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>
These functions already free it themselves before they allocate the new
extradata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It is not uncommon to find code where the caller thinks to know better
what the return value should be than the callee. E.g. something like
"if (av_new_packet(pkt, size) < 0) return AVERROR(ENOMEM);". This commit
changes several instances of this to instead forward the actual error.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
=== PROBLEM ===
I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line:
ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4
FFmpeg then fail to record audio and output this logs:
[rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40)
[rtsp @ 0xcda1f0] Error parsing AU headers
...
[rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format
In SDP provided by my RTSP server I had this fmtp line:
a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3;
In FFmpeg code, I found a check introduced by commit 24130234cd. It disallows values greater than 32 for fmtp line parameters.
RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line.
However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55.
=== FIX ===
As each parameter may have its own min and max values
I propose to introduce a range for each parameter.
For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges.
This patch fix my problem and I now can record my RTSP AAC stream to mp4.
It has passed the full fate tests suite sucessfully.
Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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.
* commit 'bb4a310bb85f43e62240145a656b1e5285b14239':
rtpdec: Don't free the payload context in the .free function
Conflicts:
libavformat/rtpdec_latm.c
libavformat/rtpdec_mpeg4.c
libavformat/rtpdec_mpegts.c
libavformat/rtpdec_xiph.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5d8cae45737bed6239bd6b6e0698802dbe1463c8':
rtpdec: Get rid of all trivial .alloc/.free functions
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Many of these functions were named foo_free_context, and since
the functions no longer should free the context itself, only
allocated elements within it, the previous naming was slightly
misleading.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes it more consistent with depacketizers that don't have any
.free function at all, where the payload context is freed by the
surrounding framework. Always free the context in the surrounding
framework, having the individual depacketizers only free any data
they've specifically allocated themselves.
This is similar to how this works for demuxer/muxers/codecs - a
component shouldn't free the priv_data that the framework has
allocated for it.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '0307cc2253e76772b1c645ac6117d08da87a147c':
rtpdec: pass an AVFormatContext to ff_parse_fmtp()
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c6f1dc8e4cd967ae056698eafb891a08003c211c':
rtpdec: Move setting the parsing flags to the actual depacketizers
rtpdec: Split handling of mpeg12 audio/video to a separate depacketizer
Conflicts:
libavformat/rtpdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a717f9904227d7979473bad40c50eb40af41d01d':
mpegts: Share the cleanup code between the demuxer and lavf-internal parser functions
rtpdec_mpeg4: Return one AAC AU per AVPacket
ppc: Include string.h for memset
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes the returned data valid to stream copy into other
containers as well, not only for decoding straight away.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
lavc: Move vector_fmul_window to AVFloatDSPContext
rtpdec_mpeg4: Check the remaining amount of data before reading
Conflicts:
libavcodec/dsputil.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '977d4a3b8a2dbc2fb5e747c7072485016c9cdfaa':
rtpdec_mpeg4: Check the return value from malloc
srtp: Mark a few variables as uninitialized
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '90c784cc13f6bf21a8eb69f3b88b50c7a70f6c59':
rtpdec: Pass the sequence number to depacketizers
configure: Make avconv depend on null, anull and resample filters
Conflicts:
configure
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Andreas Rheinhardt