diff --git a/debian/Release b/debian/Release
index 7fb4d2cd9ab47a7db36db62a5fbbdea976a4655d..5bbfec2cc9ae328804760f3a3782a036edea9761 100644
--- a/debian/Release
+++ b/debian/Release
@@ -1,7 +1,7 @@
Archive : unstable
-Version: 0.9
+Version: 0.9.1
Component : universe
Origin : SFLphone
Label : sflphone
-Architecture : all
+Architecture : i386
diff --git a/debian/arch b/debian/arch
deleted file mode 100644
index 5a9a476a8b7b3a362fa7925b74bda4a771cba262..0000000000000000000000000000000000000000
--- a/debian/arch
+++ /dev/null
@@ -1 +0,0 @@
-i386
diff --git a/debian/autopackage.sh b/debian/autopackage.sh
index 4d6248ca2dd49d3256469f817de477754a8b04d8..b3cb825990fc7ea91066b8cab1ec5d60e7a745cb 100755
--- a/debian/autopackage.sh
+++ b/debian/autopackage.sh
@@ -42,7 +42,7 @@ cp $bindir/sflphoned $sfldir$bindir
strip $sfldir$bindir/sflphoned
cp $bindir/sflphone-gtk $sfldir$bindir
strip $sfldir$bindir/sflphone-gtk
-ln -sf $sfldir$bindir/sflphone-gtk $sfldir$bindir/sflphone
+ln -sf $bindir/sflphone-gtk $sfldir$bindir/sflphone
#/usr/lib
mkdir -p $sfldir/usr/lib/sflphone/codecs
@@ -76,8 +76,10 @@ cp changelog.Debian.gz $sfldir$sharedir/doc/sflphone
cp copyright $sfldir$sharedir/doc/sflphone
cp TODO $sfldir$sharedir/doc/sflphone
-# DEBIAN files
mkdir -p $debdir
+cp debian-binary $sfldir
+cp postrm $debdir
+# DEBIAN files
# Create control file
control="$debdir/control"
touch $control
@@ -99,4 +101,4 @@ fakeroot dpkg --build $sfldir ${sfldir}_$2.deb
# Clean up the generated stuff
echo "Clean up ... "
-#rm -rf $sfldir
+rm -rf $sfldir
diff --git a/debian/changelog b/debian/changelog
deleted file mode 100644
index b2b6d3bc6cbd494a5becc148f8774450befdb577..0000000000000000000000000000000000000000
--- a/debian/changelog
+++ /dev/null
@@ -1,70 +0,0 @@
-sflphone (0.9.1) unstable; urgency=low
- * Add a search tool in the history
- * Migrate some gtk_entry_new to sexy_icon_entry_new
- * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
- the history tab
- * Add the SIP registration expire value in the user file.
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
-
-sflphone (0.9.0) unstable; urgency=low
- * Add history features
- * Call date
- * Call duration
- * Mouse events in the history tab
- * Smooth switch from the history tab to the calls tab
- * Remove most of GTK-Critical warnings
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-06-06) unstable; urgency=low
- * Audio bug correction: capture stopped after a few minutes of conversation
- with USB Plantronics sound card
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
-
-sflphone (0.9-2008-05-06) unstable; urgency=low
- * Bug correction: account creation with the assistant
- * GTK+ warnings removal
- * libnotify warnings removal
- * Remove aliasing on the SFLphone logo
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
-
-sflphone (0.9) unstable; urgency=low
- * Clean dependencies ( removal of libboost )
- * Several GTK improvement and updates
- -account window
- -configuration window
- * Migrate from GtkCheckMenuItem to GtkImageMenuItem
- * ALSA standard I/O transfers: MMAP instead of R/W
- * Fix speex audio quality
- * IAX2 protocol
- -Fix hold/unhold situation
- -Add on hold music
- * SIP protocol
- -Ringtone on incoming call
- -Fix transfer situation
- * Add desktop notification ( libnotify )
- * Improve the system tray icon behaviour
- * Improve registration error handling
- * Register/unregister from the account window takes effect without starting back SFLphone
- * Compilation warnings removal
- * Call history
- * Add an account configuration wizard
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
-
-
-
-sflphone (0.8.2 ) unstable; urgency=low
-
- * Internationalization of the GTK GUI
- * English / French
- * STUN support
- * Slight modifications of the graphical interface ( tooltips, dialpad, ...)
-
-
- -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
-
-
diff --git a/debian/control b/debian/control
index aa80a7f5a06dafd35c55efe11a5bb68577f51864..bf0d3fd666e70203a245c34ab00b4b6b6f0ee61b 100644
--- a/debian/control
+++ b/debian/control
@@ -2,18 +2,18 @@ Source: sflphone
Maintainer: SavoirFaireLinux Inc <emmanuel.milou@savoirfairelinux.com>
Section: gnome
Priority: optional
-Standards-Version: 0.9
+Build-Depends: debhelper (>> 3.0.0)
+Standards-Version: 0.9.1
Package: sflphone
-Architecture: i386
Section: gnome
Priority: optional
+Architecture: i386
Essential: no
-Depends: libgcc1 , libsamplerate0 (>=0.1.2) , libdbus-glib-1-2 (>= 0.73), libexpat1 , libgtk2.0-0 , gnome-common , libc6 (>= 2.3.6-6) , libglib2.0-0 (>= 2.12.0) , libosip2-3, libexosip2-5, libcommoncpp2-1.5.3-0 , libccrtp1-1.5-1 , libiax0 , libgtkglext1
+Depends: libgcc1 , libsamplerate0 (>=0.1.2) , libdbus-glib-1-2 (>= 0.73), libexpat1 , libgtk2.0-0 , libc6 (>= 2.3.6-6) , libglib2.0-0 (>= 2.12.0) , libosip2-2, libexosip2-4, libcommoncpp2-1.6-0 , libccrtp1-1.6-0 , sflphone-iax2 , libgsm1 (>=1.0.10) , libspeex1 (>=1.1.12) , dbus-c++-1 (>=0.5.0) , libsexy2 (>=0.1.11)
Homepage: http://www.sflphone.org
-Description: SFLphone - Answer the call
- SFLphone is meant to be a robust enterprise-class desktop phone. It is design with a hundred-calls-a-day receptionist in mind. It can work for you, too.
- .
+Description: SIP and IAX2 compatible softphone
+ SFLphone is meant to be a robust enterprise-class desktop phone.
SFLphone is released under the GNU General Public License.
- .
- SFLphone is being developed by the global community, and maintained by Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
+ SFLphone is being developed by the global community, and maintained by
+ Savoir-faire Linux, a Montreal, Quebec, Canada-based Linux consulting company.
diff --git a/debian/debian-binary b/debian/debian-binary
new file mode 100644
index 0000000000000000000000000000000000000000..cd5ac039d67e0bdadb17976e4ac39f0ffe6bb6e4
--- /dev/null
+++ b/debian/debian-binary
@@ -0,0 +1 @@
+2.0
diff --git a/debian/postrm b/debian/postrm
new file mode 100755
index 0000000000000000000000000000000000000000..56801250786e23c392e1598602ed8593b4c64ff5
--- /dev/null
+++ b/debian/postrm
@@ -0,0 +1,15 @@
+#!/bin/sh
+
+# debian/postrm postremoval script for SFLphone
+
+if [ "$1" = "purge" ]
+then
+
+ # remove the user config file
+ rm -f $HOME/.sflphone/sflphonedrc
+
+fi
+
+exit 0
+
+## -------------------------------------------------
diff --git a/debian/rules b/debian/rules
index efde01dd03e95702fef9e90f0c6e0c0cfd5ed9bf..110e88d2256c17cad8b2f0c36a9cddc8b2c05a54 100755
