* Buffer size config for video and audio.
There was only one buffer size in config.
In upstream, config value was used for video.
Audio used a hard coded value of 200 packets.
But, in the down stream sequencer, the config value was used for both
video and audio. So, if video was set up for high bit rate (deep
buffers), audio sequencer ended up using a lot of memory too in
sequencer.
Split config to be able to control that and also not hard code audio.
Another optimisation here would be to not instantiate sequencer unkess
NACK is negotiated.
* deprecate packet_buffer_size
* Use marshal + unmarshal to ensure unmarshallable fields are not copied.
Need to ensure that config structs/fields are marshallable.
There was a use of a = b copy of struct and some of the embeded structs
had locks and copying was not good.
* update protocol
* Update deps
* Add option to issue full reconnect on data channel error.
There are situations where send data packet fails because of "stream
closed". It is unclear when that happens. Seems to be after an
ICERestart after ICE failed and connection type switching to TURN
from ICE.
Once the failure happens, it is not recoverable. Potentially, it is
recoverable, but unclear where the problem lies. Attempts to reproduce
looking at the pattern of failures has been unsuccesful.
In the mean time, adding an option to issue full reconnect
when send data packet fails.
* typo
* 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
* Make congestion controller probe config
* Wait for enough estimate samples
* fixes
* format
* limit number of times a packet is ACKed
* ramp up probe duration
* go format
* correct comment
* restore default
* add float64 type to generated CLI
* Make signal close async.
Left notes about async close in code.
Also reducing retry config timeout
- Timeout to 7.5 seconds (making it 1/4th of current config)
- max retry to 4 seconds
- so, it can do 4 tries now in 7.5 seconds (with retries ending at 0.5
seconds, 1.5 seconds, 3.5 seconds, 7.5 seconds). The change of max to
4 seconds is not really needed, but it lined up with 7.5. So, made the
change.
* update comments a bit
* 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
