diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index 3a7550e0adeebc4f88695f77284df981adef2f0a..7f310dd39cc134d167a25270fc32754506ab8743 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -92,11 +92,11 @@ AlsaLayer::AlsaLayer (ManagerImpl* manager)
{
_debug ("Audio: Build ALSA layer");
- _urgentRingBuffer.createReadPointer();
+ urgentRingBuffer_.createReadPointer();
- audioPlugin_ = AudioLayer::_manager->audioPreference.getPlugin();
+ audioPlugin_ = manager_->audioPreference.getPlugin();
- AudioLayer::_noisesuppressstate = true;
+ noiseSuppressState_ = true;
}
// Destructor
@@ -146,26 +146,26 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
else if ( (stream == SFL_PCM_PLAYBACK or stream == SFL_PCM_BOTH) and is_playback_open_)
closePlaybackStream ();
- _indexIn = indexIn;
- _indexOut = indexOut;
- _indexRing = indexRing;
+ indexIn_ = indexIn;
+ indexOut_ = indexOut;
+ indexRing_ = indexRing;
- _audioSampleRate = sampleRate;
- _frameSize = frameSize;
+ audioSampleRate_ = sampleRate;
+ frameSize_ = frameSize;
audioPlugin_ = plugin;
- _debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", _indexIn, _indexOut, _indexRing);
+ _debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", indexIn_, indexOut_, indexRing_);
_debugAlsa (" : alsa plugin=%s", audioPlugin_.c_str());
- _debugAlsa (" : nb channel in=%2d, out=%2d", _inChannel, _outChannel);
- _debugAlsa (" : sample rate=%5d, format=%s", _audioSampleRate, SFLDataFormatString);
+ _debugAlsa (" : nb channel in=%2d, out=%2d", inChannel_, outChannel_);
+ _debugAlsa (" : sample rate=%5d, format=%s", audioSampleRate_, SFLDataFormatString);
audioThread_ = NULL;
// use 1 sec buffer for resampling
- converter_ = new SamplerateConverter (_audioSampleRate);
- _dcblocker = new DcBlocker;
- _audiofilter = new AudioProcessing (_dcblocker);
+ converter_ = new SamplerateConverter (audioSampleRate_);
+ dcblocker_ = new DcBlocker;
+ audiofilter_ = new AudioProcessing (dcblocker_);
}
void
@@ -173,8 +173,8 @@ AlsaLayer::startStream (void)
{
_debug ("Audio: Start stream");
- if (_audiofilter)
- _audiofilter->resetAlgorithm();
+ if (audiofilter_)
+ audiofilter_->resetAlgorithm();
if (is_playback_running_ and is_capture_running_)
return;
@@ -184,18 +184,18 @@ AlsaLayer::startStream (void)
std::string pcmc;
if (audioPlugin_ == PCM_DMIX_DSNOOP) {
- pcmp = buildDeviceTopo (PCM_DMIX, _indexOut, 0);
- pcmr = buildDeviceTopo (PCM_DMIX, _indexRing, 0);
- pcmc = buildDeviceTopo (PCM_DSNOOP, _indexIn, 0);
+ pcmp = buildDeviceTopo (PCM_DMIX, indexOut_, 0);
+ pcmr = buildDeviceTopo (PCM_DMIX, indexRing_, 0);
+ pcmc = buildDeviceTopo (PCM_DSNOOP, indexIn_, 0);
} else {
- pcmp = buildDeviceTopo (audioPlugin_, _indexOut, 0);
- pcmr = buildDeviceTopo (audioPlugin_, _indexRing, 0);
- pcmc = buildDeviceTopo (audioPlugin_, _indexIn, 0);
+ pcmp = buildDeviceTopo (audioPlugin_, indexOut_, 0);
+ pcmr = buildDeviceTopo (audioPlugin_, indexRing_, 0);
+ pcmc = buildDeviceTopo (audioPlugin_, indexIn_, 0);
}
- _debug ("pcmp: %s, index %d", pcmp.c_str(), _indexOut);
- _debug ("pcmr: %s, index %d", pcmr.c_str(), _indexRing);
- _debug ("pcmc: %s, index %d", pcmc.c_str(), _indexIn);
+ _debug ("pcmp: %s, index %d", pcmp.c_str(), indexOut_);
