From b9d248dc08ce00bc1eb1dc9ab39631ef88c953f5 Mon Sep 17 00:00:00 2001
From: asavard <asavard@asavard-KT378AA-A2L-a6552f.(none)>
Date: Wed, 27 Oct 2010 14:15:13 -0400
Subject: [PATCH] [#4367] Remove debug in rtp session
---
.../src/audio/audiortp/AudioRtpRecordHandler.cpp | 4 ----
sflphone-common/src/audio/audiortp/AudioRtpSession.cpp | 8 --------
2 files changed, 12 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index 28db02a4d9..1712c887dd 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -387,7 +387,6 @@ void AudioRtpRecordHandler::putDtmfEvent(int digit)
int AudioRtpRecordHandler::processDataEncode(void)
{
- _debug("Process data encode");
AudioCodec *audioCodec = getAudioCodec();
AudioLayer *audioLayer = Manager::instance().getAudioDriver();
@@ -407,11 +406,8 @@ int AudioRtpRecordHandler::processDataEncode(void)
// compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
int bytesToGet = computeNbByteAudioLayer(fixedCodecFramesize);
- _debug(" byte to get %d", bytesToGet);
-
// available bytes inside ringbuffer
int availBytesFromMic = audioLayer->getMainBuffer()->availForGet(_ca->getCallId());
- _debug(" avail byte from mic %d", availBytesFromMic);
if(availBytesFromMic < bytesToGet)
return 0;
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index 40b6f419d0..2e213d4346 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -235,8 +235,6 @@ bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
void AudioRtpSession::sendMicData()
{
- _debug("============== sendMicData ===============");
-
int compSize = processDataEncode();
// If no data, return
@@ -254,9 +252,6 @@ void AudioRtpSession::sendMicData()
// Increment timestamp for outgoing packet
_timestamp += _timestampIncrement;
- _debug(" compSize: %d", compSize);
- _debug(" timestamp: %d", _timestamp);
-
// putData put the data on RTP queue, sendImmediate bypass this queue
putData (_timestamp, getMicDataEncoded(), compSize);
}
@@ -302,8 +297,6 @@ void AudioRtpSession::notifyIncomingCall()
_countNotificationTime += _time->getSecond();
int countTimeModulo = _countNotificationTime % 5000;
- // _debug("countNotificationTime: %d\n", countNotificationTime);
- // _debug("countTimeModulo: %d\n", countTimeModulo);
if ( (countTimeModulo - _countNotificationTime) < 0) {
Manager::instance().notificationIncomingCall();
}
@@ -346,7 +339,6 @@ void AudioRtpSession::run ()
// Start audio stream (if not started) AND flush all buffers (main and urgent)
_manager->getAudioDriver()->startStream();
- // startRunning();
_debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
--
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