Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: James Almer <jamrial@gmail.com>

The packet counting based approach caused excessive sdt/pat/pmt for VBR, so
let's use a timestamp based approach instead similar to how we emit PCRs.
SDT/PAT/PMT period should be consistent for both VBR and CBR from now on.

Also change the type of sdt_period and pat_period to AV_OPT_TYPE_DURATION so no
floating point math is necessary.

Fixes ticket #3714.
Signed-off-by: Marton Balint <cus@passwd.hu>

Fixes #8031.

English was used before.
Signed-off-by: Marton Balint <cus@passwd.hu>

better quality without speedloss
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>

this gives better frequency response

update swresample fate and other fates
that depend on resampling
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>

According to specification max value allowed is 0x6000.
Fixes #5862.
Signed-off-by: Paul B Mahol <onemda@gmail.com>

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>

Compute individual stream durations in matroska muxer.
Write them as string tags in the same format as mkvmerge tool does.
Signed-off-by: Sasi Inguva <isasi@google.com>

Fixes Ticket4540
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The prediction used in the encoder was not correct
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The actual predictor value, set by the trellis code, never
was written back into the variable that was written into
the block header. This was accidentally removed in b304244b.

This significantly improves the audio quality of the trellis
case, which was plain broken since b304244b.

Encoding IMA QT with trellis still actually gives a slightly
worse quality than without trellis, since the trellis encoder
doesn't use the exact same way of rounding as in
adpcm_ima_qt_compress_sample and adpcm_ima_qt_expand_nibble.

Fixes part of Ticket3701
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

adpcm_ima_qt does not produce reproducible results, so it is temporarily
disabled (see #3701).
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This avoids misleading encoder names like "encoder = prores"
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Partially undoes commit 2c4e08d8:

    riff: always generate a proper WAVEFORMATEX structure in
    ff_put_wav_header

A new flag, FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX, is added to force the
use of WAVEFORMATEX rather than PCMWAVEFORMAT even for PCM codecs.

This flag is used in the Matroska muxer (the cause of the original
change) and in the ASF muxer, because the specifications for
these formats indicate explicitly that WAVEFORMATEX should be used.

Muxers for other formats will return to the original behavior of writing
PCMWAVEFORMAT when writing a header for raw PCM.

In particular, this causes raw PCM in WAV to generate the canonical
44-byte header expected by some tools.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This fixes rounding differences between platforms
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

If 384k is too high for the samplerate, choose the closest
possible

Idea to increase the bitrate from: 46439e15Signed-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>

with this change QuickTime is able to play u8 aiff file generated by FFmpeg
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

This fixes a regression that apparently was missed when switching to the
in af resampler
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Inspired-by code from af_resample.c written by Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This fixes a bogus bitrate value in the header of WAV files with
alaw/ulaw audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>

Some of the FATE changes are due to off-by-one different rounding being used
(lrintf vs av_rescale_q).
Some fate changes are due to 1 audio frame less being encoded (the new variant seems
matching what qatar does and according to ffprobe its closer to the requested duration)
the mapchan feature sadly is lost in this commit because it depends on resampling
being done in ffmpeg.c which is now moved completely into the av filter layer
-async is broken after this commit, this will be fixed in subsequent commits
the new filter reconfiguration system is flawed and will drop a frame on each
parameter change which is why the nelly moser checksums need updating.

Conflicts:

	ffmpeg.c
	tests/ref/fate/smjpeg

This avoids resampling and channel mixing by using a source with
the correct channel layout and sample rate.

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.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>

Since we cannot specify decode parameters (and also because
it is better in principle) the 1-channel reference file
needs to be enabled, too.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>