| Commit message (Collapse) | Author | Age | Files | Lines |
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Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Tested to be bit-exact across x86-64, x86-32 and ppc.
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Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Update FATE references due to encoder delay.
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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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5 FATE test references updated due to using demuxer-generated timestamps that
are either not sample-accurate or are slightly off in the input file.
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The packet duration is always 28 samples.
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
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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Signed-off-by: Mans Rullgard <mans@mansr.com>
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This allows for testing floating-point audio encoders across different
platforms where exact comparisons are unreliable due to float rounding
differences.
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This will allow for comparing decoded output to the original source when the
decoded size is not exactly the same as the original size.
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This will allow comparison to original pre-encoded content instead of
comparing to expected decoded output.
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This will allow adjusting for any encoder or decoder delay when doing
comparisons.
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
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Also, do not give AVCodecContext.frame_size priority for muxing.
Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
using the packet size and average bit rate.
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It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
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Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
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Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.
CC:libav-stable@libav.org
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Update FATE test to reflect delayed video due to the file having audio-only
frames prior to the first frame with video.
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Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.
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Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
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Update some demuxing and seeking fate tests.
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This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.
This makes the decoder output have double the magnitude
compared to before.
The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.
Signed-off-by: Martin Storsjö <martin@martin.st>
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The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.
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Demuxers are not supposed to set it.
Set stream timebase and framerates instead (this is a cfr container with
no timestamps).
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We already have sufficient coverage for 16-bit pcm.
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related to b7165426917f91ebcad84bdff366824f03b32bfe
Error messages and audible artifacts were fixed in that commit.
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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This avoids breaking fate every time the lavc version is bumped.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Based on the patch by Phil Barrett.
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ProRes codes chroma blocks in 444 mode in different order than luma blocks,
so make both decoder and encoder read/write chroma blocks in right order.
Reported by Phil Barrett
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Deprecate corresponding AVCodecContext fields.
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