--- a/debian/rules
+++ b/debian/rules
@@ -35,10 +35,12 @@ endif
config.status: configure
dh_testdir
# Add here commands to configure the package.
- autoreconf --install
- CFLAGS="$(CFLAGS)" CXX="$(CXX)" ./configure --prefix=/usr --with-debug
+#autoreconf --install
+#CFLAGS="$(CFLAGS)" CXX="$(CXX)" ./configure --prefix=/usr --with-debu
+ ./configure --prefix=/
+#gnome-autogen.sh --prefix=/usr --with-debug
cd sflphone-gtk
- gnome-autogen.sh --prefix=/usr
+ ./configure --prefix=/
#Architecture
@@ -48,7 +50,8 @@ build-arch: build-arch-stamp
build-arch-stamp: config.status
# Add here commands to compile the arch part of the package.
- #$(MAKE)
+ $(MAKE) && (cd sflphone-gtk)
+ $(MAKE)
touch build-arch-stamp
build-indep: build-indep-stamp
@@ -96,7 +99,7 @@ install-arch:
# Add here commands to install the arch part of the package into
# debian/tmp.
- $(MAKE) install DESTDIR=$(CURDIR)/debian/sflphoned
+ $(MAKE) install DESTDIR=$(CURDIR)/debian/sflphone/usr
dh_install -s
# Must not depend on anything. This is to be called by
diff --git a/debian/sflphone.install b/debian/sflphone.install
new file mode 100644
index 0000000000000000000000000000000000000000..7e7915dafd83894ad61e23a94368b0baca82fa00
--- /dev/null
+++ b/debian/sflphone.install
@@ -0,0 +1,29 @@
+usr/bin
+usr/bin/sflphoned
+usr/bin/sflphone-gtk
+usr/bin/sflphone
+
+#/usr/lib
+usr/lib/sflphone/codecs
+usr/lib/sflphone/codecs/libcodec_*
+
+#/usr/share/applications
+usr/share/applications
+usr/share/applications/sflphone.desktop
+usr/share/dbus-1/services
+usr/share/dbus-1/services/org.sflphone.SFLphone.service
+usr/share/pixmaps
+usr/share/pixmaps/sflphone.png
+
+usr/share/sflphone/*
+usr/share/sflphone/ringtones/*
+
+usr/share/locale/fr/LC_MESSAGES
+usr/share/locale/fr/LC_MESSAGES/sflphone.mo
+usr/share/locale/es/LC_MESSAGES
+usr/share/locale/es/LC_MESSAGES/sflphone.mo
+
+changelog.Debian.gz
+copyright
+TODO
+
diff --git a/sflphone-gtk/src/assistant.c b/sflphone-gtk/src/assistant.c
index e50d043c89c298d21a00b3480fc64f189c17d279..27c066cc6e4c8fd888af2ea69fd8a8a6a8c22be3 100644
--- a/sflphone-gtk/src/assistant.c
+++ b/sflphone-gtk/src/assistant.c
@@ -41,11 +41,13 @@ set_account_type( GtkWidget* widget , gpointer data )
static void close_callback( void )
{
gtk_widget_destroy(wiz->assistant);
+ g_free(wiz); wiz = NULL;
}
static void cancel_callback( void )
{
gtk_widget_destroy(wiz->assistant);
+ g_free(wiz); wiz = NULL;
}
static void
@@ -98,6 +100,7 @@ enable_stun( GtkWidget* widget )
void
build_wizard( void )
{
+if(!wiz){
wiz = ( struct _wizard* )g_malloc( sizeof( struct _wizard));
current = g_new0(account_t, 1);
current->properties = g_hash_table_new(NULL, g_str_equal);
@@ -116,14 +119,14 @@ build_wizard( void )
build_iax_account_configuration();
build_registration_error();
build_summary();
-
+
g_signal_connect(G_OBJECT(wiz->assistant), "close" , G_CALLBACK(close_callback), NULL);
g_signal_connect(G_OBJECT(wiz->assistant), "cancel" , G_CALLBACK(cancel_callback), NULL);
gtk_widget_show_all(wiz->assistant);
gtk_assistant_update_buttons_state(GTK_ASSISTANT(wiz->assistant));
-
+ }
}
GtkWidget*
diff --git a/src/Makefile.am b/src/Makefile.am
index d5e84d57d0c0e921bbc8ca3b315efd894203168f..5915d0166af82089b66206bcbe3e8750b86c9cda 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -31,10 +31,10 @@ endif
ALSAFLAG= -lasound
PULSEAUDIO_LIBS=-lpulse
-SUBDIRS = audio config dbus $(ZEROCONFDIR)# contact memmanager video mixer
+SUBDIRS = audio config dbus $(ZEROCONFDIR) #contact memmanager video mixer
sflphoned_SOURCES = eventthread.cpp main.cpp voiplink.cpp \
- managerimpl.cpp observer.cpp \
+ managerimpl.cpp observer.cpp samplerateconverter.cpp \
account.cpp sipaccount.cpp accountcreator.cpp \
sipvoiplink.cpp call.cpp sipcall.cpp \
$(IAXSOURCES)
@@ -61,7 +61,7 @@ libsflphone_la_SOURCES =
noinst_LTLIBRARIES = libsflphone.la
noinst_HEADERS = managerimpl.h manager.h global.h observer.h eventthread.h user_cfg.h \
- voiplink.h \
+ voiplink.h samplerateconverter.h \
account.h sipaccount.h accountcreator.h \
sipvoiplink.h call.h sipcall.h \
$(IAXHEADERS)
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index cb30737f116a1af7dc9242722248b747137f630c..641eb4f841a5ebfc05ea3fb7c06828ca6a36d4fa 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -38,7 +38,6 @@
#include "ringbuffer.h"
#include "../user_cfg.h"
#include "../sipcall.h"
-#include <samplerate.h>
////////////////////////////////////////////////////////////////////////////////
// AudioRtp
@@ -102,9 +101,6 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
_ca = sipcall;
_sym = sym;
// AudioRtpRTX should be close if we change sample rate
- //int IDcodec= _ca->getAudioCodec();
- //_codecSampleRate = _ca->getAudioCodec()->getClockRate();
- //_codecDesc = Manager::instance().getCodecDescriptorMap();
// TODO: Change bind address according to user settings.
// TODO: this should be the local ip not the external (router) IP
std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
@@ -118,13 +114,6 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
_sessionRecv = NULL;
_sessionSend = NULL;
}
-
- // libsamplerate-related
- // Set the converter type for the upsampling and the downsampling
- // interpolator SRC_SINC_BEST_QUALITY
- _src_state_mic = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
- _src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
-
}
AudioRtpRTX::~AudioRtpRTX () {
@@ -145,39 +134,34 @@ AudioRtpRTX::~AudioRtpRTX () {
delete _session; _session = NULL;
}
- delete [] _intBufferDown; _intBufferDown = NULL;
- delete [] _floatBufferUp; _floatBufferUp = NULL;
- delete [] _floatBufferDown; _floatBufferDown = NULL;
- delete [] _dataAudioLayer; _dataAudioLayer = NULL;
+ delete [] micData; micData = NULL;
+ delete [] micDataConverted; micDataConverted = NULL;
+ delete [] micDataEncoded; micDataEncoded = NULL;
- delete [] _sendDataEncoded; _sendDataEncoded = NULL;
- delete [] _receiveDataDecoded; _receiveDataDecoded = NULL;
+ delete [] spkrDataDecoded; spkrDataDecoded = NULL;
+ delete [] spkrDataConverted; spkrDataConverted = NULL;
delete time; time = NULL;
-
- // libsamplerate-related
- _src_state_mic = src_delete(_src_state_mic);
- _src_state_spkr = src_delete(_src_state_spkr);
}
void
AudioRtpRTX::initBuffers()
{
+ converter = new SamplerateConverter( _layerSampleRate , _layerFrameSize );
+
int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
- _dataAudioLayer = new SFLDataFormat[nbSamplesMax];
- _receiveDataDecoded = new int16[nbSamplesMax];
- _floatBufferDown = new float32[nbSamplesMax];
- _floatBufferUp = new float32[nbSamplesMax];
- _sendDataEncoded = new unsigned char[nbSamplesMax];
- _intBufferDown = new int16[nbSamplesMax];
+
+ micData = new SFLDataFormat[nbSamplesMax];
+ micDataConverted = new SFLDataFormat[nbSamplesMax];
+ micDataEncoded = new unsigned char[nbSamplesMax];
+
+ spkrDataConverted = new SFLDataFormat[nbSamplesMax];
+ spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
}
void
AudioRtpRTX::initAudioRtpSession (void)
{
-
-
-
try {
if (_ca == 0) { return; }
_audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( _ca->getAudioCodec() );
@@ -254,7 +238,6 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
// 3. encode it
// 4. send it
try {
- int16* toSIP = NULL;
timestamp += time->getSecond();
if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; } // no call, so we do nothing
@@ -267,50 +250,38 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
int maxBytesToGet = _layerSampleRate * _layerFrameSize * sizeof(SFLDataFormat) / 1000;
// available bytes inside ringbuffer
int availBytesFromMic = audiolayer->canGetMic();
- //printf("availBytesFromMic = %i \n", availBytesFromMic);
// take the lowest
int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
- //printf("max bytes to get: %i\n", bytesAvail);
// Get bytes from micRingBuffer to data_from_mic
- int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
+ //_debug("get data from mic\n");
+ int nbSample = audiolayer->getMic( micData , bytesAvail ) / sizeof(SFLDataFormat);
int nb_sample_up = nbSample;
int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000;
+ //_debug("resample data\n");
nbSample = reSampleData(_audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
- toSIP = _intBufferDown;
-
if ( nbSample < nbSamplesMax - 10 ) { // if only 10 is missing, it's ok
// fill end with 0...
