bool LADPlayBuffer::LADPlayBufferInit(LADPlayer *playPointer,
uint32 idx,
uint32 sampleRate,
uint32 chNum,
int32 format,
uint32 targetSR)
{
//keep the format
mFormat = format;
//keep LADPlayer pointer and idx to buffer table
pLadPlayer = playPointer;
mIdx = idx;
LOGD("LADPlayBufferInit sampleRate=%d ,ch=%d, format=%d",sampleRate,chNum,format);
//set blisrc
uint32 srcBufLen;
BLI_GetMemSize( sampleRate, chNum, targetSR, LAD_CHANNEL_NUM, &srcBufLen);
pSrcBuf = new int8[srcBufLen];
pSrcHdl = BLI_Open( sampleRate, chNum, targetSR, LAD_CHANNEL_NUM, pSrcBuf);
if ( !pSrcHdl ) return false;
//buffer pointer setting
mBuf.pBufBase = new int8[LAD_PLAY_BUFFER_LEN];
mBuf.pRead = mBuf.pBufBase;
mBuf.pWrite = mBuf.pBufBase;
mBuf.bufLen = LAD_PLAY_BUFFER_LEN;
LOGD("pBufBase: %p, pRead: %p, pWrite: %p, bufLen:%d \n",mBuf.pBufBase,mBuf.pRead,mBuf.pWrite,mBuf.bufLen);
//initial mutex for buffer access
#if defined(PC_EMULATION)
InitializeCriticalSection(&bMutex);
#else
int ret;
ret = pthread_mutex_init(&bMutex, NULL);
if ( ret != 0 ) { return -1; }
#endif
return true;
}
void AVB_Open()
{
kal_uint32 uWorkingBuffer;
kal_trace( TRACE_GROUP_SCO, L1AUDIO_AVB_OPEN, AVB.uState);
ASSERT(AVB.uState == AVB_STATE_IDLE);
ASSERT(AVB.uSampleRate);
PcmSink_StartBT(PCMSINK_BT_SCO);
mhdlHookPcmRut();
BLI_GetMemSize(AVB.uSampleRate, AVB.uChannelNumber, AVB_SAMPLE_RATE, AVB_CHANNEL_NUM, &uWorkingBuffer);
uWorkingBuffer = (uWorkingBuffer+3) & ~3;
AVB.pBuffer = (kal_uint16 *)audio_alloc_mem_cacheable(uWorkingBuffer + AVB_DSP_BUFFER_SIZE + AVB_SRC_BUFFER_SIZE);
ASSERT(AVB.pBuffer);
AVB.pBLISRC = BLI_Open(AVB.uSampleRate, AVB.uChannelNumber, AVB_SAMPLE_RATE, AVB_CHANNEL_NUM, ((kal_int8*)AVB.pBuffer)+AVB_DSP_BUFFER_SIZE, audio_alloc_ext_mem_cacheable_simplified);
ASSERT(AVB.pBLISRC);
AVB.PCM.rb_base = ((kal_int8*)AVB.pBuffer)+AVB_DSP_BUFFER_SIZE+uWorkingBuffer;
AVB.PCM.rb_size = AVB_SRC_BUFFER_SIZE;
AVB.PCM.write = 0;
AVB.PCM.read = 0;
AVB.uAudID = L1Audio_GetAudioID();
L1Audio_SetFlag( AVB.uAudID );
L1Audio_SetEventHandler( AVB.uAudID, AVB_Process );
AVB.uState = AVB_STATE_RUNING;
L1Audio_HookHisrHandler( D2C_AUDIO_VIA_8KBT_ID, AVB_HISR, 0);
}
bool AudioYusuStreamIn::RecOpen()
{
bool ret = true;
m_no_data_count = 0;
ALOGD("+RecOpen, mSampleRateModem:%d mSampleRate = %d",mSampleRateModem,mSampleRate);
// this will set Mic gain base on current routing
if(mSetSphEnh && mHw->SetMicGain(mInputSource) == false){
ALOGD("RecOpen SetMicGain -- IN_CALL MODE");
}
if(mSetSphEnh == false){
// VOICE_RECOGNITION, must disable sph enhancement
// Remember to turn on again if we turn off it.
int Device= mHw->Audio_Match_Input_device(mDevice,true); // turn on this deivce
ALOGD("+RecOpen, mSetSphEnh=false mDevice = %x",Device);
if(Device == android_audio_legacy::AudioSystem::DEVICE_IN_WIRED_HEADSET){
mHw->mVolumeController->SetLadMicGain(Voice_Rec_Mic_Headset);
}
else{
mHw->mVolumeController->SetLadMicGain(Voice_Rec_Mic_Handset);
}
//pLad->LAD_Set_Speech_Enhancement(mSetSphEnh);
pLad->LAD_SetSpeechMode((ENUM_Speech_Mode)SPH_MODE_BT_CORDLESS);
pLad->LAD_SetInputSource (LADIN_Microphone1);
pLad->LAD_SetOutputDevice (LADOUT_SPEAKER2);
}
//Get Wake Lock for phone call recording
acquire_wake_lock(PARTIAL_WAKE_LOCK, MD_RECORD_WAKELOCK_NAME);
//Set Modem Sampling rate
int32 mode;
mHw->getMode(&mode);
if ( mode == android_audio_legacy::AudioSystem::MODE_IN_CALL ) {
mSampleRateModem = pLad->LAD_RecordSR()/*8000*/;
}
else if ( mSampleRate <= 8000 ) {
mSampleRateModem = 8000;
}
else {
mSampleRateModem = 16000;
}
//Modem Sampling rate convert to index
int32 srIdx;
if ( mSampleRateModem == 8000 )
srIdx = Record_8k;
else if ( mSampleRateModem == 16000 )
srIdx = Record_16k;
else {
YAD_LOGE("RecOpen modem SR error, use 8k Hz \n");
srIdx = Record_8k;
}
// call this function to decide set whick mode to modem.
