9dsp.o vp56rac.o vp9dsp_8bpp.o \
vp9dsp_10bpp.o vp9dsp_12bpp.o
OBJS-$(CONFIG_VP9_DECODER) += vp9.o vp9dsp.o vp56rac.o
OBJS-$(CONFIG_VPLAYER_DECODER) += textdec.o ass.o
OBJS-$(CONFIG_VQA_DECODER) += vqavideo.o
OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o
@@ -673,7 +661,6 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_VIMA_DECODER) += vima.o adpcm_data.o
# hardware accelerators
OBJS-$(CONFIG_D3D11VA) += dxva2.o
OBJS-$(CONFIG_DXVA2) += dxva2.o
OBJS-$(CONFIG_VAAPI) += vaapi.o
OBJS-$(CONFIG_VDA) += vda.o
@@ -681,23 +668,19 @@ OBJS-$(CONFIG_VDPAU) += vdpau.o
OBJS-$(CONFIG_H263_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_H263_VDPAU_HWACCEL) += vdpau_mpeg4.o
OBJS-$(CONFIG_H264_D3D11VA_HWACCEL) += dxva2_h264.o
OBJS-$(CONFIG_H264_DXVA2_HWACCEL) += dxva2_h264.o
OBJS-$(CONFIG_H264_VAAPI_HWACCEL) += vaapi_h264.o
OBJS-$(CONFIG_H264_VDA_HWACCEL) += vda_h264.o
OBJS-$(CONFIG_H264_VDPAU_HWACCEL) += vdpau_h264.o
OBJS-$(CONFIG_HEVC_D3D11VA_HWACCEL) += dxva2_hevc.o
OBJS-$(CONFIG_HEVC_DXVA2_HWACCEL) += dxva2_hevc.o
OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL) += vdpau_mpeg12.o
OBJS-$(CONFIG_MPEG1_XVMC_HWACCEL) += mpegvideo_xvmc.o
OBJS-$(CONFIG_MPEG2_D3D11VA_HWACCEL) += dxva2_mpeg2.o
OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL) += dxva2_mpeg2.o
OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL) += vaapi_mpeg2.o
OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL) += vdpau_mpeg12.o
OBJS-$(CONFIG_MPEG2_XVMC_HWACCEL) += mpegvideo_xvmc.o
OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL) += vdpau_mpeg4.o
OBJS-$(CONFIG_VC1_D3D11VA_HWACCEL) += dxva2_vc1.o
OBJS-$(CONFIG_VC1_DXVA2_HWACCEL) += dxva2_vc1.o
OBJS-$(CONFIG_VC1_VAAPI_HWACCEL) += vaapi_vc1.o
OBJS-$(CONFIG_VC1_VDPAU_HWACCEL) += vdpau_vc1.o
@@ -733,6 +716,7 @@ OBJS-$(CONFIG_OGG_DEMUXER) += mpeg12data.o \
OBJS-$(CONFIG_OGG_MUXER) += flac.o flacdata.o \
vorbis_data.o
OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_RTPDEC) += mjpeg.o
OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_SPDIF_MUXER) += dca.o
OBJS-$(CONFIG_TAK_DEMUXER) += tak.o
@@ -747,7 +731,6 @@ OBJS-$(CONFIG_ELBG_FILTER) += elbg.o
# external codec libraries
OBJS-$(CONFIG_LIBAACPLUS_ENCODER) += libaacplus.o
OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
OBJS-$(CONFIG_LIBDCADEC_DECODER) += libdcadec.o dca.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
OBJS-$(CONFIG_LIBFDK_AAC_DECODER) += libfdk-aacdec.o
OBJS-$(CONFIG_LIBFDK_AAC_ENCODER) += libfdk-aacenc.o
@@ -790,8 +773,7 @@ OBJS-$(CONFIG_LIBVPX_VP8_ENCODER) += libvpxenc.o
OBJS-$(CONFIG_LIBVPX_VP9_DECODER) += libvpxdec.o libvpx.o
OBJS-$(CONFIG_LIBVPX_VP9_ENCODER) += libvpxenc.o libvpx.o
OBJS-$(CONFIG_LIBWAVPACK_ENCODER) += libwavpackenc.o
OBJS-$(CONFIG_LIBWEBP_ENCODER) += libwebpenc_common.o libwebpenc.o
OBJS-$(CONFIG_LIBWEBP_ANIM_ENCODER) += libwebpenc_common.o libwebpenc_animencoder.o
OBJS-$(CONFIG_LIBWEBP_ENCODER) += libwebpenc.o
OBJS-$(CONFIG_LIBX264_ENCODER) += libx264.o
OBJS-$(CONFIG_LIBX265_ENCODER) += libx265.o
OBJS-$(CONFIG_LIBXAVS_ENCODER) += libxavs.o
@@ -850,9 +832,8 @@ OBJS-$(CONFIG_CHOMP_BSF) += chomp_bsf.o
OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += dump_extradata_bsf.o
OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF) += h264_mp4toannexb_bsf.o
OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF) += imx_dump_header_bsf.o
OBJS-$(CONFIG_MJPEG2JPEG_BSF) += mjpeg2jpeg_bsf.o
OBJS-$(CONFIG_MJPEG2JPEG_BSF) += mjpeg2jpeg_bsf.o mjpeg.o
OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF) += mjpega_dump_header_bsf.o
OBJS-$(CONFIG_MPEG4_UNPACK_BFRAMES_BSF) += mpeg4_unpack_bframes_bsf.o
OBJS-$(CONFIG_MOV2TEXTSUB_BSF) += movsub_bsf.o
OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o \
mpegaudiodata.o
@@ -879,24 +860,19 @@ SKIPHEADERS += %_tablegen.h \
tableprint_vlc.h \
$(ARCH)/vp56_arith.h \
SKIPHEADERS-$(CONFIG_D3D11VA) += d3d11va.h dxva2_internal.h
SKIPHEADERS-$(CONFIG_DXVA2) += dxva2.h dxva2_internal.h
SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER) += libschroedinger.h
SKIPHEADERS-$(CONFIG_LIBUTVIDEO) += libutvideo.h
SKIPHEADERS-$(CONFIG_QSV) += qsv.h qsv_internal.h
SKIPHEADERS-$(CONFIG_QSVDEC) += qsvdec.h
SKIPHEADERS-$(CONFIG_QSVENC) += qsvenc.h
SKIPHEADERS-$(CONFIG_XVMC) += xvmc.h
SKIPHEADERS-$(CONFIG_VAAPI) += vaapi_internal.h
SKIPHEADERS-$(CONFIG_VDA) += vda.h vda_internal.h
SKIPHEADERS-$(CONFIG_VDPAU) += vdpau.h vdpau_internal.h
TESTPROGS = imgconvert \
mathops \
options \
avfft \
TESTPROGS += api-flac
TESTPROGS-$(CONFIG_CABAC) += cabac
TESTPROGS-$(CONFIG_FFT) += fft fft-fixed fft-fixed32

View File

@@ -141,10 +141,6 @@ typedef struct PredictorState {
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
#define NOISE_PRE 256 ///< preamble for NOISE_BT, put in bitstream with the first noise band
#define NOISE_PRE_BITS 9 ///< length of preamble
#define NOISE_OFFSET 90 ///< subtracted from global gain, used as offset for the preamble
/**
* Long Term Prediction
*/

