diff --git a/sflphone-client-gnome/src/config/audioconf.c b/sflphone-client-gnome/src/config/audioconf.c
index 9031c99febfd62ef4e740d56ef73b79f4b9cc88e..5dd23a4dd68a8251a7bcd0c89d409210acde3d6a 100644
--- a/sflphone-client-gnome/src/config/audioconf.c
+++ b/sflphone-client-gnome/src/config/audioconf.c
@@ -397,6 +397,8 @@ codec_active_toggled(GtkCellRendererToggle *renderer UNUSED, gchar *path, gpoint
codec = codec_list_get_by_payload(110);
else if ((g_strcasecmp(name,"speex")==0) && (g_strcasecmp(srate,"16 kHz")==0))
codec = codec_list_get_by_payload(111);
+ else if ((g_strcasecmp(name,"speex")==0) && (g_strcasecmp(srate,"32 kHz")==0))
+ codec = codec_list_get_by_payload(112);
else
codec = codec_list_get_by_name(name);
diff --git a/sflphone-client-gnome/src/contacts/calltree.c b/sflphone-client-gnome/src/contacts/calltree.c
index f87206be01e35176d50da316777963fe9ff60d52..d1bf035968d4411f05e90c0fdea0ec5a0e461871 100644
--- a/sflphone-client-gnome/src/contacts/calltree.c
+++ b/sflphone-client-gnome/src/contacts/calltree.c
@@ -412,8 +412,6 @@ calltree_update_call (calltab_t* tab, callable_obj_t * c)
void calltree_add_history_entry (callable_obj_t * c)
{
- printf("calltree_add_history_entry\n");
-
if (dbus_get_history_enabled () == 0)
return;
@@ -473,8 +471,6 @@ void calltree_add_history_entry (callable_obj_t * c)
void calltree_add_call (calltab_t* tab, callable_obj_t * c)
{
- printf("calltree_add_call\n");
-
if (tab == history)
{
calltree_add_history_entry (c);
diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index cd9906b0ac370f6f068a001a5211330f43c2ea0b..80731ac081f03ee950a4aca5e6f32151a977c4ed 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -98,7 +98,7 @@ AudioRtp::closeRtpSession () {
throw;
}
AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- audiolayer->stopStream();
+ // audiolayer->stopStream();
return true;
}
@@ -398,52 +398,6 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
int compSize = processDataEncode(audiolayer);
- /*
- // compute codec framesize in ms
- float fixed_codec_framesize = computeCodecFrameSize(_audiocodec->getFrameSize(), _audiocodec->getClockRate());
-
- // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
- int maxBytesToGet = computeNbByteAudioLayer(fixed_codec_framesize);
-
- // available bytes inside ringbuffer
- int availBytesFromMic = audiolayer->canGetMic();
-
- // set available byte to maxByteToGet
- int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
-
- if (bytesAvail == 0)
- return;
-
- // Get bytes from micRingBuffer to data_from_mic
- int nbSample = audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
-
- // nb bytes to be sent over RTP
- int compSize = 0;
-
- // test if resampling is required
- if (_audiocodec->getClockRate() != _layerSampleRate) {
-
- int nb_sample_up = nbSample;
- // _debug("_nbSample audiolayer->getMic(): %i \n", nbSample);
-
- // Store the length of the mic buffer in samples for recording
- _nSamplesMic = nbSample;
-
-
- // int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000;
- nbSample = reSampleData(micData , micDataConverted, _audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
-
- compSize = _audiocodec->codecEncode( micDataEncoded, micDataConverted, nbSample*sizeof(int16));
-
- } else {
- // no resampling required
-
- // int nbSamplesMax = _codecFrameSize;
