* Seed snapshots
- For one cycle after seeding, delta snap shot can get a huge gap
because of snapshot iitializing from start if not present. Not
a huge deal sa it should not affect functionality, but saving/restoring
(at least with down track) snap shot is a big deal. So just do it.
- Have been seeing a bunch of cases of delta stats getting a lot of
packets due to out-of-order (what seems like) receiver report. So,
save the receiver report and log it when out-of-order is detected
to understand if they are closely spaced or something else could be
happening.
* Remove comment that does not apply anymore
* log current time and RR
* Notify max layer taking into account overshoot.
An attempt to handle case of FF stopping layer 0, but not layer 1.
When max subscribed layer is layer 0, server allows overshoot to layer 1
to ensure continued streaming when the channel is not congested.
But, dynacast could have reported maximum subscribed layer as layer 0.
This is a very simple attempt to address that by taking overshoot
into account. Needs testing if this works well or not.
NOTE: When subsriber/down track is unmuted, it will report
the max subscribed layer as the max required layer. In those cases,
if the client does not start layer 0, there will still be an issue.
IOW, server is not keeping track of client behaviour that the client
has stopped layer 0 and publishing only layer 1. Server is just
accommodating overshoot with this change.
* Fix a couple of tests and change reflect.DeepEqual -> require.Equal as
much as possible
- ticker.Stop always
- clean up timer func (if they are added) on participant close
- sequencer test enhancement to add a real packet after a pdding packet
Have been seeing a few instances of "too many packets expected in delta"
when trying to generate RTCP SR on down track. Actual sequence numbers
indicate that start is after the end.
As down track RTPStats are driven by receiver report, wondering if we
are getting RTCP_RR out-of-order somehow causing this to happen.
Cannot find any other reason for this.
So, accepting RTCP_RR based update only if the sequence number is higher
than existing and also logging a warning with sequence numbers if they
look out-of-order.
Padding packets do not need the full structure. They just
need a placeholder in the sequencer array. So, use pointers
(with padding slots filled by nil) to save some memory.
Also, don't need padding for audio (yet). As padding packets
are used only for probing and we do not probe using audio tracks (yet).
* Cache RTPStats and seed on re-use
When a cached down track is re-used, RTPStats was not cached.
This caused sender reports getting out-of-sync with the remote side.
Cache RTPStats and seed it on re-use.
* staticcheck
* WIP commit
* WIP commit
* fix copy pasta
* setting PC with previous answer has to happen synchronously
* static check
* WIP commit
* WIP commit
* fixing transport tests
* fix tests and clean up
* minor renaming
* FIx test race
* log event when channel is full
* Sample RTP write error logs.
When a client goes away without notice, the media connection
will be severed. Writes will fail till connection is closed.
Avoid log spam in those scenarios. Just log one in 100 failures.
* Change reminder so that first failures gets logged
* log failure count
* Promoting a few logs to Info
Also, adding a couple of more info logs which I will remove later
after some debugging.
* mime type
* Protect pause/max layer
* notify even if not bound
- Do not update jitter on padding only packet.
Padding only packet may not have proper timestamp.
If it does, it probably has the time stamp of the
last packet with payload. That will also affect
jitter calculation, i. e. wall clock time is moving,
but RTP time is the same.
- Do not send `onMaxLayer` changed on bind.
It was probably racing with update when max layer
is updated when adaptive stream is off. There is
no need to send that update as the default would
be OFF. It will be enabled when adaptive stream
subscription turns it on or when max layer is
set when down track bind happens and adaptive stream
is off.
* WIP commit
* Connection quality changes
- Fix Firefox showing poor quality
o The issue was that we were using max available layer and
calculating quality. The rationale being that even if
server sends dynacast messages, client may not implement
dynacast and still stream all layers. But, with Firefox
(maybe a Firefox bug), it sends some small amount of
data on layer 2 even when that layer is disabled.
Guessing it is probing (or actually we might be using
some small value for high layers as Firefox cannot turn off
layers). That higher layer gets used in quality calculation.
As the bit rate on that layer is extremely low, it yields low
score.
Fixed by considering the max expected layer. That is of most
interest. Yes, clients may ignore dynacast and stream all layers,
but, max expected is the one of interest. So, look for
quality in the max expected layer and not max available layer.
- Lots of clean up around connection quality stuff
o Use a dynamic scaling thing to ensure that we do not get bitten
by absolute values. Calculate best possible scenario score and
map that to maximum MOS score. This will ensure that different
codecs, different settings do not mess up the scoring. For example,
a client might use 1 Mbps for 720p, but a different client could
use 2 Mbps for 720p. As an SFU/infrastructure middlebox, we do
not have control over quality at those rates. We can only ensure
that streaming happens smoothly at those rates. So, in that
example, for client 1, 1 Mbps will map to MOS 5.0 and for client 2,
2 Mbps will map to MOS 5.0. Any impairments after that will
reflect in the score.
o Penalise for missing target layer by one level for one layer missed.
o Move tests to connection quality directory. The participant test
was not super useful.
* Add missed file
* Remove debug code
* use more constants and initialise normalisation factor
* rtcscore pointer
* Use media payload size in scoring.
Subtract out header bytes when calculating score.
This does not seem to affect the score (under perfect conditions),
but, using header bytes will inflate the bit rate and
will affect scoring.
* Add header bytes to ToProto
* protocol pointer
* fix test
* WIP commit
* Refactor media loss proxy
* Use DynacastQuality and MediaLossProxy from MediaTrack
* fix test
* Remove unused param
* Remove unused interfaces
* Move interface methods to local
* Split out DynacastManager
* have to add codec to dynacast manager
* RUnlock
* fix restart
* Adding API to force quality and also maintain closed state
* Address PR comments
