If 384k is too high for the samplerate, choose the closest
possible
Idea to increase the bitrate from: 46439e1562
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes the USE_FLOATS == 0 available to the end user
More float optimizations can easily be added as well now
common code should be factored out into a common file once all
fixed point & floating point optimizations are done, this is to
avoid having to move code back and forth between files.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
QuickTime will play multiple audio tracks concurrently if this flag is
set for multiple audio tracks. And if no subtitle track has this flag
set, QuickTime will show no subtitles in the subtitle menu.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
After making some blind tests on a small collection of music
samples for home usage. It turned out that the default cutoff
was too low.
The impact of filter_size was not clearly distinguishable (the
results were on the edge) with the music samples but turned out
to be clearly audible in some synthetic samples.
Thanks to Daniel for helping out with the listening tests.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
* commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6':
fate-seek: remove use of gnu make 3.82 only private modifier
fate: move vsynth reference files to their own directory
fate: move fate-acodec reference files to their own dir
configure: avplay now depends on avresample
fate: split dependencies for fate-seek tests
Conflicts:
configure
tests/fate/seek.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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.
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.
If either input or output layout is known and the channel counts match,
use the known layout for both. Otherwise choose the default layout based on
av_get_default_channel_layout().
Changed some FATE references due to some WAVE files now having a non-zero
channel mask.
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
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>
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
The Zork PCM decoder does not decode the 1 sample we have correctly, therefore
the encoder based on the decoder is also incorrect. There is no good reason to
keep the encoder.
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
These tests create reference files used for psnr calculation in
the other codec tests. Treating them as (mostly) regular tests
simplifies the makefile and makes them visible in the fate reports.
The latter makes errors in these runs easier to identify.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
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.
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>
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>
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
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
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
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
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
