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-rwxr-xr-xlibaudio/AudioHardwareALSA.cpp258
-rwxr-xr-xlibaudio/AudioHardwareALSA.h38
2 files changed, 183 insertions, 113 deletions
diff --git a/libaudio/AudioHardwareALSA.cpp b/libaudio/AudioHardwareALSA.cpp
index f1b23f8..98ad027 100755
--- a/libaudio/AudioHardwareALSA.cpp
+++ b/libaudio/AudioHardwareALSA.cpp
@@ -208,7 +208,8 @@ AudioHardwareALSA::AudioHardwareALSA() :
mOutput(0),
mInput(0),
mSecRilLibHandle(NULL),
- mRilClient(0)
+ mRilClient(0),
+ mVrModeEnabled(false)
{
snd_lib_error_set_handler(&ALSAErrorHandler);
mMixer = new ALSAMixer;
@@ -289,14 +290,6 @@ status_t AudioHardwareALSA::initCheck()
return NO_INIT;
}
-status_t AudioHardwareALSA::standby()
-{
- if (mOutput)
- return mOutput->standby();
-
- return NO_ERROR;
-}
-
status_t AudioHardwareALSA::connectRILDIfRequired(void)
{
@@ -322,11 +315,15 @@ status_t AudioHardwareALSA::setVoiceVolume(float volume)
{
LOGI("### setVoiceVolume");
+ AutoMutex lock(mLock);
// sangsu fix : transmic volume level IPC to modem
if ( (AudioSystem::MODE_IN_CALL == mMode) && (mSecRilLibHandle) &&
(connectRILDIfRequired() == OK) ) {
- uint32_t routes = mRoutes[mMode];
+ uint32_t routes = AudioSystem::ROUTE_EARPIECE;
+ if (mOutput != NULL) {
+ routes = mOutput->device();
+ }
int int_volume = (int)(volume * 5);
LOGI("### route(%d) call volume(%f)", routes, volume);
@@ -395,32 +392,40 @@ AudioHardwareALSA::openOutputStream(
uint32_t *sampleRate,
status_t *status)
{
- AutoMutex lock(mLock);
+ AudioStreamOutALSA *out = NULL;
+ status_t ret = NO_ERROR;
+ {
+ AutoMutex lock(mLock);
- // only one output stream allowed
- if (mOutput) {
- *status = ALREADY_EXISTS;
- return 0;
- }
+ // only one output stream allowed
+ if (mOutput) {
+ ret = ALREADY_EXISTS;
+ goto exit;
+ }
- LOGV("[[[[[[[[\n%s - format = %d, channels = %d, sampleRate = %d, devices = %d]]]]]]]]\n", __func__, *format, *channels, *sampleRate,devices);
+ LOGV("[[[[[[[[\n%s - format = %d, channels = %d, sampleRate = %d, devices = %d]]]]]]]]\n", __func__, *format, *channels, *sampleRate,devices);
- AudioStreamOutALSA *out = new AudioStreamOutALSA(this);
+ out = new AudioStreamOutALSA(this);
- *status = out->set(format, channels, sampleRate);
+ ret = out->set(format, channels, sampleRate);
- if (*status == NO_ERROR) {
- mOutput = out;
+ if (ret == NO_ERROR) {
+ mOutput = out;
+ }
+ }
+exit:
+ if (ret == NO_ERROR) {
// 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 */
LOGI("%s] Setting ALSA device.", __func__);
mOutput->setDevice(mMode, devices, PLAYBACK); /* tushar - Enable all devices as of now */
- }
- else {
+ } else if (out) {
delete out;
}
-
+ if (status) {
+ *status = ret;
+ }
return mOutput;
}
@@ -429,16 +434,16 @@ AudioHardwareALSA::closeOutputStream(AudioStreamOut* out)
{
/* TODO:Tushar: May lead to segmentation fault - check*/
//delete out;
- AutoMutex lock(mLock);
+ {
+ AutoMutex lock(mLock);
- if (mOutput == 0 || mOutput != out) {
- LOGW("Attempt to close invalid output stream");
- }
- else {
- delete mOutput;
+ if (mOutput == 0 || mOutput != out) {
+ LOGW("Attempt to close invalid output stream");
+ return;
+ }
mOutput = 0;
}
-
+ delete out;
}
@@ -451,28 +456,36 @@ AudioHardwareALSA::openInputStream(
status_t *status,
AudioSystem::audio_in_acoustics acoustics)
{
- AutoMutex lock(mLock);
+ AudioStreamInALSA *in = NULL;
