diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.cpp b/daemon/src/audio/audiortp/audio_rtp_factory.cpp
index aaac2ef4feb0028427887d21b9f96f110c4105a7..ee57e4ec085b6a0c15149358ac4a66a248245638 100644
--- a/daemon/src/audio/audiortp/audio_rtp_factory.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_factory.cpp
@@ -42,17 +42,17 @@
namespace sfl {
-AudioRtpFactory::AudioRtpFactory(SIPCall *ca) : _rtpSession(NULL), remoteContext(NULL), localContext(NULL), ca_(ca)
+AudioRtpFactory::AudioRtpFactory(SIPCall *ca) : rtpSession_(NULL), remoteContext_(NULL), localContext_(NULL), ca_(ca)
{}
AudioRtpFactory::~AudioRtpFactory()
{
- delete _rtpSession;
+ delete rtpSession_;
}
void AudioRtpFactory::initAudioRtpConfig()
{
- if (_rtpSession != NULL)
+ if (rtpSession_ != NULL)
stop();
std::string accountId(Manager::instance().getAccountFromCall(ca_->getCallId()));
@@ -60,150 +60,136 @@ void AudioRtpFactory::initAudioRtpConfig()
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(accountId));
if (account) {
- _srtpEnabled = account->getSrtpEnabled();
+ srtpEnabled_ = account->getSrtpEnabled();
std::string key(account->getSrtpKeyExchange());
if (key == "sdes")
- _keyExchangeProtocol = Sdes;
+ keyExchangeProtocol_ = Sdes;
else if (key == "zrtp")
- _keyExchangeProtocol = Zrtp;
+ keyExchangeProtocol_ = Zrtp;
else
- _keyExchangeProtocol = Symmetric;
+ keyExchangeProtocol_ = Symmetric;
- _helloHashEnabled = account->getZrtpHelloHash();
+ helloHashEnabled_ = account->getZrtpHelloHash();
} else {
- _srtpEnabled = false;
- _keyExchangeProtocol = Symmetric;
- _helloHashEnabled = false;
+ srtpEnabled_ = false;
+ keyExchangeProtocol_ = Symmetric;
+ helloHashEnabled_ = false;
}
}
void AudioRtpFactory::initAudioSymmetricRtpSession()
{
- ost::MutexLock m(_audioRtpThreadMutex);
+ ost::MutexLock m(audioRtpThreadMutex_);
- if (_srtpEnabled) {
+ if (srtpEnabled_) {
std::string zidFilename(Manager::instance().voipPreferences.getZidFile());
- switch (_keyExchangeProtocol) {
+ switch (keyExchangeProtocol_) {
case Zrtp:
- _rtpSession = new AudioZrtpSession(ca_, zidFilename);
-
- if (_helloHashEnabled) {
- // TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
- // not even available at that point.
- ca_->getLocalSDP()->setZrtpHash(static_cast<AudioZrtpSession *>(_rtpSession)->getHelloHash());
- }
-
+ rtpSession_ = new AudioZrtpSession(ca_, zidFilename);
+ // TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
+ // not even available at that point.
+ if (helloHashEnabled_)
+ ca_->getLocalSDP()->setZrtpHash(static_cast<AudioZrtpSession *>(rtpSession_)->getHelloHash());
break;
case Sdes:
- _rtpSession = new AudioSrtpSession(ca_);
+ rtpSession_ = new AudioSrtpSession(ca_);
break;
default:
throw UnsupportedRtpSessionType("Unsupported Rtp Session Exception Type!");
}
- } else {
- _rtpSession = new AudioSymmetricRtpSession(ca_);
- }
+ } else
+ rtpSession_ = new AudioSymmetricRtpSession(ca_);
}
void AudioRtpFactory::start(AudioCodec* audiocodec)
{
- if (_rtpSession == NULL) {
+ if (rtpSession_ == NULL)
throw AudioRtpFactoryException("AudioRtpFactory: Error: RTP session was null when trying to start audio thread");
- }
- if (_rtpSession->getAudioRtpType() == Sdes) {
- if (localContext && remoteContext) {
- static_cast<AudioSrtpSession *>(_rtpSession)->restoreCryptoContext(localContext, remoteContext);
- }
- }
+ if (rtpSession_->getAudioRtpType() == Sdes)
+ if (localContext_ and remoteContext_)
+ static_cast<AudioSrtpSession *>(rtpSession_)->restoreCryptoContext(localContext_, remoteContext_);
- if (_rtpSession->startRtpThread(audiocodec) != 0) {
+ if (rtpSession_->startRtpThread(audiocodec) != 0)
throw AudioRtpFactoryException("AudioRtpFactory: Error: Failed to start AudioZrtpSession thread");
- }
-
}
-void AudioRtpFactory::stop(void)
+void AudioRtpFactory::stop()
{
- ost::MutexLock mutex(_audioRtpThreadMutex);
+ ost::MutexLock mutex(audioRtpThreadMutex_);
- if (_rtpSession == NULL)
+ if (rtpSession_ == NULL)
return;
- if (_rtpSession->getAudioRtpType() == Sdes) {
- localContext = static_cast<AudioSrtpSession *>(_rtpSession)->_localCryptoCtx;
- remoteContext = static_cast<AudioSrtpSession *>(_rtpSession)->_remoteCryptoCtx;
+ if (rtpSession_->getAudioRtpType() == Sdes) {
+ localContext_ = static_cast<AudioSrtpSession*>(rtpSession_)->localCryptoCtx_;
+ remoteContext_ = static_cast<AudioSrtpSession*>(rtpSession_)->remoteCryptoCtx_;
}
- delete _rtpSession;
- _rtpSession = NULL;
+ delete rtpSession_;
+ rtpSession_ = NULL;
}
int AudioRtpFactory::getSessionMedia()
{
- if (_rtpSession == NULL) {
+ if (rtpSession_ == NULL)
throw AudioRtpFactoryException("AudioRtpFactory: Error: RTP session was null when trying to get session media type");
- }
- return _rtpSession->getCodecPayloadType();
+ return rtpSession_->getCodecPayloadType();
}
void AudioRtpFactory::updateSessionMedia(AudioCodec *audiocodec)
{
- if (_rtpSession == NULL) {
- throw AudioRtpFactoryException("AudioRtpFactory: Error: _rtpSession was null when trying to update IP address");
- }
+ if (rtpSession_ == NULL)
+ throw AudioRtpFactoryException("AudioRtpFactory: Error: rtpSession_ was null when trying to update IP address");
- _rtpSession->updateSessionMedia(audiocodec);
+ rtpSession_->updateSessionMedia(audiocodec);
}
-void AudioRtpFactory::updateDestinationIpAddress(void)
+void AudioRtpFactory::updateDestinationIpAddress()
{
- if (_rtpSession)
- _rtpSession->updateDestinationIpAddress();
+ if (rtpSession_)
+ rtpSession_->updateDestinationIpAddress();
}
sfl::AudioZrtpSession * AudioRtpFactory::getAudioZrtpSession()
{
- if (_rtpSession->getAudioRtpType() == Zrtp) {
- return static_cast<AudioZrtpSession *>(_rtpSession);
- } else {
- throw AudioRtpFactoryException("RTP: Error: _rtpSession is NULL in getAudioZrtpSession");
- }
+ if (rtpSession_->getAudioRtpType() == Zrtp)
+ return static_cast<AudioZrtpSession *>(rtpSession_);
+ else
+ throw AudioRtpFactoryException("RTP: Error: rtpSession_ is NULL in getAudioZrtpSession");
}
void sfl::AudioRtpFactory::initLocalCryptoInfo()
{
- if (_rtpSession && _rtpSession->getAudioRtpType() == Sdes) {
- static_cast<AudioSrtpSession *>(_rtpSession)->initLocalCryptoInfo();
-
- ca_->getLocalSDP()->setLocalSdpCrypto(static_cast<AudioSrtpSession *>(_rtpSession)->getLocalCryptoInfo());
+ if (rtpSession_ && rtpSession_->getAudioRtpType() == Sdes) {
+ static_cast<AudioSrtpSession *>(rtpSession_)->initLocalCryptoInfo();
+ ca_->getLocalSDP()->setLocalSdpCrypto(static_cast<AudioSrtpSession *>(rtpSession_)->getLocalCryptoInfo());
}
}
void AudioRtpFactory::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
{
- if (_rtpSession && _rtpSession->getAudioRtpType() == Sdes) {
- static_cast<AudioSrtpSession *>(_rtpSession)->setRemoteCryptoInfo(nego);
- } else {
- throw AudioRtpFactoryException("RTP: Error: _rtpSession is NULL in setRemoteCryptoInfo");
- }
+ if (rtpSession_ && rtpSession_->getAudioRtpType() == Sdes)
+ static_cast<AudioSrtpSession *>(rtpSession_)->setRemoteCryptoInfo(nego);
+ else
+ throw AudioRtpFactoryException("RTP: Error: rtpSession_ is NULL in setRemoteCryptoInfo");
}
void AudioRtpFactory::setDtmfPayloadType(unsigned int payloadType)
{
- if (_rtpSession)
- _rtpSession->setDtmfPayloadType(payloadType);
+ if (rtpSession_)
+ rtpSession_->setDtmfPayloadType(payloadType);
}
void AudioRtpFactory::sendDtmfDigit(int digit)
{
- _rtpSession->putDtmfEvent(digit);
+ rtpSession_->putDtmfEvent(digit);
}
}
diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.h b/daemon/src/audio/audiortp/audio_rtp_factory.h
index 80adaae98aab9a641bd2a04a284a81a076e23c8e..0e33ae57c7d4f063122a23b60b260ad10a9a1b6a 100644
--- a/daemon/src/audio/audiortp/audio_rtp_factory.h
+++ b/daemon/src/audio/audiortp/audio_rtp_factory.h
@@ -99,17 +99,17 @@ class AudioRtpFactory {
* Update current RTP destination address with one stored in call
* @param None
*/
- void updateDestinationIpAddress(void);
+ void updateDestinationIpAddress();
- bool isSdesEnabled(void) const {
- return _srtpEnabled && _keyExchangeProtocol == sfl::Sdes;
+ bool isSdesEnabled() const {
+ return srtpEnabled_ and keyExchangeProtocol_ == sfl::Sdes;
}
/**
* Manually set the srtpEnable option (usefull for RTP fallback)
*/
void setSrtpEnabled(bool enable) {
- _srtpEnabled = enable;
+ srtpEnabled_ = enable;
}
/**
@@ -138,26 +138,26 @@ class AudioRtpFactory {
void sendDtmfDigit(int digit);
private:
- AudioRtpSession *_rtpSession;
- ost::Mutex _audioRtpThreadMutex;
+ AudioRtpSession *rtpSession_;
+ ost::Mutex audioRtpThreadMutex_;
// Field used when initializinga udio rtp session
// May be set manually or from config using initAudioRtpConfig
- bool _srtpEnabled;
+ bool srtpEnabled_;
// Field used when initializinga udio rtp session
// May be set manually or from config using initAudioRtpConfig
- RtpMethod _keyExchangeProtocol;
+ RtpMethod keyExchangeProtocol_;
// Field used when initializinga udio rtp session
// May be set manually or from config using initAudioRtpConfig
- bool _helloHashEnabled;
+ bool helloHashEnabled_;
/** Remote srtp crypto context to be set into incoming data queue. */
- ost::CryptoContext *remoteContext;
+ ost::CryptoContext *remoteContext_;
/** Local srtp crypto context to be set into outgoing data queue. */
- ost::CryptoContext *localContext;
+ ost::CryptoContext *localContext_;
SIPCall *ca_;
};
diff --git a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
index 9ed316885fca496c6b86a4ad476d3fe8f7196897..f29196b78801d027d9218f9b3c70ef29cf170a7c 100644
--- a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
@@ -38,76 +38,66 @@ namespace sfl {
static const SFLDataFormat initFadeinFactor = 32000;
-AudioRtpRecord::AudioRtpRecord() : _audioCodec(NULL)
- , _hasDynamicPayloadType(false)
- , _converter(NULL)
- , _codecSampleRate(0)
- , _codecFrameSize(0)
- , _micAmplFactor(initFadeinFactor)
- , _noiseSuppress(NULL)
- , _callId("")
- , _dtmfPayloadType(101) // same as Asterisk
-{
-
-}
+AudioRtpRecord::AudioRtpRecord() : audioCodec_(NULL)
+ , hasDynamicPayloadType_(false)
+ , converter_(NULL)
+ , codecSampleRate_(0)
+ , codecFrameSize_(0)
+ , micAmplFactor_(initFadeinFactor)
+ , noiseSuppress_(NULL)
+ , callId_("")
+ , dtmfPayloadType_(101) // same as Asterisk
+{}
AudioRtpRecord::~AudioRtpRecord()
{
- delete _converter;
- delete _audioCodec;
- delete _noiseSuppress;
+ delete converter_;
+ delete audioCodec_;
+ delete noiseSuppress_;
}
-AudioRtpRecordHandler::AudioRtpRecordHandler(SIPCall *ca) : _audioRtpRecord(), id_(ca->getCallId()), echoCanceller(ca->getMemoryPool()), gainController(8000, -10.0)
-{
-
-}
+AudioRtpRecordHandler::AudioRtpRecordHandler(SIPCall *ca) : audioRtpRecord_(), id_(ca->getCallId()), echoCanceller(ca->getMemoryPool()), gainController(8000, -10.0)
+{}
AudioRtpRecordHandler::~AudioRtpRecordHandler() {}
void AudioRtpRecordHandler::setRtpMedia(AudioCodec* audioCodec)
{
- _audioRtpRecord.audioCodecMutex.enter();
+ ost::MutexLock lock(audioRtpRecord_.audioCodecMutex_);
- delete _audioRtpRecord._audioCodec;
+ delete audioRtpRecord_.audioCodec_;
// Set varios codec info to reduce indirection
- _audioRtpRecord._audioCodec = audioCodec;
- _audioRtpRecord._codecPayloadType = audioCodec->getPayloadType();
- _audioRtpRecord._codecSampleRate = audioCodec->getClockRate();
- _audioRtpRecord._codecFrameSize = audioCodec->getFrameSize();
- _audioRtpRecord._hasDynamicPayloadType = audioCodec->hasDynamicPayload();
-
- _audioRtpRecord.audioCodecMutex.leave();
+ audioRtpRecord_.audioCodec_ = audioCodec;
+ audioRtpRecord_.codecPayloadType_ = audioCodec->getPayloadType();
+ audioRtpRecord_.codecSampleRate_ = audioCodec->getClockRate();
+ audioRtpRecord_.codecFrameSize_ = audioCodec->getFrameSize();
+ audioRtpRecord_.hasDynamicPayloadType_ = audioCodec->hasDynamicPayload();
}
void AudioRtpRecordHandler::initBuffers()
{
// Set sampling rate, main buffer choose the highest one
- Manager::instance().audioSamplingRateChanged(_audioRtpRecord._codecSampleRate);
+ Manager::instance().audioSamplingRateChanged(audioRtpRecord_.codecSampleRate_);
// initialize SampleRate converter using AudioLayer's sampling rate
// (internal buffers initialized with maximal sampling rate and frame size)
- delete _audioRtpRecord._converter;
- _audioRtpRecord._converter = new SamplerateConverter(getCodecSampleRate());
+ delete audioRtpRecord_.converter_;
+ audioRtpRecord_.converter_ = new SamplerateConverter(getCodecSampleRate());
}
void AudioRtpRecordHandler::initNoiseSuppress()
{
- _audioRtpRecord.audioProcessMutex.enter();
-
- delete _audioRtpRecord._noiseSuppress;
-
- _audioRtpRecord._noiseSuppress = new NoiseSuppress(getCodecFrameSize(), getCodecSampleRate());
-
- _audioRtpRecord.audioProcessMutex.leave();
+ ost::MutexLock lock(audioRtpRecord_.audioProcessMutex_);
+ delete audioRtpRecord_.noiseSuppress_;
+ audioRtpRecord_.noiseSuppress_ = new NoiseSuppress(getCodecFrameSize(), getCodecSampleRate());
}
void AudioRtpRecordHandler::putDtmfEvent(int digit)
{
- _audioRtpRecord._dtmfQueue.push_back(digit);
+ audioRtpRecord_.dtmfQueue_.push_back(digit);
}
#ifdef DUMP_PROCESS_DATA_ENCODE
@@ -116,9 +106,9 @@ std::ofstream teststream("test_process_data_encode.raw");
int AudioRtpRecordHandler::processDataEncode(void)
{
- SFLDataFormat *micData = _audioRtpRecord.decData;
- unsigned char *micDataEncoded = _audioRtpRecord.encodedData;
- SFLDataFormat *micDataConverted = _audioRtpRecord.resampledData;
+ SFLDataFormat *micData = audioRtpRecord_.decData_;
+ unsigned char *micDataEncoded = audioRtpRecord_.encodedData_;
+ SFLDataFormat *micDataConverted = audioRtpRecord_.resampledData_;
int codecSampleRate = getCodecSampleRate();
int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
@@ -129,9 +119,8 @@ int AudioRtpRecordHandler::processDataEncode(void)
int samplesToGet = resampleFactor * getCodecFrameSize();
int bytesToGet = samplesToGet * sizeof(SFLDataFormat);
- if (Manager::instance().getMainBuffer()->availForGet(id_) < bytesToGet) {
+ if (Manager::instance().getMainBuffer()->availForGet(id_) < bytesToGet)
return 0;
- }
int bytes = Manager::instance().getMainBuffer()->getData(micData, bytesToGet, id_);
@@ -142,7 +131,7 @@ int AudioRtpRecordHandler::processDataEncode(void)
int samples = bytesToGet / sizeof(SFLDataFormat);
- fadeIn(micData, samples, &_audioRtpRecord._micAmplFactor);
+ fadeIn(micData, samples, &audioRtpRecord_.micAmplFactor_);
if (Manager::instance().getEchoCancelState())
echoCanceller.getData(micData);
@@ -155,18 +144,19 @@ int AudioRtpRecordHandler::processDataEncode(void)
if (codecSampleRate != mainBufferSampleRate) {
out = micDataConverted;
- _audioRtpRecord._converter->resample(micData, micDataConverted, codecSampleRate, mainBufferSampleRate, samplesToGet);
+ audioRtpRecord_.converter_->resample(micData, micDataConverted, codecSampleRate, mainBufferSampleRate, samplesToGet);
}
if (Manager::instance().audioPreference.getNoiseReduce()) {
- _audioRtpRecord.audioProcessMutex.enter();
- _audioRtpRecord._noiseSuppress->process(micData, getCodecFrameSize());
- _audioRtpRecord.audioProcessMutex.leave();
+ ost::MutexLock lock(audioRtpRecord_.audioProcessMutex_);
+ audioRtpRecord_.noiseSuppress_->process(micData, getCodecFrameSize());
}
- _audioRtpRecord.audioCodecMutex.enter();
- int compSize = _audioRtpRecord._audioCodec->encode(micDataEncoded, out, getCodecFrameSize());
- _audioRtpRecord.audioCodecMutex.leave();
+ int compSize;
+ {
+ ost::MutexLock lock(audioRtpRecord_.audioCodecMutex_);
+ compSize = audioRtpRecord_.audioCodec_->encode(micDataEncoded, out, getCodecFrameSize());
+ }
return compSize;
}
@@ -178,19 +168,19 @@ void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, unsigned
int codecSampleRate = getCodecSampleRate();
- SFLDataFormat *spkrDataDecoded = _audioRtpRecord.decData;
- SFLDataFormat *spkrDataConverted = _audioRtpRecord.resampledData;
+ SFLDataFormat *spkrDataDecoded = audioRtpRecord_.decData_;
+ SFLDataFormat *spkrDataConverted = audioRtpRecord_.resampledData_;
int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
- _audioRtpRecord.audioCodecMutex.enter();
-
- // Return the size of data in samples
- int inSamples = _audioRtpRecord._audioCodec->decode(spkrDataDecoded , spkrData , size);
-
- _audioRtpRecord.audioCodecMutex.leave();
+ int inSamples;
+ {
+ ost::MutexLock lock(audioRtpRecord_.audioCodecMutex_);
+ // Return the size of data in samples
+ inSamples = audioRtpRecord_.audioCodec_->decode(spkrDataDecoded , spkrData , size);
+ }
- fadeIn(spkrDataDecoded, inSamples, &_audioRtpRecord._micAmplFactor);
+ fadeIn(spkrDataDecoded, inSamples, &audioRtpRecord_.micAmplFactor_);
// Normalize incomming signal
gainController.process(spkrDataDecoded, inSamples);
@@ -202,7 +192,7 @@ void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, unsigned
if (codecSampleRate != mainBufferSampleRate) {
// Do sample rate conversion
outSamples = ((float) inSamples * ((float) mainBufferSampleRate / (float) codecSampleRate));
- _audioRtpRecord._converter->resample(spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, inSamples);
+ audioRtpRecord_.converter_->resample(spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, inSamples);
out = spkrDataConverted;
}
@@ -223,5 +213,4 @@ void AudioRtpRecordHandler::fadeIn(SFLDataFormat *audio, int size, SFLDataFormat
*factor /= FADEIN_STEP_SIZE;
}
-
}
diff --git a/daemon/src/audio/audiortp/audio_rtp_record_handler.h b/daemon/src/audio/audiortp/audio_rtp_record_handler.h
index b249a004b7feba12392f171d07c67fd3a08c8dcf..50f23a5a88cdc55dbf9d3fe48f0f3af6169ee451 100644
--- a/daemon/src/audio/audiortp/audio_rtp_record_handler.h
+++ b/daemon/src/audio/audiortp/audio_rtp_record_handler.h
@@ -78,24 +78,23 @@ class AudioRtpRecord {
AudioRtpRecord();
~AudioRtpRecord();
- AudioCodec *_audioCodec;
- ost::Mutex audioCodecMutex;
- int _codecPayloadType;
- bool _hasDynamicPayloadType;
- SFLDataFormat decData[DEC_BUFFER_SIZE];
- SFLDataFormat resampledData[DEC_BUFFER_SIZE];
- unsigned char encodedData[DEC_BUFFER_SIZE];
- SamplerateConverter *_converter;
- int _codecSampleRate;
- int _codecFrameSize;
- int _converterSamplingRate;
- std::list<int> _dtmfQueue;
- SFLDataFormat _micAmplFactor;
- NoiseSuppress *_noiseSuppress;
- ost::Mutex audioProcessMutex;
- std::string _callId;
- unsigned int _dtmfPayloadType;
-
+ AudioCodec *audioCodec_;
+ ost::Mutex audioCodecMutex_;
+ int codecPayloadType_;
+ bool hasDynamicPayloadType_;
+ SFLDataFormat decData_[DEC_BUFFER_SIZE];
+ SFLDataFormat resampledData_[DEC_BUFFER_SIZE];
+ unsigned char encodedData_[DEC_BUFFER_SIZE];
+ SamplerateConverter *converter_;
+ int codecSampleRate_;
+ int codecFrameSize_;
+ int converterSamplingRate_;
+ std::list<int> dtmfQueue_;
+ SFLDataFormat micAmplFactor_;
+ NoiseSuppress *noiseSuppress_;
+ ost::Mutex audioProcessMutex_;
+ std::string callId_;
+ unsigned int dtmfPayloadType_;
};
@@ -107,45 +106,44 @@ class AudioRtpRecordHandler {
/**
* Set rtp media for this session
*/
-
void setRtpMedia(AudioCodec* audioCodec);
- AudioCodec *getAudioCodec(void) const {
- return _audioRtpRecord._audioCodec;
+ AudioCodec *getAudioCodec() const {
+ return audioRtpRecord_.audioCodec_;
}
- int getCodecPayloadType(void) const {
- return _audioRtpRecord._codecPayloadType;
+ int getCodecPayloadType() const {
+ return audioRtpRecord_.codecPayloadType_;
}
- int getCodecSampleRate(void) const {
- return _audioRtpRecord._codecSampleRate;
+ int getCodecSampleRate() const {
+ return audioRtpRecord_.codecSampleRate_;
}
- int getCodecFrameSize(void) const {
- return _audioRtpRecord._codecFrameSize;
+ int getCodecFrameSize() const {
+ return audioRtpRecord_.codecFrameSize_;
}
- bool getHasDynamicPayload(void) const {
- return _audioRtpRecord._hasDynamicPayloadType;
+ bool getHasDynamicPayload() const {
+ return audioRtpRecord_.hasDynamicPayloadType_;
}
- int DtmfPending(void) const {
- return _audioRtpRecord._dtmfQueue.size() > 0;
+ int DtmfPending() const {
+ return audioRtpRecord_.dtmfQueue_.size() > 0;
}
- const unsigned char *getMicDataEncoded(void) const {
- return _audioRtpRecord.encodedData;
+ const unsigned char *getMicDataEncoded() const {
+ return audioRtpRecord_.encodedData_;
}
- void initBuffers(void);
+ void initBuffers();
- void initNoiseSuppress(void);
+ void initNoiseSuppress();
/**
* Encode audio data from mainbuffer
*/
- int processDataEncode(void);
+ int processDataEncode();
/**
* Decode audio data received from peer
@@ -158,18 +156,18 @@ class AudioRtpRecordHandler {
void fadeIn(SFLDataFormat *audio, int size, SFLDataFormat *factor);
void setDtmfPayloadType(unsigned int payloadType) {
- _audioRtpRecord._dtmfPayloadType = payloadType;
+ audioRtpRecord_.dtmfPayloadType_ = payloadType;
}
- unsigned int getDtmfPayloadType(void) const {
- return _audioRtpRecord._dtmfPayloadType;
+ unsigned int getDtmfPayloadType() const {
+ return audioRtpRecord_.dtmfPayloadType_;
}
void putDtmfEvent(int digit);
protected:
- AudioRtpRecord _audioRtpRecord;
+ AudioRtpRecord audioRtpRecord_;
private:
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.cpp b/daemon/src/audio/audiortp/audio_rtp_session.cpp
index aaf292c938b16effcf3d9ae6f36cc061f8f7b41c..201731872caa08e3ddebd5d3f51043bfa32de183 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_session.cpp
@@ -36,6 +36,7 @@
#include "audio_symmetric_rtp_session.h"
#include "sip/sdp.h"
+#include "sip/sipcall.h"
#include "audio/audiolayer.h"
#include <ccrtp/rtp.h>
#include <ccrtp/oqueue.h>
@@ -44,33 +45,33 @@
namespace sfl {
AudioRtpSession::AudioRtpSession(SIPCall * sipcall, RtpMethod type, ost::RTPDataQueue *queue, ost::Thread *thread) :
AudioRtpRecordHandler(sipcall)
- , _ca(sipcall)
- , _type(type)
- , _timestamp(0)
- , _timestampIncrement(0)
- , _timestampCount(0)
- , _isStarted(false)
- , _queue(queue)
- , _thread(thread)
+ , ca_(sipcall)
+ , type_(type)
+ , timestamp_(0)
+ , timestampIncrement_(0)
+ , timestampCount_(0)
+ , isStarted_(false)
+ , queue_(queue)
+ , thread_(thread)
{
- assert(_ca);
- _queue->setTypeOfService(ost::RTPDataQueue::tosEnhanced);
+ assert(ca_);
+ queue_->setTypeOfService(ost::RTPDataQueue::tosEnhanced);
}
AudioRtpSession::~AudioRtpSession()
{
- _queue->disableStack();
+ queue_->disableStack();
}
void AudioRtpSession::updateSessionMedia(AudioCodec *audioCodec)
{
- int lastSamplingRate = _audioRtpRecord._codecSampleRate;
+ int lastSamplingRate = audioRtpRecord_.codecSampleRate_;
setSessionMedia(audioCodec);
- Manager::instance().audioSamplingRateChanged(_audioRtpRecord._codecSampleRate);
+ Manager::instance().audioSamplingRateChanged(audioRtpRecord_.codecSampleRate_);
- if (lastSamplingRate != _audioRtpRecord._codecSampleRate) {
+ if (lastSamplingRate != audioRtpRecord_.codecSampleRate_) {
_debug("AudioRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize());
initNoiseSuppress();
}
@@ -89,65 +90,65 @@ void AudioRtpSession::setSessionMedia(AudioCodec *audioCodec)
// G722 requires timestamp to be incremented at 8kHz
if (payloadType == g722PayloadType)
- _timestampIncrement = g722RtpTimeincrement;
+ timestampIncrement_ = g722RtpTimeincrement;
else
- _timestampIncrement = frameSize;
+ timestampIncrement_ = frameSize;
_debug("AudioRptSession: Codec payload: %d", payloadType);
_debug("AudioSymmetricRtpSession: Codec sampling rate: %d", smplRate);
_debug("AudioSymmetricRtpSession: Codec frame size: %d", frameSize);
- _debug("AudioSymmetricRtpSession: RTP timestamp increment: %d", _timestampIncrement);
+ _debug("AudioSymmetricRtpSession: RTP timestamp increment: %d", timestampIncrement_);
if (payloadType == g722PayloadType) {
_debug("AudioSymmetricRtpSession: Setting G722 payload format");
- _queue->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, g722RtpClockRate));
+ queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, g722RtpClockRate));
} else {
if (dynamic) {
_debug("AudioSymmetricRtpSession: Setting dynamic payload format");
- _queue->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, smplRate));
+ queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, smplRate));
} else {
_debug("AudioSymmetricRtpSession: Setting static payload format");
- _queue->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) payloadType));
+ queue_->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) payloadType));
}
}
- if (_type != Zrtp)
- _ca->setRecordingSmplRate(getCodecSampleRate());
+ if (type_ != Zrtp)
+ ca_->setRecordingSmplRate(getCodecSampleRate());
}
void AudioRtpSession::sendDtmfEvent()
{
ost::RTPPacket::RFC2833Payload payload;
- payload.event = _audioRtpRecord._dtmfQueue.front();
+ payload.event = audioRtpRecord_.dtmfQueue_.front();
payload.ebit = false; // end of event bit
payload.rbit = false; // reserved bit
payload.duration = 1; // duration for this event
- _audioRtpRecord._dtmfQueue.pop_front();
+ audioRtpRecord_.dtmfQueue_.pop_front();
_debug("AudioRtpSession: Send RTP Dtmf (%d)", payload.event);
- _timestamp += (_type == Zrtp) ? 160 : _timestampIncrement;
+ timestamp_ += (type_ == Zrtp) ? 160 : timestampIncrement_;
// discard equivalent size of audio
processDataEncode();
// change Payload type for DTMF payload
- _queue->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) getDtmfPayloadType(), 8000));
+ queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) getDtmfPayloadType(), 8000));
- _queue->setMark(true);
- _queue->sendImmediate(_timestamp, (const unsigned char *)(&payload), sizeof(payload));
- _queue->setMark(false);
+ queue_->setMark(true);
+ queue_->sendImmediate(timestamp_, (const unsigned char *)(&payload), sizeof(payload));
+ queue_->setMark(false);
// get back the payload to audio
- _queue->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) getCodecPayloadType()));
+ queue_->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) getCodecPayloadType()));
}
void AudioRtpSession::receiveSpeakerData()
{
- const ost::AppDataUnit* adu = _queue->getData(_queue->getFirstTimestamp());
+ const ost::AppDataUnit* adu = queue_->getData(queue_->getFirstTimestamp());
if (!adu)
return;
@@ -173,57 +174,57 @@ void AudioRtpSession::sendMicData()
return;
// Increment timestamp for outgoing packet
- _timestamp += _timestampIncrement;
+ timestamp_ += timestampIncrement_;
- if (_type == Zrtp)
- _queue->putData(_timestamp, getMicDataEncoded(), compSize);
+ if (type_ == Zrtp)
+ queue_->putData(timestamp_, getMicDataEncoded(), compSize);
// putData put the data on RTP queue, sendImmediate bypass this queue
- _queue->sendImmediate(_timestamp, getMicDataEncoded(), compSize);
+ queue_->sendImmediate(timestamp_, getMicDataEncoded(), compSize);
}
-void AudioRtpSession::setSessionTimeouts(void)
+void AudioRtpSession::setSessionTimeouts()
{
_debug("AudioRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
- _queue->setSchedulingTimeout(sfl::schedulingTimeout);
- _queue->setExpireTimeout(sfl::expireTimeout);
+ queue_->setSchedulingTimeout(sfl::schedulingTimeout);
+ queue_->setExpireTimeout(sfl::expireTimeout);
}
-void AudioRtpSession::setDestinationIpAddress(void)
+void AudioRtpSession::setDestinationIpAddress()
{
_info("AudioRtpSession: Setting IP address for the RTP session");
// Store remote ip in case we would need to forget current destination
- _remote_ip = ost::InetHostAddress(_ca->getLocalSDP()->getRemoteIP().c_str());
+ remote_ip_ = ost::InetHostAddress(ca_->getLocalSDP()->getRemoteIP().c_str());
- if (!_remote_ip) {
+ if (!remote_ip_) {
_warn("AudioRtpSession: Target IP address (%s) is not correct!",
- _ca->getLocalSDP()->getRemoteIP().data());
+ ca_->getLocalSDP()->getRemoteIP().data());
return;
}
// Store remote port in case we would need to forget current destination
- _remote_port = (unsigned short) _ca->getLocalSDP()->getRemoteAudioPort();
+ remote_port_ = (unsigned short) ca_->getLocalSDP()->getRemoteAudioPort();
_info("AudioRtpSession: New remote address for session: %s:%d",
- _ca->getLocalSDP()->getRemoteIP().data(), _remote_port);
+ ca_->getLocalSDP()->getRemoteIP().data(), remote_port_);
- if (!_queue->addDestination(_remote_ip, _remote_port)) {
+ if (!queue_->addDestination(remote_ip_, remote_port_)) {
_warn("AudioRtpSession: Can't add new destination to session!");
return;
}
}
-void AudioRtpSession::updateDestinationIpAddress(void)
+void AudioRtpSession::updateDestinationIpAddress()
{
_debug("AudioRtpSession: Update destination ip address");
// Destination address are stored in a list in ccrtp
// This method remove the current destination entry
- if (!_queue->forgetDestination(_remote_ip, _remote_port, _remote_port+1))
+ if (!queue_->forgetDestination(remote_ip_, remote_port_, remote_port_ + 1))
_debug("AudioRtpSession: Did not remove previous destination");
// new destination is stored in call
@@ -234,21 +235,21 @@ void AudioRtpSession::updateDestinationIpAddress(void)
int AudioRtpSession::startRtpThread(AudioCodec* audiocodec)
{
- if (_isStarted)
+ if (isStarted_)
return 0;
_debug("AudioSymmetricRtpSession: Starting main thread");
- _isStarted = true;
+ isStarted_ = true;
setSessionTimeouts();
setSessionMedia(audiocodec);
initBuffers();
initNoiseSuppress();
- _queue->enableStack();
- int ret = _thread->start();
+ queue_->enableStack();
+ int ret = thread_->start();
- if (_type == Zrtp)
+ if (type_ == Zrtp)
return ret;
AudioSymmetricRtpSession *self = dynamic_cast<AudioSymmetricRtpSession*>(this);
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.h b/daemon/src/audio/audiortp/audio_rtp_session.h
index 64ebd2cbd7da276150c7f1e8698b8903743f0862..35733fa49141217f89e5e208d9f49b217ae7504f 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_rtp_session.h
@@ -63,7 +63,7 @@ class AudioRtpSession : public AudioRtpRecordHandler {
virtual ~AudioRtpSession();
RtpMethod getAudioRtpType() {
- return _type;
+ return type_;
}
void updateSessionMedia(AudioCodec *audioCodec);
@@ -72,7 +72,7 @@ class AudioRtpSession : public AudioRtpRecordHandler {
/**
* Used mostly when receiving a reinvite
*/
- void updateDestinationIpAddress(void);
+ void updateDestinationIpAddress();
protected:
@@ -91,9 +91,9 @@ class AudioRtpSession : public AudioRtpRecordHandler {
*/
void sendMicData();
- SIPCall *_ca;
+ SIPCall *ca_;
- RtpMethod _type;
+ RtpMethod type_;
private:
@@ -105,13 +105,12 @@ class AudioRtpSession : public AudioRtpRecordHandler {
/**
* Set RTP Sockets send/receive timeouts
*/
- void setSessionTimeouts(void);
+ void setSessionTimeouts();
/**
* Retreive destination address for this session. Stored in CALL
*/
- void setDestinationIpAddress(void);
-
+ void setDestinationIpAddress();
/**
* Receive data from peer
@@ -121,35 +120,34 @@ class AudioRtpSession : public AudioRtpRecordHandler {
// Main destination address for this rtp session.
