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Fix handling of input if not in native endianness, and add support for
9/10-bit output. This allows us to force endianness of YUV420P 9/10bit
in the H264/10bit fate tests, which should fix them on big-endian
systems.
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This should fix behavior introduced by commit
96573c0d7605672d69b42ae1dcf18764ce47c71a. Av_rescale_rnd() is not
lossless so if two timestamps are equal after being rescaled they are
not always actually identical. This patch use av_compare_ts() to get
always a correct result.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 78431098f9e306ebe27e7698d0ae539e3df2afe9)
Tested with mplayer based on this report
http://thread.gmane.org/gmane.comp.video.mplayer.user/66043/focus=66063
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
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Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Signed-off-by: Anton Khirnov <anton@khirnov.net>
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The string "FFmpeg" was replaced by "Libav" in metadata that
got encoded in file headers.
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This allows for more reproducible results when using multi-threading.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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full-bandwidth channels.
This reduces high-frequency artifacts and improves the quality of the lower
frequency audio at low bit rates.
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This makes the AC3 encoder use the shared fixed-point MDCT rather
than its own implementation. The checksum changes are due to
different rounding in the MDCT.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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in the ac3_fixed encoder.
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This updates the wmv8-drm reference after c47d383.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This updates the seek test reference to match de11ee9. Before this
change, most of the seeks requested positions before the supposed
start of the file (the preroll time), resulting in the first packet
being returned. With the preroll subtracted, some of these seeks
will land within the file and some beyond the end, thus returning
a different set of packets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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There is no need to expand to 16-bits. Just use memcpy() to copy the raw data.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.
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 patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder. I tested lowering in
increments of 100. From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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The output was changed by a7827a17c6b3388322350456d445c94b3a82cd25.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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The reserved bits between PCR base and extension fields must be
set to 1.
Signed-off-by: Mans Rullgard <mans@mansr.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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Instead of saving huge raw files, use the md5: output pseudo-protocol
to calculate the checksum of the file directly. This is especially
useful when testing on remote targets as it avoids transferring 3.6GB
over the network.
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Originally committed as revision 26342 to svn://svn.ffmpeg.org/ffmpeg/trunk
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This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.
Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 26231 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.
Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
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This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.
Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.
Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 26119 to svn://svn.ffmpeg.org/ffmpeg/trunk
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code.
Originally committed as revision 26112 to svn://svn.ffmpeg.org/ffmpeg/trunk
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This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.
Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
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due to VC-1 decoder overreads resulting in different output.
Originally committed as revision 26055 to svn://svn.ffmpeg.org/ffmpeg/trunk
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to complement the existing video-only decode test.
Originally committed as revision 26054 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Whether the behaviour for streams using scrambling makes sense
is unclear.
Originally committed as revision 26053 to svn://svn.ffmpeg.org/ffmpeg/trunk
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The references changed due to r25956.
Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.
Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
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This includes a revert of r25840
Originally committed as revision 25842 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Originally committed as revision 25840 to svn://svn.ffmpeg.org/ffmpeg/trunk
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