status_t AudioALSAStreamIn::open()
{
// call open() only when mLock is locked.
ASSERT(mLock.tryLock() != 0);
ALOGD("%s()", __FUNCTION__);
status_t status = NO_ERROR;
if (mStandby == true)
{
mStandby = false;
// create capture handler
ASSERT(mCaptureHandler == NULL);
mCaptureHandler = mStreamManager->createCaptureHandler(&mStreamAttributeTarget);
// open audio hardware
status = mCaptureHandler->open();
ASSERT(status == NO_ERROR);
OpenPCMDump();
}
return status;
}
status_t AudioALSAStreamOut::open()
{
// call open() only when mLock is locked.
ASSERT(mLock.tryLock() != 0);
ALOGD("%s()", __FUNCTION__);
status_t status = NO_ERROR;
if (mStandby == true)
{
mStandby = false;
// HDMI stereo + HDMI multi-channel => disable HDMI stereo
if (mStreamOutType == STREAM_OUT_HDMI_MULTI_CHANNEL)
{
ALOGD("Force disable mStreamOutHDMIStereo");
AudioALSAStreamOut::setSuspendStreamOutHDMIStereo(true);
if (mStreamOutHDMIStereo != NULL) { mStreamOutHDMIStereo->standby(); }
}
// create playback handler
ASSERT(mPlaybackHandler == NULL);
AudioALSASampleRateController::getInstance()->setScenarioStatus(PLAYBACK_SCENARIO_STREAM_OUT);
mPlaybackHandler = mStreamManager->createPlaybackHandler(&mStreamAttributeSource);
// open audio hardware
status = mPlaybackHandler->open();
ASSERT(status == NO_ERROR);
OpenPCMDump(LOG_TAG);
}
return status;
}
status_t AudioALSAPlaybackHandlerBTSCO::open()
{
ALOGD("+%s(), mDevice = 0x%x", __FUNCTION__, mStreamAttributeSource->output_devices);
// debug pcm dump
OpenPCMDump(LOG_TAG);
int pcmindex = AudioALSADeviceParser::getInstance()->GetPcmIndexByString(keypcmVOIPCallBTPlayback);
int cardindex = AudioALSADeviceParser::getInstance()->GetCardIndexByString(keypcmVOIPCallBTPlayback);
struct pcm_params *params;
params = pcm_params_get(cardindex, pcmindex, PCM_OUT);
if (params == NULL)
{
ALOGD("Device does not exist.\n");
}
mStreamAttributeTarget.buffer_size = pcm_params_get_max(params, PCM_PARAM_BUFFER_BYTES);
ALOGD("buffersizemax = %d", mStreamAttributeTarget.buffer_size);
pcm_params_free(params);
// HW attribute config // TODO(Harvey): query this
#ifdef PLAYBACK_USE_24BITS_ONLY
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_8_24_BIT;
#else
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_16_BIT;
#endif
mStreamAttributeTarget.audio_channel_mask = AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeTarget.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeTarget.audio_channel_mask);
mStreamAttributeTarget.sample_rate = mWCNChipController->GetBTCurrentSamplingRateNumber();
// HW pcm config
mConfig.channels = mStreamAttributeTarget.num_channels;
mConfig.rate = mStreamAttributeTarget.sample_rate;
mConfig.period_count = 2;
mConfig.period_size = (mStreamAttributeTarget.buffer_size / (mConfig.channels * mConfig.period_count)) / ((mStreamAttributeTarget.audio_format == AUDIO_FORMAT_PCM_16_BIT) ? 2 : 4);
mConfig.format = transferAudioFormatToPcmFormat(mStreamAttributeTarget.audio_format);
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), mConfig: channels = %d, rate = %d, period_size = %d, period_count = %d, format = %d",
__FUNCTION__, mConfig.channels, mConfig.rate, mConfig.period_size, mConfig.period_count, mConfig.format);
// SRC
initBliSrc();
// bit conversion
initBitConverter();
// open pcm driver
openPcmDriver(pcmindex);
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
