Update FATE references due to encoder delay.

Update FATE references due to encoder delay.

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.

Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.

Update some demuxing and seeking fate tests.

This should have been updated in b590f3a7.

Current code compares the desired recording time with InputStream.pts,
which has a very unclear meaning. Change the code to use actual
timestamps of the frames passed to the encoder.

In several tests, one less frame is encoded, which is more correct.

In the idroq test one more frame is encoded, which is again more
correct.

Behavior with stream copy should be unchanged.

There is no point in this test using the RM format.
Signed-off-by: Mans Rullgard <mans@mansr.com>

This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.

The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we
do not use delta entries or slices, only StreamOffsets. OPAtom seeking
basically works like before.

This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.

Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and
Reimar Döffinger <Reimar.Doeffinger@gmx.de>.

Changed av_calloc to av_mallocz, added overflow checks.

this removes 2 redundant tests for pcm in mkv.
we can add the coverage back in later as fate-lavf tests if needed.

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

this was forgotten when the encoder was removed

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>

The input file for this test is no longer generated.
Signed-off-by: Mans Rullgard <mans@mansr.com>

Update FATE references accordingly.

They were replaced by (de)muxer private options.

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>

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.

in the ac3_fixed encoder.

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.

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 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

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

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

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 25204 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 24345 to svn://svn.ffmpeg.org/ffmpeg/trunk

Originally committed as revision 23506 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