- //_debug("begin: %p, nbSample: %d\n", toSIP, nbSample);
- memset(toSIP + nbSample, 0, (nbSamplesMax-nbSample)*sizeof(int16));
+ memset( micDataConverted + nbSample, 0, (nbSamplesMax-nbSample)*sizeof(int16));
nbSample = nbSamplesMax;
}
- // debug - dump sound in a file
- //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toSIP[0], toSIP[1], toSIP[2]);
- // for the mono: range = 0 to RTP_FRAME2SEND * sizeof(int16)
- // codecEncode(char *dest, int16* src, size in bytes of the src)
- int compSize = _audiocodec->codecEncode(_sendDataEncoded, toSIP, nbSample*sizeof(int16));
- //printf("jusqu'ici tout vas bien\n");
-
+ int compSize = _audiocodec->codecEncode( micDataEncoded , micDataConverted , nbSample*sizeof(int16));
// encode divise by two
// Send encoded audio sample over the network
if (compSize > nbSamplesMax) { _debug("! ARTP: %d should be %d\n", compSize, nbSamplesMax);}
if (!_sym) {
- _sessionSend->putData(timestamp, _sendDataEncoded, compSize);
+ _sessionSend->putData(timestamp, micDataEncoded, compSize);
} else {
- _session->putData(timestamp, _sendDataEncoded, compSize);
+ _session->putData(timestamp, micDataEncoded, compSize);
}
- toSIP = NULL;
} catch(...) {
_debugException("! ARTP: sending failed");
throw;
}
}
-
-
void
AudioRtpRTX::receiveSessionForSpkr (int& countTime)
{
@@ -334,54 +305,44 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
return;
}
- int payload = adu->getType(); // codec type
- unsigned char* data = (unsigned char*)adu->getData(); // data in char
+ //int payload = adu->getType(); // codec type
+ unsigned char* spkrData = (unsigned char*)adu->getData(); // data in char
unsigned int size = adu->getSize(); // size in char
-
- //_fstream.write((char*) data, size);
// Decode data with relevant codec
- int max = (int)(_codecSampleRate * _layerFrameSize / 1000);
+ unsigned int max = (unsigned int)(_codecSampleRate * _layerFrameSize / 1000);
if ( size > max ) {
- _debug("We have received from RTP a packet larger than expected: %s VS %s\n", size, max);
+ _debug("We have received from RTP a packet larger than expected: %d VS %d\n", size, max);
_debug("The packet size has been cropped\n");
size=max;
}
- //printf("size = %i\n", size);
-
if (_audiocodec != NULL) {
- int expandedSize = _audiocodec->codecDecode(_receiveDataDecoded, data, size);
- // printf("%i\n", expandedSize);
- //_fstream.write((char*) _receiveDataDecoded, );
+ int expandedSize = _audiocodec->codecDecode( spkrDataDecoded , spkrData , size );
//buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
int nbInt16 = expandedSize / sizeof(int16);
//nbInt16 represents the number of samples we just decoded
if (nbInt16 > max) {
- _debug("We have decoded an RTP packet larger than expected: %s VS %s. Cropping.\n", nbInt16, max);
+ _debug("We have decoded an RTP packet larger than expected: %d VS %d. Cropping.\n", nbInt16, max);
nbInt16=max;
}
-
- SFLDataFormat* toAudioLayer;
int nbSample = nbInt16;
// Do sample rate conversion
int nb_sample_down = nbSample;
nbSample = reSampleData(_codecSampleRate , nb_sample_down, UP_SAMPLING);
#ifdef DATAFORMAT_IS_FLOAT
- toAudioLayer = _floatBufferUp;
#else
- toAudioLayer = _dataAudioLayer;
#endif
int layer = audiolayer->getLayerType();
//_debug(" interface %i - ALSA = %i\n" , layer, ALSA);
if( CHECK_INTERFACE( layer, ALSA ) )
- audiolayer->playSamples(toAudioLayer, nbSample * sizeof(SFLDataFormat), true);
+ audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
else
- audiolayer->putMain( toAudioLayer, nbSample * sizeof(SFLDataFormat) );
+ audiolayer->putMain( spkrDataConverted, nbSample * sizeof(SFLDataFormat) );
// Notify (with a beep) an incoming call when there is already a call
countTime += time->getSecond();
if (Manager::instance().incomingCallWaiting() > 0) {
@@ -394,7 +355,6 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
} else {
countTime += time->getSecond();
}
-
delete adu; adu = NULL;
} catch(...) {
_debugException("! ARTP: receiving failed");
@@ -407,72 +367,21 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
int
AudioRtpRTX::reSampleData(int sampleRate_codec, int nbSamples, int status)
{
- if(status==UP_SAMPLING)
- return upSampleData(sampleRate_codec, nbSamples);
- else if(status==DOWN_SAMPLING)
- return downSampleData(sampleRate_codec, nbSamples);
- else
- return 0;
-}
-
-////////////////////////////////////////////////////////////////////
-//////////// RESAMPLING FUNCTIONS /////////////////////////////////
-//////////////////////////////////////////////////////////////////
-
- int
-AudioRtpRTX::upSampleData(int sampleRate_codec, int nbSamples)
-{
- double upsampleFactor = (double) _layerSampleRate / sampleRate_codec;
- int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
- if( upsampleFactor != 1 )
- {
- SRC_DATA src_data;
- src_data.data_in = _floatBufferDown;
- src_data.data_out = _floatBufferUp;
- src_data.input_frames = nbSamples;
- src_data.output_frames = (int) floor(upsampleFactor * nbSamples);
- src_data.src_ratio = upsampleFactor;
- src_data.end_of_input = 0; // More data will come
- src_short_to_float_array(_receiveDataDecoded, _floatBufferDown, nbSamples);
- src_process(_src_state_spkr, &src_data);
- nbSamples = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
- src_float_to_short_array(_floatBufferUp, _dataAudioLayer, nbSamples);
+ if(status==UP_SAMPLING){
+ return converter->upsampleData( spkrDataDecoded , spkrDataConverted , sampleRate_codec , _layerSampleRate , nbSamples );
}
-
- return nbSamples;
-}
-
- int
-AudioRtpRTX::downSampleData(int sampleRate_codec, int nbSamples)
-{
- double downsampleFactor = (double) sampleRate_codec / _layerSampleRate;
- int nbSamplesMax = (int) (sampleRate_codec * _layerFrameSize / 1000);
- if ( downsampleFactor != 1)
- {
- SRC_DATA src_data;
- src_data.data_in = _floatBufferUp;
- src_data.data_out = _floatBufferDown;
- src_data.input_frames = nbSamples;
- src_data.output_frames = (int) floor(downsampleFactor * nbSamples);
- src_data.src_ratio = downsampleFactor;
- src_data.end_of_input = 0; // More data will come
- src_short_to_float_array(_dataAudioLayer, _floatBufferUp, nbSamples);
- src_process(_src_state_mic, &src_data);
- nbSamples = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
- src_float_to_short_array(_floatBufferDown, _intBufferDown, nbSamples);
+ else if(status==DOWN_SAMPLING){
+ return converter->downsampleData( micData , micDataConverted , sampleRate_codec , _layerSampleRate , nbSamples );
}
- return nbSamples;
-
+ else
+ return 0;
}
-//////////////////////// END RESAMPLING //////////////////////////////////////////////////////
-
void
AudioRtpRTX::run () {
//mic, we receive from soundcard in stereo, and we send encoded
//encoding before sending
AudioLayer *audiolayer = Manager::instance().getAudioDriver();
- //loadCodec(_ca->getAudioCodec());
_layerFrameSize = audiolayer->getFrameSize(); // en ms
_layerSampleRate = audiolayer->getSampleRate();
initBuffers();
@@ -496,7 +405,6 @@ AudioRtpRTX::run () {
int countTime = 0; // for receive