// Stored in case or reINVITE, which may require to forget
// this destination and update a new one.
- ost::InetHostAddress _remote_ip;
+ ost::InetHostAddress remote_ip_;
// Main destination port for this rtp session.
// Stored in case reINVITE, which may require to forget
// this destination and update a new one
- unsigned short _remote_port;
+ unsigned short remote_port_;
/**
* Timestamp for this session
*/
- int _timestamp;
+ int timestamp_;
/**
* Timestamp incrementation value based on codec period length (framesize)
* except for G722 which require a 8 kHz incrementation.
*/
- int _timestampIncrement;
+ int timestampIncrement_;
/**
* Timestamp reset frequency specified in number of packet sent
*/
- short _timestampCount;
-
+ short timestampCount_;
- bool _isStarted;
+ bool isStarted_;
- ost::RTPDataQueue *_queue;
+ ost::RTPDataQueue *queue_;
- ost::Thread *_thread;
+ ost::Thread *thread_;
};
}
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.cpp b/daemon/src/audio/audiortp/audio_srtp_session.cpp
index 01e9173411f8392735db8256ac85000ae4f4d822..fee63e9bc5250b44e6bf480480ba5312f4163d11 100644
--- a/daemon/src/audio/audiortp/audio_srtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_srtp_session.cpp
@@ -46,17 +46,17 @@ namespace sfl {
AudioSrtpSession::AudioSrtpSession(SIPCall * sipcall) :
AudioSymmetricRtpSession(sipcall),
- _remoteCryptoCtx(NULL),
- _localCryptoCtx(NULL),
- _localCryptoSuite(0),
- _remoteCryptoSuite(0),
- _localMasterKeyLength(0),
- _localMasterSaltLength(0),
- _remoteMasterKeyLength(0),
- _remoteMasterSaltLength(0),
- _remoteOfferIsSet(false)
+ remoteCryptoCtx_(NULL),
+ localCryptoCtx_(NULL),
+ localCryptoSuite_(0),
+ remoteCryptoSuite_(0),
+ localMasterKeyLength_(0),
+ localMasterSaltLength_(0),
+ remoteMasterKeyLength_(0),
+ remoteMasterSaltLength_(0),
+ remoteOfferIsSet_(false)
{
- this->_type = Sdes;
+ type_ = Sdes;
}
AudioSrtpSession::~AudioSrtpSession()
@@ -74,10 +74,9 @@ void AudioSrtpSession::initLocalCryptoInfo()
initializeLocalCryptoContext();
// Set local crypto context in ccrtp
- _localCryptoCtx->deriveSrtpKeys(0);
-
- setOutQueueCryptoContext(_localCryptoCtx);
+ localCryptoCtx_->deriveSrtpKeys(0);
+ setOutQueueCryptoContext(localCryptoCtx_);
}
std::vector<std::string> AudioSrtpSession::getLocalCryptoInfo()
@@ -91,7 +90,7 @@ std::vector<std::string> AudioSrtpSession::getLocalCryptoInfo()
// cryptographic context tagged 1, 2, 3...
std::string tag = "1";
- std::string crypto_suite = sfl::CryptoSuites[_localCryptoSuite].name;
+ std::string crypto_suite = sfl::CryptoSuites[localCryptoSuite_].name;
// srtp keys formated as the following as the following
// inline:keyParameters|keylifetime|MasterKeyIdentifier
@@ -112,18 +111,16 @@ std::vector<std::string> AudioSrtpSession::getLocalCryptoInfo()
return crypto_vector;
}
-
void AudioSrtpSession::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
{
- if (!_remoteOfferIsSet) {
-
+ if (not remoteOfferIsSet_) {
_debug("%s", nego.getKeyInfo().c_str());
// Use second crypto suite if key length is 32 bit, default is 80;
if (nego.getAuthTagLength() == "32") {
- _localCryptoSuite = 1;
- _remoteCryptoSuite = 1;
+ localCryptoSuite_ = 1;
+ remoteCryptoSuite_ = 1;
}
// decode keys
@@ -131,85 +128,76 @@ void AudioSrtpSession::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
// init crypto content in Srtp session
initializeRemoteCryptoContext();
- setInQueueCryptoContext(_remoteCryptoCtx);
+ setInQueueCryptoContext(remoteCryptoCtx_);
// initLocalCryptoInfo();
- _remoteOfferIsSet = true;
+ remoteOfferIsSet_ = true;
}
-
}
-
void AudioSrtpSession::initializeLocalMasterKey(void)
{
_debug("AudioSrtp: Init local master key");
// @TODO key may have different length depending on cipher suite
- _localMasterKeyLength = sfl::CryptoSuites[_localCryptoSuite].masterKeyLength / 8;
+ localMasterKeyLength_ = sfl::CryptoSuites[localCryptoSuite_].masterKeyLength / 8;
- _debug("AudioSrtp: Local master key length %d", _localMasterKeyLength);
+ _debug("AudioSrtp: Local master key length %d", localMasterKeyLength_);
// Allocate memory for key
- unsigned char *random_key = new unsigned char[_localMasterKeyLength];
+ unsigned char *random_key = new unsigned char[localMasterKeyLength_];
// Generate ryptographically strong pseudo-random bytes
int err;
- if ((err = RAND_bytes(random_key, _localMasterKeyLength)) != 1)
+ if ((err = RAND_bytes(random_key, localMasterKeyLength_)) != 1)
_debug("Error occured while generating cryptographically strong pseudo-random key");
- memcpy(_localMasterKey, random_key, _localMasterKeyLength);
-
+ memcpy(localMasterKey_, random_key, localMasterKeyLength_);
}
-
void AudioSrtpSession::initializeLocalMasterSalt(void)
{
-
// @TODO key may have different length depending on cipher suite
- _localMasterSaltLength = sfl::CryptoSuites[_localCryptoSuite].masterSaltLength / 8;
+ localMasterSaltLength_ = sfl::CryptoSuites[localCryptoSuite_].masterSaltLength / 8;
// Allocate memory for key
- unsigned char *random_key = new unsigned char[_localMasterSaltLength];
+ unsigned char *random_key = new unsigned char[localMasterSaltLength_];
- _debug("AudioSrtp: Local master salt length %d", _localMasterSaltLength);
+ _debug("AudioSrtp: Local master salt length %d", localMasterSaltLength_);
// Generate ryptographically strong pseudo-random bytes
int err;
- if ((err = RAND_bytes(random_key, _localMasterSaltLength)) != 1)
+ if ((err = RAND_bytes(random_key, localMasterSaltLength_)) != 1)
_debug("Error occured while generating cryptographically strong pseudo-random key");
- memcpy(_localMasterSalt, random_key, _localMasterSaltLength);
-
-
+ memcpy(localMasterSalt_, random_key, localMasterSaltLength_);
}
-
std::string AudioSrtpSession::getBase64ConcatenatedKeys()
{
_debug("AudioSrtp: Get base64 concatenated keys");
// compute concatenated master and salt length
- int concatLength = _localMasterKeyLength + _localMasterSaltLength;
+ int concatLength = localMasterKeyLength_ + localMasterSaltLength_;
uint8 concatKeys[concatLength];
_debug("AudioSrtp: Concatenated length %d", concatLength);
// concatenate keys
- memcpy((void*) concatKeys, (void*) _localMasterKey, _localMasterKeyLength);
- memcpy((void*)(concatKeys + _localMasterKeyLength), (void*) _localMasterSalt, _localMasterSaltLength);
+ memcpy((void*) concatKeys, (void*) localMasterKey_, localMasterKeyLength_);
+ memcpy((void*)(concatKeys + localMasterKeyLength_), (void*) localMasterSalt_, localMasterSaltLength_);
// encode concatenated keys in base64
return encodeBase64((unsigned char*) concatKeys, concatLength);
}
-
void AudioSrtpSession::unBase64ConcatenatedKeys(std::string base64keys)
{
- _remoteMasterKeyLength = sfl::CryptoSuites[_remoteCryptoSuite].masterKeyLength / 8;
- _remoteMasterSaltLength = sfl::CryptoSuites[_remoteCryptoSuite].masterSaltLength / 8;
+ remoteMasterKeyLength_ = sfl::CryptoSuites[remoteCryptoSuite_].masterKeyLength / 8;
+ remoteMasterSaltLength_ = sfl::CryptoSuites[remoteCryptoSuite_].masterSaltLength / 8;
// pointer to binary data
char *dataptr = (char*) base64keys.data();
@@ -218,33 +206,32 @@ void AudioSrtpSession::unBase64ConcatenatedKeys(std::string base64keys)
char *output = decodeBase64((unsigned char*) dataptr, strlen(dataptr));
// copy master and slt respectively
- memcpy((void*) _remoteMasterKey, (void*) output, _remoteMasterKeyLength);
- memcpy((void*) _remoteMasterSalt, (void*)(output + _remoteMasterKeyLength), _remoteMasterSaltLength);
+ memcpy((void*) remoteMasterKey_, (void*) output, remoteMasterKeyLength_);
+ memcpy((void*) remoteMasterSalt_, (void*)(output + remoteMasterKeyLength_), remoteMasterSaltLength_);
delete[] output;
}
-
void AudioSrtpSession::initializeRemoteCryptoContext(void)
{
_debug("AudioSrtp: Initialize remote crypto context");
- CryptoSuiteDefinition crypto = sfl::CryptoSuites[_remoteCryptoSuite];
+ CryptoSuiteDefinition crypto = sfl::CryptoSuites[remoteCryptoSuite_];
- if (_remoteCryptoCtx) {
- delete _remoteCryptoCtx;
- _remoteCryptoCtx = NULL;
+ if (remoteCryptoCtx_) {
+ delete remoteCryptoCtx_;
+ remoteCryptoCtx_ = NULL;
}
- _remoteCryptoCtx = new ost::CryptoContext(0x0,
+ remoteCryptoCtx_ = new ost::CryptoContext(0x0,
0, // roc,
0L, // keydr,
SrtpEncryptionAESCM, // encryption algo
SrtpAuthenticationSha1Hmac, // authtication algo
- _remoteMasterKey,
- _remoteMasterKeyLength,
- _remoteMasterSalt,
- _remoteMasterSaltLength,
+ remoteMasterKey_,
+ remoteMasterKeyLength_,
+ remoteMasterSalt_,
+ remoteMasterSaltLength_,
crypto.encryptionKeyLength / 8,
crypto.srtpAuthKeyLength / 8,
crypto.masterSaltLength / 8, // session salt len
@@ -256,27 +243,26 @@ void AudioSrtpSession::initializeLocalCryptoContext(void)
{
_debug("AudioSrtp: Initialize local crypto context");
- CryptoSuiteDefinition crypto = sfl::CryptoSuites[_localCryptoSuite];
+ CryptoSuiteDefinition crypto = sfl::CryptoSuites[localCryptoSuite_];
- if (_localCryptoCtx) {
- delete _localCryptoCtx;
- _localCryptoCtx = NULL;
+ if (localCryptoCtx_) {
+ delete localCryptoCtx_;
+ localCryptoCtx_ = NULL;
}
- _localCryptoCtx = new ost::CryptoContext(OutgoingDataQueue::getLocalSSRC(),
+ localCryptoCtx_ = new ost::CryptoContext(OutgoingDataQueue::getLocalSSRC(),
0, // roc,
0L, // keydr,
SrtpEncryptionAESCM, // encryption algo
SrtpAuthenticationSha1Hmac, // authtication algo
- _localMasterKey,
- _localMasterKeyLength,
- _localMasterSalt,
- _localMasterSaltLength,
+ localMasterKey_,
+ localMasterKeyLength_,
+ localMasterSalt_,
+ localMasterSaltLength_,
crypto.encryptionKeyLength / 8,
crypto.srtpAuthKeyLength / 8,
crypto.masterSaltLength / 8, // session salt len
crypto.srtpAuthTagLength / 8);
-
}
void AudioSrtpSession::restoreCryptoContext(ost::CryptoContext *localContext, ost::CryptoContext *remoteContext)
@@ -336,5 +322,4 @@ char* AudioSrtpSession::decodeBase64(unsigned char *input, int length)
return buffer;
}
-
}
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.h b/daemon/src/audio/audiortp/audio_srtp_session.h
index b4df5643550cde80866d7c1a7b1a4f0894cf4462..28d822caa4c226e5d464fb0bbe5842675bff425c 100644
--- a/daemon/src/audio/audiortp/audio_srtp_session.h
+++ b/daemon/src/audio/audiortp/audio_srtp_session.h
@@ -81,7 +81,7 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
* method must be called befor setRemoteCryptoInfo in case of an
* outgoing call or after in case of an outgoing call.
*/
- std::vector<std::string> getLocalCryptoInfo(void);
+ std::vector<std::string> getLocalCryptoInfo();
/**
* Set remote crypto header from incoming sdp offer
@@ -93,7 +93,7 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
* this method must be called before sending first Invite request
* with SDP offer.
*/
- void initLocalCryptoInfo(void);
+ void initLocalCryptoInfo();
/**
* Restore the cryptographic context. most likely useful to restore
@@ -103,10 +103,10 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
/** Remote srtp crypto context to be set into incoming data queue. */
- ost::CryptoContext* _remoteCryptoCtx;
+ ost::CryptoContext* remoteCryptoCtx_;
/** Local srtp crypto context to be set into outgoing data queue. */
- ost::CryptoContext* _localCryptoCtx;
+ ost::CryptoContext* localCryptoCtx_;
private:
@@ -114,25 +114,25 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
* Init local master key according to current crypto context
* as defined in SdesNegotiator.h
*/
- void initializeLocalMasterKey(void);
+ void initializeLocalMasterKey();
/**
* Init local master salt according to current crypto context
* as defined in SdesNegotiator.h
*/
- void initializeLocalMasterSalt(void);
+ void initializeLocalMasterSalt();
/**
* Init remote crypto context in audio srtp session. This method
* must be called after unBase64ConcatenatedKeys.
*/
- void initializeRemoteCryptoContext(void);
+ void initializeRemoteCryptoContext();
/**
* Init local crypto context in audio srtp session. Make sure remote
* crypto context is set before calling this method for incoming calls.
*/
- void initializeLocalCryptoContext(void);
+ void initializeLocalCryptoContext();
/**
* Used to generate local keys to be included in SDP offer/answer.
@@ -155,33 +155,33 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
char* decodeBase64(unsigned char *input, int length);
/** Default local crypto suite is AES_CM_128_HMAC_SHA1_80*/
- int _localCryptoSuite;
+ int localCryptoSuite_;
/** Remote crypto suite is initialized at AES_CM_128_HMAC_SHA1_80*/
- int _remoteCryptoSuite;
+ int remoteCryptoSuite_;
- uint8 _localMasterKey[16];
+ uint8 localMasterKey_[16];
/** local master key length in byte */
- int _localMasterKeyLength;
+ int localMasterKeyLength_;
- uint8 _localMasterSalt[14];
+ uint8 localMasterSalt_[14];
/** local master salt length in byte */
- int _localMasterSaltLength;
+ int localMasterSaltLength_;
- uint8 _remoteMasterKey[16];
+ uint8 remoteMasterKey_[16];
/** remote master key length in byte */
- int _remoteMasterKeyLength;
+ int remoteMasterKeyLength_;
- uint8 _remoteMasterSalt[14];
+ uint8 remoteMasterSalt_[14];
/** remote master salt length in byte */
- int _remoteMasterSaltLength;
+ int remoteMasterSaltLength_;
/** Used to make sure remote crypto context not initialized wice. */
- bool _remoteOfferIsSet;
+ bool remoteOfferIsSet_;
};
}
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
index 1d5f57a262b39d7b9bdf48df92111fbad9182489..71ad81f6ea1fee8e434018105181d606f9691f16 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
@@ -36,6 +36,7 @@
#include "audio_symmetric_rtp_session.h"
#include "audio_rtp_record_handler.h"
#include "sip/sdp.h"
+#include "sip/sipcall.h"
#include "audio/audiolayer.h"
namespace sfl {
@@ -43,30 +44,20 @@ namespace sfl {
AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall * sipcall) :
ost::SymmetricRTPSession(ost::InetHostAddress(sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort())
, AudioRtpSession(sipcall, Symmetric, this, this)
- , _rtpThread(new AudioRtpThread(this))
+ , rtpThread_(new AudioRtpThread(this))
{
- _info("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort());
-
- _audioRtpRecord._callId = _ca->getCallId();
+ _info("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
+ audioRtpRecord_.callId_ = ca_->getCallId();
}
AudioSymmetricRtpSession::~AudioSymmetricRtpSession()
{
- _info("AudioSymmetricRtpSession: Delete AudioSymmetricRtpSession instance");
-
- _rtpThread->running = false;
- delete _rtpThread;
+ rtpThread_->running = false;
+ delete rtpThread_;
}
AudioSymmetricRtpSession::AudioRtpThread::AudioRtpThread(AudioSymmetricRtpSession *session) : running(true), rtpSession(session)
-{
- _debug("AudioSymmetricRtpSession: Create new rtp thread");
-}
-
-AudioSymmetricRtpSession::AudioRtpThread::~AudioRtpThread()
-{
- _debug("AudioSymmetricRtpSession: Delete rtp thread");
-}
+{}
void AudioSymmetricRtpSession::AudioRtpThread::run()
{
@@ -77,7 +68,6 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
_debug("AudioRtpThread: Entering Audio rtp thread main loop");
while (running) {
-
// Send session
if (rtpSession->DtmfPending())
rtpSession->sendDtmfEvent();
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
index 8eee8ab6defb3669d1312cb876b4f0f90a1c825d..26fdcf5f46418895dd116b39e78633027d297297 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
@@ -34,25 +34,19 @@
#ifndef __AUDIO_SYMMETRIC_RTP_SESSION_H__
#define __AUDIO_SYMMETRIC_RTP_SESSION_H__
-#include <iostream>
#include <exception>
-#include <list>
#include <cassert>
#include <cstddef>
-#include "global.h"
-
#include "audio_rtp_session.h"
-#include "audio_rtp_record_handler.h"
-#include "sip/sipcall.h"
-#include "audio/codecs/audiocodec.h"
using std::ptrdiff_t;
#include <ccrtp/rtp.h>
#include <ccrtp/iqueue.h>
#include <cc++/numbers.h> // ost::Time
-#include <fstream>
+class SIPCall;
+
namespace sfl {
class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTPSession, public AudioRtpSession {
@@ -69,9 +63,9 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
return AudioRtpSession::onRTPPacketRecv(pkt);
}
- int startSymmetricRtpThread(void) {
- assert(_rtpThread);
- return _rtpThread->start();
+ int startSymmetricRtpThread() {
+ assert(rtpThread_);
+ return rtpThread_->start();
}
private:
@@ -79,23 +73,25 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
class AudioRtpThread : public ost::Thread, public ost::TimerPort {
public:
AudioRtpThread(AudioSymmetricRtpSession *session);
- ~AudioRtpThread();
+ ~AudioRtpThread(){}
virtual void run();
bool running;
private:
+ AudioRtpThread(const AudioRtpThread &);
+ AudioRtpThread& operator=(const AudioRtpThread &);
+
AudioSymmetricRtpSession *rtpSession;
};
SpeexEchoCancel echoCanceller;
protected:
- AudioRtpThread *_rtpThread;
+ AudioRtpThread *rtpThread_;
public:
-
friend class AudioRtpThread;
};
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.cpp b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
index a09d8256c83edf4323fcc453bc3cb0fb3a665f76..6dbc9751c95cf6fb2053851fb28aa652e65c441d 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
@@ -56,28 +56,25 @@ AudioZrtpSession::AudioZrtpSession(SIPCall * sipcall, const std::string& zidFile
0,
ost::MembershipBookkeeping::defaultMembersHashSize,
ost::defaultApplication()),
- _zidFilename(zidFilename)
+ zidFilename_(zidFilename)
{
_debug("AudioZrtpSession initialized");
initializeZid();
setCancel(cancelDefault);
- _info("AudioZrtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort());
+ _info("AudioZrtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
}
AudioZrtpSession::~AudioZrtpSession()
{
- _debug("AudioZrtpSession: Delete AudioSymmetricRtpSession instance");
-
try {
terminate();
} catch (...) {
- _debug("AudioZrtpSession: Thread destructor didn't terminate correctly");
throw;
}
- Manager::instance().getMainBuffer()->unBindAll(_ca->getCallId());
+ Manager::instance().getMainBuffer()->unBindAll(ca_->getCallId());
}
void AudioZrtpSession::final()
@@ -85,23 +82,21 @@ void AudioZrtpSession::final()
delete this;
}
-void AudioZrtpSession::initializeZid(void)
+void AudioZrtpSession::initializeZid()
{
-
- if (_zidFilename.empty()) {
+ if (zidFilename_.empty())
throw ZrtpZidException("zid filename empty");
- }
std::string zidCompleteFilename;
// xdg_config = std::string (HOMEDIR) + DIR_SEPARATOR_STR + ".cache/sflphone";
- std::string xdg_config = std::string(HOMEDIR) + DIR_SEPARATOR_STR + ".cache" + DIR_SEPARATOR_STR + PACKAGE + "/" + _zidFilename;
+ std::string xdg_config = std::string(HOMEDIR) + DIR_SEPARATOR_STR + ".cache" + DIR_SEPARATOR_STR + PACKAGE + "/" + zidFilename_;
_debug(" xdg_config %s", xdg_config.c_str());
if (XDG_CACHE_HOME != NULL) {
- std::string xdg_env = std::string(XDG_CACHE_HOME) + _zidFilename;
+ std::string xdg_env = std::string(XDG_CACHE_HOME) + zidFilename_;
_debug(" xdg_env %s", xdg_env.c_str());
(xdg_env.length() > 0) ? zidCompleteFilename = xdg_env : zidCompleteFilename = xdg_config;
} else
@@ -111,40 +106,32 @@ void AudioZrtpSession::initializeZid(void)
if (initialize(zidCompleteFilename.c_str()) >= 0) {
_debug("Register callbacks");
setEnableZrtp(true);
- setUserCallback(new ZrtpSessionCallback(_ca));
+ setUserCallback(new ZrtpSessionCallback(ca_));
return;
}
_debug("Initialization from ZID file failed. Trying to remove...");
- if (remove(zidCompleteFilename.c_str()) !=0) {
- _debug("Failed to remove zid file: %m");
+ if (remove(zidCompleteFilename.c_str()) != 0)
throw ZrtpZidException("zid file deletion failed");
- }
- if (initialize(zidCompleteFilename.c_str()) < 0) {
- _debug("ZRTP initialization failed");
+ if (initialize(zidCompleteFilename.c_str()) < 0)
throw ZrtpZidException("zid initialization failed");
- }
return;
}
void AudioZrtpSession::run()
{
-
// Set recording sampling rate
- _ca->setRecordingSmplRate(getCodecSampleRate());
-
- _debug("AudioZrtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
+ ca_->setRecordingSmplRate(getCodecSampleRate());
+ _debug("AudioZrtpSession: Entering mainloop for call %s", ca_->getCallId().c_str());
uint32 timeout = 0;
while (isActive()) {
-
- if (timeout < 1000) { // !(timeout/1000)
+ if (timeout < 1000)
timeout = getSchedulingTimeout();
- }
// Send session
if (DtmfPending())
@@ -171,19 +158,16 @@ void AudioZrtpSession::run()
if (isPendingData(timeout/1000)) {
setCancel(cancelDeferred);
- if (isActive()) { // take in only if active
+ if (isActive())
takeInDataPacket();
- }
setCancel(cancelImmediate);
}
-
timeout = 0;
}
-
}
- _debug("AudioZrtpSession: Left main loop for call %s", _ca->getCallId().c_str());
+ _debug("AudioZrtpSession: Left main loop for call %s", ca_->getCallId().c_str());
}
}
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.h b/daemon/src/audio/audiortp/audio_zrtp_session.h
index 920d10b3d2ea2303d41ea460bfebdc2298dc78d9..3a19a5e3fd148e2e9090965a0b058bb56296407f 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.h
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.h
@@ -48,7 +48,7 @@ namespace sfl {
class ZrtpZidException: public std::runtime_error {
public:
- ZrtpZidException(const std::string& str="") :
+ ZrtpZidException(const std::string& str = "") :
std::runtime_error("ZRTP ZID initialization failed." + str) {}
};
@@ -56,7 +56,6 @@ class ZrtpZidException: public std::runtime_error {
class AudioZrtpSession : public AudioRtpSession, protected ost::Thread, public ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue> {
public:
AudioZrtpSession(SIPCall * sipcall, const std::string& zidFilename);
-
~AudioZrtpSession();
virtual void final();
@@ -69,10 +68,8 @@ class AudioZrtpSession : public AudioRtpSession, protected ost::Thread, public o
}
private:
-
- void initializeZid(void);
-
- std::string _zidFilename;
+ void initializeZid();
+ std::string zidFilename_;
};
}
diff --git a/daemon/src/audio/dcblocker.cpp b/daemon/src/audio/dcblocker.cpp
index 7578347ff548e5622074982d959e98f3c0b004a1..3aa1169a36fa0ffbf0fe7f48df336a8ee6408643 100644
--- a/daemon/src/audio/dcblocker.cpp
+++ b/daemon/src/audio/dcblocker.cpp
@@ -30,27 +30,24 @@
#include "dcblocker.h"
-DcBlocker::DcBlocker() : _y(0), _x(0), _xm1(0), _ym1(0) {}
-
-DcBlocker::~DcBlocker() {}
-
void DcBlocker::reset()
{
- _y = 0;
- _x = 0;
- _xm1 = 0;
- _ym1 = 0;
+ y_ = 0;
+ x_ = 0;
+ xm1_ = 0;
+ ym1_ = 0;
}
void DcBlocker::process(SFLDataFormat *out, SFLDataFormat *in, int samples)
{
- for (int i = 0; i < samples; i++) {
- _x = in[i];
+ for (int i = 0; i < samples; ++i) {
+ x_ = in[i];
- _y = (SFLDataFormat)((float) _x - (float) _xm1 + 0.9999 * (float) _y);
- _xm1 = _x;
- _ym1 = _y;
+ y_ = (SFLDataFormat) ((float) x_ - (float) xm1_ + 0.9999 * (float) y_);
+ xm1_ = x_;
+ ym1_ = y_;
- out[i] = _y;
+ out[i] = y_;
}
}
+
diff --git a/daemon/src/audio/dcblocker.h b/daemon/src/audio/dcblocker.h
index 56bd15f56aa6a14cc9062badc2671ea3ccf041ff..6dbb073ea50330eef90b3308d20a99b1e2c4f524 100644
--- a/daemon/src/audio/dcblocker.h
+++ b/daemon/src/audio/dcblocker.h
@@ -34,20 +34,13 @@
#include "global.h"
class DcBlocker {
-
public:
-
- DcBlocker();
-
- ~DcBlocker();
-
- void reset(void);
-
+ void reset();
void process(SFLDataFormat *out, SFLDataFormat *in, int samples);
private:
- SFLDataFormat _y, _x, _xm1, _ym1;
+ SFLDataFormat y_, x_, xm1_, ym1_;
};
#endif
diff --git a/daemon/src/audio/delaydetection.cpp b/daemon/src/audio/delaydetection.cpp
index 1b6bdb9c8050b1c2398e7b418400d815cedcff4e..23837e0e61604293121a11a50a8973eb3cf6a427 100644
--- a/daemon/src/audio/delaydetection.cpp
+++ b/daemon/src/audio/delaydetection.cpp
@@ -35,8 +35,9 @@
#include <string.h>
#include <samplerate.h>
+namespace {
// decimation filter coefficient
-float decimationCoefs[] = {-0.09870257, 0.07473655, 0.05616626, 0.04448337, 0.03630817, 0.02944626,
+const float decimationCoefs[] = {-0.09870257, 0.07473655, 0.05616626, 0.04448337, 0.03630817, 0.02944626,
0.02244098, 0.01463477, 0.00610982, -0.00266367, -0.01120109, -0.01873722,
-0.02373243, -0.02602213, -0.02437806, -0.01869834, -0.00875287, 0.00500204,
0.02183252, 0.04065763, 0.06015944, 0.0788299, 0.09518543, 0.10799179,
@@ -50,78 +51,75 @@ std::vector<double> ird(decimationCoefs, decimationCoefs + sizeof(decimationCoef
// decimation filter coefficient
-float bandpassCoefs[] = {0.06278034, -0.0758545, -0.02274943, -0.0084497, 0.0702427, 0.05986113,
+const float bandpassCoefs[] = {0.06278034, -0.0758545, -0.02274943, -0.0084497, 0.0702427, 0.05986113,
0.06436469, -0.02412049, -0.03433526, -0.07568665, -0.03214543, -0.07236507,
-0.06979052, -0.12446371, -0.05530828, 0.00947243, 0.15294699, 0.17735563,
0.15294699, 0.00947243, -0.05530828, -0.12446371, -0.06979052, -0.07236507,
-0.03214543, -0.07568665, -0.03433526, -0.02412049, 0.06436469, 0.05986113,
