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We would like to add two new audio receiver network metrics called relativePacketArrivalDelay and totalPacketsReceived.
relativePacketArrivalDelay measures the difference between the expected arriv…
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Hi,
I'm getting a bug where when 2 people speak at the same time, the packets become glitchy and the audio is not understable when both are talking. If they aren't speaking at the same time, audio …
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It is desirable in some situations to have access to audio data without it being decoded first. This is not possible with the current exposed API. This is incredibly useful for storing audio in memory…
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### Command
ffmpeg
### Description
A complete, cross-platform solution to record, convert and stream audio and video.
### Homepage
https://ffmpeg.org/
### Documentation
https://ff…
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So I've observed a significant latency when using one ffmpeg process to feed both video (from v4l2) and audio (from ALSA) to the included nodejs relay. With latest master. So I tried switching to havi…
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```
What steps will reproduce the problem?
1.From chromium or chrome or firefox, do a sip register to Asterisk(13.3), then
make a call to any number
2.In asterisk dialPlan, place a playback(), you wi…
-
```
What steps will reproduce the problem?
1.From chromium or chrome or firefox, do a sip register to Asterisk(13.3), then
make a call to any number
2.In asterisk dialPlan, place a playback(), you wi…
-
```
What steps will reproduce the problem?
1.From chromium or chrome or firefox, do a sip register to Asterisk(13.3), then
make a call to any number
2.In asterisk dialPlan, place a playback(), you wi…
-
```
What steps will reproduce the problem?
1.From chromium or chrome or firefox, do a sip register to Asterisk(13.3), then
make a call to any number
2.In asterisk dialPlan, place a playback(), you wi…
-
```
What steps will reproduce the problem?
1.From chromium or chrome or firefox, do a sip register to Asterisk(13.3), then
make a call to any number
2.In asterisk dialPlan, place a playback(), you wi…