+ _debug ("pcmr: %s, index %d", pcmr.c_str(), indexRing_);
+ _debug ("pcmc: %s, index %d", pcmc.c_str(), indexIn_);
if (not is_capture_open_)
open_device (pcmp, pcmc, pcmr, SFL_PCM_CAPTURE);
@@ -222,7 +222,7 @@ AlsaLayer::startStream (void)
}
}
- _isStarted = true;
+ isStarted_ = true;
}
void
@@ -230,7 +230,7 @@ AlsaLayer::stopStream (void)
{
_debug ("Audio: Stop stream");
- _isStarted = false;
+ isStarted_ = false;
try {
/* Stop the audio thread first */
@@ -256,12 +256,6 @@ AlsaLayer::stopStream (void)
flushMain ();
}
-void AlsaLayer::setNoiseSuppressState (bool state)
-{
- // if a stream already opened
- AudioLayer::_noisesuppressstate = state;
-}
-
//////////////////////////////////////////////////////////////////////////////////////////////
///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////////////////////
@@ -430,17 +424,17 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
/* Set sample rate. If we can't set to the desired exact value, we set to the nearest acceptable */
int dir = 0;
- unsigned int exact_ivalue = _audioSampleRate;
+ unsigned int exact_ivalue = audioSampleRate_;
if ((err = snd_pcm_hw_params_set_rate_near (pcm_handle, hwparams, &exact_ivalue, &dir) < 0)) {
_debugAlsa ("Audio: Error: Cannot set sample rate (%s)", snd_strerror (err));
return false;
} else
- _debug ("Audio: Set audio rate to %d", _audioSampleRate);
+ _debug ("Audio: Set audio rate to %d", audioSampleRate_);
if (dir != 0) {
- _debugAlsa ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", type , _audioSampleRate, exact_ivalue);
- _audioSampleRate = exact_ivalue;
+ _debugAlsa ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", type , audioSampleRate_, exact_ivalue);
+ audioSampleRate_ = exact_ivalue;
}
/* Set the number of channels */
@@ -808,19 +802,19 @@ namespace
void AlsaLayer::audioCallback (void)
{
- unsigned int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
- bool resample = _audioSampleRate != _mainBufferSampleRate;
+ unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ bool resample = audioSampleRate_ != mainBufferSampleRate;
notifyincomingCall();
- unsigned short spkrVolume = _manager->getSpkrVolume();
+ unsigned short spkrVolume = manager_->getSpkrVolume();
- AudioLoop *tone = _manager->getTelephoneTone();
- AudioLoop *file_tone = _manager->getTelephoneFile();
+ AudioLoop *tone = manager_->getTelephoneTone();
+ AudioLoop *file_tone = manager_->getTelephoneFile();
// AvailForGet tell the number of chars inside the buffer
// framePerBuffer are the number of data for one channel (left)
- int urgentAvailBytes = _urgentRingBuffer.AvailForGet();
+ int urgentAvailBytes = urgentRingBuffer_.AvailForGet();
if (!playbackHandle_ or !captureHandle_)
return;
@@ -836,7 +830,7 @@ void AlsaLayer::audioCallback (void)
if (toGet > playbackAvailBytes)
toGet = playbackAvailBytes;
SFLDataFormat *out = (SFLDataFormat*) malloc (toGet);
- _urgentRingBuffer.Get (out, toGet);
+ urgentRingBuffer_.Get (out, toGet);
adjustVolume(out, toGet / sizeof(SFLDataFormat), spkrVolume);
write (out, toGet, playbackHandle_);
@@ -867,7 +861,7 @@ void AlsaLayer::audioCallback (void)
// Compute maximal value to get from the ring buffer
double resampleFactor = 1.0;
if (resample) {
