/**
* \brief mute output
* \param audec pointer to audec
* \param en 1 = mute, 0 = unmute
* \return 0 on success otherwise negative error code
*/
extern "C" int android_mute(struct aml_audio_dec* audec, adec_bool_t en)
{
adec_print("android out mute");
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->mute(en);
return 0;
}
/**
* \brief set output volume
* \param audec pointer to audec
* \param vol volume value
* \return 0 on success otherwise negative error code
*/
extern "C" int android_set_volume(struct aml_audio_dec* audec, float vol)
{
adec_print("android set volume");
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(vol, vol);
return 0;
}
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;
}
PyObject* getAudioTrackVolume(PyObject*, PyObject* args)
{
const char* trackname;
if (!PyArg_ParseTuple(args, "s", &trackname))
{
return NULL;
}
Track* t = song->findTrack(QString(trackname));
if (t == NULL)
return NULL;
if (t->type() == Track::DRUM || t->type() == Track::MIDI)
return NULL;
AudioTrack* track = (AudioTrack*) t;
return Py_BuildValue("d", track->volume());
}
int APlaybackDevice::handlePlayback()
{
int bufferSizeInSamples;
bufferSizeInSamples = mBuffer->size() / aluFrameSizeFromFormat(mDevice->Format);
mAudioTrack.start();
while(mPlaybackEnabled) {
aluMixData(mDevice, mBuffer->data(), bufferSizeInSamples);
if(!write(mBuffer)) {
LOGE("Can't write audio buffer into audio track");
mPlaybackEnabled = false;
}
}
mAudioTrack.stop();
mAudioTrack.flush();
return 0;
}
void AudioTrackList::append(PassRefPtr<AudioTrack> prpTrack)
{
RefPtr<AudioTrack> track = prpTrack;
// Insert tracks in the media file order.
size_t index = track->inbandTrackIndex();
size_t insertionIndex;
for (insertionIndex = 0; insertionIndex < m_inbandTracks.size(); ++insertionIndex) {
AudioTrack* otherTrack = static_cast<AudioTrack*>(m_inbandTracks[insertionIndex].get());
if (otherTrack->inbandTrackIndex() > index)
break;
}
m_inbandTracks.insert(insertionIndex, track);
ASSERT(!track->mediaElement() || track->mediaElement() == mediaElement());
track->setMediaElement(mediaElement());
scheduleAddTrackEvent(track.release());
}
/**
* \brief stop output
* \param audec pointer to audec
* \return 0 on success otherwise negative error code
*/
extern "C" int android_stop(struct aml_audio_dec* audec)
{
adec_print("android out stop");
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->stop();
/* release AudioTrack */
delete track;
out_ops->private_data = NULL;
return 0;
}
bool APlaybackDevice::write(AAudioBuffer* buffer)
{
ssize_t length, size;
length = 0;
while(length < buffer->size()) {
size = mAudioTrack.write(buffer->data() + length, buffer->size() - length);
if(size < 0) {
return false;
}
length += size;
}
return true;
}
void AudioPortConfig::routingChanged()
{
//---------------------------------------------------
// populate lists
//---------------------------------------------------
routeList->clear();
newSrcList->clear();
newDstList->clear();
tracksList->clear();
btnConnectOut->setEnabled(false);
connectButton->setEnabled(false);
removeButton->setEnabled(false);
TrackList* tl = song->tracks();
for (ciTrack i = tl->begin(); i != tl->end(); ++i)
{
if ((*i)->isMidiTrack())
continue;
AudioTrack* track = (AudioTrack*) (*i);
if (track->type() == Track::WAVE_OUTPUT_HELPER || track->type() == Track::WAVE_INPUT_HELPER)
{
for (int channel = 0; channel < track->channels(); ++channel)
{
Route r(track, channel);
tracksList->addItem(r.name());
}
}
else
tracksList->addItem(Route(track, -1).name());
}
if(selectedIndex < tracksList->count())
tracksList->setCurrentRow(selectedIndex, QItemSelectionModel::ClearAndSelect);
//if(_selected)
// setSelected(_selected->name());
}
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 MediaSource::removeSourceBuffer(SourceBuffer& buffer, ExceptionCode& ec)
{
LOG(MediaSource, "MediaSource::removeSourceBuffer() %p", this);
Ref<SourceBuffer> protect(buffer);
// 2. If sourceBuffer specifies an object that is not in sourceBuffers then
// throw a NOT_FOUND_ERR exception and abort these steps.
