The first frame pts must be saved until we have 8 frames since RoQ audio
requires 8 frames in the first packet.

Update FATE references due to encoder delay.

Encoder output is delayed by several frames, so we keep a queue of input
frame timing info to match up with corresponding output packets.

FATE reference updated due timestamp rounding because of resampling from
44100 Hz to 16000 Hz in avconv.

Update FATE references due to encoder delay.

This simplifies matching of timestamps between input frames and output
packets.

Fixes a memleak when using the -frames option with audio.

There is no reason for it to be public, it's only meant to be used
internally.

The caller can achieve the same effect (i.e. getting raw unparsed/mangled
packets) with av_read_frame() and AVFMT_FLAG_NOPARSE |
AVFMT_FLAG_NOFILLIN

Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.

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.

Also fixes stream duration calculation.

The values in the header refer to decoded data, not compressed data.

Do not output an extra packet when out_size is reached.
Also return AVERROR_EOF instead of AVERROR(EIO).

The packet duration is always 28 samples.

This fixes decoding of Bink files with non-multiple-of-16 width.

This fixes doc generation in build tree separate from source.

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

This allows specifying additional directories to search for
@include files.

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

The other fragmentation options (frag_duration, frag_size and
frag_keyframe) are combined with OR, cutting fragments at the
first of the conditions being fulfilled.

Signed-off-by: Martin Storsjö <martin@martin.st>

This enables reordering of UDP packets by default, unless the caller
explicitly sets -max_delay 0.

Signed-off-by: Martin Storsjö <martin@martin.st>

Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.

This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.

Signed-off-by: Martin Storsjö <martin@martin.st>

Documentation for those will be generated automatically.

This will allow us to automatically generate manpages for them.

This allows the caller to get them without special code even after
seeking before receiving any data.

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>

This change avoids accessing the segment map of the previous frame if
segmentation is not enabled for the current frame. The caller of
decode_mb_mode() only calls ff_thread_await_progress() on the reference
segmentation index array if segmentation is enabled, so Chromium's TSAN
will report a race when accessing this data while segmentation is not
enabled.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>