0.0702427, -0.0084497, -0.02274943, -0.0758545, 0.06278034
};
-std::vector<double> irb(bandpassCoefs, bandpassCoefs + sizeof(bandpassCoefs) /sizeof(float));
+std::vector<double> irb(bandpassCoefs, bandpassCoefs + sizeof(bandpassCoefs) / sizeof(float));
+} // end anonymous namespace
-FirFilter::FirFilter(std::vector<double> ir) : _length(ir.size()),
- _impulseResponse(ir),
- _count(0)
+FirFilter::FirFilter(const std::vector<double> &ir) : length_(ir.size()),
+ impulseResponse_(ir),
+ counter_(0)
{
- memset(_taps, 0, sizeof(double) *MAXFILTERSIZE);
+ memset(taps_, 0, sizeof(double) * MAXFILTERSIZE);
}
-FirFilter::~FirFilter() {}
-
float FirFilter::getOutputSample(float inputSample)
{
- _taps[_count] = inputSample;
+ taps_[counter_] = inputSample;
double result = 0.0;
- int index = _count;
+ int index = counter_;
- for (int i = 0; i < _length; i++) {
- result = result + _impulseResponse[i] * _taps[index--];
+ for (int i = 0; i < length_; ++i) {
+ result = result + impulseResponse_[i] * taps_[index--];
if (index < 0)
- index = _length-1;
+ index = length_ - 1;
}
- _count++;
+ counter_++;
- if (_count >= _length)
- _count = 0;
+ if (counter_ >= length_)
+ counter_ = 0;
return result;
}
-void FirFilter::reset(void)
+void FirFilter::reset()
{
- for (int i = 0; i < _length; i++) {
- _impulseResponse[i] = 0.0;
- }
+ for (int i = 0; i < length_; ++i)
+ impulseResponse_[i] = 0.0;
}
-DelayDetection::DelayDetection() : _internalState(WaitForSpeaker), _decimationFilter(ird), _bandpassFilter(irb), _segmentSize(DELAY_BUFF_SIZE), _downsamplingFactor(8)
+DelayDetection::DelayDetection() : internalState_(WaitForSpeaker), decimationFilter_(ird), bandpassFilter_(irb), segmentSize_(DELAY_BUFF_SIZE), downsamplingFactor_(8)
{
- _micDownSize = WINDOW_SIZE / _downsamplingFactor;
- _spkrDownSize = DELAY_BUFF_SIZE / _downsamplingFactor;
+ micDownSize_ = WINDOW_SIZE / downsamplingFactor_;
+ spkrDownSize_ = DELAY_BUFF_SIZE / downsamplingFactor_;
- memset(_spkrReference, 0, sizeof(float) *WINDOW_SIZE*2);
- memset(_capturedData, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
- memset(_spkrReferenceDown, 0, sizeof(float) *WINDOW_SIZE*2);
- memset(_captureDataDown, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
- memset(_spkrReferenceFilter, 0, sizeof(float) *WINDOW_SIZE*2);
- memset(_captureDataFilter, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
- memset(_correlationResult, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
+ memset(spkrReference_, 0, sizeof(float) *WINDOW_SIZE*2);
+ memset(capturedData_, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
+ memset(spkrReferenceDown_, 0, sizeof(float) *WINDOW_SIZE*2);
+ memset(captureDataDown_, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
+ memset(spkrReferenceFilter_, 0, sizeof(float) *WINDOW_SIZE*2);
+ memset(captureDataFilter_, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
+ memset(correlationResult_, 0, sizeof(float) *DELAY_BUFF_SIZE*2);
}
void DelayDetection::putData(SFLDataFormat *inputData, int nbSamples)
{
// Machine may already got a spkr and is waiting for mic or computing correlation
- if (_nbSpkrSampleStored == WINDOW_SIZE)
+ if (nbSpkrSampleStored_ == WINDOW_SIZE)
return;
- if ((_nbSpkrSampleStored + nbSamples) > WINDOW_SIZE)
- nbSamples = WINDOW_SIZE - _nbSpkrSampleStored;
-
+ if ((nbSpkrSampleStored_ + nbSamples) > WINDOW_SIZE)
+ nbSamples = WINDOW_SIZE - nbSpkrSampleStored_;
if (nbSamples) {
@@ -129,67 +127,58 @@ void DelayDetection::putData(SFLDataFormat *inputData, int nbSamples)
float down[nbSamples];
convertInt16ToFloat32(inputData, tmp, nbSamples);
- memcpy(_spkrReference+_nbSpkrSampleStored, tmp, nbSamples*sizeof(float));
-
- downsampleData(tmp, down, nbSamples, _downsamplingFactor);
- bandpassFilter(down, nbSamples/_downsamplingFactor);
- memcpy(_spkrReferenceDown+ (_nbSpkrSampleStored/_downsamplingFactor), down, (nbSamples/_downsamplingFactor) *sizeof(float));
+ memcpy(spkrReference_ + nbSpkrSampleStored_, tmp, nbSamples * sizeof(float));
- _nbSpkrSampleStored += nbSamples;
+ downsampleData(tmp, down, nbSamples, downsamplingFactor_);
+ bandpassFilter(down, nbSamples / downsamplingFactor_);
+ memcpy(spkrReferenceDown_+ (nbSpkrSampleStored_ / downsamplingFactor_), down, (nbSamples / downsamplingFactor_) * sizeof(float));
+ nbSpkrSampleStored_ += nbSamples;
}
// Update the state
- _internalState = WaitForMic;
-
+ internalState_ = WaitForMic;
}
void DelayDetection::process(SFLDataFormat *inputData, int nbSamples)
{
- if (_internalState != WaitForMic)
+ if (internalState_ != WaitForMic)
return;
- if ((_nbMicSampleStored + nbSamples) > DELAY_BUFF_SIZE)
- nbSamples = DELAY_BUFF_SIZE - _nbMicSampleStored;
+ if ((nbMicSampleStored_ + nbSamples) > DELAY_BUFF_SIZE)
+ nbSamples = DELAY_BUFF_SIZE - nbMicSampleStored_;
if (nbSamples) {
float tmp[nbSamples];
float down[nbSamples];
convertInt16ToFloat32(inputData, tmp, nbSamples);
- memcpy(_capturedData+_nbMicSampleStored, tmp, nbSamples);
-
- downsampleData(tmp, down, nbSamples, _downsamplingFactor);
+ memcpy(capturedData_ + nbMicSampleStored_, tmp, nbSamples);
- memcpy(_captureDataDown+ (_nbMicSampleStored/_downsamplingFactor), down, (nbSamples/_downsamplingFactor) *sizeof(float));
+ downsampleData(tmp, down, nbSamples, downsamplingFactor_);
- _nbMicSampleStored += nbSamples;
+ memcpy(captureDataDown_ + (nbMicSampleStored_ / downsamplingFactor_), down, (nbSamples / downsamplingFactor_) * sizeof(float));
+ nbMicSampleStored_ += nbSamples;
}
- if (_nbMicSampleStored == DELAY_BUFF_SIZE)
- _internalState = ComputeCorrelation;
+ if (nbMicSampleStored_ == DELAY_BUFF_SIZE)
+ internalState_ = ComputeCorrelation;
else
return;
- _debug("_spkrDownSize: %d, _micDownSize: %d", _spkrDownSize, _micDownSize);
- crossCorrelate(_spkrReferenceDown, _captureDataDown, _correlationResult, _micDownSize, _spkrDownSize);
+ crossCorrelate(spkrReferenceDown_, captureDataDown_, correlationResult_, micDownSize_, spkrDownSize_);
- int maxIndex = getMaxIndex(_correlationResult, _spkrDownSize);
-
- _debug("MaxIndex: %d", maxIndex);
+ int maxIndex = getMaxIndex(correlationResult_, spkrDownSize_);
}
void DelayDetection::crossCorrelate(float *ref, float *seg, float *res, int refSize, int segSize)
{
-
- _debug("CrossCorrelate");
-
// Output has same size as the
int rsize = refSize;
int ssize = segSize;
- int tmpsize = segSize-refSize+1;
+ int tmpsize = segSize - refSize + 1;
// perform autocorrelation on reference signal
float acref = correlate(ref, ref, rsize);
@@ -215,13 +204,13 @@ void DelayDetection::crossCorrelate(float *ref, float *seg, float *res, int refS
while (rsize) {
acseg = correlate(seg, seg, rsize);
- res[ssize-1] = correlate(ref+i, seg, rsize);
- r = sqrt(acref*acseg);
+ res[ssize - 1] = correlate(ref + i, seg, rsize);
+ r = sqrt(acref * acseg);
if (r < 0.0001)
- res[ssize-1] = 0.0;
+ res[ssize - 1] = 0.0;
else
- res[ssize-1] = res[ssize-1] / r;
+ res[ssize - 1] = res[ssize-1] / r;
--rsize;
--ssize;
@@ -231,13 +220,12 @@ void DelayDetection::crossCorrelate(float *ref, float *seg, float *res, int refS
double DelayDetection::correlate(float *sig1, float *sig2, short size)
{
-
short s = size;
double ac = 0.0;
while (s--)
- ac += sig1[s]*sig2[s];
+ ac += sig1[s] * sig2[s];
return ac;
}
@@ -245,8 +233,7 @@ double DelayDetection::correlate(float *sig1, float *sig2, short size)
void DelayDetection::convertInt16ToFloat32(SFLDataFormat *input, float *output, int nbSamples)
{
-
-#define S2F_FACTOR .000030517578125f;
+ static const float S2F_FACTOR = .000030517578125f;
int len = nbSamples;
while (len) {
@@ -258,10 +245,8 @@ void DelayDetection::convertInt16ToFloat32(SFLDataFormat *input, float *output,
void DelayDetection::downsampleData(float *input, float *output, int nbSamples, int factor)
{
-
- int _src_err;
-
- SRC_STATE *_src_state = src_new(SRC_LINEAR, 1, &_src_err);
+ int src_err;
+ SRC_STATE *src_state = src_new(SRC_LINEAR, 1, &src_err);
double downfactor = 1.0 / (double) factor;
@@ -274,21 +259,20 @@ void DelayDetection::downsampleData(float *input, float *output, int nbSamples,
src_data.src_ratio = downfactor;
src_data.end_of_input = 0; // More data will come
- src_process(_src_state, &src_data);
+ src_process(src_state, &src_data);
}
}
void DelayDetection::bandpassFilter(float *input, int nbSamples)
{
- for (int i = 0; i < nbSamples; i++)
- input[i] = _bandpassFilter.getOutputSample(input[i]);
+ for (int i = 0; i < nbSamples; ++i)
+ input[i] = bandpassFilter_.getOutputSample(input[i]);
}
int DelayDetection::getMaxIndex(float *data, int size)
{
-
float max = 0.0;
int k = 0;
diff --git a/daemon/src/audio/delaydetection.h b/daemon/src/audio/delaydetection.h
index dd2adde18a44142cee51a1abe280e14cde8519d0..caa99db8551fc3ad9cb9d57bd2fa5eea6cffcf56 100644
--- a/daemon/src/audio/delaydetection.h
+++ b/daemon/src/audio/delaydetection.h
@@ -42,61 +42,42 @@
#define MAX_DELAY 150
// Size of internal buffers in samples
-#define DELAY_BUFF_SIZE MAX_DELAY*8000/1000
+#define DELAY_BUFF_SIZE MAX_DELAY * 8000 / 1000
#define MAXFILTERSIZE 100
-
-
class FirFilter {
public:
-
- /**
- * Constructor for this class
- */
- FirFilter(std::vector<double> ir);
-
- /**
- * SDestructor for this class
- */
- ~FirFilter();
-
+ FirFilter(const std::vector<double> &ir);
/**
* Perform filtering on one sample
*/
float getOutputSample(float inputSample);
- void reset(void);
-
+ void reset();
private:
/**
* Length of the filter
*/
- int _length;
+ int length_;
/**
* Coefficient of the filter
*/
- std::vector<double> _impulseResponse;
+ std::vector<double> impulseResponse_;
/**
* Circular buffer
*/
- double _taps[MAXFILTERSIZE];
-
- /**
- * Counter
- */
- int _count;
-
+ double taps_[MAXFILTERSIZE];
+ int counter_;
};
class DelayDetection {
-
public:
DelayDetection();
@@ -134,47 +115,47 @@ class DelayDetection {
int getMaxIndex(float *data, int size);
- State _internalState;
+ State internalState_;
- FirFilter _decimationFilter;
+ FirFilter decimationFilter_;
- FirFilter _bandpassFilter;
+ FirFilter bandpassFilter_;
/**
* Segment size in samples for correlation
*/
- short _segmentSize;
+ short segmentSize_;
- int _downsamplingFactor;
+ int downsamplingFactor_;
/**
* Resulting correlation size (s + w -1)
*/
- short _correlationSize;
+ short correlationSize_;
- float _spkrReference[WINDOW_SIZE*2];
+ float spkrReference_[WINDOW_SIZE*2];
- float _capturedData[DELAY_BUFF_SIZE*2];
+ float capturedData_[DELAY_BUFF_SIZE*2];
- float _spkrReferenceDown[WINDOW_SIZE*2];
+ float spkrReferenceDown_[WINDOW_SIZE*2];
- float _captureDataDown[DELAY_BUFF_SIZE*2];
+ float captureDataDown_[DELAY_BUFF_SIZE*2];
- float _spkrReferenceFilter[WINDOW_SIZE*2];
+ float spkrReferenceFilter_[WINDOW_SIZE*2];
- float _captureDataFilter[DELAY_BUFF_SIZE*2];
+ float captureDataFilter_[DELAY_BUFF_SIZE*2];
- float _correlationResult[DELAY_BUFF_SIZE*2];
+ float correlationResult_[DELAY_BUFF_SIZE*2];
- int _remainingIndex;
+ int remainingIndex_;
- int _spkrDownSize;
+ int spkrDownSize_;
- int _micDownSize;
+ int micDownSize_;
- int _nbMicSampleStored;
+ int nbMicSampleStored_;
- int _nbSpkrSampleStored;
+ int nbSpkrSampleStored_;
public:
diff --git a/daemon/src/audio/gaincontrol.cpp b/daemon/src/audio/gaincontrol.cpp
index 5955d52cba2058f50f63789074bfb5aca824b8c8..f9df37193a72ffc41cb0524f08b53b0a29506fcf 100644
--- a/daemon/src/audio/gaincontrol.cpp
+++ b/daemon/src/audio/gaincontrol.cpp
@@ -1,7 +1,6 @@
-#include <math.h>
-#include <limits.h>
+#include <cmath>
+#include <climits>
#include <fstream>
-#include <iostream>
#include "global.h"
#include "gaincontrol.h"
@@ -14,34 +13,18 @@
#define SFL_GAIN_LOGe10 2.30258509299404568402
-// #define DUMP_GAIN_CONTROL_SIGNAL
-
-GainControl::GainControl(double sr, double target) : averager(sr, SFL_GAIN_ATTACK_TIME, SFL_GAIN_RELEASE_TIME)
- , limiter(SFL_GAIN_LIMITER_RATIO, SFL_GAIN_LIMITER_THRESHOLD)
- , targetLeveldB(target)
- , targetLevelLinear(0.0)
- , currentGain(1.0)
- , previousGain(0.0)
- , maxIncreaseStep(0.0)
- , maxDecreaseStep(0.0)
+GainControl::GainControl(double sr, double target) : averager_(sr, SFL_GAIN_ATTACK_TIME, SFL_GAIN_RELEASE_TIME)
+ , limiter_(SFL_GAIN_LIMITER_RATIO, SFL_GAIN_LIMITER_THRESHOLD)
+ , targetLeveldB_(target)
+ , targetLevelLinear_(exp(targetLeveldB_ * 0.05 * SFL_GAIN_LOGe10))
+ , currentGain_(1.0)
+ , previousGain_(0.0)
+ , maxIncreaseStep_(exp(0.11513 * 12. * 160 / 8000)) // Computed on 12 frames (240 ms)
+ , maxDecreaseStep_(exp(-0.11513 * 40. * 160 / 8000))// Computed on 40 frames (800 ms)
{
- targetLevelLinear = exp(targetLeveldB * 0.05 * SFL_GAIN_LOGe10);
-
- maxIncreaseStep = exp(0.11513 * 12. * 160 / 8000); // Computed on 12 frames (240 ms)
- maxDecreaseStep = exp(-0.11513 * 40. * 160 / 8000); // Computed on 40 frames (800 ms)
-
- _debug("GainControl: Target gain %f dB (%f linear)", targetLeveldB, targetLevelLinear);
-
+ _debug("GainControl: Target gain %f dB (%f linear)", targetLeveldB_, targetLevelLinear_);
}
-GainControl::~GainControl() {}
-
-#ifdef DUMP_GAIN_CONTROL_SIGNAL
-std::fstream tmpRms("gaintestrms.raw", std::fstream::out);
-std::fstream tmpIn("gaintestin.raw", std::fstream::out);
-std::fstream tmpOut("gaintestout.raw", std::fstream::out);
-#endif
-
void GainControl::process(SFLDataFormat *buf, int samples)
{
double rms, rmsAvgLevel, in, out, diffRms, maxRms;
@@ -52,82 +35,56 @@ void GainControl::process(SFLDataFormat *buf, int samples)
// linear conversion
in = (double)buf[i] / (double)SHRT_MAX;
- out = currentGain * in;
-
- rms = detector.getRms(out);
- rmsAvgLevel = sqrt(averager.getAverage(rms));
+ out = currentGain_ * in;
-#ifdef DUMP_GAIN_CONTROL_SIGNAL
- tmpRms.write(reinterpret_cast<char *>(&rmsAvgLevel), sizeof(double));
- tmpIn.write(reinterpret_cast<char *>(&in), sizeof(double));
-#endif
+ rms = out * out;
+ rmsAvgLevel = sqrt(averager_.getAverage(rms));
- if (rmsAvgLevel > maxRms) {
+ if (rmsAvgLevel > maxRms)
maxRms = rmsAvgLevel;
- }
-
- out = limiter.limit(out);
-#ifdef DUMP_GAIN_CONTROL_SIGNAL
- tmpOut.write(reinterpret_cast<char *>(&out), sizeof(double));
-#endif
+ out = limiter_.limit(out);
buf[i] = (short)(out * (double)SHRT_MAX);
}
- diffRms = maxRms - targetLevelLinear;
-
- if ((diffRms > 0.0) && (maxRms > 0.1)) {
- currentGain *= maxDecreaseStep;
- } else if ((diffRms <= 0.0) && (maxRms > 0.1)) {
- currentGain *= maxIncreaseStep;
- } else if (maxRms <= 0.1) {
- currentGain = 1.0;
- }
+ diffRms = maxRms - targetLevelLinear_;
- currentGain = 0.5 * (currentGain + previousGain);
+ if ((diffRms > 0.0) && (maxRms > 0.1))
+ currentGain_ *= maxDecreaseStep_;
+ else if ((diffRms <= 0.0) && (maxRms > 0.1))
+ currentGain_ *= maxIncreaseStep_;
+ else if (maxRms <= 0.1)
+ currentGain_ = 1.0;
- previousGain = currentGain;
+ currentGain_ = 0.5 * (currentGain_ + previousGain_);
- // _debug("GainControl: current gain: %f, target gain: %f, rmsAvgLevel %f, target level %f",
- // currentGain, gainTargetLevel, rmsAvgLevel, targetLevelLinear);
-}
-
-GainControl::RmsDetection::RmsDetection() {}
-
-double GainControl::RmsDetection::getRms(double in)
-{
- return in * in;
+ previousGain_ = currentGain_;
}
GainControl::DetectionAverage::DetectionAverage(double sr, double ta, double tr) :
- g_a(0.0), teta_a(ta), g_r(0.0), teta_r(tr), samplingRate(sr), previous_y(0.0)
+ g_a_(0.0), teta_a_(ta), g_r_(0.0), teta_r_(tr), samplingRate_(sr), previous_y_(0.0)
{
- g_a = exp(-1.0 / (samplingRate * (teta_a / 1000.0)));
- g_r = exp(-1.0 / (samplingRate * (teta_r / 1000.0)));
+ g_a_ = exp(-1.0 / (samplingRate_ * (teta_a_ / 1000.0)));
+ g_r_ = exp(-1.0 / (samplingRate_ * (teta_r_ / 1000.0)));
}
double GainControl::DetectionAverage::getAverage(double in)
{
- if (in > previous_y) {
- previous_y = ((1.0 - g_a) * in) + (g_a * previous_y);
- } else {
- previous_y = ((1.0 - g_r) * in) + (g_r * previous_y);
- }
-
- return previous_y;
+ if (in > previous_y_)
+ previous_y_ = ((1.0 - g_a_) * in) + (g_a_ * previous_y_);
+ else
+ previous_y_ = ((1.0 - g_r_) * in) + (g_r_ * previous_y_);
+ return previous_y_;
}
-GainControl::Limiter::Limiter(double r, double thresh) : ratio(r), threshold(thresh)
-{
-}
+GainControl::Limiter::Limiter(double r, double thresh) : ratio_(r), threshold_(thresh)
+{}
double GainControl::Limiter::limit(double in)
{
- double out;
-
- out = (in > threshold ? (ratio * (in - threshold)) + threshold :
- in < -threshold ? (ratio * (in + threshold)) - threshold : in);
+ double out = (in > threshold_ ? (ratio_ * (in - threshold_)) + threshold_ :
+ in < -threshold_ ? (ratio_ * (in + threshold_)) - threshold_ : in);
return out;
}
diff --git a/daemon/src/audio/gaincontrol.h b/daemon/src/audio/gaincontrol.h
index d0455368a8ac6406377387f93cf0e8e6e2b2ee33..e8f0ecd685103c14d4232805d7f17b2cf79ebec7 100644
--- a/daemon/src/audio/gaincontrol.h
+++ b/daemon/src/audio/gaincontrol.h
@@ -3,8 +3,6 @@
#include "global.h"
-#define SFL_GAIN_BUFFER_LENGTH 160
-
class GainControl {
public:
@@ -15,11 +13,6 @@ class GainControl {
*/
GainControl(double, double);
- /**
- * Destructor for this class
- */
- ~GainControl(void);
-
/**
* Apply addaptive gain factor on input signal
* /param Input audio buffer
@@ -28,25 +21,6 @@ class GainControl {
void process(SFLDataFormat *, int samples);
private:
-
- /**
- * Rms detector
- */
- class RmsDetection {
- public:
- /**
- * Constructor for this class
- */
- RmsDetection(void);
-
- /**
- * Get rms value
- * /param Audio sample
- */
- double getRms(double);
-
- };
-
class DetectionAverage {
public:
/**
@@ -66,32 +40,32 @@ class GainControl {
/**
* Average factor for attack
*/
- double g_a;
+ double g_a_;
/**
* Attack ramp time (in ms)
*/
- double teta_a;
+ double teta_a_;
/**
* Average factor for release
*/
- double g_r;
+ double g_r_;
/**
* Release ramp time (in ms)
*/
- double teta_r;
+ double teta_r_;
/**
* Samplig rate
*/
- double samplingRate;
+ double samplingRate_;
/**
* Previous gain (first order memory)
*/
- double previous_y;
+ double previous_y_;
};
class Limiter {
@@ -109,54 +83,49 @@ class GainControl {
double limit(double);
private:
- double ratio;
- double threshold;
+ double ratio_;
+ double threshold_;
};
- /**
- * Current audio level detection
- */
- RmsDetection detector;
-
/**
* First order mean filter
*/
- DetectionAverage averager;
+ DetectionAverage averager_;
/**
* Post processing compression
*/
- Limiter limiter;
+ Limiter limiter_;
/**
* Target audio level in dB
*/
- double targetLeveldB;
+ double targetLeveldB_;
/**
* Target audio level in linear scale
*/
- double targetLevelLinear;
+ double targetLevelLinear_;
/**
* Current gain
*/
- double currentGain;
+ double currentGain_;
/**
* Previou gain for smoothing
*/
- double previousGain;
+ double previousGain_;
/**
* Maximum incrementation stop of current gain
*/
- double maxIncreaseStep;
+ double maxIncreaseStep_;
/**
* Maximum decrease step
*/
- double maxDecreaseStep;
+ double maxDecreaseStep_;
};
diff --git a/daemon/src/audio/pulseaudio/audiostream.cpp b/daemon/src/audio/pulseaudio/audiostream.cpp
index 316e2a92e82c6ad5d578ad353a80e6f97fb7b99f..530784d291d90f1d3d32c837e0aa7fe867d5ce9a 100644
--- a/daemon/src/audio/pulseaudio/audiostream.cpp
+++ b/daemon/src/audio/pulseaudio/audiostream.cpp
@@ -32,7 +32,7 @@
#include "pulselayer.h"
AudioStream::AudioStream(pa_context *c, pa_threaded_mainloop *m, const char *desc, int type, int smplrate, std::string *deviceName)
- : _mainloop(m)
+ : mainloop_(m)
{
static const pa_channel_map channel_map = {
1,
@@ -48,9 +48,9 @@ AudioStream::AudioStream(pa_context *c, pa_threaded_mainloop *m, const char *des
assert(pa_sample_spec_valid(&sample_spec));
assert(pa_channel_map_valid(&channel_map));
- _audiostream = pa_stream_new(c, desc, &sample_spec, &channel_map);
+ audiostream_ = pa_stream_new(c, desc, &sample_spec, &channel_map);
- if (!_audiostream) {
+ if (!audiostream_) {
_error("Pulse: %s: pa_stream_new() failed : %s" , desc, pa_strerror(pa_context_errno(c)));
throw std::runtime_error("Pulse : could not create stream\n");
}
@@ -64,33 +64,33 @@ AudioStream::AudioStream(pa_context *c, pa_threaded_mainloop *m, const char *des
const char *name = deviceName ? deviceName->c_str() : NULL;
- pa_threaded_mainloop_lock(_mainloop);
+ pa_threaded_mainloop_lock(mainloop_);
if (type == PLAYBACK_STREAM || type == RINGTONE_STREAM)
- pa_stream_connect_playback(_audiostream , name, &attributes, (pa_stream_flags_t)(PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE), NULL, NULL);
+ pa_stream_connect_playback(audiostream_, name, &attributes, (pa_stream_flags_t)(PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE), NULL, NULL);
else if (type == CAPTURE_STREAM)
- pa_stream_connect_record(_audiostream, name, &attributes, (pa_stream_flags_t)(PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE));
+ pa_stream_connect_record(audiostream_, name, &attributes, (pa_stream_flags_t)(PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE));
- pa_threaded_mainloop_unlock(_mainloop);
+ pa_threaded_mainloop_unlock(mainloop_);
- pa_stream_set_state_callback(_audiostream , stream_state_callback, NULL);
+ pa_stream_set_state_callback(audiostream_, stream_state_callback, NULL);
}
AudioStream::~AudioStream()
{
- pa_threaded_mainloop_lock(_mainloop);
+ pa_threaded_mainloop_lock(mainloop_);
- pa_stream_disconnect(_audiostream);
+ pa_stream_disconnect(audiostream_);
// make sure we don't get any further callback
- pa_stream_set_state_callback(_audiostream, NULL, NULL);
- pa_stream_set_write_callback(_audiostream, NULL, NULL);
- pa_stream_set_underflow_callback(_audiostream, NULL, NULL);
- pa_stream_set_overflow_callback(_audiostream, NULL, NULL);
+ pa_stream_set_state_callback(audiostream_, NULL, NULL);
+ pa_stream_set_write_callback(audiostream_, NULL, NULL);
+ pa_stream_set_underflow_callback(audiostream_, NULL, NULL);
+ pa_stream_set_overflow_callback(audiostream_, NULL, NULL);
- pa_stream_unref(_audiostream);
+ pa_stream_unref(audiostream_);
- pa_threaded_mainloop_unlock(_mainloop);
+ pa_threaded_mainloop_unlock(mainloop_);
}
void
@@ -99,7 +99,6 @@ AudioStream::stream_state_callback(pa_stream* s, void* user_data UNUSED)
char str[PA_SAMPLE_SPEC_SNPRINT_MAX];
switch (pa_stream_get_state(s)) {
-
case PA_STREAM_CREATING:
_info("Pulse: Stream is creating...");
break;
@@ -129,10 +128,10 @@ AudioStream::stream_state_callback(pa_stream* s, void* user_data UNUSED)
}
}
-bool AudioStream::isReady(void)
+bool AudioStream::isReady()
{
- if (!_audiostream)
+ if (!audiostream_)
return false;
- return pa_stream_get_state(_audiostream) == PA_STREAM_READY;
+ return pa_stream_get_state(audiostream_) == PA_STREAM_READY;
}
diff --git a/daemon/src/audio/pulseaudio/audiostream.h b/daemon/src/audio/pulseaudio/audiostream.h
index bacfc6f9baac6c67c15468235d2365c6657e815f..7e67db5fc305f6d83b2d47511bc8a2fd4ed353c2 100644
--- a/daemon/src/audio/pulseaudio/audiostream.h
+++ b/daemon/src/audio/pulseaudio/audiostream.h
@@ -63,10 +63,10 @@ class AudioStream {
* @return pa_stream* The stream
*/
pa_stream* pulseStream() {
- return _audiostream;
+ return audiostream_;
}
- bool isReady(void);
+ bool isReady();
private:
@@ -84,12 +84,12 @@ class AudioStream {
/**
* The pulse audio object
*/
- pa_stream* _audiostream;
+ pa_stream* audiostream_;
/**
* A pointer to the opaque threaded main loop object
*/
- pa_threaded_mainloop * _mainloop;
+ pa_threaded_mainloop * mainloop_;
};
#endif // _AUDIO_STREAM_H
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/audio/pulseaudio/pulselayer.cpp
index db5d35008f2194623c0e702503d31504f099e9b3..bbfb3fbbe0d374ee164ba9840152a24dca518701 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/audio/pulseaudio/pulselayer.cpp
@@ -231,7 +231,7 @@ PulseLayer::PulseLayer()
isStarted_ = true;
}
-PulseLayer::~PulseLayer(void)
+PulseLayer::~PulseLayer()
{
disconnectAudioStream();
@@ -287,13 +287,13 @@ void PulseLayer::context_state_callback(pa_context* c, void* user_data)
}
-void PulseLayer::updateSinkList(void)
+void PulseLayer::updateSinkList()
{
getSinkList()->clear();
pa_context_get_sink_info_list(context_, sink_input_info_callback, this);
}
-void PulseLayer::updateSourceList(void)
+void PulseLayer::updateSourceList()
{
getSourceList()->clear();
pa_context_get_source_info_list(context_, source_input_info_callback, this);
@@ -345,7 +345,7 @@ void PulseLayer::createStreams(pa_context* c)
}
-void PulseLayer::disconnectAudioStream(void)
+void PulseLayer::disconnectAudioStream()
{
if (playback_) {
if (playback_->pulseStream()) {
@@ -384,7 +384,7 @@ void PulseLayer::disconnectAudioStream(void)
}
}
-void PulseLayer::startStream(void)
+void PulseLayer::startStream()
{
// Create Streams
if (!playback_ or !record_)
