* 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.
* Don't update dependency info if unordered packet received
* Trace all active svc chains for downtrack
* Try to keep lower decode target decodable
* remove comments
* Test case
* clean code
* solve comments
It is possible that publisher paces the media.
So, RTCP sender report from publisher could be ahead of
what is being fowarded by a good amount (have seen up to 2 seconds
ahead). Using the forwarded time stamp for RTCP sender report
in the down stream leads to jumps back and forth in the down track
RTCP sender report.
So, look at the publisher's RTCP sender report to check for it being
ahead and use the publisher rate as a guide.
Actually, was not filtering the not last sender report error before.
Previous PR did that. This PR restores the old no last sender report
filter. Both are filterable errors.
1. Completely removing RTT and jitter from score calculation.
Need to do more work there.
a. Jitter is slow moving (RFC 3550 formula is designed that way).
But, we still get high values at times. Ideally, that should
penalise the score, but due to jitter buffer, effect may not be
too bad.
b. Need to smooth RTT. It is based on receiver report and if one
sample causes a high number, score could be penalised
(this was being used in down track direction only). One option
is to smooth it like the jitter formula above and try using it.
But, for now, disabling that also.
2. When receiving lesser number of packets (for example DTX), reduce the
weight of packet loss with a quadratic relationship to packet loss
ratio. Previously using a square root and it was potentially
weighting it too high. For example, if only 5 packets were received
due to DTX instead of 50, we were still giving 30% weight
(sqrt(0.1)). Now, it gets 1% weight. So, if one of those 5 packets
were lost (20% packet loss ratio), it still does not get much weight
as the number of packets is low.,
3. Slightly slower decrease in score (in EWMA)
4. When using RED, increase packet loss weight thresholds to be able to
take more loss before penalizing score.
Two things
- Somehow the publisher RTCP sender report time stamp goes back some
times. Log it differently. Also, use signed type for logging so
that negative is easy to see.
- On down track, because of silence frame injection on mute, the RTCP
sender report time stamp might be ahead of timestamp we will use
on unmute. If so, ensure that next timestamp is also not before
what was sent in RTCP sender report.
The PID controller seems to be working well. But, it is unclear where
it can be applied as some of the data shows significant jumps
(either caused by BT devices or possibly noise cancellation/cpu
constraint) and although PID controller is slowly pulling things
to expected sample rate, it could be a bit slow.
Unfortunately, cannot munge too much in a middle box.
However leaving the controller in there as it is doing its job
for cases where things slip slowly.
Changing things to log significant jumps (more than 200 ms away
from expected) at Infow level.
Also, recording drift and sample rate in RTP stats proto and string
representation.
With short term measurements, the adjustment itself was causing
some oscillations and drift tend to settle at some small value
and oscillated around it due to push/pull affecting small window
measurement.
* Experimental flag to try time stamp adjustment to control drift.
There is a config to enable this.
Using a PID controller to try and keep the sample rate at expected
value. Need to be seen if this works well. Adjustment are limited
to 25 ms max at a time to ensure there are no large jumps.
And it is applied when doing RTCP sender report which happens
once in 5 seconds currently for both audio and video tracks.
A nice introduction to PID controllers - https://alphaville.github.io/qub/pid-101/#/
Implementation borrowed from - https://github.com/pms67/PID
A few things TODO
1. PID controller tuning is a process. Have picked values from test from
that implementation above. May not be the best. Need to try.
2. Can potentially run this more often. Rather than running it only when
running RTCP sender report (which is once in 5 seconds now), can
potentially run it every second and limit the amount of change to
something like 10 ms max.
* remove unused variable
* debug log a bit more
* Keep track of expected RTP time stamp and control drift.
- Use monotonic clock in RTCP Sender Report and packet times
- Keep the time stamp close to expected time stamp on layer/SSRC
switches
* clean up
* fix test compile
* more test compile failures
* anticipatory clean up
* further clean up
* add received sender report logging
