Commit 3889f125 authored by Tristan Matthews's avatar Tristan Matthews
Browse files

* #7131: renamed logging macros

parent 765508ff
......@@ -113,7 +113,7 @@ bool AlsaLayer::openDevice(snd_pcm_t **pcm, const std::string &dev, snd_pcm_stre
}
if (err < 0) {
_error("Alsa: couldn't open device %s : %s", dev.c_str(),
ERROR("Alsa: couldn't open device %s : %s", dev.c_str(),
snd_strerror(err));
return false;
}
......@@ -218,7 +218,7 @@ AlsaLayer::stopStream()
#define ALSA_CALL(call, error) ({ \
int err_code = call; \
if (err_code < 0) \
_error("ALSA: "error": %s", snd_strerror(err_code)); \
ERROR("ALSA: "error": %s", snd_strerror(err_code)); \
err_code; \
})
......@@ -320,7 +320,7 @@ bool AlsaLayer::alsa_set_params(snd_pcm_t *pcm_handle)
TRY(snd_pcm_hw_params(HW), "hwparams");
#undef HW
_debug("ALSA: %s using sampling rate %dHz",
DEBUG("ALSA: %s using sampling rate %dHz",
(snd_pcm_stream(pcm_handle) == SND_PCM_STREAM_PLAYBACK) ? "playback" : "capture",
audioSampleRate_);
......@@ -369,7 +369,7 @@ AlsaLayer::write(void* buffer, int length, snd_pcm_t * handle)
}
default:
_error("ALSA: unknown write error, dropping frames: %s", snd_strerror(err));
ERROR("ALSA: unknown write error, dropping frames: %s", snd_strerror(err));
stopPlaybackStream();
break;
}
......@@ -404,12 +404,12 @@ AlsaLayer::read(void* buffer, int toCopy)
startCaptureStream();
}
_error("ALSA: XRUN capture ignored (%s)", snd_strerror(err));
ERROR("ALSA: XRUN capture ignored (%s)", snd_strerror(err));
break;
}
case EPERM:
_error("ALSA: Can't capture, EPERM (%s)", snd_strerror(err));
ERROR("ALSA: Can't capture, EPERM (%s)", snd_strerror(err));
prepareCaptureStream();
startCaptureStream();
break;
......@@ -459,9 +459,9 @@ AlsaLayer::getSoundCardsInfo(int stream)
snd_pcm_info_set_stream(pcminfo, (stream == SFL_PCM_CAPTURE) ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK);
if (snd_ctl_pcm_info(handle ,pcminfo) < 0)
_debug(" Cannot get info");
DEBUG(" Cannot get info");
else {
_debug("card %i : %s [%s]",
DEBUG("card %i : %s [%s]",
numCard,
snd_ctl_card_info_get_id(info),
snd_ctl_card_info_get_name(info));
......@@ -531,7 +531,7 @@ void AlsaLayer::capture()
int toGetSamples = snd_pcm_avail_update(captureHandle_);
if (toGetSamples < 0)
_error("Audio: Mic error: %s", snd_strerror(toGetSamples));
ERROR("Audio: Mic error: %s", snd_strerror(toGetSamples));
if (toGetSamples <= 0)
return;
......@@ -545,7 +545,7 @@ void AlsaLayer::capture()
SFLDataFormat* in = (SFLDataFormat*) malloc(toGetBytes);
if (read(in, toGetBytes) != toGetBytes) {
_error("ALSA MIC : Couldn't read!");
ERROR("ALSA MIC : Couldn't read!");
goto end;
}
......
......@@ -52,7 +52,7 @@ AudioLoop::getNext(SFLDataFormat* output, size_t total_samples, short volume)
size_t pos = pos_;
if (size_ == 0) {
_error("AudioLoop: Error: Audio loop size is 0");
ERROR("AudioLoop: Error: Audio loop size is 0");
return;
}
......
