diff options
Diffstat (limited to 'libaudio/AudioHardwareALSA.cpp')
| -rwxr-xr-x[-rw-r--r--] | libaudio/AudioHardwareALSA.cpp | 1013 |
1 files changed, 321 insertions, 692 deletions
diff --git a/libaudio/AudioHardwareALSA.cpp b/libaudio/AudioHardwareALSA.cpp index 52d0418..f1b23f8 100644..100755 --- a/libaudio/AudioHardwareALSA.cpp +++ b/libaudio/AudioHardwareALSA.cpp @@ -34,13 +34,10 @@ #include <alsa/asoundlib.h> #include "AudioHardwareALSA.h" -#define READ_FRAME_SIZE 2080 -#define READ_FRAME_SIZE_STANDARD 4160 +// #define READ_FRAME_SIZE 2080 +// #define READ_FRAME_SIZE_STANDARD 4160 -#if defined SEC_IPC -// sangsu fix : headers for IPC -#include <telephony/ril.h> -#endif +#include <dlfcn.h> #define SND_MIXER_VOL_RANGE_MIN (0) #define SND_MIXER_VOL_RANGE_MAX (100) @@ -51,21 +48,6 @@ if (strlen(x) + strlen(y) < ALSA_NAME_MAX) \ strcat(x, y); - -// If you want to dump PCM data, activate this feature -//#define PCM_INPUT_DUMP -//#define PCM_OUTPUT_DUMP - -#ifdef PCM_INPUT_DUMP -#define PCM_INPUT_DUMP_PATH "/data/Read_PCM_Dump.dat" -FILE *fpInput = NULL ; -#endif - -#ifdef PCM_OUTPUT_DUMP -#define PCM_OUTPUT_DUMP_PATH "/data/Write_PCM_Dump.dat" -FILE *fpOutput = NULL ; -#endif - extern "C" { extern int ffs(int i); @@ -90,20 +72,11 @@ extern "C" namespace android { -#if 0 -typedef AudioSystem::audio_routes audio_routes; -#define ROUTE_ALL AudioSystem::ROUTE_ALL -#define ROUTE_EARPIECE AudioSystem::ROUTE_EARPIECE -#define ROUTE_SPEAKER AudioSystem::ROUTE_SPEAKER -#define ROUTE_BLUETOOTH_SCO AudioSystem::ROUTE_BLUETOOTH_SCO -#define ROUTE_HEADSET AudioSystem::ROUTE_HEADSET -#define ROUTE_BLUETOOTH_A2DP AudioSystem::ROUTE_BLUETOOTH_A2DP -#elif defined SEC_SWP_SOUND typedef AudioSystem::audio_devices audio_routes; #define ROUTE_ALL AudioSystem::DEVICE_OUT_ALL #define ROUTE_EARPIECE AudioSystem::DEVICE_OUT_EARPIECE #define ROUTE_HEADSET AudioSystem::DEVICE_OUT_WIRED_HEADSET -#define ROUTE_HEADPHONE AudioSystem::DEVICE_OUT_WIRED_HEADPHONE +#define ROUTE_HEADPHONE AudioSystem::DEVICE_OUT_WIRED_HEADPHONE #define ROUTE_SPEAKER AudioSystem::DEVICE_OUT_SPEAKER #define ROUTE_BLUETOOTH_SCO AudioSystem::DEVICE_OUT_BLUETOOTH_SCO #define ROUTE_BLUETOOTH_SCO_HEADSET AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET @@ -111,15 +84,7 @@ typedef AudioSystem::audio_devices audio_routes; #define ROUTE_BLUETOOTH_A2DP AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP #define ROUTE_BLUETOOTH_A2DP_HEADPHONES AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES #define ROUTE_BLUETOOTH_A2DP_SPEAKER AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER -#else -typedef AudioSystem::audio_devices audio_routes; -#define ROUTE_ALL AudioSystem::DEVICE_OUT_ALL -#define ROUTE_EARPIECE AudioSystem::DEVICE_OUT_EARPIECE -#define ROUTE_SPEAKER AudioSystem::DEVICE_OUT_SPEAKER -#define ROUTE_BLUETOOTH_SCO AudioSystem::DEVICE_OUT_BLUETOOTH_SCO -#define ROUTE_HEADSET AudioSystem::DEVICE_OUT_WIRED_HEADSET -#define ROUTE_BLUETOOTH_A2DP AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP -#endif + // ---------------------------------------------------------------------------- @@ -148,46 +113,36 @@ static void ALSAErrorHandler(const char *file, /* The following table(s) need to match in order of the route bits */ -#if defined SEC_SWP_SOUND -static const char *deviceSuffix[] = { - // output devices - /* ROUTE_EARPIECE */ "_Earpiece", - /* ROUTE_SPEAKER */ "_Speaker", - /* ROUTE_HEADSET */ "_Headset", - /* ROUTE_HEADPHONE */ "_Headset", - /* ROUTE_BLUETOOTH_SCO */ "_Bluetooth", - /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth", - /* ROUTE_BLUETOOTH_SCO_CARKIT */ "_Bluetooth", //"_Bluetooth_Carkit" - /* ROUTE_BLUETOOTH_A2DP */ "_Bluetooth", //"_Bluetooth-A2DP" - /* ROUTE_BLUETOOTH_A2DP_HEADPHONES */ "_Bluetooth", //"_Bluetooth-A2DP_HeadPhone" - /* ROUTE_BLUETOOTH_A2DP_SPEAKER */ "_Bluetooth", // "_Bluetooth-A2DP_Speaker" - /* ROUTE_AUX_DIGITAL */ "_AuxDigital", - /* ROUTE_TV_OUT */ "_TvOut", - /* ROUTE_AUX_DIGITAL */ "_ExtraDockSpeaker", - /* ROUTE_NULL */ "_Null", - /* ROUTE_NULL */ "_Null", - /* ROUTE_DEFAULT */ "_OutDefault", - - // input devices - /* ROUTE_COMMUNICATION */ "_Communication", - /* ROUTE_AMBIENT */ "_Ambient", - /* ROUTE_BUILTIN_MIC */ "_Speaker", - /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth", - /* ROUTE_WIRED_HEADSET */ "_Headset", - /* ROUTE_AUX_DIGITAL */ "_AuxDigital", - /* ROUTE_VOICE_CALL */ "_VoiceCall", - /* ROUTE_BACK_MIC */ "_BackMic", - /* ROUTE_IN_DEFAULT */ "_InDefault", -}; -#else static const char *deviceSuffix[] = { + // output devices /* ROUTE_EARPIECE */ "_Earpiece", /* ROUTE_SPEAKER */ "_Speaker", - /* ROUTE_BLUETOOTH_SCO */ "_Bluetooth", /* ROUTE_HEADSET */ "_Headset", - /* ROUTE_BLUETOOTH_A2DP */ "_Bluetooth-A2DP", + /* ROUTE_HEADPHONE */ "_Headset", + /* ROUTE_BLUETOOTH_SCO */ "_Bluetooth", + /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth", + /* ROUTE_BLUETOOTH_SCO_CARKIT */ "_Bluetooth", //"_Bluetooth_Carkit" + /* ROUTE_BLUETOOTH_A2DP */ "_Bluetooth", //"_Bluetooth-A2DP" + /* ROUTE_BLUETOOTH_A2DP_HEADPHONES */ "_Bluetooth", //"_Bluetooth-A2DP_HeadPhone" + /* ROUTE_BLUETOOTH_A2DP_SPEAKER */ "_Bluetooth", // "_Bluetooth-A2DP_Speaker" + /* ROUTE_AUX_DIGITAL */ "_AuxDigital", + /* ROUTE_TV_OUT */ "_TvOut", + /* ROUTE_AUX_DIGITAL */ "_ExtraDockSpeaker", + /* ROUTE_NULL */ "_Null", + /* ROUTE_NULL */ "_Null", + /* ROUTE_DEFAULT */ "_OutDefault", + + // input devices + /* ROUTE_COMMUNICATION */ "_Communication", + /* ROUTE_AMBIENT */ "_Ambient", + /* ROUTE_BUILTIN_MIC */ "_Speaker", + /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth", + /* ROUTE_WIRED_HEADSET */ "_Headset", + /* ROUTE_AUX_DIGITAL */ "_AuxDigital", + /* ROUTE_VOICE_CALL */ "_VoiceCall", + /* ROUTE_BACK_MIC */ "_BackMic", + /* ROUTE_IN_DEFAULT */ "_InDefault", }; -#endif static const int deviceSuffixLen = (sizeof(deviceSuffix) / sizeof(char *)); @@ -251,23 +206,14 @@ const uint32_t AudioHardwareALSA::inputSamplingRates[] = { AudioHardwareALSA::AudioHardwareALSA() : mOutput(0), - mInput(0) -#if defined SEC_IPC - ,mIPC(0) //for IPC -#endif -#if defined TURN_ON_DEVICE_ONLY_USE - ,mActivatedInputDevice(false) -#endif -#if defined SYNCHRONIZE_CP - ,mActivatedCP(false) -#endif - + mInput(0), + mSecRilLibHandle(NULL), + mRilClient(0) { snd_lib_error_set_handler(&ALSAErrorHandler); mMixer = new ALSAMixer; -#if defined SEC_IPC - mIPC = new AudioHardwareIPC; // IPC init -#endif + + loadRILD(); } AudioHardwareALSA::~AudioHardwareALSA() @@ -275,11 +221,66 @@ AudioHardwareALSA::~AudioHardwareALSA() if (mOutput) delete mOutput; if (mInput) delete mInput; if (mMixer) delete mMixer; -#if defined SEC_IPC - if (mIPC) delete mIPC; // for IPC -#endif + + if (mSecRilLibHandle) { + if (disconnectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) + LOGE("Disconnect_RILD() error"); + + if (closeClientRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) + LOGE("CloseClient_RILD() error"); + + mRilClient = 0; + + dlclose(mSecRilLibHandle); + mSecRilLibHandle = NULL; + } } + +void AudioHardwareALSA::loadRILD(void) +{ + mSecRilLibHandle = dlopen("libsecril-client.so", RTLD_NOW); + + if (mSecRilLibHandle) { + LOGV("libsecril-client.so is loaded"); + + openClientRILD = (HRilClient (*)(void)) + dlsym(mSecRilLibHandle, "OpenClient_RILD"); + disconnectRILD = (int (*)(HRilClient)) + dlsym(mSecRilLibHandle, "Disconnect_RILD"); + closeClientRILD = (int (*)(HRilClient)) + dlsym(mSecRilLibHandle, "CloseClient_RILD"); + isConnectedRILD = (int (*)(HRilClient)) + dlsym(mSecRilLibHandle, "isConnected_RILD"); + connectRILD = (int (*)(HRilClient)) + dlsym(mSecRilLibHandle, "Connect_RILD"); + setCallVolume = (int (*)(HRilClient, SoundType, int)) + dlsym(mSecRilLibHandle, "SetCallVolume"); + setCallAudioPath = (int (*)(HRilClient, AudioPath)) + dlsym(mSecRilLibHandle, "SetCallAudioPath"); + + if (!openClientRILD || !disconnectRILD || !closeClientRILD || + !isConnectedRILD || !connectRILD || + !setCallVolume || !setCallAudioPath) { + LOGE("Can't load all functions from libsecril-client.so"); + + dlclose(mSecRilLibHandle); + mSecRilLibHandle = NULL; + } else { + mRilClient = openClientRILD(); + if (!mRilClient) { + LOGE("OpenClient_RILD() error"); + + dlclose(mSecRilLibHandle); + mSecRilLibHandle = NULL; + } + } + } else { + LOGE("Can't load libsecril-client.so"); + } +} + + status_t AudioHardwareALSA::initCheck() { if (mMixer && mMixer->isValid()) @@ -296,48 +297,70 @@ status_t AudioHardwareALSA::standby() return NO_ERROR; } + +status_t AudioHardwareALSA::connectRILDIfRequired(void) +{ + if (!mSecRilLibHandle) { + LOGE("connectIfRequired() lib is not loaded"); + return INVALID_OPERATION; + } + + if (isConnectedRILD(mRilClient) == 0) { + return OK; + } + + if (connectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) { + LOGE("Connect_RILD() error"); + return INVALID_OPERATION; + } + + return OK; +} + + status_t AudioHardwareALSA::setVoiceVolume(float volume) { - LOGI("### setVoiceVolume"); -#if defined SEC_IPC + LOGI("### setVoiceVolume"); + // sangsu fix : transmic volume level IPC to modem - if (AudioSystem::MODE_IN_CALL == mMode) - { - uint32_t routes = mRoutes[mMode]; - - LOGI("### route(%d) call volume(%f)", routes, volume); - switch (routes){ - case AudioSystem::ROUTE_EARPIECE: - case AudioSystem::ROUTE_HEADPHONE: // Use receive path with 3 pole headset. - LOGI("### earpiece call volume"); - mIPC->transmitVolumeIPC(OEM_SOUND_TYPE_VOICE, volume); - break; - - case AudioSystem::ROUTE_SPEAKER: - LOGI("### speaker call volume"); - mIPC->transmitVolumeIPC(OEM_SOUND_TYPE_SPEAKER, volume); - break; - - case AudioSystem::ROUTE_BLUETOOTH_SCO: - case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: - case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - case AudioSystem::ROUTE_BLUETOOTH_A2DP: - LOGI("### bluetooth call volume"); - mIPC->transmitVolumeIPC(OEM_SOUND_TYPE_BTVOICE, volume); - break; - - case AudioSystem::ROUTE_HEADSET: - LOGI("### headset call volume"); - mIPC->transmitVolumeIPC(OEM_SOUND_TYPE_HEADSET, volume); - break; - - default: - LOGE("### Call volume setting error!!!0x%08x \n", routes); - break; - } - } - // sangsu fix end -#endif + if ( (AudioSystem::MODE_IN_CALL == mMode) && (mSecRilLibHandle) && + (connectRILDIfRequired() == OK) ) { + + uint32_t routes = mRoutes[mMode]; + int int_volume = (int)(volume * 5); + + LOGI("### route(%d) call volume(%f)", routes, volume); + switch (routes) { + case AudioSystem::ROUTE_EARPIECE: + case AudioSystem::ROUTE_HEADPHONE: // Use receive path with 3 pole