- compSize = _audiocodec->codecEncode( micDataEncoded, micData, nbSample*sizeof(int16));
-
- }
- */
-
// putData put the data on RTP queue, sendImmediate bypass this queue
if (!_sym) {
// _sessionSend->putData(timestamp, micDataEncoded, compSize);
@@ -483,54 +437,10 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
unsigned int size = adu->getSize(); // size in char
processDataDecode(audiolayer, spkrData, size, countTime);
- /*
- if (_audiocodec != NULL) {
-
- // Return the size of data in bytes
- int expandedSize = _audiocodec->codecDecode( spkrDataDecoded , spkrData , size );
-
- // buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
- int nbSample = expandedSize / sizeof(SFLDataFormat);
-
- // test if resampling is required
- if (_audiocodec->getClockRate() != _layerSampleRate) {
-
- // Do sample rate conversion
- int nb_sample_down = nbSample;
- nbSample = reSampleData(spkrDataDecoded , spkrDataConverted, _codecSampleRate , nb_sample_down, UP_SAMPLING);
-
- // Store the number of samples for recording
- _nSamplesSpkr = nbSample;
-
- // put data in audio layer, size in byte
- audiolayer->putMain (spkrDataConverted, nbSample * sizeof(SFLDataFormat));
-
- } else {
-
- // Stor the number of samples for recording
- _nSamplesSpkr = nbSample;
-
- // put data in audio layer, size in byte
- audiolayer->putMain (spkrDataDecoded, nbSample * sizeof(SFLDataFormat));
- }
-
- // Notify (with a beep) an incoming call when there is already a call
- countTime += time->getSecond();
- if (Manager::instance().incomingCallWaiting() > 0) {
- countTime = countTime % 500; // more often...
- if (countTime == 0) {
- Manager::instance().notificationIncomingCall();
- }
- }
-
- } else {
- countTime += time->getSecond();
- }
- delete adu; adu = NULL;
- */
-
+
}
+
int
AudioRtpRTX::reSampleData(SFLDataFormat *input, SFLDataFormat *output, int sampleRate_codec, int nbSamples, int status)
{
@@ -634,7 +544,7 @@ AudioRtpRTX::run () {
}
- audiolayer->stopStream();
+ // audiolayer->stopStream();
_debug("- ARTP Action: Stop call %s\n",_ca->getCallId().c_str());
//} catch(std::exception &e) {
//_start.post();
diff --git a/sflphone-common/src/audio/audiostream.cpp b/sflphone-common/src/audio/audiostream.cpp
index 08587fdb310803fe561ae208c025869f6b7ab86b..abfb5ab34f92f598d327899751a6f7ed609d88e7 100644
--- a/sflphone-common/src/audio/audiostream.cpp
+++ b/sflphone-common/src/audio/audiostream.cpp
@@ -49,10 +49,6 @@ AudioStream::connectStream()
if(!_audiostream)
_audiostream = createStream( _context );
- else {
- disconnectStream();
- _audiostream = createStream( _context );
- }
return true;
}
diff --git a/sflphone-common/src/audio/codecs/Makefile.am b/sflphone-common/src/audio/codecs/Makefile.am
index 028a178cfd0342945ada0fcd9a169923e6c3f621..10a005f1c85bcd86a54a1997627352cd30d07cbd 100644
--- a/sflphone-common/src/audio/codecs/Makefile.am
+++ b/sflphone-common/src/audio/codecs/Makefile.am
@@ -31,6 +31,14 @@ libcodec_speex_wb_so_LDFLAGS = --shared -lc -lspeex $(SPEEX_NIMP)
INSTALL_SPEEX_WB_RULE = install-libcodec_speex_wb_so
endif
+if BUILD_SPEEX
+SPEEX_UB_LIB = libcodec_speex_ub.so
+libcodec_speex_ub_so_SOURCES = speexcodec_ub.cpp
+libcodec_speex_ub_so_CFLAGS = -fPIC -g -Wall
+libcodec_speex_ub_so_CXXFLAGS = -fPIC -g -Wall $(SPEEXDSP)
+libcodec_speex_ub_so_LDFLAGS = --shared -lc -lspeex $(SPEEX_NIMP)
+INSTALL_SPEEX_UB_RULE = install-libcodec_speex_ub_so
