C++ AudioTrack::setVolume方法代码示例

status_t LibmediaPlayback::Play(int duration_secs) {
  audio_channel_mask_t audio_mask =
      audio_channel_out_mask_from_count(num_channels_);
  const audio_stream_type_t kStreamType = AUDIO_STREAM_MUSIC;
  size_t frame_count = 0;  // Use default value for frame count.
  audio_output_flags_t audio_output_flags = AUDIO_OUTPUT_FLAG_NONE;

  AudioTrack* track = new AudioTrack(
      kStreamType, sample_rate_, audio_format_, audio_mask, frame_count,
      audio_output_flags, LibmediaPlayback::AudioCallback,
      reinterpret_cast<void*>(this));
  status_t status = track->initCheck();
  if (status != OK) {
    LOG(ERROR) << "Audio track initialization failed.";
    return status;
  }

  float volume = 0.9;
  track->setVolume(volume);
  status = track->start();
  if (status != OK) {
    LOG(ERROR) << "Audio track failed to start.";
    return status;
  }

  sleep(duration_secs);
  track->stop();

  if (in_file_)
    sf_close(in_file_);
  else
    sine_data_buffer_->release();

  return status;
}

/**
 * \brief set left/right output volume
 * \param audec pointer to audec
 * \param lvol refer to left volume value
 * \param rvol refer to right volume value
 * \return 0 on success otherwise negative error code
 */
extern "C" int android_set_lrvolume(struct aml_audio_dec* audec, float lvol,float rvol)
{
    adec_print("android set left and right volume separately");

    audio_out_operations_t *out_ops = &audec->aout_ops;
    AudioTrack *track = (AudioTrack *)out_ops->private_data;

    Mutex::Autolock _l(mLock);

    if (!track) {
        adec_print("No track instance!\n");
        return -1;
    }

    track->setVolume(lvol, rvol);

    return 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;
}

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;
}

void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume,
        float rightVolume, int priority, int loop, float rate)
{
    AudioTrack* oldTrack;

    LOGV("play %p: sampleID=%d, channelID=%d, leftVolume=%f, rightVolume=%f, priority=%d, loop=%d, rate=%f",
            this, sample->sampleID(), nextChannelID, leftVolume, rightVolume, priority, loop, rate);

    // if not idle, this voice is being stolen
    if (mState != IDLE) {
        LOGV("channel %d stolen - event queued for channel %d", channelID(), nextChannelID);
        mNextEvent.set(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate);
        stop();
        return;
    }

    // initialize track
    int afFrameCount;
    int afSampleRate;
    int streamType = mSoundPool->streamType();
    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
        afFrameCount = kDefaultFrameCount;
    }
    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
        afSampleRate = kDefaultSampleRate;
    }
    int numChannels = sample->numChannels();
    uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5);
    uint32_t bufferFrames = (afFrameCount * sampleRate) / afSampleRate;
    uint32_t frameCount = 0;

    if (loop) {
        frameCount = sample->size()/numChannels/((sample->format() == AudioSystem::PCM_16_BIT) ? sizeof(int16_t) : sizeof(uint8_t));
    }

#ifndef USE_SHARED_MEM_BUFFER
    // Ensure minimum audio buffer size in case of short looped sample
    if(frameCount < kDefaultBufferCount * bufferFrames) {
        frameCount = kDefaultBufferCount * bufferFrames;
    }
#endif

    AudioTrack* newTrack;
    
    // mToggle toggles each time a track is started on a given channel.
    // The toggle is concatenated with the SoundChannel address and passed to AudioTrack
    // as callback user data. This enables the detection of callbacks received from the old
    // audio track while the new one is being started and avoids processing them with 
    // wrong audio audio buffer size  (mAudioBufferSize)
    unsigned long toggle = mToggle ^ 1;
    void *userData = (void *)((unsigned long)this | toggle);
    
