Also move the declaration to internal.h, and add restrict qualifiers
to the declaration (as in the implementation).
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '2eba9087f3031c6050f8dcd996225490be6c2410':
lavc: make up a fake frame channel layout when there is no real one.
Conflicts:
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7ecc2d403ce5c7b6ea3b1f368dccefd105209c7e':
Move AVFrame from lavc to lavu.
Conflicts:
libavcodec/avcodec.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is the most that can be represented with the current channel layout
system. This limit should be raised/removed when a better system is
implemented.
* commit '5a4e9fe855282a99586050a507d0a486ad39df5b':
avcodec/internal: Fix #if DECODE_AUDIO / ENCODE_AUDIO name mismatch
shorten: use the unsigned type where needed
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit also makes sure the extradata and subtitle_header are NUL
terminated, without taking into account the trailing '\0' in account in
the size.
At the same time, it should fix 'warning: dereferencing type-punned
pointer will break strict-aliasing rules' warning for compilers who
don't consider uint8_t** and char** compatibles.
Will be used in future patches, together with the
variable that allows checking whether the lock is held.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* commit 'e57c4706e969afa1f2384481b955ccd9494cddb5':
lavc: don't reuse audio buffers
This commit causes a 0.5% speedloss for mp3 and 2% for raw pcm, that is
"time ffmpeg" style tested thus includes disk IO, demux and parsing
I would not have merged it if it wasnt required for the "new" buffer API
but sadly it is.
Once the new API is in ill reimplement proper buffer reuse, which will
fix this speedloss. In case we choose not to merge the "new" buffer
API then this commit here should be reverted.
Conflicts:
libavcodec/internal.h
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8ac0f6767bf63d3e6b308ee6648ff02598b81e03':
dcadec: allow the decoder to change the channel layout mid-stream
cook: use av_dlog() for debug logging instead of av_log() with AV_LOG_ERROR
cook: move samples_per_frame from COOKSubpacket to where it is used
cook: use av_get_channel_layout_nb_channels() instead of cook_count_channels()
cook: reverse a condition so that the code makes more sense
cook: remove unneeded COOKContext variable, sample_rate
cook: remove unneeded COOKContext variable, bit_rate
cook: use AVCodecContext.channels instead of keeping a private copy
bmvaudio: set channel layout at init() rather than validating it
atrac1: do not keep a copy of channel count in the private context
dsicinaudio: set channels and channel layout
g722dec: set channel layout at initialization instead of validating it
amrwbdec: set channels, channel_layout, and sample_rate
amrnbdec: set channels, channel_layout, and sample_rate
dca_parser: allow the parser to change the sample rate
lavc: check channel count after decoder init
lavc: move SANE_NB_CHANNELS to internal.h and use it in the PCM decoders
Conflicts:
libavcodec/dcadec.c
libavcodec/pcm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Compared to the decoder side, this code is able to change both the
delay and the number of threads seamlessly during encoding. Also
any idle thread can pick up tasks, the strict round robin in order
limit is gone too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
libxvid: Give more suitable names to libxvid-related files.
libxvid: Separate libxvid encoder from libxvid rate control code.
jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse().
fate: cosmetics: lowercase some comments
fate: Give more consistent names to some RealVideo/RealAudio tests.
lavfi: add avfilter_get_audio_buffer_ref_from_arrays().
lavfi: add extended_data to AVFilterBuffer.
lavc: check that extended_data is properly set in avcodec_encode_audio2().
lavc: pad last audio frame with silence when needed.
samplefmt: add a function for filling a buffer with silence.
samplefmt: add a function for copying audio samples.
lavr: do not try to copy to uninitialized output audio data.
lavr: make avresample_read() with NULL output discard samples.
fate: split idroq audio and video into separate tests
fate: improve dependencies
fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests
fate: split some combined tests into separate audio and video tests
fate: fix dependencies for probe tests
mips: intreadwrite: fix inline asm for gcc 4.8
mips: intreadwrite: remove unnecessary inline asm
...
