Commit graph

12 commits

Author SHA1 Message Date
Diego Nieto
c10c55bc5a rtpsource: include config.h header to avoid g_memdup2 link issue
Without adding the header a link issue related g_memdup2 might happen.
In versions below 2.67.4 that symbol is manually introduced in the
meson config files.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7933>
2024-11-21 01:11:22 +00:00
Ognyan Tonchev
03b6226772 rtpmanager: skip RTPSources which are not ready in the RTCP generation
If a stream has an 'irregular' frame rate (e.g. metadata) RTCP SR
may be generated way too early, before the RTPSource has received
the first packet after Latency was configured in the pipeline.
We skip such RTPSources in the RTCP generation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7740>
2024-10-29 02:10:47 +00:00
Jan Schmidt
ef8dfd7873 rtpmanager: save the report block statistics in each RTPSource
Move RB info from receiver reports into the internal source that the RR
are about, and deprecate (but retain) the old mapping where each
external source has only a single RB entry in the rtp statistics.

The old method is broken if a remote peer uses a single ssrc to send
receiver reports for more than one of our internal sources, other
as multiple RB in a single packet, or alternate RB in different reports.
In each case only the most recent entry was kept, overwriting data for
other internal sources.

In multicast scenarios each internal source may receive multiple
receiver reports from different peers. To support that, all received
RR's are now stored into a hash table indexed by the sender's SSRC,
and all RRs are placed into an array when generating statistics, so
that the information from all peers is retrievable.

The current deficient behaviour (adding RB info into non-internal RTPSources) is
deprecated but kept in order to be backward compatible, and retained
that way in the generated statistics structure.

Refs
[1] https://tools.ietf.org/html/rfc3550#section-6.4.1

Based on a patch by Fede Claramonte <fclaramonte@twilio.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7424>
2024-10-11 05:20:22 +00:00
Sebastian Dröge
6233eb0ff3 common: Stop using GQuark-based GstStructure field name API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432>
2024-09-26 19:21:29 +03:00
Sebastian Dröge
2a2ef23829 rtpsource: Don't store invalid running times and calculate with it
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet
then this can't be used for bitrate estimation, and also not for
constructing the next RTCP SR. Both would end up with completely wrong
values, and an RTCP SR with wrong values can easily break
synchronization in receivers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
2023-09-23 07:39:00 +00:00
Tim-Philipp Müller
8222b97331 rtpmanager: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Jan Schmidt
cb225b3682 rtpsource: Track the seqnum for senders
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.

Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
2022-11-23 10:26:29 +00:00
Sebastian Dröge
bd5a4d321b rtpsource: Don't do probation for RTX sources
Disable probation for RTX sources as packets will arrive very
irregularly and waiting for a second packet usually exceeds the deadline
of the retransmission.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:18 +00:00
Sebastian Dröge
72b6dabd32 rtpsession: Remember the corresponding media SSRC for RTX sources
This allows timing out the RTX source and sending BYE for it when the
actual media source belonging to it is timed out.

This change only applies to sending sources from this session.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Sebastian Dröge
d5c072fadd rtpsource: Rename rtp_source_update_caps to rtp_source_update_send_caps
To make it clear that this is only used for sending RTP sources.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Guillaume Desmottes
8bbdd9addb rtpsource: fix rtp_source_get_nack_deadlines doc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1775>
2022-02-22 09:40:35 +00:00
Thibault Saunier
5ff769d731 Move files from gst-plugins-good into the "subprojects/gst-plugins-good/" subdir 2021-09-24 16:13:50 -03:00
Renamed from gst/rtpmanager/rtpsource.c (Browse further)