/**
 * \brief get output latency in ms
 * \param audec pointer to audec
 * \return output latency
 */
extern "C" unsigned long android_latency(struct aml_audio_dec* audec)
{
    unsigned long latency;
    audio_out_operations_t *out_ops = &audec->aout_ops;
    AudioTrack *track = (AudioTrack *)out_ops->private_data;

    if (track) {
        latency = track->latency();
        return latency;
    }

    return 0;
}
示例#2
0
status_t MediaPlayerService::AudioOutput::open(uint32_t sampleRate, int channelCount, int format, int bufferCount)
{
    // Check argument "bufferCount" against the mininum buffer count
    if (bufferCount < mMinBufferCount) {
        LOGD("bufferCount (%d) is too small and increased to %d", bufferCount, mMinBufferCount);
        bufferCount = mMinBufferCount;

    }
    LOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
    if (mTrack) close();
    int afSampleRate;
    int afFrameCount;
    int frameCount;

    if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != NO_ERROR) {
        return NO_INIT;
    }
    if (AudioSystem::getOutputSamplingRate(&afSampleRate, mStreamType) != NO_ERROR) {
        return NO_INIT;
    }

    frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate;
    AudioTrack *t = new AudioTrack(mStreamType, sampleRate, format, channelCount, frameCount);
    if ((t == 0) || (t->initCheck() != NO_ERROR)) {
        LOGE("Unable to create audio track");
        delete t;
        return NO_INIT;
    }

    LOGV("setVolume");
    t->setVolume(mLeftVolume, mRightVolume);
    mMsecsPerFrame = 1.e3 / (float) sampleRate;
    mLatency = t->latency() + kAudioVideoDelayMs;
    mTrack = t;
    return NO_ERROR;
}
示例#3
0
qboolean SNDDMA_Init(void)
{
  if ( ! enableSound() ) {
    return false;
  }

  gDMAByteIndex = 0;

  // Initialize the AudioTrack.

  status_t result = gAudioTrack.set(
    AudioSystem::DEFAULT, // stream type
    SAMPLE_RATE,   // sample rate
    BITS_PER_SAMPLE == 16 ? AudioSystem::PCM_16_BIT : AudioSystem::PCM_8_BIT,      // format (8 or 16)
    (CHANNEL_COUNT > 1) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,       // channel mask
    0,       // default buffer size
    0, // flags
    AndroidQuakeSoundCallback, // callback
    0,  // user
    0); // default notification size

  LOGI("AudioTrack status  = %d (%s)\n", result, result == NO_ERROR ? "success" : "error");

  if ( result == NO_ERROR ) {
    LOGI("AudioTrack latency = %u ms\n", gAudioTrack.latency());
    LOGI("AudioTrack format = %u bits\n", gAudioTrack.format() == AudioSystem::PCM_16_BIT ? 16 : 8);
    LOGI("AudioTrack sample rate = %u Hz\n", gAudioTrack.getSampleRate());
    LOGI("AudioTrack frame count = %d\n", int(gAudioTrack.frameCount()));
    LOGI("AudioTrack channel count = %d\n", gAudioTrack.channelCount());

    // Initialize Quake's idea of a DMA buffer.

    shm = &sn;
    memset((void*)&sn, 0, sizeof(sn));

    shm->splitbuffer = false;	// Not used.
    shm->samplebits = gAudioTrack.format() == AudioSystem::PCM_16_BIT ? 16 : 8;
    shm->speed = gAudioTrack.getSampleRate();
    shm->channels = gAudioTrack.channelCount();
    shm->samples = TOTAL_BUFFER_SIZE / BYTES_PER_SAMPLE;
    shm->samplepos = 0; // Not used.
    shm->buffer = (unsigned char*) Hunk_AllocName(TOTAL_BUFFER_SIZE, (char*) "shmbuf");
    shm->submission_chunk = 1; // Not used.

    shm->soundalive = true;

    if ( (shm->samples & 0x1ff) != 0 ) {
      LOGE("SNDDDMA_Init: samples must be power of two.");
      return false;
    }

    if ( shm->buffer == 0 ) {
      LOGE("SNDDDMA_Init: Could not allocate sound buffer.");
      return false;
    }

    gAudioTrack.setVolume(1.0f, 1.0f);
    gAudioTrack.start();
  }

  return result == NO_ERROR;
}