+ status_t ret = NO_ERROR;
+ {
+ AutoMutex lock(mLock);
- // only one input stream allowed
- if (mInput) {
- *status = ALREADY_EXISTS;
- return 0;
- }
+ // only one input stream allowed
+ if (mInput) {
+ ret = ALREADY_EXISTS;
+ goto exit;
+ }
- AudioStreamInALSA *in = new AudioStreamInALSA(this);
+ in = new AudioStreamInALSA(this);
- *status = in->set(format, channels, sampleRate);
- if (*status == NO_ERROR) {
- mInput = in;
+ ret = in->set(format, channels, sampleRate);
+ if (ret == NO_ERROR) {
+ mInput = in;
+ }
+ }
+exit:
+ if (ret == NO_ERROR) {
// Some information is expected to be available immediately after
// the device is open.
mInput->setDevice(mMode, devices, CAPTURE); /* Tushar - as per modified arch */
setMicStatus(1);
- return mInput;
- }
- else {
+ } else if (in != NULL) {
delete in;
}
+ if (status) {
+ *status = ret;
+ }
return mInput;
}
@@ -481,29 +494,39 @@ AudioHardwareALSA::closeInputStream(AudioStreamIn* in)
{
/* TODO:Tushar: May lead to segmentation fault - check*/
//delete in;
- AutoMutex lock(mLock);
+ {
+ AutoMutex lock(mLock);
- if (mInput == 0 || mInput != in) {
- LOGW("Attempt to close invalid input stream");
- } else {
- delete mInput;
- mInput = 0;
- setMicStatus(0);
+ if (mInput == 0 || mInput != in) {
+ LOGW("Attempt to close invalid input stream");
+ return;
+ } else {
+ mInput = 0;
+ setMicStatus(0);
+ }
}
+ delete in;
}
-status_t AudioHardwareALSA::doRouting(uint32_t device)
+status_t AudioHardwareALSA::doRouting(uint32_t device, bool force)
{
- status_t ret;
-
AutoMutex lock(mLock);
+ return doRouting_l(device, force);
+}
+
+status_t AudioHardwareALSA::doRouting_l(uint32_t device, bool force)
+{
+ status_t ret;
int mode = mMode; // Prevent to changing mode on setup sequence.
- LOGV("doRouting (%d)", device);
+ LOGV("doRouting: device %x, force %d", device, force);
if (mOutput) {
//device = 0; /* Tushar - temp implementation */
+ if (device == AudioSystem::DEVICE_OUT_DEFAULT) {
+ device = mOutput->device();
+ }
// Setup sound path for CP clocking
if ( (AudioSystem::MODE_IN_CALL == mode) && (mSecRilLibHandle) &&
@@ -546,7 +569,7 @@ status_t AudioHardwareALSA::doRouting(uint32_t device)
}
}
- ret = mOutput->setDevice(mode, device, PLAYBACK);
+ ret = mOutput->setDevice(mode, device, PLAYBACK, force);
return ret;
}
@@ -597,14 +620,6 @@ size_t AudioHardwareALSA::getInputBufferSize(uint32_t sampleRate, int format, in
LOGV("getInputBufferSize() rate %d, shift %d, size %d", sampleRate, shift, size);
return size;
-//#if defined SEC_SWP_SOUND
-// if (sampleRate == 32000 || sampleRate == 44100 || sampleRate == 48000)
-// return READ_FRAME_SIZE_STANDARD;
-// else
-// return READ_FRAME_SIZE;
-//#else /* SEC_SWP_SOUND */
-// return 320;
-//#endif /* SEC_SWP_SOUND */
}
uint32_t AudioHardwareALSA::checkInputSampleRate(uint32_t sampleRate)
@@ -621,6 +636,53 @@ uint32_t AudioHardwareALSA::checkInputSampleRate(uint32_t sampleRate)
return i-1;
}
+status_t AudioHardwareALSA::setMode(int mode)
+{
+ AutoMutex lock(mLock);
+ int prevMode = mMode;
+ status_t status = AudioHardwareBase::setMode(mode);
+ LOGV("setMode() : new %d, old %d", mMode, prevMode);
+ if (status == NO_ERROR) {
+ // make sure that doAudioRouteOrMute() is called by doRouting()
+ // when entering or exiting in call mode even if the new device
+ // selected is the same as current one.