......@@ -100,12 +100,12 @@ void AudioRecord::initFileName(std::string peerNumber)
if (fileType_ == FILE_RAW) {
if (strstr(fileName_, ".raw") == NULL) {
_debug("AudioRecord: concatenate .raw file extension: name : %s", fileName_);
DEBUG("AudioRecord: concatenate .raw file extension: name : %s", fileName_);
fName.append(".raw");
}
} else if (fileType_ == FILE_WAV) {
if (strstr(fileName_, ".wav") == NULL) {
_debug("AudioRecord: concatenate .wav file extension: name : %s", fileName_);
DEBUG("AudioRecord: concatenate .wav file extension: name : %s", fileName_);
fName.append(".wav");
}
}
......@@ -122,10 +122,10 @@ bool AudioRecord::openFile()
{
bool result = false;
_debug("AudioRecord: Open file()");
DEBUG("AudioRecord: Open file()");
if (not fileExists()) {
_debug("AudioRecord: Filename does not exist, creating one");
DEBUG("AudioRecord: Filename does not exist, creating one");
byteCounter_ = 0;
if (fileType_ == FILE_RAW)
......@@ -133,7 +133,7 @@ bool AudioRecord::openFile()
else if (fileType_ == FILE_WAV)
result = setWavFile();
} else {
_debug("AudioRecord: Filename already exists, opening it");
DEBUG("AudioRecord: Filename already exists, opening it");
if (fileType_ == FILE_RAW)
result = openExistingRawFile();
else if (fileType_ == FILE_WAV)
......@@ -163,7 +163,7 @@ bool AudioRecord::isOpenFile()
bool AudioRecord::fileExists()
{
_info("AudioRecord: Trying to open %s ", fileName_);
INFO("AudioRecord: Trying to open %s ", fileName_);
return fopen(fileName_,"rb") != 0;
}
......@@ -177,10 +177,10 @@ bool AudioRecord::setRecording()
{
if (isOpenFile()) {
if (!recordingEnabled_) {
_info("AudioRecording: Start recording");
INFO("AudioRecording: Start recording");
recordingEnabled_ = true;
} else {
_info("AudioRecording: Stop recording");
INFO("AudioRecording: Stop recording");
recordingEnabled_ = false;
}
} else {
......@@ -195,7 +195,7 @@ bool AudioRecord::setRecording()
void AudioRecord::stopRecording()
{
_info("AudioRecording: Stop recording");
INFO("AudioRecording: Stop recording");
recordingEnabled_ = false;
}
......@@ -249,7 +249,7 @@ void AudioRecord::createFilename()
// fileName_ = out.str();
strncpy(fileName_, out.str().c_str(), 8192);
_info("AudioRecord: create filename for this call %s ", fileName_);
INFO("AudioRecord: create filename for this call %s ", fileName_);
}
bool AudioRecord::setRawFile()
......@@ -257,11 +257,11 @@ bool AudioRecord::setRawFile()
fileHandle_ = fopen(savePath_.c_str(), "wb");
if (!fileHandle_) {
_warn("AudioRecord: Could not create RAW file!");
WARN("AudioRecord: Could not create RAW file!");
return false;
}
_debug("AudioRecord:setRawFile() : created RAW file.");
DEBUG("AudioRecord:setRawFile() : created RAW file.");
return true;
}
......@@ -269,12 +269,12 @@ bool AudioRecord::setRawFile()
bool AudioRecord::setWavFile()
{
_debug("AudioRecord: Create new wave file %s, sampling rate: %d", savePath_.c_str(), sndSmplRate_);
DEBUG("AudioRecord: Create new wave file %s, sampling rate: %d", savePath_.c_str(), sndSmplRate_);
fileHandle_ = fopen(savePath_.c_str(), "wb");
if (!fileHandle_) {
_warn("AudioRecord: Error: could not create WAV file.");
WARN("AudioRecord: Error: could not create WAV file.");
return false;
}
......@@ -300,11 +300,11 @@ bool AudioRecord::setWavFile()
if (fwrite(&hdr, 4, 11, fileHandle_) != 11) {
_warn("AudioRecord: Error: could not write WAV header for file. ");