- resampleFactor = (double) _audioSampleRate / _mainBufferSampleRate;
+ resampleFactor = (double) audioSampleRate_ / mainBufferSampleRate;
maxNbBytesToGet = (double) toGet / resampleFactor;
}
@@ -882,7 +876,7 @@ void AlsaLayer::audioCallback (void)
int inSamples = toGet / sizeof(SFLDataFormat);
int outSamples = inSamples * resampleFactor;
SFLDataFormat *rsmpl_out = (SFLDataFormat*) malloc (outSamples * sizeof(SFLDataFormat));
- converter_->resample (out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, inSamples);
+ converter_->resample (out, rsmpl_out, mainBufferSampleRate, audioSampleRate_, inSamples);
write (rsmpl_out, outSamples * sizeof(SFLDataFormat), playbackHandle_);
free (rsmpl_out);
} else {
@@ -932,19 +926,19 @@ void AlsaLayer::audioCallback (void)
free(in);
return;
}
- adjustVolume (in, toPutSamples, _manager->getSpkrVolume());
+ adjustVolume (in, toPutSamples, manager_->getSpkrVolume());
if (resample) {
- int outSamples = toPutSamples * ((double) _audioSampleRate / _mainBufferSampleRate);
+ int outSamples = toPutSamples * ((double) audioSampleRate_ / mainBufferSampleRate);
int outBytes = outSamples * sizeof (SFLDataFormat);
SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (outBytes);
- converter_->resample ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toPutSamples);
- _audiofilter->processAudio (rsmpl_out, outBytes);
+ converter_->resample ( (SFLDataFormat*) in, rsmpl_out, mainBufferSampleRate, audioSampleRate_, toPutSamples);
+ audiofilter_->processAudio (rsmpl_out, outBytes);
getMainBuffer()->putData (rsmpl_out, outBytes);
free (rsmpl_out);
} else {
SFLDataFormat* filter_out = (SFLDataFormat*) malloc (toPutBytes);
- _audiofilter->processAudio (in, filter_out, toPutBytes);
+ audiofilter_->processAudio (in, filter_out, toPutBytes);
getMainBuffer()->putData (filter_out, toPutBytes);
free (filter_out);
}
diff --git a/daemon/src/audio/alsa/alsalayer.h b/daemon/src/audio/alsa/alsalayer.h
index ff4f8ace624342b6730e6792204c5fdbae38a40a..46c5c92edbeefcabcb17ed572cb79a5c1f2adc50 100644
--- a/daemon/src/audio/alsa/alsalayer.h
+++ b/daemon/src/audio/alsa/alsalayer.h
@@ -144,19 +144,6 @@ class AlsaLayer : public AudioLayer
void audioCallback (void);
- /**
- * Get the noise suppressor state
- * @return true if noise suppressor activated
- */
- virtual bool getNoiseSuppressState (void) const {
- return AudioLayer::_noisesuppressstate;
- }
-
- /**
- * Set the noise suppressor state
- * @param state true if noise suppressor active, false elsewhere
- */
- virtual void setNoiseSuppressState (bool state);
private:
void closeLayer (void);
diff --git a/daemon/src/audio/audiolayer.cpp b/daemon/src/audio/audiolayer.cpp
index 973f0272e13dc78783a83f19dda6e0feafcaee5c..dc19e57380e72e92ecba7d370ac492b6b9c4d6e2 100644
--- a/daemon/src/audio/audiolayer.cpp
+++ b/daemon/src/audio/audiolayer.cpp
@@ -36,66 +36,66 @@
#include <cc++/numbers.h>
AudioLayer::AudioLayer (ManagerImpl* manager , int type)
- : _layerType (type)
- , _isStarted(false)
- , _manager (manager)
- , _urgentRingBuffer (SIZEBUF, Call::DEFAULT_ID)
- , _mainBuffer(0)
- , _recorder(0)
- , _indexIn (0)
- , _indexOut (0)
- , _indexRing(0)
- , _audioSampleRate (0)
- , _frameSize (0)
- , _inChannel (1)
- , _outChannel (1)