@@ -398,7 +398,7 @@ void PulseLayer::startStream(void)
void
-PulseLayer::stopStream(void)
+PulseLayer::stopStream()
{
pa_threaded_mainloop_lock(mainloop_);
@@ -413,7 +413,7 @@ PulseLayer::stopStream(void)
disconnectAudioStream();
}
-void PulseLayer::writeToSpeaker(void)
+void PulseLayer::writeToSpeaker()
{
if (!playback_ or !playback_->isReady())
return;
@@ -505,7 +505,7 @@ void PulseLayer::writeToSpeaker(void)
pa_stream_write(s, data, inBytes, NULL, 0, PA_SEEK_RELATIVE);
}
-void PulseLayer::readFromMic(void)
+void PulseLayer::readFromMic()
{
if (!record_ or !record_->isReady())
return;
@@ -545,7 +545,7 @@ void PulseLayer::readFromMic(void)
}
-void PulseLayer::ringtoneToSpeaker(void)
+void PulseLayer::ringtoneToSpeaker()
{
if (!ringtone_ or !ringtone_->isReady())
return;
diff --git a/daemon/src/audio/pulseaudio/pulselayer.h b/daemon/src/audio/pulseaudio/pulselayer.h
index 7e078bde878eb6bd84ac898099391aae2fd32619..a57fe30a390d23a08f67dca8bc28ec2f83521f3e 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.h
+++ b/daemon/src/audio/pulseaudio/pulselayer.h
@@ -44,35 +44,35 @@ class AudioStream;
class PulseLayer : public AudioLayer {
public:
PulseLayer();
- ~PulseLayer(void);
+ ~PulseLayer();
- std::list<std::string>* getSinkList(void) {
+ std::list<std::string>* getSinkList() {
return &sinkList_;
}
- std::list<std::string>* getSourceList(void) {
+ std::list<std::string>* getSourceList() {
return &sourceList_;
}
/**
* Write data from the ring buffer to the harware and read data from the hardware
*/
- void readFromMic(void);
- void writeToSpeaker(void);
- void ringtoneToSpeaker(void);
+ void readFromMic();
+ void writeToSpeaker();
+ void ringtoneToSpeaker();
- void updateSinkList(void);
+ void updateSinkList();
- void updateSourceList(void);
+ void updateSourceList();
bool inSinkList(const std::string &deviceName) const;
bool inSourceList(const std::string &deviceName) const;
- void startStream(void);
+ void startStream();
- void stopStream(void);
+ void stopStream();
static void context_state_callback(pa_context* c, void* user_data);
@@ -92,7 +92,7 @@ class PulseLayer : public AudioLayer {
/**
* Close the connection with the local pulseaudio server
*/
- void disconnectAudioStream(void);
+ void disconnectAudioStream();
/** PulseAudio context and asynchronous loop */
pa_context* context_;
diff --git a/daemon/src/audio/recordable.cpp b/daemon/src/audio/recordable.cpp
index ca4813cede78d2f1365669ecfc1877bce1645776..4b9ea8f41bdeeb2ad900ad5e1a5d291813faccd6 100644
--- a/daemon/src/audio/recordable.cpp
+++ b/daemon/src/audio/recordable.cpp
@@ -32,21 +32,15 @@
Recordable::Recordable() : recorder(&recAudio, Manager::instance().getMainBuffer())
{
- _debug("-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Recordable Constructor -=-=-=-=-=-=-=-=-=--=-=-=-");
recAudio.setRecordingOption(AudioRecord::FILE_WAV, 8000, Manager::instance().audioPreference.getRecordpath());
}
-
Recordable::~Recordable()
{
- _debug("-=-=-=-=-=-=-=-=-=-=-=-=-=-=-= Recordable Destructor -=-=-=-=-=-=-=-=-=-=-=-=-=-");
-
- if (recAudio.isOpenFile()) {
+ if (recAudio.isOpenFile())
recAudio.closeFile();
- }
}
-
void Recordable::initRecFileName(std::string filename)
{
recAudio.initFileName(filename);
@@ -66,5 +60,3 @@ int Recordable::getRecordingSmplRate() const
{
return recAudio.getSndSamplingRate();
}
-
-
diff --git a/daemon/src/audio/recordable.h b/daemon/src/audio/recordable.h
index efbb411e54b99f49bf20a26c21aaaf86f12fc958..3e900a2f479dbc0dfb868b62befa7c656b720c2e 100644
--- a/daemon/src/audio/recordable.h
+++ b/daemon/src/audio/recordable.h
@@ -57,7 +57,7 @@ class Recordable {
/**
* Stop recording
*/
- void stopRecording(void) {
+ void stopRecording() {
recAudio.stopRecording();
}
@@ -69,7 +69,7 @@ class Recordable {
/**
* Return the file path for this recording
*/
- std::string getFileName(void);
+ std::string getFileName();
/**
* Set recording sampling rate.
@@ -79,7 +79,7 @@ class Recordable {
/**
* Return the recording sampling rate
*/
- int getRecordingSmplRate(void) const;
+ int getRecordingSmplRate() const;
/**
* Virtual method to be implemented in order to the main
diff --git a/daemon/src/audio/sound/audiofile.cpp b/daemon/src/audio/sound/audiofile.cpp
index dcf6d911422d30f844479cbb0036151b95e0d8f9..11c9172a8417760692572746c6db173a64dcff48 100644
--- a/daemon/src/audio/sound/audiofile.cpp
+++ b/daemon/src/audio/sound/audiofile.cpp
@@ -44,7 +44,7 @@
#include "manager.h"
RawFile::RawFile(const std::string& name, sfl::AudioCodec* codec, unsigned int sampleRate)
- : audioCodec(codec)
+ : audioCodec_(codec)
{
filepath_ = name;
@@ -66,9 +66,9 @@ RawFile::RawFile(const std::string& name, sfl::AudioCodec* codec, unsigned int s
file.read(fileBuffer,length);
file.close();
- unsigned int frameSize = audioCodec->getFrameSize();
- unsigned int bitrate = audioCodec->getBitRate() * 1000 / 8;
- unsigned int audioRate = audioCodec->getClockRate();
+ unsigned int frameSize = audioCodec_->getFrameSize();
+ unsigned int bitrate = audioCodec_->getBitRate() * 1000 / 8;
+ unsigned int audioRate = audioCodec_->getClockRate();
unsigned int encFrameSize = frameSize * bitrate / audioRate;
unsigned int decodedSize = length * (frameSize / encFrameSize);
@@ -78,7 +78,7 @@ RawFile::RawFile(const std::string& name, sfl::AudioCodec* codec, unsigned int s
size_ = decodedSize;
while (length >= encFrameSize) {
- bufpos += audioCodec->decode(bufpos, filepos, encFrameSize);
+ bufpos += audioCodec_->decode(bufpos, filepos, encFrameSize);
filepos += encFrameSize;
length -= encFrameSize;
}
diff --git a/daemon/src/audio/sound/audiofile.h b/daemon/src/audio/sound/audiofile.h
index 1a6d2ff182a5b3c45bb31ec53d3d6a28e8223a9f..27cd35672b0342182d93aeac1b61da18b86577c6 100644
--- a/daemon/src/audio/sound/audiofile.h
+++ b/daemon/src/audio/sound/audiofile.h
@@ -35,8 +35,6 @@
#define __AUDIOFILE_H__
#include <stdexcept>
-#include <fstream>
-
#include "audio/audioloop.h"
namespace sfl {
@@ -45,7 +43,7 @@ class AudioCodec;
class AudioFileException : public std::runtime_error {
public:
- AudioFileException(const std::string& str="") :
+ AudioFileException(const std::string& str = "") :
std::runtime_error("AudioFile: AudioFileException occured: " + str) {}
};
@@ -54,7 +52,7 @@ class AudioFileException : public std::runtime_error {
*/
class AudioFile : public AudioLoop {
public:
- std::string getFilePath(void) const {
+ std::string getFilePath() const {
return filepath_;
}
@@ -63,13 +61,6 @@ class AudioFile : public AudioLoop {
std::string filepath_;
};
-
-
-/**
- * @file audiofile.h
- * @brief A class to manage sound files
- */
-
class RawFile : public AudioFile {
public:
/**
@@ -78,24 +69,21 @@ class RawFile : public AudioFile {
RawFile(const std::string& name, sfl::AudioCodec* codec, unsigned int sampleRate = 8000);
private:
- // Copy Constructor
+ // noncopyable
RawFile(const RawFile& rh);
-
- // Assignment Operator
- const RawFile& operator= (const RawFile& rh);
+ RawFile& operator= (const RawFile& rh);
/** Your preferred codec */
- sfl::AudioCodec* audioCodec;
+ sfl::AudioCodec* audioCodec_;
};
class WaveFile : public AudioFile {
-
public:
/**
* Load a sound file in memory
- * @param filename The absolute path to the file
- * @param sampleRate The sample rate to read it
- */
+ * @param filename The absolute path to the file
+ * @param sampleRate The sample rate to read it
+ */
WaveFile(const std::string&, unsigned int);
};
diff --git a/daemon/src/audio/sound/dtmf.cpp b/daemon/src/audio/sound/dtmf.cpp
index 015eaa6a48c40fc885a26fbf9c635d012bfcbc44..914e616efcec7c2ade4e7c500b860069931ef576 100644
--- a/daemon/src/audio/sound/dtmf.cpp
+++ b/daemon/src/audio/sound/dtmf.cpp
@@ -35,54 +35,47 @@
#include "dtmf.h"
DTMF::DTMF(unsigned int sampleRate)
- : currentTone(0), newTone(0), dtmfgenerator(sampleRate)
-{
-}
+ : currentTone_(0), newTone_(0), dtmfgenerator_(sampleRate)
+{}
-DTMF::~DTMF(void)
+void DTMF::startTone(char code)
{
+ newTone_ = code;
}
-void
-DTMF::startTone(char code)
-{
- newTone = code;
-}
-
-bool
-DTMF::generateDTMF(SFLDataFormat* buffer, size_t n)
+bool DTMF::generateDTMF(SFLDataFormat* buffer, size_t n)
{
if (!buffer) return false;
try {
- if (currentTone != 0) {
+ if (currentTone_ != 0) {
// Currently generating a DTMF tone
- if (currentTone == newTone) {
+ if (currentTone_ == newTone_) {
// Continue generating the same tone
- dtmfgenerator.getNextSamples(buffer, n);
+ dtmfgenerator_.getNextSamples(buffer, n);
return true;
- } else if (newTone != 0) {
+ } else if (newTone_ != 0) {
// New tone requested
- dtmfgenerator.getSamples(buffer, n, newTone);
- currentTone = newTone;
+ dtmfgenerator_.getSamples(buffer, n, newTone_);
+ currentTone_ = newTone_;
return true;
} else {
// Stop requested
- currentTone = newTone;
+ currentTone_ = newTone_;
return false;
}
} else {
// Not generating any DTMF tone
- if (newTone) {
+ if (newTone_) {
// Requested to generate a DTMF tone
- dtmfgenerator.getSamples(buffer, n, newTone);
- currentTone = newTone;
+ dtmfgenerator_.getSamples(buffer, n, newTone_);
+ currentTone_ = newTone_;
return true;
}
-
- return false;
+ else
+ return false;
}
- } catch (DTMFException e) {
+ } catch (const DTMFException &e) {
// invalid key
return false;
}
diff --git a/daemon/src/audio/sound/dtmf.h b/daemon/src/audio/sound/dtmf.h
index 2be2c49f1cad12e0837ca234e6b1385f0dcb74dd..90b17715b82eaee2cc7c0fd636497e7543f78c47 100644
--- a/daemon/src/audio/sound/dtmf.h
+++ b/daemon/src/audio/sound/dtmf.h
@@ -50,11 +50,6 @@ class DTMF {
*/
DTMF(unsigned int sampleRate);
- /**
- * Destructor
- */
- ~DTMF(void);
-
/**
* Start the done for th given dtmf
* @param code The DTMF code
@@ -68,10 +63,11 @@ class DTMF {
*/
bool generateDTMF(SFLDataFormat* buffer, size_t n);
- char currentTone;
- char newTone;
+ private:
+ char currentTone_;
+ char newTone_;
- DTMFGenerator dtmfgenerator;
+ DTMFGenerator dtmfgenerator_;
};
#endif // __KEY_DTMF_H_
diff --git a/daemon/src/audio/sound/tone.h b/daemon/src/audio/sound/tone.h
index 5d82b2c465d0b5b3435d0b65e6d6e027e2494a2a..be9c6c0cbe270308fb924cf45c4d3ceabac2755f 100644
--- a/daemon/src/audio/sound/tone.h
+++ b/daemon/src/audio/sound/tone.h
@@ -71,7 +71,7 @@ class Tone : public AudioLoop {
/**
*
*/
- void fillWavetable(void);
+ void fillWavetable();
/**
*
diff --git a/daemon/src/audio/sound/tonelist.cpp b/daemon/src/audio/sound/tonelist.cpp
index b529c2452e760945d4c4d13b2b0dd409a5602017..4878db4db427d7d8ea3d27abb4e6c641d3386957 100644
--- a/daemon/src/audio/sound/tonelist.cpp
+++ b/daemon/src/audio/sound/tonelist.cpp
@@ -88,54 +88,47 @@ static const char *toneZone[TelephoneTone::ZID_COUNTRIES][Tone::TONE_NULL] = {
TelephoneTone::COUNTRYID
TelephoneTone::getCountryId(const std::string& countryName)
{
- if (countryName == "North America") return ZID_NORTH_AMERICA;
-
- if (countryName == "France") return ZID_FRANCE;
-
- if (countryName == "Australia") return ZID_AUSTRALIA;
-
- if (countryName == "United Kingdom") return ZID_UNITED_KINGDOM;
-
- if (countryName == "Spain") return ZID_SPAIN;
-
- if (countryName == "Italy") return ZID_ITALY;
-
- if (countryName == "Japan") return ZID_JAPAN;
-
- return ZID_NORTH_AMERICA; // default
+ if (countryName == "North America") return ZID_NORTH_AMERICA;
+ else if (countryName == "France") return ZID_FRANCE;
+ else if (countryName == "Australia") return ZID_AUSTRALIA;
+ else if (countryName == "United Kingdom") return ZID_UNITED_KINGDOM;
+ else if (countryName == "Spain") return ZID_SPAIN;
+ else if (countryName == "Italy") return ZID_ITALY;
+ else if (countryName == "Japan") return ZID_JAPAN;
+ else return ZID_NORTH_AMERICA; // default
}
TelephoneTone::TelephoneTone(const std::string& countryName, unsigned int sampleRate) :
- _currentTone(Tone::TONE_NULL)
+ currentTone_(Tone::TONE_NULL)
{
TelephoneTone::COUNTRYID countryId = getCountryId(countryName);
- _tone[Tone::TONE_DIALTONE] = new Tone(toneZone[countryId][Tone::TONE_DIALTONE], sampleRate);
- _tone[Tone::TONE_BUSY] = new Tone(toneZone[countryId][Tone::TONE_BUSY], sampleRate);
- _tone[Tone::TONE_RINGTONE] = new Tone(toneZone[countryId][Tone::TONE_RINGTONE], sampleRate);
- _tone[Tone::TONE_CONGESTION] = new Tone(toneZone[countryId][Tone::TONE_CONGESTION], sampleRate);
+ tone_[Tone::TONE_DIALTONE] = new Tone(toneZone[countryId][Tone::TONE_DIALTONE], sampleRate);
+ tone_[Tone::TONE_BUSY] = new Tone(toneZone[countryId][Tone::TONE_BUSY], sampleRate);
+ tone_[Tone::TONE_RINGTONE] = new Tone(toneZone[countryId][Tone::TONE_RINGTONE], sampleRate);
+ tone_[Tone::TONE_CONGESTION] = new Tone(toneZone[countryId][Tone::TONE_CONGESTION], sampleRate);
}
TelephoneTone::~TelephoneTone()
{
for (size_t i=0; i < Tone::TONE_NULL; i++)
- delete _tone[i];
+ delete tone_[i];
}
void
TelephoneTone::setCurrentTone(Tone::TONEID toneId)
{
- if (toneId != Tone::TONE_NULL && _currentTone != toneId)
- _tone[toneId]->reset();
+ if (toneId != Tone::TONE_NULL && currentTone_ != toneId)
+ tone_[toneId]->reset();
- _currentTone = toneId;
+ currentTone_ = toneId;
}
Tone*
TelephoneTone::getCurrentTone()
{
- if (_currentTone == Tone::TONE_NULL)
+ if (currentTone_ == Tone::TONE_NULL)
return NULL;
- return _tone[_currentTone];
+ return tone_[currentTone_];
}
diff --git a/daemon/src/audio/sound/tonelist.h b/daemon/src/audio/sound/tonelist.h
index 8df0c4cc02f7f37e7ed7ee8a985e6fe12368a2ec..5c43b6ab52b196009f637e496291c96f36606368 100644
--- a/daemon/src/audio/sound/tonelist.h
+++ b/daemon/src/audio/sound/tonelist.h
@@ -39,7 +39,6 @@
class TelephoneTone {
public:
-
/** Countries */
enum COUNTRYID {
ZID_NORTH_AMERICA = 0,
@@ -61,14 +60,13 @@ class TelephoneTone {
private:
// Copy Constructor
TelephoneTone(const TelephoneTone& rh);
-
// Assignment Operator
TelephoneTone& operator= (const TelephoneTone& rh);
static COUNTRYID getCountryId(const std::string& countryName);
- Tone* _tone[Tone::TONE_NULL];
- Tone::TONEID _currentTone;
+ Tone* tone_[Tone::TONE_NULL];
+ Tone::TONEID currentTone_;
};
#endif
diff --git a/daemon/src/config/yamlnode.h b/daemon/src/config/yamlnode.h
index 5d2ae0a16ceaf44d9a5cd8f1e72edb0300a0a4f9..a183ebe04dbb9863b882e3fa2234fcb1101a38c0 100644
--- a/daemon/src/config/yamlnode.h
+++ b/daemon/src/config/yamlnode.h
@@ -38,7 +38,6 @@
namespace Conf {
-
class YamlNode;
typedef std::list<YamlNode *> Sequence;
@@ -47,87 +46,67 @@ typedef std::map<std::string, YamlNode *> Mapping;
enum NodeType { DOCUMENT, SCALAR, MAPPING, SEQUENCE };
class YamlNode {
-
public:
YamlNode(NodeType t, YamlNode *top=NULL) : type(t), topNode(top) {}
- virtual ~YamlNode(void) {}
+ virtual ~YamlNode() {}
- NodeType getType(void) {
+ NodeType getType() {
return type;
}
- YamlNode *getTopNode(void) {
+ YamlNode *getTopNode() {
return topNode;
}
- virtual void deleteChildNodes(void) = 0;
+ virtual void deleteChildNodes() = 0;
private:
-
NodeType type;
-
YamlNode *topNode;
-
};
class YamlDocument : YamlNode {
-
public:
-
YamlDocument(YamlNode* top=NULL) : YamlNode(DOCUMENT, top) {}
- ~YamlDocument() {}
-
void addNode(YamlNode *node);
- YamlNode *popNode(void);
+ YamlNode *popNode();
- Sequence *getSequence(void) {
+ Sequence *getSequence() {
return &doc;
}
- virtual void deleteChildNodes(void);
+ virtual void deleteChildNodes();
private:
-
Sequence doc;
-
};
class SequenceNode : public YamlNode {
-
public:
-
SequenceNode(YamlNode *top) : YamlNode(SEQUENCE, top) {}
- ~SequenceNode() {}
-
Sequence *getSequence() {
return &seq;
}
void addNode(YamlNode *node);
- virtual void deleteChildNodes(void);
+ virtual void deleteChildNodes();
private:
-
Sequence seq;
-
};
class MappingNode : public YamlNode {
-
public:
-
MappingNode(YamlNode *top) : YamlNode(MAPPING, top) {}
- ~MappingNode() {}
-
Mapping *getMapping() {
return ↦
}
@@ -147,26 +126,18 @@ class MappingNode : public YamlNode {
void getValue(const std::string &key, int *i);
void getValue(const std::string &key, std::string *s);
- virtual void deleteChildNodes(void);
+ virtual void deleteChildNodes();
private:
-
Mapping map;
-
std::string tmpKey;
-
};
-
class ScalarNode : public YamlNode {
-
public:
-
ScalarNode(std::string s="", YamlNode *top=NULL) : YamlNode(SCALAR, top), str(s) {}
ScalarNode(bool b, YamlNode *top=NULL) : YamlNode(SCALAR, top), str(b ? "true" : "false") {}
- ~ScalarNode() {}
-
const std::string &getValue() {
return str;
}
@@ -175,17 +146,13 @@ class ScalarNode : public YamlNode {
str = s;
}
- virtual void deleteChildNodes(void) {}
+ virtual void deleteChildNodes() {}
private:
-
std::string str;
-
};
-
}
-
-
#endif
+
diff --git a/daemon/src/config/yamlparser.cpp b/daemon/src/config/yamlparser.cpp
index 008c8919c9ed3284e4a586aab845a468b06d9062..166abd1b94577f8ae9307b36645411cb1d0ac2ee 100644
--- a/daemon/src/config/yamlparser.cpp
+++ b/daemon/src/config/yamlparser.cpp
@@ -77,14 +77,13 @@ YamlParser::~YamlParser()
}
}
-void YamlParser::serializeEvents() throw(YamlParserException)
+void YamlParser::serializeEvents()
{
bool done = false;
yaml_event_t event, copiedEvent;
try {
-
- while (!done) {
+ while (not done) {
if (!yaml_parser_parse(&parser, &event))
throw YamlParserException("Error while parsing");
@@ -99,30 +98,27 @@ void YamlParser::serializeEvents() throw(YamlParserException)
yaml_event_delete(&event);
}
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
}
-void YamlParser::copyEvent(yaml_event_t *event_to, yaml_event_t *event_from) throw(YamlParserException)
+void YamlParser::copyEvent(yaml_event_t *event_to, yaml_event_t *event_from)
{
-
switch (event_from->type) {
case YAML_STREAM_START_EVENT: {
if (yaml_stream_start_event_initialize(event_to,
- event_from->data.stream_start.encoding) == 0) {
+ event_from->data.stream_start.encoding) == 0)
throw YamlParserException("Error stream start event");
- }
break;
}
case YAML_STREAM_END_EVENT: {
- if (yaml_stream_end_event_initialize(event_to) == 0) {
+ if (yaml_stream_end_event_initialize(event_to) == 0)
throw YamlParserException("Error stream end event");
- }
break;
}
@@ -132,26 +128,23 @@ void YamlParser::copyEvent(yaml_event_t *event_to, yaml_event_t *event_from) thr
event_from->data.document_start.version_directive,
event_from->data.document_start.tag_directives.start,
event_from->data.document_start.tag_directives.end,
- event_from->data.document_start.implicit) == 0) {
+ event_from->data.document_start.implicit) == 0)
throw YamlParserException("Error document start event");
- }
break;
}
case YAML_DOCUMENT_END_EVENT: {
if (yaml_document_end_event_initialize(event_to,
- event_from->data.document_end.implicit) == 0) {
+ event_from->data.document_end.implicit) == 0)
throw YamlParserException("Error document end event");
- }
break;
}
case YAML_ALIAS_EVENT: {
if (yaml_alias_event_initialize(event_to,
- event_from->data.alias.anchor) == 0) {
+ event_from->data.alias.anchor) == 0)
throw YamlParserException("Error alias event initialize");
- }
break;
}
@@ -163,9 +156,8 @@ void YamlParser::copyEvent(yaml_event_t *event_to, yaml_event_t *event_from) thr
event_from->data.scalar.length,
event_from->data.scalar.plain_implicit,
event_from->data.scalar.quoted_implicit,
- event_from->data.scalar.style) == 0) {
+ event_from->data.scalar.style) == 0)
throw YamlParserException("Error scalar event initialize");
- }
break;
}
@@ -174,16 +166,14 @@ void YamlParser::copyEvent(yaml_event_t *event_to, yaml_event_t *event_from) thr
event_from->data.sequence_start.anchor,
event_from->data.sequence_start.tag,
event_from->data.sequence_start.implicit,
- event_from->data.sequence_start.style) == 0) {
+ event_from->data.sequence_start.style) == 0)
throw YamlParserException("Error sequence start event");
- }
break;
}
case YAML_SEQUENCE_END_EVENT: {
- if (yaml_sequence_end_event_initialize(event_to) == 0) {
+ if (yaml_sequence_end_event_initialize(event_to) == 0)
throw YamlParserException("Error sequence end event");
- }
break;
}
@@ -192,16 +182,13 @@ void YamlParser::copyEvent(yaml_event_t *event_to, yaml_event_t *event_from) thr
event_from->data.mapping_start.anchor,
event_from->data.mapping_start.tag,
event_from->data.mapping_start.implicit,
- event_from->data.mapping_start.style) == 0) {
+ event_from->data.mapping_start.style) == 0)
throw YamlParserException("Error mapping start event");
- }
-
break;
}
case YAML_MAPPING_END_EVENT: {
- if (yaml_mapping_end_event_initialize(event_to) == 0) {
+ if (yaml_mapping_end_event_initialize(event_to) == 0)
throw YamlParserException("Error mapping end event");
- }
break;
}
@@ -211,7 +198,7 @@ void YamlParser::copyEvent(yaml_event_t *event_to, yaml_event_t *event_from) thr
}
-YamlDocument *YamlParser::composeEvents() throw(YamlParserException)
+YamlDocument *YamlParser::composeEvents()
{
try {
if (eventNumber == 0)
@@ -223,7 +210,7 @@ YamlDocument *YamlParser::composeEvents() throw(YamlParserException)
eventIndex = 0;
processStream();
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
@@ -231,7 +218,7 @@ YamlDocument *YamlParser::composeEvents() throw(YamlParserException)
return doc;
}
-void YamlParser::processStream() throw(YamlParserException)
+void YamlParser::processStream()
{
try {
while ((eventIndex < eventNumber) && (events[eventIndex].type != YAML_STREAM_END_EVENT)) {
@@ -244,12 +231,12 @@ void YamlParser::processStream() throw(YamlParserException)
if (events[eventIndex].type != YAML_STREAM_END_EVENT)
throw YamlParserException("Did not found end of stream");
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
}
-void YamlParser::processDocument() throw(YamlParserException)
+void YamlParser::processDocument()
{
try {
@@ -280,13 +267,13 @@ void YamlParser::processDocument() throw(YamlParserException)
if (events[eventIndex].type != YAML_DOCUMENT_END_EVENT)
throw YamlParserException("Did not found end of document");
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
}
-void YamlParser::processScalar(YamlNode *topNode) throw(YamlParserException)
+void YamlParser::processScalar(YamlNode *topNode)
{
try {
@@ -308,17 +295,15 @@ void YamlParser::processScalar(YamlNode *topNode) throw(YamlParserException)
default:
break;
}
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
}
-void YamlParser::processSequence(YamlNode *topNode) throw(YamlParserException)
+void YamlParser::processSequence(YamlNode *topNode)
{
-
try {
-
if (!topNode)
throw YamlParserException("No container for sequence");
@@ -362,13 +347,13 @@ void YamlParser::processSequence(YamlNode *topNode) throw(YamlParserException)
if (events[eventIndex].type != YAML_SEQUENCE_END_EVENT)
throw YamlParserException("Did not found end of sequence");
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
}
-void YamlParser::processMapping(YamlNode *topNode) throw(YamlParserException)
+void YamlParser::processMapping(YamlNode *topNode)
{
try {
@@ -422,14 +407,13 @@ void YamlParser::processMapping(YamlNode *topNode) throw(YamlParserException)
if (events[eventIndex].type != YAML_MAPPING_END_EVENT)
throw YamlParserException("Did not found end of mapping");
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
}
-void YamlParser::constructNativeData() throw(YamlParserException)
+void YamlParser::constructNativeData()
{
-
try {
Sequence *seq;
@@ -462,7 +446,7 @@ void YamlParser::constructNativeData() throw(YamlParserException)
iter++;
}
- } catch (YamlParserException &e) {
+ } catch (const YamlParserException &e) {
throw;
}
diff --git a/daemon/src/config/yamlparser.h b/daemon/src/config/yamlparser.h
index ca8e64622c2914e53d47994bbefbd8a5fbbf205b..43b54f466b0647c8166afc05b893c4522d59d431 100644
--- a/daemon/src/config/yamlparser.h
+++ b/daemon/src/config/yamlparser.h
@@ -59,37 +59,37 @@ class YamlParser {
~YamlParser();
- void serializeEvents() throw(YamlParserException);
+ void serializeEvents();
- YamlDocument *composeEvents() throw(YamlParserException);
+ YamlDocument *composeEvents();
- void constructNativeData() throw(YamlParserException);
+ void constructNativeData();
- SequenceNode *getAccountSequence(void) {
+ SequenceNode *getAccountSequence() {
return accountSequence;
};
- MappingNode *getPreferenceNode(void) {
+ MappingNode *getPreferenceNode() {
return preferenceNode;
}
- MappingNode *getAddressbookNode(void) {
+ MappingNode *getAddressbookNode() {
return addressbookNode;
}
- MappingNode *getAudioNode(void) {
+ MappingNode *getAudioNode() {
return audioNode;
}
- MappingNode *getHookNode(void) {
+ MappingNode *getHookNode() {
return hooksNode;
}
- MappingNode *getVoipPreferenceNode(void) {
+ MappingNode *getVoipPreferenceNode() {
return voiplinkNode;
}
- MappingNode *getShortcutNode(void) {
+ MappingNode *getShortcutNode() {
return shortcutNode;
}
@@ -98,17 +98,17 @@ class YamlParser {
/**
* Copy yaml parser event in event_to according to their type.
