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;
}
