| Commit message (Collapse) | Author | Age |
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This is a new library for audio sample format, channel layout, and sample rate
conversion.
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This partially reverts acb1730218f1c614dc8ca3ba45d9de1e05059515
which would only have needed to change the checksums if channel mixing had
been properly avoided. This changes the output file size reference and the
seek test reference back to the previous values.
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DTS LBR identifier ('dtse') is not included since libavcodec doesn't support it yet.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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The new incremental parser doesn't always clear prev_pkt,
however the packet queue is cleared when seeking. Which leads
to a use-after-free.
Verified using Valgrind.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Aligning nb_samples will give both correct plane pointer alignment and enough
padding for SIMD-optimized functions.
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Fixes a division by 0.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Martin Storsjö <martin@martin.st>
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libschroedinger is the preferred way to decode Dirac video.
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Reduces the amount of upfront data required for cluster parsing
thus decreasing latency on seek and startup.
The change in the seek-lavf_mkv FATE test is due to incremental
parsing no longer reading as much data as the old parser and
thus not having that additional data to generate index entries
based on keyframes. Index entries are added correctly as the
file is parsed.
All FATE tests pass and Chrome has been using this patch for ~6
months without issue.
Currently incremental parsing is not supported for files with
SSA tracks since they require merging packets between clusters.
In this case the code falls back to non-incremental parsing.
Signed-off-by: Aaron Colwell <acolwell@chromium.org>
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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nb_input_files is already an invalid index.
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The mpegts demuxer reads 5 KB at startup just for discovering
the packet size. Since the default avio buffer size is 32 KB,
the seek back to the start will in most cases be within the
avio buffer, and will in most cases succeed even if the actual
protocol isn't seekable.
This makes the demuxer startup faster/with less data when
reading data from a non-seekable input, by not skipping
the first few KB.
If it fails, don't warn if the protocol isn't seekable, making
it behave as before in the failure case.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This allows masking CPU features with the -cpuflags avconv option
which is useful for testing different optimisations without rebuilding.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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These functions were left unused by the lowres removal.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Splitted files don't start always from a sync sample.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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This feature is complex, of questionable utility, and slows down
normal decoding.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This removes all references to AVCodecContext.dsp_mask and marks
it for eviction at the next version bump. It has been superseded
by av_set_cpu_flag_mask() which, unlike this field, works everywhere.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Fail earlier and with a more descriptive error message.
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General cosmetics, such as keeping lines under 80 characters,
fixing a couple of typos (predition -> prediction) and a
general style fix that was pointed out by Derek when I was having
my sliced multithreading patch in review by him.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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rate field in stsd to 0.
Libisomediafile appears to always set this field to zero.
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libavcodec/xxan.c:293:13: warning: variable ‘corr_end’ set but not used
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This simplifies the code a bit and avoids an uninitialized variable warning.
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Fixes jittery video playback of rm files with cook audio.
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Change some lavf tests to avoid resampling and channel mixing.
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This avoids resampling and channel mixing by using a source with
the correct channel layout and sample rate.
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Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
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Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
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The functions operate on the sample level rather than the byte level and work
with all audio sample formats.
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This will allow decoding to md5 and doing a diff comparison to a reference
checksum instead of a fuzzy stddev or oneoff comparison.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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The output format is not always the same as the file extension,
which is sometimes required for correct probing. We can avoid
probing by specifying the format since it is already known.
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Add cvtdq2ps and cvtps2dq to the AVX instruction list.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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The decoder can change the layout and channel count during decoding,
but currently we only validate that the two are compatible when opening
the codec. This checks for incompatibilities after each decoded frame.
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Fix this warning:
libavformat/aviobuf.c:663:20: warning: assignment discards qualifiers from pointer target type
Although this is a public header, it should remain source and
binary compatible.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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This also allows dropping some PPC-specific ugliness from dsputil.[ch].
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This fixes the warning:
libavcodec/mpegaudiodec.c:1704:14: warning: variable ‘out_size’ set but not used
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If a video track specifies a zero frame rate (invalid but occurs),
this results in a division by zero and subsequent undefined conversion
to integer. Setting the default duration from the frame rate only
if the latter is greater than zero avoids such problems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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It avoids some rounding errors.
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This is in line with other demuxers and overall seems more correct
than assuming codec time base.
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This improves output for formats with more than one AVCodec.
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Do not pointlessly call ff_alloc_packet multiple times,
and fix an infinite loop by clamping the maximum
number of bits to target in the algorithm that does
not use lambda.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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