status_t AudioALSACaptureDataProviderBTSCO::open()
{
ALOGD("%s()", __FUNCTION__);
ASSERT(mClientLock.tryLock() != 0); // lock by base class attach
AudioAutoTimeoutLock _l(mEnableLock);
ASSERT(mEnable == false);
// config attribute (will used in client SRC/Enh/... later) // TODO(Harvey): query this
mStreamAttributeSource.audio_format = AUDIO_FORMAT_PCM_16_BIT;
mStreamAttributeSource.audio_channel_mask = AUDIO_CHANNEL_IN_MONO;
mStreamAttributeSource.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeSource.audio_channel_mask);
mStreamAttributeSource.sample_rate = mWCNChipController->GetBTCurrentSamplingRateNumber();
// pcm config
mConfig.channels = mStreamAttributeSource.num_channels;
mConfig.rate = mStreamAttributeSource.sample_rate;
// Buffer size: 2048(period_size) * 1(ch) * 2(byte) * 4(period_count) = 16 kb
mConfig.period_size = 2048;
mConfig.period_count = 4;
mConfig.format = PCM_FORMAT_S16_LE;
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), audio_format = %d, audio_channel_mask=%x, num_channels=%d, sample_rate=%d", __FUNCTION__,
mStreamAttributeSource.audio_format, mStreamAttributeSource.audio_channel_mask, mStreamAttributeSource.num_channels, mStreamAttributeSource.sample_rate);
ALOGD("%s(), format = %d, channels=%d, rate=%d", __FUNCTION__,
mConfig.format, mConfig.channels, mConfig.rate);
OpenPCMDump(LOG_TAG);
// enable pcm
ASSERT(mPcm == NULL);
mPcm = pcm_open(0, 12, PCM_IN | PCM_MONOTONIC, &mConfig);
ASSERT(mPcm != NULL && pcm_is_ready(mPcm) == true);
ALOGV("%s(), mPcm = %p", __FUNCTION__, mPcm);
pcm_start(mPcm);
// create reading thread
mEnable = true;
int ret = pthread_create(&hReadThread, NULL, AudioALSACaptureDataProviderBTSCO::readThread, (void *)this);
if (ret != 0)
{
ALOGE("%s() create thread fail!!", __FUNCTION__);
return UNKNOWN_ERROR;
}
return NO_ERROR;
}
status_t AudioALSACaptureDataProviderVoiceDL::open()
{
ALOGD("%s()", __FUNCTION__);
ASSERT(mClientLock.tryLock() != 0); // lock by base class attach
AudioAutoTimeoutLock _l(mEnableLock);
ASSERT(mEnable == false);
SpeechDataProcessingHandler::getInstance()->getStreamAttributeSource(&mStreamAttributeSource);
ALOGD("%s(), mStreamAttributeSource: audio_format = %d, num_channels = %d, audio_channel_mask = %x, sample_rate = %d\n", __FUNCTION__, mStreamAttributeSource.audio_format, mStreamAttributeSource.num_channels, mStreamAttributeSource.audio_channel_mask, mStreamAttributeSource.sample_rate);
mEnable = true;
OpenPCMDump(LOG_TAG);
return SpeechDataProcessingHandler::getInstance()->recordOn(RECORD_TYPE_DL);
}
status_t AudioALSACaptureDataProviderBTCVSD::open()
{
ALOGD("%s()", __FUNCTION__);
ASSERT(mClientLock.tryLock() != 0); // lock by base class attach
AudioAutoTimeoutLock _l(mEnableLock);
ASSERT(mEnable == false);
// config attribute (will used in client SRC/Enh/... later)
mStreamAttributeSource.audio_format = AUDIO_FORMAT_PCM_16_BIT;
mStreamAttributeSource.audio_channel_mask = AUDIO_CHANNEL_IN_MONO;
mStreamAttributeSource.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeSource.audio_channel_mask);
mStreamAttributeSource.sample_rate = mWCNChipController->GetBTCurrentSamplingRateNumber();
if (mAudioBTCVSDControl->BT_SCO_isWideBand() == true)
{
mReadBufferSize = MSBC_PCM_FRAME_BYTE * 6 * 2; // 16k mono->48k stereo
}
else
{
mReadBufferSize = SCO_RX_PCM8K_BUF_SIZE * 12 * 2; // 8k mono->48k stereo
}
