From 908e1fb31f60d40675de9b35cfdaaccc728b32d7 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
Date: Wed, 22 Aug 2012 12:32:36 -0400
Subject: [PATCH] #14848: Remove explicite reference to
audioCodec->getFrameSize move transportRate to audioRtpSession instead of
audioSymmetricRtpSession
---
daemon/src/audio/audiortp/audio_rtp_session.cpp | 7 +++++++
daemon/src/audio/audiortp/audio_rtp_session.h | 8 +++++++-
.../audio/audiortp/audio_symmetric_rtp_session.cpp | 11 -----------
.../src/audio/audiortp/audio_symmetric_rtp_session.h | 4 +---
daemon/src/audio/audiortp/audio_zrtp_session.cpp | 12 ++----------
daemon/src/audio/audiortp/audio_zrtp_session.h | 1 -
6 files changed, 17 insertions(+), 26 deletions(-)
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.cpp b/daemon/src/audio/audiortp/audio_rtp_session.cpp
index d1bfd45167..473db11a75 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_session.cpp
@@ -45,6 +45,7 @@ AudioRtpSession::AudioRtpSession(SIPCall &call, ost::RTPDataQueue &queue, ost::T
, call_(call)
, timestamp_(0)
, timestampIncrement_(0)
+ , transportRate_(20)
, queue_(queue)
, isStarted_(false)
, remote_ip_()
@@ -98,6 +99,12 @@ void AudioRtpSession::setSessionMedia(AudioCodec &audioCodec)
else
queue_.setPayloadFormat(ost::StaticPayloadFormat(static_cast<ost::StaticPayloadType>(payloadType)));
}
+
+ call_.setRecordingSmplRate(getCodecSampleRate());
+
+ int transportRate = getCodecFrameSize() / (getCodecSampleRate() / 1000);
+ transportRate_ = (transportRate > 0)?transportRate:20;
+ DEBUG("Switching to a transport rate of %d ms",transportRate_);
}
void AudioRtpSession::sendDtmfEvent()
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.h b/daemon/src/audio/audiortp/audio_rtp_session.h
index ca4f406bba..f62687da32 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_rtp_session.h
@@ -80,7 +80,7 @@ class AudioRtpSession : public AudioRtpRecordHandler {
/**
* Set the audio codec for this RTP session
*/
- virtual void setSessionMedia(AudioCodec &codec) = 0;
+ void setSessionMedia(AudioCodec &codec);
bool onRTPPacketRecv(ost::IncomingRTPPkt&);
@@ -111,6 +111,12 @@ class AudioRtpSession : public AudioRtpRecordHandler {
*/
int timestampIncrement_;
+ /**
+ * Rate at which the transport layer handle packets, should be
+ * synchronized with codec requirements.
+ */
+ unsigned int transportRate_;
+
ost::RTPDataQueue &queue_;
bool isStarted_;
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
index e7b6326121..84f3772086 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
@@ -43,7 +43,6 @@ AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall &call) :
, ost::SymmetricRTPSession(ost::InetHostAddress(call.getLocalIp().c_str()), call.getLocalAudioPort())
, AudioRtpSession(call, *this, *this)
, rtpThread_(*this)
- , transportRate_(20)
{
DEBUG("Setting new RTP session with destination %s:%d",
call_.getLocalIp().c_str(), call_.getLocalAudioPort());
@@ -78,16 +77,6 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
}
}
-void AudioSymmetricRtpSession::setSessionMedia(AudioCodec &audioCodec)
-{
- AudioRtpSession::setSessionMedia(audioCodec);
- call_.setRecordingSmplRate(getCodecSampleRate());
-
- int transportRate = audioCodec.getFrameSize() / (audioCodec.getClockRate()/1000);
- transportRate_ = (transportRate > 0)?transportRate:20;
- DEBUG("Switching to a transport rate of %d ms",transportRate_);
-}
-
int AudioSymmetricRtpSession::startRtpThread(AudioCodec &audiocodec)
{
DEBUG("Starting main thread");
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
index ba364a16a5..b7b9e58f57 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
@@ -96,11 +96,9 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
NON_COPYABLE(AudioRtpThread);
AudioSymmetricRtpSession &rtpSession_;
};
- void setSessionMedia(AudioCodec &codec);
int startRtpThread(AudioCodec &audiocodec);
- AudioRtpThread rtpThread_ ;
- unsigned int transportRate_;
+ AudioRtpThread rtpThread_;
};
}
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.cpp b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
index 0a109bb0e6..469e04e445 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
@@ -126,12 +126,9 @@ AudioZrtpSession::AudioZrtpThread::AudioZrtpThread(AudioZrtpSession &session) :
void AudioZrtpSession::AudioZrtpThread::run()
{
- // Set recording sampling rate
- int threadSleep = 20;
-
DEBUG("Entering Audio zrtp thread main loop %s", running_ ? "running" : "not running");
- TimerPort::setTimer(threadSleep);
+ TimerPort::setTimer(zrtpSession_.transportRate_);
while (running_) {
// Send session
@@ -142,7 +139,7 @@ void AudioZrtpSession::AudioZrtpThread::run()
Thread::sleep(TimerPort::getTimer());
- TimerPort::incTimer(threadSleep);
+ TimerPort::incTimer(zrtpSession_.transportRate_);
}
DEBUG("Leaving audio rtp thread loop");
@@ -153,11 +150,6 @@ int AudioZrtpSession::getIncrementForDTMF() const
return 160;
}
-void AudioZrtpSession::setSessionMedia(AudioCodec &audioCodec)
-{
- AudioRtpSession::setSessionMedia(audioCodec);
-}
-
int AudioZrtpSession::startRtpThread(AudioCodec &audiocodec)
{
if(isStarted_)
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.h b/daemon/src/audio/audiortp/audio_zrtp_session.h
index 8ddb53d7e1..827ce72741 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.h
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.h
@@ -96,7 +96,6 @@ class AudioZrtpSession :
void sendMicData();
void initializeZid();
std::string zidFilename_;
- void setSessionMedia(AudioCodec &codec);
int startRtpThread(AudioCodec &audiocodec);
virtual int getIncrementForDTMF() const;
--
GitLab