if (!m_sourceBuffers->length() || !m_sourceBuffers->contains(buffer)) {
ec = NOT_FOUND_ERR;
return;
}
// 3. If the sourceBuffer.updating attribute equals true, then run the following steps: ...
buffer.abortIfUpdating();
// 4. Let SourceBuffer audioTracks list equal the AudioTrackList object returned by sourceBuffer.audioTracks.
RefPtr<AudioTrackList> audioTracks = buffer.audioTracks();
// 5. If the SourceBuffer audioTracks list is not empty, then run the following steps:
if (audioTracks->length()) {
// 5.1 Let HTMLMediaElement audioTracks list equal the AudioTrackList object returned by the audioTracks
// attribute on the HTMLMediaElement.
// 5.2 Let the removed enabled audio track flag equal false.
bool removedEnabledAudioTrack = false;
// 5.3 For each AudioTrack object in the SourceBuffer audioTracks list, run the following steps:
while (audioTracks->length()) {
AudioTrack* track = audioTracks->lastItem();
// 5.3.1 Set the sourceBuffer attribute on the AudioTrack object to null.
track->setSourceBuffer(nullptr);
// 5.3.2 If the enabled attribute on the AudioTrack object is true, then set the removed enabled
// audio track flag to true.
if (track->enabled())
removedEnabledAudioTrack = true;
// 5.3.3 Remove the AudioTrack object from the HTMLMediaElement audioTracks list.
// 5.3.4 Queue a task to fire a trusted event named removetrack, that does not bubble and is not
// cancelable, and that uses the TrackEvent interface, at the HTMLMediaElement audioTracks list.
if (mediaElement())
mediaElement()->removeAudioTrack(track);
// 5.3.5 Remove the AudioTrack object from the SourceBuffer audioTracks list.
// 5.3.6 Queue a task to fire a trusted event named removetrack, that does not bubble and is not
// cancelable, and that uses the TrackEvent interface, at the SourceBuffer audioTracks list.
audioTracks->remove(track);
}
// 5.4 If the removed enabled audio track flag equals true, then queue a task to fire a simple event
// named change at the HTMLMediaElement audioTracks list.
if (removedEnabledAudioTrack)
mediaElement()->audioTracks()->scheduleChangeEvent();
}
// 6. Let SourceBuffer videoTracks list equal the VideoTrackList object returned by sourceBuffer.videoTracks.
RefPtr<VideoTrackList> videoTracks = buffer.videoTracks();
// 7. If the SourceBuffer videoTracks list is not empty, then run the following steps:
if (videoTracks->length()) {
// 7.1 Let HTMLMediaElement videoTracks list equal the VideoTrackList object returned by the videoTracks
// attribute on the HTMLMediaElement.
// 7.2 Let the removed selected video track flag equal false.
bool removedSelectedVideoTrack = false;
// 7.3 For each VideoTrack object in the SourceBuffer videoTracks list, run the following steps:
while (videoTracks->length()) {
VideoTrack* track = videoTracks->lastItem();
// 7.3.1 Set the sourceBuffer attribute on the VideoTrack object to null.
track->setSourceBuffer(nullptr);
// 7.3.2 If the selected attribute on the VideoTrack object is true, then set the removed selected
// video track flag to true.
if (track->selected())
removedSelectedVideoTrack = true;
// 7.3.3 Remove the VideoTrack object from the HTMLMediaElement videoTracks list.
// 7.3.4 Queue a task to fire a trusted event named removetrack, that does not bubble and is not
// cancelable, and that uses the TrackEvent interface, at the HTMLMediaElement videoTracks list.
if (mediaElement())
mediaElement()->removeVideoTrack(track);
// 7.3.5 Remove the VideoTrack object from the SourceBuffer videoTracks list.
