From a701fe3bba0e6c16224d2cd8e41ba65998a073b8 Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Sat, 29 Oct 2005 17:48:23 +0000
Subject: [PATCH] Don't close sflphone when portaudio failed to load Remove
enable_audio variable (non-protected) in SipCall
---
src/audio/audiolayer.cpp | 60 ++++++++++++++++++++++++---------
src/audio/audiolayer.h | 23 +++++++------
src/audio/audiortp.cpp | 13 +++-----
src/managerimpl.cpp | 72 +++++++++++++++++++---------------------
src/sipcall.cpp | 11 +++---
src/sipcall.h | 1 -
src/sipvoiplink.cpp | 29 ++++++++--------
7 files changed, 112 insertions(+), 97 deletions(-)
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 2e9650a560..d370b52982 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -33,10 +33,9 @@ AudioLayer::AudioLayer()
, _mainSndRingBuffer(SIZEBUF)
, _micRingBuffer(SIZEBUF)
, _stream(NULL)
+ , _errorMessage("")
{
- _debugInit(" portaudio initialization...");
portaudio::System::initialize();
- _debugInit(" portaudio end initialization.");
NBCHARFORTWOINT16 = sizeof(int16)/sizeof(unsigned char) * CHANNELS;
}
@@ -61,7 +60,7 @@ void
AudioLayer::listDevices()
{
ost::MutexLock guard(_mutex);
- portaudio::System::DeviceIterator pos = portaudio::System::instance().devicesBegin();
+ portaudio::System::DeviceIterator pos = portaudio::System::instance().devicesBegin();
while(pos != portaudio::System::instance().devicesEnd()) {
_debug("AudioLayer: Device (%d) %s\n", pos->index(), pos->name());
pos++;
@@ -72,8 +71,8 @@ AudioLayer::listDevices()
void
AudioLayer::openDevice (int index)
{
- ost::MutexLock guard(_mutex);
closeStream();
+
// Set up the parameters required to open a (Callback)Stream:
portaudio::DirectionSpecificStreamParameters
outParams(portaudio::System::instance().deviceByIndex(index),
@@ -93,6 +92,7 @@ AudioLayer::openDevice (int index)
SAMPLING_RATE, FRAME_PER_BUFFER /*paFramesPerBufferUnspecified*/, paNoFlag /*paPrimeOutputBuffersUsingStreamCallback | paNeverDropInput*/);
// Create (and open) a new Stream, using the AudioLayer::audioCallback
+ ost::MutexLock guard(_mutex);
_stream = new portaudio::MemFunCallbackStream<AudioLayer>(params,
*this,
&AudioLayer::audioCallback);
@@ -128,7 +128,9 @@ AudioLayer::stopStream(void)
void
AudioLayer::sleep(int msec)
{
- portaudio::System::instance().sleep(msec);
+ if (_stream) {
+ portaudio::System::instance().sleep(msec);
+ }
}
bool
@@ -143,16 +145,19 @@ AudioLayer::isStreamActive (void)
}
}
-void
+int
AudioLayer::putMain(void* buffer, int toCopy)
{
ost::MutexLock guard(_mutex);
- int a = _mainSndRingBuffer.AvailForPut();
- if ( a >= toCopy ) {
- _mainSndRingBuffer.Put(buffer, toCopy);
- } else {
- _mainSndRingBuffer.Put(buffer, a);
+ if (_stream) {
+ int a = _mainSndRingBuffer.AvailForPut();
+ if ( a >= toCopy ) {
+ return _mainSndRingBuffer.Put(buffer, toCopy);
+ } else {
+ return _mainSndRingBuffer.Put(buffer, a);
+ }
}
+ return 0;
}
void
@@ -162,14 +167,37 @@ AudioLayer::flushMain()
_mainSndRingBuffer.flush();
}
-void
+int
AudioLayer::putUrgent(void* buffer, int toCopy)
{
- int a = _urgentRingBuffer.AvailForPut();
- if ( a >= toCopy ) {
- _urgentRingBuffer.Put(buffer, toCopy);
+ if (_stream) {
+ int a = _urgentRingBuffer.AvailForPut();
+ if ( a >= toCopy ) {