*/
- void copyEvent(yaml_event_t *event_to, yaml_event_t *event_from) throw(YamlParserException);
+ void copyEvent(yaml_event_t *event_to, yaml_event_t *event_from);
- void processStream(void) throw(YamlParserException);
+ void processStream();
- void processDocument(void) throw(YamlParserException);
+ void processDocument();
- void processScalar(YamlNode *topNode) throw(YamlParserException);
+ void processScalar(YamlNode *topNode);
- void processSequence(YamlNode *topNode) throw(YamlParserException);
+ void processSequence(YamlNode *topNode);
- void processMapping(YamlNode *topNode) throw(YamlParserException);
+ void processMapping(YamlNode *topNode);
void mainNativeDataMapping(MappingNode *map);
diff --git a/daemon/src/dbus/callmanager.cpp b/daemon/src/dbus/callmanager.cpp
index a8b449caadcb53a0cc8e927d5509d7929ee59cfc..aea7355ef46bcdb0f67122be16cb87e6bbd10e8d 100644
--- a/daemon/src/dbus/callmanager.cpp
+++ b/daemon/src/dbus/callmanager.cpp
@@ -202,7 +202,7 @@ CallManager::getConferenceDetails(const std::string& callID)
}
std::vector< std::string >
-CallManager::getConferenceList(void)
+CallManager::getConferenceList()
{
return Manager::instance().getConferenceList();
}
@@ -245,14 +245,14 @@ CallManager::getCurrentAudioCodecName(const std::string& callID)
}
-std::map< std::string, std::string >
+std::map<std::string, std::string>
CallManager::getCallDetails(const std::string& callID)
{
return Manager::instance().getCallDetails(callID);
}
-std::vector< std::string >
-CallManager::getCallList(void)
+std::vector<std::string>
+CallManager::getCallList()
{
return Manager::instance().getCallList();
}
@@ -301,7 +301,7 @@ CallManager::getAudioZrtpSession(const std::string& callID)
throw CallManagerException("Call id " + callID + " is not valid");
}
- sfl::AudioZrtpSession * zSession = call->getAudioRtp()->getAudioZrtpSession();
+ sfl::AudioZrtpSession * zSession = call->getAudioRtp().getAudioZrtpSession();
if (!zSession)
throw CallManagerException("Failed to get AudioZrtpSession");
diff --git a/daemon/src/dbus/callmanager.h b/daemon/src/dbus/callmanager.h
index e1650ecf96bff0c643d31464c701a809e88b97b3..e0592f32b4ec5f5a9134e9238c6a85c1c370f6e5 100644
--- a/daemon/src/dbus/callmanager.h
+++ b/daemon/src/dbus/callmanager.h
@@ -85,7 +85,7 @@ class CallManager
void transfer(const std::string& callID, const std::string& to);
void attendedTransfer(const std::string& transferID, const std::string& targetID);
std::map< std::string, std::string > getCallDetails(const std::string& callID);
- std::vector< std::string > getCallList(void);
+ std::vector< std::string > getCallList();
std::string getCurrentCallID();
/* Conference related methods */
@@ -98,7 +98,7 @@ class CallManager
void hangUpConference(const std::string& confID);
void holdConference(const std::string& confID);
void unholdConference(const std::string& confID);
- std::vector< std::string > getConferenceList(void);
+ std::vector< std::string > getConferenceList();
std::vector< std::string > getParticipantList(const std::string& confID);
std::map< std::string, std::string > getConferenceDetails(const std::string& callID);
diff --git a/daemon/src/dbus/configurationmanager.cpp b/daemon/src/dbus/configurationmanager.cpp
index 88806dd6a1ac3371b0af8093e77148346e4a48fa..4abab73d4d1076e3b5473050ae47cd271b95eb56 100644
--- a/daemon/src/dbus/configurationmanager.cpp
+++ b/daemon/src/dbus/configurationmanager.cpp
@@ -53,7 +53,7 @@ ConfigurationManager::ConfigurationManager(DBus::Connection& connection) :
shortcutsKeys.push_back("toggle_hold");
}
-std::map<std::string, std::string> ConfigurationManager::getIp2IpDetails(void)
+std::map<std::string, std::string> ConfigurationManager::getIp2IpDetails()
{
std::map<std::string, std::string> ip2ipAccountDetails;
SIPAccount *sipaccount = static_cast<SIPAccount *>(Manager::instance().getAccount(IP2IP_PROFILE));
@@ -161,7 +161,7 @@ std::vector<std::string> ConfigurationManager::getAccountList()
* Send the list of all codecs loaded to the client through DBus.
* Can stay global, as only the active codecs will be set per accounts
*/
-std::vector<int32_t > ConfigurationManager::getAudioCodecList(void)
+std::vector<int32_t > ConfigurationManager::getAudioCodecList()
{
std::vector<int32_t> list(Manager::instance().audioCodecFactory.getAudioCodecList());
@@ -171,7 +171,7 @@ std::vector<int32_t > ConfigurationManager::getAudioCodecList(void)
return list;
}
-std::vector<std::string> ConfigurationManager::getSupportedTlsMethod(void)
+std::vector<std::string> ConfigurationManager::getSupportedTlsMethod()
{
std::vector<std::string> method;
method.push_back("Default");
@@ -265,14 +265,14 @@ int32_t ConfigurationManager::getAudioDeviceIndex(const std::string& name)
return Manager::instance().getAudioDeviceIndex(name);
}
-std::string ConfigurationManager::getCurrentAudioOutputPlugin(void)
+std::string ConfigurationManager::getCurrentAudioOutputPlugin()
{
_debug("ConfigurationManager: Get audio plugin %s", Manager::instance().getCurrentAudioOutputPlugin().c_str());
return Manager::instance().getCurrentAudioOutputPlugin();
}
-std::string ConfigurationManager::getNoiseSuppressState(void)
+std::string ConfigurationManager::getNoiseSuppressState()
{
return Manager::instance().getNoiseSuppressState();
}
@@ -282,7 +282,7 @@ void ConfigurationManager::setNoiseSuppressState(const std::string& state)
Manager::instance().setNoiseSuppressState(state);
}
-std::string ConfigurationManager::getEchoCancelState(void)
+std::string ConfigurationManager::getEchoCancelState()
{
return Manager::instance().getEchoCancelState() ? "enabled" : "disabled";
}
@@ -292,7 +292,7 @@ void ConfigurationManager::setEchoCancelState(const std::string& state)
Manager::instance().setEchoCancelState(state);
}
-int ConfigurationManager::getEchoCancelTailLength(void)
+int ConfigurationManager::getEchoCancelTailLength()
{
return Manager::instance().getEchoCancelTailLength();
}
@@ -302,7 +302,7 @@ void ConfigurationManager::setEchoCancelTailLength(const int32_t& length)
Manager::instance().setEchoCancelTailLength(length);
}
-int ConfigurationManager::getEchoCancelDelay(void)
+int ConfigurationManager::getEchoCancelDelay()
{
return Manager::instance().getEchoCancelDelay();
}
@@ -312,12 +312,12 @@ void ConfigurationManager::setEchoCancelDelay(const int32_t& delay)
Manager::instance().setEchoCancelDelay(delay);
}
-int32_t ConfigurationManager::isIax2Enabled(void)
+int32_t ConfigurationManager::isIax2Enabled()
{
return HAVE_IAX;
}
-std::string ConfigurationManager::getRecordPath(void)
+std::string ConfigurationManager::getRecordPath()
{
return Manager::instance().getRecordPath();
}
@@ -327,7 +327,7 @@ void ConfigurationManager::setRecordPath(const std::string& recPath)
Manager::instance().setRecordPath(recPath);
}
-bool ConfigurationManager::getIsAlwaysRecording(void)
+bool ConfigurationManager::getIsAlwaysRecording()
{
return Manager::instance().getIsAlwaysRecording();
}
@@ -337,7 +337,7 @@ void ConfigurationManager::setIsAlwaysRecording(const bool& rec)
Manager::instance().setIsAlwaysRecording(rec);
}
-int32_t ConfigurationManager::getHistoryLimit(void)
+int32_t ConfigurationManager::getHistoryLimit()
{
return Manager::instance().getHistoryLimit();
}
@@ -352,22 +352,22 @@ void ConfigurationManager::setAudioManager(const std::string& api)
Manager::instance().setAudioManager(api);
}
-std::string ConfigurationManager::getAudioManager(void)
+std::string ConfigurationManager::getAudioManager()
{
return Manager::instance().getAudioManager();
}
-void ConfigurationManager::setMailNotify(void)
+void ConfigurationManager::setMailNotify()
{
Manager::instance().setMailNotify();
}
-int32_t ConfigurationManager::getMailNotify(void)
+int32_t ConfigurationManager::getMailNotify()
{
return Manager::instance().getMailNotify();
}
-std::map<std::string, int32_t> ConfigurationManager::getAddressbookSettings(void)
+std::map<std::string, int32_t> ConfigurationManager::getAddressbookSettings()
{
return Manager::instance().getAddressbookSettings();
}
@@ -377,7 +377,7 @@ void ConfigurationManager::setAddressbookSettings(const std::map<std::string, in
Manager::instance().setAddressbookSettings(settings);
}
-std::vector<std::string> ConfigurationManager::getAddressbookList(void)
+std::vector<std::string> ConfigurationManager::getAddressbookList()
{
return Manager::instance().getAddressbookList();
}
@@ -388,7 +388,7 @@ void ConfigurationManager::setAddressbookList(
Manager::instance().setAddressbookList(list);
}
-std::map<std::string, std::string> ConfigurationManager::getHookSettings(void)
+std::map<std::string, std::string> ConfigurationManager::getHookSettings()
{
return Manager::instance().getHookSettings();
}
@@ -404,7 +404,7 @@ void ConfigurationManager::setAccountsOrder(const std::string& order)
Manager::instance().setAccountsOrder(order);
}
-std::vector<std::string> ConfigurationManager::getHistory(void)
+std::vector<std::string> ConfigurationManager::getHistory()
{
return Manager::instance().getHistorySerialized();
}
@@ -420,12 +420,12 @@ std::string ConfigurationManager::getAddrFromInterfaceName(
return SIPVoIPLink::getInterfaceAddrFromName(interface);
}
-std::vector<std::string> ConfigurationManager::getAllIpInterface(void)
+std::vector<std::string> ConfigurationManager::getAllIpInterface()
{
return SIPVoIPLink::getAllIpInterface();
}
-std::vector<std::string> ConfigurationManager::getAllIpInterfaceByName(void)
+std::vector<std::string> ConfigurationManager::getAllIpInterfaceByName()
{
return SIPVoIPLink::getAllIpInterfaceByName();
}
diff --git a/daemon/src/dbus/configurationmanager.h b/daemon/src/dbus/configurationmanager.h
index d8574e4f788ab6399e24e17df21e2e64e83b7678..f1516b0ad1f828bf0eec760ee2d2c41f5db5625b 100644
--- a/daemon/src/dbus/configurationmanager.h
+++ b/daemon/src/dbus/configurationmanager.h
@@ -70,10 +70,10 @@ class ConfigurationManager
std::vector< std::string > getAccountList();
void sendRegister(const std::string& accoundID , const int32_t& expire);
- std::map< std::string, std::string > getTlsSettingsDefault(void);
+ std::map< std::string, std::string > getTlsSettingsDefault();
- std::vector< int32_t > getAudioCodecList(void);
- std::vector< std::string > getSupportedTlsMethod(void);
+ std::vector< int32_t > getAudioCodecList();
+ std::vector< std::string > getSupportedTlsMethod();
std::vector< std::string > getAudioCodecDetails(const int32_t& payload);
std::vector< int32_t > getActiveAudioCodecList(const std::string& accountID);
void setActiveAudioCodecList(const std::vector< std::string >& list, const std::string& accountID);
@@ -87,60 +87,60 @@ class ConfigurationManager
std::vector< std::string > getAudioInputDeviceList();
std::vector< std::string > getCurrentAudioDevicesIndex();
int32_t getAudioDeviceIndex(const std::string& name);
- std::string getCurrentAudioOutputPlugin(void);
- std::string getNoiseSuppressState(void);
+ std::string getCurrentAudioOutputPlugin();
+ std::string getNoiseSuppressState();
void setNoiseSuppressState(const std::string& state);
- std::string getEchoCancelState(void);
+ std::string getEchoCancelState();
void setEchoCancelState(const std::string& state);
void setEchoCancelTailLength(const int32_t& length);
- int getEchoCancelTailLength(void);
+ int getEchoCancelTailLength();
void setEchoCancelDelay(const int32_t& length);
- int getEchoCancelDelay(void);
+ int getEchoCancelDelay();
- std::string getAudioManager(void);
+ std::string getAudioManager();
void setAudioManager(const std::string& api);
- int32_t isIax2Enabled(void);
- std::string getRecordPath(void);
+ int32_t isIax2Enabled();
+ std::string getRecordPath();
void setRecordPath(const std::string& recPath);
- bool getIsAlwaysRecording(void);
+ bool getIsAlwaysRecording();
void setIsAlwaysRecording(const bool& rec);
void setHistoryLimit(const int32_t& days);
- int32_t getHistoryLimit(void);
+ int32_t getHistoryLimit();
- int32_t getMailNotify(void);
- void setMailNotify(void);
+ int32_t getMailNotify();
+ void setMailNotify();
- std::map<std::string, int32_t> getAddressbookSettings(void);
+ std::map<std::string, int32_t> getAddressbookSettings();
void setAddressbookSettings(const std::map<std::string, int32_t>& settings);
- std::vector< std::string > getAddressbookList(void);
+ std::vector< std::string > getAddressbookList();
void setAddressbookList(const std::vector< std::string >& list);
void setAccountsOrder(const std::string& order);
- std::map<std::string, std::string> getHookSettings(void);
+ std::map<std::string, std::string> getHookSettings();
void setHookSettings(const std::map<std::string, std::string>& settings);
- std::vector<std::string> getHistory(void);
+ std::vector<std::string> getHistory();
void setHistory(const std::vector<std::string> &entries);
- std::map<std::string, std::string> getTlsSettings(void);
+ std::map<std::string, std::string> getTlsSettings();
void setTlsSettings(const std::map< std::string, std::string >& details);
- std::map< std::string, std::string > getIp2IpDetails(void);
+ std::map< std::string, std::string > getIp2IpDetails();
std::vector< std::map< std::string, std::string > > getCredentials(const std::string& accountID);
void setCredentials(const std::string& accountID, const std::vector< std::map< std::string, std::string > >& details);
std::string getAddrFromInterfaceName(const std::string& interface);
- std::vector<std::string> getAllIpInterface(void);
- std::vector<std::string> getAllIpInterfaceByName(void);
+ std::vector<std::string> getAllIpInterface();
+ std::vector<std::string> getAllIpInterfaceByName();
std::map< std::string, std::string > getShortcuts();
void setShortcuts(const std::map< std::string, std::string >& shortcutsMap);
};
-
#endif//CONFIGURATIONMANAGER_H
+
diff --git a/daemon/src/dbus/instance.h b/daemon/src/dbus/instance.h
index cc64ef7a391f805fb64474d35d2bd8472b097ab9..31c82d29e96652267ef03eb6680baf43ac22797a 100644
--- a/daemon/src/dbus/instance.h
+++ b/daemon/src/dbus/instance.h
@@ -58,7 +58,7 @@ class Instance
void Register(const int32_t& pid, const std::string& name);
void Unregister(const int32_t& pid);
- int32_t getRegistrationCount(void);
+ int32_t getRegistrationCount();
};
diff --git a/daemon/src/fileutils.cpp b/daemon/src/fileutils.cpp
index 29d1149bed5bd69042cb22535196884365f79e0d..60279ec6393284627c650804b739a785ac15c0c7 100644
--- a/daemon/src/fileutils.cpp
+++ b/daemon/src/fileutils.cpp
@@ -38,7 +38,7 @@ void set_program_dir(char *program_path)
program_dir = dirname(program_path);
}
-const char *get_program_dir(void)
+const char *get_program_dir()
{
return program_dir;
}
diff --git a/daemon/src/history/historyitem.cpp b/daemon/src/history/historyitem.cpp
index 1221c82fa42c54a27150f7d2e8ae91822e292a7b..130094aaa810a84cb6ca657670f8894b24d77f7d 100644
--- a/daemon/src/history/historyitem.cpp
+++ b/daemon/src/history/historyitem.cpp
@@ -38,18 +38,17 @@
#define ITEM_SEPARATOR "|"
HistoryItem::HistoryItem(std::string timestamp_start, CallType call_type, std::string timestamp_stop, std::string name, std::string number, std::string id, std::string account_id, std::string recording, std::string confID, std::string timeAdded)
- : _timestamp_start(timestamp_start),
- _timestamp_stop(timestamp_stop),
- _call_type(call_type),
- _name(name),
- _number(number),
- _id(id),
- _account_id(account_id),
- _recording_file(recording),
- _confID(confID),
- _timeAdded(timeAdded)
-{
-}
+ : timestamp_start_(timestamp_start),
+ timestamp_stop_(timestamp_stop),
+ call_type_(call_type),
+ name_(name),
+ number_(number),
+ id_(id),
+ account_id_(account_id),
+ recording_file_(recording),
+ confID_(confID),
+ timeAdded_(timeAdded)
+{}
HistoryItem::HistoryItem(std::string serialized_form)
@@ -63,38 +62,37 @@ HistoryItem::HistoryItem(std::string serialized_form)
switch (indice) {
case 0: // The call type
- _call_type = (CallType) atoi(tmp.c_str());
+ call_type_ = (CallType) atoi(tmp.c_str());
break;
case 1: // The number field
- _number = tmp;
+ number_ = tmp;
break;
case 2: // The name field
- _name = tmp;
-
- if (_name == "empty")
- _name = "";
+ name_ = tmp;
+ if (name_ == "empty")
+ name_ = "";
break;
case 3: // The start timestamp
- _timestamp_start = tmp;
+ timestamp_start_ = tmp;
break;
case 4: // The end timestamp
- _timestamp_stop = tmp;
+ timestamp_stop_ = tmp;
break;
case 5: // The ID
- _id = tmp;
+ id_ = tmp;
break;
case 6: // The account ID
- _account_id = tmp;
+ account_id_ = tmp;
break;
case 7: // The recorded file name
- _recording_file = tmp;
+ recording_file_ = tmp;
break;
case 8: // The conference ID
- _confID = tmp;
+ confID_ = tmp;
break;
case 9: // The time
- _timeAdded = tmp;
+ timeAdded_ = tmp;
break;
default: // error
_error("Unserialized form %d not recognized\n", indice);
@@ -105,11 +103,6 @@ HistoryItem::HistoryItem(std::string serialized_form)
}
}
-HistoryItem::~HistoryItem()
-{
- // TODO
-}
-
bool HistoryItem::save(Conf::ConfigTree **history)
{
std::stringstream section;
@@ -118,53 +111,39 @@ bool HistoryItem::save(Conf::ConfigTree **history)
// The section is : "[" + timestamp = "]"
section << rand();
std::string sectionstr = section.str();
- call_type << _call_type;
-
- /*
- _error("-- Unserialized type: %s", call_type.str().c_str());
- _error("-- Unserialized time start: %s", _timestamp_start.c_str());
- _error("-- Unserialized time stop: %s", _timestamp_stop.c_str());
- _error("-- Unserialized number: %s", _number.c_str());
- _error("-- Unserialized id: %s", _id.c_str());
- _error("-- Unserialized account: %s", _account_id.c_str());
- _error("-- Unserialized name: %s", _name.c_str());
- _error("-- Unserialized record file: %s", _recording_file.c_str());
- _error("-- Unserialized conference id:%s", _confID.c_str());
- _error("-- Unserialized time added: %s", _timeAdded.c_str());
- */
+ call_type << call_type_;
return (*history)->setConfigTreeItem(sectionstr, "type", call_type.str())
- && (*history)->setConfigTreeItem(sectionstr, "timestamp_start", _timestamp_start)
- && (*history)->setConfigTreeItem(sectionstr, "timestamp_stop", _timestamp_stop)
- && (*history)->setConfigTreeItem(sectionstr, "number", _number)
- && (*history)->setConfigTreeItem(sectionstr, "id", _id)
- && (*history)->setConfigTreeItem(sectionstr, "accountid", _account_id)
- && (*history)->setConfigTreeItem(sectionstr, "name", _name)
- && (*history)->setConfigTreeItem(sectionstr, "recordfile", _recording_file)
- && (*history)->setConfigTreeItem(sectionstr, "confid", _confID)
- && (*history)->setConfigTreeItem(sectionstr, "timeadded", _timeAdded);
+ && (*history)->setConfigTreeItem(sectionstr, "timestamp_start", timestamp_start_)
+ && (*history)->setConfigTreeItem(sectionstr, "timestamp_stop", timestamp_stop_)
+ && (*history)->setConfigTreeItem(sectionstr, "number", number_)
+ && (*history)->setConfigTreeItem(sectionstr, "id", id_)
+ && (*history)->setConfigTreeItem(sectionstr, "accountid", account_id_)
+ && (*history)->setConfigTreeItem(sectionstr, "name", name_)
+ && (*history)->setConfigTreeItem(sectionstr, "recordfile", recording_file_)
+ && (*history)->setConfigTreeItem(sectionstr, "confid", confID_)
+ && (*history)->setConfigTreeItem(sectionstr, "timeadded", timeAdded_);
}
-std::string HistoryItem::serialize(void)
+std::string HistoryItem::serialize()
{
- std::stringstream res;
-
// Replace empty string with a valid standard string value
- std::string name(_name);
+ std::string name(name_);
- if (name == "")
+ if (name.empty())
name = "empty";
// For the account ID, check also if the accountID corresponds to an existing account
// ie the account may have been removed
- std::string accountID(_account_id);
+ std::string accountID(account_id_);
- if (_account_id == "" || not valid_account(_account_id))
+ if (account_id_.empty() or not valid_account(account_id_))
accountID = "empty";
+ std::stringstream res;
// Serialize it
- res << _call_type << ITEM_SEPARATOR << _number << ITEM_SEPARATOR << name << ITEM_SEPARATOR << _timestamp_start << ITEM_SEPARATOR << _timestamp_stop
- << ITEM_SEPARATOR << _id << ITEM_SEPARATOR << accountID << ITEM_SEPARATOR << _recording_file << ITEM_SEPARATOR << _confID << ITEM_SEPARATOR << _timeAdded;
+ res << call_type_ << ITEM_SEPARATOR << number_ << ITEM_SEPARATOR << name << ITEM_SEPARATOR << timestamp_start_ << ITEM_SEPARATOR << timestamp_stop_
+ << ITEM_SEPARATOR << id_ << ITEM_SEPARATOR << accountID << ITEM_SEPARATOR << recording_file_ << ITEM_SEPARATOR << confID_ << ITEM_SEPARATOR << timeAdded_;
return res.str();
}
diff --git a/daemon/src/history/historyitem.h b/daemon/src/history/historyitem.h
index d4788fe8ed18979baa0abdfc3dfabeb923772fe9..83b136db8c17a94b2ec5a8f77797f0a0496a09b3 100644
--- a/daemon/src/history/historyitem.h
+++ b/daemon/src/history/historyitem.h
@@ -30,12 +30,11 @@
* as that of the covered work.
*/
-#ifndef _HISTORY_ITEM
-#define _HISTORY_ITEM
+#ifndef HISTORY_ITEM_H_
+#define HISTORY_ITEM_H_
#include <string>
#include <config/config.h>
-#include <iostream>
typedef enum CallType {
CALL_MISSED,
@@ -43,7 +42,6 @@ typedef enum CallType {
CALL_OUTGOING
} CallType;
-
class HistoryItem {
public:
@@ -69,21 +67,15 @@ class HistoryItem {
*/
HistoryItem(std::string="");
- /*
- * Destructor
- */
- ~HistoryItem();
-
std::string get_timestamp() const {
- return _timestamp_start;
+ return timestamp_start_;
}
bool save(Conf::ConfigTree **history);
- std::string serialize(void);
+ std::string serialize();
private:
-
/*
* @return true if the account ID corresponds to a loaded account
*/
@@ -92,45 +84,45 @@ class HistoryItem {
/*
* Timestamp representing the date of the call
*/
- std::string _timestamp_start;
- std::string _timestamp_stop;
+ std::string timestamp_start_;
+ std::string timestamp_stop_;
/*
* Represents the type of call
* Has be either CALL_MISSED, CALL_INCOMING or CALL_OUTGOING
*/
- CallType _call_type;
+ CallType call_type_;
/*
* The information about the callee/caller, depending on the type of call.