headset. + LOGI("### earpiece call volume"); + setCallVolume(mRilClient, SOUND_TYPE_VOICE, int_volume); + break; + + case AudioSystem::ROUTE_SPEAKER: + LOGI("### speaker call volume"); + setCallVolume(mRilClient, SOUND_TYPE_SPEAKER, int_volume); + break; + + case AudioSystem::ROUTE_BLUETOOTH_SCO: + case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: + case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: + case AudioSystem::ROUTE_BLUETOOTH_A2DP: + LOGI("### bluetooth call volume"); + setCallVolume(mRilClient, SOUND_TYPE_BTVOICE, int_volume); + break; + + case AudioSystem::ROUTE_HEADSET: + LOGI("### headset call volume"); + setCallVolume(mRilClient, SOUND_TYPE_HEADSET, int_volume); + break; + + default: + LOGE("### Call volume setting error!!!0x%08x \n", routes); + break; + } + } + // sangsu fix end // The voice volume is used by the VOICE_CALL audio stream. if (mMixer) @@ -354,34 +377,23 @@ status_t AudioHardwareALSA::setMasterVolume(float volume) return INVALID_OPERATION; } -#if defined TURN_ON_DEVICE_ONLY_USE int AudioHardwareALSA::setMicStatus(int on) { - LOGI("[%s], on=%d", __func__, on); - ALSAControl *mALSAControl = new ALSAControl(); - status_t ret = mALSAControl->set("Mic Status", on); - delete mALSAControl; - return NO_ERROR; + LOGI("[%s], on=%d", __func__, on); + ALSAControl *mALSAControl = new ALSAControl(); + status_t ret = mALSAControl->set("Mic Status", on); + delete mALSAControl; + return NO_ERROR; } -#endif -#if 0 -AudioStreamOut * -AudioHardwareALSA::AudioStreamOut* openOutputStream( - int format=0, - int channelCount=0, - uint32_t sampleRate=0, - status_t *status=0) -#else AudioStreamOut * AudioHardwareALSA::openOutputStream( - uint32_t devices, - int *format, - uint32_t *channels, - uint32_t *sampleRate, - status_t *status) -#endif + uint32_t devices, + int *format, + uint32_t *channels, + uint32_t *sampleRate, + status_t *status) { AutoMutex lock(mLock); @@ -397,25 +409,13 @@ AudioHardwareALSA::openOutputStream( *status = out->set(format, channels, sampleRate); -#ifdef PCM_OUTPUT_DUMP - if(fpOutput == NULL) - { - fpOutput = fopen(PCM_OUTPUT_DUMP_PATH, "w"); - if (fpOutput == NULL) - LOGE("fpOutput File Open Error!!"); - } -#endif - if (*status == NO_ERROR) { mOutput = out; // Some information is expected to be available immediately after // the device is open. /* Tushar - Sets the current device output here - we may set device here */ - //uint32_t routes = mRoutes[mMode]; - //mOutput->setDevice(mMode, routes); - LOGI("%s] Setting ALSA device.", __func__); + LOGI("%s] Setting ALSA device.", __func__); mOutput->setDevice(mMode, devices, PLAYBACK); /* tushar - Enable all devices as of now */ - mOutput->setWakeLock(); } else { delete out; @@ -427,15 +427,10 @@ AudioHardwareALSA::openOutputStream( void AudioHardwareALSA::closeOutputStream(AudioStreamOut* out) { - /* TODO:Tushar: May lead to segmentation fault - check*/ - //delete out; + /* TODO:Tushar: May lead to segmentation fault - check*/ + //delete out; AutoMutex lock(mLock); -#ifdef PCM_OUTPUT_DUMP - fclose(fpOutput); - fpOutput = NULL; -#endif - if (mOutput == 0 || mOutput != out) { LOGW("Attempt to close invalid output stream"); } @@ -446,14 +441,7 @@ AudioHardwareALSA::closeOutputStream(AudioStreamOut* out) } -#if 0 -AudioStreamIn * -AudioHardwareALSA::openInputStream(int format, - int channelCount, - uint32_t sampleRate, - status_t *status, - AudioSystem::audio_in_acoustics acoustics) -#else + AudioStreamIn* AudioHardwareALSA::openInputStream( uint32_t devices, @@ -462,7 +450,6 @@ AudioHardwareALSA::openInputStream( uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) -#endif { AutoMutex lock(mLock); @@ -479,23 +466,8 @@ AudioHardwareALSA::openInputStream( mInput = in; // Some information is expected to be available immediately after // the device is open. - //uint32_t routes = mRoutes[mMode]; - //mInput->setDevice(mMode, routes); mInput->setDevice(mMode, devices, CAPTURE); /* Tushar - as per modified arch */ - mInput->setWakeLock(); -#if defined TURN_ON_DEVICE_ONLY_USE - mActivatedInputDevice = true; setMicStatus(1); -#endif -#ifdef PCM_INPUT_DUMP - if(fpInput == NULL) - { - fpInput = fopen(PCM_INPUT_DUMP_PATH, "w"); - if (fpInput == NULL) - LOGE("fpInput File Open Error!!"); - } -#endif - return mInput; } else { @@ -507,147 +479,81 @@ AudioHardwareALSA::openInputStream( void AudioHardwareALSA::closeInputStream(AudioStreamIn* in) { - /* TODO:Tushar: May lead to segmentation fault - check*/ - //delete in; - AutoMutex lock(mLock); + /* TODO:Tushar: May lead to segmentation fault - check*/ + //delete in; + AutoMutex lock(mLock); if (mInput == 0 || mInput != in) { LOGW("Attempt to close invalid input stream"); - } - else { + } else { delete mInput; mInput = 0; -#ifdef PCM_INPUT_DUMP - fclose(fpInput); - fpInput = NULL; -#endif -#if defined TURN_ON_DEVICE_ONLY_USE - mActivatedInputDevice = false; setMicStatus(0); -#endif - } } -#if defined SEC_SWP_SOUND + status_t AudioHardwareALSA::doRouting(uint32_t device) { - uint32_t routes; - status_t ret; - - AutoMutex lock(mLock); - int mode = mMode; // Prevent to changing mode on setup sequence. - - if (mOutput) { - routes = device; - //routes = 0; /* Tushar - temp implementation */ - - // Setup sound path for CP clocking - -#if