* commit '2df0c32ea12ddfa72ba88309812bfb13b674130f':
lavc: use a separate field for exporting audio encoder padding
Conflicts:
libavcodec/audio_frame_queue.c
libavcodec/avcodec.h
libavcodec/libvorbisenc.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmaenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
codec context is used for encoding or decoding (and has yet another
different meaning for video), preventing generic handling of the codec
context.
Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
As indicated in the function documentation, the header MUST be
checked prior to calling it because no consistency check is done
there.
CC:libav-stable@libav.org
this prevents the creation of a packet even though no single sample has ever
been input, which some confusion in the timestamp generation
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0f24a3ca999a702f83af9307f9f47b6fdeb546a5':
lavc: remove disabled FF_API_OLD_ENCODE_VIDEO cruft
lavc: remove disabled FF_API_OLD_ENCODE_AUDIO cruft
lavc: remove disabled FF_API_OLD_DECODE_AUDIO cruft
Conflicts:
libavcodec/flacenc.c
libavcodec/libgsm.c
libavcodec/utils.c
libavcodec/version.h
The compatibility wrapers are left as they likely sre still
in wide use. They will be removed when they break or otherwise
cause work without an volunteer being available.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6327c10702922eabcb1c6170abd3f03d23ce4c51':
atomic: fix CAS with armcc.
png: use av_mallocz_array() for the zlib zalloc function
libmp3lame: use the correct remaining buffer size when flushing
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '284ea790d89441fa1e6b2d72d3c1ed6d61972f0b':
dsputil: move vector_fmul_scalar() to AVFloatDSPContext in libavutil
aacenc: use the correct output buffer
aacdec: fix signed overflows in lcg_random()
base64: fix signed overflow in shift
Conflicts:
libavcodec/dsputil.c
libavutil/base64.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '292d1e78743855404c7d07e3e7cb3f9c9ae6275b':
fate: dependencies for acodec tests
fate: dependencies for vsynth tests
fate: add macros useful for conditionally enabling things
libmp3lame: resize the output buffer if needed
Conflicts:
tests/fate/acodec.mak
tests/fate/vcodec.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The LAME API documentation for the required buffer size refers to the size for
a single encode call. However, we store multiple frames in the same output
buffer but only read 1 frame at a time out of it. As a result, the buffer size
given in lame_encode_buffer() is actually smaller than what it should be.
Since we do not know how many frames it will end up buffering, it is best to
just reallocate if needed.
* qatar/master:
wmaenc: use float planar sample format
(e)ac3enc: use planar sample format
aacenc: use planar sample format
adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
adpcmenc: move 'ch' variable to higher scope
adpcmenc: fix 3 instances of variable shadowing
adpcm_ima_wav: simplify encoding
libvorbis: use planar sample format
libmp3lame: use planar sample formats
vorbisenc: use float planar sample format
ffm: do not write or read the audio sample format
parseutils: fix parsing of invalid alpha values
doc/RELEASE_NOTES: update for the 9 release.
smoothstreamingenc: Add a more verbose error message
smoothstreamingenc: Ignore the return value from mkdir
smoothstreamingenc: Try writing a manifest when opening the muxer
smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
smoothstreamingenc: Properly return errors from ism_flush to the caller
smoothstreamingenc: Check the output UrlContext before accessing it
Conflicts:
doc/RELEASE_NOTES
libavcodec/aacenc.c
libavcodec/ac3enc_template.c
libavcodec/wmaenc.c
tests/ref/lavf/ffm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libmp3lame: add missing layout terminator
avconv: multithreaded demuxing.
Bump lavu minor and add an APIChanges entry for audioconvert functions.
audioconvert: add a function for extracting the channel with the given index
audioconvert: add a function for getting the name of a single channel.
audioconvert: add a function for getting channel's index in layout
audioconvert: use av_popcount64 in av_get_channel_layout_nb_channels
vf_libopencv: add missing headers.
iac: add missing dependency
Conflicts:
configure
doc/APIchanges
ffmpeg.c
libavcodec/libmp3lame.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (26 commits)
fate: use diff -b in oneline comparison
Add missing version bumps and APIchanges/Changelog entries.
lavfi: move buffer management function to a separate file.
lavfi: move formats-related functions from default.c to formats.c
lavfi: move video-related functions to a separate file.
fate: make smjpeg a demux test
fate: separate sierra-vmd audio and video tests
fate: separate smacker audio and video tests
libmp3lame: set supported channel layouts.
avconv: automatically insert asyncts when -async is used.
avconv: add support for audio filters.
lavfi: add asyncts filter.
lavfi: add aformat filter
lavfi: add an audio buffer sink.
lavfi: add an audio buffer source.
buffersrc: add av_buffersrc_write_frame().
buffersrc: fix invalid read in uninit if the fifo hasn't been allocated
lavfi: rename vsrc_buffer.c to buffersrc.c
avfiltergraph: reindent
lavfi: add channel layout/sample rate negotiation.
...
Conflicts:
Changelog
doc/APIchanges
doc/filters.texi
ffmpeg.c
ffprobe.c
libavcodec/libmp3lame.c
libavfilter/Makefile
libavfilter/af_aformat.c
libavfilter/allfilters.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/buffersrc.c
libavfilter/defaults.c
libavfilter/formats.c
libavfilter/src_buffer.c
libavfilter/version.h
libavfilter/vf_yadif.c
libavfilter/vsrc_buffer.c
libavfilter/vsrc_buffer.h
libavutil/avutil.h
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (26 commits)
adxenc: use AVCodec.encode2()
adxenc: Use the AVFrame in ADXContext for coded_frame
indeo4: fix out-of-bounds function call.
configure: Restructure help output.
configure: Internal-only components should not be command-line selectable.
vorbisenc: use AVCodec.encode2()
libvorbis: use AVCodec.encode2()
libopencore-amrnbenc: use AVCodec.encode2()
ra144enc: use AVCodec.encode2()
nellymoserenc: use AVCodec.encode2()
roqaudioenc: use AVCodec.encode2()
libspeex: use AVCodec.encode2()
libvo_amrwbenc: use AVCodec.encode2()
libvo_aacenc: use AVCodec.encode2()
wmaenc: use AVCodec.encode2()
mpegaudioenc: use AVCodec.encode2()
libmp3lame: use AVCodec.encode2()
libgsmenc: use AVCodec.encode2()
libfaac: use AVCodec.encode2()
g726enc: use AVCodec.encode2()
...
Conflicts:
configure
libavcodec/Makefile
libavcodec/ac3enc.c
libavcodec/adxenc.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/vorbisenc.c
libavcodec/wmaenc.c
tests/ref/acodec/g722
tests/ref/lavf/asf
tests/ref/lavf/ffm
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_asf
tests/ref/seek/lavf_ffm
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_mpg
tests/ref/seek/lavf_rm
tests/ref/seek/lavf_ts
Merged-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>
