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philburk avatar philburk commented on May 16, 2024

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.

philburk avatar philburk commented on May 16, 2024

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.

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