full-bandwidth channels.

This reduces high-frequency artifacts and improves the quality of the lower
frequency audio at low bit rates.

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>

Signed-off-by: Mans Rullgard <mans@mansr.com>

Signed-off-by: Mans Rullgard <mans@mansr.com>

in the ac3_fixed encoder.

Factor ffmpeg calls using run_ffmpeg()

Add a THREADS variable to fate calls.

This updates the wmv8-drm reference after c47d3835.
Signed-off-by: Mans Rullgard <mans@mansr.com>

This script was used to import Mike's old fate database and
is no longer needed.
Signed-off-by: Mans Rullgard <mans@mansr.com>

Signed-off-by: Mans Rullgard <mans@mansr.com>

This updates the seek test reference to match de11ee90.  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>

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.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

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>

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

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>

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>

Signed-off-by: Mans Rullgard <mans@mansr.com>

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>

The output was changed by a7827a17.
Signed-off-by: Mans Rullgard <mans@mansr.com>

The reserved bits between PCR base and extension fields must be
set to 1.
Signed-off-by: Mans Rullgard <mans@mansr.com>

Signed-off-by: Mans Rullgard <mans@mansr.com>

Signed-off-by: Mans Rullgard <mans@mansr.com>

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.

Signed-off-by: Mans Rullgard <mans@mansr.com>

Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>

Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>

Originally committed as revision 26342 to svn://svn.ffmpeg.org/ffmpeg/trunk

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

Originally committed as revision 26231 to svn://svn.ffmpeg.org/ffmpeg/trunk

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

it should not be necessary anymore.

Originally committed as revision 26198 to svn://svn.ffmpeg.org/ffmpeg/trunk

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

Originally committed as revision 26134 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 26119 to svn://svn.ffmpeg.org/ffmpeg/trunk