View File

@@ -40,12 +40,6 @@
#include "aacenc.h"
#include "aactab.h"
/** Frequency in Hz for lower limit of noise substitution **/
#define NOISE_LOW_LIMIT 4000
/** Total number of usable codebooks **/
#define CB_TOT 13
/** bits needed to code codebook run value for long windows */
static const uint8_t run_value_bits_long[64] = {
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
@@ -63,10 +57,6 @@ static const uint8_t * const run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
/** Map to convert values from BandCodingPath index to a codebook index **/
static const uint8_t aac_cb_out_map[CB_TOT] = {0,1,2,3,4,5,6,7,8,9,10,11,13};
/** Inverse map to convert from codebooks to BandCodingPath indices **/
static const uint8_t aac_cb_in_map[CB_TOT+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12};
/**
* Quantize one coefficient.
@@ -118,7 +108,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
const float *scaled, int size, int scale_idx,
int cb, const float lambda, const float uplim,
int *bits, int BT_ZERO, int BT_UNSIGNED,
int BT_PAIR, int BT_ESC, int BT_NOISE)
int BT_PAIR, int BT_ESC)
{
const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
const float Q = ff_aac_pow2sf_tab [q_idx];
@@ -129,6 +119,8 @@ static av_always_inline float quantize_and_encode_band_cost_template(
float cost = 0;
const int dim = BT_PAIR ? 2 : 4;
int resbits = 0;
const int range = aac_cb_range[cb];
const int maxval = aac_cb_maxval[cb];
int off;
if (BT_ZERO) {
@@ -138,22 +130,15 @@ static av_always_inline float quantize_and_encode_band_cost_template(
*bits = 0;
return cost * lambda;
}
if (BT_NOISE) {
for (i = 0; i < size; i++)
cost += in[i]*in[i];
if (bits)
*bits = 0;
return cost * lambda;
}
if (!scaled) {
abs_pow34_v(s->scoefs, in, size);
scaled = s->scoefs;
}
quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, aac_cb_maxval[cb]);
quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, maxval);
if (BT_UNSIGNED) {
off = 0;
} else {
off = aac_cb_maxval[cb];
off = maxval;
}
for (i = 0; i < size; i += dim) {
const float *vec;
@@ -162,7 +147,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
int curbits;
float rd = 0.0f;
for (j = 0; j < dim; j++) {
curidx *= aac_cb_range[cb];
curidx *= range;
curidx += quants[j] + off;
}
curbits = ff_aac_spectral_bits[cb-1][curidx];
@@ -210,7 +195,7 @@ static av_always_inline float quantize_and_encode_band_cost_template(
int len = av_log2(coef);
put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
put_sbits(pb, len, coef);
put_bits(pb, len, coef & ((1 << len) - 1));
}
}
}
@@ -222,17 +207,8 @@ static av_always_inline float quantize_and_encode_band_cost_template(
return cost;
}
static float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb,
const float *in, const float *scaled,
int size, int scale_idx, int cb,
const float lambda, const float uplim,
int *bits) {
av_assert0(0);
return 0.0f;
}
#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE) \
static float quantize_and_encode_band_cost_ ## NAME( \
#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC) \
static float quantize_and_encode_band_cost_ ## NAME( \
struct AACEncContext *s, \
PutBitContext *pb, const float *in, \
const float *scaled, int size, int scale_idx, \
@@ -241,16 +217,15 @@ static float quantize_and_encode_band_cost_ ## NAME(
return quantize_and_encode_band_cost_template( \
s, pb, in, scaled, size, scale_idx, \
BT_ESC ? ESC_BT : cb, lambda, uplim, bits, \
BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE); \
BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC); \
}
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO, 1, 0, 0, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC, 0, 1, 1, 1, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO, 1, 0, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0)
QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC, 0, 1, 1, 1)
static float (*const quantize_and_encode_band_cost_arr[])(
struct AACEncContext *s,
@@ -270,8 +245,6 @@ static float (*const quantize_and_encode_band_cost_arr[])(
quantize_and_encode_band_cost_UPAIR,
quantize_and_encode_band_cost_UPAIR,
quantize_and_encode_band_cost_ESC,
quantize_and_encode_band_cost_NONE, /* CB 12 doesn't exist */
quantize_and_encode_band_cost_NOISE,
};
#define quantize_and_encode_band_cost( \
@@ -339,7 +312,7 @@ typedef struct BandCodingPath {
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda)
{
BandCodingPath path[120][CB_TOT];
BandCodingPath path[120][12];
int w, swb, cb, start, size;
int i, j;
const int max_sfb = sce->ics.max_sfb;
@@ -352,7 +325,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
start = win*128;
for (cb = 0; cb < CB_TOT; cb++) {
for (cb = 0; cb < 12; cb++) {
path[0][cb].cost = 0.0f;
path[0][cb].prev_idx = -1;
path[0][cb].run = 0;
@@ -360,7 +333,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
for (swb = 0; swb < max_sfb; swb++) {
size = sce->ics.swb_sizes[swb];
if (sce->zeroes[win*16 + swb]) {
for (cb = 0; cb < CB_TOT; cb++) {
for (cb = 0; cb < 12; cb++) {
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = path[swb][cb].cost;
path[swb+1][cb].run = path[swb][cb].run + 1;
@@ -370,14 +343,14 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
int mincb = next_mincb;
next_minrd = INFINITY;
next_mincb = 0;
for (cb = 0; cb < CB_TOT; cb++) {
for (cb = 0; cb < 12; cb++) {
float cost_stay_here, cost_get_here;
float rd = 0.0f;
for (w = 0; w < group_len; w++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(win+w)*16+swb];
rd += quantize_band_cost(s, sce->coeffs + start + w*128,
s->scoefs + start + w*128, size,
sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
sce->sf_idx[(win+w)*16+swb], cb,
lambda / band->threshold, INFINITY, NULL);
}
cost_stay_here = path[swb][cb].cost + rd;
@@ -406,7 +379,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
//convert resulting path from backward-linked list
stack_len = 0;
idx = 0;
for (cb = 1; cb < CB_TOT; cb++)
for (cb = 1; cb < 12; cb++)
if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
idx = cb;
ppos = max_sfb;
@@ -421,13 +394,12 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
//perform actual band info encoding
start = 0;
for (i = stack_len - 1; i >= 0; i--) {
cb = aac_cb_out_map[stackcb[i]];
put_bits(&s->pb, 4, cb);
put_bits(&s->pb, 4, stackcb[i]);
count = stackrun[i];
memset(sce->zeroes + win*16 + start, !cb, count);
memset(sce->zeroes + win*16 + start, !stackcb[i], count);
//XXX: memset when band_type is also uint8_t
for (j = 0; j < count; j++) {
sce->band_type[win*16 + start] = cb;
sce->band_type[win*16 + start] = stackcb[i];
start++;
}
while (count >= run_esc) {
@@ -441,7 +413,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda)
{
BandCodingPath path[120][CB_TOT];
BandCodingPath path[120][12];
int w, swb, cb, start, size;
int i, j;
const int max_sfb = sce->ics.max_sfb;
@@ -454,7 +426,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
abs_pow34_v(s->scoefs, sce->coeffs, 1024);
start = win*128;
for (cb = 0; cb < CB_TOT; cb++) {
for (cb = 0; cb < 12; cb++) {
path[0][cb].cost = run_bits+4;
path[0][cb].prev_idx = -1;
path[0][cb].run = 0;
@@ -478,7 +450,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
}
next_minbits = path[swb+1][0].cost;
next_mincb = 0;
for (cb = 1; cb < CB_TOT; cb++) {
for (cb = 1; cb < 12; cb++) {
path[swb+1][cb].cost = 61450;
path[swb+1][cb].prev_idx = -1;
path[swb+1][cb].run = 0;
@@ -487,7 +459,6 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
float minbits = next_minbits;
int mincb = next_mincb;
int startcb = sce->band_type[win*16+swb];
startcb = aac_cb_in_map[startcb];
next_minbits = INFINITY;
next_mincb = 0;
for (cb = 0; cb < startcb; cb++) {
@@ -495,20 +466,13 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
path[swb+1][cb].prev_idx = -1;
path[swb+1][cb].run = 0;
}
for (cb = startcb; cb < CB_TOT; cb++) {
for (cb = startcb; cb < 12; cb++) {
float cost_stay_here, cost_get_here;
float bits = 0.0f;
if (cb == 12 && sce->band_type[win*16+swb] != NOISE_BT) {
path[swb+1][cb].cost = 61450;
path[swb+1][cb].prev_idx = -1;
path[swb+1][cb].run = 0;
continue;
}
for (w = 0; w < group_len; w++) {
bits += quantize_band_cost(s, sce->coeffs + start + w*128,
s->scoefs + start + w*128, size,
sce->sf_idx[(win+w)*16+swb],
aac_cb_out_map[cb],
sce->sf_idx[(win+w)*16+swb], cb,
0, INFINITY, NULL);
}
cost_stay_here = path[swb][cb].cost + bits;
@@ -537,7 +501,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
//convert resulting path from backward-linked list
stack_len = 0;
idx = 0;
for (cb = 1; cb < CB_TOT; cb++)
for (cb = 1; cb < 12; cb++)
if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
idx = cb;
ppos = max_sfb;
@@ -553,13 +517,12 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
//perform actual band info encoding
start = 0;
for (i = stack_len - 1; i >= 0; i--) {
cb = aac_cb_out_map[stackcb[i]];
put_bits(&s->pb, 4, cb);
put_bits(&s->pb, 4, stackcb[i]);
count = stackrun[i];
memset(sce->zeroes + win*16 + start, !cb, count);
memset(sce->zeroes + win*16 + start, !stackcb[i], count);
//XXX: memset when band_type is also uint8_t
for (j = 0; j < count; j++) {
sce->band_type[win*16 + start] = cb;
sce->band_type[win*16 + start] = stackcb[i];
start++;
}
while (count >= run_esc) {
@@ -748,11 +711,9 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
{
int start = 0, i, w, w2, g;
int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
const float freq_mult = avctx->sample_rate/(1024.0f/sce->ics.num_windows)/2.0f;
float dists[128] = { 0 }, uplims[128] = { 0 };
float dists[128] = { 0 }, uplims[128];
float maxvals[128];
int noise_sf[128] = { 0 };
int fflag, minscaler, minscaler_n;
int fflag, minscaler;
int its = 0;
int allz = 0;
float minthr = INFINITY;
@@ -763,14 +724,12 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
//XXX: some heuristic to determine initial quantizers will reduce search time
//determine zero bands and upper limits
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = 0;
for (g = 0; g < sce->ics.num_swb; g++) {
int nz = 0;
float uplim = 0.0f, energy = 0.0f;
float uplim = 0.0f;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
uplim += band->threshold;
energy += band->energy;
if (band->energy <= band->threshold || band->threshold == 0.0f) {
sce->zeroes[(w+w2)*16+g] = 1;
continue;
@@ -778,18 +737,10 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
nz = 1;
}
uplims[w*16+g] = uplim *512;
if (s->options.pns && start*freq_mult > NOISE_LOW_LIMIT && energy < uplim * 1.2f) {
noise_sf[w*16+g] = av_clip(4+FFMIN(log2f(energy)*2,255), -100, 155);
sce->band_type[w*16+g] = NOISE_BT;
nz= 1;
} else { /** Band type will be determined by the twoloop algorithm */
sce->band_type[w*16+g] = 0;
}
sce->zeroes[w*16+g] = !nz;
if (nz)
minthr = FFMIN(minthr, uplim);
allz |= nz;
start += sce->ics.swb_sizes[g];
}
}
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
@@ -820,7 +771,6 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
do {
int tbits, qstep;
minscaler = sce->sf_idx[0];
minscaler_n = sce->sf_idx[0];
//inner loop - quantize spectrum to fit into given number of bits
qstep = its ? 1 : 32;
do {
@@ -835,11 +785,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
int cb;
float dist = 0.0f;
if (sce->band_type[w*16+g] == NOISE_BT) {
minscaler_n = FFMIN(minscaler_n, noise_sf[w*16+g]);
start += sce->ics.swb_sizes[g];
continue;
} else if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
start += sce->ics.swb_sizes[g];
continue;
}
@@ -882,17 +828,9 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
fflag = 0;
minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for (g = 0; g < sce->ics.num_swb; g++)
if (sce->band_type[w*16+g] == NOISE_BT)
sce->sf_idx[w*16+g] = av_clip(noise_sf[w*16+g], minscaler_n, minscaler_n + SCALE_MAX_DIFF);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (g = 0; g < sce->ics.num_swb; g++) {
int prevsc = sce->sf_idx[w*16+g];
if (sce->band_type[w*16+g] == NOISE_BT)
continue;
if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) {
if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1))
sce->sf_idx[w*16+g]--;

View File

@@ -303,11 +303,6 @@ static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
if (num_back_channels < 0)
return 0;
if (num_side_channels == 0 && num_back_channels >= 4) {
num_side_channels = 2;
num_back_channels -= 2;
}
i = 0;
if (num_front_channels & 1) {
e2c_vec[i] = (struct elem_to_channel) {
@@ -429,7 +424,7 @@ 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;
@@ -462,18 +457,12 @@ static int output_configure(AACContext *ac,
AVCodecContext *avctx = ac->avctx;
int i, channels = 0, ret;
uint64_t layout = 0;
uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
uint8_t type_counts[TYPE_END] = { 0 };
if (ac->oc[1].layout_map != layout_map) {
memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
ac->oc[1].layout_map_tags = tags;
}
for (i = 0; i < tags; i++) {
int type = layout_map[i][0];
int id = layout_map[i][1];
id_map[type][id] = type_counts[type]++;
}
// Try to sniff a reasonable channel order, otherwise output the
// channels in the order the PCE declared them.
if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
@@ -481,14 +470,12 @@ static int output_configure(AACContext *ac,
for (i = 0; i < tags; i++) {
int type = layout_map[i][0];
int id = layout_map[i][1];
int iid = id_map[type][id];
int position = layout_map[i][2];
// Allocate or free elements depending on if they are in the
// current program configuration.
ret = che_configure(ac, position, type, iid, &channels);
ret = che_configure(ac, position, type, id, &channels);
if (ret < 0)
return ret;
ac->tag_che_map[type][id] = ac->che[type][iid];
}
if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
if (layout == AV_CH_FRONT_CENTER) {
@@ -498,6 +485,7 @@ static int output_configure(AACContext *ac,
}
}
memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
if (layout) avctx->channel_layout = layout;
ac->oc[1].channel_layout = layout;
avctx->channels = ac->oc[1].channels = channels;
@@ -539,8 +527,7 @@ static int set_default_channel_config(AVCodecContext *avctx,
int *tags,
int channel_config)
{
if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
channel_config > 12) {
if (channel_config < 1 || channel_config > 7) {
av_log(avctx, AV_LOG_ERROR,
"invalid default channel configuration (%d)\n",
channel_config);
@@ -620,19 +607,11 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
/* For indexed channel configurations map the channels solely based
* on position. */
switch (ac->oc[1].m4ac.chan_config) {
case 12:
case 7:
if (ac->tags_mapped == 3 && type == TYPE_CPE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
}
case 11:
if (ac->tags_mapped == 2 &&
ac->oc[1].m4ac.chan_config == 11 &&
type == TYPE_SCE) {
ac->tags_mapped++;
return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
}
case 6:
/* Some streams incorrectly code 5.1 audio as
* SCE[0] CPE[0] CPE[1] SCE[1]
@@ -921,7 +900,7 @@ 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);
@@ -964,10 +943,10 @@ static int decode_audio_specific_config(AACContext *ac,
GetBitContext gb;
int i, ret;
ff_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
for (i = 0; i < bit_size >> 3; i++)
ff_dlog(avctx, "%02x ", data[i]);
ff_dlog(avctx, "\n");
av_dlog(avctx, "%02x ", data[i]);
av_dlog(avctx, "\n");
if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
return ret;
@@ -1014,7 +993,7 @@ static int decode_audio_specific_config(AACContext *ac,
return AVERROR(ENOSYS);
}
ff_dlog(avctx,
av_dlog(avctx,
"AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
m4ac->sample_rate, m4ac->sbr,
@@ -1415,7 +1394,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
int band_type_run_end[120])
{
int g, i, idx = 0;
int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
int offset[3] = { global_gain, global_gain - 90, 0 };
int clipped_offset;
int noise_flag = 1;
for (g = 0; g < ics->num_window_groups; g++) {
@@ -1427,7 +1406,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
} else if ((band_type[idx] == INTENSITY_BT) ||
(band_type[idx] == INTENSITY_BT2)) {
for (; i < run_end; i++, idx++) {
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
clipped_offset = av_clip(offset[2], -155, 100);
if (offset[2] != clipped_offset) {
avpriv_request_sample(ac->avctx,
@@ -1440,9 +1419,9 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
} else if (band_type[idx] == NOISE_BT) {
for (; i < run_end; i++, idx++) {
if (noise_flag-- > 0)
offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
offset[1] += get_bits(gb, 9) - 256;
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
clipped_offset = av_clip(offset[1], -100, 155);
if (offset[1] != clipped_offset) {
avpriv_request_sample(ac->avctx,
@@ -1454,7 +1433,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
}
} else {
for (; i < run_end; i++, idx++) {
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (offset[0] > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
"Scalefactor (%d) out of range.\n", offset[0]);
@@ -2906,7 +2885,7 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
ac->tags_mapped = 0;
if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
if (chan_config < 0 || chan_config >= 8) {
avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
chan_config);
return AVERROR_INVALIDDATA;
@@ -2987,11 +2966,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
if (avctx->debug & FF_DEBUG_STARTCODE)
av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
if (!avctx->channels && elem_type != TYPE_PCE) {
err = AVERROR_INVALIDDATA;
goto fail;
}
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",
@@ -3081,11 +3055,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
}
}
if (!avctx->channels) {
*got_frame_ptr = 0;
return 0;
}
spectral_to_sample(ac);
multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
@@ -3104,12 +3073,7 @@ 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;
ac->frame->sample_rate = avctx->sample_rate;