- , _errorMessage (0)
- , _mutex ()
- , _dcblocker(0)
- , _audiofilter(0)
- , _noisesuppressstate(false)
- , _countNotificationTime(0)
- , _time (new ost::Time)
+ : layerType_ (type)
+ , isStarted_ (false)
+ , manager_ (manager)
+ , urgentRingBuffer_ (SIZEBUF, Call::DEFAULT_ID)
+ , mainBuffer_ (0)
+ , recorder_ (0)
+ , indexIn_ (0)
+ , indexOut_ (0)
+ , indexRing_ (0)
+ , audioSampleRate_ (0)
+ , frameSize_ (0)
+ , inChannel_ (1)
+ , outChannel_ (1)
+ , errorMessage_ (0)
+ , mutex_ ()
+ , dcblocker_ (0)
+ , audiofilter_ (0)
+ , noiseSuppressState_ (false)
+ , countNotificationTime_ (0)
+ , time_ (new ost::Time)
{}
AudioLayer::~AudioLayer ()
{
- delete _time;
- delete _audiofilter;
- delete _dcblocker;
+ delete time_;
+ delete audiofilter_;
+ delete dcblocker_;
}
void AudioLayer::flushMain (void)
{
- ost::MutexLock guard (_mutex);
+ ost::MutexLock guard (mutex_);
// should pass call id
getMainBuffer()->flushAllBuffers();
}
void AudioLayer::flushUrgent (void)
{
- ost::MutexLock guard (_mutex);
- _urgentRingBuffer.flushAll();
+ ost::MutexLock guard (mutex_);
+ urgentRingBuffer_.flushAll();
}
void AudioLayer::putUrgent (void* buffer, int toCopy)
{
- ost::MutexLock guard (_mutex);
- _urgentRingBuffer.Put (buffer, toCopy);
+ ost::MutexLock guard (mutex_);
+ urgentRingBuffer_.Put (buffer, toCopy);
}
void AudioLayer::notifyincomingCall()
{
// Notify (with a beep) an incoming call when there is already a call
if (Manager::instance().incomingCallWaiting()) {
- _countNotificationTime += _time->getSecond();
- int countTimeModulo = _countNotificationTime % 5000;
+ countNotificationTime_ += time_->getSecond();
+ int countTimeModulo = countNotificationTime_ % 5000;
- if ((countTimeModulo - _countNotificationTime) < 0)
+ if ((countTimeModulo - countNotificationTime_) < 0)
Manager::instance().notificationIncomingCall();
- _countNotificationTime = countTimeModulo;
+ countNotificationTime_ = countTimeModulo;
}
}
diff --git a/daemon/src/audio/audiolayer.h b/daemon/src/audio/audiolayer.h
index 0ba22e506243c5c48340a84bd5ec677d18785637..47a4fc70fa6d20543e9079360e48fbff982cf6d9 100644
--- a/daemon/src/audio/audiolayer.h
+++ b/daemon/src/audio/audiolayer.h
@@ -101,7 +101,7 @@ class AudioLayer
*/
virtual void stopStream (void) = 0;
- bool isStarted(void) { return _isStarted; }
+ bool isStarted(void) const { return isStarted_; }
/**
* Send a chunk of data to the hardware buffer to start the playback
@@ -121,8 +121,8 @@ class AudioLayer
* Could be: ALSA_PLAYBACK_DEVICE
* ALSA_CAPTURE_DEVICE
*/
- void setErrorMessage (const int& error) {
- _errorMessage = error;
+ void setErrorMessage (int error) {
+ errorMessage_ = error;
}
/**
@@ -130,7 +130,7 @@ class AudioLayer
* @return int The error code
*/
int getErrorMessage() const {
- return _errorMessage;
+ return errorMessage_;
}
/**
@@ -139,7 +139,7 @@ class AudioLayer
* 0 for the first available card on the system, 1 ...
*/
int getIndexIn() const {
- return _indexIn;
+ return indexIn_;
}
/**
@@ -148,7 +148,7 @@ class AudioLayer
* 0 for the first available card on the system, 1 ...
*/
int getIndexOut() const {
- return _indexOut;
+ return indexOut_;
}
/**
@@ -157,7 +157,7 @@ class AudioLayer
* 0 for the first available card on the system, 1 ...
