Ensure that the RTP payload size is at least one TS packet
(188 bytes) when packetizing MPEG-TS.
Previously, small payload sizes were silently rounded up,
which could lead to incorrect behavior. Return EINVAL
with a clear error message instead.
Signed-off-by: Devraj Ajmera <devrajajmera077@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The ssrc option used AV_OPT_TYPE_INT with range [INT_MIN, INT_MAX],
but the underlying struct field is uint32_t and RFC 3550 defines SSRC
as a 32-bit identifier covering the full [0, UINT32_MAX] range. This
caused ffmpeg to reject any SSRC value above INT_MAX (~2.1 billion),
which is roughly half of all valid values.
The auto-generated fallback path (av_get_random_seed()) already
returns uint32_t and can produce values above INT_MAX, creating an
inconsistency where the automatic path succeeds but the explicit
-ssrc option rejects the same value.
Change the option to AV_OPT_TYPE_UINT with range [0, UINT32_MAX]
to match the field type and allow the full identifier space.
Fixes ticket #9080.
Signed-off-by: Neko Asakura <neko.asakura@outlook.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix#20712
When print_sdp() fails during rtp streaming, sch_start()
returns error code without calling wirte_trailer, causing
a memory leak.
The avformat_free_context() will call the mux->deinit, so
we should add a rtp_deinit() to free all internal allocations.
Signed-off-by: Jack Lau <jacklau1222gm@gmail.com>
Not all rtp formats require the video dimensions to be available
up front. H264 and HEVC will send them as stream parameters.
The flag is restrictive and prevents RTP repacketization
without parsing the codec information out of the stream.
This change checks to see if the codec parameters are available
on the rtp formats that need it.
Signed-off-by: Koushik Dutta <koushd@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add RTP packetizer and depacketizer according to (most)
of the official AV1 RTP specification. This enables
streaming via RTSP between ffmpeg and ffmpeg and has
also been tested to work with AV1 RTSP streams via
GStreamer.
It also adds the required SDP attributes for AV1.
AV1 RTP encoding is marked as experimental due to
draft specification status, debug amount reduced
and other changes suggested by Tristan.
Added optional code for searching the sequence
header to determine the first packet for broken
AV1 encoders / parsers.
Stops depacketizing on corruption until next keyframe,
no longer prematurely issues packet on decoding if
temporal unit was not complete yet.
Change-Id: I90f5c5b9d577908a0d713606706b5654fde5f910
Signed-off-by: Chris Hodges <chrishod@axis.com>
Signed-off-by: Ronald S. Bultje <rsbultje@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>
Unnecessary since acf63d5350adeae551d412db699f8ca03f7e76b9;
also avoids relocations.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
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>
RTCP synchronization packet was broken since commit in ffmpeg version > 2.8.3
(commit: e04b039b15) Since this commit (2e814d0329)
"rtpenc: Simplify code by introducing a macro for rescaling NTP timestamps", NTP_TO_RTP_FORMAT
uses av_rescale_rnd() function to add the data to the packet.
This causes an overflow in the av_rescale_rnd() function and it will return INT64_MIN.
Causing the NTP stamp in the RTCP packet to have an invalid value.
Github: Closes#182
Reverting commit '2e814d0329aded98c811d0502839618f08642685' solves the problem.
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 '4f6cd883f06f7893a2b60a41e7a4f8ae633dac2f':
rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f8c01257f93ceda3e03bc4e540a51022d1e2bff2':
rtpenc: Always do the default initialization regardless of codecs
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1fc64e2e07787bbca82a72c146588e850e6d098a':
rtpenc: Write conditional statements on separate lines
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0662440b991361fdb5e732712d997a73e4692e34':
rtpenc_aac: Set a default value for max_frames_per_packet at init
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Instead check the timestamps while muxing, to avoid buffering a
too long timestamp range into one single packet.
This makes the AMR and AAC packetization slightly less efficient,
since we set a possibly unnecessarily high max_frames_per_packet.
(These packetizers end up doing a memmove of the TOC bytes if
sending a packet before max_frames_per_packet is achieved, and
we end up setting max_frames_per_packet to a value that should
be high enough for most uses.)
All packetizers that use max_frames_per_packet now set it either
to a default value, or to a value calculated based on other
parameters, so none of them rely on the previous default setting.
For iLBC, copy one frame at a time, to allow checking the timestamp
range for each of them - basically doing potentially multiple
loops to simplify the code instead of trying to calculate the
number of frames to buffer while honoring s1->max_delay.
This is in preparation for reducing the coupling between libavformat
and libavcodec, by not having the muxers use the encoder field
frame_size (which may not be available during e.g. stream copy).
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids having to jump to the defaultcase in the switch. Manually
override the stream time base back to 90 kHz for the few audio codecs
that don't use the sample rate as time base (mp2, mp3).
Signed-off-by: Martin Storsjö <martin@martin.st>
Marvin Scholz