View File

@@ -78,9 +78,9 @@ static const float * const tns_tmp2_map[4] = {
};
// @}
static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 4, 5, 0, 5, 0 };
static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
static const uint8_t aac_channel_layout_map[16][5][3] = {
static const uint8_t aac_channel_layout_map[7][5][3] = {
{ { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, },
{ { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
{ { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
@@ -88,16 +88,9 @@ static const uint8_t aac_channel_layout_map[16][5][3] = {
{ { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, },
{ { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE }, },
{ { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_FRONT }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE }, },
{ { 0, } },
{ { 0, } },
{ { 0, } },
{ { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE }, },
{ { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_SIDE }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE }, },
{ { 0, } },
/* TODO: Add 7+1 TOP configuration */
};
static const uint64_t aac_channel_layout[16] = {
static const uint64_t aac_channel_layout[8] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
@@ -106,12 +99,6 @@ static const uint64_t aac_channel_layout[16] = {
AV_CH_LAYOUT_5POINT1_BACK,
AV_CH_LAYOUT_7POINT1_WIDE_BACK,
0,
0,
0,
AV_CH_LAYOUT_6POINT1,
AV_CH_LAYOUT_7POINT1,
0,
/* AV_CH_LAYOUT_7POINT1_TOP, */
};
#endif /* AVCODEC_AACDECTAB_H */

View File

@@ -388,26 +388,15 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce)
{
int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
int noise_flag = 1;
int off = sce->sf_idx[0], diff;
int i, w;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (i = 0; i < sce->ics.max_sfb; i++) {
if (!sce->zeroes[w*16 + i]) {
if (sce->band_type[w*16 + i] == NOISE_BT) {
diff = sce->sf_idx[w*16 + i] - off_pns;
off_pns = sce->sf_idx[w*16 + i];
if (noise_flag-- > 0) {
put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
continue;
}
} else {
diff = sce->sf_idx[w*16 + i] - off_sf;
off_sf = sce->sf_idx[w*16 + i];
}
diff += SCALE_DIFF_ZERO;
diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
av_assert0(diff >= 0 && diff <= 120);
off = sce->sf_idx[w*16 + i];
put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
}
}
@@ -733,9 +722,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows(7);
if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
return ret;
if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))
return ret;
return 0;
@@ -842,9 +831,6 @@ static const AVOption aacenc_options[] = {
{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
{"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
{"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},
{"disable", "Disable PNS", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
{"enable", "Enable PNS (Proof of concept)", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
{NULL}
};

View File

@@ -42,7 +42,6 @@ typedef enum AACCoder {
typedef struct AACEncOptions {
int stereo_mode;
int aac_coder;
int pns;
} AACEncOptions;
struct AACEncContext;

View File

@@ -304,8 +304,6 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
const float num_bark = calc_bark((float)bandwidth);
ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
if (!ctx->model_priv_data)
return AVERROR(ENOMEM);
pctx = (AacPsyContext*) ctx->model_priv_data;
pctx->chan_bitrate = chan_bitrate;
@@ -315,7 +313,7 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
ctx->bitres.size = 6144 - pctx->frame_bits;
ctx->bitres.size -= ctx->bitres.size % 8;
pctx->fill_level = ctx->bitres.size;
minath = ath(3410 - 0.733 * ATH_ADD, ATH_ADD);
minath = ath(3410, ATH_ADD);
for (j = 0; j < 2; j++) {
AacPsyCoeffs *coeffs = pctx->psy_coef[j];
const uint8_t *band_sizes = ctx->bands[j];
@@ -357,10 +355,6 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
}
pctx->ch = av_mallocz_array(ctx->avctx->channels, sizeof(AacPsyChannel));
if (!pctx->ch) {
av_freep(&ctx->model_priv_data);
return AVERROR(ENOMEM);
}
lame_window_init(pctx, ctx->avctx);
@@ -733,10 +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

@@ -514,7 +514,7 @@ 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 +528,6 @@ 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;
@@ -641,7 +635,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb, SBRData *ch_data)
{
int i;
int bs_pointer = 0;
unsigned bs_pointer = 0;
// frameLengthFlag ? 15 : 16; 960 sample length frames unsupported; this value is numTimeSlots
int abs_bord_trail = 16;
int num_rel_lead, num_rel_trail;
@@ -734,7 +728,6 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
break;
}
av_assert0(bs_pointer >= 0);
if (bs_pointer > ch_data->bs_num_env + 1) {
av_log(ac->avctx, AV_LOG_ERROR,
"Invalid bitstream, bs_pointer points to a middle noise border outside the time borders table: %d\n",
@@ -754,11 +747,11 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
ch_data->t_q[0] = ch_data->t_env[0];
ch_data->t_q[ch_data->bs_num_noise] = ch_data->t_env[ch_data->bs_num_env];
if (ch_data->bs_num_noise > 1) {
int idx;
unsigned int idx;
if (ch_data->bs_frame_class == FIXFIX) {
idx = ch_data->bs_num_env >> 1;
} else if (ch_data->bs_frame_class & 1) { // FIXVAR or VARVAR
idx = ch_data->bs_num_env - FFMAX(bs_pointer - 1, 1);
idx = ch_data->bs_num_env - FFMAX((int)bs_pointer - 1, 1);
} else { // VARFIX
if (!bs_pointer)
idx = 1;
@@ -1019,8 +1012,6 @@ 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);
@@ -1697,12 +1688,6 @@ 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];
@@ -1726,7 +1711,6 @@ 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,

View File

@@ -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;
@@ -203,9 +200,9 @@ int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
if (band >= AC3_CRITICAL_BANDS || dba_lengths[seg] > AC3_CRITICAL_BANDS-band)
return -1;
if (dba_values[seg] >= 4) {
delta = (dba_values[seg] - 3) * 128;
delta = (dba_values[seg] - 3) << 7;
} else {
delta = (dba_values[seg] - 4) * 128;
delta = (dba_values[seg] - 4) << 7;
}
for (i = 0; i < dba_lengths[seg]; i++) {
mask[band++] += delta;

View File

@@ -140,8 +140,8 @@ int avpriv_ac3_parse_header2(GetBitContext *gbc, AC3HeaderInfo **phdr)
hdr->channel_mode = get_bits(gbc, 3);
hdr->lfe_on = get_bits1(gbc);
hdr->bit_rate = 8LL * hdr->frame_size * hdr->sample_rate /
(hdr->num_blocks * 256);
hdr->bit_rate = (uint32_t)(8.0 * hdr->frame_size * hdr->sample_rate /
(hdr->num_blocks * 256.0));
hdr->channels = ff_ac3_channels_tab[hdr->channel_mode] + hdr->lfe_on;
}
hdr->channel_layout = avpriv_ac3_channel_layout_tab[hdr->channel_mode];

View File

@@ -112,7 +112,7 @@ static const uint8_t ac3_default_coeffs[8][5][2] = {
static inline int
symmetric_dequant(int code, int levels)
{
return ((code - (levels >> 1)) * (1 << 24)) / levels;
return ((code - (levels >> 1)) << 24) / levels;
}
/*
@@ -196,10 +196,10 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
s->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
#else
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&s->fmt_conv, avctx);
#endif
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
if (USE_FIXED)
@@ -470,7 +470,7 @@ static void calc_transform_coeffs_cpl(AC3DecodeContext *s)
int cpl_coord = s->cpl_coords[ch][band] << 5;
for (bin = band_start; bin < band_end; bin++) {
s->fixed_coeffs[ch][bin] =
MULH(s->fixed_coeffs[CPL_CH][bin] * (1 << 4), cpl_coord);
MULH(s->fixed_coeffs[CPL_CH][bin] << 4, cpl_coord);
}
if (ch == 2 && s->phase_flags[band]) {
for (bin = band_start; bin < band_end; bin++)
@@ -567,7 +567,8 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
av_log(s->avctx, AV_LOG_ERROR, "bap %d is invalid in plain AC-3\n", bap);
bap = 15;
}
mantissa = (unsigned)get_sbits(gbc, quantization_tab[bap]) << (24 - quantization_tab[bap]);
mantissa = get_sbits(gbc, quantization_tab[bap]);
mantissa <<= 24 - quantization_tab[bap];
break;
}
coeffs[freq] = mantissa >> exps[freq];
@@ -601,7 +602,7 @@ static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch,
/* if AHT is used, mantissas for all blocks are encoded in the first
block of the frame. */
int bin;
if (CONFIG_EAC3_DECODER && !blk)
if (!blk && CONFIG_EAC3_DECODER)
ff_eac3_decode_transform_coeffs_aht_ch(s, ch);
for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin];
@@ -923,13 +924,14 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
bin = s->spx_src_start_freq;
for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
int bandsize = s->spx_band_sizes[bnd];
int bandsize;
int spx_coord_exp, spx_coord_mant;
INTFLOAT nratio, sblend, nblend;
#if USE_FIXED
int64_t accu;
/* calculate blending factors */
int64_t accu = ((bin << 23) + (bandsize << 22))
* (int64_t)s->spx_dst_end_freq;
bandsize = s->spx_band_sizes[bnd];
accu = (int64_t)((bin << 23) + (bandsize << 22)) * s->spx_dst_end_freq;
nratio = (int)(accu >> 32);
nratio -= spx_blend << 18;
@@ -949,6 +951,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
float spx_coord;
/* calculate blending factors */
bandsize = s->spx_band_sizes[bnd];
nratio = ((float)((bin + (bandsize >> 1))) / s->spx_dst_end_freq) - spx_blend;
nratio = av_clipf(nratio, 0.0f, 1.0f);
nblend = sqrtf(3.0f * nratio); // noise is scaled by sqrt(3)
@@ -1352,7 +1355,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
}
/* apply spectral extension to high frequency bins */
if (CONFIG_EAC3_DECODER && s->spx_in_use) {
if (s->spx_in_use && CONFIG_EAC3_DECODER) {
ff_eac3_apply_spectral_extension(s);
}