Conflicts:
cmdutils.h
configure
doc/APIchanges
doc/filters.texi
ffmpeg.c
ffplay.c
libavcodec/internal.h
libavcodec/jpeglsdec.c
libavcodec/libschroedingerdec.c
libavcodec/libxvid.c
libavcodec/libxvid_rc.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersink.h
tests/Makefile
tests/fate/aac.mak
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/ea.mak
tests/fate/image.mak
tests/fate/libavutil.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/qt.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/vqf.mak
tests/ref/fate/ea-mad
tests/ref/fate/ea-tqi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes race conditions that ultimately lead to memory corruption.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The new API allows (optionally and on by default) using a internal buffer to encode, avoiding
the need to allocate large buffers or risking failure on too small buffers.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (36 commits)
adpcmenc: Use correct frame_size for Yamaha ADPCM.
avcodec: add ff_samples_to_time_base() convenience function to internal.h
adx parser: set duration
mlp parser: set duration instead of frame_size
gsm parser: set duration
mpegaudio parser: set duration instead of frame_size
(e)ac3 parser: set duration instead of frame_size
flac parser: set duration instead of frame_size
avcodec: add duration field to AVCodecParserContext
avutil: add av_rescale_q_rnd() to allow different rounding
pnmdec: remove useless .pix_fmts
libmp3lame: support float and s32 sample formats
libmp3lame: renaming, rearrangement, alignment, and comments
libmp3lame: use the LAME default bit rate
libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
libmp3lame: cosmetics: remove some pointless comments
libmp3lame: convert some debugging code to av_dlog()
libmp3lame: remove outdated comment.
libmp3lame: do not set coded_frame->key_frame.
libmp3lame: improve error handling in MP3lame_encode_init()
...
Conflicts:
doc/APIchanges
libavcodec/libmp3lame.c
libavcodec/pcxenc.c
libavcodec/pnmdec.c
libavcodec/pnmenc.c
libavcodec/sgienc.c
libavcodec/utils.c
libavformat/hls.c
libavutil/avutil.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libx264: fix indentation.
vorbis: fix overflows in floor1[] vector and inverse db table index.
win64: add a XMM clobber test configure option.
movdec: Parse the dvc1 atom
ARM: ac3: fix ac3_bit_alloc_calc_bap_armv6
swscale: K&R formatting cosmetics for Blackfin code
frwu: lowercase the FRWU codec name
movdec: fix dts generation in fragmented files
fate: make acodec-ac3_fixed test output raw AC3
APIchanges: add missing commit hashes
swscale: implement MMX, SSE2 and AVX functions for RGB32 input.
ra144enc: drop pointless "encoder" from .long_name
bethsoftvideo: fix palette reading.
mpc7: use av_fast_padded_malloc()
mpc7: simplify handling of packet sizes that are not a multiple of 4 bytes
doc: decoding Forward Uncompressed is supported
Fix a typo in the x86 asm version of ff_vector_clip_int32()
pcmenc: Do not set avpkt->size.
ff_alloc_packet: modify the size of the packet to match the requested size
Conflicts:
doc/APIchanges
libavcodec/libx264.c
libavcodec/mpc7.c
libavformat/isom.h
libswscale/Makefile
libswscale/bfin/yuv2rgb_bfin.c
tests/ref/fate/bethsoft-vid
tests/ref/seek/ac3_ac3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rv34: add NEON rv34_idct_add
rv34: 1-pass inter MB reconstruction
add SMJPEG muxer
avformat: split out common SMJPEG code
pictordec: Use bytestream2 functions
avconv: use avcodec_encode_audio2()
pcmenc: use AVCodec.encode2()
avcodec: bump minor version and add APIChanges for the new audio encoding API
avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio()
avcodec: add a public function, avcodec_fill_audio_frame().
rv34: Intra 16x16 handling
rv34: Inter/intra MB code split
Conflicts:
Changelog
libavcodec/avcodec.h
libavcodec/pictordec.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/rv34dsp.asm
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows audio encoders to optionally take an AVFrame as input and write
encoded output to an AVPacket.
This also adds AVCodec.encode2() which will also be usable by video and
subtitle encoders once support is implemented in the public functions.
While we correctly "register" the side data when we split it,
the application (in this case FFmpeg) might not update the
AVPacket pool it uses to finally free the packet, thus
causing a leak.
This also makes the av_dup_packet unnecessary which could
cause an even worse leak in this situation.
Also change the code to not modify the user-provide AVPacket at all.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
