Comments (2)
The latency varies depending on how many frames are buffered between the app and the DAC. If the app writes N frames to AAudio then the latency will jump up immediately by N frames. The latency will then go down smoothly by sampleRate frames per second until something underflows or the app writes more data.
If the app runs with a nearly full buffer then the latency will be higher than if it runs with a nearly empty buffer. We can calculate a maximum latency that assumes full buffers all the way down. But the instantaneous latency will generally be lower.
If we add a function to Oboe then a good name might be calculateInstantaneousLatency() or calculateCurrentLatency().
from oboe.
Here is code to measure latency using timestamps.
{
int64_t presentationFrame;
int64_t presentationTime;
result = AAudioStream_getTimestamp(aaudioStream,
CLOCK_MONOTONIC,
&presentationFrame,
&presentationTime
);
if (result == AAUDIO_OK) {
int64_t elapsedNanos = getNanoseconds() - presentationTime;
int64_t elapsedFrames = actualSampleRate * elapsedNanos / NANOS_PER_SECOND;
int64_t currentFrame = presentationFrame + elapsedFrames;
int64_t framesWritten = AAudioStream_getFramesWritten(aaudioStream);
int64_t estimatedLatencyFrames = framesWritten - currentFrame;
int64_t estimatedLatencyMillis = estimatedLatencyFrames * 1000 / actualSampleRate;
printf("estimatedLatencyMillis %d\n", (int)estimatedLatencyMillis);
}
}
from oboe.
Related Issues (20)
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from oboe.