status_t AudioALSAFMController::setFmEnable(const bool enable, const audio_devices_t output_device)
{
// Lock to Protect HW Registers & AudioMode
// TODO(Harvey): get stream manager lock here?
AudioAutoTimeoutLock _l(mLock);
ALOGD("+%s(), mFmEnable = %d => enable = %d", __FUNCTION__, mFmEnable, enable);
// Check Current Status
if (enable == mFmEnable)
{
ALOGW("-%s(), enable == mFmEnable, return.", __FUNCTION__);
return INVALID_OPERATION;
}
// Update Enable Status
mFmEnable = enable;
// get current device
ALOGD("%s(), output_device = 0x%x", __FUNCTION__, output_device);
AudioALSASampleRateController *pAudioALSASampleRateController = AudioALSASampleRateController::getInstance();
if (mFmEnable == true) // Open
{
#if 0 // local print only
ALOGD("IsFMMergeInterfaceSupported = %d", WCNChipController::GetInstance()->IsFMMergeInterfaceSupported());
ALOGD("IsFmChipPadSelConnSys = %d", WCNChipController::GetInstance()->IsFmChipPadSelConnSys());
ALOGD("IsFmChipUseSlaveMode = %d", WCNChipController::GetInstance()->IsFmChipUseSlaveMode());
ALOGD("GetFmChipSamplingRate = %d", WCNChipController::GetInstance()->GetFmChipSamplingRate());
ALOGD("IsBTMergeInterfaceSupported = %d", WCNChipController::GetInstance()->IsBTMergeInterfaceSupported());
ALOGD("BTChipHWInterface = %d", WCNChipController::GetInstance()->BTChipHWInterface());
ALOGD("BTUseCVSDRemoval = %d", WCNChipController::GetInstance()->BTUseCVSDRemoval());
ALOGD("BTChipSamplingRate = %d", WCNChipController::GetInstance()->BTChipSamplingRate());
ALOGD("BTChipSamplingRateNumber = %d", WCNChipController::GetInstance()->BTChipSamplingRateNumber());
ALOGD("BTChipSyncFormat = %d", WCNChipController::GetInstance()->BTChipSyncFormat());
ALOGD("BTChipSyncLength = %d", WCNChipController::GetInstance()->BTChipSyncLength());
ALOGD("BTChipSecurityHiLo = %d", WCNChipController::GetInstance()->BTChipSecurityHiLo());
ALOGD("GetBTCurrentSamplingRateNumber = %d", WCNChipController::GetInstance()->GetBTCurrentSamplingRateNumber());
#endif
// set default 44100 Hz
pAudioALSASampleRateController->setPrimaryStreamOutSampleRate(44100);
pAudioALSASampleRateController->setScenarioStatus(PLAYBACK_SCENARIO_FM);
WCNChipController::GetInstance()->SetFmChipSampleRate(getFmDownlinkSamplingRate());
// Set Audio Digital/Analog HW Register
if (checkFmNeedUseDirectConnectionMode(output_device) == true)
{
setFmDirectConnection(true, true);
mHardwareResourceManager->startOutputDevice(output_device, getFmDownlinkSamplingRate());
}
else
{
mIsFmDirectConnectionMode = false;
}
// Set Direct/Indirect Mode to FMAudioPlayer
doDeviceChangeCallback();
}
else // Close
{
// Disable Audio Digital/Analog HW Register
if (mIsFmDirectConnectionMode == true)
{
mHardwareResourceManager->stopOutputDevice();
}
setFmDirectConnection(false, true);
// reset FM playback status
pAudioALSASampleRateController->resetScenarioStatus(PLAYBACK_SCENARIO_FM);
}
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;
}