TimerPort::setTimer(_layerFrameSize);
- //audiolayer->flushMic();
audiolayer->startStream();
_start.post();
_debug("- ARTP Action: Start\n");
@@ -514,8 +422,6 @@ AudioRtpRTX::run () {
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
}
- //_fstream.close();
- //unloadCodec();
//_debug("stop stream for audiortp loop\n");
audiolayer->stopStream();
} catch(std::exception &e) {
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 0b735ac890414ed4a27b7556432d18111b0779d1..a0cb003bf35566a6251361edf0b0a3b5ef0a5ce5 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -28,8 +28,8 @@
#include <ccrtp/rtp.h>
#include <cc++/numbers.h>
-#include <samplerate.h>
#include "../global.h"
+#include "../samplerateconverter.h"
#define UP_SAMPLING 0
#define DOWN_SAMPLING 1
@@ -83,33 +83,17 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
/** Is the session symmetric or not */
bool _sym;
- /** When we receive data, we decode it inside this buffer */
- int16* _receiveDataDecoded;
+ /** Mic-data related buffers */
+ SFLDataFormat* micData;
+ SFLDataFormat* micDataConverted;
+ unsigned char* micDataEncoded;
- /** Buffers used for send data from the mic */
- unsigned char* _sendDataEncoded;
-
- /** Downsampled int16 buffer */
- int16* _intBufferDown;
-
- /** After that we send the data inside this buffer if there is a format conversion or rate conversion */
- /** Also use for getting mic-ringbuffer data */
- SFLDataFormat* _dataAudioLayer;
-
- /** Downsampled float buffer */
- float32* _floatBufferDown;
-
- /** Upsampled float buffer */
- float32* _floatBufferUp;
-
- /** libsamplerate converter for incoming voice */
- SRC_STATE* _src_state_spkr;
+ /** Speaker-data related buffers */
+ SFLDataFormat* spkrDataDecoded;
+ SFLDataFormat* spkrDataConverted;
- /** libsamplerate converter for outgoing voice */
- SRC_STATE* _src_state_mic;
-
- /** libsamplerate error */
- int _src_err;
+ /** Sample rate converter object */
+ SamplerateConverter* converter;
/** Variables to process audio stream: sample rate for playing sound (typically 44100HZ) */
int _layerSampleRate;
@@ -154,22 +138,6 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
*/
int reSampleData(int sampleRate_codec, int nbSamples, int status);
- /**
- * Upsample the data from the clock rate of the codec to the sample rate of the layer
- * @param sampleRate_codec The sample rate of the codec selected to encode/decode the data
- * @param nbSamples Number of samples to process
- * @return int The number of samples after process
- */
- int upSampleData(int sampleRate_codec, int nbSamples);
-
- /**
- * Downsample the data from the sample rate of the layer to the clock rate of the codec
- * @param sampleRate_codec The sample rate of the codec selected to encode/decode the data
- * @param nbSamples Number of samples to process
- * @return int The number of samples after process
- */
- int downSampleData(int sampleRate_codec, int nbSamples);
-
/** The audio codec used during the session */
AudioCodec* _audiocodec;
};
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 1cc5a5d26404bfddf3848f6279e82961acc1e4c0..6b76c9c123dc6bcdfae3761b2cdb33facedd8b45 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -25,12 +25,10 @@
#include "manager.h"
#include "audio/audiolayer.h"
-#include <samplerate.h>
//#include <iax/iax-client.h>
#include <math.h>
#include <dlfcn.h>
-
#define IAX_BLOCKING 1
#define IAX_NONBLOCKING 0
@@ -40,20 +38,12 @@
#define RANDOM_IAX_PORT rand() % 64000 + 1024
#define MUSIC_ONHOLD true
-#define NO_MUSIC_ONHOLD false
-
-// from IAXC : iaxclient.h
-
-#define IAX__20S_8KHZ_MAX 320 //320 samples, IAX packets can have more than 20ms.
-#define IAX__20S_48KHZ_MAX 1920 // 320*6 samples = 1920, 6 = 48000/8000
#define CHK_VALID_CALL if (call == NULL) { _debug("IAX: Call doesn't exists\n"); \
- return false; }
-
+ return false; }
-
-IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
- : VoIPLink(accountID)
+ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
+: VoIPLink(accountID)
{
_evThread = new EventThread(this);
_regSession = NULL;
@@ -64,19 +54,16 @@ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
audiolayer = NULL;
- _receiveDataDecoded = new int16[IAX__20S_48KHZ_MAX];
- _sendDataEncoded = new unsigned char[IAX__20S_8KHZ_MAX];
+ converter = new SamplerateConverter();
- // we estimate that the number of format after a conversion 8000->48000 is expanded to 6 times
- _dataAudioLayer = new SFLDataFormat[IAX__20S_48KHZ_MAX];
- _floatBuffer8000 = new float32[IAX__20S_8KHZ_MAX];
- _floatBuffer48000 = new float32[IAX__20S_48KHZ_MAX];
- _intBuffer8000 = new int16[IAX__20S_8KHZ_MAX];
+ int nbSamplesMax = (int) ( converter->getFrequence() * converter->getFramesize() / 1000 );
- // libsamplerate-related
- _src_state_mic = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
- _src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
+ micData = new SFLDataFormat[nbSamplesMax];
+ micDataConverted = new SFLDataFormat[nbSamplesMax];
+ micDataEncoded = new unsigned char[nbSamplesMax];
+ spkrDataConverted = new SFLDataFormat[nbSamplesMax];
+ spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
}
@@ -87,20 +74,16 @@ IAXVoIPLink::~IAXVoIPLink()
terminate();
audiolayer = NULL;
- delete [] _intBuffer8000; _intBuffer8000 = NULL;
- delete [] _floatBuffer48000; _floatBuffer48000 = NULL;
- delete [] _floatBuffer8000; _floatBuffer8000 = NULL;
- delete [] _dataAudioLayer; _dataAudioLayer = NULL;
- delete [] _sendDataEncoded; _sendDataEncoded = NULL;
- delete [] _receiveDataDecoded; _receiveDataDecoded = NULL;
+ delete [] micData; micData = NULL;
+ delete [] micDataConverted; micDataConverted = NULL;
+ delete [] micDataEncoded; micDataEncoded = NULL;
- // libsamplerate-related
- _src_state_mic = src_delete(_src_state_mic);
- _src_state_spkr = src_delete(_src_state_spkr);
+ delete [] spkrDataDecoded; spkrDataDecoded = NULL;
+ delete [] spkrDataConverted; spkrDataConverted = NULL;
}
-bool
+ bool
IAXVoIPLink::init()
{
// If it was done, don't do it again, until we call terminate()
@@ -139,15 +122,14 @@ IAXVoIPLink::init()
}
if (port == IAX_FAILURE || nbTry==0) {
_debug("Fail to initialize iax\n");
-
+
_initDone = false;
}
-
return returnValue;
}
-void
+ void
IAXVoIPLink::terminate()
{
// If it was done, don't do it again, until we call init()
@@ -162,7 +144,7 @@ IAXVoIPLink::terminate()
_initDone = false;
}
-void
+ void
IAXVoIPLink::terminateIAXCall()
{
std::string reason = "Dumped Call";
@@ -183,7 +165,7 @@ IAXVoIPLink::terminateIAXCall()
_callMap.clear();
}
-void
+ void
IAXVoIPLink::getEvent()
{
IAXCall* call = NULL;
@@ -214,7 +196,7 @@ IAXVoIPLink::getEvent()
// We've got an event before it's associated with any call
iaxHandlePrecallEvent(event);
}
-
+
iax_event_free(event);
}
_mutexIAX.leaveMutex();
@@ -233,7 +215,7 @@ IAXVoIPLink::getEvent()
_evThread->sleep(3);
}
-void
+ void
IAXVoIPLink::sendAudioFromMic(void)
{
IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
@@ -242,7 +224,7 @@ IAXVoIPLink::sendAudioFromMic(void)
// Let's mind our own business.