bool bStereoRecOn = false;
SetHdrecordingMode(mode,&bStereoRecOn);
// get real ch num for BLI
uint8 mRealChNum;
#if defined(MTK_AUDIO_HD_REC_SUPPORT)
if ((mHw->GetVmFlag() == true && mode == android_audio_legacy::AudioSystem::MODE_IN_CALL ) ||
(mHw->META_Get_DualMic_Test_Mode()==TRUE) ||
(mHw->GetCtm4WayFlag() == true && mode == android_audio_legacy::AudioSystem::MODE_IN_CALL ) ||
(mode == android_audio_legacy::AudioSystem::MODE_IN_CALL)){
mRealChNum = 1;
}
else {
mRealChNum = (bStereoRecOn == true) ? 2 : 1;
ALOGD(" mRealChNum = %d bStereoRecOn = %d",mRealChNum,bStereoRecOn);
}
#else
mRealChNum = 1;
#endif
ALOGD("RecOpen Sampling Rate : %d, Modem Sample Rate : %d \n", mSampleRate, mSampleRateModem);
//If modem sampling rate different from sampling rate, SRC required
if ( (mSampleRateModem != mSampleRate) || mChNum != 1 && mFormat != android_audio_legacy::AudioSystem::VM_FMT ) {
ALOGD("mSampleRateModem=%d, mRealChNum=%d, mSampleRate=%d need src pSrcHdl = %p mChNum = %d",
mSampleRateModem,mRealChNum,mSampleRate,pSrcHdl,mChNum);
uint32 srcBufLen;
if(pSrcHdl == NULL){
BLI_GetMemSize( mSampleRateModem, mRealChNum, mSampleRate, mChNum, &srcBufLen);
pSrcBuf = new int8[srcBufLen];
pSrcHdl = BLI_Open( mSampleRateModem, mRealChNum, mSampleRate, mChNum, pSrcBuf);
}
else{
ALOGD("pSrcHdl exit = %p",pSrcHdl);
}
if ( !pSrcHdl ) return false;
}
// Add delay 10ms for Recording. Because record process is on-off-on (very quickly. about 2~8 tick counts),
// modem side DSP has control flow problem. The recorded sound is bad.
// So just add delay to work around this issue.
// Modem side will also find out the root cause. If modem side fix this issue, remove this work around.
usleep(10*1000);
if( true == CanRecordFM(mode) )
{
pLad->LAD_SetInputSource(LADIN_FM_Radio);
ALOGD("RecOpen, LAD_SetInputSource, LADIN_FM_Radio");
}
else
{
// pLad->LAD_SetInputSource(LADIN_Microphone1);
// ALOGD("RecOpen, LAD_SetInputSource, LADIN_Microphone1");
}
//reset read and write pointer of internal buffer
inBuf.pRead = inBuf.pBufBase;
inBuf.pWrite = inBuf.pBufBase;
ALOGD("SetRecDropFrameCount(%d)", m_DropFrameCount);
pLad->LAD_SetRecDropFrameCount(m_DropFrameCount);
//Start recording
if ( mHw->GetVmFlag() == true && mode == android_audio_legacy::AudioSystem::MODE_IN_CALL ){
ret &= pLad->LAD_OpenNormalRecPath(LADDATA_VM, 0);
ALOGD("-RecOpen, LADDATA_VM");
}
#if defined(MTK_DUAL_MIC_SUPPORT)||defined(MTK_AUDIO_HD_REC_SUPPORT)
else if(mHw->META_Get_DualMic_Test_Mode()==TRUE){
pLad->LAD_SetInputSource(LADIN_DualAnalogMic);
// set rec level
uint32 ladLevel = 0xffffff - (MAX_MIC_GAIN_DB - mHw->mDualMicTool_micGain) * 2; // in 0.5dB
ALOGD("set dual mic rec level level Rec: 0x%x, ladlevel: 0x%x",mHw->mDualMicTool_micGain,ladLevel);
pLad->LAD_SetMicrophoneVolume(ladLevel);
// start record
ret &= pLad->LAD_OpenNormalRecPath(LADDATA_DUAL_MIC_VM, 0);
ALOGD("-RecOpen, LADDATA_DUAL_MIC");
}
#endif
else if ( mHw->GetCtm4WayFlag() == true && mode == android_audio_legacy::AudioSystem::MODE_IN_CALL ){
ret &= pLad->LAD_OpenNormalRecPath(LADDATA_CTM_4WAY, 0);
ALOGD("-RecOpen, LADDATA_CTM_4WAY");
}
#if !defined(MTK_AUDIO_HD_REC_SUPPORT)
else{
ret &= pLad->LAD_OpenNormalRecPath(LADDATA_PCM, srIdx);
}
#else //MTK_AUDIO_HD_REC_SUPPORT = yes
else if ( mode == android_audio_legacy::AudioSystem::MODE_IN_CALL ){
