c29d81e736
=== 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 24130234cd9dd733116d17b724ea4c8e12ce097a. 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>
FFmpeg README
FFmpeg is a collection of libraries and tools to process multimedia content such as audio, video, subtitles and related metadata.
Libraries
libavcodecprovides implementation of a wider range of codecs.libavformatimplements streaming protocols, container formats and basic I/O access.libavutilincludes hashers, decompressors and miscellaneous utility functions.libavfilterprovides a mean to alter decoded Audio and Video through chain of filters.libavdeviceprovides an abstraction to access capture and playback devices.libswresampleimplements audio mixing and resampling routines.libswscaleimplements color conversion and scaling routines.
Tools
- ffmpeg is a command line toolbox to manipulate, convert and stream multimedia content.
- ffplay is a minimalistic multimedia player.
- ffprobe is a simple analysis tool to inspect multimedia content.
- Additional small tools such as
aviocat,ismindexandqt-faststart.
Documentation
The offline documentation is available in the doc/ directory.
The online documentation is available in the main website and in the wiki.
Examples
Coding examples are available in the doc/examples directory.
License
FFmpeg codebase is mainly LGPL-licensed with optional components licensed under GPL. Please refer to the LICENSE file for detailed information.
Contributing
Patches should be submitted to the ffmpeg-devel mailing list using
git format-patch or git send-email. Github pull requests should be
avoided because they are not part of our review process and will be ignored.