// 7.3.6 Queue a task to fire a trusted event named removetrack, that does not bubble and is not
// cancelable, and that uses the TrackEvent interface, at the SourceBuffer videoTracks list.
videoTracks->remove(track);
}
// 7.4 If the removed selected video track flag equals true, then queue a task to fire a simple event
// named change at the HTMLMediaElement videoTracks list.
if (removedSelectedVideoTrack)
mediaElement()->videoTracks()->scheduleChangeEvent();
}
// 8. Let SourceBuffer textTracks list equal the TextTrackList object returned by sourceBuffer.textTracks.
RefPtr<TextTrackList> textTracks = buffer.textTracks();
// 9. If the SourceBuffer textTracks list is not empty, then run the following steps:
if (textTracks->length()) {
// 9.1 Let HTMLMediaElement textTracks list equal the TextTrackList object returned by the textTracks
// attribute on the HTMLMediaElement.
// 9.2 Let the removed enabled text track flag equal false.
bool removedEnabledTextTrack = false;
// 9.3 For each TextTrack object in the SourceBuffer textTracks list, run the following steps:
while (textTracks->length()) {
TextTrack* track = textTracks->lastItem();
// 9.3.1 Set the sourceBuffer attribute on the TextTrack object to null.
track->setSourceBuffer(nullptr);
// 9.3.2 If the mode attribute on the TextTrack object is set to "showing" or "hidden", then
// set the removed enabled text track flag to true.
if (track->mode() == TextTrack::showingKeyword() || track->mode() == TextTrack::hiddenKeyword())
removedEnabledTextTrack = true;
// 9.3.3 Remove the TextTrack object from the HTMLMediaElement textTracks list.
// 9.3.4 Queue a task to fire a trusted event named removetrack, that does not bubble and is not
// cancelable, and that uses the TrackEvent interface, at the HTMLMediaElement textTracks list.
if (mediaElement())
mediaElement()->removeTextTrack(track);
// 9.3.5 Remove the TextTrack object from the SourceBuffer textTracks list.
// 9.3.6 Queue a task to fire a trusted event named removetrack, that does not bubble and is not
// cancelable, and that uses the TrackEvent interface, at the SourceBuffer textTracks list.
textTracks->remove(track);
}
// 9.4 If the removed enabled text track flag equals true, then queue a task to fire a simple event
// named change at the HTMLMediaElement textTracks list.
if (removedEnabledTextTrack)
mediaElement()->textTracks()->scheduleChangeEvent();
}
// 10. If sourceBuffer is in activeSourceBuffers, then remove sourceBuffer from activeSourceBuffers ...
m_activeSourceBuffers->remove(buffer);
// 11. Remove sourceBuffer from sourceBuffers and fire a removesourcebuffer event
// on that object.
m_sourceBuffers->remove(buffer);
// 12. Destroy all resources for sourceBuffer.
buffer.removedFromMediaSource();
}
Route name2route(const QString& rn, bool /*dst*/, int rtype)/*{{{*/
{
// printf("name2route %s\n", rn.toLatin1().constData());
int channel = -1;
QString s(rn);
// Support old route style in oom files. Obsolete.
