diff --git a/daemon/src/audio/audiorecord.cpp b/daemon/src/audio/audiorecord.cpp
index dcfe03fdde815d10fbcf4c94c39ca249cf6460ab..24226265a5348748a53ea0f35e551915f6d4b10b 100644
--- a/daemon/src/audio/audiorecord.cpp
+++ b/daemon/src/audio/audiorecord.cpp
@@ -165,7 +165,7 @@ bool AudioRecord::isOpenFile()
bool AudioRecord::fileExists()
{
- INFO("AudioRecord: Trying to open %s ", fileName_);
+ DEBUG("AudioRecord: Trying to open %s ", fileName_);
return fopen(fileName_,"rb") != 0;
}
@@ -179,10 +179,10 @@ bool AudioRecord::setRecording()
{
if (isOpenFile()) {
if (!recordingEnabled_) {
- INFO("AudioRecording: Start recording");
+ DEBUG("AudioRecording: Start recording");
recordingEnabled_ = true;
} else {
- INFO("AudioRecording: Stop recording");
+ DEBUG("AudioRecording: Stop recording");
recordingEnabled_ = false;
}
} else {
@@ -197,7 +197,7 @@ bool AudioRecord::setRecording()
void AudioRecord::stopRecording()
{
- INFO("AudioRecording: Stop recording");
+ DEBUG("AudioRecording: Stop recording");
recordingEnabled_ = false;
}
@@ -251,7 +251,7 @@ void AudioRecord::createFilename()
// fileName_ = out.str();
strncpy(fileName_, out.str().c_str(), 8192);
- INFO("AudioRecord: create filename for this call %s ", fileName_);
+ DEBUG("AudioRecord: create filename for this call %s ", fileName_);
}
bool AudioRecord::setRawFile()
@@ -327,7 +327,7 @@ bool AudioRecord::openExistingRawFile()
bool AudioRecord::openExistingWavFile()
{
- INFO("%s(%s)\n", __PRETTY_FUNCTION__, fileName_);
+ DEBUG("%s(%s)\n", __PRETTY_FUNCTION__, fileName_);
fileHandle_ = fopen(fileName_, "rb+");
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.cpp b/daemon/src/audio/audiortp/audio_rtp_session.cpp
index 5c898441773cb0dff2e2a65434ad742c7f43b7d1..2a6bea42e7e2f404cbe47751b1a2c1e3fb857a22 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_session.cpp
@@ -196,8 +196,6 @@ void AudioRtpSession::setSessionTimeouts()
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());
@@ -210,7 +208,7 @@ void AudioRtpSession::setDestinationIpAddress()
// Store remote port in case we would need to forget current destination
remote_port_ = (unsigned short) ca_->getLocalSDP()->getRemoteAudioPort();
- INFO("AudioRtpSession: New remote address for session: %s:%d",
+ DEBUG("AudioRtpSession: New remote address for session: %s:%d",
ca_->getLocalSDP()->getRemoteIP().data(), remote_port_);
if (!queue_->addDestination(remote_ip_, remote_port_)) {
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
index 4107133d34d7530b12c57e63aaceebd28f4b7498..cc5d9ba37e068ad94c070a739cde6d6079b5116d 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
@@ -48,7 +48,7 @@ AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall * sipcall) :
, echoCanceller()
, rtpThread_(new AudioRtpThread(this))
{
- INFO("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
+ DEBUG("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
audioRtpRecord_.callId_ = ca_->getCallId();
}
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.cpp b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
index 81af2f5e1026d40928fba5273f4967fcb0def7ab..ebdb8f839e421b1494341b8a19bf846f9c6218c2 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
@@ -63,7 +63,7 @@ AudioZrtpSession::AudioZrtpSession(SIPCall * sipcall, const std::string& zidFile
setCancel(cancelDefault);
- INFO("AudioZrtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
+ DEBUG("AudioZrtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
}
AudioZrtpSession::~AudioZrtpSession()
@@ -77,7 +77,7 @@ void AudioZrtpSession::final()
// tmatth:Oct 25 2011:FIXME:
// This was crashing...seems like it's not necessary. Double check