*/
- std::string _name;
- std::string _number;
+ std::string name_;
+ std::string number_;
/**
* The identifier fo this item
*/
- std::string _id;
+ std::string id_;
/*
* The account the call was made with
*/
- std::string _account_id;
+ std::string account_id_;
/**
* Wether or not a recording exist for this call
*/
- std::string _recording_file;
+ std::string recording_file_;
/**
* The conference ID for this call (if any)
*/
- std::string _confID;
+ std::string confID_;
/**
* Time added to conference
*/
- std::string _timeAdded;
+ std::string timeAdded_;
};
diff --git a/daemon/src/history/historymanager.cpp b/daemon/src/history/historymanager.cpp
index e82d8d1f92e2a262e96fc3d34a6a7ccdbf6bbd44..8989f3dc4f5d47c7c23d45525c128f2a3f1f345e 100644
--- a/daemon/src/history/historymanager.cpp
+++ b/daemon/src/history/historymanager.cpp
@@ -48,91 +48,71 @@ static void free_history(HistoryItemMap &history_items)
} // end anonymous namespace
HistoryManager::HistoryManager()
- : _history_loaded(false),
- _history_path("")
+ : history_loaded_(false),
+ history_path_("")
{
}
HistoryManager::~HistoryManager()
{
- free_history(_history_items);
+ free_history(history_items_);
}
int HistoryManager::load_history(int limit, std::string path)
{
Conf::ConfigTree history_list;
-
- _debug("HistoryManager: Load history");
-
create_history_path(path);
load_history_from_file(&history_list);
return load_history_items_map(&history_list, limit);
}
-bool HistoryManager::save_history(void)
+bool HistoryManager::save_history()
{
Conf::ConfigTree history_list;
-
- _debug("HistoryManager: Save history");
-
save_history_items_map(&history_list);
return save_history_to_file(&history_list);
}
bool HistoryManager::load_history_from_file(Conf::ConfigTree *history_list)
{
- _debug("HistoryManager: Load history from file %s", _history_path.c_str());
+ _debug("HistoryManager: Load history from file %s", history_path_.c_str());
- int exist = history_list->populateFromFile(_history_path.c_str());
- _history_loaded = (exist == 2) ? false : true;
+ int exist = history_list->populateFromFile(history_path_.c_str());
+ history_loaded_ = (exist == 2) ? false : true;
- return _history_loaded;
+ return history_loaded_;
}
int HistoryManager::load_history_items_map(Conf::ConfigTree *history_list, int limit)
{
-
- short nb_items = 0;
- Conf::TokenList sections;
- HistoryItem *item;
- Conf::TokenList::iterator iter;
- std::string number, name, callID, accountID, timestamp_start, timestamp_stop, recording_file, confID, time_added;
- CallType type;
- int history_limit;
- time_t current_timestamp;
-
- _debug("HistoryManager: Load history items");
+ using std::string;
// We want to save only the items recent enough (ie compared to CONFIG_HISTORY_LIMIT)
// Get the current timestamp
- (void) time(¤t_timestamp);
- history_limit = get_unix_timestamp_equivalent(limit);
-
- sections = history_list->getSections();
- iter = sections.begin();
-
- while (iter != sections.end()) {
-
- type = (CallType) getConfigInt(*iter, "type", history_list);
- timestamp_start = getConfigString(*iter, "timestamp_start", history_list);
- timestamp_stop = getConfigString(*iter, "timestamp_stop", history_list);
- name = getConfigString(*iter, "name", history_list);
- number = getConfigString(*iter, "number", history_list);
- callID = getConfigString(*iter, "id", history_list);
- accountID = getConfigString(*iter, "accountid", history_list);
- recording_file = getConfigString(*iter, "recordfile", history_list);
- confID = getConfigString(*iter, "confid", history_list);
- time_added = getConfigString(*iter, "timeadded", history_list);
+ time_t current_timestamp;
+ time(¤t_timestamp);
+ int history_limit = get_unix_timestamp_equivalent(limit);
+
+ Conf::TokenList sections(history_list->getSections());
+ int nb_items = 0;
+ for (Conf::TokenList::iterator iter = sections.begin(); iter != sections.end(); ++iter) {
+ CallType type = (CallType) getConfigInt(*iter, "type", history_list);
+ string timestamp_start = getConfigString(*iter, "timestamp_start", history_list);
+ string timestamp_stop = getConfigString(*iter, "timestamp_stop", history_list);
+ string name = getConfigString(*iter, "name", history_list);
+ string number = getConfigString(*iter, "number", history_list);
+ string callID = getConfigString(*iter, "id", history_list);
+ string accountID = getConfigString(*iter, "accountid", history_list);
+ string recording_file = getConfigString(*iter, "recordfile", history_list);
+ string confID = getConfigString(*iter, "confid", history_list);
+ string time_added = getConfigString(*iter, "timeadded", history_list);
// Make a check on the start timestamp to know it is loadable according to CONFIG_HISTORY_LIMIT
-
if (atoi(timestamp_start.c_str()) >= ((int) current_timestamp - history_limit)) {
- item = new HistoryItem(timestamp_start, type, timestamp_stop, name, number, callID, accountID, recording_file, confID, time_added);
+ HistoryItem *item = new HistoryItem(timestamp_start, type, timestamp_stop, name, number, callID, accountID, recording_file, confID, time_added);
add_new_history_entry(item);
- nb_items ++;
+ ++nb_items;
}
-
- iter ++;
}
return nb_items;
@@ -141,32 +121,22 @@ int HistoryManager::load_history_items_map(Conf::ConfigTree *history_list, int l
bool HistoryManager::save_history_to_file(Conf::ConfigTree *history_list)
{
- _debug("HistoryManager: Saving history in XDG directory: %s", _history_path.data());
- return history_list->saveConfigTree(_history_path.data());
+ _debug("HistoryManager: Saving history in XDG directory: %s", history_path_.c_str());
+ return history_list->saveConfigTree(history_path_.data());
}
int HistoryManager::save_history_items_map(Conf::ConfigTree *history_list)
{
- HistoryItemMap::iterator iter;
- HistoryItem *item;
int items_saved = 0;
-
- _debug("HistoryManager: Save history items map");
-
- iter = _history_items.begin();
-
- while (iter != _history_items.end()) {
- item = *iter;
+ for (HistoryItemMap::iterator iter = history_items_.begin(); iter != history_items_.end(); ++iter) {
+ HistoryItem *item = *iter;
if (item) {
if (item->save(&history_list))
- items_saved ++;
- } else {
+ ++items_saved;
+ } else
_debug("can't save NULL history item\n");
- }
-
- iter ++;
}
return items_saved;
@@ -175,7 +145,7 @@ int HistoryManager::save_history_items_map(Conf::ConfigTree *history_list)
void HistoryManager::add_new_history_entry(HistoryItem *new_item)
{
// Add it in the map
- _history_items.push_back(new_item);
+ history_items_.push_back(new_item);
}
int HistoryManager::create_history_path(std::string path)
@@ -185,8 +155,7 @@ int HistoryManager::create_history_path(std::string path)
xdg_data = std::string(HOMEDIR) + DIR_SEPARATOR_STR + ".local/share/sflphone";
- if (path == "") {
-
+ if (path.empty()) {
// If the environment variable is set (not null and not empty), we'll use it to save the history
// Else we 'll the standard one, ie: XDG_DATA_HOME = $HOMEDIR/.local/share/sflphone
if (XDG_DATA_HOME != NULL) {
@@ -205,11 +174,10 @@ int HistoryManager::create_history_path(std::string path)
}
// Load user's history
- _history_path = userdata + DIR_SEPARATOR_STR + "history";
+ history_path_ = userdata + DIR_SEPARATOR_STR + "history";
} else
set_history_path(path);
-
return 0;
}
@@ -219,7 +187,7 @@ HistoryManager::getConfigInt(const std::string& section, const std::string& name
{
try {
return history_list->getConfigTreeItemIntValue(section, name);
- } catch (Conf::ConfigTreeItemException& e) {
+ } catch (const Conf::ConfigTreeItemException& e) {
throw e;
}
@@ -231,21 +199,21 @@ HistoryManager::getConfigString(const std::string& section, const std::string& n
{
try {
return history_list->getConfigTreeItemValue(section, name);
- } catch (Conf::ConfigTreeItemException& e) {
+ } catch (const Conf::ConfigTreeItemException& e) {
throw e;
}
return "";
}
-std::vector<std::string> HistoryManager::get_history_serialized(void)
+std::vector<std::string> HistoryManager::get_history_serialized()
{
std::vector<std::string> serialized;
HistoryItemMap::iterator iter;
_debug("HistoryManager: Get history serialized");
- for (iter = _history_items.begin(); iter != _history_items.end(); ++iter) {
+ for (iter = history_items_.begin(); iter != history_items_.end(); ++iter) {
HistoryItem *current = *iter;
if (current)
@@ -258,40 +226,31 @@ std::vector<std::string> HistoryManager::get_history_serialized(void)
int HistoryManager::set_serialized_history(std::vector<std::string> history, int limit)
{
- std::vector<std::string>::iterator iter;
- HistoryItem *new_item;
- int items_added = 0;
int history_limit;
time_t current_timestamp;
_debug("HistoryManager: Set serialized history");
// Clear the existing history
- free_history(_history_items);
+ free_history(history_items_);
// We want to save only the items recent enough (ie compared to CONFIG_HISTORY_LIMIT)
// Get the current timestamp
- (void) time(¤t_timestamp);
+ time(¤t_timestamp);
history_limit = get_unix_timestamp_equivalent(limit);
- for (iter = history.begin() ; iter != history.end() ; iter ++) {
- new_item = new HistoryItem(*iter);
-
- if (new_item == NULL) {
- _error("HistoryManager: Error: Could not create history item");
- }
-
+ int items_added = 0;
+ for (std::vector<std::string>::iterator iter = history.begin() ; iter != history.end() ; ++iter) {
+ HistoryItem *new_item = new HistoryItem(*iter);
int item_timestamp = atoi(new_item->get_timestamp().c_str());
if (item_timestamp >= ((int) current_timestamp - history_limit)) {
add_new_history_entry(new_item);
- items_added ++;
- } else {
+ ++items_added;
+ } else
delete new_item;
- }
}
return items_added;
}
-
diff --git a/daemon/src/history/historymanager.h b/daemon/src/history/historymanager.h
index 578cd5e26df057ef163a40712ef08e8a7c3a1670..a27ae247a4b7935b7ee97402956a3c28a2aacf40 100644
--- a/daemon/src/history/historymanager.h
+++ b/daemon/src/history/historymanager.h
@@ -63,7 +63,7 @@ class HistoryManager {
/**
*@return bool True if the history has been successfully saved in the file
*/
- bool save_history(void);
+ bool save_history();
/*
* Load the history from a file to the dedicated data structure
@@ -88,22 +88,22 @@ class HistoryManager {
/**
*@return bool True if the history file has been successfully read
*/
- bool is_loaded(void) const {
- return _history_loaded;
+ bool is_loaded() const {
+ return history_loaded_;
}
void set_history_path(const std::string &filename) {
- _history_path = filename;
+ history_path_ = filename;
}
/*
*@return int The number of items found in the history file
*/
- int get_history_size(void) const {
- return _history_items.size();
+ int get_history_size() const {
+ return history_items_.size();
}
- std::vector<std::string> get_history_serialized(void);
+ std::vector<std::string> get_history_serialized();
int set_serialized_history(std::vector<std::string> history, int limit);
@@ -129,18 +129,18 @@ class HistoryManager {
/*
* Map containing the history items
*/
- HistoryItemMap _history_items;
+ HistoryItemMap history_items_;
/*
* History has been loaded
*/
- bool _history_loaded;
+ bool history_loaded_;
/*
* The path to the history file
*/
- std::string _history_path;
+ std::string history_path_;
friend class HistoryTest;
};
diff --git a/daemon/src/iax/iaxaccount.cpp b/daemon/src/iax/iaxaccount.cpp
index 778d56723e21e854cc825a3a3a9654cce2a61f21..a30705662ae02888796bfa9ec85552b7fa363ed9 100644
--- a/daemon/src/iax/iaxaccount.cpp
+++ b/daemon/src/iax/iaxaccount.cpp
@@ -84,7 +84,7 @@ void IAXAccount::serialize(Conf::YamlEmitter *emitter)
try {
emitter->serializeAccount(&accountmap);
- } catch (Conf::YamlEmitterException &e) {
+ } catch (const Conf::YamlEmitterException &e) {
_error("ConfigTree: %s", e.what());
}
}
diff --git a/daemon/src/iax/iaxaccount.h b/daemon/src/iax/iaxaccount.h
index dc866888e665052be2d666cd7d6d42b52036fa13..60b9a1acee27eb704046dc45685460177534d457 100644
--- a/daemon/src/iax/iaxaccount.h
+++ b/daemon/src/iax/iaxaccount.h
@@ -40,7 +40,6 @@
class IAXAccount : public Account {
public:
IAXAccount(const std::string& accountID);
-
~IAXAccount();
virtual void serialize(Conf::YamlEmitter *emitter);
@@ -51,30 +50,21 @@ class IAXAccount : public Account {
std::map<std::string, std::string> getAccountDetails() const;
- /**
- * Actually useless, since config loading is done in init()
- */
+ // Actually useless, since config loading is done in init()
void loadConfig();
- /**
- * Register an account
- */
+ // Register an account
void registerVoIPLink();
- /**
- * Unregister an account
- */
+ // Unregister an account
void unregisterVoIPLink();
- std::string getPassword(void) const {
+ std::string getPassword() const {
return password_;
}
private:
-
- /**
- * Account login information: password
- */
+ // Account login information: password
std::string password_;
};
diff --git a/daemon/src/iax/iaxcall.cpp b/daemon/src/iax/iaxcall.cpp
index 79088235b758d14547cd719b640bd760b2b3d376..afa6622310fc35875838c3120d9d7751019dd4ca 100644
--- a/daemon/src/iax/iaxcall.cpp
+++ b/daemon/src/iax/iaxcall.cpp
@@ -37,27 +37,29 @@
namespace {
int codecToASTFormat(int c)
{
- if (c == PAYLOAD_CODEC_ULAW) return AST_FORMAT_ULAW;
-
- if (c == PAYLOAD_CODEC_GSM) return AST_FORMAT_GSM;
-
- if (c == PAYLOAD_CODEC_ALAW) return AST_FORMAT_ALAW;
-
- if (c == PAYLOAD_CODEC_ILBC_20) return AST_FORMAT_ILBC;
-
- if (c == PAYLOAD_CODEC_SPEEX_8000) return AST_FORMAT_SPEEX;
-
- _error("Codec %d not supported!", c);
- return 0;
+ switch (c) {
+ case PAYLOAD_CODEC_ULAW:
+ return AST_FORMAT_ULAW;
+ case PAYLOAD_CODEC_GSM:
+ return AST_FORMAT_GSM;
+ case PAYLOAD_CODEC_ALAW:
+ return AST_FORMAT_ALAW;
+ case PAYLOAD_CODEC_ILBC_20:
+ return AST_FORMAT_ILBC;
+ case PAYLOAD_CODEC_SPEEX_8000:
+ return AST_FORMAT_SPEEX;
+
+ default:
+ _error("Codec %d not supported!", c);
+ return 0;
+ }
}
}
IAXCall::IAXCall(const std::string& id, Call::CallType type) : Call(id, type), session(NULL)
-{
-}
+{}
-int
-IAXCall::getSupportedFormat(const std::string &accountID) const
+int IAXCall::getSupportedFormat(const std::string &accountID) const
{
Account *account = Manager::instance().getAccount(accountID);
@@ -94,18 +96,21 @@ int IAXCall::getFirstMatchingFormat(int needles, const std::string &accountID) c
return 0;
}
-int IAXCall::getAudioCodec(void)
+int IAXCall::getAudioCodec() const
{
- if (format == AST_FORMAT_ULAW) return PAYLOAD_CODEC_ULAW;
-
- if (format == AST_FORMAT_GSM) return PAYLOAD_CODEC_GSM;
-
- if (format == AST_FORMAT_ALAW) return PAYLOAD_CODEC_ALAW;
-
- if (format == AST_FORMAT_ILBC) return PAYLOAD_CODEC_ILBC_20;
-
- if (format == AST_FORMAT_SPEEX) return PAYLOAD_CODEC_SPEEX_8000;
-
- _error("IAX: Format %d not supported!", format);
- return -1;
+ switch (format) {
+ case AST_FORMAT_ULAW:
+ return PAYLOAD_CODEC_ULAW;
+ case AST_FORMAT_GSM:
+ return PAYLOAD_CODEC_GSM;
+ case AST_FORMAT_ALAW:
+ return PAYLOAD_CODEC_ALAW;
+ case AST_FORMAT_ILBC:
+ return PAYLOAD_CODEC_ILBC_20;
+ case AST_FORMAT_SPEEX:
+ return PAYLOAD_CODEC_SPEEX_8000;
+ default:
+ _error("IAX: Format %d not supported!", format);
+ return -1;
+ }
}
diff --git a/daemon/src/iax/iaxcall.h b/daemon/src/iax/iaxcall.h
index fe404da08f6aedaba33ac30a959d8712d777f5ef..e81c3bb70ebe0a98167ee35c8211388b8eec93c5 100644
--- a/daemon/src/iax/iaxcall.h
+++ b/daemon/src/iax/iaxcall.h
@@ -32,7 +32,6 @@
#define IAXCALL_H
#include "call.h"
-#include "audio/codecs/audiocodecfactory.h"
/**
* @file: iaxcall.h
@@ -69,7 +68,7 @@ class IAXCall : public Call {
*/
int getFirstMatchingFormat(int needles, const std::string &accountID) const;
- int getAudioCodec(void);
+ int getAudioCodec() const;
int format;
iax_session* session;
diff --git a/daemon/src/iax/iaxvoiplink.cpp b/daemon/src/iax/iaxvoiplink.cpp
index f89e3301dc40f18036ec7e29cf225569dd4f0882..2ee740e936fe35044738d70e58e701f3312c9254 100644
--- a/daemon/src/iax/iaxvoiplink.cpp
+++ b/daemon/src/iax/iaxvoiplink.cpp
@@ -136,7 +136,7 @@ IAXVoIPLink::getEvent()
}
void
-IAXVoIPLink::sendAudioFromMic(void)
+IAXVoIPLink::sendAudioFromMic()
{
for (CallMap::const_iterator iter = callMap_.begin(); iter != callMap_.end() ; ++iter) {
IAXCall *currentCall = dynamic_cast<IAXCall*>(iter->second);
@@ -396,7 +396,8 @@ IAXVoIPLink::carryingDTMFdigits(const std::string& id, char code)
void
IAXVoIPLink::sendTextMessage(sfl::InstantMessaging *module,
- const std::string& callID, const std::string& message,
+ const std::string& callID,
+ const std::string& message,
const std::string& /*from*/)
{
IAXCall* call = getIAXCall(callID);
@@ -439,7 +440,7 @@ IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
IAXCall*
-IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
+IAXVoIPLink::iaxFindCallBySession(iax_session* session)
{
ost::MutexLock m(callMapMutex_);
@@ -537,8 +538,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
/* Handle audio event, VOICE packet received */
-void
-IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
+void IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
{
// Skip this empty packet.
if (!event->datalen)
@@ -580,8 +580,7 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
/**
* Handle the registration process
*/
-void
-IAXVoIPLink::iaxHandleRegReply(iax_event* event)
+void IAXVoIPLink::iaxHandleRegReply(iax_event* event)
{
IAXAccount *account = dynamic_cast<IAXAccount *>(Manager::instance().getAccount(accountID_));
@@ -598,8 +597,7 @@ IAXVoIPLink::iaxHandleRegReply(iax_event* event)
nextRefreshStamp_ = time(NULL) + (event->ies.refresh ? event->ies.refresh : 60);
}
-void
-IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
+void IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
{
IAXCall *call;
std::string id;
diff --git a/daemon/src/iax/iaxvoiplink.h b/daemon/src/iax/iaxvoiplink.h
index 17f21058bf17fbf0bbca05a1153be77dca990bdb..1b29e2199e0bd486c7dfb8064aafda7f1590f92a 100644
--- a/daemon/src/iax/iaxvoiplink.h
+++ b/daemon/src/iax/iaxvoiplink.h
@@ -59,31 +59,23 @@ class IAXAccount;
class IAXVoIPLink : public VoIPLink {
public:
- /**
- * Constructor
- * @param accountID The account containing the voip link
- */
IAXVoIPLink(const std::string& accountID);
-
- /**
- * Destructor
- */
~IAXVoIPLink();
/**
* Listen to events sent by the call manager ( asterisk, etc .. )
*/
- void getEvent(void);
+ void getEvent();
/**
* Init the voip link
*/
- virtual void init(void);
+ virtual void init();
/**
* Terminate a voip link by clearing the call list
*/
- virtual void terminate(void);
+ virtual void terminate();
/**
* Send out registration
@@ -236,7 +228,7 @@ class IAXVoIPLink : public VoIPLink {
/**
* Work out the audio data from Microphone to IAX2 channel
*/
- void sendAudioFromMic(void);
+ void sendAudioFromMic();
/**
* Send an outgoing call invite to iax
diff --git a/daemon/src/im/instant_messaging.cpp b/daemon/src/im/instant_messaging.cpp
index a8403cfb046cd1a09d45170f5818c88dbffdc403..1e30e9c1b56ae4a30c56fd8624f2a64cf6a63b33 100644
--- a/daemon/src/im/instant_messaging.cpp
+++ b/daemon/src/im/instant_messaging.cpp
@@ -49,14 +49,7 @@ static void XMLCALL startElementCallback(void *userData, const char *name, const
}
static void XMLCALL endElementCallback(void * /*userData*/, const char * /*name*/)
-{
-}
-
-
-InstantMessaging::InstantMessaging() {}
-
-
-InstantMessaging::~InstantMessaging() {}
+{}
bool InstantMessaging::saveMessage(const std::string& message, const std::string& author, const std::string& id, int mode)
{
diff --git a/daemon/src/im/instant_messaging.h b/daemon/src/im/instant_messaging.h
index 347293788a798765bdacb20fbcccedc7ca0bc8f0..17018b56ca0a03eef17d1620de1c4e51eb8b4267 100644
--- a/daemon/src/im/instant_messaging.h
+++ b/daemon/src/im/instant_messaging.h
@@ -67,27 +67,15 @@ class InstantMessageException : public std::runtime_error {
};
class InstantMessaging {
-
public:
-
typedef std::map <std::string, std::string> UriEntry;
typedef std::list <UriEntry> UriList;
- /*
- * Class constructor
- */
- InstantMessaging();
-
- /*
- * Class destructor
- */
- ~InstantMessaging();
-
/**
* Return the maximum number if character for a single SIP MESSAGE.
* Longer messages should be splitted in several smaller messages using split_message
*/
- size_t getMessageMaximumSize(void) {
+ static size_t getMessageMaximumSize() {
return MAXIMUM_MESSAGE_LENGTH;
}
@@ -161,11 +149,6 @@ class InstantMessaging {
* @return A string containing the actual message
*/
std::string findTextMessage(std::string& text);
-
- private:
-
- InstantMessaging(const InstantMessaging&); //No Copy Constructor
- InstantMessaging& operator= (const InstantMessaging&); //No Assignment Operator
};
}
#endif // __INSTANT_MESSAGING_H_
diff --git a/daemon/src/managerimpl.h b/daemon/src/managerimpl.h
index eda4145baf501fe05235d67dab86ecdb3f500725..fd17a44d6e9dc65bf105a7e0435877a320994c00 100644
--- a/daemon/src/managerimpl.h
+++ b/daemon/src/managerimpl.h
@@ -730,13 +730,13 @@ class ManagerImpl {
* Get the desktop mail notification level
* @return int The mail notification level
*/
- int32_t getMailNotify(void) const;
+ int32_t getMailNotify() const;
/**
* Get the list of the active codecs
* @return std::vector< ::std::string > The list of active codecs
*/
- std::vector<std::string> getActiveCodecList(void) const;
+ std::vector<std::string> getActiveCodecList() const;
/**
* Retrieve in the configuration tree the value of a parameter in a specific section
diff --git a/daemon/src/sip/sdes_negotiator.cpp b/daemon/src/sip/sdes_negotiator.cpp
index 0bb8e8d587868d0f1ab806ad79c3cfcb91ff06aa..213a676532d274b21b64cdc93801b3f7c24f05d6 100644
--- a/daemon/src/sip/sdes_negotiator.cpp
+++ b/daemon/src/sip/sdes_negotiator.cpp
@@ -41,13 +41,11 @@ using namespace sfl;
SdesNegotiator::SdesNegotiator(const std::vector<CryptoSuiteDefinition>& localCapabilites,
const std::vector<std::string>& remoteAttribute) :
- _remoteAttribute(remoteAttribute),
- _localCapabilities(localCapabilites)
-{
-
-}
+ remoteAttribute_(remoteAttribute),
+ localCapabilities_(localCapabilites)
+{}
-std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
+std::vector<CryptoAttribute *> SdesNegotiator::parse()
{
// The patterns below try to follow
// the ABNF grammar rules described in
@@ -62,7 +60,6 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
* keyParamsPattern;
try {
-
// used to match white space (which are used as separator)
generalSyntaxPattern = new Pattern("[\x20\x09]+", "g");
@@ -81,19 +78,7 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
"(?P<mkiValue>[0-9]+)\\:" \
"(?P<mkiLength>[0-9]{1,3})\\;?)?", "g");
- /*
- sessionParamPattern = new Pattern (
- "(?P<sessionParam>(kdr\\=[0-9]{1,2}|" \
- "UNENCRYPTED_SRTP|" \
- "UNENCRYPTED_SRTCP|" \
- "UNAUTHENTICATED_SRTP|" \
- "FEC_ORDER=(?P<fecOrder>FEC_SRTP|SRTP_FEC)|" \
- "FEC_KEY=(?P<fecKey>" + keyParamsPattern->getPattern() + ")|" \
- "WSH=(?P<wsh>[0-9]{1,2})|" \
- "(?<!\\-)[[:graph:]]+))*", "g"); // srtp-session-extension
- */
-
- } catch (compile_error& exception) {
+ } catch (const compile_error& exception) {
throw parse_error("A compile exception occured on a pattern.");
}
@@ -102,10 +87,10 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
// and parse its content
- std::vector<std::string>::iterator iter;
std::vector<CryptoAttribute *> cryptoAttributeVector;
- for (iter = _remoteAttribute.begin(); iter != _remoteAttribute.end(); iter++) {
+ for (std::vector<std::string>::iterator iter = remoteAttribute_.begin();
+ iter != remoteAttribute_.end(); ++iter) {
// Split the line into its component
// that we will analyze further down.
@@ -116,14 +101,12 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
try {
sdesLine = generalSyntaxPattern->split();
- if (sdesLine.size() < 3) {
+ if (sdesLine.size() < 3)
throw parse_error("Missing components in SDES line");
- }
- } catch (match_error& exception) {
+ } catch (const match_error& exception) {
throw parse_error("Error while analyzing the SDES line.");
}
-
// Check if the attribute starts with a=crypto
// and get the tag for this line
*tagPattern << sdesLine.at(0);
@@ -133,12 +116,11 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
if (tagPattern->matches()) {
try {
tag = tagPattern->group("tag");
- } catch (match_error& exception) {
+ } catch (const match_error& exception) {
throw parse_error("Error while parsing the tag field");
}
- } else {
+ } else
return cryptoAttributeVector;
- }
// Check if the crypto suite is valid and retreive
// its value.
@@ -149,12 +131,11 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
if (cryptoSuitePattern->matches()) {
try {
cryptoSuite = cryptoSuitePattern->group("cryptoSuite");
- } catch (match_error& exception) {
+ } catch (const match_error& exception) {
throw parse_error("Error while parsing the crypto-suite field");
}
- } else {
+ } else
return cryptoAttributeVector;
- }
// Parse one or more key-params field.
*keyParamsPattern << sdesLine.at(2);
@@ -173,28 +154,10 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
mkiValue = keyParamsPattern->group("mkiValue");
mkiLength = keyParamsPattern->group("mkiLength");
}
- } catch (match_error& exception) {
+ } catch (const match_error& exception) {
throw parse_error("Error while parsing the key-params field");
}
- /**
- * Parse the optional session-param fields
- * @TODO Implement this !
