From 4b565bec5a2dd574b4087ed367ab4b1f3c6ee7f5 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.net>
Date: Tue, 22 Sep 2009 16:51:02 -0400
Subject: [PATCH] [#2176] Fix fix audio stream binding in iax
---
sflphone-common/src/iax/iaxvoiplink.cpp | 23 ++++++++++++++++++++++-
1 file changed, 22 insertions(+), 1 deletion(-)
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index 728000d8d8..2e1201bfff 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -271,6 +271,8 @@ IAXVoIPLink::sendAudioFromMic (void)
AudioCodec *ac;
IAXCall *currentCall;
+
+
// We have to update the audio layer type in case we switched
// TODO Find out a better way to do it
updateAudiolayer();
@@ -278,10 +280,14 @@ IAXVoIPLink::sendAudioFromMic (void)
currentCall = getIAXCall (Manager::instance().getCurrentCallId());
if (!currentCall) {
+
+
// Let's mind our own business.
return;
}
+
+
if (currentCall -> getAudioCodec() < 0)
return;
@@ -307,9 +313,13 @@ IAXVoIPLink::sendAudioFromMic (void)
return;
}
+
+
// Send sound here
if (audiolayer) {
+
+
// we have to get 20ms of data from the mic *20/1000 = /50
// rate/50 shall be lower than IAX__20S_48KHZ_MAX
maxBytesToGet = audiolayer->getSampleRate() * audiolayer->getFrameSize() / 1000 * sizeof (SFLDataFormat);
@@ -330,6 +340,8 @@ IAXVoIPLink::sendAudioFromMic (void)
// Get bytes from micRingBuffer to data_from_mic
nbSample_ = audiolayer->getMainBuffer()->getData (micData, bytesAvail, 100, currentCall->getCallId()) / sizeof (SFLDataFormat);
+
+
// Store the number of samples for recording
nbSampleForRec_ = nbSample_;
@@ -526,6 +538,7 @@ IAXVoIPLink::peerHungup (const CallID& id)
IAXCall* call = getIAXCall (id);
std::string reason = "Dumped Call";
CHK_VALID_CALL;
+
_mutexIAX.enterMutex();
_mutexIAX.leaveMutex();
@@ -769,6 +782,9 @@ IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
case IAX_EVENT_ANSWER:
if (call->getConnectionState() != Call::Connected) {
+
+ Manager::instance().addStream(call->getCallId());
+
call->setConnectionState (Call::Connected);
call->setState (Call::Active);
audiolayer->startStream();
@@ -850,6 +866,8 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
int expandedSize, nbSample_;
AudioCodec *ac;
+
+
// If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
// were triggered
@@ -859,7 +877,8 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
return;
}
- if (audiolayer) {
+ if (audiolayer) {
+
// On-the-fly codec changing (normally, when we receive a full packet)
// as per http://tools.ietf.org/id/draft-guy-iax-03.txt
// - subclass holds the voiceformat property.
@@ -1000,6 +1019,8 @@ IAXVoIPLink::iaxHandlePrecallEvent (iax_event* event)
id = Manager::instance().getNewCallID();
+ _debug("-------------------------------------------------------------- callid %s", id.c_str());
+
call = new IAXCall (id, Call::Incoming);
if (!call) {
--
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