diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index 10f481487dab6c6f2f837b26e9162a51ee9680c2..2f0005241bb689bd35cbf1f9ef0a7fcc065ed91c 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -110,7 +110,6 @@ AudioRtp::closeRtpSession ()
// This will make RTP threads finish.
_debug ("AudioRtp::Stopping rtp session\n");
-
try {
delete _RTXThread;
_RTXThread = 0;
@@ -119,9 +118,6 @@ AudioRtp::closeRtpSession ()
throw;
}
- // AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- // audiolayer->stopStream();
-
_debug ("AudioRtp::Audio rtp stopped\n");
return true;
diff --git a/sflphone-common/src/audio/audiostream.cpp b/sflphone-common/src/audio/audiostream.cpp
index 8054c1e10ea9bb01056a214127dec0eb77b40029..4af1e26c0416bed42f3663bb3ef61f19d7446578 100644
--- a/sflphone-common/src/audio/audiostream.cpp
+++ b/sflphone-common/src/audio/audiostream.cpp
@@ -23,18 +23,21 @@
static pa_channel_map channel_map ;
-AudioStream::AudioStream (pa_context* context, int type, std::string desc, double vol UNUSED)
- : _audiostream (NULL), _streamType (type), _streamDescription (desc), flag (PA_STREAM_AUTO_TIMING_UPDATE), sample_spec(), _volume()
+AudioStream::AudioStream (PulseLayerType * driver)
+ : _audiostream (NULL),
+ _context (driver->context),
+ _streamType (driver->type),
+ _streamDescription (driver->description),
+ _volume(),
+ _mainloop(driver->mainloop),
+ flag (PA_STREAM_AUTO_TIMING_UPDATE),
+ sample_spec()
{
sample_spec.format = PA_SAMPLE_S16LE;
sample_spec.rate = 44100;
sample_spec.channels = 1;
channel_map.channels = 1;
pa_cvolume_set (&_volume , 1 , PA_VOLUME_NORM) ; // * vol / 100 ;
-
- _context = context;
-
- // connectStream();
}
AudioStream::~AudioStream()
@@ -59,10 +62,14 @@ AudioStream::disconnectStream (void)
ost::MutexLock guard (_mutex);
_debug ("Destroy audio streams\n");
- pa_stream_disconnect (_audiostream);
- pa_stream_unref (_audiostream);
-
- _audiostream = NULL;
+
+ pa_threaded_mainloop_lock(_mainloop);
+ if(_audiostream) {
+ pa_stream_disconnect (_audiostream);
+ pa_stream_unref (_audiostream);
+ _audiostream = NULL;
+ }
+ pa_threaded_mainloop_unlock(_mainloop);
return true;
}
@@ -70,13 +77,16 @@ AudioStream::disconnectStream (void)
void
-AudioStream::stream_state_callback (pa_stream* s, void* user_data UNUSED)
+AudioStream::stream_state_callback (pa_stream* s, void* user_data)
{
-
-
+ pa_threaded_mainloop *m;
+
_debug ("AudioStream::stream_state_callback :: The state of the stream changed\n");
assert (s);
+ m = (pa_threaded_mainloop*) user_data;
+ assert(m);
+
switch (pa_stream_get_state (s)) {
case PA_STREAM_CREATING:
@@ -85,7 +95,7 @@ AudioStream::stream_state_callback (pa_stream* s, void* user_data UNUSED)
case PA_STREAM_TERMINATED:
_debug ("Stream is terminating...\n");
- PulseLayer::streamState++;
+ pa_threaded_mainloop_signal(m, 0);
break;
case PA_STREAM_READY:
@@ -162,7 +172,7 @@ AudioStream::createStream (pa_context* c)
_debug ("Stream type unknown \n");
}
- pa_stream_set_state_callback (s , stream_state_callback, NULL);
+ pa_stream_set_state_callback (s , stream_state_callback, _mainloop);
free (attributes);
diff --git a/sflphone-common/src/audio/audiostream.h b/sflphone-common/src/audio/audiostream.h
index f21e4b0d27654a5efa5f28e919de6817de075d0b..2314d31670ac28fede04fb0036e9b605210e1a57 100644
--- a/sflphone-common/src/audio/audiostream.h
+++ b/sflphone-common/src/audio/audiostream.h
@@ -27,6 +27,7 @@
#include "ringbuffer.h"
#include "audioloop.h"
+
#include <cc++/thread.h>
/**
@@ -38,16 +39,23 @@ enum STREAM_TYPE {
UPLOAD_STREAM
};
+struct PulseLayerType {
+ pa_context * context;
+ pa_threaded_mainloop * mainloop;
+
+ std::string description;
+
+ int type;
+ double volume;
+};
class AudioStream {
public:
/**
* Constructor
- * @param context The pulseaudio context
- * @param type The type of audio stream
- * @param desc The stream name
+ * @param context The PulseLayerType structure containing various information.