ALOGD("%s(), audio_format = %d, audio_channel_mask=%x, num_channels=%d, sample_rate=%d", __FUNCTION__,
mStreamAttributeSource.audio_format, mStreamAttributeSource.audio_channel_mask, mStreamAttributeSource.num_channels, mStreamAttributeSource.sample_rate);
OpenPCMDump(LOG_TAG);
// enable bt cvsd driver
mAudioBTCVSDControl->BT_SCO_RX_Begin(mFd2);
// create reading thread
mEnable = true;
int ret = pthread_create(&hReadThread, NULL, AudioALSACaptureDataProviderBTCVSD::readThread, (void *)this);
if (ret != 0)
{
ALOGE("%s() create thread fail!!", __FUNCTION__);
return UNKNOWN_ERROR;
}
return NO_ERROR;
}
status_t AudioALSACaptureDataProviderVoice::open()
{
ALOGD("%s()", __FUNCTION__);
ASSERT(mClientLock.tryLock() != 0); // lock by base class attach
AudioAutoTimeoutLock _l(mEnableLock);
ASSERT(mEnable == false);
// config attribute (will used in client SRC/Enh/... later)
SpeechDriverInterface *pSpeechDriver = SpeechDriverFactory::GetInstance()->GetSpeechDriver();
mStreamAttributeSource.audio_format = AUDIO_FORMAT_PCM_16_BIT;
mStreamAttributeSource.num_channels = pSpeechDriver->GetRecordChannelNumber();
mStreamAttributeSource.audio_channel_mask = (mStreamAttributeSource.num_channels == 1) ? AUDIO_CHANNEL_IN_MONO : AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeSource.sample_rate = pSpeechDriver->GetRecordSampleRate();
mEnable = true;
OpenPCMDump(LOG_TAG);
return SpeechDriverFactory::GetInstance()->GetSpeechDriver()->RecordOn();
}
status_t AudioALSACaptureDataProviderTDM::open()
{
ALOGD("%s()", __FUNCTION__);
ASSERT(mClientLock.tryLock() != 0); // lock by base class attach
AudioAutoTimeoutLock _l(mEnableLock);
AudioAutoTimeoutLock _l2(*AudioALSADriverUtility::getInstance()->getStreamSramDramLock());
ASSERT(mEnable == false);
// config attribute (will used in client SRC/Enh/... later) // TODO(Harvey): query this
mStreamAttributeSource.audio_format = AUDIO_FORMAT_PCM_16_BIT;
mStreamAttributeSource.audio_channel_mask = AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeSource.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeSource.audio_channel_mask);
mStreamAttributeSource.sample_rate = 44100;
OpenPCMDump(LOG_TAG);
// enable pcm
ASSERT(mPcm == NULL);
mPcm = pcm_open(0, 13, PCM_IN, &mConfig);
// mPcm = pcm_open(0, 1, PCM_IN, &mConfig);
ASSERT(mPcm != NULL && pcm_is_ready(mPcm) == true);
ALOGV("%s(), mPcm = %p", __FUNCTION__, mPcm);
pcm_start(mPcm);
// create reading thread
mEnable = true;
int ret = pthread_create(&hReadThread, NULL, AudioALSACaptureDataProviderTDM::readThread, (void *)this);
if (ret != 0)
{
ALOGE("%s() create thread fail!!", __FUNCTION__);
return UNKNOWN_ERROR;
}
return NO_ERROR;
}
status_t AudioALSAPlaybackHandlerNormal::open()
{
ALOGD("+%s(), mDevice = 0x%x", __FUNCTION__, mStreamAttributeSource->output_devices);
AudioAutoTimeoutLock _l(*AudioALSADriverUtility::getInstance()->getStreamSramDramLock());
// debug pcm dump
OpenPCMDump(LOG_TAG);
// acquire pmic clk
mHardwareResourceManager->EnableAudBufClk(true);
HpImpeDanceDetect();
int pcmindex = AudioALSADeviceParser::getInstance()->GetPcmIndexByString(keypcmI2S0Dl1Playback);
int cardindex = AudioALSADeviceParser::getInstance()->GetCardIndexByString(keypcmI2S0Dl1Playback);
ALOGD("AudioALSAPlaybackHandlerNormal::open() pcmindex = %d", pcmindex);
ListPcmDriver(cardindex, pcmindex);
struct pcm_params *params;