View File

@@ -36,7 +36,6 @@
#include "libavutil/internal.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "me_cmp.h"
#include "put_bits.h"
#include "audiodsp.h"
@@ -1688,43 +1687,43 @@ static void dprint_options(AC3EncodeContext *s)
case 16: av_strlcpy(strbuf, "E-AC-3 (enhanced)", 32); break;
default: snprintf(strbuf, 32, "ERROR");
}
ff_dlog(avctx, "bitstream_id: %s (%d)\n", strbuf, s->bitstream_id);
ff_dlog(avctx, "sample_fmt: %s\n", av_get_sample_fmt_name(avctx->sample_fmt));
av_dlog(avctx, "bitstream_id: %s (%d)\n", strbuf, s->bitstream_id);
av_dlog(avctx, "sample_fmt: %s\n", av_get_sample_fmt_name(avctx->sample_fmt));
av_get_channel_layout_string(strbuf, 32, s->channels, avctx->channel_layout);
ff_dlog(avctx, "channel_layout: %s\n", strbuf);
ff_dlog(avctx, "sample_rate: %d\n", s->sample_rate);
ff_dlog(avctx, "bit_rate: %d\n", s->bit_rate);
ff_dlog(avctx, "blocks/frame: %d (code=%d)\n", s->num_blocks, s->num_blks_code);
av_dlog(avctx, "channel_layout: %s\n", strbuf);
av_dlog(avctx, "sample_rate: %d\n", s->sample_rate);
av_dlog(avctx, "bit_rate: %d\n", s->bit_rate);
av_dlog(avctx, "blocks/frame: %d (code=%d)\n", s->num_blocks, s->num_blks_code);
if (s->cutoff)
ff_dlog(avctx, "cutoff: %d\n", s->cutoff);
av_dlog(avctx, "cutoff: %d\n", s->cutoff);
ff_dlog(avctx, "per_frame_metadata: %s\n",
av_dlog(avctx, "per_frame_metadata: %s\n",
opt->allow_per_frame_metadata?"on":"off");
if (s->has_center)
ff_dlog(avctx, "center_mixlev: %0.3f (%d)\n", opt->center_mix_level,
av_dlog(avctx, "center_mixlev: %0.3f (%d)\n", opt->center_mix_level,
s->center_mix_level);
else
ff_dlog(avctx, "center_mixlev: {not written}\n");
av_dlog(avctx, "center_mixlev: {not written}\n");
if (s->has_surround)
ff_dlog(avctx, "surround_mixlev: %0.3f (%d)\n", opt->surround_mix_level,
av_dlog(avctx, "surround_mixlev: %0.3f (%d)\n", opt->surround_mix_level,
s->surround_mix_level);
else
ff_dlog(avctx, "surround_mixlev: {not written}\n");
av_dlog(avctx, "surround_mixlev: {not written}\n");
if (opt->audio_production_info) {
ff_dlog(avctx, "mixing_level: %ddB\n", opt->mixing_level);
av_dlog(avctx, "mixing_level: %ddB\n", opt->mixing_level);
switch (opt->room_type) {
case AC3ENC_OPT_NOT_INDICATED: av_strlcpy(strbuf, "notindicated", 32); break;
case AC3ENC_OPT_LARGE_ROOM: av_strlcpy(strbuf, "large", 32); break;
case AC3ENC_OPT_SMALL_ROOM: av_strlcpy(strbuf, "small", 32); break;
default: snprintf(strbuf, 32, "ERROR (%d)", opt->room_type);
}
ff_dlog(avctx, "room_type: %s\n", strbuf);
av_dlog(avctx, "room_type: %s\n", strbuf);
} else {
ff_dlog(avctx, "mixing_level: {not written}\n");
ff_dlog(avctx, "room_type: {not written}\n");
av_dlog(avctx, "mixing_level: {not written}\n");
av_dlog(avctx, "room_type: {not written}\n");
}
ff_dlog(avctx, "copyright: %s\n", opt->copyright?"on":"off");
ff_dlog(avctx, "dialnorm: %ddB\n", opt->dialogue_level);
av_dlog(avctx, "copyright: %s\n", opt->copyright?"on":"off");
av_dlog(avctx, "dialnorm: %ddB\n", opt->dialogue_level);
if (s->channel_mode == AC3_CHMODE_STEREO) {
switch (opt->dolby_surround_mode) {
case AC3ENC_OPT_NOT_INDICATED: av_strlcpy(strbuf, "notindicated", 32); break;
@@ -1732,11 +1731,11 @@ static void dprint_options(AC3EncodeContext *s)
case AC3ENC_OPT_MODE_OFF: av_strlcpy(strbuf, "off", 32); break;
default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_mode);
}
ff_dlog(avctx, "dsur_mode: %s\n", strbuf);
av_dlog(avctx, "dsur_mode: %s\n", strbuf);
} else {
ff_dlog(avctx, "dsur_mode: {not written}\n");
av_dlog(avctx, "dsur_mode: {not written}\n");
}
ff_dlog(avctx, "original: %s\n", opt->original?"on":"off");
av_dlog(avctx, "original: %s\n", opt->original?"on":"off");
if (s->bitstream_id == 6) {
if (opt->extended_bsi_1) {
@@ -1746,17 +1745,17 @@ static void dprint_options(AC3EncodeContext *s)
case AC3ENC_OPT_DOWNMIX_LORO: av_strlcpy(strbuf, "loro", 32); break;
default: snprintf(strbuf, 32, "ERROR (%d)", opt->preferred_stereo_downmix);
}
ff_dlog(avctx, "dmix_mode: %s\n", strbuf);
ff_dlog(avctx, "ltrt_cmixlev: %0.3f (%d)\n",
av_dlog(avctx, "dmix_mode: %s\n", strbuf);
av_dlog(avctx, "ltrt_cmixlev: %0.3f (%d)\n",
opt->ltrt_center_mix_level, s->ltrt_center_mix_level);
ff_dlog(avctx, "ltrt_surmixlev: %0.3f (%d)\n",
av_dlog(avctx, "ltrt_surmixlev: %0.3f (%d)\n",
opt->ltrt_surround_mix_level, s->ltrt_surround_mix_level);
ff_dlog(avctx, "loro_cmixlev: %0.3f (%d)\n",
av_dlog(avctx, "loro_cmixlev: %0.3f (%d)\n",
opt->loro_center_mix_level, s->loro_center_mix_level);
ff_dlog(avctx, "loro_surmixlev: %0.3f (%d)\n",
av_dlog(avctx, "loro_surmixlev: %0.3f (%d)\n",
opt->loro_surround_mix_level, s->loro_surround_mix_level);
} else {
ff_dlog(avctx, "extended bitstream info 1: {not written}\n");
av_dlog(avctx, "extended bitstream info 1: {not written}\n");
}
if (opt->extended_bsi_2) {
switch (opt->dolby_surround_ex_mode) {
@@ -1765,23 +1764,23 @@ static void dprint_options(AC3EncodeContext *s)
case AC3ENC_OPT_MODE_OFF: av_strlcpy(strbuf, "off", 32); break;
default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_ex_mode);
}
ff_dlog(avctx, "dsurex_mode: %s\n", strbuf);
av_dlog(avctx, "dsurex_mode: %s\n", strbuf);
switch (opt->dolby_headphone_mode) {
case AC3ENC_OPT_NOT_INDICATED: av_strlcpy(strbuf, "notindicated", 32); break;
case AC3ENC_OPT_MODE_ON: av_strlcpy(strbuf, "on", 32); break;
case AC3ENC_OPT_MODE_OFF: av_strlcpy(strbuf, "off", 32); break;
default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_headphone_mode);
}
ff_dlog(avctx, "dheadphone_mode: %s\n", strbuf);
av_dlog(avctx, "dheadphone_mode: %s\n", strbuf);
switch (opt->ad_converter_type) {
case AC3ENC_OPT_ADCONV_STANDARD: av_strlcpy(strbuf, "standard", 32); break;
case AC3ENC_OPT_ADCONV_HDCD: av_strlcpy(strbuf, "hdcd", 32); break;
default: snprintf(strbuf, 32, "ERROR (%d)", opt->ad_converter_type);
}
ff_dlog(avctx, "ad_conv_type: %s\n", strbuf);
av_dlog(avctx, "ad_conv_type: %s\n", strbuf);
} else {
ff_dlog(avctx, "extended bitstream info 2: {not written}\n");
av_dlog(avctx, "extended bitstream info 2: {not written}\n");
}
}
#endif

View File

@@ -22,7 +22,6 @@
#include <inttypes.h>
#include "libavutil/avassert.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
@@ -241,7 +240,6 @@ void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
float y = in->y[i] * scale;
if (in->pitch_lag > 0)
av_assert0(x < size);
do {
out[x] += y;
y *= in->pitch_fac;

View File

@@ -197,7 +197,7 @@ static inline int16_t adpcm_ima_wav_expand_nibble(ADPCMChannelStatus *c, GetBitC
step_index = av_clip(step_index, 0, 88);
sign = nibble & (1 << shift);
delta = av_mod_uintp2(nibble, shift);
delta = nibble & ((1 << shift) - 1);
diff = ((2 * delta + 1) * step) >> shift;
predictor = c->predictor;
if (sign) predictor -= diff;
@@ -578,8 +578,6 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
case AV_CODEC_ID_ADPCM_IMA_DK4:
if (avctx->block_align > 0)
buf_size = FFMIN(buf_size, avctx->block_align);
if (buf_size < 4 * ch)
return AVERROR_INVALIDDATA;
nb_samples = 1 + (buf_size - 4 * ch) * 2 / ch;
break;
case AV_CODEC_ID_ADPCM_IMA_RAD:
@@ -593,15 +591,13 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
int bsamples = ff_adpcm_ima_block_samples[avctx->bits_per_coded_sample - 2];
if (avctx->block_align > 0)
buf_size = FFMIN(buf_size, avctx->block_align);
if (buf_size < 4 * ch)
return AVERROR_INVALIDDATA;
nb_samples = 1 + (buf_size - 4 * ch) / (bsize * ch) * bsamples;
break;
}
case AV_CODEC_ID_ADPCM_MS:
if (avctx->block_align > 0)
buf_size = FFMIN(buf_size, avctx->block_align);
nb_samples = (buf_size - 6 * ch) * 2 / ch;
nb_samples = 2 + (buf_size - 7 * ch) * 2 / ch;
break;
case AV_CODEC_ID_ADPCM_SBPRO_2:
case AV_CODEC_ID_ADPCM_SBPRO_3:
@@ -614,8 +610,6 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
case AV_CODEC_ID_ADPCM_SBPRO_4: samples_per_byte = 2; break;
}
if (!s->status[0].step_index) {
if (buf_size < ch)
return AVERROR_INVALIDDATA;
nb_samples++;
buf_size -= ch;
}
@@ -1504,7 +1498,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
prev = 0;
}
prev = av_clip_intp2((prev + 0x20) >> 6, 21);
prev = av_clip((prev + 0x20) >> 6, -0x200000, 0x1fffff);
byte = bytestream2_get_byteu(&gb);
if (!channel)
@@ -1534,11 +1528,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
*got_frame_ptr = 1;
if (avpkt->size < bytestream2_tell(&gb)) {
av_log(avctx, AV_LOG_ERROR, "Overread of %d < %d\n", avpkt->size, bytestream2_tell(&gb));
return avpkt->size;
}
return bytestream2_tell(&gb);
}

View File

@@ -316,12 +316,6 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
int lpc_quant[2];
int rice_history_mult[2];
if (!alac->rice_limit) {
avpriv_request_sample(alac->avctx,
"Compression with rice limit 0");
return AVERROR(ENOSYS);
}
decorr_shift = get_bits(&alac->gb, 8);
decorr_left_weight = get_bits(&alac->gb, 8);
@@ -534,12 +528,6 @@ static int allocate_buffers(ALACContext *alac)
int ch;
int buf_size = alac->max_samples_per_frame * sizeof(int32_t);
for (ch = 0; ch < 2; ch++) {
alac->predict_error_buffer[ch] = NULL;
alac->output_samples_buffer[ch] = NULL;
alac->extra_bits_buffer[ch] = NULL;
}
for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],
buf_size, buf_alloc_fail);

View File

@@ -76,30 +76,24 @@ void avcodec_register_all(void)
/* hardware accelerators */
REGISTER_HWACCEL(H263_VAAPI, h263_vaapi);
REGISTER_HWACCEL(H263_VDPAU, h263_vdpau);
REGISTER_HWACCEL(H264_D3D11VA, h264_d3d11va);
REGISTER_HWACCEL(H264_DXVA2, h264_dxva2);
REGISTER_HWACCEL(H264_MMAL, h264_mmal);
REGISTER_HWACCEL(H264_QSV, h264_qsv);
REGISTER_HWACCEL(H264_VAAPI, h264_vaapi);
REGISTER_HWACCEL(H264_VDA, h264_vda);
REGISTER_HWACCEL(H264_VDA_OLD, h264_vda_old);
REGISTER_HWACCEL(H264_VDPAU, h264_vdpau);
REGISTER_HWACCEL(HEVC_D3D11VA, hevc_d3d11va);
REGISTER_HWACCEL(HEVC_DXVA2, hevc_dxva2);
REGISTER_HWACCEL(MPEG1_XVMC, mpeg1_xvmc);
REGISTER_HWACCEL(MPEG1_VDPAU, mpeg1_vdpau);
REGISTER_HWACCEL(MPEG2_XVMC, mpeg2_xvmc);
REGISTER_HWACCEL(MPEG2_D3D11VA, mpeg2_d3d11va);
REGISTER_HWACCEL(MPEG2_DXVA2, mpeg2_dxva2);
REGISTER_HWACCEL(MPEG2_VAAPI, mpeg2_vaapi);
REGISTER_HWACCEL(MPEG2_VDPAU, mpeg2_vdpau);
REGISTER_HWACCEL(MPEG4_VAAPI, mpeg4_vaapi);
REGISTER_HWACCEL(MPEG4_VDPAU, mpeg4_vdpau);
REGISTER_HWACCEL(VC1_D3D11VA, vc1_d3d11va);
REGISTER_HWACCEL(VC1_DXVA2, vc1_dxva2);
REGISTER_HWACCEL(VC1_VAAPI, vc1_vaapi);
REGISTER_HWACCEL(VC1_VDPAU, vc1_vdpau);
REGISTER_HWACCEL(WMV3_D3D11VA, wmv3_d3d11va);
REGISTER_HWACCEL(WMV3_DXVA2, wmv3_dxva2);
REGISTER_HWACCEL(WMV3_VAAPI, wmv3_vaapi);
REGISTER_HWACCEL(WMV3_VDPAU, wmv3_vdpau);
@@ -113,7 +107,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (AMV, amv);
REGISTER_DECODER(ANM, anm);
REGISTER_DECODER(ANSI, ansi);
REGISTER_ENCDEC (APNG, apng);
REGISTER_DECODER(APNG, apng);
REGISTER_ENCDEC (ASV1, asv1);
REGISTER_ENCDEC (ASV2, asv2);
REGISTER_DECODER(AURA, aura);
@@ -177,13 +171,11 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (H263P, h263p);
REGISTER_DECODER(H264, h264);
REGISTER_DECODER(H264_CRYSTALHD, h264_crystalhd);
REGISTER_DECODER(H264_MMAL, h264_mmal);
REGISTER_DECODER(H264_QSV, h264_qsv);
REGISTER_DECODER(H264_VDA, h264_vda);
REGISTER_DECODER(H264_VDPAU, h264_vdpau);
REGISTER_DECODER(HEVC, hevc);
REGISTER_DECODER(HNM4_VIDEO, hnm4_video);
REGISTER_DECODER(HQ_HQA, hq_hqa);
REGISTER_DECODER(HQX, hqx);
REGISTER_ENCDEC (HUFFYUV, huffyuv);
REGISTER_DECODER(IDCIN, idcin);
@@ -235,6 +227,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(MVC2, mvc2);
REGISTER_DECODER(MXPEG, mxpeg);
REGISTER_DECODER(NUV, nuv);
REGISTER_ENCODER(NVENC, nvenc);
REGISTER_DECODER(PAF_VIDEO, paf_video);
REGISTER_ENCDEC (PAM, pam);
REGISTER_ENCDEC (PBM, pbm);
@@ -276,7 +269,6 @@ void avcodec_register_all(void)
REGISTER_DECODER(SVQ3, svq3);
REGISTER_ENCDEC (TARGA, targa);
REGISTER_DECODER(TARGA_Y216, targa_y216);
REGISTER_DECODER(TDSC, tdsc);
REGISTER_DECODER(THEORA, theora);
REGISTER_DECODER(THP, thp);
REGISTER_DECODER(TIERTEXSEQVIDEO, tiertexseqvideo);
@@ -519,7 +511,6 @@ void avcodec_register_all(void)
/* external libraries */
REGISTER_DECODER(LIBCELT, libcelt);
REGISTER_DECODER(LIBDCADEC, libdcadec)
REGISTER_ENCODER(LIBFAAC, libfaac);
REGISTER_ENCDEC (LIBFDK_AAC, libfdk_aac);
REGISTER_ENCDEC (LIBGSM, libgsm);
@@ -543,7 +534,6 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (LIBVPX_VP8, libvpx_vp8);
REGISTER_ENCDEC (LIBVPX_VP9, libvpx_vp9);
REGISTER_ENCODER(LIBWAVPACK, libwavpack);
REGISTER_ENCODER(LIBWEBP_ANIM, libwebp_anim); /* preferred over libwebp */
REGISTER_ENCODER(LIBWEBP, libwebp);
REGISTER_ENCODER(LIBX264, libx264);
REGISTER_ENCODER(LIBX264RGB, libx264rgb);
@@ -552,20 +542,13 @@ void avcodec_register_all(void)
REGISTER_ENCODER(LIBXVID, libxvid);
REGISTER_DECODER(LIBZVBI_TELETEXT, libzvbi_teletext);
REGISTER_ENCODER(LIBAACPLUS, libaacplus);
REGISTER_ENCODER(LIBOPENH264, libopenh264);
/* text */
REGISTER_DECODER(BINTEXT, bintext);
REGISTER_DECODER(XBIN, xbin);
REGISTER_DECODER(IDF, idf);
/* external libraries, that shouldn't be used by default if one of the
* above is available */
REGISTER_ENCODER(LIBOPENH264, libopenh264);
REGISTER_ENCODER(H264_QSV, h264_qsv);
REGISTER_ENCODER(NVENC, nvenc);
REGISTER_ENCODER(NVENC_H264, nvenc_h264);
REGISTER_ENCODER(NVENC_HEVC, nvenc_hevc);
/* parsers */
REGISTER_PARSER(AAC, aac);
REGISTER_PARSER(AAC_LATM, aac_latm);
@@ -613,7 +596,6 @@ void avcodec_register_all(void)
REGISTER_BSF(MJPEG2JPEG, mjpeg2jpeg);
REGISTER_BSF(MJPEGA_DUMP_HEADER, mjpega_dump_header);
REGISTER_BSF(MP3_HEADER_DECOMPRESS, mp3_header_decompress);
REGISTER_BSF(MPEG4_UNPACK_BFRAMES, mpeg4_unpack_bframes);
REGISTER_BSF(MOV2TEXTSUB, mov2textsub);
REGISTER_BSF(NOISE, noise);
REGISTER_BSF(REMOVE_EXTRADATA, remove_extradata);