*/
- AudioStream(pa_context* context , int type, std::string desc, double vol);
+ AudioStream(PulseLayerType * driver);
/**
* Destructor
@@ -166,6 +174,8 @@ class AudioStream {
pa_sample_spec sample_spec ;
pa_cvolume _volume;
+ pa_threaded_mainloop * _mainloop;
+
ost::Mutex _mutex;
};
diff --git a/sflphone-common/src/audio/pulselayer.cpp b/sflphone-common/src/audio/pulselayer.cpp
index 9d586d2041301be4440f6631b55335459a95c87c..39d3ba887d32ab28d818043428f879ea55078013 100644
--- a/sflphone-common/src/audio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulselayer.cpp
@@ -21,8 +21,6 @@
int framesPerBuffer = 2048;
-int PulseLayer::streamState;
-
static void audioCallback (pa_stream* s, size_t bytes, void* userdata)
{
assert (s && bytes);
@@ -39,7 +37,6 @@ PulseLayer::PulseLayer (ManagerImpl* manager)
, playback()
, record()
{
- PulseLayer::streamState = 0;
_debug ("PulseLayer::Pulse audio constructor: Create context\n");
}
@@ -48,9 +45,6 @@ PulseLayer::PulseLayer (ManagerImpl* manager)
PulseLayer::~PulseLayer (void)
{
closeLayer ();
-
- // pa_context_disconnect( context );
- // pa_context_unref( context );
}
bool
@@ -61,17 +55,16 @@ PulseLayer::closeLayer (void)
playback->disconnectStream();
record->disconnectStream();
- while (PulseLayer::streamState != 2);
-
- PulseLayer::streamState = 0;
-
- //TODO Remove this ugly hack
- sleep (2);
-
- pa_context_disconnect (context);
-
- pa_context_unref (context);
-
+ pa_threaded_mainloop_lock (m);
+ pa_threaded_mainloop_wait(m);
+ if(m) {
+ pa_context_disconnect (context);
+ pa_context_unref (context);
+ }
+ pa_threaded_mainloop_unlock (m);
+
+ pa_threaded_mainloop_free (m);
+
return true;
}
@@ -167,18 +160,29 @@ bool PulseLayer::createStreams (pa_context* c)
{
_debug ("PulseLayer::createStreams \n");
- playback = new AudioStream (c, PLAYBACK_STREAM, PLAYBACK_STREAM_NAME, _manager->getSpkrVolume());
+ PulseLayerType * playbackParam = new PulseLayerType();
+ playbackParam->context = c;
+ playbackParam->type = PLAYBACK_STREAM;
+ playbackParam->description = PLAYBACK_STREAM_NAME;
+ playbackParam->volume = _manager->getSpkrVolume();
+ playbackParam->mainloop = m;
+
+ playback = new AudioStream (playbackParam);
playback->connectStream();
pa_stream_set_write_callback (playback->pulseStream(), audioCallback, this);
- // pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
- // pa_stream_set_suspended_callback( playback->pulseStream(), stream_suspended_callback, this);
-
- record = new AudioStream (c, CAPTURE_STREAM, CAPTURE_STREAM_NAME , _manager->getMicVolume());
+ delete playbackParam;
+
+ PulseLayerType * recordParam = new PulseLayerType();
+ recordParam->context = c;
+ recordParam->type = CAPTURE_STREAM;
+ recordParam->description = CAPTURE_STREAM_NAME;
+ recordParam->volume = _manager->getMicVolume();
+ recordParam->mainloop = m;
+
+ record = new AudioStream (recordParam);
record->connectStream();
