MOQT Test Ultra low latency with Webcodecs: PLAYER
MOQT Version
:
-
,
MediaPackager Version
:
-
(Encoder audio sampling frequency should be the same than audioContext (player) sampling frequency, this is almost guaranteed if you use same browser (computer) for encode and playback. The fix is simple but not done yet :-))
Data needed
WT server:
Namespace:
Track name (audio,video will be added):
Old Track name:
Full track names (based on namespace and track name):
AuthInfo (must match with publisher):
Min audio player buffer (ms):
(it waits until audio buffers this amount to start playback)
Max audio player buffer (ms):
(this + jitter is the max latency allowed)
Audio jitter buffer buffer for this player (ms):
Update
Video jitter buffer buffer for this player (ms):
Update
Start
Stop
Latency
(only valid if encoder and player clocks are synchronized, or they are the same machine)
Audio latency capture to renderer - approx (ms):
Video latency capture to renderer - approx (ms):
Video latency via overlay - exact (ms):
Receiver demuxer
Current received audio TS(ms):
Current received video TS(ms):
V-A diff(ms):
First audio TS(ms):
First video TS(ms):
V-A start diff(ms):
Receiver dejitter
Audio
Buffer current max size:
Buffer size:
Gaps detected:
Video
Buffer current max size:
Buffer size:
Gaps detected:
Decoders
Audio
Current frame TS compensated (ms):
Buffer size:
Timestamp compensation(ms):
(The Audio decoder does NOT track timestamps (bummer), it just uses the 1st one sent and at every decoded audio sample adds 1/fs (so sample time), that means if we drop and audio packet those timestamps will be collapsed creating A/V out of sync. We compensate those lost packets with this)
Video
Current frame TS(ms):
Buffer size:
V-A diff(ms):
Renderers
Audio
Current frame TS(ms):
Buffer size:
Total silence inserted (ms):
Video
Current frame TS(ms):
Buffer size:
Not printed frames:
V-A diff(ms):
Dropped data (frames / chunks):