defined SEC_IPC - - if (AudioSystem::MODE_IN_CALL == mode) - { - - LOGI("### incall mode route (%d)", routes); - switch(routes){ - case AudioSystem::ROUTE_EARPIECE: - LOGI("### incall mode earpiece route"); - mIPC->transmitAudioPathIPC(OEM_SOUND_AUDIO_PATH_HANDSET); - break; - - case AudioSystem::ROUTE_SPEAKER: - LOGI("### incall mode speaker route"); - mIPC->transmitAudioPathIPC(OEM_SOUND_AUDIO_PATH_SPEAKER); - break; - - case AudioSystem::ROUTE_BLUETOOTH_SCO: - case AudioSystem::ROUTE_BLUETOOTH_SCO_HEADSET: - case AudioSystem::ROUTE_BLUETOOTH_SCO_CARKIT: -#if defined BT_NR_EC_ONOFF - if(mBluetoothECOff) - { - LOGI("### incall mode bluetooth EC OFF route"); - mIPC->transmitAudioPathIPC(OEM_SOUND_AUDIO_PATH_BT_NSEC_OFF); - } - else - { -#endif - LOGI("### incall mode bluetooth route"); - mIPC->transmitAudioPathIPC(OEM_SOUND_AUDIO_PATH_BLUETOOTH); -#if defined BT_NR_EC_ONOFF - } -#endif - break; - - case AudioSystem::ROUTE_HEADSET : - case AudioSystem::ROUTE_HEADPHONE : - LOGI("### incall mode headset route"); - mIPC->transmitAudioPathIPC(OEM_SOUND_AUDIO_PATH_HEADSET); - break; - - case AudioSystem::ROUTE_BLUETOOTH_A2DP: - LOGI("### incall mode bluetooth route"); - mIPC->transmitAudioPathIPC(OEM_SOUND_AUDIO_PATH_BLUETOOTH); - break; - - default: - LOGE("### incall mode Error!! route = [%d]", routes); - break; - } - } -#endif// end of #if defined SEC_IPC - - ret = mOutput->setDevice(mode, routes, PLAYBACK); - -#if defined SEC_IPC - if (AudioSystem::MODE_IN_CALL == mode) - { -#if defined SYNCHRONIZE_CP - if(!mActivatedCP) - { - mIPC->transmitClock_IPC(OEM_SOUND_CLOCK_START); - mActivatedCP = true; - } -#endif - } - - if (AudioSystem::MODE_NORMAL== mode) // Call stop. - { -#if defined SYNCHRONIZE_CP - if(mActivatedCP) - mActivatedCP = false; -#endif - - } -#endif // end of #if defined SEC_IPC - -#ifndef SYNCHRONIZE_CP -// ret = mOutput->setDevice(mode, routes, PLAYBACK); -#endif - return ret; - } - - return NO_INIT; -} - -#else -/** This function is no more used */ -status_t AudioHardwareALSA::doRouting() -{ - uint32_t routes; + status_t ret; + AutoMutex lock(mLock); + int mode = mMode; // Prevent to changing mode on setup sequence. + + LOGV("doRouting (%d)", device); - LOGD("Inside AudioHardwareALSA::doRouting \n"); if (mOutput) { - //routes = mRoutes[mMode]; - routes = 0; /* Tushar - temp implementation */ - return mOutput->setDevice(mMode, routes, PLAYBACK); + //device = 0; /* Tushar - temp implementation */ + + // Setup sound path for CP clocking + if ( (AudioSystem::MODE_IN_CALL == mode) && (mSecRilLibHandle) && + (connectRILDIfRequired() == OK) ) { + + LOGI("### incall mode route (%d)", device); + + switch(device){ + case AudioSystem::ROUTE_EARPIECE: + LOGI("### incall mode earpiece route"); + setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_HANDSET); + break; + + case AudioSystem::ROUTE_SPEAKER: + LOGI("### incall mode speaker route"); + setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_SPEAKER); + break; + + case AudioSystem::ROUTE_BLUETOOTH_SCO: + case AudioSystem::ROUTE_BLUETOOTH_SCO_HEADSET: + case AudioSystem::ROUTE_BLUETOOTH_SCO_CARKIT: + LOGI("### incall mode bluetooth route"); + setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH); + break; + + case AudioSystem::ROUTE_HEADSET : + case AudioSystem::ROUTE_HEADPHONE : + LOGI("### incall mode headset route"); + setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_HEADSET); + break; + + case AudioSystem::ROUTE_BLUETOOTH_A2DP: + LOGI("### incall mode bluetooth route"); + setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH); + break; + + default: + LOGE("### incall mode Error!! route = [%d]", device); + break; + } + } + + ret = mOutput->setDevice(mode, device, PLAYBACK); + + return ret; } - return NO_INIT; + return NO_INIT; } -#endif + status_t AudioHardwareALSA::setMicMute(bool state) { @@ -721,7 +627,6 @@ ALSAStreamOps::ALSAStreamOps() : mHandle(0), mHardwareParams(0), mSoftwareParams(0), - mMode(-1), mDevice(0) { if (snd_pcm_hw_params_malloc(&mHardwareParams) < 0) { @@ -751,44 +656,44 @@ status_t ALSAStreamOps::set(int *pformat, uint32_t *pchannels, uint32_t *prate) { - int lformat = pformat ? *pformat : 0; - unsigned int lchannels = pchannels ? *pchannels : 0; - unsigned int lrate = prate ? *prate : 0; + int lformat = pformat ? *pformat : 0; + unsigned int lchannels = pchannels ? *pchannels : 0; + unsigned int lrate = prate ? *prate : 0; - LOGD("ALSAStreamOps - input - format = %d, channels = %d, rate = %d\n", lformat, lchannels, lrate); - LOGD("ALSAStreamOps - default - format = %d, channelCount = %d, rate = %d\n", mDefaults->format, mDefaults->channelCount, mDefaults->sampleRate); + LOGD("ALSAStreamOps - input - format = %d, channels = %d, rate = %d\n", lformat, lchannels, lrate); + LOGD("ALSAStreamOps - default - format = %d, channelCount = %d, rate = %d\n", mDefaults->format, mDefaults->channelCount, mDefaults->sampleRate); - if(lformat == 0) lformat = getAndroidFormat(mDefaults->format);//format(); - if(lchannels == 0) lchannels = getAndroidChannels(mDefaults->channelCount);// channelCount(); - if(lrate == 0) lrate = mDefaults->sampleRate; + if (lformat == 0) lformat = getAndroidFormat(mDefaults->format);//format(); + if (lchannels == 0) lchannels = getAndroidChannels(mDefaults->channelCount);// channelCount(); + if (lrate == 0) lrate = mDefaults->sampleRate; - if((lformat != getAndroidFormat(mDefaults->format)) || - (lchannels != getAndroidChannels(mDefaults->channelCount))) { - if(pformat) *pformat = getAndroidFormat(mDefaults->format); - if(pchannels) *pchannels = getAndroidChannels(mDefaults->channelCount); - return BAD_VALUE; - } - if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) { - if (lrate != mDefaults->sampleRate) { - if(prate) *prate = mDefaults->sampleRate; + if ( (lformat != getAndroidFormat(mDefaults->format)) || + (lchannels != getAndroidChannels(mDefaults->channelCount)) ) { + if (pformat) *pformat = getAndroidFormat(mDefaults->format); + if (pchannels) *pchannels = getAndroidChannels(mDefaults->channelCount); + return BAD_VALUE; + } + if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) { + if (lrate != mDefaults->sampleRate) { + if (prate) *prate = mDefaults->sampleRate; return BAD_VALUE; - } - } else { - mDefaults->smpRateShift = AudioHardwareALSA::checkInputSampleRate(lrate); - // audioFlinger will reopen the input stream with correct smp rate - if (AudioHardwareALSA::inputSamplingRates[mDefaults->smpRateShift] != lrate) { - if(prate) *prate = AudioHardwareALSA::inputSamplingRates[mDefaults->smpRateShift]; + } + } else { + mDefaults->smpRateShift = AudioHardwareALSA::checkInputSampleRate(lrate); + // audioFlinger will reopen the input stream with correct smp rate + if (AudioHardwareALSA::inputSamplingRates[mDefaults->smpRateShift] != lrate) { + if(prate) *prate = AudioHardwareALSA::inputSamplingRates[mDefaults->smpRateShift]; return BAD_VALUE; - } - } + } + } mDefaults->sampleRate = lrate; - if(pformat) *pformat = getAndroidFormat(mDefaults->format); - if(pchannels) *pchannels = getAndroidChannels(mDefaults->channelCount); - if(prate) *prate = mDefaults->sampleRate; + if(pformat) *pformat = getAndroidFormat(mDefaults->format); + if(pchannels) *pchannels = getAndroidChannels(mDefaults->channelCount); + if(prate) *prate = mDefaults->sampleRate; - return NO_ERROR; + return NO_ERROR; } @@ -985,7 +890,7 @@ status_t ALSAStreamOps::open(int mode, uint32_t device) int err; - LOGI("Try to open ALSA %s device %s", stream, devName); + LOGI("Try to open ALSA %s device %s", stream, devName); for(;;) { // The PCM stream is opened in blocking mode, per ALSA defaults. The @@ -1003,7 +908,7 @@ status_t ALSAStreamOps::open(int mode, uint32_t device) if (err < 0) { // None of the Android defined audio devices exist. Open a generic one. - devName = "hw:00,1"; // 090507 SMDKC110 Froyo + devName = "hw:00,1"; // 090507 SMDKC110 Froyo err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0); if (err < 0) { @@ -1013,7 +918,6 @@ status_t ALSAStreamOps::open(int mode, uint32_t device) } } - mMode = mode; mDevice = device; LOGI("Initialized ALSA %s device %s", stream, devName); @@ -1026,9 +930,8 @@ void ALSAStreamOps::close() mHandle = NULL; if (handle) { + snd_pcm_drain(handle); snd_pcm_close(handle); - mMode = -1; - mDevice = 0; } } @@ -1052,7 +955,7 @@ status_t ALSAStreamOps::setSoftwareParams() // Configure ALSA to start the transfer when the buffer is almost full. snd_pcm_get_params(mHandle, &bufferSize, &periodSize); - LOGE("bufferSize %d, periodSize %d\n", bufferSize, periodSize); + LOGE("bufferSize %d, periodSize %d\n", (int)bufferSize, (int)periodSize); if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) { // For playback, configure ALSA to start the transfer when the @@ -1221,11 +1124,6 @@ status_t ALSAStreamOps::setDevice(int mode, uint32_t device, uint audio_mode) // FIXME:: always use default sampling rate sampleRate(DEFAULT_SAMPLE_RATE); - // Disable hardware resampling. - status = setHardwareResample(false); - if (status != NO_ERROR) - return status; - snd_pcm_uframes_t bufferSize = mDefaults->bufferSize; snd_pcm_uframes_t periodSize = mDefaults->periodSize; period_val = bufferSize/periodSize; @@ -1265,12 +1163,12 @@ status_t ALSAStreamOps::setDevice(int mode, uint32_t device, uint audio_mode) // err = snd_pcm_hw_params_set_buffer_time_near (mHandle, mHardwareParams, // &latency, NULL); // if(audio_mode == PLAYBACK) { -// period_val = PERIODS_PLAYBACK; -// if(snd_pcm_hw_params_set_periods(mHandle, mHardwareParams, period_val, 0) < 0) -// LOGE("Fail to set period size %d for playback", period_val); +// period_val = PERIODS_PLAYBACK; +// if(snd_pcm_hw_params_set_periods(mHandle, mHardwareParams, period_val, 0) < 0) +// LOGE("Fail to set period size %d for playback", period_val); // } // else -// period_val = PERIODS_CAPTURE; +// period_val = PERIODS_CAPTURE; // // if (err < 0) { // LOGD("snd_pcm_hw_params_set_buffer_time_near() failed: %s", snd_strerror(err)); @@ -1413,8 +1311,8 @@ status_t AudioStreamOutALSA::setVolume(float left, float right) if (! mParent->mMixer || ! mDevice) return NO_INIT; - /** Tushar - Need to decide on the volume value - * that we pass onto the mixer. */ + /** Tushar - Need to decide on the volume value + * that we pass onto the mixer. */ return mParent->mMixer->setVolume (mDevice, (left + right)/2); } @@ -1429,7 +1327,6 @@ status_t AudioStreamOutALSA::setVolume(float volume) /* New Arch */ status_t AudioStreamOutALSA::setParameters(const