+endif
if BUILD_CELT
CELT_LIB = libcodec_celt.so
@@ -52,7 +60,7 @@ INSTALL_ILBC_RULE = install-libcodec_ilbc_so
SUBDIRS = ilbc
endif
-noinst_PROGRAMS = libcodec_ulaw.so libcodec_alaw.so libcodec_g722.so $(GSM_LIB) $(SPEEX_NB_LIB) $(SPEEX_WB_LIB) $(CELT_LIB) $(ILBC_LIB)
+noinst_PROGRAMS = libcodec_ulaw.so libcodec_alaw.so libcodec_g722.so $(GSM_LIB) $(SPEEX_NB_LIB) $(SPEEX_WB_LIB) $(SPEEX_UB_LIB) $(CELT_LIB) $(ILBC_LIB)
noinst_HEADERS = audiocodec.h
@@ -72,8 +80,8 @@ libcodec_g722_so_CFLAGS = -fPIC -g -Wall
libcodec_g722_so_CXXFLAGS = -fPIC -g -Wall
libcodec_g722_so_LDFLAGS = --shared -lc
-install-exec-local: install-libcodec_ulaw_so install-libcodec_alaw_so install-libcodec_g722_so $(INSTALL_GSM_RULE) $(INSTALL_SPEEX_NB_RULE) $(INSTALL_SPEEX_WB_RULE) $(INSTALL_CELT_RULE) $(INSTALL_ILBC_RULE)
-uninstall-local: uninstall-libcodec_ulaw_so uninstall-libcodec_alaw_so uninstall-libcodec_g722_so uninstall-libcodec_gsm_so uninstall-libcodec_speex_nb_so uninstall-libcodec_speex_wb_so uninstall-libcodec_celt_so
+install-exec-local: install-libcodec_ulaw_so install-libcodec_alaw_so install-libcodec_g722_so $(INSTALL_GSM_RULE) $(INSTALL_SPEEX_NB_RULE) $(INSTALL_SPEEX_WB_RULE) $(INSTALL_SPEEX_UB_RULE) $(INSTALL_CELT_RULE) $(INSTALL_ILBC_RULE)
+uninstall-local: uninstall-libcodec_ulaw_so uninstall-libcodec_alaw_so uninstall-libcodec_g722_so uninstall-libcodec_gsm_so uninstall-libcodec_speex_nb_so uninstall-libcodec_speex_wb_so uninstall-libcodec_speex_ub_so uninstall-libcodec_celt_so
install-libcodec_ulaw_so: libcodec_ulaw.so
@@ -89,6 +97,8 @@ install-libcodec_speex_nb_so: libcodec_speex_nb.so
$(INSTALL_PROGRAM) libcodec_speex_nb.so $(sflcodecdir)
install-libcodec_speex_wb_so: libcodec_speex_wb.so
$(INSTALL_PROGRAM) libcodec_speex_wb.so $(sflcodecdir)
+install-libcodec_speex_ub_so: libcodec_speex_ub.so
+ $(INSTALL_PROGRAM) libcodec_speex_ub.so $(sflcodecdir)
install-libcodec_celt_so: libcodec_celt.so
$(INSTALL_PROGRAM) libcodec_celt.so $(sflcodecdir)
install-libcodec_ilbc_so: libcodec_ilbc.so
@@ -107,6 +117,8 @@ uninstall-libcodec_speex_nb_so:
rm -f $(sflcodecdir)/libcodec_speex_nb.so
uninstall-libcodec_speex_wb_so:
rm -f $(sflcodecdir)/libcodec_speex_wb.so
+uninstall-libcodec_speex_ub_so:
+ rm -f $(sflcodecdir)/libcodec_speex_ub.so
rm -rf $(sflcodecdir)
uninstall-libcodec_celt_so: libcodec_celt.so
$(INSTALL_PROGRAM) libcodec_celt.so $(sflcodecdir)
diff --git a/sflphone-common/src/audio/codecs/speexcodec_nb.cpp b/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
index 22089f0454c7f7654de0a4c3f57275c1056dfda4..96f7c63db46e2d3fe180a2c9ea51f95ffbaa9416 100644
--- a/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
+++ b/sflphone-common/src/audio/codecs/speexcodec_nb.cpp
@@ -127,9 +127,11 @@ class Speex : public AudioCodec{
speex_preprocess_run(_preprocess_state, src);
#endif
+
speex_encode_int(_speex_enc_state, src, &_speex_enc_bits);
int nbBytes = speex_bits_write(&_speex_enc_bits, (char*)dst, size);
+
return nbBytes;
}
diff --git a/sflphone-common/src/audio/codecs/speexcodec_ub.cpp b/sflphone-common/src/audio/codecs/speexcodec_ub.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..bce7e952e4ee37816e89d027d1ee9e72bcecffaf
--- /dev/null
+++ b/sflphone-common/src/audio/codecs/speexcodec_ub.cpp
@@ -0,0 +1,158 @@
+/*
+ * Copyright (C) 2007-2009 Savoir-Faire Linux inc.
