Commit c9a15deb authored by Tristan Matthews's avatar Tristan Matthews
Browse files

* #9832: AudioRtpSession: cleanup logging

parent f3af752f
......@@ -33,12 +33,9 @@
*/
#include "audio_rtp_session.h"
#include "audio_symmetric_rtp_session.h"
#include "logger.h"
#include "sip/sdp.h"
#include "sip/sipcall.h"
#include "audio/audiolayer.h"
#include <ccrtp/rtp.h>
#include <ccrtp/oqueue.h>
#include "manager.h"
......@@ -72,10 +69,10 @@ 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("Update noise suppressor with sampling rate %d and frame size %d",
getCodecSampleRate(), getCodecFrameSize());
initNoiseSuppress();
}
}
void AudioRtpSession::setSessionMedia(AudioCodec &audioCodec)
......@@ -94,20 +91,20 @@ void AudioRtpSession::setSessionMedia(AudioCodec &audioCodec)
else
timestampIncrement_ = frameSize;
DEBUG("AudioRtpSession: Codec payload: %d", payloadType);
DEBUG("AudioRtpSession: Codec sampling rate: %d", smplRate);
DEBUG("AudioRtpSession: Codec frame size: %d", frameSize);
DEBUG("AudioRtpSession: RTP timestamp increment: %d", timestampIncrement_);
DEBUG("Codec payload: %d", payloadType);
DEBUG("Codec sampling rate: %d", smplRate);
DEBUG("Codec frame size: %d", frameSize);
DEBUG("RTP timestamp increment: %d", timestampIncrement_);
if (payloadType == g722PayloadType) {
DEBUG("AudioRtpSession: Setting G722 payload format");
DEBUG("Setting G722 payload format");
queue_.setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, g722RtpClockRate));
} else {
if (dynamic) {
DEBUG("AudioRtpSession: Setting dynamic payload format");
DEBUG("Setting dynamic payload format");
queue_.setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, smplRate));
} else {
DEBUG("AudioRtpSession: Setting static payload format");
DEBUG("Setting static payload format");
queue_.setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) payloadType));
}
}
......@@ -129,7 +126,7 @@ void AudioRtpSession::sendDtmfEvent()
audioRtpRecord_.dtmfQueue_.pop_front();
DEBUG("AudioRtpSession: Send RTP Dtmf (%d)", payload.event);
DEBUG("Send RTP Dtmf (%d)", payload.event);
incrementTimestampForDTMF();
......@@ -184,7 +181,8 @@ void AudioRtpSession::sendMicData()
void AudioRtpSession::setSessionTimeouts()
{
DEBUG("AudioRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
DEBUG("Set session scheduling timeout (%d) and expireTimeout (%d)",
sfl::schedulingTimeout, sfl::expireTimeout);
queue_.setSchedulingTimeout(sfl::schedulingTimeout);
queue_.setExpireTimeout(sfl::expireTimeout);
......@@ -196,7 +194,7 @@ void AudioRtpSession::setDestinationIpAddress()
remote_ip_ = ost::InetHostAddress(call_.getLocalSDP()->getRemoteIP().c_str());
if (!remote_ip_) {
WARN("AudioRtpSession: Target IP address (%s) is not correct!",
WARN("Target IP address (%s) is not correct!",
call_.getLocalSDP()->getRemoteIP().data());
return;
}
......@@ -204,24 +202,24 @@ void AudioRtpSession::setDestinationIpAddress()
// Store remote port in case we would need to forget current destination
remote_port_ = (unsigned short) call_.getLocalSDP()->getRemoteAudioPort();
DEBUG("AudioRtpSession: New remote address for session: %s:%d",
DEBUG("New remote address for session: %s:%d",
call_.getLocalSDP()->getRemoteIP().data(), remote_port_);
if (!queue_.addDestination(remote_ip_, remote_port_)) {
WARN("AudioRtpSession: Can't add new destination to session!");
WARN("Can't add new destination to session!");
return;
}
}
void AudioRtpSession::updateDestinationIpAddress()
{
DEBUG("AudioRtpSession: Update destination ip address");
DEBUG("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("Did not remove previous destination");
// new destination is stored in call
// we just need to recall this method
......@@ -234,7 +232,7 @@ int AudioRtpSession::startRtpThread(AudioCodec &audiocodec)
if (isStarted_)
return 0;
DEBUG("AudioRtpSession: Starting main thread");
DEBUG("Starting main thread");
isStarted_ = true;
setSessionTimeouts();
......
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