*/
int getIndexRing() const {
- return _indexRing;
+ return indexRing_;
}
/**
@@ -166,7 +166,7 @@ class AudioLayer
* default: 44100 HZ
*/
unsigned int getSampleRate() const {
- return _audioSampleRate;
+ return audioSampleRate_;
}
/**
@@ -175,7 +175,7 @@ class AudioLayer
* default: 20 ms
*/
unsigned int getFrameSize() const {
- return _frameSize;
+ return frameSize_;
}
/**
@@ -184,7 +184,7 @@ class AudioLayer
*
*/
int getLayerType (void) const {
- return _layerType;
+ return layerType_;
}
/**
@@ -196,54 +196,56 @@ class AudioLayer
* @return MainBuffer* a pointer to the MainBuffer instance
*/
MainBuffer* getMainBuffer (void) const {
- return _mainBuffer;
+ return mainBuffer_;
}
/**
* Set the mainbuffer once the audiolayer is created
*/
void setMainBuffer (MainBuffer* mainbuffer) {
- _mainBuffer = mainbuffer;
+ mainBuffer_ = mainbuffer;
}
/**
* Set the audio recorder
*/
void setRecorderInstance (Recordable* rec) {
- _recorder = rec;
+ recorder_ = rec;
}
/**
* Get the audio recorder
*/
Recordable* getRecorderInstance (void) const {
- return _recorder;
+ return recorder_;
}
- /**
- * Get the noise suppressor state
- * @return true if noise suppressor activated
- */
- virtual bool getNoiseSuppressState (void) const = 0;
-
/**
* Set the noise suppressor state
* @param state true if noise suppressor active, false elsewhere
*/
- virtual void setNoiseSuppressState (bool state) = 0;
+ void setNoiseSuppressState (bool state) { noiseSuppressState_ = state; }
+
+ /**
+ * Get the noise suppressor state
+ * @return true if noise suppressor activated
+ */
+ bool getNoiseSuppressState (void) const {
+ return noiseSuppressState_;
+ }
/**
* Get the mutex lock for the entire audio layer
*/
ost::Mutex* getMutexLock (void) {
- return &_mutex;
+ return &mutex_;
}
void notifyincomingCall (void);
protected:
- int _layerType;
+ int layerType_;
/**
* Drop the pending frames and close the capture device
@@ -258,15 +260,15 @@ class AudioLayer
/**
* Wether or not the audio layer stream is started
*/
- bool _isStarted;
+ bool isStarted_;
/** Augment coupling, reduce indirect access */
- ManagerImpl* _manager;
+ ManagerImpl* manager_;
/**
* Urgent ring buffer used for ringtones
*/
- RingBuffer _urgentRingBuffer;
+ RingBuffer urgentRingBuffer_;
/**
* Instance of the MainBuffer for the whole application
@@ -275,72 +277,72 @@ class AudioLayer
* Audio instances must be registered into the MainBuffer and bound together via the ManagerImpl.