+ return _urgentRingBuffer.Put(buffer, toCopy);
+ } else {
+ return _urgentRingBuffer.Put(buffer, a);
+ }
+ }
+ return 0;
+}
+
+int
+AudioLayer::canGetMic()
+{
+ if (_stream) {
+ return _micRingBuffer.AvailForGet();
+ } else {
+ return 0;
+ }
+}
+
+int
+AudioLayer::getMic(void *buffer, int toCopy)
+{
+ if(_stream) {
+ return _micRingBuffer.Get(buffer, toCopy, 100);
} else {
- _urgentRingBuffer.Put(buffer, a);
+ return 0;
}
}
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 3c0cdab759..8349d9e655 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -52,23 +52,26 @@ public:
bool isStreamStopped (void);
void flushMain();
- void putMain(void* buffer, int toCopy);
- void putUrgent(void* buffer, int toCopy);
+ int putMain(void* buffer, int toCopy);
+ int putUrgent(void* buffer, int toCopy);
+ int canGetMic();
+ int getMic(void *, int);
void flushMic();
- int audioCallback (const void *, void *, unsigned long,
- const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
+ int audioCallback (const void *, void *, unsigned long,
+ const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
- inline RingBuffer &urgentRingBuffer(void) { return _urgentRingBuffer; }
- inline RingBuffer &micRingBuffer(void) { return _micRingBuffer; }
+ void setErrorMessage(const std::string& error) { _errorMessage = error; }
+ std::string getErrorMessage() { return _errorMessage; }
private:
- void closeStream (void);
- RingBuffer _urgentRingBuffer;
- RingBuffer _mainSndRingBuffer;
- RingBuffer _micRingBuffer;
+ void closeStream (void);
+ RingBuffer _urgentRingBuffer;
+ RingBuffer _mainSndRingBuffer;
+ RingBuffer _micRingBuffer;
portaudio::MemFunCallbackStream<AudioLayer> *_stream;
+ std::string _errorMessage;
// portaudio::AutoSystem autoSys;
ost::Mutex _mutex;
int NBCHARFORTWOINT16;
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 6a65a1ffe2..7eeac7441d 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -48,7 +48,6 @@ AudioRtp::~AudioRtp (void) {
int
AudioRtp::createNewSession (SipCall *ca) {
// Start RTP Send/Receive threads
- ca->enable_audio = 1;
_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
_RTXThread = new AudioRtpRTX (ca, Manager::instance().getAudioDriver(), _symmetric);
@@ -169,7 +168,7 @@ AudioRtpRTX::initAudioRtpSession (void)
void
AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic_stereo, int16* data_from_mic_mono, int timestamp)
{
- int availBytesFromMic = Manager::instance().getAudioDriver()->micRingBuffer().AvailForGet();
+ int availBytesFromMic = Manager::instance().getAudioDriver()->canGetMic();
int maxBytesToGet = RTP_FRAMES2SEND * 2 * 2; // * channels * int16/byte
int bytesAvail;
@@ -181,7 +180,7 @@ AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_m
}
// Get bytes from micRingBuffer to data_from_mic
- Manager::instance().getAudioDriver()->micRingBuffer().Get(data_from_mic_stereo, bytesAvail, 100);
+ Manager::instance().getAudioDriver()->getMic(data_from_mic_stereo, bytesAvail);
// control volume and stereo->mono
// the j is in int16 RTP_FRAMES2SEND
// data_from_mic_mono = 0 to RTP_FRAME2SEND [in int16]
@@ -225,7 +224,6 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers_stereo, int16* data
