5 FATE test references updated due to using demuxer-generated timestamps that
are either not sample-accurate or are slightly off in the input file.

The packet duration is always 28 samples.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>

Also, do not give AVCodecContext.frame_size priority for muxing.

Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
             by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
                 using the packet size and average bit rate.

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.

Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.

The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.

compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

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

Update FATE test to reflect delayed video due to the file having audio-only
frames prior to the first frame with video.

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

Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.

Update some demuxing and seeking fate tests.

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>

The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.

Demuxers are not supposed to set it.
Set stream timebase and framerates instead (this is a cfr container with
no timestamps).

We already have sufficient coverage for 16-bit pcm.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

This avoids breaking fate every time the lavc version is bumped.
Signed-off-by: Martin Storsjö <martin@martin.st>

Based on the patch by Phil Barrett.

ProRes codes chroma blocks in 444 mode in different order than luma blocks,
so make both decoder and encoder read/write chroma blocks in right order.

Reported by Phil Barrett

Deprecate corresponding AVCodecContext fields.

WavPack has a comprehensive test suite, and a bunch
of corner cases.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

Neither of those is guaranteed to be connected to framerate in any way
(if it even exists).

Fixes bug 56.

r_frame_rate should in theory have something to do with input framerate,
but in practice it is often made up from thin air by lavf. So unless we
are targeting a constant output framerate, it's better to just use input
stream timebase.

Brings back dropped frames in nuv and cscd tests introduced in
cd1ad18a

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

It is not supposed to be done outside lavc.

This is basically a revert of 818062f2.

It is unclear what issue this was supposed to fix, if it reappears again
it will have to be fixed in a more proper place.

The wtv-demux test change is because the sample starts with a B-frame.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

Output packet size should match avctx->block_align. The target output packet
size is 1024 bytes.
Before:
mono   - 1024 samples -> 512 bytes
stereo - 2048 samples -> 2048 bytes
After:
mono   - 2048 samples -> 1024 bytes
stereo - 1024 samples -> 1024 bytes

According to unofficial documentation, the video rate is locked to the audio
sample rate. This results in proper synchronization of audio and video
timestamps from the demuxer. This only works if the first audio packet occurs
before the first video packet or the audio sample rate is the default rate of
11111 Hz, both of which are true for all samples in our archive.