- */
- /*
- if (sdesLine.size() == 3) continue;
-
- int i;
- for (i = 3; i < sdesLine.size(); i++) {
- sessionParamPattern << sdesLine.at(i);
- while (sessionpParamPattern.matches()) {
-
- } catch (match_error& exception) {
- throw parse_error("Error while parsing the crypto-suite field");
- }
- }
- } */
-
// Add the new CryptoAttribute to the vector
CryptoAttribute * cryptoAttribute = new CryptoAttribute(tag, cryptoSuite, srtpKeyMethod, srtpKeyInfo, lifetime, mkiValue, mkiLength);
@@ -204,71 +167,40 @@ std::vector<CryptoAttribute *> SdesNegotiator::parse(void)
return cryptoAttributeVector;
}
-bool SdesNegotiator::negotiate(void)
+bool SdesNegotiator::negotiate()
{
std::vector<CryptoAttribute *> cryptoAttributeVector = parse();
std::vector<CryptoAttribute *>::iterator iter_offer = cryptoAttributeVector.begin();
- std::vector<CryptoSuiteDefinition>::iterator iter_local = _localCapabilities.begin();
+ std::vector<CryptoSuiteDefinition>::iterator iter_local = localCapabilities_.begin();
bool negotiationSuccess = false;
try {
-
while (!negotiationSuccess && (iter_offer != cryptoAttributeVector.end())) {
+ iter_local = localCapabilities_.begin();
- /*
- std::cout << "Negotiate tag: " + (*iter_offer)->getTag() << std::endl;
- std::cout << "Crypto Suite: " + (*iter_offer)->getCryptoSuite() << std::endl;
- std::cout << "SRTP Key Method: " + (*iter_offer)->getSrtpKeyMethod() << std::endl;
- std::cout << "SRTP Key Info: " + (*iter_offer)->getSrtpKeyInfo() << std::endl;
- std::cout << "Lifetime: " + (*iter_offer)->getLifetime() << std::endl;
- std::cout << "MKI Value: " + (*iter_offer)->getMkiValue() << std::endl;
- std::cout << "MKI Length: " + (*iter_offer)->getMkiLength() << std::endl;
- */
-
- iter_local = _localCapabilities.begin();
-
- while (!negotiationSuccess && (iter_local != _localCapabilities.end())) {
+ while (!negotiationSuccess && (iter_local != localCapabilities_.end())) {
if ((*iter_offer)->getCryptoSuite().compare((*iter_local).name)) {
-
negotiationSuccess = true;
- _cryptoSuite = (*iter_offer)->getCryptoSuite();
- _srtpKeyMethod = (*iter_offer)->getSrtpKeyMethod();
- _srtpKeyInfo = (*iter_offer)->getSrtpKeyInfo();
- // TODO why does there return empty strings
- // _lifetime = (*iter_offer)->getLifetime();
- // _mkiValue = (*iter_offer)->getMkiValue();
- // _mkiLength = (*iter_offer)->getMkiLength();
-
- _authTagLength = _cryptoSuite.substr(_cryptoSuite.size()-2, 2);
-
- /*
- std::cout << "Negotiate tag: " + (*iter_offer)->getTag() << std::endl;
- std::cout << "Crypto Suite: " + _cryptoSuite << std::endl;
- std::cout << "SRTP Key Method: " + _srtpKeyMethod << std::endl;
- std::cout << "SRTP Key Info: " + _srtpKeyInfo << std::endl;
- std::cout << "Lifetime: " + _lifetime << std::endl;
- std::cout << "MKI Value: " + _mkiValue << std::endl;
- std::cout << "MKI Length: " + _mkiLength << std::endl;
- std::cout << "Auth tag length: " + _authTagLength << std::endl;
- */
+ cryptoSuite_ = (*iter_offer)->getCryptoSuite();
+ srtpKeyMethod_ = (*iter_offer)->getSrtpKeyMethod();
+ srtpKeyInfo_ = (*iter_offer)->getSrtpKeyInfo();
+ authTagLength_ = cryptoSuite_.substr(cryptoSuite_.size()-2, 2);
}
iter_local++;
}
-
delete(*iter_offer);
-
iter_offer++;
}
- } catch (parse_error& exception) {
+ } catch (const parse_error& exception) {
return false;
- } catch (match_error& exception) {
+ } catch (const match_error& exception) {
return false;
}
diff --git a/daemon/src/sip/sdes_negotiator.h b/daemon/src/sip/sdes_negotiator.h
index 6c63961e40df8b001ca4b0b6059d65d7915d1817..42d2b66c96aae4c207db996c37073f17682dbea0 100644
--- a/daemon/src/sip/sdes_negotiator.h
+++ b/daemon/src/sip/sdes_negotiator.h
@@ -152,57 +152,56 @@ class SdesNegotiator {
public:
SdesNegotiator(const std::vector<CryptoSuiteDefinition>& localCapabilites, const std::vector<std::string>& remoteAttribute);
- ~SdesNegotiator() {};
- bool negotiate(void);
+ bool negotiate();
/**
* Return crypto suite after negotiation
*/
- std::string getCryptoSuite(void) const {
- return _cryptoSuite;
+ std::string getCryptoSuite() const {
+ return cryptoSuite_;
}
/**
* Return key method after negotiation (most likely inline:)
*/
- std::string getKeyMethod(void) const {
- return _srtpKeyMethod;
+ std::string getKeyMethod() const {
+ return srtpKeyMethod_;
}
/**
* Return crypto suite after negotiation
*/
- std::string getKeyInfo(void) const {
- return _srtpKeyInfo;
+ std::string getKeyInfo() const {
+ return srtpKeyInfo_;
}
/**
* Return key lifetime after negotiation
*/
- std::string getLifeTime(void) const {
- return _lifetime;
+ std::string getLifeTime() const {
+ return lifetime_;
}
/**
* Return mki value after negotiation
*/
- std::string getMkiValue(void) const {
- return _mkiValue;
+ std::string getMkiValue() const {
+ return mkiValue_;
}
/**
* Return mki length after negotiation
*/
- std::string getMkiLength(void) const {
- return _mkiLength;
+ std::string getMkiLength() const {
+ return mkiLength_;
}
/**
* Authentication tag lenth
*/
- std::string getAuthTagLength(void) const {
- return _authTagLength;
+ std::string getAuthTagLength() const {
+ return authTagLength_;
}
@@ -212,46 +211,46 @@ class SdesNegotiator {
* Multiple crypto lines can be sent, and the
* prefered method is then chosen from that list.
*/
- std::vector<std::string> _remoteAttribute;
+ std::vector<std::string> remoteAttribute_;
- std::vector<CryptoSuiteDefinition> _localCapabilities;
+ std::vector<CryptoSuiteDefinition> localCapabilities_;
/**
* Selected crypto suite after negotiation
*/
- std::string _cryptoSuite;
+ std::string cryptoSuite_;
/**
* Selected key method after negotiation (most likely inline:)
*/
- std::string _srtpKeyMethod;
+ std::string srtpKeyMethod_;
/**
* Selected crypto suite after negotiation
*/
- std::string _srtpKeyInfo;
+ std::string srtpKeyInfo_;
/**
* Selected key lifetime after negotiation
*/
- std::string _lifetime;
+ std::string lifetime_;
/**
* Selected mki value after negotiation
*/
- std::string _mkiValue;
+ std::string mkiValue_;
/**
* Selected mki length after negotiation
*/
- std::string _mkiLength;
+ std::string mkiLength_;
/**
* Authenticvation tag length in byte
*/
- std::string _authTagLength;
+ std::string authTagLength_;
- std::vector<CryptoAttribute *> parse(void);
+ std::vector<CryptoAttribute *> parse();
};
}
#endif // __SDES_NEGOTIATOR_H__
diff --git a/daemon/src/sip/sdp.cpp b/daemon/src/sip/sdp.cpp
index 89aaca09b3d0a143004020ff3753db03d06c4309..3ada56f849d44b54a8109fc76c091cdff16fdccd 100644
--- a/daemon/src/sip/sdp.cpp
+++ b/daemon/src/sip/sdp.cpp
@@ -113,7 +113,7 @@ void Sdp::setActiveRemoteSdpSession(const pjmedia_sdp_session *sdp)
_error("Sdp: Error: Could not found dtmf event from remote sdp");
}
-std::string Sdp::getCodecName(void) const
+std::string Sdp::getCodecName() const
{
try {
sfl::AudioCodec *codec = getSessionMedia();
@@ -123,7 +123,7 @@ std::string Sdp::getCodecName(void) const
}
}
-sfl::AudioCodec* Sdp::getSessionMedia(void) const
+sfl::AudioCodec* Sdp::getSessionMedia() const
{
if (sessionAudioMedia_.size() < 1)
throw SdpException("No codec description for this media");
diff --git a/daemon/src/sip/sdp.h b/daemon/src/sip/sdp.h
index 0bc8d8530b0300c793b35abac587ae4b8aa2f1a7..c6a9f78ca4cfea893d1df8cb71f10ff7fd316d53 100644
--- a/daemon/src/sip/sdp.h
+++ b/daemon/src/sip/sdp.h
@@ -74,7 +74,7 @@ class Sdp {
/**
* Accessor for the internal memory pool
*/
- pj_pool_t *getMemoryPool(void) const {
+ pj_pool_t *getMemoryPool() const {
return memPool_;
}
@@ -83,7 +83,7 @@ class Sdp {
*
* @return The structure that describes a SDP session
*/
- pjmedia_sdp_session *getLocalSdpSession(void) {
+ pjmedia_sdp_session *getLocalSdpSession() {
return localSession_;
}
@@ -92,7 +92,7 @@ class Sdp {
*
* @return The structure that describe the SDP session
*/
- pjmedia_sdp_session *getRemoteSdpSession(void) {
+ pjmedia_sdp_session *getRemoteSdpSession() {
return remoteSession_;
}
@@ -108,7 +108,7 @@ class Sdp {
*
* @return pjmedia_sdp_session The negotiated offer
*/
- pjmedia_sdp_session* getActiveLocalSdpSession(void) {
+ pjmedia_sdp_session* getActiveLocalSdpSession() {
return activeLocalSession_;
}
@@ -124,7 +124,7 @@ class Sdp {
*
* @return pjmedia_sdp_session The negotiated offer
*/
- pjmedia_sdp_session* getActiveRemoteSdpSession(void) {
+ pjmedia_sdp_session* getActiveRemoteSdpSession() {
return activeRemoteSession_;
}
@@ -132,13 +132,13 @@ class Sdp {
/**
* Return whether or not the media have been determined for this sdp session
*/
- bool hasSessionMedia(void) const;
+ bool hasSessionMedia() const;
/**
* Return the codec of the first media after negotiation
* @throw SdpException
*/
- sfl::AudioCodec* getSessionMedia(void) const;
+ sfl::AudioCodec* getSessionMedia() const;
/*
* On building an invite outside a dialog, build the local offer and create the
@@ -158,17 +158,17 @@ class Sdp {
/**
* Start the sdp negotiation.
*/
- void startNegotiation(void);
+ void startNegotiation();
/**
* Remove all media in the session media vector.
*/
- void cleanSessionMedia(void);
+ void cleanSessionMedia();
/**
* Remove all media in local media capability vector
*/
- void cleanLocalMediaCapabilities(void);
+ void cleanLocalMediaCapabilities();
/*
* Write accessor. Set the local IP address that will be used in the sdp session
@@ -180,7 +180,7 @@ class Sdp {
/*
* Read accessor. Get the local IP address
*/
- std::string getLocalIP(void) const {
+ std::string getLocalIP() const {
return localIpAddr_;
}
@@ -194,7 +194,7 @@ class Sdp {
/**
* @return The published audio port
*/
- int getLocalPublishedAudioPort(void) const {
+ int getLocalPublishedAudioPort() const {
return localAudioPort_;
}
@@ -271,7 +271,7 @@ class Sdp {
void setMediaTransportInfoFromRemoteSdp();
- std::string getCodecName(void) const;
+ std::string getCodecName() const;
void receivingAnswerAfterInitialOffer(const pjmedia_sdp_session* remote);
@@ -354,9 +354,9 @@ class Sdp {
*/
unsigned int telephoneEventPayload_;
- Sdp(const Sdp&); //No Copy Constructor
-
- Sdp& operator= (const Sdp&); //No Assignment Operator
+ // noncopyable
+ Sdp(const Sdp&);
+ Sdp& operator=(const Sdp&);
/*
* Build the sdp media section
diff --git a/daemon/src/sip/sipaccount.cpp b/daemon/src/sip/sipaccount.cpp
index e210faf8e1239c3a01e5ee73ce1e458fb88df202..0f9cfcce665b9efcefc114948dfbf4e7be11f095 100644
--- a/daemon/src/sip/sipaccount.cpp
+++ b/daemon/src/sip/sipaccount.cpp
@@ -582,7 +582,7 @@ pjsip_ssl_method SIPAccount::sslMethodStringToPjEnum(const std::string& method)
return PJSIP_SSL_UNSPECIFIED_METHOD;
}
-void SIPAccount::initTlsConfiguration(void)
+void SIPAccount::initTlsConfiguration()
{
// TLS listener is unique and should be only modified through IP2IP_PROFILE
tlsListenerPort_ = tlsPort_;
@@ -605,7 +605,7 @@ void SIPAccount::initTlsConfiguration(void)
tlsSetting_.timeout.msec = atol(tlsNegotiationTimeoutMsec_.c_str());
}
-void SIPAccount::initStunConfiguration(void)
+void SIPAccount::initStunConfiguration()
{
size_t pos;
std::string stunServer, serverName, serverPort;
@@ -654,13 +654,13 @@ bool SIPAccount::hostnameMatch(const std::string& hostname) const
return hostname == hostname_;
}
-std::string SIPAccount::getLoginName(void)
+std::string SIPAccount::getLoginName()
{
struct passwd * user_info = getpwuid(getuid());
return user_info ? user_info->pw_name : "";
}
-std::string SIPAccount::getFromUri(void) const
+std::string SIPAccount::getFromUri() const
{
std::string scheme;
std::string transport;
@@ -709,7 +709,7 @@ std::string SIPAccount::getToUri(const std::string& username) const
return "<" + scheme + username + (hostname.empty() ? "" : "@") + hostname + transport + ">";
}
-std::string SIPAccount::getServerUri(void) const
+std::string SIPAccount::getServerUri() const
{
std::string scheme;
std::string transport;
@@ -843,7 +843,7 @@ void SIPAccount::setCredentials(const std::vector<std::map<std::string, std::str
}
}
-const std::vector<std::map<std::string, std::string> > &SIPAccount::getCredentials(void)
+const std::vector<std::map<std::string, std::string> > &SIPAccount::getCredentials()
{
return credentials_;
}
@@ -858,7 +858,7 @@ std::string SIPAccount::getUserAgentName() const
return result;
}
-std::map<std::string, std::string> SIPAccount::getIp2IpDetails(void) const
+std::map<std::string, std::string> SIPAccount::getIp2IpDetails() const
{
assert(accountID_ == IP2IP_PROFILE);
std::map<std::string, std::string> ip2ipAccountDetails;
diff --git a/daemon/src/sip/sipaccount.h b/daemon/src/sip/sipaccount.h
index 03b93a87aa45cef32402a7cfe997b3345f438d85..5ca66094f5a7783f9caece76dbf8a13eed3d58fc 100644
--- a/daemon/src/sip/sipaccount.h
+++ b/daemon/src/sip/sipaccount.h
@@ -125,8 +125,8 @@ class SIPAccount : public Account {
virtual void setAccountDetails(std::map<std::string, std::string> details);
virtual std::map<std::string, std::string> getAccountDetails() const;
- std::map<std::string, std::string> getIp2IpDetails(void) const;
- std::map<std::string, std::string> getTlsSettings(void) const;
+ std::map<std::string, std::string> getIp2IpDetails() const;
+ std::map<std::string, std::string> getTlsSettings() const;
void setTlsSettings(const std::map<std::string, std::string>& details);
/**
@@ -154,12 +154,12 @@ class SIPAccount : public Account {
* @param none
* @return int The number of credentials set for this account.
*/
- unsigned getCredentialCount(void) const {
+ unsigned getCredentialCount() const {
return credentials_.size();
}
void setCredentials(const std::vector<std::map<std::string, std::string> >& details);
- const std::vector<std::map<std::string, std::string> > &getCredentials(void);
+ const std::vector<std::map<std::string, std::string> > &getCredentials();
/**
* A client sendings a REGISTER request MAY suggest an expiration
@@ -168,7 +168,7 @@ class SIPAccount : public Account {
*
* @return the expiration value.
*/
- unsigned getRegistrationExpire(void) const {
+ unsigned getRegistrationExpire() const {
if (registrationExpire_ == 0)
return PJSIP_REGC_EXPIRATION_NOT_SPECIFIED;
@@ -178,7 +178,7 @@ class SIPAccount : public Account {
/**
* Doubles the Expiration Interval of Contact Addresses.
*/
- void doubleRegistrationExpire(void) {
+ void doubleRegistrationExpire() {
registrationExpire_ *= 2;
if (registrationExpire_ < 0)
@@ -204,7 +204,7 @@ class SIPAccount : public Account {
* @param void
* @return pjsip_regc* A pointer to the registration structure
*/
- pjsip_regc* getRegistrationInfo(void) const {
+ pjsip_regc* getRegistrationInfo() const {
return regc_;
}
@@ -223,7 +223,7 @@ class SIPAccount : public Account {
* file, that can be used directly by PJSIP to initialize
* TLS transport.
*/
- const pjsip_tls_setting * getTlsSetting(void) const {
+ const pjsip_tls_setting * getTlsSetting() const {
return &tlsSetting_;
}
@@ -232,14 +232,14 @@ class SIPAccount : public Account {
* file, that can be used directly by PJSIP to initialize
* an alternate UDP transport.
*/
- std::string getStunServer(void) const {
+ std::string getStunServer() const {
return stunServer_;
}
void setStunServer(const std::string &srv) {
stunServer_ = srv;
}
- pj_str_t getStunServerName(void) const {
+ pj_str_t getStunServerName() const {
return stunServerName_;
}
@@ -248,7 +248,7 @@ class SIPAccount : public Account {
* file, that can be used directly by PJSIP to initialize
* an alternate UDP transport.
*/
- pj_uint16_t getStunPort(void) const {
+ pj_uint16_t getStunPort() const {
return stunPort_;
}
void setStunPort(pj_uint16_t port) {
@@ -259,7 +259,7 @@ class SIPAccount : public Account {
* @return bool Tells if current transport for that
* account is set to TLS.
*/
- bool isTlsEnabled(void) const {
+ bool isTlsEnabled() const {
return transportType_ == PJSIP_TRANSPORT_TLS;
}
@@ -267,7 +267,7 @@ class SIPAccount : public Account {
* @return bool Tells if current transport for that
* account is set to OTHER.
*/
- bool isStunEnabled(void) const {
+ bool isStunEnabled() const {
return stunEnabled_;
}
@@ -280,7 +280,7 @@ class SIPAccount : public Account {
* of the host on which the UA is running, since these are not logical
* names."
*/
- std::string getFromUri(void) const;
+ std::string getFromUri() const;
/*
* This method adds the correct scheme, hostname and append
@@ -300,7 +300,7 @@ class SIPAccount : public Account {
* @return pj_str_t "server" uri based on @param hostPort
* @param hostPort A string formatted as : "hostname:port"
*/
- std::string getServerUri(void) const;
+ std::string getServerUri() const;
/**
* @param port Optional port. Otherwise set to the port defined for that account.
@@ -312,7 +312,7 @@ class SIPAccount : public Account {
/**
* Get the local interface name on which this account is bound.
*/
- std::string getLocalInterface(void) const {
+ std::string getLocalInterface() const {
return interface_;
}
@@ -330,7 +330,7 @@ class SIPAccount : public Account {
* actually using.
* @return pj_uint16 The port used for that account
*/
- pj_uint16_t getLocalPort(void) const {
+ pj_uint16_t getLocalPort() const {
return (pj_uint16_t) localPort_;
}
@@ -347,7 +347,7 @@ class SIPAccount : public Account {
* for the chosen SIP transport.
* @return pj_uint16 The port used for that account
*/
- pj_uint16_t getPublishedPort(void) const {
+ pj_uint16_t getPublishedPort() const {
return (pj_uint16_t) publishedPort_;
}
@@ -364,7 +364,7 @@ class SIPAccount : public Account {
* Get the local port for TLS listener.
* @return pj_uint16 The port used for that account
*/
- pj_uint16_t getTlsListenerPort(void) const {
+ pj_uint16_t getTlsListenerPort() const {
return tlsListenerPort_;
}
@@ -374,7 +374,7 @@ class SIPAccount : public Account {
* will be used.
* @return std::string The public IPV4 address formatted in the standard dot notation.
*/
- std::string getPublishedAddress(void) const {
+ std::string getPublishedAddress() const {
return publishedIpAddress_;
}
@@ -387,27 +387,27 @@ class SIPAccount : public Account {
publishedIpAddress_ = publishedIpAddress;
}
- std::string getServiceRoute(void) const {
+ std::string getServiceRoute() const {
return serviceRoute_;
}
- DtmfType getDtmfType(void) const {
+ DtmfType getDtmfType() const {
return dtmfType_;
}
- bool getSrtpEnabled(void) const {
+ bool getSrtpEnabled() const {
return srtpEnabled_;
}
- std::string getSrtpKeyExchange(void) const {
+ std::string getSrtpKeyExchange() const {
return srtpKeyExchange_;
}
- bool getSrtpFallback(void) const {
+ bool getSrtpFallback() const {
return srtpFallback_;
}
- bool getZrtpHelloHash(void) const {
+ bool getZrtpHelloHash() const {
return zrtpHelloHash_;
}
@@ -427,12 +427,12 @@ class SIPAccount : public Account {
* Initializes tls settings from configuration file.
*
*/
- void initTlsConfiguration(void);
+ void initTlsConfiguration();
/*
* Initializes STUN config from the config file
*/
- void initStunConfiguration(void);
+ void initStunConfiguration();
/**
* If username is not provided, as it happens for Direct ip calls,
@@ -440,7 +440,7 @@ class SIPAccount : public Account {
* running this program.
* @return std::string The login name under which SFLPhone is running.
*/
- static std::string getLoginName(void);
+ static std::string getLoginName();
// The pjsip client registration information
pjsip_regc *regc_;
diff --git a/daemon/src/sip/sipcall.cpp b/daemon/src/sip/sipcall.cpp
index 6c2f87cf24b0e5b0aaedaec9fb3c528d4638dbcc..dd31f3bdc387e2db9490d1602a5fc0defc73a56e 100644
--- a/daemon/src/sip/sipcall.cpp
+++ b/daemon/src/sip/sipcall.cpp
@@ -35,10 +35,15 @@
#include "audio/audiortp/audio_rtp_factory.h"
#include "sdp.h"
+namespace {
+ static const int INITIAL_SIZE = 16384;
+ static const int INCREMENT_SIZE = INITIAL_SIZE;
+}
+
SIPCall::SIPCall(const std::string& id, Call::CallType type, pj_caching_pool *caching_pool) : Call(id, type)
, inv(NULL)
- , _audiortp(this)
- , pool_(pj_pool_create(&caching_pool->factory, id.c_str(), 16384, 16384, NULL))
+ , audiortp_(this)
+ , pool_(pj_pool_create(&caching_pool->factory, id.c_str(), INITIAL_SIZE, INCREMENT_SIZE, NULL))
, local_sdp_(new Sdp(pool_))
{
}
diff --git a/daemon/src/sip/sipcall.h b/daemon/src/sip/sipcall.h
index 5f20bbf478643f2579788745a182000104323a7a..7213957f02ab2dd3d24c09f054ed171d2611844a 100644
--- a/daemon/src/sip/sipcall.h
+++ b/daemon/src/sip/sipcall.h
@@ -64,21 +64,21 @@ class SIPCall : public Call {
/**
* Return the local SDP session
*/
- Sdp* getLocalSDP(void) {
+ Sdp* getLocalSDP() {
return local_sdp_;
}
/**
* Returns a pointer to the AudioRtp object
*/
- sfl::AudioRtpFactory * getAudioRtp(void) {
- return &_audiortp;
+ sfl::AudioRtpFactory & getAudioRtp() {
+ return audiortp_;
}
/**
* Return the local memory pool for this call
*/
- pj_pool_t *getMemoryPool(void) {
+ pj_pool_t *getMemoryPool() {
return pool_;
}
@@ -98,7 +98,7 @@ class SIPCall : public Call {
/**
* Audio Rtp Session factory
*/
- sfl::AudioRtpFactory _audiortp;
+ sfl::AudioRtpFactory audiortp_;
/**
* The pool to allocate memory, released once call hang up
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index a37bc8fcb894264c503d8e95dbf5a15b895d4100..1cc6855cb34b70fe17ddf28b52e89cf3fabdc5f0 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -76,7 +76,7 @@ using namespace sfl;
namespace {
-static pjsip_transport *_localUDPTransport = NULL; /** The default transport (5060) */
+static pjsip_transport *localUDPTransport_ = NULL; /** The default transport (5060) */
/** A map to retreive SFLphone internal call id
* Given a SIP call ID (usefull for transaction sucha as transfer)*/
@@ -96,10 +96,10 @@ void setCallMediaLocal(SIPCall* call, const std::string &localIP);
*/
std::string fetchHeaderValue(pjsip_msg *msg, std::string field);
-static pj_caching_pool pool_cache, *_cp = &pool_cache;
-static pj_pool_t *_pool;
-static pjsip_endpoint *_endpt;
-static pjsip_module _mod_ua;
+static pj_caching_pool pool_cache, *cp_ = &pool_cache;
+static pj_pool_t *pool_;
+static pjsip_endpoint *endpt_;
+static pjsip_module mod_ua_;
static pj_thread_t *thread;
static void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status UNUSED);
@@ -108,7 +108,7 @@ static void sdp_create_offer_cb(pjsip_inv_session *inv, pjmedia_sdp_session **p_
static void invite_session_state_changed_cb(pjsip_inv_session *inv, pjsip_event *e);
static void outgoing_request_forked_cb(pjsip_inv_session *inv, pjsip_event *e);
static void transaction_state_changed_cb(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
-static void registration_cb(struct pjsip_regc_cbparam *param);
+static void registration_cb(pjsip_regc_cbparam *param);
static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata);
static pj_bool_t transaction_response_cb(pjsip_rx_data *rdata UNUSED) ;
@@ -180,32 +180,32 @@ SIPVoIPLink::SIPVoIPLink() : evThread_(new EventThread(this))
pj_log_set_level(6); // From 0 (min) to 6 (max)
TRY(pjnath_init());
- pj_caching_pool_init(_cp, &pj_pool_factory_default_policy, 0);
- _pool = pj_pool_create(&_cp->factory, "sflphone", 4000, 4000, NULL);
+ pj_caching_pool_init(cp_, &pj_pool_factory_default_policy, 0);
+ pool_ = pj_pool_create(&cp_->factory, "sflphone", 4000, 4000, NULL);
- if (!_pool)
+ if (!pool_)
throw VoipLinkException("UserAgent: Could not initialize memory pool");
- TRY(pjsip_endpt_create(&_cp->factory, pj_gethostname()->ptr, &_endpt));
+ TRY(pjsip_endpt_create(&cp_->factory, pj_gethostname()->ptr, &endpt_));
if (loadSIPLocalIP().empty())
throw VoipLinkException("UserAgent: Unable to determine network capabilities");
- TRY(pjsip_tsx_layer_init_module(_endpt));
- TRY(pjsip_ua_init_module(_endpt, NULL));
- TRY(pjsip_replaces_init_module(_endpt)); // See the Replaces specification in RFC 3891
- TRY(pjsip_100rel_init_module(_endpt));
+ TRY(pjsip_tsx_layer_init_module(endpt_));
+ TRY(pjsip_ua_init_module(endpt_, NULL));
+ TRY(pjsip_replaces_init_module(endpt_)); // See the Replaces specification in RFC 3891
+ TRY(pjsip_100rel_init_module(endpt_));
// Initialize and register sflphone module
- _mod_ua.name = pj_str((char*) PACKAGE);
- _mod_ua.id = -1;
- _mod_ua.priority = PJSIP_MOD_PRIORITY_APPLICATION;
- _mod_ua.on_rx_request = &transaction_request_cb;
- _mod_ua.on_rx_response = &transaction_response_cb;
- TRY(pjsip_endpt_register_module(_endpt, &_mod_ua));
+ mod_ua_.name = pj_str((char*) PACKAGE);
+ mod_ua_.id = -1;
+ mod_ua_.priority = PJSIP_MOD_PRIORITY_APPLICATION;
+ mod_ua_.on_rx_request = &transaction_request_cb;
+ mod_ua_.on_rx_response = &transaction_response_cb;
+ TRY(pjsip_endpt_register_module(endpt_, &mod_ua_));
- TRY(pjsip_evsub_init_module(_endpt));
- TRY(pjsip_xfer_init_module(_endpt));
+ TRY(pjsip_evsub_init_module(endpt_));
+ TRY(pjsip_xfer_init_module(endpt_));
static const pjsip_inv_callback inv_cb = {
invite_session_state_changed_cb,
@@ -217,20 +217,20 @@ SIPVoIPLink::SIPVoIPLink() : evThread_(new EventThread(this))
NULL,
NULL,
};
- TRY(pjsip_inv_usage_init(_endpt, &inv_cb));
+ TRY(pjsip_inv_usage_init(endpt_, &inv_cb));
static const pj_str_t allowed[] = { { (char*) "INFO", 4}, { (char*) "REGISTER", 8}, { (char*) "OPTIONS", 7}, { (char*) "MESSAGE", 7 } }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}
- pjsip_endpt_add_capability(_endpt, &_mod_ua, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
+ pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
static const pj_str_t text_plain = { (char*) "text/plain", 10 };
- pjsip_endpt_add_capability(_endpt, &_mod_ua, PJSIP_H_ACCEPT, NULL, 1, &text_plain);
+ pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, NULL, 1, &text_plain);
static const pj_str_t accepted = { (char*) "application/sdp", 15 };
- pjsip_endpt_add_capability(_endpt, &_mod_ua, PJSIP_H_ACCEPT, NULL, 1, &accepted);
+ pjsip_endpt_add_capability(endpt_, &mod_ua_, PJSIP_H_ACCEPT, NULL, 1, &accepted);
_debug("UserAgent: pjsip version %s for %s initialized", pj_get_version(), PJ_OS_NAME);
- TRY(pjsip_replaces_init_module(_endpt));
+ TRY(pjsip_replaces_init_module(endpt_));
evThread_->start();
}
@@ -242,11 +242,11 @@ SIPVoIPLink::~SIPVoIPLink()
pj_thread_destroy(thread);
const pj_time_val tv = {0, 10};
- pjsip_endpt_handle_events(_endpt, &tv);
- pjsip_endpt_destroy(_endpt);
+ pjsip_endpt_handle_events(endpt_, &tv);
+ pjsip_endpt_destroy(endpt_);
- pj_pool_release(_pool);
- pj_caching_pool_destroy(_cp);
+ pj_pool_release(pool_);
+ pj_caching_pool_destroy(cp_);
pj_shutdown();
}
@@ -275,7 +275,7 @@ SIPVoIPLink::getEvent()
pj_thread_register(NULL, desc, &thread);
static const pj_time_val timeout = {0, 10};
- pjsip_endpt_handle_events(_endpt, &timeout);
+ pjsip_endpt_handle_events(endpt_, &timeout);
}
void SIPVoIPLink::sendRegister(Account *a)
@@ -288,7 +288,7 @@ void SIPVoIPLink::sendRegister(Account *a)
pjsip_regc *regc = account->getRegistrationInfo();
- if (pjsip_regc_create(_endpt, (void *) account, ®istration_cb, ®c) != PJ_SUCCESS)
+ if (pjsip_regc_create(endpt_, (void *) account, ®istration_cb, ®c) != PJ_SUCCESS)
throw VoipLinkException("UserAgent: Unable to create regc structure.");
std::string srvUri(account->getServerUri());
@@ -306,7 +306,7 @@ void SIPVoIPLink::sendRegister(Account *a)
throw VoipLinkException("Unable to initialize account registration structure");
if (!account->getServiceRoute().empty())
- pjsip_regc_set_route_set(regc, createRouteSet(account->getServiceRoute(), _pool));
+ pjsip_regc_set_route_set(regc, createRouteSet(account->getServiceRoute(), pool_));
pjsip_regc_set_credentials(regc, account->getCredentialCount(), account->getCredInfo());
@@ -317,7 +317,7 @@ void SIPVoIPLink::sendRegister(Account *a)
pj_str_t pJuseragent = pj_str((char*)useragent.c_str());
const pj_str_t STR_USER_AGENT = { (char*) "User-Agent", 10 };
- pjsip_generic_string_hdr *h = pjsip_generic_string_hdr_create(_pool, &STR_USER_AGENT, &pJuseragent);
+ pjsip_generic_string_hdr *h = pjsip_generic_string_hdr_create(pool_, &STR_USER_AGENT, &pJuseragent);
pj_list_push_back(&hdr_list, (pjsip_hdr*) h);
pjsip_regc_add_headers(regc, &hdr_list);
@@ -327,7 +327,7 @@ void SIPVoIPLink::sendRegister(Account *a)
if (pjsip_regc_register(regc, PJ_TRUE, &tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to initialize transaction data for account registration");
- if (pjsip_regc_set_transport(regc, initTransportSelector(account->transport_, _pool)) != PJ_SUCCESS)
+ if (pjsip_regc_set_transport(regc, initTransportSelector(account->transport_, pool_)) != PJ_SUCCESS)
throw VoipLinkException("Unable to set transport");
// decrease transport's ref count, counter incrementation is managed when acquiring transport
@@ -378,7 +378,7 @@ Call *SIPVoIPLink::newOutgoingCall(const std::string& id, const std::string& toU
if (account == NULL) // TODO: We should investigate how we could get rid of this error and create a IP2IP call instead
throw VoipLinkException("Could not get account for this call");
- SIPCall* call = new SIPCall(id, Call::Outgoing, _cp);
+ SIPCall* call = new SIPCall(id, Call::Outgoing, cp_);
// If toUri is not a well formated sip URI, use account information to process it
std::string toUri;
@@ -416,10 +416,10 @@ Call *SIPVoIPLink::newOutgoingCall(const std::string& id, const std::string& toU
}
try {
- call->getAudioRtp()->initAudioRtpConfig();
- call->getAudioRtp()->initAudioSymmetricRtpSession();
- call->getAudioRtp()->initLocalCryptoInfo();
- call->getAudioRtp()->start(static_cast<sfl::AudioCodec *>(audiocodec));
+ call->getAudioRtp().initAudioRtpConfig();
+ call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().initLocalCryptoInfo();
+ call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
} catch (...) {
delete call;
throw VoipLinkException("Could not start rtp session for early media");
@@ -490,10 +490,10 @@ SIPVoIPLink::hangup(const std::string& id)
return;
// Make sure user data is NULL in callbacks
- inv->mod_data[_mod_ua.id] = NULL;
+ inv->mod_data[mod_ua_.id] = NULL;
if (Manager::instance().isCurrentCall(id))
- call->getAudioRtp()->stop();
+ call->getAudioRtp().stop();
removeCall(id);
}
@@ -513,10 +513,10 @@ SIPVoIPLink::peerHungup(const std::string& id)
return;
// Make sure user data is NULL in callbacks
- call->inv->mod_data[_mod_ua.id ] = NULL;
+ call->inv->mod_data[mod_ua_.id ] = NULL;
if (Manager::instance().isCurrentCall(id))
- call->getAudioRtp()->stop();
+ call->getAudioRtp().stop();
removeCall(id);
}
@@ -526,7 +526,7 @@ SIPVoIPLink::onhold(const std::string& id)
{
SIPCall *call = getSIPCall(id);
call->setState(Call::Hold);
- call->getAudioRtp()->stop();
+ call->getAudioRtp().stop();
Sdp *sdpSession = call->getLocalSDP();
@@ -563,10 +563,9 @@ SIPVoIPLink::offhold(const std::string& id)
if (audiocodec == NULL)
throw VoipLinkException("Could not instantiate codec");
- call->getAudioRtp()->initAudioRtpConfig();
- call->getAudioRtp()->initAudioSymmetricRtpSession();
- call->getAudioRtp()->start(static_cast<sfl::AudioCodec *>(audiocodec));
-
+ call->getAudioRtp().initAudioRtpConfig();
+ call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
} catch (const SdpException &e) {
_error("UserAgent: Exception: %s", e.what());
} catch (...) {
@@ -619,7 +618,7 @@ SIPVoIPLink::transferCommon(SIPCall *call, pj_str_t *dst)
* because after this function, we can no find the cooresponding
* voiplink from the call any more. But the voiplink is useful!