if (rn.size() >= 2 && rn[0].isNumber() && rn[1] == ':')
{
channel = rn[0].toAscii() - int('1');
s = rn.mid(2);
}
if (rtype == -1)
{
if (checkAudioDevice())
{
void* p = audioDevice->findPort(s.toLatin1().constData());
if (p)
return Route(p, channel);
}
TrackList* tl = song->tracks();
for (iTrack i = tl->begin(); i != tl->end(); ++i)
{
if ((*i)->isMidiTrack())
{
MidiTrack* track = (MidiTrack*) * i;
if (track->name() == s)
return Route(track, channel);
}
else
{
AudioTrack* track = (AudioTrack*) * i;
if (track->name() == s)
return Route(track, channel);
}
}
for (iMidiDevice i = midiDevices.begin(); i != midiDevices.end(); ++i)
{
if ((*i)->name() == s)
return Route(*i, channel);
}
// p3.3.49
if (s.left(ROUTE_MIDIPORT_NAME_PREFIX.length()) == ROUTE_MIDIPORT_NAME_PREFIX)
{
bool ok = false;
int port = s.mid(ROUTE_MIDIPORT_NAME_PREFIX.length()).toInt(&ok);
if (ok)
return Route(port, channel);
}
}
else
{
if (rtype == Route::TRACK_ROUTE)
{
TrackList* tl = song->tracks();
for (iTrack i = tl->begin(); i != tl->end(); ++i)
{
if ((*i)->isMidiTrack())
{
MidiTrack* track = (MidiTrack*) * i;
if (track->name() == s)
return Route(track, channel);
}
else
{
AudioTrack* track = (AudioTrack*) * i;
if (track->name() == s)
return Route(track, channel);
}
}
}// TODO Distinguish the device types
else if (rtype == Route::MIDI_DEVICE_ROUTE)
{
for (iMidiDevice i = midiDevices.begin(); i != midiDevices.end(); ++i)
{
if ((*i)->name() == s)
return Route(*i, channel);
}
}
else if (rtype == Route::JACK_ROUTE)
{
if (checkAudioDevice())
{
void* p = audioDevice->findPort(s.toLatin1().constData());
if (p)
return Route(p, channel);
}
}
else if (rtype == Route::MIDI_PORT_ROUTE) // p3.3.49
{
if (s.left(ROUTE_MIDIPORT_NAME_PREFIX.length()) == ROUTE_MIDIPORT_NAME_PREFIX)
{
bool ok = false;
int port = s.mid(ROUTE_MIDIPORT_NAME_PREFIX.length()).toInt(&ok);
if (ok)
return Route(port, channel);
}
}
}
printf(" name2route: <%s> not found\n", rn.toLatin1().constData());
return Route((Track*) 0, channel);
}/*}}}*/
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;
}
/*
==============
SNDDMA_Shutdown
Reset the sound device for exiting
===============
*/
void SNDDMA_Shutdown(void)
{
gAudioTrack.stop();
}
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)
{
int pdepth = pipeline->size();
if (fxid > pdepth)
return false;
pipeline->setOn(fxid, onoff);
}
break;
}
case QPybridgeEvent::SONG_ADD_TRACK:
song->addTrack(e->getP1());
song->updateTrackViews();
break;
case QPybridgeEvent::SONG_CHANGE_TRACKNAME:
{
Track* t = this->findTrack(e->getS1());
if (t == NULL)
return false;
t->setName(e->getS2());
break;
}
case QPybridgeEvent::SONG_DELETE_TRACK:
{
Track* t = this->findTrack(e->getS1());
if (t == NULL)
return false;
audio->msgRemoveTrack(t);
break;
}
default:
printf("Unknown pythonthread event received: %d\n", e->getType());
break;
}
return true;
}
int libmediacb_start(msm_ctx *ctx, int channels, int samplerate) {
__android_log_print(ANDROID_LOG_INFO,"liblossless","libmedia_ START REEEEACHED REACHEDDDDDD1");
status_t status;
int chans;
if(!ctx) return LIBLOSSLESS_ERR_NOCTX;
__android_log_print(ANDROID_LOG_INFO,"liblossless","libmediacb_start ctx=%p chans=%d rate=%d afd=%d atrack=%p",
ctx, channels, samplerate,ctx->afd,ctx->track);
AudioTrack* atrack = (AudioTrack *) ctx->track;
if(atrack && ctx->samplerate == samplerate && ctx->channels == channels) {
__android_log_print(ANDROID_LOG_INFO,"liblossless","same audio track parameters, restarting");
atrack->stop();
