Assemble it already in caf_write_packet(). This has the advantage
of reducing the amount of buffers used; it also allows to avoid
a seek when writing the trailer and avoids function call overhead
(for the avio_w8(); it also reduces codesize).
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
We're not writing a kuki chunk because its contents for Opus are currently
unknown, so it's best if we don't allow the creation of non spec compliant
files.
Signed-off-by: James Almer <jamrial@gmail.com>
caf_write_deinit() would segfault if the CAFStreamContext
couldn't be allocated. Fix this by moving everything from
CAFStreamContext to the ordinary CAFContext; the separation
doesn't make sense for a format with only one stream anyway
and removing it also avoids an indirection.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
If a codec has fixed block_align and frame_size but a given sample has either
priming or remainder frames, a pakt chunk can be written declaring zero packets
and no table, reporting only the samples to be discarded.
Signed-off-by: James Almer <jamrial@gmail.com>
st->duration is not guaranteed to be set, so store the sum of packet durations instead.
Also, set mPrimingFrames and mRemainderFrames to correct values.
Based on a patch by Jun Zhao.
Signed-off-by: James Almer <jamrial@gmail.com>
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
More exactly: Not more than one stream of each type for which
a default codec (i.e. AVOutputFormat.(audio|video|subtitle)_codec)
is set; for those types for which no such codec is set (or for
which no designated default codec in AVOutputFormat exists at all)
no streams are permitted.
Given that with this flag set the default codecs become more important,
they are now set explicitly to AV_CODEC_ID_NONE for "unset";
the earlier code relied on AV_CODEC_ID_NONE being equal to zero,
so that default static initialization set it accordingly;
but this is not how one is supposed to use an enum.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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.
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>
