* Do not restart receiver on codec change mid-session.
This is not supported and was an erroneous change during the
receiver_base/buffer_base + RTP stream restart consolidation. Also make
the codec munger creation more resilient.
* fix test
In some paths, it is better to preserve pre-recorded time. So, make the
base implementations preserve the RTCP Sender Report receive time.
Also, add a method to enable forwarding packet arrival time. Could be
used across relay.
* Rework receiver restart.
- Protect against concurrent restarts
- Clean up and consolidate code around restarts
- Use `RestartStream` of buffer rather than creating new buffers.
* fix test
* Minor refactor in buffer base and audio level
- Make a function for `restartStream`. Will be useful
when external signal needs to restart a stream. Also restart all the
bits (audio level, dd parser and frame rate calculator)
- make an audio level mode with RTP timestamp so that some state can be
moved out of buffer base
* clean up
* log restart
This is not all of it as it is not possible (or at least I do not know
of a way) to get all suggestions for a repo/project. Did this via loop
searching mainly and taking the modernize suggestions.
* Resolve RTX pair via OnTrack also.
In simulcast probing path, the interceptor chain is not invoked
for primary stream. Not sure if this is a recent change. Due to this,
the RTX pair does not get resolved.
Use the onTrack callback to resolve the pair.
* remove debug
* Return extended sequence number only and not packet.
Callers need only the extended sequence number.
Extended packet could get release if the forwarder processes it before
caller accesses it causing a data race.
* grow bucket in a go routine
* Refactor receiver and buffer into Base and higher layer.
To be able to share code/functionality with relay.
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* clean up
* deps
* fix test
* fix test
* Store buffer after creating it.
Also changing signature of creator function as it could call TrackInfo()
and get into a deadlock.
* fix double unlock
* add some more debug logging
* Refactor receiver and buffer into Base and higher layer.
To be able to share code/functionality with relay.
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* clean up
* deps
* fix test
* fix test
* Some logging changes.
Trying to chase a case of large sequence number gap on subscriber side
where packets are sent after a long time.
* return values instead of logging
* Add support for RTP stream restart.
When an unhandled packet is encountered, try a restart sequence.
Restart happens when 5 packets with contiguous sequence numbers and same
or increasing time stamps are received. Note that this does not work for
B-frame type of scenarios, but that is true for receive path handling
even before this. As WebRTC does not use B-frames, it is fine. But,
needs to be looked at again if B-frames are necessary.
It is controlled by a config that is disabled by default.
* clean up
* debug log
It is possible that the stream stops just after start and
restarts much later introducing a large gap in sequence number.
That could look like an unhandled case because the wrap back handler
does not have enough packets yet.
Let other checks based on time stamp gap take effect and only if that
also leaves the sequence number unhandled, drop the packet.
* switch participant callbacks to room to listener interface
* mage generate
* clean up
* clear listener
* clean up
* use interface in up data track manager
* tweaks
* Paul feedback - should reduce the diff as this keeps the room handlers as is except making methods for a couple of anonymous handlers
* clean up
When a participant is closing, RTCP readers should be cleaned up from
factory even if the participant is expected to resume. The resumed
participant will be a new participant session and peer connection(s) and
everything will be set up again.
- New bucket API to pass in max packet size and sequence number offset
and seequence number size generic type
- Move OWD estimator to mediatransportutil.
Denys Smirnov