atrack->flush();
ctx->cbstart = 0; ctx->cbend = 0;
atrack->start();
return 0;
}
if(!ctx->cbbuf) {
ctx->cbbuf = (unsigned char *) malloc(DEFAULT_CB_BUFSZ);
if(!ctx->cbbuf) return LIBLOSSLESS_ERR_NOMEM;
ctx->cbbuf_size = DEFAULT_CB_BUFSZ;
}
ctx->cbstart = 0; ctx->cbend = 0;
if(!atrack) {
atrack = new AudioTrack();
if(!atrack) {
__android_log_print(ANDROID_LOG_ERROR,"liblossless","could not create AudioTrack!");
return LIBLOSSLESS_ERR_INIT;
}
__android_log_print(ANDROID_LOG_INFO,"liblossless","AudioTrack created at %p. Now trying to setup (buffsz %d)",
atrack, DEFAULT_ATRACK_CONF_BUFSZ);
if(!sdk_version) {
char c[PROP_VALUE_MAX];
if(__system_property_get("ro.build.version.sdk",c) > 0) sscanf(c,"%d",&sdk_version);
else sdk_version = 8;
__android_log_print(ANDROID_LOG_INFO,"liblossless","got sdk_version %d", sdk_version);
}
if(sdk_version > 13) chans = (channels == 2) ? 3 : 1;
else if(sdk_version > 6) chans = (channels == 2) ? 12 : 4;
else chans = channels;
#ifdef BUILD_JB
status = atrack->set(_MUSIC, samplerate, FMTBPS, chans, DEFAULT_ATRACK_CONF_BUFSZ/(2*channels),AUDIO_OUTPUT_FLAG_NONE,cbf,ctx);
#else
status = atrack->set(_MUSIC, samplerate, FMTBPS, chans, DEFAULT_ATRACK_CONF_BUFSZ/(2*channels),0,cbf,ctx);
#endif
if(status != NO_ERROR) {
__android_log_print(ANDROID_LOG_INFO,"liblossless","AudioTrack setup failed");
delete atrack;
return LIBLOSSLESS_ERR_INIT;
}
ctx->track = atrack;
} else {
atrack->stop();
atrack->flush();
ctx->cbstart = 0; ctx->cbend = 0;
__android_log_print(ANDROID_LOG_INFO,"liblossless","trying to reconfigure old AudioTrack");
status = atrack->setSampleRate(samplerate);
if(status != NO_ERROR) {
__android_log_print(ANDROID_LOG_INFO,"liblossless","could not set AudioTrack sample rate");
return LIBLOSSLESS_ERR_INIT;
}
}
__android_log_print(ANDROID_LOG_INFO,"liblossless","AudioTrack setup OK, starting audio!");
ctx->conf_size = DEFAULT_CONF_BUFSZ;
atrack->start();
__android_log_print(ANDROID_LOG_INFO,"liblossless","playback started!");
atrack->setPositionUpdatePeriod(0);
atrack->setMarkerPosition(0);
static int s(0); if(!s) { print_priority(__FUNCTION__); s = 1; }
return 0;
}
void AudioPortConfig::trackSelectionChanged()
{
routeList->clear();
newSrcList->clear();
newDstList->clear();
QListWidgetItem* titem = tracksList->currentItem();
AudioTrack* atrack = (AudioTrack*)song->findTrack(titem->text());
if(atrack)
{
_selected = atrack;
selectedIndex = tracksList->row(titem);
//TrackList* tl = song->tracks();
//for(iTrack t = tl->begin(); t != tl->end(); ++t)
//{
// if((*t)->isMidiTrack())
// continue;
// AudioTrack* track = (AudioTrack*) (*t);
// if(track->name() == atrack->name())
// continue; //You cant connect a track to itself
//int channels = track->channels();
switch (atrack->type())
{
case Track::WAVE_OUTPUT_HELPER:/*{{{*/
for(iTrack t = song->tracks()->begin(); t != song->tracks()->end(); ++t)
{
if((*t)->isMidiTrack())
continue;
AudioTrack* track = (AudioTrack*) (*t);
if(track->name() == atrack->name() || track->type() == Track::WAVE_OUTPUT_HELPER)
continue; //You cant connect a track to itself
//for (int channel = 0; channel < track->channels(); ++channel)
//{
Route r(track, -1);
newSrcList->addItem(r.name());
//}
}
insertInputs();
//newDstList->addItem(Route(track, -1).name());
break;
case Track::WAVE_INPUT_HELPER:
for(iTrack t = song->tracks()->begin(); t != song->tracks()->end(); ++t)
{
if((*t)->isMidiTrack())
continue;
AudioTrack* track = (AudioTrack*) (*t);
