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>

This sends NACK for missed packets and PLI (picture loss indication)
if a depacketizer indicates that it needs a new keyframe, according
to RFC 4585.

This is only enabled if the SDP indicated that feedback is supported
(via the AVPF or SAVPF profile names).

The feedback packets are throttled to a certain maximum interval
(currently 250 ms) to make sure the feedback packets don't eat up
too much bandwidth (which might be counterproductive). The RFC
specifies a more elaborate feedback packet scheduling.

The feedback packets are currently sent independently from normal
RTCP RR packets, which is not totally spec compliant, but works
fine in the environments I've tested it in. (RFC 5506 allows this,
but requires a SDP attribute for enabling it.)
Signed-off-by: Martin Storsjö <martin@martin.st>

This allows calling other dynamic payload handler functions if
needed.
Signed-off-by: Martin Storsjö <martin@martin.st>

The warning is a false positive, but I prefer actually initializing
it over masking it with av_uninit, since the code is not performance
critical.
Signed-off-by: Martin Storsjö <martin@martin.st>

This is a bug from c7d4de3d - if the previous frame wasn't
returned yet (due to missing the final packets), but we have
enough data of it to return the first partition, we write that into
pkt and set returned_old_frame. That commit forgot returning 0 for
the case where this current packet didn't have the end_packet flag
set.
Signed-off-by: Martin Storsjö <martin@martin.st>

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

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

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

If we timed out and consumed a packet from the reordering queue,
but didn't return a packet to the caller, recheck the queue status.
Otherwise, we could end up in an infinite loop, trying to consume
a queued packet that has already been consumed.

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>

If the matrix results in an output channel not getting a contribution
from any input channel and the corresponding input channel does not
contribute to any outputs, we can skip the channel during mixing and
silence it after mixing.

If the matrix results in an input channel not contributing to any output
channels and it is not in the output mix, or if the input channel only
contributes fully to the same output channel, we can skip the channel
during mixing.

If the matrix results in an output channel only getting full
contribution from the corresponding input channel and that input channel
does not contribute to any other output channels, we can skip the
channel during mixing.