return;
}
-
+
if( currentCall -> getAudioCodec() < 0 )
return;
@@ -266,13 +248,13 @@ 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
- int maxBytesToGet = audiolayer->getSampleRate()/50*sizeof(SFLDataFormat);
+ int maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
// available bytes inside ringbuffer
int availBytesFromMic = audiolayer->canGetMic();
@@ -285,59 +267,21 @@ IAXVoIPLink::sendAudioFromMic(void)
// take the lowest
int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
//_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
-
+
// Get bytes from micRingBuffer to data_from_mic
- int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
-
- // Audio ici est PARFAIT
- int16* toIAX = NULL;
- //if (audiolayer->getSampleRate() != audiocodec->getClockRate() && nbSample) {
- if (audiolayer->getSampleRate() != ac ->getClockRate() && nbSample) {
- SRC_DATA src_data;
-#ifdef DATAFORMAT_IS_FLOAT
- src_data.data_in = _dataAudioLayer;
-#else
- src_short_to_float_array(_dataAudioLayer, _floatBuffer48000, nbSample);
- src_data.data_in = _floatBuffer48000;
-#endif
-
- // Audio parfait à ce point.
- double factord = (double) ac->getClockRate() / audiolayer->getSampleRate();
-
- src_data.src_ratio = factord;
- src_data.input_frames = nbSample;
- src_data.output_frames = (int) floor(factord * nbSample);
- src_data.data_out = _floatBuffer8000;
- src_data.end_of_input = 0;
-
- src_process(_src_state_mic, &src_data);
-
- nbSample = src_data.output_frames_gen;
-
- // Bon, l'audio en float 8000 est laid mais yé consistant.
- src_float_to_short_array (_floatBuffer8000, _intBuffer8000, nbSample);
- toIAX = _intBuffer8000;
-
- // Audio bon ici aussi..
- } else {
-#ifdef DATAFORMAT_IS_FLOAT
- // convert _receiveDataDecoded to float inside _receiveData
- src_float_to_short_array(_dataAudioLayer, _intBuffer8000, nbSample);
- toIAX = _intBuffer8000;
- //if (nbSample > IAX__20S_8KHZ_MAX) { _debug("Alert from mic, nbSample %d is bigger than expected %d\n", nbSample, IAX__20S_8KHZ_MAX); }
-#else
- toIAX = _dataAudioLayer; // int to int
-#endif
- }
+ int nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
+
+ // resample
+ nbSample = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() , (int)audiolayer->getSampleRate() , nbSample );
// for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
- int compSize = ac->codecEncode(_sendDataEncoded, toIAX, nbSample*sizeof(int16));
+ int compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
// Send it out!
_mutexIAX.enterMutex();
// Make sure the session and the call still exists.
if (currentCall->getSession()) {
- if ( iax_send_voice(currentCall->getSession(), currentCall->getFormat(), (unsigned char*)_sendDataEncoded, compSize, nbSample) == -1) {
+ if ( iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample) == -1) {
_debug("IAX: Error sending voice data.\n");
}
}
@@ -346,7 +290,7 @@ IAXVoIPLink::sendAudioFromMic(void)
}
-IAXCall*
+ IAXCall*
IAXVoIPLink::getIAXCall(const CallID& id)
{
Call* call = getCall(id);
@@ -358,7 +302,7 @@ IAXVoIPLink::getIAXCall(const CallID& id)
-bool
+ bool
IAXVoIPLink::sendRegister()
{
bool result = false;
@@ -412,7 +356,7 @@ IAXVoIPLink::sendRegister()
-bool
+ bool
IAXVoIPLink::sendUnregister()
{
_mutexIAX.enterMutex();
@@ -434,7 +378,7 @@ IAXVoIPLink::sendUnregister()
return false;
}
-Call*
+ Call*
IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
IAXCall* call = new IAXCall(id, Call::Outgoing);
@@ -456,12 +400,12 @@ IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
}
-bool
+ bool
IAXVoIPLink::answer(const CallID& id)
{
IAXCall* call = getIAXCall(id);
call->setCodecMap(Manager::instance().getCodecDescriptorMap());
-
+
CHK_VALID_CALL;
_mutexIAX.enterMutex();
@@ -477,7 +421,7 @@ IAXVoIPLink::answer(const CallID& id)
return true;
}
-bool
+ bool
IAXVoIPLink::hangup(const CallID& id)
{
IAXCall* call = getIAXCall(id);
@@ -495,7 +439,7 @@ IAXVoIPLink::hangup(const CallID& id)
return true;
}
-bool
+ bool
IAXVoIPLink::onhold(const CallID& id)
{
IAXCall* call = getIAXCall(id);
@@ -503,7 +447,7 @@ IAXVoIPLink::onhold(const CallID& id)
CHK_VALID_CALL;
//if (call->getState() == Call::Hold) { _debug("Call is already on hold\n"); return false; }
-
+
_mutexIAX.enterMutex();
iax_quelch_moh(call->getSession() , MUSIC_ONHOLD);
_mutexIAX.leaveMutex();
@@ -512,7 +456,7 @@ IAXVoIPLink::onhold(const CallID& id)
return true;
}
-bool
+ bool
IAXVoIPLink::offhold(const CallID& id)
{
IAXCall* call = getIAXCall(id);
@@ -528,7 +472,7 @@ IAXVoIPLink::offhold(const CallID& id)
return true;
}
-bool
+ bool
IAXVoIPLink::transfer(const CallID& id, const std::string& to)
{
IAXCall* call = getIAXCall(id);
@@ -537,7 +481,7 @@ IAXVoIPLink::transfer(const CallID& id, const std::string& to)
char callto[to.length()+1];
strcpy(callto, to.c_str());
-
+
_mutexIAX.enterMutex();
iax_transfer(call->getSession(), callto);
_mutexIAX.leaveMutex();
@@ -546,7 +490,7 @@ IAXVoIPLink::transfer(const CallID& id, const std::string& to)
// removeCall(id);
}
-bool
+ bool
IAXVoIPLink::refuse(const CallID& id)
{
IAXCall* call = getIAXCall(id);
@@ -560,7 +504,7 @@ IAXVoIPLink::refuse(const CallID& id)
removeCall(id);
}
-bool
+ bool
IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
IAXCall* call = getIAXCall(id);
@@ -574,7 +518,7 @@ IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
-bool
+ bool
IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
{
struct iax_session *newsession;
@@ -582,18 +526,18 @@ IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
newsession = iax_session_new();
if (!newsession) {
- _debug("IAX Error: Can't make new session for a new call\n");
- return false;
+ _debug("IAX Error: Can't make new session for a new call\n");
+ return false;
}
call->setSession(newsession);
/* reset activity and ping "timers" */
// iaxc_note_activity(callNo);
-
+
std::string strNum = _user + ":" + _pass + "@" + _host + "/" + call->getPeerNumber();
char user[_user.length()+1];
strcpy(user, _user.c_str());
-
+
char num[strNum.length()+1];
strcpy(num, strNum.c_str());
@@ -610,7 +554,7 @@ IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
}
-IAXCall*
+ IAXCall*
IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
{
// access to callMap shoud use that
@@ -628,7 +572,7 @@ IAXVoIPLink::iaxFindCallBySession(struct iax_session* session)
return NULL; // not found
}
-void
+ void
IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
{
// call should not be 0
@@ -636,192 +580,158 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
//
CallID id = call->getCallId();
int16* output = 0; // for audio output
-
+
switch(event->etype) {
- case IAX_EVENT_HANGUP:
- Manager::instance().peerHungupCall(id);
- if (Manager::instance().isCurrentCall(id)) {
- audiolayer->stopStream();
- // stop audio
- }
- removeCall(id);
- break;
-
- case IAX_EVENT_REJECT:
- //Manager::instance().peerHungupCall(id);
- if (Manager::instance().isCurrentCall(id)) {
- // stop audio
- audiolayer->stopStream();
- }
- call->setConnectionState(Call::Connected);
- call->setState(Call::Error);
- Manager::instance().callFailure(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_ACCEPT:
- // Call accepted over there by the computer, not the user yet.