View File

@@ -253,24 +253,24 @@ static av_cold void dprint_specific_config(ALSDecContext *ctx)
AVCodecContext *avctx = ctx->avctx;
ALSSpecificConfig *sconf = &ctx->sconf;
ff_dlog(avctx, "resolution = %i\n", sconf->resolution);
ff_dlog(avctx, "floating = %i\n", sconf->floating);
ff_dlog(avctx, "frame_length = %i\n", sconf->frame_length);
ff_dlog(avctx, "ra_distance = %i\n", sconf->ra_distance);
ff_dlog(avctx, "ra_flag = %i\n", sconf->ra_flag);
ff_dlog(avctx, "adapt_order = %i\n", sconf->adapt_order);
ff_dlog(avctx, "coef_table = %i\n", sconf->coef_table);
ff_dlog(avctx, "long_term_prediction = %i\n", sconf->long_term_prediction);
ff_dlog(avctx, "max_order = %i\n", sconf->max_order);
ff_dlog(avctx, "block_switching = %i\n", sconf->block_switching);
ff_dlog(avctx, "bgmc = %i\n", sconf->bgmc);
ff_dlog(avctx, "sb_part = %i\n", sconf->sb_part);
ff_dlog(avctx, "joint_stereo = %i\n", sconf->joint_stereo);
ff_dlog(avctx, "mc_coding = %i\n", sconf->mc_coding);
ff_dlog(avctx, "chan_config = %i\n", sconf->chan_config);
ff_dlog(avctx, "chan_sort = %i\n", sconf->chan_sort);
ff_dlog(avctx, "RLSLMS = %i\n", sconf->rlslms);
ff_dlog(avctx, "chan_config_info = %i\n", sconf->chan_config_info);
av_dlog(avctx, "resolution = %i\n", sconf->resolution);
av_dlog(avctx, "floating = %i\n", sconf->floating);
av_dlog(avctx, "frame_length = %i\n", sconf->frame_length);
av_dlog(avctx, "ra_distance = %i\n", sconf->ra_distance);
av_dlog(avctx, "ra_flag = %i\n", sconf->ra_flag);
av_dlog(avctx, "adapt_order = %i\n", sconf->adapt_order);
av_dlog(avctx, "coef_table = %i\n", sconf->coef_table);
av_dlog(avctx, "long_term_prediction = %i\n", sconf->long_term_prediction);
av_dlog(avctx, "max_order = %i\n", sconf->max_order);
av_dlog(avctx, "block_switching = %i\n", sconf->block_switching);
av_dlog(avctx, "bgmc = %i\n", sconf->bgmc);
av_dlog(avctx, "sb_part = %i\n", sconf->sb_part);
av_dlog(avctx, "joint_stereo = %i\n", sconf->joint_stereo);
av_dlog(avctx, "mc_coding = %i\n", sconf->mc_coding);
av_dlog(avctx, "chan_config = %i\n", sconf->chan_config);
av_dlog(avctx, "chan_sort = %i\n", sconf->chan_sort);
av_dlog(avctx, "RLSLMS = %i\n", sconf->rlslms);
av_dlog(avctx, "chan_config_info = %i\n", sconf->chan_config_info);
#endif
}
@@ -352,20 +352,16 @@ static av_cold int read_specific_config(ALSDecContext *ctx)
if (get_bits_left(&gb) < bits_needed)
return AVERROR_INVALIDDATA;
if (!(sconf->chan_pos = av_malloc_array(avctx->channels, sizeof(*sconf->chan_pos))))
if (!(sconf->chan_pos = av_malloc(avctx->channels * sizeof(*sconf->chan_pos))))
return AVERROR(ENOMEM);
ctx->cs_switch = 1;
for (i = 0; i < avctx->channels; i++) {
sconf->chan_pos[i] = -1;
}
for (i = 0; i < avctx->channels; i++) {
int idx;
idx = get_bits(&gb, chan_pos_bits);
if (idx >= avctx->channels || sconf->chan_pos[idx] != -1) {
if (idx >= avctx->channels) {
av_log(avctx, AV_LOG_WARNING, "Invalid channel reordering.\n");
ctx->cs_switch = 0;
break;
@@ -682,7 +678,7 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
if (!sconf->rlslms) {
if (sconf->adapt_order && sconf->max_order) {
if (sconf->adapt_order) {
int opt_order_length = av_ceil_log2(av_clip((bd->block_length >> 3) - 1,
2, sconf->max_order + 1));
*bd->opt_order = get_bits(gb, opt_order_length);
@@ -1246,7 +1242,6 @@ static int revert_channel_correlation(ALSDecContext *ctx, ALSBlockData *bd,
ALSChannelData *ch = cd[c];
unsigned int dep = 0;
unsigned int channels = ctx->avctx->channels;
unsigned int channel_size = ctx->sconf.frame_length + ctx->sconf.max_order;
if (reverted[c])
return 0;
@@ -1277,9 +1272,9 @@ static int revert_channel_correlation(ALSDecContext *ctx, ALSBlockData *bd,
bd->raw_samples = ctx->raw_samples[c] + offset;
for (dep = 0; !ch[dep].stop_flag; dep++) {
ptrdiff_t smp;
ptrdiff_t begin = 1;
ptrdiff_t end = bd->block_length - 1;
unsigned int smp;
unsigned int begin = 1;
unsigned int end = bd->block_length - 1;
int64_t y;
int32_t *master = ctx->raw_samples[ch[dep].master_channel] + offset;
@@ -1291,28 +1286,11 @@ static int revert_channel_correlation(ALSDecContext *ctx, ALSBlockData *bd,
if (ch[dep].time_diff_sign) {
t = -t;
if (begin < t) {
av_log(ctx->avctx, AV_LOG_ERROR, "begin %td smaller than time diff index %d.\n", begin, t);
return AVERROR_INVALIDDATA;
}
begin -= t;
} else {
if (end < t) {
av_log(ctx->avctx, AV_LOG_ERROR, "end %td smaller than time diff index %d.\n", end, t);
return AVERROR_INVALIDDATA;
}
end -= t;
}
if (FFMIN(begin - 1, begin - 1 + t) < ctx->raw_buffer - master ||
FFMAX(end + 1, end + 1 + t) > ctx->raw_buffer + channels * channel_size - master) {
av_log(ctx->avctx, AV_LOG_ERROR,
"sample pointer range [%p, %p] not contained in raw_buffer [%p, %p].\n",
master + FFMIN(begin - 1, begin - 1 + t), master + FFMAX(end + 1, end + 1 + t),
ctx->raw_buffer, ctx->raw_buffer + channels * channel_size);
return AVERROR_INVALIDDATA;
}
for (smp = begin; smp < end; smp++) {
y = (1 << 6) +
MUL64(ch[dep].weighting[0], master[smp - 1 ]) +
@@ -1325,16 +1303,6 @@ static int revert_channel_correlation(ALSDecContext *ctx, ALSBlockData *bd,
bd->raw_samples[smp] += y >> 7;
}
} else {
if (begin - 1 < ctx->raw_buffer - master ||
end + 1 > ctx->raw_buffer + channels * channel_size - master) {
av_log(ctx->avctx, AV_LOG_ERROR,
"sample pointer range [%p, %p] not contained in raw_buffer [%p, %p].\n",
master + begin - 1, master + end + 1,
ctx->raw_buffer, ctx->raw_buffer + channels * channel_size);
return AVERROR_INVALIDDATA;
}
for (smp = begin; smp < end; smp++) {
y = (1 << 6) +
MUL64(ch[dep].weighting[0], master[smp - 1]) +
@@ -1493,11 +1461,6 @@ static int read_frame_data(ALSDecContext *ctx, unsigned int ra_frame)
// TODO: read_diff_float_data
if (get_bits_left(gb) < 0) {
av_log(ctx->avctx, AV_LOG_ERROR, "Overread %d\n", -get_bits_left(gb));
return AVERROR_INVALIDDATA;
}
return 0;
}
@@ -1703,12 +1666,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
avctx->sample_fmt = sconf->resolution > 1
? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S16;
avctx->bits_per_raw_sample = (sconf->resolution + 1) * 8;
if (avctx->bits_per_raw_sample > 32) {
av_log(avctx, AV_LOG_ERROR, "Bits per raw sample %d larger than 32.\n",
avctx->bits_per_raw_sample);
ret = AVERROR_INVALIDDATA;
goto fail;
}
}
// set maximum Rice parameter for progressive decoding based on resolution
@@ -1723,14 +1680,14 @@ static av_cold int decode_init(AVCodecContext *avctx)
// allocate quantized parcor coefficient buffer
num_buffers = sconf->mc_coding ? avctx->channels : 1;
ctx->quant_cof = av_malloc_array(num_buffers, sizeof(*ctx->quant_cof));
ctx->lpc_cof = av_malloc_array(num_buffers, sizeof(*ctx->lpc_cof));
ctx->quant_cof_buffer = av_malloc_array(num_buffers * sconf->max_order,
sizeof(*ctx->quant_cof_buffer));
ctx->lpc_cof_buffer = av_malloc_array(num_buffers * sconf->max_order,
sizeof(*ctx->lpc_cof_buffer));
ctx->lpc_cof_reversed_buffer = av_malloc_array(sconf->max_order,
sizeof(*ctx->lpc_cof_buffer));
ctx->quant_cof = av_malloc(sizeof(*ctx->quant_cof) * num_buffers);
ctx->lpc_cof = av_malloc(sizeof(*ctx->lpc_cof) * num_buffers);
ctx->quant_cof_buffer = av_malloc(sizeof(*ctx->quant_cof_buffer) *
num_buffers * sconf->max_order);
ctx->lpc_cof_buffer = av_malloc(sizeof(*ctx->lpc_cof_buffer) *
num_buffers * sconf->max_order);
ctx->lpc_cof_reversed_buffer = av_malloc(sizeof(*ctx->lpc_cof_buffer) *
sconf->max_order);
if (!ctx->quant_cof || !ctx->lpc_cof ||
!ctx->quant_cof_buffer || !ctx->lpc_cof_buffer ||
@@ -1747,14 +1704,15 @@ static av_cold int decode_init(AVCodecContext *avctx)
}
// allocate and assign lag and gain data buffer for ltp mode
ctx->const_block = av_malloc_array(num_buffers, sizeof(*ctx->const_block));
ctx->shift_lsbs = av_malloc_array(num_buffers, sizeof(*ctx->shift_lsbs));
ctx->opt_order = av_malloc_array(num_buffers, sizeof(*ctx->opt_order));
ctx->store_prev_samples = av_malloc_array(num_buffers, sizeof(*ctx->store_prev_samples));
ctx->use_ltp = av_mallocz_array(num_buffers, sizeof(*ctx->use_ltp));
ctx->ltp_lag = av_malloc_array(num_buffers, sizeof(*ctx->ltp_lag));
ctx->ltp_gain = av_malloc_array(num_buffers, sizeof(*ctx->ltp_gain));
ctx->ltp_gain_buffer = av_malloc_array(num_buffers * 5, sizeof(*ctx->ltp_gain_buffer));
ctx->const_block = av_malloc (sizeof(*ctx->const_block) * num_buffers);
ctx->shift_lsbs = av_malloc (sizeof(*ctx->shift_lsbs) * num_buffers);
ctx->opt_order = av_malloc (sizeof(*ctx->opt_order) * num_buffers);
ctx->store_prev_samples = av_malloc(sizeof(*ctx->store_prev_samples) * num_buffers);
ctx->use_ltp = av_mallocz(sizeof(*ctx->use_ltp) * num_buffers);
ctx->ltp_lag = av_malloc (sizeof(*ctx->ltp_lag) * num_buffers);
ctx->ltp_gain = av_malloc (sizeof(*ctx->ltp_gain) * num_buffers);
ctx->ltp_gain_buffer = av_malloc (sizeof(*ctx->ltp_gain_buffer) *
num_buffers * 5);
if (!ctx->const_block || !ctx->shift_lsbs ||
!ctx->opt_order || !ctx->store_prev_samples ||
@@ -1770,12 +1728,12 @@ static av_cold int decode_init(AVCodecContext *avctx)
// allocate and assign channel data buffer for mcc mode
if (sconf->mc_coding) {
ctx->chan_data_buffer = av_mallocz_array(num_buffers * num_buffers,
sizeof(*ctx->chan_data_buffer));
ctx->chan_data = av_mallocz_array(num_buffers,
sizeof(*ctx->chan_data));
ctx->reverted_channels = av_malloc_array(num_buffers,
sizeof(*ctx->reverted_channels));
ctx->chan_data_buffer = av_malloc(sizeof(*ctx->chan_data_buffer) *
num_buffers * num_buffers);
ctx->chan_data = av_malloc(sizeof(*ctx->chan_data) *
num_buffers);
ctx->reverted_channels = av_malloc(sizeof(*ctx->reverted_channels) *
num_buffers);
if (!ctx->chan_data_buffer || !ctx->chan_data || !ctx->reverted_channels) {
av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
@@ -1793,9 +1751,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
channel_size = sconf->frame_length + sconf->max_order;
ctx->prev_raw_samples = av_malloc_array(sconf->max_order, sizeof(*ctx->prev_raw_samples));
ctx->raw_buffer = av_mallocz_array(avctx->channels * channel_size, sizeof(*ctx->raw_buffer));
ctx->raw_samples = av_malloc_array(avctx->channels, sizeof(*ctx->raw_samples));
ctx->prev_raw_samples = av_malloc (sizeof(*ctx->prev_raw_samples) * sconf->max_order);
ctx->raw_buffer = av_mallocz(sizeof(*ctx-> raw_buffer) * avctx->channels * channel_size);
ctx->raw_samples = av_malloc (sizeof(*ctx-> raw_samples) * avctx->channels);
// allocate previous raw sample buffer
if (!ctx->prev_raw_samples || !ctx->raw_buffer|| !ctx->raw_samples) {
@@ -1812,10 +1770,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
// allocate crc buffer
if (HAVE_BIGENDIAN != sconf->msb_first && sconf->crc_enabled &&
(avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_CAREFUL))) {
ctx->crc_buffer = av_malloc_array(ctx->cur_frame_length *
avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt),
sizeof(*ctx->crc_buffer));
ctx->crc_buffer = av_malloc(sizeof(*ctx->crc_buffer) *
ctx->cur_frame_length *
avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt));
if (!ctx->crc_buffer) {
av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
ret = AVERROR(ENOMEM);

