android notes, Audio, Alsa, buffer size & latency

在 framework/base/media/libmedia/AudioTrack.cpp (getMinFrameCount) :

    // Ensure that buffer depth covers at least audio hardware latency
    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
    if (minBufCount < 2) minBufCount = 2;

    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
              afFrameCount * minBufCount * sampleRate / afSampleRate;
    return NO_ERROR;

Latency 是問hardware.,,經由OutputDescriptor 傳回 (include/media/AudioSystem.h)

    class OutputDescriptor {
    public:
        OutputDescriptor()
        : samplingRate(0), format(0), channels(0), frameCount(0), latency(0)  {}

        uint32_t samplingRate;
        int32_t format;
        int32_t channels;
        size_t frameCount;
        uint32_t latency;
    };

或是 IAudioFlinger.

這個latency 直，是由 alsa 的 bufferSize 算出來的。
在 hardware/alsa_sound/alsa_default.cpp..

static alsa_handle_t _defaultsOut = {
    module      : 0,
    devices     : AudioSystem::DEVICE_OUT_ALL,
    curDev      : 0,
    curMode     : 0,
    handle      : 0,
    format      : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
    channels    : 2,
    sampleRate  : DEFAULT_SAMPLE_RATE,
    latency     : 40000, // Desired Delay in usec
    bufferSize  : 8192, // Desired Number of samples
    mmap        : 0,
    modPrivate  : 0,
};

這裡的bufferSize, 變更的話, latency 值就會改變。