From 921eec02b32823023156108e8c3a066e61583768 Mon Sep 17 00:00:00 2001
From: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
Date: Tue, 4 Sep 2007 17:14:02 -0400
Subject: [PATCH] IAX: Fix hangup SEGFAULTing the iax sound subsys + Add
samplerate conversion to Mic output (still non-working)
Adding samplerate conversion and sound output fixed the problems of the
Jitter Buffer. It seems it updates the timestamps and causes less empty
packets crashing the system.
* Removed some debugging output.
* Cleanups
---
src/iaxvoiplink.cpp | 211 +++++++++++++++++++++++++++++---------------
src/iaxvoiplink.h | 7 ++
2 files changed, 145 insertions(+), 73 deletions(-)
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 481e10c4e4..68897ee96f 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -177,46 +177,108 @@ IAXVoIPLink::getEvent()
continue;
}
- _debug ("Receive IAX Event: %d (0x%x)\n", event->etype, event->etype);
+ //_debug ("Receive IAX Event: %d (0x%x)\n", event->etype, event->etype);
call = iaxFindCallBySession(event->session);
if (call) {
- _debug(" - We've got an associated call, handle call event\n");
+ //_debug(" - We've got an associated call, handle call event\n");
iaxHandleCallEvent(event, call);
} else if (event->session && event->session == _regSession) {
-
- _debug(" - We've got an associated REGISTRATION session, handle registration process\n");
+
+ //_debug(" - We've got an associated REGISTRATION session, handle registration process\n");
// in iaxclient, there is many session handling, here, only one
iaxHandleRegReply(event);
-
+
} else {
-
- _debug (" - We've got some other event, deal with them alone.\n");
+
+ //_debug (" - We've got some other event, deal with them alone.\n");
iaxHandlePrecallEvent(event);
-
+
}
-
+
iax_event_free(event);
}
- // Woah, we should do that in another thread, which will always send out stuff..
- // send sound here
- if(_currentCall && audiolayer) {
- int samples = audiolayer->canGetMic();
- if (samples != 0) {
- //int datalen = audiolayer->getMic(_sendDataEncoded, samples);
- //_debug("iax_send_voice(%p, %d, ,%d, %d)\n", _currentCall->getSession(), _currentCall->getFormat(), datalen, samples);
- //if ( iax_send_voice(_currentCall->getSession(), _currentCall->getFormat(), (char*)_sendDataEncoded, datalen, samples) == -1) {
- // // error sending voice
- //}
+ _mutexIAX.leaveMutex();
+
+ // Send sound here
+ if (_currentCall && audiolayer) {
+ AudioCodec* audiocodec = _currentCall->getAudioCodec();
+
+ // 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);
+
+ // available bytes inside ringbuffer
+ int availBytesFromMic = audiolayer->canGetMic();
+
+ // 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);
+ int16* toIAX = NULL;
+ if (audiolayer->getSampleRate() != audiocodec->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
+
+ double factord = (double) audiocodec->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; /* More data to come */
+
+ src_process(_src_state_mic, &src_data);
+
+ nbSample = src_data.output_frames_gen;
+
+ src_float_to_short_array (_floatBuffer8000, _intBuffer8000, nbSample);
+ toIAX = _intBuffer8000;
+ } 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
+ }
+
+ if ( nbSample < (IAX__20S_8KHZ_MAX - 10) ) { // if only 10 is missing, it's ok
+ // fill end with 0...
+ //_debug("begin: %p, nbSample: %d\n", toIAX, nbSample);
+ //_debug("has to fill: %d chars at %p\n", (IAX__20S_8KHZ_MAX-nbSample)*sizeof(int16), toIAX + nbSample);
+ memset(toIAX + nbSample, 0, (IAX__20S_8KHZ_MAX-nbSample)*sizeof(int16));
+ nbSample = IAX__20S_8KHZ_MAX;
}
+ //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toIAX[0], toIAX[1], toIAX[2]);
+
+ // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
+ int compSize = audiocodec->codecEncode(_sendDataEncoded, toIAX, 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) {
+ _debug("IAX: Error sending voice data.\n");
+ }
+ }
+ _mutexIAX.leaveMutex();
}
- // unlock mutex here
- _mutexIAX.leaveMutex();
//iaxRefreshRegistrations();
// thread wait 5 millisecond
@@ -498,7 +560,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
case IAX_EVENT_HANGUP:
Manager::instance().peerHungupCall(id);
if (Manager::instance().isCurrentCall(id)) {
- _currentCall = 0;
+ _currentCall = NULL;
audiolayer->stopStream();
// stop audio
}
@@ -509,16 +571,14 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
Manager::instance().peerHungupCall(id);
if (Manager::instance().isCurrentCall(id)) {
// stop audio
- _currentCall = 0;
+ _currentCall = NULL;
audiolayer->stopStream();
}
removeCall(id);
break;
case IAX_EVENT_ACCEPT:
- // accept
- //
-
+ // Call accepted over there by the computer, not the user yet.
if (event->ies.format) {
call->setFormat(event->ies.format);
}
@@ -540,8 +600,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
audiolayer->startStream();
// start audio here?
} else {
- // deja connecté
- // ?
+ // deja connecté ?
}
break;
@@ -554,26 +613,62 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
break;
case IAX_EVENT_VOICE:
+ iaxHandleVoiceEvent(event, call);
+ break;
+
+ case IAX_EVENT_TEXT:
+ break;
+
+ case IAX_EVENT_RINGA:
+ 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: %d\n", event->etype);
+
+ }
+}
+
+
+/* Handle audio event, VOICE packet received */
+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 (audiolayer) {
AudioCodec* audiocodec = call->getAudioCodec();
- if (!audiocodec) {
- // libiax2 stores the voiceformat in the 'subclass' element.
- if (event->subclass) {
- // Set the format, with the first voice packet
- call->setFormat(event->subclass);
- audiocodec = call->getAudioCodec();
- } else {
- // Send a VNAK, because they sent a Mini packet before
- // a full VOICE packet (with the format inside)
- _debug("IAX: sending VNAK, received mini packet before full VOICE packet.");
- iax_vnak(event->session);
- break;
- }
+ // 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);
+ audiocodec = call->getAudioCodec();
}
-
_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
+
unsigned char* data = (unsigned char*)event->data;
unsigned int size = event->datalen;
@@ -582,14 +677,11 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
size = IAX__20S_8KHZ_MAX;
}
- // On pourrait ajuster le codec dynamiquement ici, comme c'est fait dans SIP.
- // à moins que IAX ne permette pas de changer le codec à chaque paquet.
-
int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
int nbInt16 = expandedSize/sizeof(int16);
if (nbInt16 > IAX__20S_8KHZ_MAX) {
- _debug("We have decoded a IAX VOICE packet larger than expected: %s VS %s. Cropping.\n", 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;
}
@@ -635,33 +727,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
} else {
_debug("IAX: incoming audio, but no sound card open");
}
- break;
-
- case IAX_EVENT_TEXT:
- break;
-
- case IAX_EVENT_RINGA:
- 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: %d\n", event->etype);
-
- }
}
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 481f9f9083..4910b5702f 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -105,6 +105,13 @@ private:
*/
void iaxHandleCallEvent(iax_event* event, IAXCall* call);
+ /**
+ * Handle the VOICE events specifically
+ * @param event The iax_event containing the IAX_EVENT_VOICE
+ * @param call The associated IAXCall
+ */
+ void iaxHandleVoiceEvent(iax_event* event, IAXCall* call);
+
/**
* Handle IAX Registration Reply event
* @param event An iax_event pointer
--
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