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include "audiocodec.h"
+#include <cstdio>
+#include <speex/speex.h>
+#include <speex/speex_preprocess.h>
+
+class Speex : public AudioCodec{
+ public:
+ Speex(int payload=0)
+ : AudioCodec(payload, "speex"),
+ _speexModePtr(NULL),
+ _speex_dec_bits(),
+ _speex_enc_bits(),
+ _speex_dec_state(),
+ _speex_enc_state(),
+ _speex_frame_size(),
+ _preprocess_state()
+ {
+ _clockRate = 32000;
+ _frameSize = 320; // 10 ms at 32 kHz
+ _channel = 1;
+ _bitrate = 0;
+ _bandwidth = 0;
+ initSpeex();
+ }
+
+ Speex( const Speex& );
+ Speex& operator=(const Speex&);
+
+ void initSpeex() {
+
+ // int _samplingRate = 32000;
+
+ // 8000 HZ --> Narrow-band mode
+ // TODO Manage the other modes
+ _speexModePtr = &speex_wb_mode;
+ // _speexModePtr = &speex_wb_mode;
+
+ // Init the decoder struct
+ speex_bits_init(&_speex_dec_bits);
+ _speex_dec_state = speex_decoder_init(_speexModePtr);
+
+ // Init the encoder struct
+ speex_bits_init(&_speex_enc_bits);
+ _speex_enc_state = speex_encoder_init(_speexModePtr);
+
+ speex_encoder_ctl(_speex_enc_state,SPEEX_SET_SAMPLING_RATE,&_clockRate);
+
+ speex_decoder_ctl(_speex_dec_state, SPEEX_GET_FRAME_SIZE, &_speex_frame_size);
+
+#ifdef HAVE_SPEEXDSP_LIB
+
+ int enable = 1;
+ int quality = 10;
+ int complex = 10;
+ int attenuation = -10;
+
+ speex_encoder_ctl(_speex_enc_state, SPEEX_SET_VAD, &enable);
+ speex_encoder_ctl(_speex_enc_state, SPEEX_SET_DTX, &enable);
+ // speex_encoder_ctl(_speex_enc_state, SPEEX_SET_VBR_QUALITY, &quality);
+ speex_encoder_ctl(_speex_enc_state, SPEEX_SET_COMPLEXITY, &complex);
+
+ // Init the decoder struct
+ speex_decoder_ctl(_speex_dec_state, SPEEX_GET_FRAME_SIZE, &_speex_frame_size);
+
+ // Init the preprocess struct
+ _preprocess_state = speex_preprocess_state_init(_speex_frame_size,_clockRate);
+ speex_preprocess_ctl(_preprocess_state, SPEEX_PREPROCESS_SET_DENOISE, &enable);
+ speex_preprocess_ctl(_preprocess_state, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &attenuation);
+ speex_preprocess_ctl(_preprocess_state, SPEEX_PREPROCESS_SET_VAD, &enable);
+ speex_preprocess_ctl(_preprocess_state, SPEEX_PREPROCESS_SET_AGC, &enable);
+#endif
+
+ }
+
+ ~Speex()
+ {
+ terminateSpeex();
+ }
+
+ void terminateSpeex() {
+ // Destroy the decoder struct
+ speex_bits_destroy(&_speex_dec_bits);
+ speex_decoder_destroy(_speex_dec_state);
+ _speex_dec_state = 0;
+
+ // Destroy the encoder struct
+ speex_bits_destroy(&_speex_enc_bits);
+ speex_encoder_destroy(_speex_enc_state);
+ _speex_enc_state = 0;
+ }
+
+ virtual int codecDecode (short *dst, unsigned char *src, unsigned int size)
+ {
+
+ // int ratio = 320 / _speex_frame_size;
+ speex_bits_read_from(&_speex_dec_bits, (char*)src, size);
+ speex_decode_int(_speex_dec_state, &_speex_dec_bits, dst);