+ if ((prevMode != AudioSystem::MODE_IN_CALL) && (mMode == AudioSystem::MODE_IN_CALL)) {
+ LOGV("setMode() entering call");
+ doRouting_l(AudioSystem::DEVICE_OUT_DEFAULT, true);
+ setVoiceRecordGain_l(false);
+ }
+ if ((prevMode == AudioSystem::MODE_IN_CALL) && (mMode != AudioSystem::MODE_IN_CALL)) {
+ LOGV("setMode() exiting call");
+ doRouting_l(AudioSystem::DEVICE_OUT_DEFAULT, true);
+ if (mOutput != NULL && !mOutput->isActive()) {
+ mOutput->close();
+ }
+ }
+ }
+
+ return status;
+}
+
+int AudioHardwareALSA::setVoiceRecordGain(bool enable)
+{
+ AutoMutex lock(mLock);
+ return setVoiceRecordGain_l(enable);
+}
+
+int AudioHardwareALSA::setVoiceRecordGain_l(bool enable)
+{
+ LOGI("[%s], enable=%d", __func__, enable);
+ if (enable != mVrModeEnabled &&
+ !(enable && (mMode == AudioSystem::MODE_IN_CALL))) {
+ ALSAControl *alsaControl = new ALSAControl();
+ status_t ret = alsaControl->set("Codec Status", enable ? 5 : 4);
+ delete alsaControl;
+ mVrModeEnabled = enable;
+ }
+
+ return NO_ERROR;
+}
+
// ----------------------------------------------------------------------------
ALSAStreamOps::ALSAStreamOps() :
@@ -699,15 +761,6 @@ status_t ALSAStreamOps::set(int *pformat,
uint32_t ALSAStreamOps::sampleRate() const
{
-// unsigned int rate;
-// int err;
-//
-// if (! mHandle)
-// return NO_INIT;
-//
-// return snd_pcm_hw_params_get_rate(mHardwareParams, &rate, 0) < 0
-// ? 0 : static_cast<uint32_t>(rate);
-
return mDefaults->sampleRate;
}
@@ -790,9 +843,6 @@ int ALSAStreamOps::format() const
int pcmFormatBitWidth;
int audioSystemFormat;
-// if (!mHandle)
-// return -1;
-
if (snd_pcm_hw_params_get_format(mHardwareParams, &ALSAFormat) < 0) {
return -1;
}
@@ -868,8 +918,6 @@ status_t ALSAStreamOps::channelCount(int channelCount) {
if (!mHandle)
return NO_INIT;
- // if(channelCount == 1) channelCount = 2; //Kamat: This is a fix added to avoid audioflinger crash (current audio driver does not support mono). Please check and modify suitably later.
