* sfu: parse VP8 and VP9 payload descriptors without an allocation
Replace ExtPacket.Payload with typed VP8 and VP9 pointer fields that point
at backing structs inside the ExtPacket, so parsing does not allocate and a
wrong codec type is a compile error.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* sfu: copy the VP8 descriptor in the test packet helper
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
* sfu/buffer: stop allocating per ingested packet
Buffer.Write allocated six objects for every RTP packet. They were
the parsed rtp.Packet, a copy of it in patchExtPacket, and the flow
state, whose address two rare log calls took. The rest were an
AbsCaptureTime, present on about one packet per second, and the
extension slice that pion grows with append.
Unmarshal into a pooled scratch packet. Give ExtPacket its own backing
rtp.Packet and an [8]rtp.Extension array, and copy the header into
them. Patch the payload in place. If the abs-capture-time extension is
present, parse it. Log a copy of the flow state.
Per packet, Write plus ReadExtended. Opus: 6 allocs, 448 B -> 0.
VP8: 6 allocs, 416 B -> 1 alloc, 32 B. Opus 240 -> 142 ns, VP8 269 -> 204 ns.
Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>
* sfu/buffer: point extension payloads at the bucket copy on read
Queued ExtPackets kept extension payload slices into the write input,
which the transport reuses for the next packet. ReadExtended now parses
the header again from the bucket copy, so extensions and CSRCs point
into RawPacket like the payload does.
The test writes every packet through one input buffer and checks the
audio level of each packet read back.
Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>
* sfu/buffer: test that a dropped padding packet keeps the shifted sequence number on read
The bucket stores the shifted number, so re-parsing the header on read
returns it. This test guards that.
Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>
---------
Co-authored-by: Claude Fable 5.1 <noreply@anthropic.com>
* Fix skipped packets accounting.
No need to copy unskipped packet RTP header to skipped packet.
That was causing padding bytes to be counted.
Also use Header.PaddingSize as base PaddingSize is deprecated.
* PaddingSize in header in utils
Not a major issue, but just avoiding duplicate creation of NACK module.
RTCP feedback of `nack` and `nack pli` end up getting treated as `nack`
and was double creating.
* Add debug for receiver restart.
Have a suspicion that something is deadlocking between restart receiver
and buffer bind during replay. Adding debug to get a better picture of
state of receiver restart.
* consistent logging
* Set up audio config in audio level module when config is updated.
It is possible to get audio config after bind (bind is where the audio
level module is created) for remote tracks. So, split out setting audio
level config in audio level module and invoke it when config is updated.
* coderabbit review
* prevent divide-by-0
* not active before config
* Reducing some info level logs.
Also, relaxing the check for runaway RTCP receiver report to allow for
rollover to catch up if it is not too far away.
* set logger
With audio simulcast codecs, it is possible that the clock rate of the
primary codec is different from the secondary codec. If a subscriber
binds to the secondary codec, the clock rate should be set correctly. Do
it at bind time.
* 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
* 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
David Chen