params = pcm_params_get(cardindex, pcmindex, PCM_OUT);
if (params == NULL)
{
ALOGD("Device does not exist.\n");
}
mStreamAttributeTarget.buffer_size = pcm_params_get_max(params, PCM_PARAM_BUFFER_BYTES);
pcm_params_free(params);
// HW attribute config // TODO(Harvey): query this
#ifdef PLAYBACK_USE_24BITS_ONLY
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_8_24_BIT;
#else
mStreamAttributeTarget.audio_format = (mStreamAttributeSource->audio_format == AUDIO_FORMAT_PCM_32_BIT) ? AUDIO_FORMAT_PCM_8_24_BIT : AUDIO_FORMAT_PCM_16_BIT;
#endif
mStreamAttributeTarget.audio_channel_mask = AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeTarget.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeTarget.audio_channel_mask);
mStreamAttributeTarget.sample_rate = ChooseTargetSampleRate(mStreamAttributeSource->sample_rate,mStreamAttributeSource->output_devices);
// HW pcm config
mConfig.channels = mStreamAttributeTarget.num_channels;
mConfig.rate = mStreamAttributeTarget.sample_rate;
// Buffer size: 1536(period_size) * 2(ch) * 4(byte) * 2(period_count) = 24 kb
mConfig.period_count = 2;
mConfig.period_size = (mStreamAttributeTarget.buffer_size / (mConfig.channels * mConfig.period_count)) / ((mStreamAttributeTarget.audio_format == AUDIO_FORMAT_PCM_16_BIT) ? 2 : 4);
mConfig.format = transferAudioFormatToPcmFormat(mStreamAttributeTarget.audio_format);
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), mConfig: channels = %d, rate = %d, period_size = %d, period_count = %d, format = %d",
__FUNCTION__, mConfig.channels, mConfig.rate, mConfig.period_size, mConfig.period_count, mConfig.format);
// post processing
initPostProcessing();
#if defined(MTK_SPEAKER_MONITOR_SUPPORT)
unsigned int fc, bw;
int th;
if (mAudioFilterManagerHandler)
{
AudioALSASpeakerMonitor::getInstance()->GetFilterParam(&fc, &bw, &th);
ALOGD("%s(), fc %d bw %d, th %d", __FUNCTION__, fc, bw, th);
mAudioFilterManagerHandler->setSpkFilterParam(fc, bw, th);
}
#endif
// SRC
initBliSrc();
// bit conversion
initBitConverter();
initDataPending();
// disable lowjitter mode
SetLowJitterMode(true, mStreamAttributeTarget.sample_rate);
// open pcm driver
openPcmDriver(pcmindex);
// open codec driver
mHardwareResourceManager->startOutputDevice(mStreamAttributeSource->output_devices, mStreamAttributeTarget.sample_rate);
//============Voice UI&Unlock REFERECE=============
AudioVUnlockDL *VUnlockhdl = AudioVUnlockDL::getInstance();
if (VUnlockhdl != NULL)
{
VUnlockhdl->SetInputStandBy(false);
VUnlockhdl-> GetSRCInputParameter(mStreamAttributeTarget.sample_rate, mStreamAttributeTarget.num_channels, mStreamAttributeTarget.audio_format);
VUnlockhdl->GetFirstDLTime();
}
//===========================================
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
status_t AudioALSACaptureDataProviderEchoRefExt::open()
{
ALOGD("%s()", __FUNCTION__);
ASSERT(mClientLock.tryLock() != 0); // lock by base class attach
AudioAutoTimeoutLock _l(mEnableLock);
AudioAutoTimeoutLock _l2(*AudioALSADriverUtility::getInstance()->getStreamSramDramLock());
ASSERT(mEnable == false);
AudioALSASampleRateController *pAudioALSASampleRateController = AudioALSASampleRateController::getInstance();
pAudioALSASampleRateController->setScenarioStatus(PLAYBACK_SCENARIO_ECHO_REF_EXT);
// config attribute (will used in client SRC/Enh/... later) // TODO(Sam): query the mConfig?