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@@ -399,17 +399,17 @@ static void decode_pitch_vector(AMRContext *p,
decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
amr_subframe->p_lag, p->pitch_lag_int,
subframe);
} else {
} else
ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
amr_subframe->p_lag,
p->pitch_lag_int, subframe,
mode != MODE_4k75 && mode != MODE_5k15,
mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
pitch_lag_frac *= 2;
}
p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
pitch_lag_int += pitch_lag_frac > 0;
/* Calculate the pitch vector by interpolating the past excitation at the

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@@ -592,14 +592,14 @@ static void decode_array_0000(APEContext *ctx, GetBitContext *gb,
int ksummax, ksummin;
rice->ksum = 0;
for (i = 0; i < FFMIN(blockstodecode, 5); i++) {
for (i = 0; i < 5; i++) {
out[i] = get_rice_ook(&ctx->gb, 10);
rice->ksum += out[i];
}
rice->k = av_log2(rice->ksum / 10) + 1;
if (rice->k >= 24)
return;
for (; i < FFMIN(blockstodecode, 64); i++) {
for (; i < 64; i++) {
out[i] = get_rice_ook(&ctx->gb, rice->k);
rice->ksum += out[i];
rice->k = av_log2(rice->ksum / ((i + 1) * 2)) + 1;
@@ -1461,13 +1461,13 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
nblocks);
return AVERROR_INVALIDDATA;
}
s->samples = nblocks;
/* Initialize the frame decoder */
if (init_frame_decoder(s) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error reading frame header\n");
return AVERROR_INVALIDDATA;
}
s->samples = nblocks;
}
if (!s->data) {

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@@ -1,266 +0,0 @@
/*
* Copyright (c) 2015 Ludmila Glinskih
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/*
* FLAC codec test.
* Encodes raw data to FLAC format and decodes it back to raw. Compares raw-data
* after that.
*/
#include "avcodec.h"
#include "libavutil/common.h"
#include "libavutil/samplefmt.h"
#define NUMBER_OF_FRAMES 200
#define NAME_BUFF_SIZE 100
/* generate i-th frame of test audio */
static int generate_raw_frame(uint16_t *frame_data, int i, int sample_rate,
int channels, int frame_size)
{
int j, k;
for (j = 0; j < frame_size; j++) {
frame_data[channels * j] = 10000 * ((j / 10 * i) % 2);
for (k = 1; k < channels; k++)
frame_data[channels * j + k] = frame_data[channels * j] * (k + 1);
}
return 0;
}
static int init_encoder(AVCodec *enc, AVCodecContext **enc_ctx,
int64_t ch_layout, int sample_rate)
{
AVCodecContext *ctx;
int result;
char name_buff[NAME_BUFF_SIZE];
av_get_channel_layout_string(name_buff, NAME_BUFF_SIZE, 0, ch_layout);
av_log(NULL, AV_LOG_INFO, "channel layout: %s, sample rate: %i\n", name_buff, sample_rate);
ctx = avcodec_alloc_context3(enc);
if (!ctx) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate encoder context\n");
return AVERROR(ENOMEM);
}
ctx->sample_fmt = AV_SAMPLE_FMT_S16;
ctx->sample_rate = sample_rate;
ctx->channel_layout = ch_layout;
result = avcodec_open2(ctx, enc, NULL);
if (result < 0) {
av_log(ctx, AV_LOG_ERROR, "Can't open encoder\n");
return result;
}
*enc_ctx = ctx;
return 0;
}
static int init_decoder(AVCodec *dec, AVCodecContext **dec_ctx,
int64_t ch_layout)
{
AVCodecContext *ctx;
int result;
ctx = avcodec_alloc_context3(dec);
if (!ctx) {
av_log(NULL, AV_LOG_ERROR , "Can't allocate decoder context\n");
return AVERROR(ENOMEM);
}
ctx->request_sample_fmt = AV_SAMPLE_FMT_S16;
/* XXX: FLAC ignores it for some reason */
ctx->request_channel_layout = ch_layout;
ctx->channel_layout = ch_layout;
result = avcodec_open2(ctx, dec, NULL);
if (result < 0) {
av_log(ctx, AV_LOG_ERROR, "Can't open decoder\n");
return result;
}
*dec_ctx = ctx;
return 0;
}
static int run_test(AVCodec *enc, AVCodec *dec, AVCodecContext *enc_ctx,
AVCodecContext *dec_ctx)
{
AVPacket enc_pkt;
AVFrame *in_frame, *out_frame;
uint8_t *raw_in = NULL, *raw_out = NULL;
int in_offset = 0, out_offset = 0;
int frame_data_size = 0;
int result = 0;
int got_output = 0;
int i = 0;
in_frame = av_frame_alloc();
if (!in_frame) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate input frame\n");
return AVERROR(ENOMEM);
}
in_frame->nb_samples = enc_ctx->frame_size;
in_frame->format = enc_ctx->sample_fmt;
in_frame->channel_layout = enc_ctx->channel_layout;
if (av_frame_get_buffer(in_frame, 32) != 0) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate a buffer for input frame\n");
return AVERROR(ENOMEM);
}
out_frame = av_frame_alloc();
if (!out_frame) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate output frame\n");
return AVERROR(ENOMEM);
}
raw_in = av_malloc(in_frame->linesize[0] * NUMBER_OF_FRAMES);
if (!raw_in) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for raw_in\n");
return AVERROR(ENOMEM);
}
raw_out = av_malloc(in_frame->linesize[0] * NUMBER_OF_FRAMES);
if (!raw_out) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for raw_out\n");
return AVERROR(ENOMEM);
}
for (i = 0; i < NUMBER_OF_FRAMES; i++) {
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
generate_raw_frame((uint16_t*)(in_frame->data[0]), i, enc_ctx->sample_rate,
enc_ctx->channels, enc_ctx->frame_size);
memcpy(raw_in + in_offset, in_frame->data[0], in_frame->linesize[0]);
in_offset += in_frame->linesize[0];
result = avcodec_encode_audio2(enc_ctx, &enc_pkt, in_frame, &got_output);
if (result < 0) {
av_log(NULL, AV_LOG_ERROR, "Error encoding audio frame\n");
return result;
}
/* if we get an encoded packet, feed it straight to the decoder */
if (got_output) {
result = avcodec_decode_audio4(dec_ctx, out_frame, &got_output, &enc_pkt);
if (result < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio packet\n");
return result;
}
if (got_output) {
if (result != enc_pkt.size) {
av_log(NULL, AV_LOG_INFO, "Decoder consumed only part of a packet, it is allowed to do so -- need to update this test\n");
return AVERROR_UNKNOWN;
}
if (in_frame->nb_samples != out_frame->nb_samples) {
av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different number of samples\n");
return AVERROR_UNKNOWN;
}
if (in_frame->channel_layout != out_frame->channel_layout) {
av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different channel layout\n");
return AVERROR_UNKNOWN;
}
if (in_frame->format != out_frame->format) {
av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different sample format\n");
return AVERROR_UNKNOWN;
}
memcpy(raw_out + out_offset, out_frame->data[0], out_frame->linesize[0]);
out_offset += out_frame->linesize[0];
}
}
av_free_packet(&enc_pkt);
}
if (memcmp(raw_in, raw_out, frame_data_size * NUMBER_OF_FRAMES) != 0) {
av_log(NULL, AV_LOG_ERROR, "Output differs\n");
return 1;
}
av_log(NULL, AV_LOG_INFO, "OK\n");
av_freep(&raw_in);
av_freep(&raw_out);
av_frame_free(&in_frame);
av_frame_free(&out_frame);
return 0;
}
static int close_encoder(AVCodecContext **enc_ctx)
{
avcodec_close(*enc_ctx);
av_freep(enc_ctx);
return 0;
}
static int close_decoder(AVCodecContext **dec_ctx)
{
avcodec_close(*dec_ctx);
av_freep(dec_ctx);
return 0;
}
int main(void)
{
AVCodec *enc = NULL, *dec = NULL;
AVCodecContext *enc_ctx = NULL, *dec_ctx = NULL;
uint64_t channel_layouts[] = {AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1_BACK, AV_CH_LAYOUT_SURROUND, AV_CH_LAYOUT_STEREO_DOWNMIX};
int sample_rates[] = {8000, 44100, 48000, 192000};
int cl, sr;
avcodec_register_all();
enc = avcodec_find_encoder(AV_CODEC_ID_FLAC);
if (!enc) {
av_log(NULL, AV_LOG_ERROR, "Can't find encoder\n");
return 1;
}
dec = avcodec_find_decoder(AV_CODEC_ID_FLAC);
if (!dec) {
av_log(NULL, AV_LOG_ERROR, "Can't find decoder\n");
return 1;
}
for (cl = 0; cl < FF_ARRAY_ELEMS(channel_layouts); cl++) {
for (sr = 0; sr < FF_ARRAY_ELEMS(sample_rates); sr++) {
if (init_encoder(enc, &enc_ctx, channel_layouts[cl], sample_rates[sr]) != 0)
return 1;
if (init_decoder(dec, &dec_ctx, channel_layouts[cl]) != 0)
return 1;
if (run_test(enc, dec, enc_ctx, dec_ctx) != 0)
return 1;
close_encoder(&enc_ctx);
close_decoder(&dec_ctx);
}
}
return 0;
}