if(track->name() == atrack->name())
continue; //You cant connect a track to itself
switch(track->type())
{
case Track::WAVE_OUTPUT_HELPER:
case Track::WAVE:
newDstList->addItem(Route(track, -1).name());
break;
default:
break;
}
}
insertOutputs();
break;
case Track::WAVE:
for(iTrack t = song->tracks()->begin(); t != song->tracks()->end(); ++t)
{
if((*t)->isMidiTrack())
continue;
AudioTrack* track = (AudioTrack*) (*t);
if(track->name() == atrack->name())
continue; //You cant connect a track to itself
if(track->type() == Track::WAVE_INPUT_HELPER)
{
newSrcList->addItem(Route(track, -1).name());
}
else if(track->type() == Track::WAVE_OUTPUT_HELPER)
{
newDstList->addItem(Route(track, -1).name());
}
}
break;
default:
break;/*}}}*/
}
//}
QTreeWidgetItem* widgetItem;
const RouteList* rl = atrack->outRoutes();
for (ciRoute r = rl->begin(); r != rl->end(); ++r)
{
QString src("");
if (atrack->type() == Track::WAVE_OUTPUT_HELPER)
{
widgetItem = new QTreeWidgetItem(routeList, QStringList() << src << QString("") << atrack->name() << r->name() << QString::number(r->channel), Track::WAVE_OUTPUT_HELPER);
}
else
{
widgetItem = new QTreeWidgetItem(routeList, QStringList() << src << QString("") << atrack->name() << r->name() << QString::number(0), Track::WAVE_OUTPUT_HELPER);
}
widgetItem->setTextAlignment(1, Qt::AlignHCenter);
widgetItem->setTextAlignment(4, Qt::AlignHCenter);
}
const RouteList* rli = atrack->inRoutes();
for (ciRoute ri = rli->begin(); ri != rli->end(); ++ri)
{
QString src("");
if (atrack->type() == Track::WAVE_INPUT_HELPER)
{
widgetItem = new QTreeWidgetItem(routeList, QStringList() << ri->name() << QString::number(ri->channel) << atrack->name() << src << QString(""), Track::WAVE_INPUT_HELPER);
}
else
{
widgetItem = new QTreeWidgetItem(routeList, QStringList() << ri->name() << QString::number(0) << atrack->name() << src << QString(""), Track::WAVE_INPUT_HELPER);
}
widgetItem->setTextAlignment(1, Qt::AlignHCenter);
widgetItem->setTextAlignment(4, Qt::AlignHCenter);
}
routeSelectionChanged(); // init remove button
srcSelectionChanged(); // init select button
}
}
/**
* \brief output initialization
* \param audec pointer to audec
* \return 0 on success otherwise negative error code
*/
extern "C" int android_init(struct aml_audio_dec* audec)
{
adec_print("android out init");
status_t status;
AudioTrack *track;
audio_out_operations_t *out_ops = &audec->aout_ops;
Mutex::Autolock _l(mLock);
track = new AudioTrack();
if (track == NULL) {
adec_print("AudioTrack Create Failed!");
return -1;
}
int SessionID = audec->SessionID;
adec_print("SessionID = %d",SessionID);
#if defined(_VERSION_JB)
status = track->set(AUDIO_STREAM_MUSIC,
audec->samplerate,
AUDIO_FORMAT_PCM_16_BIT,
(audec->channels == 1) ? AUDIO_CHANNEL_OUT_MONO : AUDIO_CHANNEL_OUT_STEREO,
0, // frameCount
AUDIO_OUTPUT_FLAG_NONE, // flags
audioCallback,
audec, // user when callback
0, // notificationFrames
0, // shared buffer
false, // threadCanCallJava
SessionID); // sessionId
#elif defined(_VERSION_ICS)
status = track->set(AUDIO_STREAM_MUSIC,
audec->samplerate,
AUDIO_FORMAT_PCM_16_BIT,
(audec->channels == 1) ? AUDIO_CHANNEL_OUT_MONO : AUDIO_CHANNEL_OUT_STEREO,
0, // frameCount
0, // flags
audioCallback,
audec, // user when callback
0, // notificationFrames
0, // shared buffer
false, // threadCanCallJava
SessionID); // sessionId
#else // GB or lower:
status = track->set(AudioSystem::MUSIC,
audec->samplerate,
AudioSystem::PCM_16_BIT,