WARN("AudioRecord: Error: could not write WAV header for file. ");
return false;
}
_debug("AudioRecord: created WAV file successfully.");
DEBUG("AudioRecord: created WAV file successfully.");
return true;
}
......@@ -315,7 +315,7 @@ bool AudioRecord::openExistingRawFile()
fileHandle_ = fopen(fileName_, "ab+");
if (!fileHandle_) {
_warn("AudioRecord: could not create RAW file!");
WARN("AudioRecord: could not create RAW file!");
return false;
}
......@@ -325,37 +325,37 @@ bool AudioRecord::openExistingRawFile()
bool AudioRecord::openExistingWavFile()
{
_info("%s(%s)\n", __PRETTY_FUNCTION__, fileName_);
INFO("%s(%s)\n", __PRETTY_FUNCTION__, fileName_);
fileHandle_ = fopen(fileName_, "rb+");
if (!fileHandle_) {
_warn("AudioRecord: Error: could not open WAV file!");
WARN("AudioRecord: Error: could not open WAV file!");
return false;
}
if (fseek(fileHandle_, 40, SEEK_SET) != 0) // jump to data length
_warn("AudioRecord: Error: Couldn't seek offset 40 in the file ");
WARN("AudioRecord: Error: Couldn't seek offset 40 in the file ");
if (fread(&byteCounter_, 4, 1, fileHandle_))
_warn("AudioRecord: Error: bytecounter Read successfully ");
WARN("AudioRecord: Error: bytecounter Read successfully ");
if (fseek(fileHandle_, 0 , SEEK_END) != 0)
_warn("AudioRecord: Error: Couldn't seek at the en of the file ");
WARN("AudioRecord: Error: Couldn't seek at the en of the file ");
if (fclose(fileHandle_) != 0)
_warn("AudioRecord: Error: Can't close file r+ ");
WARN("AudioRecord: Error: Can't close file r+ ");
fileHandle_ = fopen(fileName_, "ab+");
if (!fileHandle_) {
_warn("AudioRecord: Error: Could not createopen WAV file ab+!");
WARN("AudioRecord: Error: Could not createopen WAV file ab+!");
return false;
}
if (fseek(fileHandle_, 4 , SEEK_END) != 0)
_warn("AudioRecord: Error: Couldn't seek at the en of the file ");
WARN("AudioRecord: Error: Couldn't seek at the en of the file ");
return true;
......@@ -364,38 +364,38 @@ bool AudioRecord::openExistingWavFile()
void AudioRecord::closeWavFile()
{
if (fileHandle_ == 0) {
_debug("AudioRecord: Can't closeWavFile, a file has not yet been opened!");
DEBUG("AudioRecord: Can't closeWavFile, a file has not yet been opened!");
return;
}
_debug("AudioRecord: Close wave file");
DEBUG("AudioRecord: Close wave file");
SINT32 bytes = byteCounter_ * channels_;
fseek(fileHandle_, 40, SEEK_SET); // jump to data length
if (ferror(fileHandle_))
_warn("AudioRecord: Error: can't reach offset 40 while closing");
WARN("AudioRecord: Error: can't reach offset 40 while closing");
fwrite(&bytes, sizeof(SINT32), 1, fileHandle_);
if (ferror(fileHandle_))
_warn("AudioRecord: Error: can't write bytes for data length ");
WARN("AudioRecord: Error: can't write bytes for data length ");
bytes = byteCounter_ * channels_ + 44; // + 44 for the wave header
fseek(fileHandle_, 4, SEEK_SET); // jump to file size
if (ferror(fileHandle_))
_warn("AudioRecord: Error: can't reach offset 4");
WARN("AudioRecord: Error: can't reach offset 4");
fwrite(&bytes, 4, 1, fileHandle_);
if (ferror(fileHandle_))
_warn("AudioRecord: Error: can't reach offset 4");
WARN("AudioRecord: Error: can't reach offset 4");
if (fclose(fileHandle_) != 0)
_warn("AudioRecord: Error: can't close file");
WARN("AudioRecord: Error: can't close file");
}
void AudioRecord::recSpkrData(SFLDataFormat* buffer, int nSamples)