View File

@@ -27,6 +27,6 @@ void ff_flac_lpc_16_arm(int32_t *samples, const int coeffs[32], int order,
av_cold void ff_flacdsp_init_arm(FLACDSPContext *c, enum AVSampleFormat fmt, int channels,
int bps)
{
if (CONFIG_FLAC_DECODER)
c->lpc16 = ff_flac_lpc_16_arm;
if (bps <= 16 && CONFIG_FLAC_DECODER)
c->lpc = ff_flac_lpc_16_arm;
}

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@@ -36,8 +36,8 @@ int ff_ass_subtitle_header(AVCodecContext *avctx,
"[Script Info]\r\n"
"; Script generated by FFmpeg/Lavc%s\r\n"
"ScriptType: v4.00+\r\n"
"PlayResX: %d\r\n"
"PlayResY: %d\r\n"
"PlayResX: 384\r\n"
"PlayResY: 288\r\n"
"\r\n"
"[V4+ Styles]\r\n"
@@ -67,7 +67,6 @@ int ff_ass_subtitle_header(AVCodecContext *avctx,
"[Events]\r\n"
"Format: Layer, Start, End, Style, Name, MarginL, MarginR, MarginV, Effect, Text\r\n",
!(avctx->flags & CODEC_FLAG_BITEXACT) ? AV_STRINGIFY(LIBAVCODEC_VERSION) : "",
ASS_DEFAULT_PLAYRESX, ASS_DEFAULT_PLAYRESY,
font, font_size, color, color, back_color, back_color,
-bold, -italic, -underline, alignment);

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@@ -25,9 +25,6 @@
#include "avcodec.h"
#include "libavutil/bprint.h"
#define ASS_DEFAULT_PLAYRESX 384
#define ASS_DEFAULT_PLAYRESY 288
/**
* @name Default values for ASS style
* @{

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@@ -103,7 +103,7 @@ typedef struct ASSSplitContext ASSSplitContext;
* Split a full ASS file or a ASS header from a string buffer and store
* the split structure in a newly allocated context.
*
* @param buf String containing the ASS formatted data.
* @param buf String containing the ASS formated data.
* @return Newly allocated struct containing split data.
*/
ASSSplitContext *ff_ass_split(const char *buf);

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@@ -333,8 +333,6 @@ static av_cold int encode_init(AVCodecContext *avctx)
avctx->global_quality / 2) / avctx->global_quality;
avctx->extradata = av_mallocz(8);
if (!avctx->extradata)
return AVERROR(ENOMEM);
avctx->extradata_size = 8;
((uint32_t *) avctx->extradata)[0] = av_le2ne32(a->inv_qscale);
((uint32_t *) avctx->extradata)[1] = av_le2ne32(AV_RL32("ASUS"));
@@ -363,8 +361,6 @@ AVCodec ff_asv1_encoder = {
.encode2 = encode_frame,
.pix_fmts = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUV420P,
AV_PIX_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE |
FF_CODEC_CAP_INIT_CLEANUP,
};
#endif
@@ -379,7 +375,5 @@ AVCodec ff_asv2_encoder = {
.encode2 = encode_frame,
.pix_fmts = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUV420P,
AV_PIX_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE |
FF_CODEC_CAP_INIT_CLEANUP,
};
#endif

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@@ -820,7 +820,7 @@ static void decode_qu_spectra(GetBitContext *gb, const Atrac3pSpecCodeTab *tab,
int num_coeffs = tab->num_coeffs;
int bits = tab->bits;
int is_signed = tab->is_signed;
unsigned val;
unsigned val, mask = (1 << bits) - 1;
for (pos = 0; pos < num_specs;) {
if (group_size == 1 || get_bits1(gb)) {
@@ -828,7 +828,7 @@ static void decode_qu_spectra(GetBitContext *gb, const Atrac3pSpecCodeTab *tab,
val = get_vlc2(gb, vlc_tab->table, vlc_tab->bits, 1);
for (i = 0; i < num_coeffs; i++) {
cf = av_mod_uintp2(val, bits);
cf = val & mask;
if (is_signed)
cf = sign_extend(cf, bits);
else if (cf && get_bits1(gb))
@@ -1724,7 +1724,11 @@ static int decode_tones_info(GetBitContext *gb, Atrac3pChanUnitCtx *ctx,
if (num_channels == 2) {
get_subband_flags(gb, ctx->waves_info->tone_sharing, ctx->waves_info->num_tone_bands);
get_subband_flags(gb, ctx->waves_info->tone_master, ctx->waves_info->num_tone_bands);
get_subband_flags(gb, ctx->waves_info->invert_phase, ctx->waves_info->num_tone_bands);
if (get_subband_flags(gb, ctx->waves_info->phase_shift,
ctx->waves_info->num_tone_bands)) {
avpriv_report_missing_feature(avctx, "GHA Phase shifting");
return AVERROR_PATCHWELCOME;
}
}
ctx->waves_info->tones_index = 0;

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@@ -122,7 +122,7 @@ typedef struct Atrac3pWaveSynthParams {
int num_tone_bands; ///< number of PQF bands with tones
uint8_t tone_sharing[ATRAC3P_SUBBANDS]; ///< 1 - subband-wise tone sharing flags
uint8_t tone_master[ATRAC3P_SUBBANDS]; ///< 1 - subband-wise tone channel swapping
uint8_t invert_phase[ATRAC3P_SUBBANDS]; ///< 1 - subband-wise phase inversion
uint8_t phase_shift[ATRAC3P_SUBBANDS]; ///< 1 - subband-wise 180° phase shifting
int tones_index; ///< total sum of tones in this unit
Atrac3pWaveParam waves[48];
} Atrac3pWaveSynthParams;

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@@ -72,9 +72,6 @@ static av_cold int atrac3p_decode_close(AVCodecContext *avctx)
av_freep(&ctx->ch_units);
av_freep(&ctx->fdsp);
ff_mdct_end(&ctx->mdct_ctx);
ff_mdct_end(&ctx->ipqf_dct_ctx);
return 0;
}
@@ -384,7 +381,7 @@ static int atrac3p_decode_frame(AVCodecContext *avctx, void *data,
*got_frame_ptr = 1;
return FFMIN(avctx->block_align, avpkt->size);
return avctx->block_align;
}
AVCodec ff_atrac3p_decoder = {

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@@ -116,16 +116,14 @@ av_cold void ff_atrac3p_init_wave_synth(void)
* @param[in] synth_param ptr to common synthesis parameters
* @param[in] waves_info parameters for each sine wave
* @param[in] envelope envelope data for all waves in a group
* @param[in] fdsp ptr to floating-point DSP context
* @param[in] invert_phase flag indicating 180° phase shift
* @param[in] phase_shift flag indicates 180° phase shift
* @param[in] reg_offset region offset for trimming envelope data
* @param[out] out receives sythesized data
*/
static void waves_synth(Atrac3pWaveSynthParams *synth_param,
Atrac3pWavesData *waves_info,
Atrac3pWaveEnvelope *envelope,
AVFloatDSPContext *fdsp,
int invert_phase, int reg_offset, float *out)
int phase_shift, int reg_offset, float *out)
{
int i, wn, inc, pos;
double amp;
@@ -148,10 +146,6 @@ static void waves_synth(Atrac3pWaveSynthParams *synth_param,
}
}
/* invert phase if requested */
if (invert_phase)
fdsp->vector_fmul_scalar(out, out, -1.0f, 128);
/* fade in with steep Hann window if requested */
if (envelope->has_start_point) {
pos = (envelope->start_pos << 2) - reg_offset;
@@ -222,12 +216,12 @@ void ff_atrac3p_generate_tones(Atrac3pChanUnitCtx *ch_unit, AVFloatDSPContext *f
/* synthesize waves for both overlapping regions */
if (tones_now->num_wavs && reg1_env_nonzero)
waves_synth(ch_unit->waves_info_prev, tones_now, &tones_now->curr_env,
fdsp, ch_unit->waves_info_prev->invert_phase[sb] & ch_num,
ch_unit->waves_info_prev->phase_shift[sb] & ch_num,
128, wavreg1);
if (tones_next->num_wavs && reg2_env_nonzero)
waves_synth(ch_unit->waves_info, tones_next, &tones_next->curr_env, fdsp,
ch_unit->waves_info->invert_phase[sb] & ch_num, 0, wavreg2);
waves_synth(ch_unit->waves_info, tones_next, &tones_next->curr_env,
ch_unit->waves_info->phase_shift[sb] & ch_num, 0, wavreg2);
/* Hann windowing for non-faded wave signals */
if (tones_now->num_wavs && tones_next->num_wavs &&
@@ -605,8 +599,8 @@ void ff_atrac3p_ipqf(FFTContext *dct_ctx, Atrac3pIPQFChannelCtx *hist,
const float *in, float *out)
{
int i, s, sb, t, pos_now, pos_next;
LOCAL_ALIGNED(32, float, idct_in, [ATRAC3P_SUBBANDS]);
LOCAL_ALIGNED(32, float, idct_out, [ATRAC3P_SUBBANDS]);
DECLARE_ALIGNED(32, float, idct_in)[ATRAC3P_SUBBANDS];
DECLARE_ALIGNED(32, float, idct_out)[ATRAC3P_SUBBANDS];
memset(out, 0, ATRAC3P_FRAME_SAMPLES * sizeof(*out));

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@@ -291,8 +291,6 @@ enum AVCodecID {
AV_CODEC_ID_MVC1_DEPRECATED,
AV_CODEC_ID_MVC2_DEPRECATED,
AV_CODEC_ID_HQX,
AV_CODEC_ID_TDSC,
AV_CODEC_ID_HQ_HQA,
AV_CODEC_ID_BRENDER_PIX= MKBETAG('B','P','I','X'),
AV_CODEC_ID_Y41P = MKBETAG('Y','4','1','P'),
@@ -1825,7 +1823,7 @@ typedef struct AVCodecContext {
/**
* precision of the intra DC coefficient - 8
* - encoding: Set by user.
* - decoding: Set by libavcodec
* - decoding: unused
*/
int intra_dc_precision;
@@ -2852,7 +2850,6 @@ typedef struct AVCodecContext {
#define FF_PROFILE_DTS_96_24 40
#define FF_PROFILE_DTS_HD_HRA 50
#define FF_PROFILE_DTS_HD_MA 60
#define FF_PROFILE_DTS_EXPRESS 70
#define FF_PROFILE_MPEG2_422 0
#define FF_PROFILE_MPEG2_HIGH 1
@@ -2912,11 +2909,6 @@ typedef struct AVCodecContext {
#define FF_PROFILE_HEVC_MAIN_STILL_PICTURE 3
#define FF_PROFILE_HEVC_REXT 4
#define FF_PROFILE_VP9_0 0
#define FF_PROFILE_VP9_1 1
#define FF_PROFILE_VP9_2 2
#define FF_PROFILE_VP9_3 3
/**
* level
* - encoding: Set by user.
@@ -3270,11 +3262,6 @@ typedef struct AVCodec {
* Will be called when seeking
*/
void (*flush)(AVCodecContext *);
/**
* Internal codec capabilities.
* See FF_CODEC_CAP_* in internal.h
*/
int caps_internal;
} AVCodec;
int av_codec_get_max_lowres(const AVCodec *codec);
@@ -3852,8 +3839,6 @@ int av_packet_merge_side_data(AVPacket *pkt);
int av_packet_split_side_data(AVPacket *pkt);
const char *av_packet_side_data_name(enum AVPacketSideDataType type);
/**
* Pack a dictionary for use in side_data.
*

