| Commit message (Collapse) | Author | Age | Files | Lines |
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* cus/stable:
ffplay: use frame count based queueing for audio queue
ffplay: reset audio_pkt_temp when opening audio
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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This reduces the number of queued frames for audio data but also reduces the
amount of A-V difference after changing the audio stream (because less frames
are queued). Fixes bug #1035.
Signed-off-by: Marton Balint <cus@passwd.hu>
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Otherwise we may use the remaining data of the last packet from the previous
audio stream. Fixes bug #951.
Signed-off-by: Marton Balint <cus@passwd.hu>
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* 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>
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
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Break some of the longer sections into smaller sensible pieces;
make some option descriptions and option ordering more consistent.
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The first frame pts must be saved until we have 8 frames since RoQ audio
requires 8 frames in the first packet.
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Update FATE references due to encoder delay.
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Encoder output is delayed by several frames, so we keep a queue of input
frame timing info to match up with corresponding output packets.
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FATE reference updated due timestamp rounding because of resampling from
44100 Hz to 16000 Hz in avconv.
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Update FATE references due to encoder delay.
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This simplifies matching of timestamps between input frames and output
packets.
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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This parly reverts 85469f.
Fixes ticket #1091.
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Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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This is even potentially faster in this use-case.
Should fix AAC SBR decoding on machines with SSE but not
SSE2, fixing track issue #1041.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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This new API stores useful data in a dedicated structure
and has clearly delimited init functions.
Hopefully, uses of the old API can be replaced quickly.
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Author of the code from qatar is Ronald S. Bultje
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Allow to specify the display aspect ratio adopted for rendering
subtitles.
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Previously it was using the same value of the input video DAR, which is
inconsistent with most implementations.
Fix trac ticket #1098.
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Check on < 0 rather than on != 0, this is more correct as a positive
error code from av_parse_ratio() value doesn't mean an error.
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