From 39c656f24c5f67db367bb8a77beb6339e5c960a5 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
Date: Mon, 1 Nov 2010 12:11:10 -0400
Subject: [PATCH] [#4367] Reduce packet receive/sent timeout
---
.../src/audio/audiortp/AudioRtpRecordHandler.h | 4 ++--
.../src/audio/audiortp/AudioRtpSession.cpp | 18 ++++++++++++------
2 files changed, 14 insertions(+), 8 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
index 6a92ab5b25..28ce2ed088 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
@@ -47,8 +47,8 @@ namespace sfl
// Factor use to increase volume in fade in
#define FADEIN_STEP_SIZE 4;
-static const int schedulingTimeout = 100000;
-static const int expireTimeout = 1000000;
+static const int schedulingTimeout = 10000;
+static const int expireTimeout = 100000;
class AudioRtpSessionException: public std::exception
{
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index e12142b5b0..fbe0079f6d 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -72,7 +72,7 @@ AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
// static_cast<ost::DualRTPUDPIPv4Channel>(dso)->sendSocket->setTypeOfService(ost::Socket::tosLowDelay);
// static_cast<ost::DualRTPChannel<ost::DualRTPUDPIPv4Channel> >(dso)->sendSocket->setTypeOfService(ost::Socket::tosLowDelay);
- setTypeOfService(tosEnhanced);
+ // setTypeOfService(tosEnhanced);
}
AudioRtpSession::~AudioRtpSession()
@@ -261,6 +261,7 @@ void AudioRtpSession::sendMicData()
// putData put the data on RTP queue, sendImmediate bypass this queue
putData (_timestamp, getMicDataEncoded(), compSize);
+ // sendData()
}
@@ -347,10 +348,12 @@ void AudioRtpSession::run ()
uint32 timeout = 0;
while ( isActive() ) {
- if ( timeout < 100 ){ // !(timeout/1000)
+ if ( timeout < 1000 ){ // !(timeout/1000)
timeout = getSchedulingTimeout();
}
+ // timeout = ;
+
_manager->getAudioLayerMutex()->enter();
// Send session
@@ -360,6 +363,8 @@ void AudioRtpSession::run ()
sendMicData ();
}
+ receiveSpeakerData();
+
// This also should be moved
notifyIncomingCall();
@@ -373,14 +378,17 @@ void AudioRtpSession::run ()
// make sure the scheduling timeout is
// <= the check interval for RTCP
// packets
+ _debug("timeout before: %d, maxwait %d", timeout, maxWait);
timeout = (timeout > maxWait)? maxWait : timeout;
- if ( timeout < 100 ) { // !(timeout/1000)
+ _debug("timeout after: %d", timeout);
+
+ if ( timeout < 1000 ) { // !(timeout/1000)
setCancel(cancelDeferred);
dispatchDataPacket();
setCancel(cancelImmediate);
timerTick();
} else {
- if ( isPendingData(timeout/100) ) {
+ if ( isPendingData(timeout/1000) ) {
setCancel(cancelDeferred);
if (isActive()) { // take in only if active
takeInDataPacket();
@@ -389,8 +397,6 @@ void AudioRtpSession::run ()
}
timeout = 0;
}
-
- receiveSpeakerData();
}
_debug ("AudioRtpSession: Left main loop for call %s", _ca->getCallId().c_str());
--
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