adu = _session->getData(_session->getFirstTimestamp());
}
if (adu == NULL) {
- //Manager::instance().getAudioDriver()->flushMain();
return;
}
@@ -256,8 +254,6 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers_stereo, int16* data
// Set decoded data to sound device
// expandedSize is in mono/bytes, since we double in stereo, we send two time more
Manager::instance().getAudioDriver()->putMain(data_for_speakers_stereo, expandedSize*2);
- //}
-
Manager::instance().getAudioDriver()->startStream();
// Notify (with a beep) an incoming call when there is already a call
@@ -309,7 +305,7 @@ AudioRtpRTX::run (void) {
TimerPort::setTimer(frameSize);
audiolayer->flushMic();
- while (!testCancel() && _ca != NULL && _ca->enable_audio != -1) {
+ while (!testCancel() && _ca != NULL) {
////////////////////////////
// Send session
////////////////////////////
@@ -325,14 +321,13 @@ AudioRtpRTX::run (void) {
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(frameSize); // 'frameSize' ms
}
+ audiolayer->stopStream();
delete [] data_for_speakers_stereo; data_for_speakers_stereo = 0;
delete [] data_for_speakers_recv; data_for_speakers_recv = 0;
delete [] char_to_send; char_to_send = 0;
delete [] data_from_mic_mono; data_from_mic_mono = 0;
delete [] data_from_mic_stereo; data_from_mic_stereo = 0;
-
- audiolayer->stopStream();
}
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index bebae80cab..baec39cc25 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -128,26 +128,19 @@ ManagerImpl::init()
try {
selectAudioDriver();
}
- catch (const portaudio::PaException &e)
- {
- displayError(e.paErrorText());
- throw e;
- }
- catch (const portaudio::PaCppException &e)
- {
- displayError(e.what());
- throw e;
- }
- catch (const std::runtime_error &e)
- {
- displayError(e.what());
- throw e;
- }
- catch (...)
- {
- displayError("An unknown exception occured.");
+ catch (const portaudio::PaException &e) {
+ getAudioDriver()->setErrorMessage(e.paErrorText());
+ }
+ catch (const portaudio::PaCppException &e) {
+ getAudioDriver()->setErrorMessage(e.what());
+ }
+ catch (const std::runtime_error &e) {
+ getAudioDriver()->setErrorMessage(e.what());
+ }
+ catch (...) {
+ displayError("An unknown exception occured while selecting audio driver.");
throw;
- }
+ }
initAudioCodec();
_debugInit("Adding new VoIP Link");
@@ -584,11 +577,7 @@ ManagerImpl::playDtmf(char code)
// put the size in bytes...
// so size * CHANNELS * 2 (bytes for the int16)
int nbInt16InChar = sizeof(int16)/sizeof(char);
- int toSend = audiolayer->urgentRingBuffer().AvailForPut();
- if (toSend > (size * CHANNELS * nbInt16InChar)) {
- toSend = size * CHANNELS * nbInt16InChar;
- }
- audiolayer->urgentRingBuffer().Put(buf_ctrl_vol, toSend);
+ audiolayer->putUrgent(buf_ctrl_vol, size * CHANNELS * nbInt16InChar);
// We activate the stream if it's not active yet.
if (!audiolayer->isStreamActive()) {
@@ -1509,24 +1498,31 @@ ManagerImpl::getConfigList(const std::string& sequenceId, const std::string& nam
bool
ManagerImpl::getAudioDeviceList(const std::string& sequenceId)
{
- TokenList tk;
- portaudio::System& sys = portaudio::System::instance();
+ bool returnValue = false;
+ try {
+ // TODO: test when there is an error on initializing...