mStreamAttributeSource.audio_format = AUDIO_FORMAT_PCM_16_BIT;
mStreamAttributeSource.audio_channel_mask = AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeSource.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeSource.audio_channel_mask);
mStreamAttributeSource.sample_rate = AudioALSASampleRateController::getInstance()->getPrimaryStreamOutSampleRate();
mConfig.channels = mStreamAttributeSource.num_channels;
mConfig.rate = mStreamAttributeSource.sample_rate;
// Buffer size: 2048(period_size) * 2(ch) * 2(byte) * 8(period_count) = 64 kb
mConfig.period_size = 2048;
mConfig.period_count = 8;
mConfig.format = PCM_FORMAT_S16_LE;
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
#if 0
//latency time, set as DataProvider buffer size
mStreamAttributeSource.latency = (kReadBufferSize * 1000) / (mStreamAttributeSource.num_channels * mStreamAttributeSource.sample_rate *
(mStreamAttributeSource.audio_format == AUDIO_FORMAT_PCM_8_BIT ? 1 : //8 1byte/frame
(mStreamAttributeSource.audio_format == AUDIO_FORMAT_PCM_32_BIT ? 4 : //24bit 3bytes/frame
2))); //default 2bytes/sample
#else
//latency time, set as hardware buffer size
mStreamAttributeSource.latency = (mConfig.period_size * mConfig.period_count * 1000) / mConfig.rate;
#endif
ALOGD("%s(), audio_format = %d, audio_channel_mask=%x, num_channels=%d, sample_rate=%d, latency=%dms", __FUNCTION__,
mStreamAttributeSource.audio_format, mStreamAttributeSource.audio_channel_mask, mStreamAttributeSource.num_channels, mStreamAttributeSource.sample_rate, mStreamAttributeSource.latency);
ALOGD("%s(), format = %d, channels=%d, rate=%d", __FUNCTION__,
mConfig.format, mConfig.channels, mConfig.rate);
#ifdef NXP_SMARTPA_SUPPORT
MTK_Tfa98xx_EchoReferenceConfigure(1);
#endif
OpenPCMDump(LOG_TAG);
// enable pcm
ASSERT(mPcm == NULL);
mPcm = pcm_open(0, 16, PCM_IN | PCM_MONOTONIC, &mConfig);
ASSERT(mPcm != NULL && pcm_is_ready(mPcm) == true);
pcm_start(mPcm);
// create reading thread
mEnable = true;
int ret = pthread_create(&hReadThread, NULL, AudioALSACaptureDataProviderEchoRefExt::readThread, (void *)this);
if (ret != 0)
{
ALOGE("%s() create thread fail!!", __FUNCTION__);
return UNKNOWN_ERROR;
}
return NO_ERROR;
}
status_t AudioALSAPlaybackHandlerSphDL::open()
{
ALOGD("+%s(), mDevice = 0x%x", __FUNCTION__, mStreamAttributeSource->output_devices);
// debug pcm dump
OpenPCMDump(LOG_TAG);
int pcmindex = AudioALSADeviceParser::getInstance()->GetPcmIndexByString(keypcmI2S0Dl1Playback);
int cardindex = AudioALSADeviceParser::getInstance()->GetCardIndexByString(keypcmI2S0Dl1Playback);
ALOGD("AudioALSAPlaybackHandlerSphDL::open() pcmindex = %d", pcmindex);
ListPcmDriver(cardindex, pcmindex);
struct pcm_params *params;
params = pcm_params_get(cardindex, pcmindex, PCM_OUT);
if (params == NULL)
{
ALOGD("Device does not exist.\n");
}
mStreamAttributeTarget.buffer_size = pcm_params_get_max(params, PCM_PARAM_BUFFER_BYTES);
ALOGD("buffersizemax = %d", mStreamAttributeTarget.buffer_size);
pcm_params_free(params);
// HW attribute config // TODO(Harvey): query this
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_16_BIT;
mStreamAttributeTarget.audio_channel_mask = AUDIO_CHANNEL_IN_MONO;
mStreamAttributeTarget.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeTarget.audio_channel_mask);
mStreamAttributeTarget.sample_rate = mStreamAttributeSource->sample_rate; // same as source stream
