/**
* \brief start output
* \param audec pointer to audec
* \return 0 on success otherwise negative error code
*
* Call android_start(), then the callback will start being called.
*/
extern "C" int android_start(struct aml_audio_dec* audec)
{
adec_print("android out start");
status_t status;
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;
}
status = track->initCheck();
if (status != NO_ERROR) {
delete track;
out_ops->private_data = NULL;
return -1;
}
track->start();
adec_print("AudioTrack initCheck OK and started.");
return 0;
}
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;
}
int
sa_stream_open(sa_stream_t *s) {
if (s == NULL) {
return SA_ERROR_NO_INIT;
}
if (s->output_unit != NULL) {
return SA_ERROR_INVALID;
}
int32_t chanConfig = s->channels == 1 ?
AudioSystem::CHANNEL_OUT_MONO : AudioSystem::CHANNEL_OUT_STEREO;
int frameCount;
if (AudioTrack::getMinFrameCount(&frameCount, AudioSystem::DEFAULT,
s->rate) != NO_ERROR) {
return SA_ERROR_INVALID;
}
int minsz = frameCount * s->channels * sizeof(int16_t);
s->bufferSize = s->rate * s->channels * sizeof(int16_t);
if (s->bufferSize < minsz) {
s->bufferSize = minsz;
}
AudioTrack *track =
new AudioTrack(AudioSystem::SYSTEM,
s->rate,
AudioSystem::PCM_16_BIT,
chanConfig,
frameCount,
0,
NULL, NULL,
0,
0);
if (track->initCheck() != NO_ERROR) {
delete track;
return SA_ERROR_INVALID;
}
s->output_unit = track;
ALOG("%p - New stream %u %u bsz=%u min=%u", s, s->rate, s->channels,
s->bufferSize, minsz);
return SA_SUCCESS;
}
status_t APlaybackDevice::open()
{
status_t err;
int sampleRateInHz;
int channelConfig;
int audioFormat;
int bufferSizeInBytes;
LOG_FUNC_START
sampleRateInHz = mDevice->Frequency;
channelConfig = aluChannelsFromFormat(mDevice->Format) == 1 ?
AUDIO_CHANNEL_OUT_MONO : AUDIO_CHANNEL_OUT_STEREO;
audioFormat = aluBytesFromFormat(mDevice->Format) == 1 ?
AUDIO_FORMAT_PCM_8_BIT : AUDIO_FORMAT_PCM_16_BIT;
err = AudioTrack::getMinFrameCount(&bufferSizeInBytes, audioFormat, sampleRateInHz);
RETURN_IF(err);
LOGV("rate(%i), channel(%i), format(%i), buffSize(%i), numUpdates(%i)",
sampleRateInHz, channelConfig, audioFormat, bufferSizeInBytes, mDevice->NumUpdates);
err = mAudioTrack.set(AUDIO_STREAM_MUSIC,
sampleRateInHz,
audioFormat,
channelConfig,
bufferSizeInBytes, // frameCount
0, // flags
0, 0); // callback, callback data (user)
RETURN_IF(err);
err = mAudioTrack.initCheck();
RETURN_IF(err);
if(mBuffer) {
delete mBuffer;
}
mBuffer = new AAudioBuffer(bufferSizeInBytes);
LOG_FUNC_END
return NO_ERROR;
}
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;
}
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;
}