String8& keyValuePairs) { -#if defined SLSI_S5PC110 AudioParameter param = AudioParameter(keyValuePairs); status_t status = NO_ERROR; int device; @@ -1438,11 +1335,7 @@ status_t AudioStreamOutALSA::setParameters(const String8& keyValuePairs) if (param.getInt(String8(AudioParameter::keyRouting), device) == NO_ERROR) { - mDevice = device; - -// if (mParent->mInput) mParent->mInput->mDevice = device; - mParent->mRoutes[mParent->mMode] = mDevice; - mParent->doRouting(mDevice); + mParent->doRouting(device); param.remove(String8(AudioParameter::keyRouting)); } @@ -1457,23 +1350,11 @@ status_t AudioStreamOutALSA::setParameters(const String8& keyValuePairs) status = BAD_VALUE; } return status; -#else - /* TODO: Implement as per new arch */ - - LOGD("AudioStreamOutAlsa::setParameters... %s \n\n",keyValuePairs.string()); - if (! mParent->mOutput )//|| ! mMode) - return NO_INIT; +} - int device = keyValuePairs.string()[keyValuePairs.length()-1] - 48 -1 ; //easy conversion frm ascii to int and then to required number - LOGV("\n\n-------->> ALSA SET PARAMS device %d \n\n",(1<<device)); - mParent->mOutput->setDevice(mMode, 1<<device, PLAYBACK); - return NO_ERROR; -#endif -} String8 AudioStreamOutALSA::getParameters(const String8& keys) { -#if defined SLSI_S5PC110 AudioParameter param = AudioParameter(keys); String8 value; String8 key = String8(AudioParameter::keyRouting); @@ -1484,20 +1365,17 @@ String8 AudioStreamOutALSA::getParameters(const String8& keys) LOGD("AudioStreamOutALSA::getParameters() %s", param.toString().string()); return param.toString(); -#else - /* TODO: Implement as per new arch */ - return keys; -#endif } + status_t AudioStreamOutALSA::getRenderPosition(uint32_t *dspFrames) { - //TODO: enable when supported by driver - return INVALID_OPERATION; + //TODO: enable when supported by driver + return INVALID_OPERATION; } -#if 1 // Fix for underrun error + ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes) { snd_pcm_sframes_t n; @@ -1508,17 +1386,11 @@ ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes) if (!mPowerLock) { LOGD("Calling setDevice from write @..%d.\n",__LINE__); - ALSAStreamOps::setDevice(mMode, mDevice, PLAYBACK); + ALSAStreamOps::setDevice(mParent->mode(), mDevice, PLAYBACK); acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock"); mPowerLock = true; } - // if (isStandby()) - // return 0; -#ifdef PCM_OUTPUT_DUMP - fwrite(buffer, bytes, 1, fpOutput); - LOGD("Output PCM dumped!!"); -#endif do { // write correct number of bytes per attempt n = snd_pcm_writei(mHandle, (char *) buffer + sent, snd_pcm_bytes_to_frames(mHandle, bytes @@ -1527,7 +1399,7 @@ ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes) LOGD("Calling setDevice.. pcm_write returned error @..%d.\n",__LINE__); // Somehow the stream is in a bad state. The driver probably // has a bug and snd_pcm_recover() doesn't seem to handle this. - ALSAStreamOps::setDevice(mMode, mDevice, PLAYBACK); + ALSAStreamOps::setDevice(mParent->mode(), mDevice, PLAYBACK); } else if (n < 0) { if (mHandle) { // snd_pcm_recover() will return 0 if successful in recovering from @@ -1543,41 +1415,6 @@ ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes) //LOGI("Request Bytes=%d, Actual Written=%d",bytes,sent); return snd_pcm_frames_to_bytes(mHandle, sent); } -#else -ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes) -{ - snd_pcm_sframes_t n; - status_t err; - - AutoMutex lock(mLock); -#if 0 - if (isStandby()) - return 0; -#endif - if (!mPowerLock) { - acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioLock"); - ALSAStreamOps::setDevice(mMode, mDevice,PLAYBACK); - mPowerLock = true; - } - if (!mHandle) { - return -1; - } - n = snd_pcm_writei(mHandle, - buffer, - snd_pcm_bytes_to_frames(mHandle, bytes)); - if (n < 0 && mHandle) { - // snd_pcm_recover() will return 0 if successful in recovering from - // an error, or -errno if the error was unrecoverable. - //device driver sometimes does not recover -vladi - n = snd_pcm_recover(mHandle, n, 0); - if(n < 0) //if recover fails - ALSAStreamOps::setDevice(mMode, mDevice, PLAYBACK); - } - - return static_cast<ssize_t>(n); -} -#endif - status_t AudioStreamOutALSA::dump(int fd, const Vector<String16>& args) @@ -1589,41 +1426,28 @@ status_t AudioStreamOutALSA::setDevice(int mode, uint32_t newDevice, uint32_t au { AutoMutex lock(mLock); - return ALSAStreamOps::setDevice(mode, newDevice, audio_mode); + LOGV("AudioStreamOutALSA::setDevice(mode %d, newDevice %x, audio_mode %d), mDevice %x", + mode, newDevice, audio_mode, mDevice); + if (newDevice != mDevice) { + mParent->mRoutes[mParent->mMode] = newDevice; + return ALSAStreamOps::setDevice(mode, newDevice, audio_mode); + } + return NO_ERROR; } status_t AudioStreamOutALSA::standby() { AutoMutex lock(mLock); LOGD("Inside AudioStreamOutALSA::standby\n"); - if (mHandle) - snd_pcm_drain ( mHandle); + + ALSAStreamOps::close(); if (mPowerLock) { - if (!mParent->mActivatedInputDevice) { // Let PCM device alive on activating input stream. - snd_pcm_close( mHandle); - mHandle = NULL; -#if 1 // Fix for underrun error - release_wake_lock("AudioOutLock"); -#else - release_wake_lock ("AudioLock"); -#endif - mPowerLock = false; - } + release_wake_lock("AudioOutLock"); + mPowerLock = false; } - // close(); //Don't