+ TokenList tk;
+ portaudio::System& sys = portaudio::System::instance();
- const char *hostApiName;
- const char *deviceName;
+ const char *hostApiName;
+ const char *deviceName;
- for (int index = 0; index < sys.deviceCount(); index++ ) {
- portaudio::Device& device = sys.deviceByIndex(index);
- hostApiName = device.hostApi().name();
- deviceName = device.name();
+ for (int index = 0; index < sys.deviceCount(); index++ ) {
+ portaudio::Device& device = sys.deviceByIndex(index);
+ hostApiName = device.hostApi().name();
+ deviceName = device.name();
- tk.clear();
- std::ostringstream str; str << index; tk.push_back(str.str());
- tk.push_back(deviceName);
- tk.push_back(std::string(hostApiName));
- _gui->sendMessage("100", sequenceId, tk);
+ tk.clear();
+ std::ostringstream str; str << index; tk.push_back(str.str());
+ tk.push_back(deviceName);
+ tk.push_back(std::string(hostApiName));
+ _gui->sendMessage("100", sequenceId, tk);
+ }
+ returnValue = true;
+ } catch (...) {
+ returnValue = false;
}
- return true;
+ return returnValue;
}
bool
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index c6eb8e8744..644657793f 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -44,7 +44,6 @@ SipCall::SipCall (CALLID id, CodecDescriptorVector* cdv) : _localIp("127.0.0.1")
alloc(); // char* allocation
_cdv = cdv;
_audiocodec = NULL;
- enable_audio = false;
_local_audio_port = 0;
_remote_sdp_audio_port = 0;
@@ -262,7 +261,6 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
eXosip_call_send_answer (_tid, 415, NULL);
_debug("< Sending Answer 415\n");
} else {
- enable_audio = false;
sdp_message_t *local_sdp = eXosip_get_sdp_info(answer);
sdp_media_t *local_med = NULL;
@@ -397,18 +395,17 @@ SipCall::answeredCall(eXosip_event_t *event) {
void
SipCall::answeredCall_without_hold (eXosip_event_t *event)
{
- if (enable_audio == true && event->response != NULL) {
+ if (event->response == NULL ) { return; }
+
+ if (_cid!=0) {
sdp_message_t *sdp = eXosip_get_sdp_info (event->response);
if (sdp != NULL) {
/* audio is started and session has just been modified */
- enable_audio = false;
sdp_message_free (sdp);
}
}
- if (enable_audio == false &&
- event->request != NULL &&
- event->response != NULL) { /* audio is started */
+ if (event->request != NULL) { /* audio is started */
sdp_message_t *local_sdp = eXosip_get_sdp_info (event->request);
sdp_message_t *remote_sdp = eXosip_get_sdp_info (event->response);
diff --git a/src/sipcall.h b/src/sipcall.h
index 87f2e6fd84..bbeea534a2 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -44,7 +44,6 @@ public:
~SipCall (void);
int payload;
- bool enable_audio; /* true = started, false = stopped */
/*
* Store information about incoming call and negociate payload
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index d68d0b5deb..25e53f2f52 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -402,7 +402,6 @@ SipVoIPLink::hangup (CALLID id)
eXosip_unlock();
// Release RTP channels
- sipcall->enable_audio = false;
_audiortp.closeRtpSession();
deleteSipCall(id);
@@ -482,15 +481,14 @@ SipVoIPLink::onhold (CALLID id)
osip_message_set_content_type (invite, "application/sdp");
}
- eXosip_lock ();
// Send request
_audiortp.closeRtpSession();
+ eXosip_lock ();
i = eXosip_call_send_request (did, invite);
eXosip_unlock ();
// Disable audio
- sipcall->enable_audio = false;
return i;
}
@@ -682,7 +680,18 @@ SipVoIPLink::getEvent (void)
// proceeding call...
break;
- // The peer-user answers
+ case EXOSIP_CALL_RINGING: // 9 peer call is ringing
+ id = findCallIdInitial(event);
+ _debug("Call is ringing [id = %d, cid = %d, did = %d]\n", id, event->cid, event->did);
+ if (id != 0) {
+ getSipCall(id)->ringingCall(event);
+ Manager::instance().peerRingingCall(id);
+ } else {
+ returnValue = -1;
+ }
+ break;
+
+ // The peer-user answers
case EXOSIP_CALL_ANSWERED: // 10
{
id = findCallIdInitial(event);
@@ -717,17 +726,6 @@ SipVoIPLink::getEvent (void)
returnValue = -1;
}
}
- case EXOSIP_CALL_RINGING: //peer call is ringing
- id = findCallIdInitial(event);
- _debug("Call is ringing [id = %d, cid = %d, did = %d]\n", id, event->cid, event->did);
- if (id != 0) {
- getSipCall(id)->ringingCall(event);
- Manager::instance().peerRingingCall(id);
- } else {
- returnValue = -1;
- }
- break;
-
case EXOSIP_CALL_REDIRECTED:
break;
@@ -748,7 +746,6 @@ SipVoIPLink::getEvent (void)
if (id != 0) {
if (Manager::instance().callCanBeClosed(id)) {
sipcall = getSipCall(id);
- if ( sipcall != NULL ) { sipcall->enable_audio = false; }
_audiortp.closeRtpSession();
}
Manager::instance().peerHungupCall(id);
--
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