-
err = snd_pcm_hw_params_set_channels(mHandle, mHardwareParams, channelCount);
if (err < 0) {
LOGE("Unable to set channel count to %i: %s",
@@ -930,6 +978,7 @@ void ALSAStreamOps::close()
mHandle = NULL;
if (handle) {
+ LOGV("ALSAStreamOps::close()");
snd_pcm_drain(handle);
snd_pcm_close(handle);
}
@@ -1301,7 +1350,6 @@ AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent) :
AudioStreamOutALSA::~AudioStreamOutALSA()
{
standby();
- mParent->mOutput = NULL;
}
@@ -1339,12 +1387,6 @@ status_t AudioStreamOutALSA::setParameters(const String8& keyValuePairs)
param.remove(String8(AudioParameter::keyRouting));
}
- else if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR)
- {
- mParent->mOutput->mDefaults->sampleRate = value;
- mParent->doRouting(mDevice);
- param.remove(String8(AudioParameter::keySamplingRate));
- }
if (param.size()) {
status = BAD_VALUE;
@@ -1382,6 +1424,7 @@ ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
size_t sent = 0;
status_t err;
+ mParent->lock().lock();
AutoMutex lock(mLock);
if (!mPowerLock) {
@@ -1390,6 +1433,7 @@ ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock");
mPowerLock = true;
}
+ mParent->lock().unlock();
do {
// write correct number of bytes per attempt
@@ -1422,24 +1466,29 @@ status_t AudioStreamOutALSA::dump(int fd, const Vector<String16>& args)
return NO_ERROR;
}
-status_t AudioStreamOutALSA::setDevice(int mode, uint32_t newDevice, uint32_t audio_mode)
+status_t AudioStreamOutALSA::setDevice(int mode,
+ uint32_t newDevice,
+ uint32_t audio_mode,
+ bool force)
{
AutoMutex lock(mLock);
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;
+ if (newDevice != mDevice || force) {
return ALSAStreamOps::setDevice(mode, newDevice, audio_mode);
}
return NO_ERROR;
}
status_t AudioStreamOutALSA::standby() {
+ AutoMutex _l(mParent->lock());
AutoMutex lock(mLock);
LOGD("Inside AudioStreamOutALSA::standby\n");
- ALSAStreamOps::close();
+ if (mParent->mode() != AudioSystem::MODE_IN_CALL) {
+ ALSAStreamOps::close();
+ }
if (mPowerLock) {
release_wake_lock("AudioOutLock");
@@ -1481,7 +1530,6 @@ AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent) :
AudioStreamInALSA::~AudioStreamInALSA()
{
standby();
- mParent->mInput = NULL;
}
status_t AudioStreamInALSA::setGain(float gain)
@@ -1497,17 +1545,19 @@ ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes)
snd_pcm_sframes_t n;
status_t err;
+ mParent->lock().lock();
AutoMutex lock(mLock);
-
if (!mPowerLock) {
acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock");
// setMicStatus(1);
LOGD("Calling setDevice from read@..%d.\n",__LINE__);
- ALSAStreamOps::setDevice(mParent->mode(), mDevice,CAPTURE);
+ ALSAStreamOps::setDevice(mParent->mode(), mDevice, CAPTURE);
mPowerLock = true;
}
+ mParent->lock().unlock();
+
if (!mHandle) {
return -1;
}
@@ -1560,7 +1610,10 @@ status_t AudioStreamInALSA::dump(int fd, const Vector<String16>& args)
return NO_ERROR;
}
-status_t AudioStreamInALSA::setDevice(int mode, uint32_t newDevice, uint32_t audio_mode)
+status_t AudioStreamInALSA::setDevice(int mode,
+ uint32_t newDevice,
+ uint32_t audio_mode,
+ bool force)
{
AutoMutex lock(mLock);
@@ -1569,7 +1622,9 @@ status_t AudioStreamInALSA::setDevice(int mode, uint32_t newDevice, uint32_t aud
status_t AudioStreamInALSA::standby()
{
+ AutoMutex _l(mParent->lock());
AutoMutex lock(mLock);
+
LOGD("Entering AudioStreamInALSA::standby\n");
ALSAStreamOps::close();
@@ -1586,14 +1641,21 @@ status_t AudioStreamInALSA::standby()
status_t AudioStreamInALSA::setParameters(const String8& keyValuePairs)
{
AudioParameter param = AudioParameter(keyValuePairs);
- String8 key = String8(AudioParameter::keyRouting);
+ String8 key = String8("vr_mode");
status_t status = NO_ERROR;
- int device;
+ int value;
LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string());
- if (param.getInt(key, device) == NO_ERROR) {
- if(mHandle != NULL && device != 0)
- setDevice(mParent->mode(), device, CAPTURE);