*
*/
- MainBuffer* _mainBuffer;
+ MainBuffer* mainBuffer_;
/**
* A pointer to the recordable instance (may be a call or a conference)
*/
- Recordable* _recorder;
+ Recordable* recorder_;
/**
* Number of audio cards on which capture stream has been opened
*/
- int _indexIn;
+ int indexIn_;
/**
* Number of audio cards on which playback stream has been opened
*/
- int _indexOut;
+ int indexOut_;
/**
* Number of audio cards on which ringtone stream has been opened
*/
- int _indexRing;
+ int indexRing_;
/**
* Sample Rate SFLphone should send sound data to the sound card
* The value can be set in the user config file- now: 44100HZ
*/
- unsigned int _audioSampleRate;
+ unsigned int audioSampleRate_;
/**
* Length of the sound frame we capture or read in ms
* The value can be set in the user config file - now: 20ms
*/
- unsigned int _frameSize;
+ unsigned int frameSize_;
/**
* Input channel (mic) should be 1 mono
*/
- unsigned int _inChannel;
+ unsigned int inChannel_;
/**
* Output channel (stereo) should be 1 mono
*/
- unsigned int _outChannel;
+ unsigned int outChannel_;
/** Contains the current error code */
- int _errorMessage;
+ int errorMessage_;
/**
* Lock for the entire audio layer
*/
- ost::Mutex _mutex;
+ ost::Mutex mutex_;
- DcBlocker *_dcblocker;
- AudioProcessing *_audiofilter;
+ DcBlocker *dcblocker_;
+ AudioProcessing *audiofilter_;
- bool _noisesuppressstate;
+ bool noiseSuppressState_;
/**
* Time counter used to trigger incoming call notification
*/
- int _countNotificationTime;
+ int countNotificationTime_;
/**
* Used to get formated system time in order to compute incoming call notification
*/
- ost::Time * _time;
+ ost::Time * time_;
};
#endif // _AUDIO_LAYER_H_
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/audio/pulseaudio/pulselayer.cpp
index d46b0016bf55e9cab9911d67ea838845c2d91d92..da0d04fa451968cef82e4766db12cf09001e0551 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/audio/pulseaudio/pulselayer.cpp
@@ -227,9 +227,9 @@ PulseLayer::PulseLayer (ManagerImpl* manager)
, ringtone_(0)
, converter_(0)
{
- _urgentRingBuffer.createReadPointer();
+ urgentRingBuffer_.createReadPointer();
- AudioLayer::_noisesuppressstate = true;
+ noiseSuppressState_ = true;
openLayer();
}
@@ -244,14 +244,14 @@ PulseLayer::~PulseLayer (void)
void
PulseLayer::openLayer (void)
{
- if(_isStarted)
+ if (isStarted_)
return;
_info ("Audio: Open Pulseaudio layer");
connectPulseAudioServer();
- _isStarted = true;
+ isStarted_ = true;
}
void
@@ -259,7 +259,7 @@ PulseLayer::closeLayer (void)
{
_info ("Audio: Close Pulseaudio layer");
- _isStarted = false;
+ isStarted_ = false;
disconnectAudioStream();
@@ -375,19 +375,19 @@ void PulseLayer::context_state_callback (pa_context* c, void* user_data)
void PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int indexRing UNUSED, int sampleRate, int frameSize , int stream UNUSED, const std::string &plugin UNUSED)
{
- _debug ("Audio: Open device sampling rate %d, frame size %d", _audioSampleRate, _frameSize);
+ _debug ("Audio: Open device sampling rate %d, frame size %d", audioSampleRate_, frameSize_);
- _audioSampleRate = sampleRate;
- _frameSize = frameSize;
+ audioSampleRate_ = sampleRate;
+ frameSize_ = frameSize;
flushUrgent();
// use 1 sec buffer for resampling
- converter_ = new SamplerateConverter (_audioSampleRate);
+ converter_ = new SamplerateConverter (audioSampleRate_);
// Instantiate the algorithm
- AudioLayer::_dcblocker = new DcBlocker;
- AudioLayer::_audiofilter = new AudioProcessing (_dcblocker);
+ dcblocker_ = new DcBlocker;
+ audiofilter_ = new AudioProcessing (dcblocker_);
}
@@ -426,11 +426,11 @@ void PulseLayer::createStreams (pa_context* c)
{
_info ("Audio: Create streams");
- playback_ = new AudioStream (c, mainloop_, PLAYBACK_STREAM_NAME, PLAYBACK_STREAM, _audioSampleRate);
