Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
It will not be set if the stream codec context is not the encoding
context. Use av_get_audio_frame_duration() instead, it should work for
all audio codecs supported by the muxer.
The chunk size is limited to UINT16_MAX (written by avio_wb16), so make
sure that the packet size is not too large.
Such large frames need to be split into slices smaller than 64 kB, but
that is currently supported neither by the rv10/rv20 encoders nor the rm
muxer.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The chunk size is limited to 0xFFFF (written by avio_wb16), so make
sure that the packet size is not too large.
Such large frames need to be split into slices smaller than 64 kB, but
that is currently supported neither by the rv10/rv20 encoders nor the rm
muxer.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
See Ticket244
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
Since 596e5d4783, this is not necessary anymore. It also allows to
actually disable the flushing, improving write performance (but
possibly giving worse latency in real-time streaming).
Signed-off-by: Martin Storsjö <martin@martin.st>
This is consistent with stdio and is what we want to do in all cases.
Fixes a bug in the voc muxer which didn't flush in write_trailer()
previously. This is the cause of the change in the test results.
It is possible that just extending the RMMuxContext.streams
array would avoid it.
It is also possible that two audio streams will fail to mux
correctly as well, though at least it should not crash for
this reason.
I do not feel like checking either of these.
This patch fixes trac issue #1022 (at least it makes it
exit with a proper error message instead of crashing).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
In the name of consistency:
put_byte -> avio_w8
put_<type> -> avio_w<type>
put_buffer -> avio_write
put_nbyte will be made private
put_tag will be merged with avio_put_str
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 77eb5504d3)
In the name of consistency:
put_byte -> avio_w8
put_<type> -> avio_w<type>
put_buffer -> avio_write
put_nbyte will be made private
put_tag will be merged with avio_put_str
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This also lists the objects from those two libraries as internal (by adding
the ff_ prefix) so that they can then be hidden via linker scripts.
(cherry picked from commit c6610a216e)
other audio codecs rather than only AC-3, so add some code that makes
word byte-swapping only happen for AC-3, not for all audio codecs.
Patch by Francesco Lavra <francescolavra interfree it>.
Originally committed as revision 23361 to svn://svn.ffmpeg.org/ffmpeg/trunk
single index chunk) and is always empty anyway. See "[PATCH] rmenc.c: remove
index writing" thread.
Originally committed as revision 18119 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reason for this is that there are no shared entries in the demuxer/muxer
context, making it a mystery as to why it was shared between the two. See
"[PATCH] clean rmdemux/muxcontext" patch on mailinglist.
Originally committed as revision 16111 to svn://svn.ffmpeg.org/ffmpeg/trunk
patch by Stefano Sabatini, stefano.sabatini-lala poste.it
along with some spelling/consistency fixes for the long names by me
Originally committed as revision 13649 to svn://svn.ffmpeg.org/ffmpeg/trunk
Vittorio Giovara
Anton Khirnov
Michael Niedermayer
Luca Barbato
Federico Tomassetti
Andreas Cadhalpun
Diego Biurrun
Clément Bœsch
Piotr Bandurski
Reimar Döffinger
Martin Storsjö
Mans Rullgard
Diego Elio Pettenò
Francesco Lavra
Jean-Daniel Dupas
Stefano Sabatini
Ronald S. Bultje
Aurelien Jacobs
Justin Ruggles