+
+ if (param.getInt(key, value) == NO_ERROR) {
+ mParent->setVoiceRecordGain((value != 0));
+ param.remove(key);
+ }
+
+ key = String8(AudioParameter::keyRouting);
+ if (param.getInt(key, value) == NO_ERROR) {
+ if(mHandle != NULL && value != 0)
+ setDevice(mParent->mode(), value, CAPTURE);
param.remove(key);
}
diff --git a/libaudio/AudioHardwareALSA.h b/libaudio/AudioHardwareALSA.h
index f65aa68..22821ab 100755
--- a/libaudio/AudioHardwareALSA.h
+++ b/libaudio/AudioHardwareALSA.h
@@ -89,6 +89,10 @@ namespace android
class ALSAStreamOps
{
+ public:
+ uint32_t device() { return mDevice; }
+ void close();
+
protected:
friend class AudioStreamOutALSA;
friend class AudioStreamInALSA;
@@ -124,13 +128,12 @@ namespace android
uint32_t getAndroidChannels(int channelCount) const;
status_t open(int mode, uint32_t device);
- void close();
status_t setSoftwareParams();
status_t setPCMFormat(snd_pcm_format_t format);
status_t setHardwareResample(bool resample);
const char *streamName();
- virtual status_t setDevice(int mode, uint32_t device, uint32_t audio_mode);
+ status_t setDevice(int mode, uint32_t device, uint32_t audio_mode);
const char *deviceName(int mode, uint32_t device);
@@ -187,7 +190,8 @@ namespace android
virtual ssize_t write(const void *buffer, size_t bytes);
virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode);
+ status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode,
+ bool force = false);
virtual status_t setVolume(float left, float right); //Tushar: New arch
status_t setVolume(float volume);
@@ -198,7 +202,7 @@ namespace android
virtual String8 getParameters(const String8& keys);
virtual status_t getRenderPosition(uint32_t *dspFrames);
-
+ bool isActive() { return mPowerLock; }
private:
AudioHardwareALSA *mParent;
@@ -238,7 +242,8 @@ namespace android
virtual ssize_t read(void* buffer, ssize_t bytes);
virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode);
+ status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode,
+ bool force = false);
virtual status_t setGain(float gain);
@@ -249,6 +254,8 @@ namespace android
virtual unsigned int getInputFramesLost() const { return 0; }
+ bool isActive() { return mPowerLock; }
+
private:
AudioHardwareALSA *mParent;
bool mPowerLock;
@@ -267,15 +274,11 @@ namespace android
*/
virtual status_t initCheck();
- /**
- * put the audio hardware into standby mode to conserve power. Returns
- * status based on include/utils/Errors.h
- */
- virtual status_t standby();
-
/** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
virtual status_t setVoiceVolume(float volume);
+ virtual status_t setMode(int mode);
+
/**
* set the audio volume for all audio activities other than voice call.
* Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
@@ -312,6 +315,10 @@ namespace android
static const uint32_t inputSamplingRates[];
int mode() { return mMode; }
+ Mutex& lock() { return mLock; }
+
+ int setVoiceRecordGain(bool enable);
+ int setVoiceRecordGain_l(bool enable);
protected:
/**
@@ -320,7 +327,8 @@ namespace android
* doRouting when required. If the device has any special requirements these
* methods can be overriden.
*/
- virtual status_t doRouting(uint32_t device);
+ status_t doRouting(uint32_t device, bool force = false);
+ status_t doRouting_l(uint32_t device, bool force = false);
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -331,13 +339,12 @@ namespace android
AudioStreamOutALSA *mOutput;
AudioStreamInALSA *mInput;
- uint32_t mRoutes[AudioSystem::NUM_MODES];
private:
Mutex mLock;
- bool mActivatedInputDevice;
-
void *mSecRilLibHandle;
HRilClient mRilClient;
+ bool mVrModeEnabled;
+
HRilClient (*openClientRILD) (void);
int (*disconnectRILD) (HRilClient);
int (*closeClientRILD) (HRilClient);
@@ -348,6 +355,7 @@ namespace android
void loadRILD(void);
status_t connectRILDIfRequired(void);
+
};
}; // namespace android
generated by cgit v1.1 at 2024-12-23 13:47:17 +0000