Handle pred_flag parameter not given to get_mvdata_interlaced()
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Previously, for broken frames, we only returned the first partition
of the frame (we would append all the received packets to the packet
buffer, then set pkt->size to the size of the first partition, since
the rest of the frame could have lost data inbetween) - now instead
return the full buffered data we have, but don't append anything more
to the buffer after the lost packet discontinuity. Decoding the
truncated packet should hopefully get better quality than trimming out
everything after the first partition.
Signed-off-by: Martin Storsjö <martin@martin.st>

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

This is required by RFC 3550 (section 6.5):

   The list of items in each chunk MUST be terminated by one or more
   null octets, the first of which is interpreted as an item type of
   zero to denote the end of the list.

This was implicitly added as padding before, unless the host name
length matched up so no padding was added.

This makes wireshark parse the packets properly if other RTCP items
are appended to the same packet.
Signed-off-by: Martin Storsjö <martin@martin.st>

Use the libavutil replacement.

This matches the AVInputFormat.read_packet() API.

Add some additional checks for EOF and print error messages on an incomplete
header or packet.

FATE reference updated for id-cin-video due to the demuxer no longer
returning a partial video packet at EOF.

chunk_size is unsigned 32-bit, but av_get_packet() takes a signed int as the
packet size.

Also, do not allow seek-by-byte, as there is no way to find the next packet
boundary.

Also, use 1 / sample_rate for audio stream time_base.

Avoids using unsupported parameters and signed integer overflows.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Also do not unnecessarily skip 0 bytes.

Nothing in the AU specification sets a limit on channel count.
We only need to avoid an overflow in the packet size calculation.

It is already set by av_get_packet() even for partial reads.

This allows the caller to either include them (and get more packets
decoded, but possibly some nonperfect frames), or discard them (by
setting fflags=discardcorrupt).
Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Diego Biurrun <diego@biurrun.de>

This uses page duration instead of byte size to determine when to buffer
the page. Also, it tries to avoid continued pages by buffering the current
page if there are already packets in the page and adding the next packet
would require it to be continued on a new page. This can improve seeking
performance.

The default page duration is 1 second, which is much saner than filling
all page segments by default.

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