// with valgrind/helgrind
-// delete this;
+// delete this;
}
void AudioZrtpSession::initializeZid()
diff --git a/daemon/src/audio/audiortp/zrtp_session_callback.cpp b/daemon/src/audio/audiortp/zrtp_session_callback.cpp
index 24e6a9bbf8e6953a98d55be2edeaa9950ebbc3f1..270468a80ea2a8fc8e683d515610d31dce7d0a17 100644
--- a/daemon/src/audio/audiortp/zrtp_session_callback.cpp
+++ b/daemon/src/audio/audiortp/zrtp_session_callback.cpp
@@ -55,7 +55,7 @@ ZrtpSessionCallback::ZrtpSessionCallback(SIPCall *sipcall) :
if (not infoMap_.empty())
return;
- INFO("Zrtp: Initialize callbacks");
+ DEBUG("Zrtp: Initialize callbacks");
// Information Map
infoMap_[InfoHelloReceived] = "Hello received, preparing a Commit";
diff --git a/daemon/src/audio/pulseaudio/audiostream.cpp b/daemon/src/audio/pulseaudio/audiostream.cpp
index c0ffe920db965b05725231dd6689e8a256767e50..3f29c187baccea7b2f011f70d2d11187b54f8267 100644
--- a/daemon/src/audio/pulseaudio/audiostream.cpp
+++ b/daemon/src/audio/pulseaudio/audiostream.cpp
@@ -100,15 +100,15 @@ AudioStream::stream_state_callback(pa_stream* s, void* user_data UNUSED)
switch (pa_stream_get_state(s)) {
case PA_STREAM_CREATING:
- INFO("Pulse: Stream is creating...");
+ DEBUG("Pulse: Stream is creating...");
break;
case PA_STREAM_TERMINATED:
- INFO("Pulse: Stream is terminating...");
+ DEBUG("Pulse: Stream is terminating...");
break;
case PA_STREAM_READY:
- INFO("Pulse: Stream successfully created, connected to %s", pa_stream_get_device_name(s));
+ DEBUG("Pulse: Stream successfully created, connected to %s", pa_stream_get_device_name(s));
DEBUG("Pulse: maxlength %u", pa_stream_get_buffer_attr(s)->maxlength);
DEBUG("Pulse: tlength %u", pa_stream_get_buffer_attr(s)->tlength);
DEBUG("Pulse: prebuf %u", pa_stream_get_buffer_attr(s)->prebuf);
@@ -118,7 +118,7 @@ AudioStream::stream_state_callback(pa_stream* s, void* user_data UNUSED)
break;
case PA_STREAM_UNCONNECTED:
- INFO("Pulse: Stream unconnected");
+ DEBUG("Pulse: Stream unconnected");
break;
case PA_STREAM_FAILED:
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index 5625855a3c5c9a59f18041190373bad5cabe7729..95874f23c3d03f9e9c4fdccc210b85d92f84f85c 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -340,7 +340,7 @@ bool ManagerImpl::answerCall(const std::string& call_id)
//THREAD=Main
void ManagerImpl::hangupCall(const std::string& callId)
{
- INFO("Manager: Hangup call %s", callId.c_str());
+ DEBUG("Manager: Hangup call %s", callId.c_str());
// store the current call id
std::string currentCallId(getCurrentCallId());
@@ -2776,7 +2776,7 @@ ManagerImpl::getAccount(const std::string& accountID)
std::string ManagerImpl::getAccountIdFromNameAndServer(const std::string& userName, const std::string& server) const
{
- INFO("Manager : username = %s, server = %s", userName.c_str(), server.c_str());
+ DEBUG("Manager : username = %s, server = %s", userName.c_str(), server.c_str());
// Try to find the account id from username and server name by full match
for (AccountMap::const_iterator iter = accountMap_.begin(); iter != accountMap_.end(); ++iter) {
diff --git a/daemon/src/preferences.cpp b/daemon/src/preferences.cpp
index 8844bf3256cc86fad25ddd0f7fec5db73125338a..98f36f2a2a5dd9e3476b49634af9fbf3958f941a 100644
--- a/daemon/src/preferences.cpp
+++ b/daemon/src/preferences.cpp
@@ -282,7 +282,7 @@ void checkSoundCard(int &card, int stream)
AudioLayer* AudioPreference::createAudioLayer()
{
- if (audioApi_ == PULSEAUDIO_API_STR and system("ps -C pulseaudio") == 0)
+ if (audioApi_ == PULSEAUDIO_API_STR and system("ps -C pulseaudio > /dev/null") == 0)
return new PulseLayer;
else {
audioApi_ = ALSA_API_STR;