* Throttle packet errors/warns a bit.
In very bad network conditions, it is possible that packets
arrive out-of-order, have choppy behaviour.
Use some counters and temper logs.
* slight change in comment
Streaming could start after 16-bits has rolled over. So, have to add
that base back to what is received in receiver report.
Otherwise, it looks like there are not packets received in window
leading to poor quality.
* Split RTPStats into receiver and sender.
For receiver, short types are input and need to calculate extended type.
For sender (subscriber), it can operate only in extended type.
This makes the subscriber side a little simpler and should make it more
efficient as it can do simple comparisons in extended type space.
There was also an issue with subscriber using shorter type and
calculating extended type. When subscriber starts after the publisher
has already rolled over in sequence number OR timestamp, when
subsequent publisher side sender reports are used to adjust subscriber
time stamps, they were out of whack. Using extended type on subscriber
does not face that.
* fix test
* extended types from sequencer
* log
* Fix time stamp adjustment when starting with dmummy packets.
- Populated extended values in ExtPacket on dummy packet.
- Have to pass reference time stamp offset to first packet time
adjustment.
* display participant version info
* Sequencer small optimisations
1. Use range map to exclude padding only packets. Should take lesser
space as we are not using slice to hold pointer to actual data.
2. Avoid `time.Now()` when adding each packet. Just use the arrival time
as it should be close enough. `time.Now()` was showing up in
profile.
* remove debug
* correct comment
Profiling showed updating jitter going through the snapshot maps.
With the reduction of one, there should only be one snapshot
and hopefully that should gain some cycles back.
* Do NACKs and reports always.
With padding packet drops, it is possible that a lot of packets
go by without RTCP RR.
Do NACKs and RTCP RR always.
* remove local variable
Seeing the case of a stream starting with padding packets
on migration. As publisher in that case is always sending packets.
it is possible that the new node gets padding packets at the start.
Processing them in buffer leads to trying to drop that padding packet
and adding an exclusion range. That fails because the extended
sequence number is not available for unprocessed packets.
It is okay to drop them as they will be dropped anyway.
But, they are useful for bandwidth estimation. So, headers are processed
even if the packet is RTPStats unprocessed.
* Handle duplicate padding packet in the up stream.
The following sequence would have produce incorrect results
- Sequence number 39 - regular packet - offset = 0
- Sequence number 40 - padding only - drop - offset = 1
- Sequence number 40 - padding only duplicate - was not dropped (this is
the bug) - apply offet - sequence number becomes 39 and clashes with
previous packet
- Sequence number 41 - regular packet - apply offset - goes through as 40.
- Sequence number 40 again - does not get dropped - will pass through as 39.
* fix duplicate dropping
* fix tests
* accept repeat last value as padding injection could cause that
* use exclusion ranges
* more UT and more specific errors
* Remove sender report warp logs.
They are not useful.
Also replacing drift report with proper protocol and reporting
both packet ad report drift.
Need to dig more into out-of-order sender report sending.
That requires some digging and understanding.
* record time of anachronous report
* more logging around out-of-order repair
* log time of out-of-order received sender report
* Update deps and place holder StartParticipantEgress
* Drop padding only packet on publisher side.
* add UT
* update deps
* remove debug
* add fast path short cut
* correct comment
* fix test
* fix for Linux
* Use VP9 Key frame detection from Galene.
With ffmpeg generated VP9 file with single layer
and publishing via Go SDK, the key picture determination
outlined at https://datatracker.ietf.org/doc/html/draft-ietf-payload-vp9-16#page-13
under the F bit explanation does not work. It declares kay frame for
pretty much all frames. Unclear if ffmpeg generated bitstream has issues
or if that procedure in the above document does not work for single
layer case.
Using the bit stream explained here
https://storage.googleapis.com/downloads.webmproject.org/docs/vp9/vp9-bitstream-specification-v0.6-20160331-draft.pdf
(pages 28, 62, 63) implemented in Galene.
That is more expensive as it has to parse more, but works in all cases.
* Add AV1-TODo
* add some TODOs
* Adjust TS and cycles when adjusting start.
Chasing some AddPacket errors across relay.
Noticed that in one case the start/end sequence was flipped.
There is a known issue of it happening with resync.
Unclear if this instance was due to resync or not.
The start was close to the edge (64513). So, thought maybe
adjust at start and noticed that it needs to maybe increase
cycle count if start is wrapping back. In this case, the
start is 1000 before wrap point. So, may not be a wrap back
issue, but addressing what I found anyway.
* fix test
Seeing a good chunk of logs using default offset.
And it is concentrated heavily on few tracks.
Logging more to understand this better before
potentially demoting this log.
* Use 32-bit ts for first packet adjustment.
Otherwise, a new subscriber on a long running sees a huge difference
if the publisher side has rolled over.
As this happens only in the first two minutes of a track's lifecycle
it is fine to not consider rollover.
* log RTP in anachronous report
* Integrate logger components
Dividing into the following components
* pub - publisher
* pub.sfu
* sub - subscriber
* transport
* transport.pion
* transport.cc
* api
* webhook
* update go modules
* Add control of playout delay
Add config to enable playout delay. The delay will be limited by
[min,max] in the config option and calculated by upstream & downstream
RTT.
* check protocol version to enable playout delay
* Move config to room, limit playout-delay update interval, solve comments
* Remove adaptive playout-delay
* Remove unused config
* Add debug log for RTCP sender report.
Temporary to collect more data. Hitting scenarios under congestion
where the sender report gets off sync. Need some data to pore through
and understand and implement changes.
* Debugw
* WIP commit
* WIP commit
* WIP commit
* Some clean up
- Removed a chatty debug log
- some spelling, punctuation correction in comments
- missed an `Abs` in check, add it.