......@@ -425,12 +425,12 @@ void AudioRecord::recData(SFLDataFormat* buffer, int nSamples)
{
if (recordingEnabled_) {
if (fileHandle_ == 0) {
_debug("AudioRecord: Can't record data, a file has not yet been opened!");
DEBUG("AudioRecord: Can't record data, a file has not yet been opened!");
return;
}
if (fwrite(buffer, sizeof(SFLDataFormat), nSamples, fileHandle_) != (unsigned int) nSamples)
_warn("AudioRecord: Could not record data! ");
WARN("AudioRecord: Could not record data! ");
else {
fflush(fileHandle_);
byteCounter_ += (unsigned long)(nSamples*sizeof(SFLDataFormat));
......@@ -444,7 +444,7 @@ void AudioRecord::recData(SFLDataFormat* buffer_1, SFLDataFormat* buffer_2,
{
if (recordingEnabled_) {
if (fileHandle_ == 0) {
_debug("AudioRecord: Can't record data, a file has not yet been opened!");
DEBUG("AudioRecord: Can't record data, a file has not yet been opened!");
return;
}
......@@ -452,7 +452,7 @@ void AudioRecord::recData(SFLDataFormat* buffer_1, SFLDataFormat* buffer_2,
mixBuffer_[k] = (buffer_1[k]+buffer_2[k]);
if (fwrite(&mixBuffer_[k], 2, 1, fileHandle_) != 1)
_warn("AudioRecord: Could not record data!");
WARN("AudioRecord: Could not record data!");
else
fflush(fileHandle_);
}
......
......@@ -125,7 +125,7 @@ int AudioRtpRecordHandler::processDataEncode()
int bytes = Manager::instance().getMainBuffer()->getData(micData, bytesToGet, id_);
if (bytes != bytesToGet) {
_error("%s : asked %d bytes from mainbuffer, got %d", __PRETTY_FUNCTION__, bytesToGet, bytes);
ERROR("%s : asked %d bytes from mainbuffer, got %d", __PRETTY_FUNCTION__, bytesToGet, bytes);
return 0;
}
......
......@@ -72,7 +72,7 @@ void AudioRtpSession::updateSessionMedia(AudioCodec *audioCodec)
Manager::instance().audioSamplingRateChanged(audioRtpRecord_.codecSampleRate_);
if (lastSamplingRate != audioRtpRecord_.codecSampleRate_) {
_debug("AudioRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize());
DEBUG("AudioRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize());
initNoiseSuppress();
}
......@@ -94,20 +94,20 @@ void AudioRtpSession::setSessionMedia(AudioCodec *audioCodec)
else
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("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_);
if (payloadType == g722PayloadType) {
_debug("AudioSymmetricRtpSession: Setting G722 payload format");
DEBUG("AudioSymmetricRtpSession: Setting G722 payload format");
queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, g722RtpClockRate));
} else {
if (dynamic) {
_debug("AudioSymmetricRtpSession: Setting dynamic payload format");
DEBUG("AudioSymmetricRtpSession: Setting dynamic payload format");
queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, smplRate));
} else {
_debug("AudioSymmetricRtpSession: Setting static payload format");
DEBUG("AudioSymmetricRtpSession: Setting static payload format");
queue_->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) payloadType));
}
}
......@@ -127,7 +127,7 @@ void AudioRtpSession::sendDtmfEvent()
audioRtpRecord_.dtmfQueue_.pop_front();
_debug("AudioRtpSession: Send RTP Dtmf (%d)", payload.event);
DEBUG("AudioRtpSession: Send RTP Dtmf (%d)", payload.event);
timestamp_ += (type_ == Zrtp) ? 160 : timestampIncrement_;
......@@ -186,7 +186,7 @@ void AudioRtpSession::sendMicData()
void AudioRtpSession::setSessionTimeouts()
{
_debug("AudioRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
