Merge "aaudio: use minimum of one burst in buffer" into rvc-dev

aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {

    int32_t adjustedFrames = requestedFrames;

    const int32_t maximumSize = getBufferCapacity() - mFramesPerBurst;

    // The buffer size can be set to zero.

    // This means that the callback may be called when the internal buffer becomes empty.

    // This will be fine on some devices in ideal circumstances and will result in the

    // lowest possible latency.

    // If there are glitches then they should be detected as XRuns and the size can be increased.

    static const int32_t minimumSize = 0;

    // Minimum size should be a multiple number of bursts.

    const int32_t minimumSize = 1 * mFramesPerBurst;

    // Clip to minimum size so that rounding up will work better.

    adjustedFrames = std::max(minimumSize, adjustedFrames);

        // Round to the next highest burst size.

        int32_t numBursts = (adjustedFrames + mFramesPerBurst - 1) / mFramesPerBurst;

        adjustedFrames = numBursts * mFramesPerBurst;

        // Clip just in case maximumSize is not a multiple of mFramesPerBurst.

        adjustedFrames = std::min(maximumSize, adjustedFrames);

    }

    // Clip against the actual size from the endpoint.

    int32_t actualFrames = 0;

    mAudioEndpoint.setBufferSizeInFrames(maximumSize, &actualFrames);

    // actualFrames should be <= maximumSize

    // actualFrames should be <= actual maximum size of endpoint

    adjustedFrames = std::min(actualFrames, adjustedFrames);

    mBufferSizeInFrames = adjustedFrames;