// HW pcm config
mConfig.channels = mStreamAttributeTarget.num_channels;
mConfig.rate = mStreamAttributeTarget.sample_rate;
// Buffer size: 1536(period_size) * 2(ch) * 4(byte) * 2(period_count) = 24 kb
mConfig.period_count = 4;
//mConfig.period_size = (0x4000/(mConfig.channels*mConfig.period_count))/((mStreamAttributeTarget.audio_format == AUDIO_FORMAT_PCM_16_BIT) ? 2 : 4);
mConfig.period_size = 512;
mConfig.format = transferAudioFormatToPcmFormat(mStreamAttributeTarget.audio_format);
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), mConfig: channels = %d, rate = %d, period_size = %d, period_count = %d, format = %d",
__FUNCTION__, mConfig.channels, mConfig.rate, mConfig.period_size, mConfig.period_count, mConfig.format);
// post processing
initPostProcessing();
#if defined(MTK_SPEAKER_MONITOR_SUPPORT)
unsigned int fc, bw;
int th;
if (mAudioFilterManagerHandler)
{
AudioALSASpeakerMonitor::getInstance()->GetFilterParam(&fc, &bw, &th);
ALOGD("%s(), fc %d bw %d, th %d", __FUNCTION__, fc, bw, th);
mAudioFilterManagerHandler->setSpkFilterParam(fc, bw, th);
}
#endif
// SRC
initBliSrc();
// bit conversion
initBitConverter();
// open pcm driver
openPcmDriver(pcmindex);
//Echo reference path
if (mixer_ctl_set_enum_by_string(mixer_get_ctl_by_name(mMixer, "Audio_Dl1_MD_Echo_Ref_Switch"), "On"))
{
ALOGE("Error: Audio_Dl1_MD_Echo_Ref_Switch invalid value");
}
// open codec driver
// Don't startoutputDevice here, let speech driver to open.
mHardwareResourceManager->startOutputDevice(mStreamAttributeSource->output_devices, mStreamAttributeTarget.sample_rate);
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
status_t AudioALSAPlaybackHandlerOffload::open()
{
ALOGD("+%s(), mDevice = 0x%x", __FUNCTION__, mStreamAttributeSource->output_devices);
ALOGD("%s(), mStreamAttributeSource: format = %d",__FUNCTION__, mStreamAttributeSource->audio_format);
AudioAutoTimeoutLock _l(*AudioALSADriverUtility::getInstance()->getStreamSramDramLock());
// debug pcm dump
OpenPCMDump(LOG_TAG);
// acquire pmic clk
mHardwareResourceManager->EnableAudBufClk(true);
//doug to check
//HpImpeDanceDetect(); //doug to check
#if 1
int pcmindex = AudioALSADeviceParser::getInstance()->GetPcmIndexByString(keypcmDl1Meida);
int cardindex = AudioALSADeviceParser::getInstance()->GetCardIndexByString(keypcmDl1Meida);
ALOGD("AudioALSAPlaybackHandlerOffload::open() pcmindex = %d", pcmindex);
ListPcmDriver(cardindex, pcmindex);
struct pcm_params *params;
params = pcm_params_get(cardindex, pcmindex, PCM_OUT);
if (params == NULL)
{
ALOGD("Device does not exist.\n");
}
mStreamAttributeTarget.buffer_size = pcm_params_get_max(params, PCM_PARAM_BUFFER_BYTES);
ALOGD("%s buffersizemax = %d", __FUNCTION__, mStreamAttributeTarget.buffer_size);
pcm_params_free(params);
#endif
//mStreamAttributeTarget.buffer_size = 32768;
//#ifdef PLAYBACK_USE_24BITS_ONLY
//mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_32_BIT;
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_8_24_BIT;
//mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_16_BIT;
//#else
// mStreamAttributeTarget.audio_format = (mStreamAttributeSource->audio_format == AUDIO_FORMAT_PCM_32_BIT) ? AUDIO_FORMAT_PCM_8_24_BIT : AUDIO_FORMAT_PCM_16_BIT;
//#endif
mStreamAttributeTarget.audio_channel_mask = AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeTarget.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeTarget.audio_channel_mask);