DEBUG("AudioRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
queue_->setSchedulingTimeout(sfl::schedulingTimeout);
queue_->setExpireTimeout(sfl::expireTimeout);
......@@ -194,13 +194,13 @@ void AudioRtpSession::setSessionTimeouts()
void AudioRtpSession::setDestinationIpAddress()
{
_info("AudioRtpSession: Setting IP address for the RTP session");
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());
if (!remote_ip_) {
_warn("AudioRtpSession: Target IP address (%s) is not correct!",
WARN("AudioRtpSession: Target IP address (%s) is not correct!",
ca_->getLocalSDP()->getRemoteIP().data());
return;
}
......@@ -208,24 +208,24 @@ 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",
INFO("AudioRtpSession: New remote address for session: %s:%d",
ca_->getLocalSDP()->getRemoteIP().data(), remote_port_);
if (!queue_->addDestination(remote_ip_, remote_port_)) {
_warn("AudioRtpSession: Can't add new destination to session!");
WARN("AudioRtpSession: Can't add new destination to session!");
return;
}
}
void AudioRtpSession::updateDestinationIpAddress()
{
_debug("AudioRtpSession: Update destination ip address");
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))
_debug("AudioRtpSession: Did not remove previous destination");
DEBUG("AudioRtpSession: Did not remove previous destination");
// new destination is stored in call
// we just need to recall this method
......@@ -238,7 +238,7 @@ int AudioRtpSession::startRtpThread(AudioCodec* audiocodec)
if (isStarted_)
return 0;
_debug("AudioSymmetricRtpSession: Starting main thread");
DEBUG("AudioSymmetricRtpSession: Starting main thread");
isStarted_ = true;
setSessionTimeouts();
......
......@@ -61,12 +61,12 @@ AudioSrtpSession::AudioSrtpSession(SIPCall * sipcall) :
AudioSrtpSession::~AudioSrtpSession()
{
_debug("AudioSrtp: Destroy audio srtp session");
DEBUG("AudioSrtp: Destroy audio srtp session");
}
void AudioSrtpSession::initLocalCryptoInfo()
{
_debug("AudioSrtp: Set cryptographic info for this rtp session");
DEBUG("AudioSrtp: Set cryptographic info for this rtp session");
// Initialize local Crypto context
initializeLocalMasterKey();
......@@ -82,7 +82,7 @@ void AudioSrtpSession::initLocalCryptoInfo()
std::vector<std::string> AudioSrtpSession::getLocalCryptoInfo()
{
_debug("AudioSrtp: Get Cryptographic info from this rtp session");
DEBUG("AudioSrtp: Get Cryptographic info from this rtp session");
std::vector<std::string> crypto_vector;
......@@ -104,7 +104,7 @@ std::vector<std::string> AudioSrtpSession::getLocalCryptoInfo()
crypto_attr += crypto_suite.append(" ");
crypto_attr += srtp_keys;
_debug("%s", crypto_attr.c_str());
DEBUG("%s", crypto_attr.c_str());
crypto_vector.push_back(crypto_attr);
......@@ -114,7 +114,7 @@ std::vector<std::string> AudioSrtpSession::getLocalCryptoInfo()
void AudioSrtpSession::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
{
if (not remoteOfferIsSet_) {
_debug("%s", nego.getKeyInfo().c_str());
DEBUG("%s", nego.getKeyInfo().c_str());
// Use second crypto suite if key length is 32 bit, default is 80;
......@@ -137,12 +137,12 @@ void AudioSrtpSession::setRemoteCryptoInfo(sfl::SdesNegotiator& nego)
void AudioSrtpSession::initializeLocalMasterKey()
{
_debug("AudioSrtp: Init local master key");
DEBUG("AudioSrtp: Init local master key");
// @TODO key may have different length depending on cipher suite
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_];