pa_stream_set_read_callback (record->pulseStream() , audioCallback, this);
- // pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
- // pa_stream_set_suspended_callback(record->pulseStream(), stream_suspended_callback, this);
-
+ delete recordParam;
pa_threaded_mainloop_signal (m , 0);
diff --git a/sflphone-common/src/audio/pulselayer.h b/sflphone-common/src/audio/pulselayer.h
index 698cc9bf6b68b6ff7009b35efcccba3c664d2729..8c1fa083842f9990b73f141730cf849146655280 100644
--- a/sflphone-common/src/audio/pulselayer.h
+++ b/sflphone-common/src/audio/pulselayer.h
@@ -197,7 +197,6 @@ class PulseLayer : public AudioLayer {
// private:
public:
- static int streamState;
friend class AudioLayerTest;
};
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index aa1e2f9a410b4a267212a636d4249d40bb7d0a79..af41b6d268c760870fcca500204e488df9331432 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -601,8 +601,6 @@ SIPVoIPLink::answer (const CallID& id)
call->setConnectionState (Call::Connected);
call->setState (Call::Active);
- ;
-
return true;
} else {
// Create and send a 488/Not acceptable here
@@ -617,6 +615,7 @@ SIPVoIPLink::answer (const CallID& id)
_debug ("SIPVoIPLink::answer: fail terminate call %s \n",call->getCallId().c_str());
terminateOneCall (call->getCallId());
removeCall (call->getCallId());
+ _audiortp->closeRtpSession ();
return false;
}
}
@@ -1195,39 +1194,25 @@ SIPVoIPLink::SIPCheckUrl (const std::string& url UNUSED)
void
SIPVoIPLink::SIPCallServerFailure (SIPCall *call)
{
- //if (!event->response) { return; }
- //switch(event->response->status_code) {
- //case SIP_SERVICE_UNAVAILABLE: // 500
- //case SIP_BUSY_EVRYWHERE: // 600
- //case SIP_DECLINE: // 603
- //SIPCall* call = findSIPCallWithCid(event->cid);
if (call != 0) {
_debug ("Server error!\n");
CallID id = call->getCallId();
Manager::instance().callFailure (id);
terminateOneCall (id);
removeCall (id);
+ _audiortp->closeRtpSession();
}
-
- //break;
- //}
}
void
SIPVoIPLink::SIPCallClosed (SIPCall *call)
{
-
-
- // it was without did before
- //SIPCall* call = findSIPCallWithCid(event->cid);
if (!call) {
return;
}
CallID id = call->getCallId();
- //call->setDid(event->did);
-
if (Manager::instance().isCurrentCall (id)) {
call->setAudioStart (false);
_debug ("* SIP Info: Stopping AudioRTP when closing\n");
@@ -1245,9 +1230,6 @@ SIPVoIPLink::SIPCallClosed (SIPCall *call)
void
SIPVoIPLink::SIPCallReleased (SIPCall *call)
{
- // do cleanup if exists
- // only cid because did is always 0 in these case..
- //SIPCall* call = findSIPCallWithCid(event->cid);
if (!call) {
return;
}
@@ -2050,7 +2032,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
break;
default:
- _debug ("sipvoiplink.cpp - line 1635 : Unhandled call state. This is probably a bug.\n");
+ _debug ("sipvoiplink.cpp - line %d : Unhandled call state. This is probably a bug.\n", __LINE__);
break;
}
}