- if (event->ies.format) {
- call->setFormat(event->ies.format);
- }
- break;
-
- case IAX_EVENT_ANSWER:
- if (call->getConnectionState() != Call::Connected){
+ case IAX_EVENT_HANGUP:
+ Manager::instance().peerHungupCall(id);
+ if (Manager::instance().isCurrentCall(id)) {
+ audiolayer->stopStream();
+ // stop audio
+ }
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_REJECT:
+ //Manager::instance().peerHungupCall(id);
+ if (Manager::instance().isCurrentCall(id)) {
+ // stop audio
+ audiolayer->stopStream();
+ }
call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
+ call->setState(Call::Error);
+ Manager::instance().callFailure(id);
+ removeCall(id);
+ break;
+ case IAX_EVENT_ACCEPT:
+ // Call accepted over there by the computer, not the user yet.
if (event->ies.format) {
- // Should not get here, should have been set in EVENT_ACCEPT
call->setFormat(event->ies.format);
}
-
- Manager::instance().peerAnsweredCall(id);
- //audiolayer->flushMic();
- audiolayer->startStream();
- // start audio here?
- } else {
- // deja connecté ?
- }
- break;
-
- case IAX_EVENT_BUSY:
- call->setConnectionState(Call::Connected);
- call->setState(Call::Busy);
- Manager::instance().callBusy(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_VOICE:
- //_debug("Should have a decent value!!!!!! = %i\n" , call -> getAudioCodec());
- if( !audiolayer -> isCaptureActive())
- audiolayer->startStream();
- iaxHandleVoiceEvent(event, call);
- break;
-
- case IAX_EVENT_TEXT:
- break;
-
- case IAX_EVENT_RINGA:
- call->setConnectionState(Call::Ringing);
- Manager::instance().peerRingingCall(call->getCallId());
- break;
-
- case IAX_IE_MSGCOUNT:
- // _debug("messssssssssssssssssssssssssssssssssssssssssssssssages\n");
- break;
- case IAX_EVENT_PONG:
- break;
-
- case IAX_EVENT_URL:
- break;
-
- // case IAX_EVENT_CNG: ??
- // break;
-
- case IAX_EVENT_TIMEOUT:
- break;
-
- case IAX_EVENT_TRANSFER:
- break;
-
- default:
- _debug("Unknown event type (in call event): %d\n", event->etype);
-
+ break;
+
+ case IAX_EVENT_ANSWER:
+ if (call->getConnectionState() != Call::Connected){
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+
+ if (event->ies.format) {
+ // Should not get here, should have been set in EVENT_ACCEPT
+ call->setFormat(event->ies.format);
+ }
+
+ Manager::instance().peerAnsweredCall(id);
+ //audiolayer->flushMic();
+ audiolayer->startStream();
+ // start audio here?
+ } else {
+ // deja connecté ?
+ }
+ break;
+
+ case IAX_EVENT_BUSY:
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Busy);
+ Manager::instance().callBusy(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_VOICE:
+ //_debug("Should have a decent value!!!!!! = %i\n" , call -> getAudioCodec());
+ //if( !audiolayer -> isCaptureActive())
+ //audiolayer->startStream();
+ iaxHandleVoiceEvent(event, call);
+ break;
+
+ case IAX_EVENT_TEXT:
+ break;
+
+ case IAX_EVENT_RINGA:
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+ break;
+
+ case IAX_IE_MSGCOUNT:
+ break;
+ case IAX_EVENT_PONG:
+ break;
+
+ case IAX_EVENT_URL:
+ break;
+
+ // case IAX_EVENT_CNG: ??
+ // break;
+
+ case IAX_EVENT_TIMEOUT:
+ break;
+
+ case IAX_EVENT_TRANSFER:
+ break;
+
+ default:
+ _debug("Unknown event type (in call event): %d\n", event->etype);
+
}
}
/* Handle audio event, VOICE packet received */
-void
+ void
IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
{
- // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
- // were triggered
- if (!event->datalen) {
- // Skip this empty packet.
- //_debug("IAX: Skipping empty jitter-buffer interpolated packet\n");
- return;
+ // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
+ // were triggered
+ if (!event->datalen) {
+ // Skip this empty packet.
+ //_debug("IAX: Skipping empty jitter-buffer interpolated packet\n");
+ return;
+ }
+
+ 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.
+ if (event->subclass && event->subclass != call->getFormat()) {
+ call->setFormat(event->subclass);
}
+ //_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
+ AudioCodec* ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
- if (audiolayer) {
- // _debug("codec = %i\n" , call->getFormat());
- // _debug("codec = %i\n" , _audiocodec->getPayload());
- //_debug("codec = %s\n" , _audiocodec->getCodecName().c_str());
- // 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.
- if (event->subclass && event->subclass != call->getFormat()) {
- call->setFormat(event->subclass);
- }
- //_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
- AudioCodec* ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
+ unsigned char* data = (unsigned char*)event->data;
+ unsigned int size = event->datalen;
- unsigned char* data = (unsigned char*)event->data;
- unsigned int size = event->datalen;
+ // Decode data with relevant codec
+ int max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
- if (size > IAX__20S_8KHZ_MAX) {
- _debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, IAX__20S_8KHZ_MAX);
- size = IAX__20S_8KHZ_MAX;
- }
+ if (size > max) {
+ _debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, max);
+ size = max;
+ }
- int expandedSize = ac->codecDecode(_receiveDataDecoded, data, size);
- int nbInt16 = expandedSize/sizeof(int16);
+ int expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
+ int nbInt16 = expandedSize/sizeof(int16);
- if (nbInt16 > IAX__20S_8KHZ_MAX) {
- _debug("We have decoded an IAX VOICE packet larger than expected: %s VS %s. Cropping.\n", nbInt16, IAX__20S_8KHZ_MAX);
- nbInt16 = IAX__20S_8KHZ_MAX;
- }
-
- SFLDataFormat* toAudioLayer;
- int nbSample = nbInt16;
- int nbSampleMaxRate = nbInt16 * 6;
-
- if ( audiolayer->getSampleRate() != ac->getClockRate() && nbSample ) {
- // Do sample rate conversion
- double factord = (double) audiolayer->getSampleRate() / ac->getClockRate();
- // SRC_DATA from samplerate.h
- SRC_DATA src_data;
- src_data.data_in = _floatBuffer8000;
- src_data.data_out = _floatBuffer48000;
- src_data.input_frames = nbSample;
- src_data.output_frames = (int) floor(factord * nbSample);
- src_data.src_ratio = factord;
- src_data.end_of_input = 0;
- src_short_to_float_array(_receiveDataDecoded, _floatBuffer8000, nbSample);
-
- // samplerate convert, go!
- src_process(_src_state_spkr, &src_data);
-
- nbSample = ( src_data.output_frames_gen > IAX__20S_48KHZ_MAX) ? IAX__20S_48KHZ_MAX : src_data.output_frames_gen;
-#ifdef DATAFORMAT_IS_FLOAT
- toAudioLayer = _floatBuffer48000;
-#else
- src_float_to_short_array(_floatBuffer48000, _dataAudioLayer, nbSample);
- toAudioLayer = _dataAudioLayer;
-#endif
-
- } else {
- nbSample = nbInt16;
-#ifdef DATAFORMAT_IS_FLOAT
- // convert _receiveDataDecoded to float inside _receiveData
- src_short_to_float_array(_receiveDataDecoded, _floatBuffer8000, nbSample);
- toAudioLayer = _floatBuffer8000;
-#else
- toAudioLayer = _receiveDataDecoded; // int to int
-#endif
- }
- audiolayer->playSamples(toAudioLayer, nbSample * sizeof(SFLDataFormat), true);
- } else {
- _debug("IAX: incoming audio, but no sound card open");
+ if (nbInt16 > max) {
+ _debug("We have decoded an IAX VOICE packet larger than expected: %s VS %s. Cropping.\n", nbInt16, max);
+ nbInt16 = max;
}
+ int nbSample = nbInt16;
+ // resample
+ nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
+
+ audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
+
+ } else {
+ _debug("IAX: incoming audio, but no sound card open");
+ }
+
}
/**
* Handle the registration process
*/
-void
+ void
IAXVoIPLink::iaxHandleRegReply(iax_event* event)
{
if (event->etype == IAX_EVENT_REGREJ) {
@@ -846,7 +756,7 @@ IAXVoIPLink::iaxHandleRegReply(iax_event* event)
}
}
-void
+ void
IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
{
IAXCall* call = NULL;
@@ -854,96 +764,96 @@ IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
std::string reason = "Error ringing user.";
switch(event->etype) {
- case IAX_EVENT_REGACK:
- case IAX_EVENT_REGREJ:
- _debug("IAX Registration Event in a pre-call setup\n");
- break;
-
- case IAX_EVENT_REGREQ:
- // Received when someone wants to register to us!?!
