From 96f64a4f77ecce3c5b04e034bd19429643547825 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.net>
Date: Mon, 19 Oct 2009 18:03:12 -0400
Subject: [PATCH] [2308] Fix latency at the beginning of a call, when playing
DTMF and wehn starting tone
---
sflphone-common/src/audio/audiolayer.cpp | 4 ++--
sflphone-common/src/audio/audiortp/AudioRtpSession.h | 4 +++-
sflphone-common/src/managerimpl.cpp | 9 ++++++++-
3 files changed, 13 insertions(+), 4 deletions(-)
diff --git a/sflphone-common/src/audio/audiolayer.cpp b/sflphone-common/src/audio/audiolayer.cpp
index 9f1119ff69..8966eb6aba 100644
--- a/sflphone-common/src/audio/audiolayer.cpp
+++ b/sflphone-common/src/audio/audiolayer.cpp
@@ -24,13 +24,13 @@ void AudioLayer::flushMain (void)
ost::MutexLock guard (_mutex);
// should pass call id
- _mainBuffer.flush();
+ _mainBuffer.flushAllBuffers();
}
void AudioLayer::flushUrgent (void)
{
ost::MutexLock guard (_mutex);
- _urgentRingBuffer.flush();
+ _urgentRingBuffer.flushAll();
}
void AudioLayer::flushMic (void)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index bed84b1c3d..3ac8916b3a 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -498,10 +498,12 @@ namespace sfl {
}
_ca->setRecordingSmplRate(_audiocodec->getClockRate());
-
+
_audiolayer->startStream();
static_cast<D*>(this)->startRunning();
+ _audiolayer->flushUrgent();
+
_debug ("Entering RTP mainloop for callid %s\n",_ca->getCallId().c_str());
while (!testCancel()) {
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 052b87fadf..006ef35f3e 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -372,6 +372,7 @@ ManagerImpl::answerCall (const CallID& call_id)
}
+
if (!getAccountLink (account_id)->answer (call_id)) {
// error when receiving...
removeCallAccount (call_id);
@@ -777,7 +778,7 @@ ManagerImpl::refuseCall (const CallID& id)
AccountID accountid;
bool returnValue;
- stopTone (true);
+ stopTone (false);
int nbCalls = getCallList().size();
@@ -1700,6 +1701,8 @@ ManagerImpl::playDtmf (char code, bool isTalking)
// Put buffer to urgentRingBuffer
// put the size in bytes...
// so size * 1 channel (mono) * sizeof (bytes for the data)
+ audiolayer->flushUrgent();
+
audiolayer->startStream();
audiolayer->putUrgent (buf, size * sizeof (SFLDataFormat));
}
@@ -2030,6 +2033,8 @@ bool ManagerImpl::playATone (Tone::TONEID toneId)
if (audiolayer)
audiolayer->startStream();
+ audiolayer->flushUrgent();
+
if (_telephoneTone != 0) {
_toneMutex.enterMutex();
_telephoneTone->setCurrentTone (toneId);
@@ -2039,6 +2044,7 @@ bool ManagerImpl::playATone (Tone::TONEID toneId)
nbSamples = audioloop->getSize();
SFLDataFormat buf[nbSamples];
+
if (audiolayer) {
audiolayer->putUrgent (buf, nbSamples);
} else
@@ -2218,6 +2224,7 @@ void ManagerImpl::notificationIncomingCall (void)
SFLDataFormat buf[nbSampling];
tone.getNext (buf, tone.getSize());
/* Put the data in the urgent ring buffer */
+ audiolayer->flushUrgent();
audiolayer->putUrgent (buf, sizeof (SFLDataFormat) *nbSampling);
}
}
--
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