call this as this function will reset the mode also return NO_ERROR; } -bool AudioStreamOutALSA::isStandby() -{ - return (!mPowerLock); -} - -void AudioStreamOutALSA::setWakeLock() -{ - acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock"); - mPowerLock = true; -} #define USEC_TO_MSEC(x) ((x + 999) / 1000) @@ -1656,7 +1480,7 @@ AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent) : AudioStreamInALSA::~AudioStreamInALSA() { - standby_l(); + standby(); mParent->mInput = NULL; } @@ -1678,17 +1502,10 @@ ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes) if (!mPowerLock) { acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock"); -#ifdef PCM_INPUT_DUMP - fwrite(buffer, readBytes, 1, fpInput); - LOGD("Input PCM dumped!!"); -#endif -#if defined TURN_ON_DEVICE_ONLY_USE - mParent->mActivatedInputDevice = true; // setMicStatus(1); -#endif - LOGD("Calling setDevice from read@..%d.\n",__LINE__); - ALSAStreamOps::setDevice(mMode, mDevice,CAPTURE); + LOGD("Calling setDevice from read@..%d.\n",__LINE__); + ALSAStreamOps::setDevice(mParent->mode(), mDevice,CAPTURE); mPowerLock = true; } if (!mHandle) { @@ -1696,8 +1513,8 @@ ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes) } // FIXME: only support reads of exactly bufferSize() for now - if (bytes != bufferSize()) { - LOGW("AudioStreamInALSA::read bad read size %d expected %d", bytes, bufferSize()); + if (bytes != (ssize_t)bufferSize()) { + LOGW("AudioStreamInALSA::read bad read size %d expected %d", (int)bytes, bufferSize()); return -1; } @@ -1708,9 +1525,9 @@ ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes) (uint8_t *)mBuffer, frames << shift); if (n < 0) { - LOGD("AudioStreamInALSA::read error %d", n); + LOGD("AudioStreamInALSA::read error %d", (int)n); n = snd_pcm_recover(mHandle, n, 0); - LOGD("AudioStreamInALSA::snd_pcm_recover error %d", n); + LOGD("AudioStreamInALSA::snd_pcm_recover error %d", (int)n); if (n) return static_cast<ssize_t> (n); } else { @@ -1725,19 +1542,15 @@ ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes) // downsampler in AudioFlinger (SR in < 2 * SR out) int16_t *out = (int16_t *)buffer; if (mDefaults->channelCount == 1) { - for (size_t i = 0; i < n; i++) { + for (ssize_t i = 0; i < n; i++) { out[i] = mBuffer[i << shift]; } } else { - for (size_t i = 0; i < n; i++) { + for (ssize_t i = 0; i < n; i++) { out[i] = mBuffer[i << shift]; out[i + 1] = mBuffer[(i << shift) + 1]; } } -#ifdef PCM_INPUT_DUMP - fwrite(buffer, bytes, 1, fpInput); - LOGD("Input PCM dumped!!"); -#endif return snd_pcm_frames_to_bytes(mHandle, n); } @@ -1757,86 +1570,52 @@ status_t AudioStreamInALSA::setDevice(int mode, uint32_t newDevice, uint32_t aud status_t AudioStreamInALSA::standby() { AutoMutex lock(mLock); + LOGD("Entering AudioStreamInALSA::standby\n"); - return standby_l(); -} + ALSAStreamOps::close(); -status_t AudioStreamInALSA::standby_l() -{ - LOGD("Entering AudioStreamInALSA::standby\n"); if (mPowerLock) { - mParent->mActivatedInputDevice = false; - snd_pcm_close(mHandle); - LOGD("AudioStreamInALSA::standby snd_pcm_close()"); - mHandle = NULL; - release_wake_lock ("AudioInLock"); - mPowerLock = false; + release_wake_lock ("AudioInLock"); + mPowerLock = false; } return NO_ERROR; } -void AudioStreamInALSA::setWakeLock() -{ - acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock"); - mPowerLock = true; -} - /* New Arch */ status_t AudioStreamInALSA::setParameters(const String8& keyValuePairs) { -#if defined SLSI_S5PC110 - AudioParameter param = AudioParameter(keyValuePairs); - String8 key = String8(AudioParameter::keyRouting); - status_t status = NO_ERROR; - int device; - LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string()); - - if (param.getInt(key, device) == NO_ERROR) { - mDevice = device; - if(mPowerLock && mDevice != 0) - setDevice(mMode, mDevice, CAPTURE); - param.remove(key); - } - - if (param.size()) { - status = BAD_VALUE; - } - return status; -#else - /* TODO: Implement as per new arch */ - - if (! mParent->mInput )//|| ! mMode) - return NO_INIT; + AudioParameter param = AudioParameter(keyValuePairs); + String8 key = String8(AudioParameter::keyRouting); + status_t status = NO_ERROR; + int device; + LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string()); - // yman.seo use setDevice temp. - int device = keyValuePairs.string()[keyValuePairs.length()-1] - 48 -1 ; //easy conversion frm ascii to int and then to required number - LOGD("\n\n-------->> ALSA AudioStreamIn SET PARAMS device %d \n\n",(1<<device)); - if(mParent->mActivatedInputDevice )//Recording stopped with Alarm ,then don't call setDevice() of record, just return - return mParent->mInput->setDevice(mMode, 1<<device, CAPTURE); + if (param.getInt(key, device) == NO_ERROR) { + if(mHandle != NULL && device != 0) + setDevice(mParent->mode(), device, CAPTURE); + param.remove(key); + } - return NO_ERROR; -// yman.seo return NO_ERROR; -#endif + if (param.size()) { + status = BAD_VALUE; + } + return status; } + String8 AudioStreamInALSA::getParameters(const String8& keys) { -#if defined SLSI_S5PC110 - AudioParameter param = AudioParameter(keys); - String8 value; - String8 key = String8(AudioParameter::keyRouting); + AudioParameter param = AudioParameter(keys); + String8 value; + String8 