This commit does for AVOutputFormat what commit
20f9727018 did for AVCodec:
It adds a new type FFOutputFormat, moves all the internals
of AVOutputFormat to it and adds a now reduced AVOutputFormat
as first member.
This does not affect/improve extensibility of both public
or private fields for muxers (it is still a mess due to lavd).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Use the stream duration as last resort, as an off-by-one result of the
"st->duration / (caf->packets - 1)" calculation can break playback on some
devices.
Also, don't write the sample_rate value propagated by encoders like libopus.
The sample rate of the audio fed to it is irrelevant after being encoded.
Fixes ticket #9930.
Signed-off-by: James Almer <jamrial@gmail.com>
Do this by using the AVStream's priv_data for the buffer holding
the packet size data.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
If an array for the packet sizes could not be successfully reallocated
when writing a packet, the CAF muxer frees said array, but does not
reset the number of valid bytes. As a result, when the trailer is
written later, avio_write tries to read that many bytes from NULL,
which segfaults.
Fix this by not freeing the array in case of error; also, postpone
writing the packet data after having successfully (re)allocated the
array, so that even on allocation error the file can be correctly
finalized.
Also remove an unnecessary resetting of the number of size entries
used at the end.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
(As long as avio_write() only accepts an int, it makes no sense
to try to support sizes that don't fit into an int.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Frame size of Opus stream was previously presumed here to be 960 samples
(20ms), however sizes of 120, 240, 480, 1920, and 2880 are also allowed.
It can also alter on a per-packet basis and even multiple frames may be
present in a single packet according to the specification, for the sake
of simplicity however, let us assume that this doesn't occur.
Because the mFramesPerPacket field, representing the number of samples
per packet in the ffmpeg terminilogy, is the key factor in calculating
packet durations and all that follows from that (index, bitrate, ...),
it is crucial to get right.
Therefore, if the packet size is not available ahead of time (as it is in
the case of Opus), calculate an average from the stream duration once we
know how many packets there are and update the filed in the header.
This is possible now that the next-API is gone.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
To make it consistent with other muxers.
The user can still control the generic flushing behaviour after write_header
(same way as after packets) using the -flush_packets option, the default
typically means to flush unless a non-streamed file output is used.
Therefore this change should have no adverse effect on streaming, even if it is
assumed that the first packet has a clean buffer, so small seekbacks within the
output buffer work even when the IO context is not seekable.
Signed-off-by: Marton Balint <cus@passwd.hu>