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>

Partially fixes bug 44.

Deprecate AVCodecContext.directpred

Deprecate AVCodecContext.partitions.

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

Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

They were replaced by (de)muxer private options.

AVStream is no place for it and it's unused outside of ffmpeg anyway.

YUV planes were marked as uint16_t, but they contained signed data.
Fixes issue 1108 and 675.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

ipod shuffle doesn't support anything else.
Signed-off-by: Anton Khirnov <anton@khirnov.net>

This should fix behavior introduced by commit
96573c0d. 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>

The string "FFmpeg" was replaced by "Libav" in metadata that
got encoded in file headers.

This allows for more reproducible results when using multi-threading.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

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>

in the ac3_fixed encoder.

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.

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>

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>

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

The references changed due to r25956.

Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk

Fixes a scr issue reported with dvdauthor ([FFmpeg-user] FFMPEG encoded MPEG-2 video causes error in DVDAuthor)

Originally committed as revision 25512 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 25400 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 25395 to svn://svn.ffmpeg.org/ffmpeg/trunk

This fixes issues if the first DTS is far away from 0.

Originally committed as revision 25383 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 25012 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 25011 to svn://svn.ffmpeg.org/ffmpeg/trunk

Patch by Thierry Foucu, tfoucu at gmail

Originally committed as revision 24379 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 23967 to svn://svn.ffmpeg.org/ffmpeg/trunk

with me that it's unnecessary.

Originally committed as revision 23478 to svn://svn.ffmpeg.org/ffmpeg/trunk

Start them on keyframes when reasonable, and delay writing audio packets
to help ensure that there's audio samples available for the first frame in
clusters.

Patch by James Zern <jzern at google>

Originally committed as revision 23473 to svn://svn.ffmpeg.org/ffmpeg/trunk

This isn't exactly semantically equivalent, but the field has already been
long abused to mean this, and writing it helps in determining a decent cfr
time base when transcoding from a mkv where the video codec stores none (VP8).

Originally committed as revision 23284 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 23248 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 23231 to svn://svn.ffmpeg.org/ffmpeg/trunk