+
+ // return the nuber of byte, not the number of sample
+ return _speex_frame_size * 2;
+ }
+
+ virtual int codecEncode (unsigned char *dst, short *src, unsigned int size)
+ {
+ speex_bits_reset(&_speex_enc_bits);
+
+#ifdef HAVE_SPEEXDSP_LIB
+
+ speex_preprocess_run(_preprocess_state, src);
+#endif
+
+
+ speex_encode_int(_speex_enc_state, src, &_speex_enc_bits);
+ speex_bits_nbytes(&_speex_enc_bits);
+ int nbBytes = speex_bits_write(&_speex_enc_bits, (char*)dst, size);
+
+ return nbBytes;
+ }
+
+ private:
+ const SpeexMode* _speexModePtr;
+ SpeexBits _speex_dec_bits;
+ SpeexBits _speex_enc_bits;
+ void *_speex_dec_state;
+ void *_speex_enc_state;
+ int _speex_frame_size;
+ SpeexPreprocessState *_preprocess_state;
+};
+
+// the class factories
+extern "C" AudioCodec* create() {
+ return new Speex(112);
+}
+
+extern "C" void destroy(AudioCodec* a) {
+ delete a;
+}
+
diff --git a/sflphone-common/src/audio/codecs/speexcodec_wb.cpp b/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
index 81d27bbcb0eebee7d7a75687bf82c0a8e410b9b6..dc93f39ec1a063732ef64b718d5fe48abf572f70 100644
--- a/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
+++ b/sflphone-common/src/audio/codecs/speexcodec_wb.cpp
@@ -111,8 +111,9 @@ class Speex : public AudioCodec{
virtual int codecDecode (short *dst, unsigned char *src, unsigned int size)
{
-
+
int ratio = 320 / _speex_frame_size;
+
speex_bits_read_from(&_speex_dec_bits, (char*)src, size);
speex_decode_int(_speex_dec_state, &_speex_dec_bits, dst);
@@ -128,9 +129,10 @@ class Speex : public AudioCodec{
speex_preprocess_run(_preprocess_state, src);
#endif
+ printf("Codec::codecEncode() size %i\n", size);
speex_encode_int(_speex_enc_state, src, &_speex_enc_bits);
int nbBytes = speex_bits_write(&_speex_enc_bits, (char*)dst, size);
- printf("Codec::codecEncode() nbBytes: %i \n",nbBytes);
+ printf("Codec::codecEncode() nbBytes %i\n", nbBytes);
return nbBytes;
}
diff --git a/sflphone-common/src/audio/pulselayer.cpp b/sflphone-common/src/audio/pulselayer.cpp
index a5b07f277b2280da0b5fea1ad6131ae5849b73e8..c8410651b536ddddecb21dc5ce513700295430fd 100644
--- a/sflphone-common/src/audio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulselayer.cpp
@@ -166,12 +166,15 @@ bool PulseLayer::createStreams( pa_context* c )
pa_threaded_mainloop_signal(m , 0);
+ isCorked = true;
+
return true;
}
bool PulseLayer::openDevice(int indexIn UNUSED, int indexOut UNUSED, int sampleRate, int frameSize , int stream UNUSED, std::string plugin UNUSED)
{
+
_debug("PulseLayer::openDevice \n");
_sampleRate = sampleRate;
_frameSize = frameSize;
@@ -225,35 +228,46 @@ int PulseLayer::getMic(void *buffer, int toCopy)
void PulseLayer::startStream (void)
{
// flushMic();
+ if (isCorked) {