*/
- pjsip_evsub_set_mod_data(sub, _mod_ua.id, this);
+ pjsip_evsub_set_mod_data(sub, mod_ua_.id, this);
/*
* Create REFER request.
@@ -697,7 +696,7 @@ SIPVoIPLink::refuse(const std::string& id)
if (!call->isIncoming() or call->getConnectionState() == Call::Connected)
return;
- call->getAudioRtp()->stop();
+ call->getAudioRtp().stop();
pjsip_tx_data *tdata;
@@ -708,7 +707,7 @@ SIPVoIPLink::refuse(const std::string& id)
return;
// Make sure the pointer is NULL in callbacks
- call->inv->mod_data[_mod_ua.id] = NULL;
+ call->inv->mod_data[mod_ua_.id] = NULL;
removeCall(id);
}
@@ -736,7 +735,7 @@ void
SIPVoIPLink::dtmfSend(SIPCall *call, char code, DtmfType dtmf)
{
if (dtmf == OVERRTP) {
- call->getAudioRtp()->sendDtmfDigit(code - '0');
+ call->getAudioRtp().sendDtmfDigit(code - '0');
return;
}
@@ -765,7 +764,7 @@ SIPVoIPLink::dtmfSend(SIPCall *call, char code, DtmfType dtmf)
if (tdata->msg->body == NULL)
pjsip_tx_data_dec_ref(tdata);
else
- pjsip_dlg_send_request(call->inv->dlg, tdata, _mod_ua.id, NULL);
+ pjsip_dlg_send_request(call->inv->dlg, tdata, mod_ua_.id, NULL);
}
bool
@@ -804,7 +803,7 @@ SIPVoIPLink::SIPStartCall(SIPCall *call)
pjsip_auth_clt_set_credentials(&dialog->auth_sess, account->getCredentialCount(), account->getCredInfo());
- call->inv->mod_data[_mod_ua.id] = call;
+ call->inv->mod_data[mod_ua_.id] = call;
pjsip_tx_data *tdata;
@@ -840,7 +839,7 @@ SIPVoIPLink::SIPCallClosed(SIPCall *call)
std::string id(call->getCallId());
if (Manager::instance().isCurrentCall(id))
- call->getAudioRtp()->stop();
+ call->getAudioRtp().stop();
Manager::instance().peerHungupCall(id);
removeCall(id);
@@ -875,7 +874,7 @@ bool SIPVoIPLink::SIPNewIpToIpCall(const std::string& id, const std::string& to)
if (!account)
return false;
- SIPCall *call = new SIPCall(id, Call::Outgoing, _cp);
+ SIPCall *call = new SIPCall(id, Call::Outgoing, cp_);
call->setCallConfiguration(Call::IPtoIP);
call->initRecFileName(to);
@@ -893,10 +892,10 @@ bool SIPVoIPLink::SIPNewIpToIpCall(const std::string& id, const std::string& to)
// Audio Rtp Session must be initialized before creating initial offer in SDP session
// since SDES require crypto attribute.
- call->getAudioRtp()->initAudioRtpConfig();
- call->getAudioRtp()->initAudioSymmetricRtpSession();
- call->getAudioRtp()->initLocalCryptoInfo();
- call->getAudioRtp()->start(static_cast<sfl::AudioCodec *>(audiocodec));
+ call->getAudioRtp().initAudioRtpConfig();
+ call->getAudioRtp().initAudioSymmetricRtpSession();
+ call->getAudioRtp().initLocalCryptoInfo();
+ call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
// Building the local SDP offer
call->getLocalSDP()->setLocalIP(localAddress);
@@ -950,7 +949,7 @@ static pj_bool_t stun_sock_on_rx_data_cb(pj_stun_sock *stun_sock UNUSED, void *p
pj_status_t SIPVoIPLink::stunServerResolve(SIPAccount *account)
{
pj_stun_config stunCfg;
- pj_stun_config_init(&stunCfg, &_cp->factory, 0, pjsip_endpt_get_ioqueue(_endpt), pjsip_endpt_get_timer_heap(_endpt));
+ pj_stun_config_init(&stunCfg, &cp_->factory, 0, pjsip_endpt_get_ioqueue(endpt_), pjsip_endpt_get_timer_heap(endpt_));
static const pj_stun_sock_cb stun_sock_cb = {
stun_sock_on_rx_data_cb,
@@ -989,7 +988,7 @@ void SIPVoIPLink::createDefaultSipUdpTransport()
createUdpTransport(account);
assert(account->transport_);
- _localUDPTransport = account->transport_;
+ localUDPTransport_ = account->transport_;
}
void SIPVoIPLink::createTlsListener(SIPAccount *account, pjsip_tpfactory **listener)
@@ -1007,7 +1006,7 @@ void SIPVoIPLink::createTlsListener(SIPAccount *account, pjsip_tpfactory **liste
local_addr.sin_port
};
- pjsip_tls_transport_start(_endpt, account->getTlsSetting(), &local_addr, &a_name, 1, listener);
+ pjsip_tls_transport_start(endpt_, account->getTlsSetting(), &local_addr, &a_name, 1, listener);
}
@@ -1024,7 +1023,8 @@ void SIPVoIPLink::createTlsTransport(SIPAccount *account, std::string remoteAddr
if (localTlsListener == NULL)
createTlsListener(account, &localTlsListener);
- pjsip_endpt_acquire_transport(_endpt, PJSIP_TRANSPORT_TLS, &rem_addr, sizeof(rem_addr), NULL, &account->transport_);
+ pjsip_endpt_acquire_transport(endpt_, PJSIP_TRANSPORT_TLS, &rem_addr,
+ sizeof rem_addr, NULL, &account->transport_);
}
@@ -1051,8 +1051,8 @@ void SIPVoIPLink::createSipTransport(SIPAccount *account)
if (account->transport_)
pjsip_transport_add_ref(account->transport_);
else {
- account->transport_ = _localUDPTransport;
- account->setLocalPort(_localUDPTransport->local_name.port);
+ account->transport_ = localUDPTransport_;
+ account->setLocalPort(localUDPTransport_->local_name.port);
}
}
}
@@ -1092,8 +1092,8 @@ void SIPVoIPLink::createUdpTransport(SIPAccount *account)
listeningPort
};
- pjsip_udp_transport_start(_endpt, &bound_addr, &a_name, 1, &account->transport_);
- pjsip_tpmgr_dump_transports(pjsip_endpt_get_tpmgr(_endpt)); // dump debug information to stdout
+ pjsip_udp_transport_start(endpt_, &bound_addr, &a_name, 1, &account->transport_);
+ pjsip_tpmgr_dump_transports(pjsip_endpt_get_tpmgr(endpt_)); // dump debug information to stdout
if (account->transport_)
transportMap_[account->getLocalPort()] = account->transport_;
@@ -1137,7 +1137,7 @@ void SIPVoIPLink::createStunTransport(SIPAccount *account)
// Query the mapped IP address and port on the 'outside' of the NAT
pj_sockaddr_in pub_addr;
- if (pjstun_get_mapped_addr(&_cp->factory, 1, &sock, &stunServer, stunPort, &stunServer, stunPort, &pub_addr) != PJ_SUCCESS) {
+ if (pjstun_get_mapped_addr(&cp_->factory, 1, &sock, &stunServer, stunPort, &stunServer, stunPort, &pub_addr) != PJ_SUCCESS) {
_error("Can't contact STUN server");
pj_sock_close(sock);
return;
@@ -1153,9 +1153,10 @@ void SIPVoIPLink::createStunTransport(SIPAccount *account)
account->setPublishedAddress(listeningAddress);
account->setPublishedPort(a_name.port);
- pjsip_udp_transport_attach2(_endpt, PJSIP_TRANSPORT_UDP, sock, &a_name, 1, &account->transport_);
+ pjsip_udp_transport_attach2(endpt_, PJSIP_TRANSPORT_UDP, sock, &a_name, 1,
+ &account->transport_);
- pjsip_tpmgr_dump_transports(pjsip_endpt_get_tpmgr(_endpt));
+ pjsip_tpmgr_dump_transports(pjsip_endpt_get_tpmgr(endpt_));
}
@@ -1174,7 +1175,7 @@ void SIPVoIPLink::findLocalAddressFromUri(const std::string& uri, pjsip_transpor
ss << DEFAULT_SIP_PORT;
port = ss.str();
- pjsip_uri *genericUri = pjsip_parse_uri(_pool, (char*)uri.data(), uri.size(), 0);
+ pjsip_uri *genericUri = pjsip_parse_uri(pool_, (char*)uri.data(), uri.size(), 0);
const pj_str_t *pjMachineName = pj_gethostname();
addr = std::string(pjMachineName->ptr, pjMachineName->slen);
@@ -1196,25 +1197,25 @@ void SIPVoIPLink::findLocalAddressFromUri(const std::string& uri, pjsip_transpor
port = ss.str();
} else {
if (transport == NULL)
- transport = _localUDPTransport;
+ transport = localUDPTransport_;
transportType = PJSIP_TRANSPORT_UDP;
}
- pjsip_tpmgr *tpmgr = pjsip_endpt_get_tpmgr(_endpt);
+ pjsip_tpmgr *tpmgr = pjsip_endpt_get_tpmgr(endpt_);
if (!tpmgr)
return;
pjsip_tpselector *tp_sel = NULL;
- if (transportType == PJSIP_TRANSPORT_UDP && transport)
- tp_sel = initTransportSelector(transport, _pool);
+ if (transportType == PJSIP_TRANSPORT_UDP and transport)
+ tp_sel = initTransportSelector(transport, pool_);
pj_str_t localAddress;
int i_port;
- if (pjsip_tpmgr_find_local_addr(tpmgr, _pool, transportType, tp_sel, &localAddress, &i_port) != PJ_SUCCESS)
+ if (pjsip_tpmgr_find_local_addr(tpmgr, pool_, transportType, tp_sel, &localAddress, &i_port) != PJ_SUCCESS)
return;
addr = std::string(localAddress.ptr, localAddress.slen);
@@ -1276,7 +1277,7 @@ int SIPSessionReinvite(SIPCall *call)
void invite_session_state_changed_cb(pjsip_inv_session *inv, pjsip_event *e)
{
- SIPCall *call = reinterpret_cast<SIPCall*>(inv->mod_data[_mod_ua.id]);
+ SIPCall *call = reinterpret_cast<SIPCall*>(inv->mod_data[mod_ua_.id]);
if (call == NULL)
return;
@@ -1332,7 +1333,7 @@ void invite_session_state_changed_cb(pjsip_inv_session *inv, pjsip_event *e)
void sdp_request_offer_cb(pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
{
- SIPCall *call = (SIPCall*) inv->mod_data[_mod_ua.id ];
+ SIPCall *call = (SIPCall*) inv->mod_data[mod_ua_.id ];
if (!call)
return;
@@ -1348,7 +1349,7 @@ void sdp_request_offer_cb(pjsip_inv_session *inv, const pjmedia_sdp_session *off
static void sdp_create_offer_cb(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
{
- SIPCall *call = reinterpret_cast<SIPCall*>(inv->mod_data[_mod_ua.id]);
+ SIPCall *call = reinterpret_cast<SIPCall*>(inv->mod_data[mod_ua_.id]);
std::string accountid(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(accountid));
@@ -1376,7 +1377,7 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
const pjmedia_sdp_session *remote_sdp;
const pjmedia_sdp_session *local_sdp;
- SIPCall *call = reinterpret_cast<SIPCall *>(inv->mod_data[_mod_ua.id]);
+ SIPCall *call = reinterpret_cast<SIPCall *>(inv->mod_data[mod_ua_.id]);
if (call == NULL) {
_debug("UserAgent: Call declined by peer, SDP negotiation stopped");
@@ -1419,8 +1420,8 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
// Update internal field for
sdpSession->setMediaTransportInfoFromRemoteSdp();
- call->getAudioRtp()->updateDestinationIpAddress();
- call->getAudioRtp()->setDtmfPayloadType(sdpSession->getTelephoneEventType());
+ call->getAudioRtp().updateDestinationIpAddress();
+ call->getAudioRtp().setDtmfPayloadType(sdpSession->getTelephoneEventType());
// Get the crypto attribute containing srtp's cryptographic context (keys, cipher)
CryptoOffer crypto_offer;
@@ -1441,7 +1442,7 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
nego_success = true;
try {
- call->getAudioRtp()->setRemoteCryptoInfo(sdesnego);
+ call->getAudioRtp().setRemoteCryptoInfo(sdesnego);
} catch (...) {}
Manager::instance().getDbusManager()->getCallManager()->secureSdesOn(call->getCallId());
@@ -1452,14 +1453,14 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
// We did not found any crypto context for this media, RTP fallback
- if (!nego_success && call->getAudioRtp()->isSdesEnabled()) {
- call->getAudioRtp()->stop();
- call->getAudioRtp()->setSrtpEnabled(false);
+ if (!nego_success && call->getAudioRtp().isSdesEnabled()) {
+ call->getAudioRtp().stop();
+ call->getAudioRtp().setSrtpEnabled(false);
std::string accountID = Manager::instance().getAccountFromCall(call->getCallId());
if (((SIPAccount *) Manager::instance().getAccount(accountID))->getSrtpFallback())
- call->getAudioRtp()->initAudioSymmetricRtpSession();
+ call->getAudioRtp().initAudioSymmetricRtpSession();
}
if (!sdpSession)
@@ -1477,9 +1478,9 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
int pl = sessionMedia->getPayloadType();
- if (pl != call->getAudioRtp()->getSessionMedia()) {
+ if (pl != call->getAudioRtp().getSessionMedia()) {
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec(pl);
- call->getAudioRtp()->updateSessionMedia(static_cast<sfl::AudioCodec *>(audiocodec));
+ call->getAudioRtp().updateSessionMedia(static_cast<sfl::AudioCodec *>(audiocodec));
}
} catch (const SdpException &e) {
_error("UserAgent: Exception: %s", e.what());
@@ -1533,7 +1534,7 @@ void transaction_state_changed_cb(pjsip_inv_session *inv UNUSED, pjsip_transacti
std::string formatedMessage = (char*) r_data->msg_info.msg->body->data;
// Try to determine who is the recipient of the message
- SIPCall *call = reinterpret_cast<SIPCall *>(inv->mod_data[_mod_ua.id]);
+ SIPCall *call = reinterpret_cast<SIPCall *>(inv->mod_data[mod_ua_.id]);
if (!call)
return;
@@ -1644,7 +1645,7 @@ static void handleIncomingOptions(pjsip_rx_data *rdata)
{
pjsip_tx_data *tdata;
- if (pjsip_endpt_create_response(_endpt, rdata, PJSIP_SC_OK, NULL, &tdata) != PJ_SUCCESS)
+ if (pjsip_endpt_create_response(endpt_, rdata, PJSIP_SC_OK, NULL, &tdata) != PJ_SUCCESS)
return;
#define ADD_HDR(hdr) do { \
@@ -1652,7 +1653,7 @@ static void handleIncomingOptions(pjsip_rx_data *rdata)
if (cap_hdr) \
pjsip_msg_add_hdr (tdata->msg, (pjsip_hdr*) pjsip_hdr_clone (tdata->pool, cap_hdr)); \
} while(0)
-#define ADD_CAP(cap) ADD_HDR(pjsip_endpt_get_capability(_endpt, cap, NULL));
+#define ADD_CAP(cap) ADD_HDR(pjsip_endpt_get_capability(endpt_, cap, NULL));
ADD_CAP(PJSIP_H_ALLOW);
ADD_CAP(PJSIP_H_ACCEPT);
@@ -1662,7 +1663,7 @@ static void handleIncomingOptions(pjsip_rx_data *rdata)
pjsip_response_addr res_addr;
pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
- if (pjsip_endpt_send_response(_endpt, &res_addr, tdata, NULL, NULL) != PJ_SUCCESS)
+ if (pjsip_endpt_send_response(endpt_, &res_addr, tdata, NULL, NULL) != PJ_SUCCESS)
pjsip_tx_data_dec_ref(tdata);
}
@@ -1693,14 +1694,14 @@ static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
Manager::instance().startVoiceMessageNotification(account_id, voicemail);
}
- pjsip_endpt_respond_stateless(_endpt, rdata, PJSIP_SC_OK, NULL, NULL, NULL);
+ pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_OK, NULL, NULL, NULL);
return true;
} else if (method->id == PJSIP_OPTIONS_METHOD) {
handleIncomingOptions(rdata);
return true;
} else if (method->id != PJSIP_INVITE_METHOD && method->id != PJSIP_ACK_METHOD) {
- pjsip_endpt_respond_stateless(_endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
+ pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return true;
}
@@ -1713,15 +1714,17 @@ static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
r_sdp = NULL;
if (account->getActiveCodecs().empty()) {
- pjsip_endpt_respond_stateless(_endpt, rdata, PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, NULL, NULL);
+ pjsip_endpt_respond_stateless(endpt_, rdata,
+ PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, NULL,
+ NULL);
return false;
}
// Verify that we can handle the request
unsigned options = 0;
- if (pjsip_inv_verify_request(rdata, &options, NULL, NULL, _endpt, NULL) != PJ_SUCCESS) {
- pjsip_endpt_respond_stateless(_endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
+ if (pjsip_inv_verify_request(rdata, &options, NULL, NULL, endpt_, NULL) != PJ_SUCCESS) {
+ pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return true;
}
@@ -1730,7 +1733,7 @@ static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
UrlHook::runAction(Manager::instance().hookPreference.getUrlCommand(), header_value);
}
- SIPCall* call = new SIPCall(Manager::instance().getNewCallID(), Call::Incoming, _cp);
+ SIPCall* call = new SIPCall(Manager::instance().getNewCallID(), Call::Incoming, cp_);
Manager::instance().associateCallToAccount(call->getCallId(), account_id);
// May use the published address as well
@@ -1761,8 +1764,8 @@ static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
call->getLocalSDP()->setLocalIP(addrSdp);
- call->getAudioRtp()->initAudioRtpConfig();
- call->getAudioRtp()->initAudioSymmetricRtpSession();
+ call->getAudioRtp().initAudioRtpConfig();
+ call->getAudioRtp().initAudioSymmetricRtpSession();
if (rdata->msg_info.msg->body) {
char sdpbuffer[1000];
@@ -1787,8 +1790,8 @@ static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
sfl::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
if (sdesnego.negotiate()) {
- call->getAudioRtp()->setRemoteCryptoInfo(sdesnego);
- call->getAudioRtp()->initLocalCryptoInfo();
+ call->getAudioRtp().setRemoteCryptoInfo(sdesnego);
+ call->getAudioRtp().initLocalCryptoInfo();
}
}
}
@@ -1796,22 +1799,21 @@ static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
call->getLocalSDP()->receiveOffer(r_sdp, account->getActiveCodecs());
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec(PAYLOAD_CODEC_ULAW);
- call->getAudioRtp()->start(static_cast<sfl::AudioCodec *>(audiocodec));
+ call->getAudioRtp().start(static_cast<sfl::AudioCodec *>(audiocodec));
pjsip_dialog* dialog;
if (pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, NULL, &dialog) != PJ_SUCCESS) {
delete call;
- pjsip_endpt_respond_stateless(_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
+ pjsip_endpt_respond_stateless(endpt_, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
return false;
-
}
pjsip_inv_create_uas(dialog, rdata, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv);
PJ_ASSERT_RETURN(pjsip_dlg_set_transport(dialog, tp) == PJ_SUCCESS, 1);
- call->inv->mod_data[_mod_ua.id] = call;
+ call->inv->mod_data[mod_ua_.id] = call;
// Check whether Replaces header is present in the request and process accordingly.
pjsip_dialog *replaced_dlg;
@@ -1819,7 +1821,7 @@ static pj_bool_t transaction_request_cb(pjsip_rx_data *rdata)
if (pjsip_replaces_verify_request(rdata, &replaced_dlg, PJ_FALSE, &response) != PJ_SUCCESS) {
_error("Something wrong with Replaces request.");
- pjsip_endpt_respond_stateless(_endpt, rdata, 500 /* internal server error */, NULL, NULL, NULL);
+ pjsip_endpt_respond_stateless(endpt_, rdata, 500 /* internal server error */, NULL, NULL, NULL);
}
// Check if call has been transfered
@@ -1859,7 +1861,7 @@ static pj_bool_t transaction_response_cb(pjsip_rx_data *rdata)
pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
- if (!tsx || tsx->method.id != PJSIP_INVITE_METHOD)
+ if (!tsx or tsx->method.id != PJSIP_INVITE_METHOD)
return PJ_SUCCESS;
if (tsx->status_code / 100 == 2) {
@@ -1877,7 +1879,7 @@ static pj_bool_t transaction_response_cb(pjsip_rx_data *rdata)
void onCallTransfered(pjsip_inv_session *inv, pjsip_rx_data *rdata)
{
- SIPCall *currentCall = reinterpret_cast<SIPCall *>(inv->mod_data[_mod_ua.id]);
+ SIPCall *currentCall = reinterpret_cast<SIPCall *>(inv->mod_data[mod_ua_.id]);
if (currentCall == NULL)
return;
@@ -1903,12 +1905,12 @@ void transfer_client_cb(pjsip_evsub *sub, pjsip_event *event)
break;
case PJSIP_EVSUB_STATE_TERMINATED:
- pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ pjsip_evsub_set_mod_data(sub, mod_ua_.id, NULL);
break;
case PJSIP_EVSUB_STATE_ACTIVE: {
- SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *>(pjsip_evsub_get_mod_data(sub, _mod_ua.id));
+ SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *>(pjsip_evsub_get_mod_data(sub, mod_ua_.id));
if (!link or !event)
return;
@@ -1950,7 +1952,7 @@ void transfer_client_cb(pjsip_evsub *sub, pjsip_event *event)
pjsip_inv_send_msg(call->inv, tdata);
Manager::instance().hangupCall(call->getCallId());
- pjsip_evsub_set_mod_data(sub, _mod_ua.id, NULL);
+ pjsip_evsub_set_mod_data(sub, mod_ua_.id, NULL);
}
break;
@@ -1997,7 +1999,7 @@ std::string fetchHeaderValue(pjsip_msg *msg, std::string field)
} // end anonymous namespace
-std::vector<std::string> SIPVoIPLink::getAllIpInterfaceByName(void)
+std::vector<std::string> SIPVoIPLink::getAllIpInterfaceByName()
{
static ifreq ifreqs[20];
ifconf ifconf;
@@ -2042,7 +2044,7 @@ std::string SIPVoIPLink::getInterfaceAddrFromName(const std::string &ifaceName)
return inet_ntoa(saddr_in->sin_addr);
}
-std::vector<std::string> SIPVoIPLink::getAllIpInterface(void)
+std::vector<std::string> SIPVoIPLink::getAllIpInterface()
{
pj_sockaddr addrList[16];
unsigned addrCnt = PJ_ARRAY_SIZE(addrList);
diff --git a/daemon/src/sip/sipvoiplink.h b/daemon/src/sip/sipvoiplink.h
index f11b182bc7d37ca219fb041335f22fe7d739b45d..cf5fdfb613c8eb1cd98e14209caf0f17165b2c62 100644
--- a/daemon/src/sip/sipvoiplink.h
+++ b/daemon/src/sip/sipvoiplink.h
@@ -32,8 +32,8 @@
* as that of the covered work.
*/
-#ifndef SIPVOIPLINK_H
-#define SIPVOIPLINK_H
+#ifndef SIPVOIPLINK_H_
+#define SIPVOIPLINK_H_
#include <map>
@@ -66,10 +66,6 @@ class SIPAccount;
class SIPVoIPLink : public VoIPLink {
public:
-
- /**
- * Destructor
- */
~SIPVoIPLink();
/**
@@ -81,17 +77,17 @@ class SIPVoIPLink : public VoIPLink {
/**
* Try to initiate the pjsip engine/thread and set config
*/
- virtual void init(void);
+ virtual void init();
/**
* Shut the library and clean up
*/
- virtual void terminate(void);
+ virtual void terminate();
/**
* Event listener. Each event send by the call manager is received and handled from here
*/
- virtual void getEvent(void);
+ virtual void getEvent();
/**
* Build and send SIP registration request
@@ -226,7 +222,7 @@ class SIPVoIPLink : public VoIPLink {
* of IPV4 address available on all of the interfaces on
* the system.
*/
- static std::vector<std::string> getAllIpInterface(void);
+ static std::vector<std::string> getAllIpInterface();
/**
* List all the interfaces on the system and return
@@ -236,7 +232,7 @@ class SIPVoIPLink : public VoIPLink {
* of interface name available on all of the interfaces on
* the system.
*/
- static std::vector<std::string> getAllIpInterfaceByName(void);
+ static std::vector<std::string> getAllIpInterfaceByName();
/**
* List all the interfaces on the system and return
@@ -353,5 +349,4 @@ class SIPVoIPLink : public VoIPLink {
friend class SIPTest;
};
-
-#endif
+#endif // SIPVOIPLINK_H_
diff --git a/daemon/test/historytest.cpp b/daemon/test/historytest.cpp
index 670bcc58b9747f7c6e1665dd7d903fec08869858..529237360ef6117d73c7b4f574a47b9724fc2f81 100644
--- a/daemon/test/historytest.cpp
+++ b/daemon/test/historytest.cpp
@@ -58,7 +58,7 @@ void HistoryTest::test_create_history_path()
result = history->create_history_path(path);
CPPUNIT_ASSERT(result == 0);
CPPUNIT_ASSERT(!history->is_loaded());
- CPPUNIT_ASSERT(history->_history_path == path);
+ CPPUNIT_ASSERT(history->history_path_ == path);
}
void HistoryTest::test_load_history_from_file()