......@@ -151,7 +151,7 @@ void AudioSrtpSession::initializeLocalMasterKey()
int err;
if ((err = RAND_bytes(random_key, localMasterKeyLength_)) != 1)
_debug("Error occured while generating cryptographically strong pseudo-random key");
DEBUG("Error occured while generating cryptographically strong pseudo-random key");
memcpy(localMasterKey_, random_key, localMasterKeyLength_);
}
......@@ -164,27 +164,27 @@ void AudioSrtpSession::initializeLocalMasterSalt()
// Allocate memory for key
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)
_debug("Error occured while generating cryptographically strong pseudo-random key");
DEBUG("Error occured while generating cryptographically strong pseudo-random key");
memcpy(localMasterSalt_, random_key, localMasterSaltLength_);
}
std::string AudioSrtpSession::getBase64ConcatenatedKeys()
{
_debug("AudioSrtp: Get base64 concatenated keys");
DEBUG("AudioSrtp: Get base64 concatenated keys");
// compute concatenated master and salt length
int concatLength = localMasterKeyLength_ + localMasterSaltLength_;
uint8 concatKeys[concatLength];
_debug("AudioSrtp: Concatenated length %d", concatLength);
DEBUG("AudioSrtp: Concatenated length %d", concatLength);
// concatenate keys
memcpy((void*) concatKeys, (void*) localMasterKey_, localMasterKeyLength_);
......@@ -214,7 +214,7 @@ void AudioSrtpSession::unBase64ConcatenatedKeys(std::string base64keys)
void AudioSrtpSession::initializeRemoteCryptoContext()
{
_debug("AudioSrtp: Initialize remote crypto context");
DEBUG("AudioSrtp: Initialize remote crypto context");
CryptoSuiteDefinition crypto = sfl::CryptoSuites[remoteCryptoSuite_];
......@@ -241,7 +241,7 @@ void AudioSrtpSession::initializeRemoteCryptoContext()
void AudioSrtpSession::initializeLocalCryptoContext()
{
_debug("AudioSrtp: Initialize local crypto context");
DEBUG("AudioSrtp: Initialize local crypto context");
CryptoSuiteDefinition crypto = sfl::CryptoSuites[localCryptoSuite_];
......
......@@ -46,7 +46,7 @@ AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall * sipcall) :
, AudioRtpSession(sipcall, Symmetric, this, this)
, rtpThread_(new AudioRtpThread(this))
{
_info("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
INFO("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
audioRtpRecord_.callId_ = ca_->getCallId();
}
......@@ -65,7 +65,7 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
TimerPort::setTimer(threadSleep);
_debug("AudioRtpThread: Entering Audio rtp thread main loop");
DEBUG("AudioRtpThread: Entering Audio rtp thread main loop");
while (running) {
// Send session
......@@ -79,7 +79,7 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
TimerPort::incTimer(threadSleep);
}
_debug("AudioRtpThread: Leaving audio rtp thread loop");
DEBUG("AudioRtpThread: Leaving audio rtp thread loop");
}
}
......@@ -58,12 +58,12 @@ AudioZrtpSession::AudioZrtpSession(SIPCall * sipcall, const std::string& zidFile
ost::defaultApplication()),
zidFilename_(zidFilename)
{
_debug("AudioZrtpSession initialized");
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()
......@@ -93,24 +93,24 @@ void AudioZrtpSession::initializeZid()
std::string xdg_config = std::string(HOMEDIR) + DIR_SEPARATOR_STR + ".cache" + DIR_SEPARATOR_STR + PACKAGE + "/" + zidFilename_;
_debug(" xdg_config %s", xdg_config.c_str());
DEBUG(" xdg_config %s", xdg_config.c_str());
if (XDG_CACHE_HOME != NULL) {