#ifdef USE_SHARED_MEM_BUFFER
    newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
            numChannels, sample->getIMemory(), 0, callback, userData);
#else
    newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
            numChannels, frameCount, 0, callback, userData, bufferFrames);
#endif
    if (newTrack->initCheck() != NO_ERROR) {
        LOGE("Error creating AudioTrack");
        delete newTrack;
        return;
    }
    LOGV("setVolume %p", newTrack);
    newTrack->setVolume(leftVolume, rightVolume);
    newTrack->setLoop(0, frameCount, loop);

    {
        Mutex::Autolock lock(&mLock);
        // From now on, AudioTrack callbacks recevieved with previous toggle value will be ignored.
        mToggle = toggle;
        oldTrack = mAudioTrack;
        mAudioTrack = newTrack;
        mPos = 0;
        mSample = sample;
        mChannelID = nextChannelID;
        mPriority = priority;
        mLoop = loop;
        mLeftVolume = leftVolume;
        mRightVolume = rightVolume;
        mNumChannels = numChannels;
        mRate = rate;
        clearNextEvent();
        mState = PLAYING;
        mAudioTrack->start();
        mAudioBufferSize = newTrack->frameCount()*newTrack->frameSize();
    }

    LOGV("delete oldTrack %p", oldTrack);
    delete oldTrack;
}

bool Song::event(QEvent* _e)
{
	if (_e->type() != QEvent::User)
		return false; //ignore all events except user events, which are events from Python bridge subsystem

	QPybridgeEvent* e = (QPybridgeEvent*) _e;
	switch (e->getType())
	{
		case QPybridgeEvent::SONG_UPDATE:
			this->update(e->getP1());
			break;
		case QPybridgeEvent::SONGLEN_CHANGE:
			this->setLen(e->getP1());
			break;
		case QPybridgeEvent::SONG_POSCHANGE:
			this->setPos(e->getP1(), e->getP2());
			break;
		case QPybridgeEvent::SONG_SETPLAY:
			this->setPlay(true);
			break;
		case QPybridgeEvent::SONG_SETSTOP:
			this->setStop(true);
			break;
		case QPybridgeEvent::SONG_REWIND:
			this->rewindStart();
			break;
		case QPybridgeEvent::SONG_SETMUTE:
		{
			Track* track = this->findTrack(e->getS1());
			if (track == NULL)
				return false;

			bool muted = e->getP1() == 1;
			track->setMute(muted);
			this->update(SC_MUTE | SC_TRACK_MODIFIED);
			break;
		}
		case QPybridgeEvent::SONG_SETCTRL:
		{
			Track* t = this->findTrack(e->getS1());
			if (t == NULL)
				return false;

			if (t->isMidiTrack() == false)
				return false;

			MidiTrack* track = (MidiTrack*) t;
			int chan = track->outChannel();

			int num = e->getP1();
			int val = e->getP2();
			int tick = song->cpos();
			MidiPlayEvent ev(tick, track->outPort(), chan, ME_CONTROLLER, num, val, t);
			audio->msgPlayMidiEvent(&ev);
			song->update(SC_MIDI_CONTROLLER);
			break;
		}
		case QPybridgeEvent::SONG_SETAUDIOVOL:
		{
			Track* t = this->findTrack(e->getS1());
			if (t == NULL)
				return false;

			if (t->type() == Track::DRUM || t->type() == Track::MIDI)
				return false;

			AudioTrack* track = (AudioTrack*) t;
			track->setVolume(e->getD1());
			break;
		}
		case QPybridgeEvent::SONG_IMPORT_PART:
		{
			Track* track = this->findTrack(e->getS1());
			QString filename = e->getS2();
			unsigned int tick = e->getP1();
			if (track == NULL)
				return false;

			oom->importPartToTrack(filename, tick, track);
			break;
		}
		case QPybridgeEvent::SONG_TOGGLE_EFFECT:
		{
			Track* t = this->findTrack(e->getS1());
			if (t == NULL)
				return false;

			if (t->type() != Track::WAVE)
				return false;

			int fxid = e->getP1();


			int onoff = (e->getP2() == 1);

			AudioTrack* track = (AudioTrack*) t;
			Pipeline* pipeline = track->efxPipe();
			const Pipeline* pipeline = track->efxPipe();
			if(pipeline)
			{
//.........这里部分代码省略.........