| Commit message (Collapse) | Author | Age | Files | Lines |
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Rely on frame_count. Also more consistent with audio path.
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Fixes use of the example with encoders which use tha AVFrame w/h/pix_fmt fields
FFV1 is one of these codecs
We cannot easily workaround the not set fields in common code because the API
has AVFrame constant for the encoders.
Alternatives would be to fix the API or to duplicate the struct and fill in
missing fields. Or as is to require all user apps to set this correctly and
maybe simplify for that case
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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In particular, fix trac ticket #3231.
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Simplify logic, avoid multiple unnecessary alloc/free operations.
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timebase
Fix PTS set on the frame when encoding, which must be specified in the
encoder timebase or this will confuse the encoder.
When muxing the packet, the PTS/DTS generated by the encoder is then
rescaled to the stream timebase.
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Avoid the need of tweaking, also show how to get list of supported sample
formats.
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- do not allocate resample dst buffer when resample is off
- free sample buffers in addition to freeing data pointer arrays
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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sample_fmt
We generate S16 samples and we should allocate the right buffer
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Hi list! Since my last patch (fix 2 memleaks in doc/examples/muxing.c)
I found more problems to fix.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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negative value
Fix broken != 0 check.
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Fixes CID1135756.
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
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Fixes CID1135757.
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
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The open_codec_context function, when it fails to find a codec, now
return AVERROR(EINVAL) to signal an error.
Before it would return the stream index, which was always >= 0, and
continue as if a codec was found. This change make it fail faster,
instead of repeated failed tries to decode frames with no codec.
Signed-off-by: Even Wiik Thomassen <e.thomassen@sportradar.com>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
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* commit '48d17ee6dc2b2a552f645484f200c2946bf24607':
api-example: remove an unneeded call to avcodec_get_frame_defaults().
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* commit 'eb891b3114f499e96b9faddd0b0ae856345dfbd9':
Replace all uses of avcodec_free_frame with av_frame_free().
Conflicts:
doc/examples/decoding_encoding.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Fixes use of uinitialized data and crash
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Support the case when multiple frames are contained in a single packet.
In particular, fix fate-samples/lossless-audio/luckynight-partial.shn
sample decoding.
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Set the value on the filter context instead. Simplify.
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Simplify.
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This is required to build with FFmpeg compilation options.
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This codepath is not implemented and just crashes, also its simpler
without special cases, which makes sense for an example
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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* qatar/master:
Add an audio transcoding example.
Conflicts:
configure
doc/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Anton Khirnov <anton@khirnov.net>
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This includes moving libavformat/output-example to doc/examples/output.
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avcodec_close() does nothing in case the argument is NULL. Simplify.
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* commit '5b9c3b4505206143d85398c1410949319fa1180f':
Replace all instances of avcodec_alloc_frame() with av_frame_alloc().
Conflicts:
doc/examples/decoding_encoding.c
doc/examples/muxing.c
ffmpeg.c
libavcodec/alacenc.c
libavcodec/libopenjpegenc.c
libavcodec/libvpxenc.c
libavcodec/pcm.c
libavcodec/xbmenc.c
libavcodec/xwdenc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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That example shows how the decoding process works, not only the
demuxing.
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Fix infinite loop at flushing.
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It uses at least sin()
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"into the doc/examples directory" vs "into doc/examples".
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Adjust the code so that a working ffplay command is printed in the
planar audio case.
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There is no reason why this should copy the audio data in a very
complicated way. Also, strictly write the first plane, instead of
writing the whole buffer. This is more helpful in context of the
example. This way a user can clearly confirm that it works by playing
the written data as raw audio.
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This assumes one audio packet is decoded one time. This is not true:
packets can be partially decoded. Then you have to "adjust" the packet
and pass the undecoded part of the packet to the decode function again.
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Allows to encode to output in case the destination sample format is
different from AV_SAMPLE_FMT_S16.
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the next bump
Add function avfilter_graph_parse_ptr() and favor it in place of
avfilter_graph_parse(), which will be restored with the old/Libav
signature at the next bump.
If HAVE_INCOMPATIBLE_LIBAV_API is enabled it will use the
Libav-compatible signature for avfilter_graph_parse().
At the next major bump the current implementation of
avfilter_graph_parse() should be dropped in favor of the Libav/old
implementation.
Should address trac ticket #2672.
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