key = String8(AudioParameter::keyRouting); - if (param.get(key, value) == NO_ERROR) { - param.addInt(key, (int)mDevice); - } + if (param.get(key, value) == NO_ERROR) { + param.addInt(key, (int)mDevice); + } - LOGD("AudioStreamInALSA::getParameters() %s", param.toString().string()); - return param.toString(); -#else - /* TODO: Implement as per new arch */ - return keys; -#endif + LOGD("AudioStreamInALSA::getParameters() %s", param.toString().string()); + return param.toString(); } @@ -2158,7 +1937,7 @@ status_t ALSAMixer::getCaptureMuteState(uint32_t device, bool *state) status_t ALSAMixer::setPlaybackMuteState(uint32_t device, bool state) { - LOGE("\n set playback mute device %d, state %d \n", device,state); + LOGE("\n set playback mute device %d, state %d \n", device,state); for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++) if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) { @@ -2319,155 +2098,5 @@ status_t ALSAControl::set(const char *name, unsigned int value, int index) return (ret < 0) ? BAD_VALUE : NO_ERROR; } -// ---------------------------------------------------------------------------- - -#if defined SEC_IPC -AudioHardwareIPC::AudioHardwareIPC() : - mClient(NULL) -{ -LOGD("### %s", __func__); - int err = 0; - - mClient = OpenClient_RILD(); - if (mClient == NULL){ - LOGE("[*] OpenClient_RILD() error\n"); - err = 1; - } - - if (RegisterRequestCompleteHandler(mClient, RIL_REQUEST_OEM_HOOK_RAW, - onRawReqComplete) != RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] RegisterRequestCompleteHandler() error\n"); - err = 1; - } - - if (RegisterUnsolicitedHandler(mClient, 11004, onUnsol) != - RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] RegisterUnsolicitedHandler() error\n"); - err = 1; - } - - if (!err) LOGD("Success Initializing IPC"); - else LOGE("Failed Initializing IPC"); -} - -AudioHardwareIPC::~AudioHardwareIPC() -{ - LOGD("### %s", __func__); - if (RegisterRequestCompleteHandler(mClient, RIL_REQUEST_OEM_HOOK_RAW, NULL) != RIL_CLIENT_ERR_SUCCESS) - LOGE("RegisterRequestCompleteHandler(NULL) error\n"); - - if (RegisterUnsolicitedHandler(mClient, 11004, NULL) != RIL_CLIENT_ERR_SUCCESS) - LOGE("RegisterUnsolicitedHandler(NULL) error\n"); - - if (Disconnect_RILD(mClient) != RIL_CLIENT_ERR_SUCCESS) - LOGE("[*] Disconnect_RILD() error\n"); - - if (CloseClient_RILD(mClient) != RIL_CLIENT_ERR_SUCCESS) - LOGE("[*] CloseClient_RILD() error\n"); - mClient = NULL; -} - -status_t AudioHardwareIPC::transmitVolumeIPC(uint32_t type, float volume) -{ - int ret = 0; - uint32_t level = (uint32_t)(volume * 5); - - if (isConnected_RILD(mClient) == 0) { - if (Connect_RILD(mClient) != RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] Connect_RILD() error\n"); - return INVALID_OPERATION; - } - } - - memset(data, 0, 100); - data[0] = OEM_FUNCTION_ID_SOUND; - data[1] = OEM_SOUND_SET_VOLUME_CTRL; - data[2] = 0x00; // data length - data[3] = 0x06; // data length - data[4] = type; // volume type - data[5] = level; // volume level - - ret = InvokeOemRequestHookRaw(mClient, data, 6); //sizeof(data)); - if (ret != RIL_CLIENT_ERR_AGAIN && ret != RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] InvokeOemRequestHookRaw() error ret = %d\n", ret); - return INVALID_OPERATION; - } - return NO_ERROR; -} - -status_t AudioHardwareIPC::transmitAudioPathIPC(uint32_t path) -{ - int ret = 0; - - LOGI("### %s %d ", __func__, path); - if (isConnected_RILD(mClient) == 0) { - if (Connect_RILD(mClient) != RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] Connect_RILD() error\n"); - return INVALID_OPERATION; - } - } - - memset(data, 0, 100); - data[0] = OEM_FUNCTION_ID_SOUND; - data[1] = OEM_SOUND_SET_AUDIO_PATH_CTRL; - data[2] = 0x00; // data length - data[3] = 0x05; // data length - data[4] = path; // audio path - - ret = InvokeOemRequestHookRaw(mClient, data, 5); //sizeof(data)); - if (ret != RIL_CLIENT_ERR_AGAIN && ret != RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] InvokeOemRequestHookRaw() error ret = %d\n", ret); - return INVALID_OPERATION; - } - return NO_ERROR; -} - - -#if defined SYNCHRONIZE_CP -status_t AudioHardwareIPC::transmitClock_IPC(uint32_t condition) -{ - int ret = 0; - - LOGV("### %s %d ", __func__, condition); - - if (isConnected_RILD(mClient) == 0) { - if (Connect_RILD(mClient) != RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] Connect_RILD() error\n"); - return INVALID_OPERATION; - } - } - - memset(data, 0, 100); - data[0] = OEM_FUNCTION_ID_SOUND; - data[1] = OEM_SOUND_SET_CLOCK_CTRL; - data[2] = 0x00; // data length - data[3] = 0x05; // data length - data[4] = condition; - - ret = InvokeOemRequestHookRaw(mClient, data, 5); //sizeof(data)); - - if (ret != RIL_CLIENT_ERR_AGAIN && ret != RIL_CLIENT_ERR_SUCCESS){ - LOGE("[*] InvokeOemRequestHookRaw() error ret = %d\n", ret); - return INVALID_OPERATION; - } - - return NO_ERROR; -} -#endif - -static int onRawReqComplete(HRilClient client, const void *data, size_t datalen) -{ - LOGV("[*] %s(): datalen(%d)\n", __FUNCTION__, datalen); - return 0; -} -static int onUnsol(HRilClient client, const void *data, size_t datalen) -{ - int a; - a = ((int *)data)[0]; - LOGV("%s(): a(%d)\n", __FUNCTION__, a); - - return 0; -} -#endif //SEC_IPC }; // namespace android |