* Disable audio loss proxying.
Added a config which is off by default.
With audio NACKs, that is the preferred repair mechanism.
With RED, repair is built in via packet redundancy to recover from
isolated losses.
So, proxying is not required. But, leaving it in there with a config
that is disabled by default.
* fix test
* Forward publisher sender report.
Publisher side RTCP sernfer report is rebased to SFU time base
and used to send sender rerport to subscriber.
Will wait to merge till previous versions are out as this will require a
bunch of testing.
* - Add rebased report drift
- update protocol dep
- fix path change check, it has to check against delta of propagation
delay and not propagation delay as the two side clocks could be way
off.
Firefox on Windows 10 seems to be producing simulcast tracks with
duplicate RID. That causes a leak as only one buffer is processed.
Ignore duplicate rid.
NOTE: This is not perfect as the actual layer -> rid is indeterminable
at addition time. It would require looking at packets to determine the
video dimensions and match to rid/layer to figure out which one is
correct and which one is duplicate.
To simplify though, taking the first one and dropping later ones.
This could mean the correct resolution is not streamed, but that should
be okay. The leak is far more destructive.
Was at 20 when LOST was introduced, but was going to 20 even when under
not LOST conditions. When there are packets, want the min to be at 30.
Going down to 20 resulted in reporting LOST quality even when packets
were flowing (although they were experiencing heavy loss and quality
would have been very bad, yet they are not lost).
Also, sample warning about adding packet to bucket even more.
* Add debug to understand VP9 freezes.
Have reports of VP9 freezing in some rooms.
Some data indicates that NACKs are received by SFU, but cannot get RTP
packet when that happens. It is possible that the NACKs are all from
dropped packets. Adding some debug to understand drops/NACKs better.
* enable DD debug
* comment out DD debug
* markers
* add back log about diff length mismatch
* add back key frame mismatch logging
* log skipped drops also
* Log cleanup pass
Demoted a bunch of logs to DEBUG, consolidated logs.
* use context logger and fix context var usage
* moved common error types, fixed tests
* Declare audio inactive if stale.
Stale samples were used to declare audio active.
Maintain last update time and declare inactive if samples are stale.
* correct comment
* spelling
* check level in test
* 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
* 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
* 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
* 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
* 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
* 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.
* 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
* 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
* Decode chains
* clean up
* clean up
* decode targets only on publisher side
* comment out supported codecs
* fix test compile
* fix another test compile
* Adding TODO notes
* chainID -> chainIdx
* do not need to check for switch up point when using chains, as long as chain integrity is good, can switch
* more comments
* address comments