- // Asterisk receives and answers to that, not us, we're a phone.
- _debug("Registration by a peer, don't allow it\n");
- break;
-
- case IAX_EVENT_CONNECT:
- // We've got an incoming call! Yikes!
- _debug("> IAX_EVENT_CONNECT (receive)\n");
-
- id = Manager::instance().getNewCallID();
-
-
- call = new IAXCall(id, Call::Incoming);
-
- if (!call) {
- _debug("! IAX Failure: unable to create an incoming call");
- return;
- }
+ case IAX_EVENT_REGACK:
+ case IAX_EVENT_REGREJ:
+ _debug("IAX Registration Event in a pre-call setup\n");
+ break;
- // Setup the new IAXCall
- // Associate the call to the session.
- call->setSession(event->session);
+ case IAX_EVENT_REGREQ:
+ // Received when someone wants to register to us!?!
+ // Asterisk receives and answers to that, not us, we're a phone.
+ _debug("Registration by a peer, don't allow it\n");
+ break;
- // setCallAudioLocal(call);
- call->setCodecMap(Manager::instance().getCodecDescriptorMap());
- call->setConnectionState(Call::Progressing);
+ case IAX_EVENT_CONNECT:
+ // We've got an incoming call! Yikes!
+ _debug("> IAX_EVENT_CONNECT (receive)\n");
+ id = Manager::instance().getNewCallID();
- if (event->ies.calling_number)
- call->setPeerNumber(std::string(event->ies.calling_number));
- if (event->ies.calling_name)
- call->setPeerName(std::string(event->ies.calling_name));
- if (Manager::instance().incomingCall(call, getAccountID())) {
- /** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
- * l'établissement du codec de transmission */
+ call = new IAXCall(id, Call::Incoming);
- // Remote lists its capabilities
- int format = call->getFirstMatchingFormat(event->ies.capability);
- // Remote asks for preferred codec voiceformat
- int pref_format = call->getFirstMatchingFormat(event->ies.format);
+ if (!call) {
+ _debug("! IAX Failure: unable to create an incoming call");
+ return;
+ }
- // Priority to remote's suggestion. In case it's a forwarding, no transcoding
- // will be needed from the server, thus less latency.
- if (pref_format)
- format = pref_format;
+ // Setup the new IAXCall
+ // Associate the call to the session.
+ call->setSession(event->session);
- iax_accept(event->session, format);
- iax_ring_announce(event->session);
+ // setCallAudioLocal(call);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ call->setConnectionState(Call::Progressing);
- addCall(call);
- } else {
- // reject call, unable to add it
- iax_reject(event->session, (char*)reason.c_str());
- delete call; call = NULL;
- }
+ if (event->ies.calling_number)
+ call->setPeerNumber(std::string(event->ies.calling_number));
+ if (event->ies.calling_name)
+ call->setPeerName(std::string(event->ies.calling_name));
- break;
-
- case IAX_EVENT_HANGUP:
- // Remote peer hung up
- call = iaxFindCallBySession(event->session);
- id = call->getCallId();
-
- Manager::instance().peerHungupCall(id);
- removeCall(id);
- break;
-
- case IAX_EVENT_TIMEOUT: // timeout for an unknown session
-
- break;
-
- case IAX_IE_MSGCOUNT:
- //_debug("messssssssssssssssssssssssssssssssssssssssssssssssages\n");
- break;
-
- default:
- _debug("Unknown event type (in precall): %d\n", event->etype);
+ if (Manager::instance().incomingCall(call, getAccountID())) {
+ /** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
+ * l'établissement du codec de transmission */
+
+ // Remote lists its capabilities
+ int format = call->getFirstMatchingFormat(event->ies.capability);
+ // Remote asks for preferred codec voiceformat
+ int pref_format = call->getFirstMatchingFormat(event->ies.format);
+
+ // Priority to remote's suggestion. In case it's a forwarding, no transcoding
+ // will be needed from the server, thus less latency.
+ if (pref_format)
+ format = pref_format;
+
+ iax_accept(event->session, format);
+ iax_ring_announce(event->session);
+
+ addCall(call);
+ } else {
+ // reject call, unable to add it
+ iax_reject(event->session, (char*)reason.c_str());
+
+ delete call; call = NULL;
+ }
+
+ break;
+
+ case IAX_EVENT_HANGUP:
+ // Remote peer hung up
+ call = iaxFindCallBySession(event->session);
+ id = call->getCallId();
+
+ Manager::instance().peerHungupCall(id);
+ removeCall(id);
+ break;
+
+ case IAX_EVENT_TIMEOUT: // timeout for an unknown session
+
+ break;
+
+ case IAX_IE_MSGCOUNT:
+ //_debug("messssssssssssssssssssssssssssssssssssssssssssssssages\n");
+ break;
+
+ default:
+ _debug("Unknown event type (in precall): %d\n", event->etype);
}
-
+
}
-int
+ int
IAXVoIPLink::iaxCodecMapToFormat(IAXCall* call)
{
CodecOrder map = call->getCodecMap().getActiveCodecs();
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index ea67145ede0534d38fd60a0dca6d392dd3b04250..3a8c3fbd4a11b183e6cd58b4f99a0cce81cc238d 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -1,8 +1,8 @@
/*
- * Copyright (C) 2006-2007 Savoir-Faire Linux inc.