mStreamAttributeTarget.sample_rate = ChooseTargetSampleRate(mStreamAttributeSource->sample_rate);
// HW pcm config
mConfig.channels = mStreamAttributeTarget.num_channels;
mConfig.rate = mStreamAttributeTarget.sample_rate;
// Buffer size: 1536(period_size) * 2(ch) * 4(byte) * 2(period_count) = 24 kb
mConfig.period_count = 2;
mConfig.period_size = (mStreamAttributeTarget.buffer_size / (mConfig.channels * mConfig.period_count)) / ((mStreamAttributeTarget.audio_format == AUDIO_FORMAT_PCM_16_BIT) ? 2 : 4);
mConfig.format = transferAudioFormatToPcmFormat(mStreamAttributeTarget.audio_format);
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), mConfig: channels = %d, rate = %d, period_size = %d, period_count = %d, format = %d",
__FUNCTION__, mConfig.channels, mConfig.rate, mConfig.period_size, mConfig.period_count, mConfig.format);
mComprConfig.codec = (struct snd_codec*)malloc(sizeof(struct snd_codec));
if(mComprConfig.codec == NULL)
ALOGE("%s(), allocate mComprConfig.codec fail");
mComprConfig.fragments = 1024;
mComprConfig.fragment_size = 8192;
//mComprConfig.fragment_size = mStreamAttributeTarget.buffer_size;
mComprConfig.codec->sample_rate = mStreamAttributeTarget.sample_rate;
mComprConfig.codec->reserved[0] = mConfig.period_size;
mComprConfig.codec->reserved[1] = mComprConfig.fragments*mComprConfig.fragment_size;
if(mConfig.format == PCM_FORMAT_S16_LE)
mComprConfig.codec->format = SNDRV_PCM_FORMAT_S16_LE;
else
mComprConfig.codec->format = SNDRV_PCM_FORMAT_S32_LE;
mComprConfig.codec->id = SND_AUDIOCODEC_MP3;
mComprConfig.codec->ch_in = 2;
mComprConfig.codec->ch_out = 2;
//init decoder
mDecHandler = AudioDecHandlerCreate();
if(mDecHandler == NULL)
{
ALOGE("+%s(), DecHandler create fail", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
if(!mDecHandler->InitAudioDecoder())
{
ALOGE("+%s(), Decoder IP init fail", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
// post processing
initPostProcessing();
// SRC
initBliSrc();
// bit conversion
initBitConverter();
initDataPending();
// disable lowjitter mode //doug to check
SetLowJitterMode(true, mStreamAttributeTarget.sample_rate);
openComprDriver(23);
if( compress_set_gapless_metadata(mComprStream, &offload_stream.offload_mdata) != 0)
ALOGE("%s(), compress_set_gapless_metadata() error= %s", __FUNCTION__, compress_get_error(mComprStream));
mHardwareResourceManager->startOutputDevice(mStreamAttributeSource->output_devices, mStreamAttributeTarget.sample_rate);
offload_stream.tmpBuffer = (void*)malloc(mComprConfig.fragment_size);
mWritebytes = mComprConfig.fragment_size;
list_init(&offload_stream.offload_cmd_list);
int ret = pthread_mutex_init(&offload_stream.offload_mutex, NULL);
if (ret != 0)
{
ALOGE("%s, Failed to initialize Mutex!", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
ret = pthread_cond_init(&offload_stream.offload_cond, NULL);
if (ret != 0)
{
ALOGE("%s, Failed to initialize Cond!", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
threadExit = false;
ret = pthread_create(&offload_stream.offload_pthread, NULL, &offload_threadloop, this);
if (ret != 0)
{
ALOGE("%s() create thread OffloadWrite fail!!", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
usleep(1 * 1000);
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