(audec->channels == 1) ? AudioSystem::CHANNEL_OUT_MONO : AudioSystem::CHANNEL_OUT_STEREO,
0, // frameCount
0, // flags
audioCallback,
audec, // user when callback
0, // notificationFrames
0, // shared buffer
SessionID);
#endif
if (status != NO_ERROR) {
adec_print("track->set returns %d", status);
adec_print("audio out samplet %d", audec->samplerate);
adec_print("audio out channels %d", audec->channels);
delete track;
track = NULL;
return -1;
}
af_resample_linear_init();
out_ops->private_data = (void *)track;
return 0;
}
bool APlaybackDevice::isPlaying()
{
return mPlaybackEnabled && !mAudioTrack.stopped();
}
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;
}
std::vector<WebMediaPlayer::TrackId> SourceBuffer::initializationSegmentReceived(const std::vector<MediaTrackInfo>& newTracks)
{
WTF_LOG(Media, "SourceBuffer::initializationSegmentReceived %p tracks=%zu", this, newTracks.size());
ASSERT(m_source);
ASSERT(m_source->mediaElement());
ASSERT(m_updating);
// TODO(servolk): Implement proper 'initialization segment received' algorithm according to MSE spec:
// https://w3c.github.io/media-source/#sourcebuffer-init-segment-received
std::vector<WebMediaPlayer::TrackId> result;
for (const auto& trackInfo : newTracks) {
const auto& trackType = std::get<0>(trackInfo);
const auto& id = std::get<1>(trackInfo);
const auto& kind = std::get<2>(trackInfo);
const auto& label = std::get<3>(trackInfo);
const auto& language = std::get<4>(trackInfo);
if (!RuntimeEnabledFeatures::audioVideoTracksEnabled()) {
static WebMediaPlayer::TrackId nextTrackId = 0;
result.push_back(++nextTrackId);
continue;
}
const TrackBase* trackBase = nullptr;
if (trackType == WebMediaPlayer::AudioTrack) {
AudioTrack* audioTrack = nullptr;
if (!m_firstInitializationSegmentReceived) {
audioTrack = AudioTrack::create(id, kind, label, language, false);
audioTracks().add(audioTrack);
m_source->mediaElement()->audioTracks().add(audioTrack);
} else {
audioTrack = findExistingTrackById(audioTracks(), id);
ASSERT(audioTrack);
}
trackBase = audioTrack;
result.push_back(audioTrack->trackId());
} else if (trackType == WebMediaPlayer::VideoTrack) {
VideoTrack* videoTrack = nullptr;
if (!m_firstInitializationSegmentReceived) {
videoTrack = VideoTrack::create(id, kind, label, language, false);
videoTracks().add(videoTrack);
m_source->mediaElement()->videoTracks().add(videoTrack);
} else {
videoTrack = findExistingTrackById(videoTracks(), id);
ASSERT(videoTrack);
}
trackBase = videoTrack;
result.push_back(videoTrack->trackId());
} else {
NOTREACHED();
}
(void)trackBase;
#if !LOG_DISABLED
const char* logActionStr = m_firstInitializationSegmentReceived ? "using existing" : "added";
const char* logTrackTypeStr = (trackType == WebMediaPlayer::AudioTrack) ? "audio" : "video";
WTF_LOG(Media, "Tracks (sb=%p): %s %sTrack %p trackId=%d id=%s label=%s lang=%s", this, logActionStr, logTrackTypeStr, trackBase, trackBase->trackId(), trackBase->id().utf8().data(), trackBase->label().utf8().data(), trackBase->language().utf8().data());
#endif
}
if (!m_firstInitializationSegmentReceived) {
// 5. If active track flag equals true, then run the following steps:
// 5.1. Add this SourceBuffer to activeSourceBuffers.
// 5.2. Queue a task to fire a simple event named addsourcebuffer at
// activesourcebuffers.
m_source->setSourceBufferActive(this);
// 6. Set first initialization segment received flag to true.
m_firstInitializationSegmentReceived = true;
}
return result;
}
