The minimum valid packet length is 28, given that the length includes
the packet header.
This didn't cause any issues so far as the code did not care about the
last two fields in the SR section, but will be relevant in a future
commit.
Change delta_timestamp to int32_t and add explicit cast to handle
RTP timestamp wraparound correctly. This fixes implementation-defined
behavior when computing negative timestamp differences due to 32-bit
wraparound.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The rtp_set_prft() function incorrectly calculates the timestamp delta
when RTP timestamps wrap around the 32-bit boundary. The current code:
delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp;
treats both timestamps as large positive values, causing wraparound to
produce a large negative delta instead of the correct small positive delta.
For example, with a 90kHz video clock:
- last_rtcp_timestamp = 0xFFFFFF00 (near wraparound)
- timestamp = 0x00000100 (after wraparound)
- Current result: delta ≈ -4.3 billion ticks ≈ -47,721 seconds
- Expected result: delta ≈ +512 ticks ≈ +0.006 seconds
This causes prft->wallclock to jump backward by approximately:
- 90kHz video: ~47,721 seconds (~13.25 hours)
- 48kHz audio: ~89,478 seconds (~24.9 hours)
- 8kHz audio: ~536,871 seconds (~6.2 days)
Fix by casting the subtraction result to int32_t, which correctly
handles wraparound through modular arithmetic:
delta_timestamp = (int32_t)(timestamp - s->last_rtcp_timestamp);
This ensures the delta is always in the range [-2^31, 2^31-1], making
wraparound produce the correct small positive values.
Fixes timing jumps in applications that rely on Producer Reference Time
for media synchronization.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit will properly set the duration field of Opus AVPackets.
Currently, duration is set to 0 on Opus packets from the RTP demuxer.
The Ogg muxer depends on the duration field to properly compute the page granule
value. Without a proper duration, the granule will be wrong, and result in
negative pts values in ogg files.
See oggenc.c:657 (ogg_write_packet_internal)
This commit calculates using the opus_duration function, which was copied
from oggparseopus.c
I moved this functionality and the existing opus extradata functionality
(added by me in 6c24f2b) into a new rtpdec_opus.c file.
Reviewed-by: Tristan Matthews <tmatth@videolan.org>
Signed-off-by: Marvin Scholz <epirat07@gmail.com>
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>
Timestamp difference is available in media timebase (1/90K) where as
rtcp time is in the default microseconds timebase. This patch fixes
the calculated prft wallclock time by rescaling the timestamp delta
to the microseconds timebase.
Signed-off-by: James Almer <jamrial@gmail.com>
Besides avoiding allocations this also fixes a design defect of
ff_rtp_send_punch_packets: It did not return an error in case of
these allocations failed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes two warnings:
libavformat/rtpdec.c:155:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
libavformat/rtpdec.c:168:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
This adds partial support for the RFC 4175 (raw video over RTP). The
only supported formats are the YCbCr-4:2:2 8 bit because it's natively
supported by FFmpeg with pixel format UYVY, and 10 bit which requires
the vrawdepay codec to convert the payload in a format handled by
FFmpeg.
Signed-off-by: Damien Riegel <damien.riegel@savoirfairelinux.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
When ffplay is used to play from the RTSP URL that serves 24 bit audio
content, ffplay fails to recognize the audio codec format. The attached
patch adds support for playing 24 bit audio content over RTSP by
defining a dynamic payload handler for "L24".
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add new mime types AAL2-G726 for g726 as suggested in rfc 3551.
This patch will break interaction with applications that incorrectly
use big-endian G.726 with mime type G726 but we know of at least one
device (DVTel camera) that correctly implements the rfc, so do the same.
Fixes ticket #5890.
It doesn't matter what the actual reason for not returning
an AVPacket was - if we didn't return any packet and we have
the next one queued, parse it immediately. (rtp_parse_queued_packet
always consumes a queued packet if one exists, so there's no risk
for infinite loops.)
Signed-off-by: Martin Storsjö <martin@martin.st>
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.
This commit print as AV_LOG_VERBOSE the jitter buffer
size. It might be the default value or the value set by application.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit adds a warning trace when jitter buffer
is full. It helps to understand leading decoding issues.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This commit adds an error trace when jitter buffer
is full. It helps to understand leading decoding issues.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit print as AV_LOG_INFO the jitter buffer
size. It might be the default value or the value set by application.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This makes more sense than mapping to AV_CODEC_ID_SUBRIP. Nothing
indicates that a T.140 track contains subrip sub-titles.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '353b492d0f2a21ae8eb829db1ac01b54b2a4d202':
rtpdec: Change enc_name to a pointer instead of a fixed-size buffer
Merged-by: Michael Niedermayer <michaelni@gmx.at>