- _urgentRingBuffer.flush();
- _micRingBuffer.flush();
- _voiceRingBuffer.flush();
+ _urgentRingBuffer.flush();
+ _micRingBuffer.flush();
+ _voiceRingBuffer.flush();
- _debug("PulseLayer::Start stream\n");
- pa_threaded_mainloop_lock(m);
+ _debug("PulseLayer::Start stream\n");
+ pa_threaded_mainloop_lock(m);
- pa_stream_cork( playback->pulseStream(), 0, NULL, NULL);
- pa_stream_cork( record->pulseStream(), 0, NULL, NULL);
+ pa_stream_cork( playback->pulseStream(), 0, NULL, NULL);
+ pa_stream_cork( record->pulseStream(), 0, NULL, NULL);
- pa_threaded_mainloop_unlock(m);
+ pa_threaded_mainloop_unlock(m);
+
+ isCorked = false;
+ }
}
void
PulseLayer::stopStream (void)
{
- _debug("PulseLayer::Stop stream\n");
- pa_stream_flush( playback->pulseStream(), NULL, NULL );
- pa_stream_flush( record->pulseStream(), NULL, NULL );
+
+ if (!isCorked) {
+
+ _debug("PulseLayer::Stop stream\n");
+ pa_stream_flush( playback->pulseStream(), NULL, NULL );
+ pa_stream_flush( record->pulseStream(), NULL, NULL );
- flushMic();
- flushMain();
- flushUrgent();
+ flushMic();
+ flushMain();
+ flushUrgent();
- pa_stream_cork( playback->pulseStream(), 1, NULL, NULL);
- pa_stream_cork( record->pulseStream(), 1, NULL, NULL);
+ pa_stream_cork( playback->pulseStream(), 1, NULL, NULL);
+ pa_stream_cork( record->pulseStream(), 1, NULL, NULL);
+ isCorked = true;
+
+ }
+
}
diff --git a/sflphone-common/src/audio/pulselayer.h b/sflphone-common/src/audio/pulselayer.h
index beae9eacc2b5b7400aca9ccd6d25a79b98dcd1c8..698cc9bf6b68b6ff7009b35efcccba3c664d2729 100644
--- a/sflphone-common/src/audio/pulselayer.h
+++ b/sflphone-common/src/audio/pulselayer.h
@@ -189,6 +189,8 @@ class PulseLayer : public AudioLayer {
*/
AudioStream* record;
+ bool isCorked;
+
int spkrVolume;
int micVolume;
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index d4d67ad216c9c8164997206ec125a0bee50c00c1..5e01cd2d71683df845e1296c5818473a1a503f65 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -143,7 +143,7 @@ ManagerImpl::init()
}
if (audiolayer == 0)
- audiolayer->stopStream();
+ audiolayer->stopStream();
// Load the history
@@ -326,7 +326,14 @@ ManagerImpl::hangupCall(const CallID& id)
_debug("Stop audio stream\n");
audiolayer = getAudioDriver();
- audiolayer->stopStream();
+
+ int nbCalls = getCallList().size();
+
+ _debug("nbCalls %i \n", nbCalls);
+
+ // stop stream
+ if (!(nbCalls > 1))
+ audiolayer->stopStream();
/* Direct IP to IP call */
if (getConfigFromCall (id) == Call::IPtoIP) {
@@ -952,11 +959,12 @@ void ManagerImpl::stopTone (bool stopAudio=true)
hasToPlayTone = getConfigInt(SIGNALISATION, PLAY_TONES);
if (!hasToPlayTone)
return;
-
+
if (stopAudio) {
audiolayer = getAudioDriver();
if (audiolayer) audiolayer->stopStream();
}
+
_toneMutex.enterMutex();
if (_telephoneTone != 0) {