+ * Copyright (C) 2006-2007-2008 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
* Author: Yan Morin <yan.morin@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
@@ -24,9 +24,9 @@
#include "voiplink.h"
#include <iax2/iax-client.h>
#include "global.h"
-#include <samplerate.h>
#include "audio/codecDescriptor.h"
+#include "samplerateconverter.h"
class EventThread;
class IAXCall;
@@ -279,31 +279,17 @@ class IAXVoIPLink : public VoIPLink
/** Connection to audio card/device */
AudioLayer* audiolayer;
- /** When we receive data, we decode it inside this buffer */
- int16* _receiveDataDecoded;
- /** When we send data, we encode it inside this buffer*/
- unsigned char* _sendDataEncoded;
-
- /** After that we send the data inside this buffer if there is a format conversion or rate conversion. */
- /* Also use for getting mic-ringbuffer data */
- SFLDataFormat* _dataAudioLayer;
-
- /** Buffer for 8000hz samples in conversion */
- float32* _floatBuffer8000;
- /** Buffer for 48000hz samples in conversion */
- float32* _floatBuffer48000;
-
- /** Buffer for 8000hz samples for mic conversion */
- int16* _intBuffer8000;
-
- /** libsamplerate converter for incoming voice */
- SRC_STATE* _src_state_spkr;
+ /** Mic-data related buffers */
+ SFLDataFormat* micData;
+ SFLDataFormat* micDataConverted;
+ unsigned char* micDataEncoded;
- /** libsamplerate converter for outgoing voice */
- SRC_STATE* _src_state_mic;
+ /** Speaker-data related buffers */
+ SFLDataFormat* spkrDataDecoded;
+ SFLDataFormat* spkrDataConverted;
- /** libsamplerate error */
- int _src_err;
+ /** Sample rate converter object */
+ SamplerateConverter* converter;
};
diff --git a/src/samplerateconverter.cpp b/src/samplerateconverter.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..126c5d957cf7a9712bfbf4b105a0d817d6f653aa
--- /dev/null
+++ b/src/samplerateconverter.cpp
@@ -0,0 +1,116 @@
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * 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 "samplerateconverter.h"
+
+SamplerateConverter::SamplerateConverter( void ) {
+ // Default values
+ _frequence = Manager::instance().getConfigInt( AUDIO , ALSA_SAMPLE_RATE ); // 44100;
+ _framesize = Manager::instance().getConfigInt( AUDIO , ALSA_FRAME_SIZE );
+
+ init();
+}
+
+SamplerateConverter::SamplerateConverter( int freq , int fs ) {
+
+ _frequence = freq ;
+ _framesize = fs ;
+
+ init();
+}
+
+SamplerateConverter::~SamplerateConverter( void ) {
+
+ delete [] _floatBufferUpMic; _floatBufferUpMic = NULL;
+ delete [] _floatBufferDownMic; _floatBufferDownMic = NULL;
+
+ delete [] _floatBufferUpSpkr; _floatBufferUpSpkr = NULL;
+ delete [] _floatBufferDownSpkr; _floatBufferDownSpkr = NULL;
+
+ // libSamplerateConverter-related
+ _src_state_mic = src_delete(_src_state_mic);
+ _src_state_spkr = src_delete(_src_state_spkr);
+}
+
+void SamplerateConverter::init( void ) {
+
+ // libSamplerateConverter-related
+ // Set the converter type for the upsampling and the downsampling
+ // interpolator SRC_SINC_BEST_QUALITY
+ _src_state_mic = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
+ _src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
+
+ int nbSamplesMax = (int) ( getFrequence() * getFramesize() / 1000 );
+ _floatBufferDownMic = new float32[nbSamplesMax];
+ _floatBufferUpMic = new float32[nbSamplesMax];
+ _floatBufferDownSpkr = new float32[nbSamplesMax];
+ _floatBufferUpSpkr = new float32[nbSamplesMax];
+}
+
+//TODO Add ifdef for int16 or float32 type
+int SamplerateConverter::upsampleData( SFLDataFormat* dataIn , SFLDataFormat* dataOut, int samplerate1 , int samplerate2 , int nbSamples ){
+
+ double upsampleFactor = (double)samplerate2 / samplerate1 ;
+ int nbSamplesMax = (int) ( samplerate2 * getFramesize() / 1000 );
+ if( upsampleFactor != 1 && dataIn != NULL )
+ {
+ SRC_DATA src_data;
+ src_data.data_in = _floatBufferDownSpkr;
+ src_data.data_out = _floatBufferUpSpkr;
+ src_data.input_frames = nbSamples;
+ src_data.output_frames = (int) floor(upsampleFactor * nbSamples);
+ src_data.src_ratio = upsampleFactor;
+ src_data.end_of_input = 0; // More data will come
+ //_debug("upsample %d %d %f %d\n" , src_data.input_frames , src_data.output_frames, src_data.src_ratio , nbSamples);
+ src_short_to_float_array( dataIn , _floatBufferDownSpkr, nbSamples);
+ //_debug("upsample %d %f %d\n" , src_data.output_frames, src_data.src_ratio , nbSamples);
+ src_process(_src_state_spkr, &src_data);
+ //_debug("upsample %d %d %d\n" , samplerate1, samplerate2 , nbSamples);
+ nbSamples = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
+ src_float_to_short_array(_floatBufferUpSpkr, dataOut, nbSamples);
+ //_debug("upsample %d %d %d\n" , samplerate1, samplerate2 , nbSamples);
+ }
+ return nbSamples;
+}
+
+//TODO Add ifdef for int16 or float32 type
+int SamplerateConverter::downsampleData( SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples ){
+
+ double downsampleFactor = (double)samplerate1 / samplerate2;
+ //_debug("factor = %f\n" , downsampleFactor);
+ int nbSamplesMax = (int) ( samplerate1 * getFramesize() / 1000 );
+ if ( downsampleFactor != 1)
+ {
+ SRC_DATA src_data;
+ src_data.data_in = _floatBufferUpMic;
+ src_data.data_out = _floatBufferDownMic;
+ src_data.input_frames = nbSamples;
+ src_data.output_frames = (int) floor(downsampleFactor * nbSamples);
+ src_data.src_ratio = downsampleFactor;
+ src_data.end_of_input = 0; // More data will come
+ //_debug("downsample %d %f %d\n" , src_data.output_frames, src_data.src_ratio , nbSamples);
+ src_short_to_float_array( dataIn, _floatBufferUpMic, nbSamples );
+ //_debug("downsample %d %f %d\n" , src_data.output_frames, src_data.src_ratio , nbSamples);
+ src_process(_src_state_mic, &src_data);
+ //_debug("downsample %d %f %d\n" , src_data.output_frames, src_data.src_ratio , nbSamples);
+ nbSamples = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
+ //_debug("downsample %d %f %d\n" , src_data.output_frames, src_data.src_ratio , nbSamples);
+ src_float_to_short_array( _floatBufferDownMic , dataOut , nbSamples );
+ }
+ return nbSamples;
+}
diff --git a/src/samplerateconverter.h b/src/samplerateconverter.h
new file mode 100644
index 0000000000000000000000000000000000000000..86ea175319c44fd36660d3fdf8b33a7ef9de1e46
--- /dev/null
+++ b/src/samplerateconverter.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * 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.
+ */
+
+#ifndef _SAMPLE_RATE_H
+#define _SAMPLE_RATE_H
+
+#include <samplerate.h>
+#include <math.h>
+
+#include "global.h"
+#include "manager.h"
+
+class SamplerateConverter {
+ public:
+ /** Constructor */
+ SamplerateConverter( void );
+ SamplerateConverter( int freq , int fs );
+ /** Destructor */
+ ~SamplerateConverter( void );
+
+ /**
+ * Upsample from the samplerate1 to the samplerate2
+ * @param data The data buffer
+ * @param SamplerateConverter1 The lower sample rate
+ * @param SamplerateConverter2 The higher sample rate
+ * @param nbSamples The number of samples to process
+ * @return int The number of samples after the operation
+ */
+ int upsampleData( SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples );
+
+ /**
+ * Downsample from the samplerate1 to the samplerate2
+ * @param data The data buffer
+ * @param SamplerateConverter1 The lower sample rate
+ * @param SamplerateConverter2 The higher sample rate
+ * @param nbSamples The number of samples to process
+ * @return int The number of samples after the operation
+ */
+ int downsampleData( SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples );
+
+ int getFrequence( void ) { return _frequence; }
+
+ int getFramesize( void ) { return _framesize; }
+
+ private:
+ void init( void );
+
+ /** Audio layer caracteristics */
+ int _frequence;
+ int _framesize;
+
+ /** Downsampled/Upsampled float buffers for the mic data processing */
+ float32* _floatBufferDownMic;
+ float32* _floatBufferUpMic;
+ /** libSamplerateConverter converter for outgoing voice */
+ SRC_STATE* _src_state_mic;
+
+ /** Downsampled/Upsampled float buffers for the speaker data processing */
+ float32* _floatBufferDownSpkr;
+ float32* _floatBufferUpSpkr;
+ /** libSamplerateConverter converter for incoming voice */
+ SRC_STATE* _src_state_spkr;
+ /** libSamplerateConverter error */
+ int _src_err;
+};
+
+#endif //_SAMPLE_RATE_H
+
diff --git a/src/voiplink.cpp b/src/voiplink.cpp
index 59aca9d19989c81e01e7a5da13eb410cf880366d..1a762036ed9a44f0c84b2b0458da5dfabc09d22e 100644
--- a/src/voiplink.cpp
+++ b/src/voiplink.cpp
@@ -25,8 +25,9 @@
#include "voiplink.h"
#include "manager.h"
-VoIPLink::VoIPLink(const AccountID& accountID) : _accountID(accountID), _localIPAddress("127.0.0.1"), _localPort(0), _registrationError(NO_ERROR), _initDone(false)
+VoIPLink::VoIPLink(const AccountID& accountID) : _accountID(accountID), _localIPAddress("127.0.0.1"), _localPort(0), _initDone(false)
{
+ _registrationError = NO_ERROR;
}
VoIPLink::~VoIPLink (void)