* 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
* Update go deps
Generated by renovateBot
* use generics with Deque
---------
Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>
Co-authored-by: David Zhao <dz@livekit.io>
* Use local time base for NTP in RTCP Sender Report for downtracks.
More details in comments in code.
* Remove debug
* RTCPSenderReportInfo -> RTCPSenderReportDataExt
* Get rid of sender report data pointer checks
* WIP commit
* comment
* clean up
* remove unused stuff
* cleaner comment
* remove unused stuff
* remove unused stuff
* more comments
* TrackSender method to handle RTCP sender report data
* fix test
* push rtcp sender report data to down tracks
* Need payload type for codec id mapping in relay protocol
* rename variable a bit
* Split stream tracker impl from base
* slight re-arrangement of code
* fps based stream tracker
* MinFPS config
* switch back to packet based tracker
* use video config by default to handle sources without type
* Prevent rtx buffer and forwarding path colliding
Received packets are put into RTX buffer which is
a circular buffer and the packet (sequence number) is
queued for forwarding. If the RTX buffer fills up
and cycles before forwarding happens, forwarding
would pick the wrong packet (as it is holding a
reference to a byte slice in the RTX buffer) to forward.
Prevent it by moving reading from RTX buffer just
before forwarding. Adds an extra copy from RTX buffer
-> temp buffer for forwarding, but ensures that forwarding
buffer is not used by another go routine.
* Revert some changes from previous commit
Details:
- Do all forward processing as before.
- One difference is not load raw packet into ExtPacket.
- Load raw packet into provided buffer when module that reads
using ReadExtended calls that function. If the packet is
not there in the retransmission buffer, that packet will be
dropped. This is the case we are trying to fix, i. e. the RTX
buffer has cycled before ReadExtended could pull the packet.
This makes a copy into the provided buffer so that the data
does not change underneath.
* Remove debug comment
* Oops missed a function call
