ALSA detect (and create) buffer overrunsHow to stream semi-live audio over internetNode.js Live Streaming: Avoid bufferingAudio mixing with alsa's dmix plugin in c++Method for audio playback with known output latency on WindowsReal time audio streaming to mobile device - dynamic playback speedAlsa Buffer overrun while playback of audioALSA buffer xrun induced by low quality source in ffmpeg capturePortAudio unreliable: Expression '…' failedWhen do I know to cause a rebuffer on streaming audio when playing over ALSA?Strange ALSA monitor device behaviour
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ALSA detect (and create) buffer overruns
How to stream semi-live audio over internetNode.js Live Streaming: Avoid bufferingAudio mixing with alsa's dmix plugin in c++Method for audio playback with known output latency on WindowsReal time audio streaming to mobile device - dynamic playback speedAlsa Buffer overrun while playback of audioALSA buffer xrun induced by low quality source in ffmpeg capturePortAudio unreliable: Expression '…' failedWhen do I know to cause a rebuffer on streaming audio when playing over ALSA?Strange ALSA monitor device behaviour
I am writing a real time voice streaming application that, for reasons too depressing to explain here, uses ALSA. I have a problem that one sides' voice keeps getting delayed and delayed (i.e. - as the streaming progresses, the voice is issued with higher and higher latency).
I suspect that a clock drift between the two sides causes side A to generate more samples per second than side B plays, causing extra samples to be produced each time.
I have not been able to locate how to ask ALSA how many samples are currently queued, nor limit the buffer size so that ALSA reject my new samples instead of increasing its buffer to play them.
Is what I'm trying to do possible?
audio-streaming alsa
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
add a comment |
I am writing a real time voice streaming application that, for reasons too depressing to explain here, uses ALSA. I have a problem that one sides' voice keeps getting delayed and delayed (i.e. - as the streaming progresses, the voice is issued with higher and higher latency).
I suspect that a clock drift between the two sides causes side A to generate more samples per second than side B plays, causing extra samples to be produced each time.
I have not been able to locate how to ask ALSA how many samples are currently queued, nor limit the buffer size so that ALSA reject my new samples instead of increasing its buffer to play them.
Is what I'm trying to do possible?
audio-streaming alsa
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
ALSA's buffer size is fixed. You have to measure the relative speeds, and resample accordingly.
– CL.
Mar 7 at 7:02
add a comment |
I am writing a real time voice streaming application that, for reasons too depressing to explain here, uses ALSA. I have a problem that one sides' voice keeps getting delayed and delayed (i.e. - as the streaming progresses, the voice is issued with higher and higher latency).
I suspect that a clock drift between the two sides causes side A to generate more samples per second than side B plays, causing extra samples to be produced each time.
I have not been able to locate how to ask ALSA how many samples are currently queued, nor limit the buffer size so that ALSA reject my new samples instead of increasing its buffer to play them.
Is what I'm trying to do possible?
audio-streaming alsa
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
I am writing a real time voice streaming application that, for reasons too depressing to explain here, uses ALSA. I have a problem that one sides' voice keeps getting delayed and delayed (i.e. - as the streaming progresses, the voice is issued with higher and higher latency).
I suspect that a clock drift between the two sides causes side A to generate more samples per second than side B plays, causing extra samples to be produced each time.
I have not been able to locate how to ask ALSA how many samples are currently queued, nor limit the buffer size so that ALSA reject my new samples instead of increasing its buffer to play them.
Is what I'm trying to do possible?
audio-streaming alsa
audio-streaming alsa
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
asked Mar 7 at 6:26
Shachar ShemeshShachar Shemesh
5,89631742
5,89631742
ALSA's buffer size is fixed. You have to measure the relative speeds, and resample accordingly.
– CL.
Mar 7 at 7:02
add a comment |
ALSA's buffer size is fixed. You have to measure the relative speeds, and resample accordingly.
– CL.
Mar 7 at 7:02
ALSA's buffer size is fixed. You have to measure the relative speeds, and resample accordingly.
– CL.
Mar 7 at 7:02
ALSA's buffer size is fixed. You have to measure the relative speeds, and resample accordingly.
– CL.
Mar 7 at 7:02
add a comment |
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ALSA's buffer size is fixed. You have to measure the relative speeds, and resample accordingly.
– CL.
Mar 7 at 7:02