+ playback_ = new AudioStream (c, mainloop_, PLAYBACK_STREAM_NAME, PLAYBACK_STREAM, audioSampleRate_);
- std::string playbackDevice(_manager->audioPreference.getDevicePlayback());
- std::string recordDevice(_manager->audioPreference.getDeviceRecord());
- std::string ringtoneDevice(_manager->audioPreference.getDeviceRingtone());
+ std::string playbackDevice(manager_->audioPreference.getDevicePlayback());
+ std::string recordDevice(manager_->audioPreference.getDeviceRecord());
+ std::string ringtoneDevice(manager_->audioPreference.getDeviceRingtone());
_debug ("Audio: Device for playback: %s", playbackDevice.c_str());
_debug ("Audio: Device for record: %s", recordDevice.c_str());
@@ -447,7 +447,7 @@ void PulseLayer::createStreams (pa_context* c)
pa_stream_set_moved_callback (playback_->pulseStream(), stream_moved_callback, this);
pa_stream_set_latency_update_callback (playback_->pulseStream(), latency_update_callback, this);
- record_ = new AudioStream (c, mainloop_, CAPTURE_STREAM_NAME, CAPTURE_STREAM, _audioSampleRate);
+ record_ = new AudioStream (c, mainloop_, CAPTURE_STREAM_NAME, CAPTURE_STREAM, audioSampleRate_);
if (inSourceList (recordDevice))
record_->connectStream (&recordDevice);
@@ -458,7 +458,7 @@ void PulseLayer::createStreams (pa_context* c)
pa_stream_set_moved_callback (record_->pulseStream(), stream_moved_callback, this);
pa_stream_set_latency_update_callback (record_->pulseStream(), latency_update_callback, this);
- ringtone_ = new AudioStream (c, mainloop_, RINGTONE_STREAM_NAME, RINGTONE_STREAM, _audioSampleRate);
+ ringtone_ = new AudioStream (c, mainloop_, RINGTONE_STREAM_NAME, RINGTONE_STREAM, audioSampleRate_);
if (inSourceList (ringtoneDevice))
ringtone_->connectStream (&ringtoneDevice);
@@ -493,7 +493,7 @@ void PulseLayer::closeCaptureStream (void)
if (name && strlen (name)) {
_debug ("Audio: record device to be stored in config: %s", name);
- _manager->audioPreference.setDeviceRecord (name);
+ manager_->audioPreference.setDeviceRecord (name);
}
}
@@ -511,7 +511,7 @@ void PulseLayer::closePlaybackStream (void)
if (name && strlen (name)) {
_debug ("Audio: playback device to be stored in config: %s", name);
- _manager->audioPreference.setDevicePlayback (name);
+ manager_->audioPreference.setDevicePlayback (name);
}
}
@@ -525,7 +525,7 @@ void PulseLayer::closePlaybackStream (void)
if (name && strlen (name)) {
_debug ("Audio: ringtone device to be stored in config: %s", name);
- _manager->audioPreference.setDeviceRingtone (name);
+ manager_->audioPreference.setDeviceRingtone (name);
}
}
@@ -622,14 +622,6 @@ void PulseLayer::processData (void)
readFromMic();
}
-void PulseLayer::setNoiseSuppressState (bool state)
-{
- // if a stream already opened
- AudioLayer::_noisesuppressstate = state;
-
-}
-
-
void PulseLayer::writeToSpeaker (void)
{
notifyincomingCall();
@@ -641,12 +633,12 @@ void PulseLayer::writeToSpeaker (void)
return;
}
- int urgentBytes = _urgentRingBuffer.AvailForGet();
+ int urgentBytes = urgentRingBuffer_.AvailForGet();
if (urgentBytes > writeableSizeBytes)
urgentBytes = writeableSizeBytes;
if (urgentBytes) {
SFLDataFormat *out = (SFLDataFormat*) pa_xmalloc (urgentBytes);
- _urgentRingBuffer.Get (out, urgentBytes);
+ urgentRingBuffer_.Get (out, urgentBytes);
pa_stream_write (playback_->pulseStream(), out, urgentBytes, NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (out);
// Consume the regular one as well (same amount of bytes)
@@ -654,7 +646,7 @@ void PulseLayer::writeToSpeaker (void)
return;
}
- AudioLoop *toneToPlay = _manager->getTelephoneTone();
+ AudioLoop *toneToPlay = manager_->getTelephoneTone();
if (toneToPlay) {
if (playback_->getStreamState() == PA_STREAM_READY) {
SFLDataFormat *out = (SFLDataFormat*) pa_xmalloc (writeableSizeBytes);
