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Media/WebRTC Audio Issues: Difference between revisions
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L16
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(L16) |
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*** Inputs need to either sync to the MSG frequency (system now, to be output clock), or they need to have optional resampling (see above about long-term mismatches) | *** Inputs need to either sync to the MSG frequency (system now, to be output clock), or they need to have optional resampling (see above about long-term mismatches) | ||
** Extra clock domains in MediaStreamGraph<br>MediaStreamGraph currently is clocked on the system clock; ongoing work is moving it to be clocked on the audio output clock. This will reduce total delay in MSG. Note that the output clock frequency may both drift and also suddenly change when the output is re-routed, and the code needs to adapt smoothly to this. | ** Extra clock domains in MediaStreamGraph<br>MediaStreamGraph currently is clocked on the system clock; ongoing work is moving it to be clocked on the audio output clock. This will reduce total delay in MSG. Note that the output clock frequency may both drift and also suddenly change when the output is re-routed, and the code needs to adapt smoothly to this. | ||
** Basic sample rate low<br>Right now, we're clocking everything at 16000Hz; we should be using higher clockrates | ** Basic sample rate low<br>Right now, we're clocking everything at 16000Hz; we should be using higher clockrates. | ||
*** The "L16" pseudo-codec in GIPS only supports 8000/16000/32000 sampling rates (including in 3.30) | |||
* MediaStreamGraph fundamental latency | * MediaStreamGraph fundamental latency | ||
** Because MediaStreamGraph reclocks and plays out from the MSG, it has to keep a minimal buffering level to avoid underflow. This adds 15-30ish ms of input-side latency (and output latency, though output clocking MSG will reduce that). Note that correcting this may require different audio streams for internal versus PeerConnection/"realtime" streams. | ** Because MediaStreamGraph reclocks and plays out from the MSG, it has to keep a minimal buffering level to avoid underflow. This adds 15-30ish ms of input-side latency (and output latency, though output clocking MSG will reduce that). Note that correcting this may require different audio streams for internal versus PeerConnection/"realtime" streams. | ||