Explicitly identify decoder/encoder wrappers with a common name. This
saves API users from guessing by the name suffix. For example, they
don't have to guess that "h264_qsv" is the h264 QSV implementation, and
instead they can just check the AVCodec .codec and .wrapper_name fields.
Explicitly mark AVCodec entries that are hardware decoders or most
likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing
API users listing hardware decoders in a more generic way. The proposed
AVCodecHWConfig does not provide this information fully, because it's
concerned with decoder configuration, not information about the fact
whether the hardware is used or not.
AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software
implementations in case the hardware is not capable.
Based on a patch by Philip Langdale <philipl@overt.org>.
Merges Libav commit 47687a2f8a.
Pass the cutoff option from lavc's avcodec_options[] to libmp3lame's
lowpass option, without allowing to adjust its default behavior.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '955aec3c7c7be39b659197e1ec379a09f2b7c41c':
mpegaudiodecheader: check the header in avpriv_mpegaudio_decode_header
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Almost all the places from which this function is called already check
the header manually and in the two that don't (the mp3 muxer) the check
should not cause any problems.
This parameter can be used to inform the allocation code about how much
downsizing might occur, and can be used to optimize how to allocate the
packet
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* 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>