@@ -677,7 +669,7 @@ void PulseLayer::writeToSpeaker (void)
}
unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
- bool resample = _audioSampleRate != mainBufferSampleRate;
+ bool resample = audioSampleRate_ != mainBufferSampleRate;
// how much samples we can write in the output
int outSamples = writeableSizeBytes / sizeof(SFLDataFormat);
@@ -687,7 +679,7 @@ void PulseLayer::writeToSpeaker (void)
double resampleFactor = 1.;
if (resample) {
- resampleFactor = (double) _audioSampleRate / mainBufferSampleRate;
+ resampleFactor = (double) audioSampleRate_ / mainBufferSampleRate;
inSamples = (double) inSamples / resampleFactor;
}
@@ -702,7 +694,7 @@ void PulseLayer::writeToSpeaker (void)
// test if resampling is required
if (resample) {
SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (outBytes);
- converter_->resample (out, rsmpl_out, mainBufferSampleRate, _audioSampleRate, inSamples);
+ converter_->resample (out, rsmpl_out, mainBufferSampleRate, audioSampleRate_, inSamples);
pa_stream_write (playback_->pulseStream(), rsmpl_out, outBytes, NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (rsmpl_out);
} else
@@ -717,7 +709,7 @@ void PulseLayer::readFromMic (void)
size_t r;
unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
- bool resample = _audioSampleRate != mainBufferSampleRate;
+ bool resample = audioSampleRate_ != mainBufferSampleRate;
if (pa_stream_peek (record_->pulseStream() , (const void**) &data , &r) < 0 or !data) {
_error("Audio: Error capture stream peek failed: %s" , pa_strerror (pa_context_errno (context_)));
@@ -726,20 +718,20 @@ void PulseLayer::readFromMic (void)
if (resample) {
int inSamples = r / sizeof(SFLDataFormat);
- double resampleFactor = (double) _audioSampleRate / mainBufferSampleRate;
+ double resampleFactor = (double) audioSampleRate_ / mainBufferSampleRate;
int outSamples = (double) inSamples * resampleFactor;
int outBytes = outSamples * sizeof(SFLDataFormat);
SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (outBytes);
- converter_->resample ( (SFLDataFormat *) data, rsmpl_out, mainBufferSampleRate, _audioSampleRate, inSamples);
+ converter_->resample ( (SFLDataFormat *) data, rsmpl_out, mainBufferSampleRate, audioSampleRate_, inSamples);
// remove dc offset
- _audiofilter->processAudio (rsmpl_out, outBytes);
+ audiofilter_->processAudio (rsmpl_out, outBytes);
getMainBuffer()->putData (rsmpl_out, outBytes);
pa_xfree (rsmpl_out);
} else {
SFLDataFormat* filter_out = (SFLDataFormat*) pa_xmalloc (r);
// remove dc offset
- _audiofilter->processAudio ( (SFLDataFormat *) data, filter_out, r);
+ audiofilter_->processAudio ( (SFLDataFormat *) data, filter_out, r);
getMainBuffer()->putData (filter_out, r);
pa_xfree (filter_out);
}
@@ -752,7 +744,7 @@ end:
void PulseLayer::ringtoneToSpeaker (void)
{
- AudioLoop* fileToPlay = _manager->getTelephoneFile();
+ AudioLoop* fileToPlay = manager_->getTelephoneFile();
int writableSize = pa_stream_writable_size (ringtone_->pulseStream());
if (fileToPlay) {
diff --git a/daemon/src/audio/pulseaudio/pulselayer.h b/daemon/src/audio/pulseaudio/pulselayer.h
index 05ed174a8ad31e5169419eba7f1417bbd9966709..353f1668667ebffa83f31b12dc2983ac7b8a2403 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.h
+++ b/daemon/src/audio/pulseaudio/pulselayer.h
@@ -173,20 +173,6 @@ class PulseLayer : public AudioLayer
*/
void processData (void);
- /**
- * Get the noise suppressor state
- * @return true if noise suppressor activated
- */
- bool getNoiseSuppressState (void) const {
- return AudioLayer::_noisesuppressstate;
- }
-
- /**
- * Set the noise suppressor state
- * @param state true if noise suppressor active, false elsewhere
- */
- void setNoiseSuppressState (bool state);
-
private:
// Copy Constructor
PulseLayer (const PulseLayer& rh);