View File

@@ -336,29 +336,6 @@ uint8_t *av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
return NULL;
}
const char *av_packet_side_data_name(enum AVPacketSideDataType type)
{
switch(type) {
case AV_PKT_DATA_PALETTE: return "Palette";
case AV_PKT_DATA_NEW_EXTRADATA: return "New Extradata";
case AV_PKT_DATA_PARAM_CHANGE: return "Param Change";
case AV_PKT_DATA_H263_MB_INFO: return "H263 MB Info";
case AV_PKT_DATA_REPLAYGAIN: return "Replay Gain";
case AV_PKT_DATA_DISPLAYMATRIX: return "Display Matrix";
case AV_PKT_DATA_STEREO3D: return "Stereo 3D";
case AV_PKT_DATA_AUDIO_SERVICE_TYPE: return "Audio Service Type";
case AV_PKT_DATA_SKIP_SAMPLES: return "Skip Samples";
case AV_PKT_DATA_JP_DUALMONO: return "JP Dual Mono";
case AV_PKT_DATA_STRINGS_METADATA: return "Strings Metadata";
case AV_PKT_DATA_SUBTITLE_POSITION: return "Subtitle Position";
case AV_PKT_DATA_MATROSKA_BLOCKADDITIONAL: return "Matroska BlockAdditional";
case AV_PKT_DATA_WEBVTT_IDENTIFIER: return "WebVTT ID";
case AV_PKT_DATA_WEBVTT_SETTINGS: return "WebVTT Settings";
case AV_PKT_DATA_METADATA_UPDATE: return "Metadata Update";
}
return NULL;
}
#define FF_MERGE_MARKER 0x8c4d9d108e25e9feULL
int av_packet_merge_side_data(AVPacket *pkt){

View File

@@ -950,32 +950,15 @@ static int binkb_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
return 0;
}
static int bink_put_pixels(BinkContext *c,
uint8_t *dst, uint8_t *prev, int stride,
uint8_t *ref_start,
uint8_t *ref_end)
{
int xoff = get_value(c, BINK_SRC_X_OFF);
int yoff = get_value(c, BINK_SRC_Y_OFF);
uint8_t *ref = prev + xoff + yoff * stride;
if (ref < ref_start || ref > ref_end) {
av_log(c->avctx, AV_LOG_ERROR, "Copy out of bounds @%d, %d\n",
xoff, yoff);
return AVERROR_INVALIDDATA;
}
c->hdsp.put_pixels_tab[1][0](dst, ref, stride, 8);
return 0;
}
static int bink_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
int plane_idx, int is_chroma)
{
int blk, ret;
int i, j, bx, by;
uint8_t *dst, *prev, *ref_start, *ref_end;
uint8_t *dst, *prev, *ref, *ref_start, *ref_end;
int v, col[2];
const uint8_t *scan;
int xoff, yoff;
LOCAL_ALIGNED_16(int16_t, block, [64]);
LOCAL_ALIGNED_16(uint8_t, ublock, [64]);
LOCAL_ALIGNED_16(int32_t, dctblock, [64]);
@@ -1097,10 +1080,15 @@ static int bink_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
prev += 8;
break;
case MOTION_BLOCK:
ret = bink_put_pixels(c, dst, prev, stride,
ref_start, ref_end);
if (ret < 0)
return ret;
xoff = get_value(c, BINK_SRC_X_OFF);
yoff = get_value(c, BINK_SRC_Y_OFF);
ref = prev + xoff + yoff * stride;
if (ref < ref_start || ref > ref_end) {
av_log(c->avctx, AV_LOG_ERROR, "Copy out of bounds @%d, %d\n",
bx*8 + xoff, by*8 + yoff);
return AVERROR_INVALIDDATA;
}
c->hdsp.put_pixels_tab[1][0](dst, ref, stride, 8);
break;
case RUN_BLOCK:
scan = bink_patterns[get_bits(gb, 4)];
@@ -1126,10 +1114,15 @@ static int bink_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
dst[coordmap[*scan++]] = get_value(c, BINK_SRC_COLORS);
break;
case RESIDUE_BLOCK:
ret = bink_put_pixels(c, dst, prev, stride,
ref_start, ref_end);
if (ret < 0)
return ret;
xoff = get_value(c, BINK_SRC_X_OFF);
yoff = get_value(c, BINK_SRC_Y_OFF);
ref = prev + xoff + yoff * stride;
if (ref < ref_start || ref > ref_end) {
av_log(c->avctx, AV_LOG_ERROR, "Copy out of bounds @%d, %d\n",
bx*8 + xoff, by*8 + yoff);
return AVERROR_INVALIDDATA;
}
c->hdsp.put_pixels_tab[1][0](dst, ref, stride, 8);
c->bdsp.clear_block(block);
v = get_bits(gb, 7);
read_residue(gb, block, v);
@@ -1146,10 +1139,15 @@ static int bink_decode_plane(BinkContext *c, AVFrame *frame, GetBitContext *gb,
c->bdsp.fill_block_tab[1](dst, v, stride, 8);
break;
case INTER_BLOCK:
ret = bink_put_pixels(c, dst, prev, stride,
ref_start, ref_end);
if (ret < 0)
return ret;
xoff = get_value(c, BINK_SRC_X_OFF);
yoff = get_value(c, BINK_SRC_Y_OFF);
ref = prev + xoff + yoff * stride;
if (ref < ref_start || ref > ref_end) {
av_log(c->avctx, AV_LOG_ERROR, "Copy out of bounds @%d, %d\n",
bx*8 + xoff, by*8 + yoff);
return -1;
}
c->hdsp.put_pixels_tab[1][0](dst, ref, stride, 8);
memset(dctblock, 0, sizeof(*dctblock) * 64);
dctblock[0] = get_value(c, BINK_SRC_INTER_DC);
read_dct_coeffs(gb, dctblock, bink_scan, bink_inter_quant, -1);

View File

@@ -31,7 +31,6 @@
#include "libavutil/atomic.h"
#include "libavutil/avassert.h"
#include "avcodec.h"
#include "internal.h"
#include "mathops.h"
#include "get_bits.h"
#include "put_bits.h"
@@ -70,8 +69,6 @@ void avpriv_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
if (length == 0)
return;
av_assert0(length <= put_bits_left(pb));
if (CONFIG_SMALL || words < 16 || put_bits_count(pb) & 7) {
for (i = 0; i < words; i++)
put_bits(pb, 16, AV_RB16(src + 2 * i));
@@ -172,7 +169,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
if (table_nb_bits > 30)
return -1;
table_index = alloc_table(vlc, table_size, flags & INIT_VLC_USE_NEW_STATIC);
ff_dlog(NULL, "new table index=%d size=%d\n", table_index, table_size);
av_dlog(NULL, "new table index=%d size=%d\n", table_index, table_size);
if (table_index < 0)
return table_index;
table = (volatile VLC_TYPE (*)[2])&vlc->table[table_index];
@@ -182,7 +179,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
n = codes[i].bits;
code = codes[i].code;
symbol = codes[i].symbol;
ff_dlog(NULL, "i=%d n=%d code=0x%x\n", i, n, code);
av_dlog(NULL, "i=%d n=%d code=0x%x\n", i, n, code);
if (n <= table_nb_bits) {
/* no need to add another table */
j = code >> (32 - table_nb_bits);
@@ -194,7 +191,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
}
for (k = 0; k < nb; k++) {
int bits = table[j][1];
ff_dlog(NULL, "%4x: code=%d n=%d\n", j, i, n);
av_dlog(NULL, "%4x: code=%d n=%d\n", j, i, n);
if (bits != 0 && bits != n) {
av_log(NULL, AV_LOG_ERROR, "incorrect codes\n");
return AVERROR_INVALIDDATA;
@@ -224,7 +221,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
subtable_bits = FFMIN(subtable_bits, table_nb_bits);
j = (flags & INIT_VLC_LE) ? bitswap_32(code_prefix) >> (32 - table_nb_bits) : code_prefix;
table[j][1] = -subtable_bits;
ff_dlog(NULL, "%4x: n=%d (subtable)\n",
av_dlog(NULL, "%4x: n=%d (subtable)\n",
j, codes[i].bits + table_nb_bits);
index = build_table(vlc, subtable_bits, k-i, codes+i, flags);
if (index < 0)
@@ -247,7 +244,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
/* Build VLC decoding tables suitable for use with get_vlc().
'nb_bits' set the decoding table size (2^nb_bits) entries. The
'nb_bits' set thee decoding table size (2^nb_bits) entries. The
bigger it is, the faster is the decoding. But it should not be too
big to save memory and L1 cache. '9' is a good compromise.
@@ -265,7 +262,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
'xxx_size' : gives the number of bytes of each entry of the 'bits'
or 'codes' tables.
'wrap' and 'size' make it possible to use any memory configuration and types
'wrap' and 'size' allows to use any memory configuration and types
(byte/word/long) to store the 'bits', 'codes', and 'symbols' tables.
'use_static' should be set to 1 for tables, which should be freed

View File

@@ -71,10 +71,8 @@ static av_always_inline type bytestream2_get_ ## name ## u(GetByteContext *g) \
} \
static av_always_inline type bytestream2_get_ ## name(GetByteContext *g) \
{ \
if (g->buffer_end - g->buffer < bytes) { \
g->buffer = g->buffer_end; \
if (g->buffer_end - g->buffer < bytes) \
return 0; \
} \
return bytestream2_get_ ## name ## u(g); \
} \
static av_always_inline type bytestream2_peek_ ## name(GetByteContext *g) \

View File

@@ -174,7 +174,7 @@ int main(void){
CABACContext c;
uint8_t b[9*SIZE];
uint8_t r[9*SIZE];
int i, ret = 0;
int i;
uint8_t state[10]= {0};
AVLFG prng;
@@ -188,11 +188,15 @@ int main(void){
}
for(i=0; i<SIZE; i++){
START_TIMER
put_cabac_bypass(&c, r[i]&1);
STOP_TIMER("put_cabac_bypass")
}
for(i=0; i<SIZE; i++){
START_TIMER
put_cabac(&c, state, r[i]&1);
STOP_TIMER("put_cabac")
}
put_cabac_terminate(&c, 1);
@@ -202,24 +206,22 @@ int main(void){
memset(state, 0, sizeof(state));
for(i=0; i<SIZE; i++){
if( (r[i]&1) != get_cabac_bypass(&c) ) {
START_TIMER
if( (r[i]&1) != get_cabac_bypass(&c) )
av_log(NULL, AV_LOG_ERROR, "CABAC bypass failure at %d\n", i);
ret = 1;
}
STOP_TIMER("get_cabac_bypass")
}
for(i=0; i<SIZE; i++){
if( (r[i]&1) != get_cabac_noinline(&c, state) ) {
START_TIMER
if( (r[i]&1) != get_cabac_noinline(&c, state) )
av_log(NULL, AV_LOG_ERROR, "CABAC failure at %d\n", i);
ret = 1;
}
STOP_TIMER("get_cabac")
}
if(!get_cabac_terminate(&c)) {
if(!get_cabac_terminate(&c))
av_log(NULL, AV_LOG_ERROR, "where's the Terminator?\n");
ret = 1;
}
return ret;
return 0;
}
#endif /* TEST */

View File

@@ -1,63 +0,0 @@
/*
* Canopus common routines
* Copyright (c) 2015 Vittorio Giovara <vittorio.giovara@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "libavutil/rational.h"
#include "avcodec.h"
#include "bytestream.h"
#include "canopus.h"
int ff_canopus_parse_info_tag(AVCodecContext *avctx,
const uint8_t *src, size_t size)
{
GetByteContext gbc;
int par_x, par_y, field_order;
bytestream2_init(&gbc, src, size);
/* Parse aspect ratio. */
bytestream2_skip(&gbc, 8); // unknown, 16 bits 1
par_x = bytestream2_get_le32(&gbc);
par_y = bytestream2_get_le32(&gbc);
if (par_x && par_y)
av_reduce(&avctx->sample_aspect_ratio.num,
&avctx->sample_aspect_ratio.den,
par_x, par_y, 255);
/* Short INFO tag (used in CLLC) has only AR data. */
if (size == 0x18)
return 0;
bytestream2_skip(&gbc, 16); // unknown RDRT tag
/* Parse FIEL tag. */
bytestream2_skip(&gbc, 8); // 'FIEL' and 4 bytes 0
field_order = bytestream2_get_le32(&gbc);
switch (field_order) {
case 0: avctx->field_order = AV_FIELD_TT; break;
case 1: avctx->field_order = AV_FIELD_BB; break;
case 2: avctx->field_order = AV_FIELD_PROGRESSIVE; break;
}
return 0;
}

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