From 6745c09a7e220b986b8940ea234b5daf136186a9 Mon Sep 17 00:00:00 2001
From: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
Date: Wed, 22 Oct 2008 17:45:37 -0400
Subject: [PATCH] remove warnings in iaxvoiplink, iaxcall
---
src/audio/alsalayer.cpp | 4 +++
src/audio/pulselayer.cpp | 1 +
src/iaxcall.cpp | 10 ++++---
src/iaxvoiplink.cpp | 63 +++++++++++++++++++++++++---------------
src/iaxvoiplink.h | 4 +--
5 files changed, 52 insertions(+), 30 deletions(-)
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index 72673ec566..b379daaad4 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -195,6 +195,10 @@ AlsaLayer::putUrgent(void* buffer, int toCopy)
int
AlsaLayer::canGetMic()
{
+
+
+ _debug("alsa\n");
+
int avail;
if ( _CaptureHandle ) {
avail = snd_pcm_avail_update( _CaptureHandle );
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index a2da80d467..59b1d4b95f 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -199,6 +199,7 @@ PulseLayer::putUrgent(void* buffer, int toCopy)
int
PulseLayer::canGetMic()
{
+ _debug("pulseaudio");
if( record )
return _micRingBuffer.AvailForGet();
else
diff --git a/src/iaxcall.cpp b/src/iaxcall.cpp
index 04c5928152..05327d77d6 100644
--- a/src/iaxcall.cpp
+++ b/src/iaxcall.cpp
@@ -55,9 +55,11 @@ IAXCall::setFormat(int format)
int
IAXCall::getSupportedFormat()
{
- CodecOrder map = getCodecMap().getActiveCodecs();
- int format = 0;
- int iter;
+ CodecOrder map;
+ int format = 0;
+ unsigned int iter;
+
+ map = getCodecMap().getActiveCodecs();
for(iter=0 ; iter < map.size() ; iter++){
switch(map[iter]) {
@@ -85,7 +87,7 @@ IAXCall::getFirstMatchingFormat(int needles)
{
CodecOrder map = getCodecMap().getActiveCodecs();
int format = 0;
- int iter;
+ unsigned int iter;
for(iter=0 ; iter < map.size() ; iter++) {
switch(map[iter]) {
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 7e295da797..73b42a0492 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -216,16 +216,18 @@ IAXVoIPLink::getEvent()
void
IAXVoIPLink::sendAudioFromMic(void)
{
- IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
-
- if (!currentCall) {
- // Let's mind our own business.
- return;
- }
-
- if( currentCall -> getAudioCodec() < 0 )
- return;
+ int maxBytesToGet, availBytesFromMic, bytesAvail, nbSample, compSize;
+ AudioCodec *ac;
+
+ IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
+
+ if (!currentCall) {
+ // Let's mind our own business.
+ return;
+ }
+ if( currentCall -> getAudioCodec() < 0 )
+ return;
// Just make sure the currentCall is in state to receive audio right now.
//_debug("Here we get: connectionState: %d state: %d \n",
@@ -237,7 +239,7 @@ IAXVoIPLink::sendAudioFromMic(void)
return;
}
- AudioCodec* ac = currentCall -> getCodecMap().getCodec( currentCall -> getAudioCodec() );
+ ac = currentCall -> getCodecMap().getCodec( currentCall -> getAudioCodec() );
if (!ac) {
// Audio codec still not determined.
if (audiolayer) {
@@ -250,12 +252,13 @@ IAXVoIPLink::sendAudioFromMic(void)
// Send sound here
if (audiolayer) {
+ _debug("hum");
// 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()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
+ maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
// available bytes inside ringbuffer
- int availBytesFromMic = audiolayer->canGetMic();
+ availBytesFromMic = audiolayer->canGetMic();
if (availBytesFromMic < maxBytesToGet) {
// We need packets full!
@@ -263,17 +266,17 @@ IAXVoIPLink::sendAudioFromMic(void)
}
// take the lowest
- int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+ 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( micData, bytesAvail ) / sizeof(SFLDataFormat);
+ 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( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
+ compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
// Send it out!
_mutexIAX.enterMutex();
@@ -482,6 +485,8 @@ IAXVoIPLink::transfer(const CallID& id, const std::string& to)
iax_transfer(call->getSession(), callto);
_mutexIAX.leaveMutex();
+ return true;
+
// should we remove it?
// removeCall(id);
}
@@ -498,6 +503,8 @@ IAXVoIPLink::refuse(const CallID& id)
iax_reject(call->getSession(), (char*) reason.c_str());
_mutexIAX.leaveMutex();
removeCall(id);
+
+ return true;
}
bool
@@ -510,6 +517,8 @@ IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
_mutexIAX.enterMutex();
iax_send_dtmf(call->getSession(), code);
_mutexIAX.leaveMutex();
+
+ return true;
}
@@ -575,7 +584,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
// note activity?
//
CallID id = call->getCallId();
- int16* output = 0; // for audio output
switch(event->etype) {
case IAX_EVENT_HANGUP:
@@ -675,6 +683,13 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
void
IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
{
+
+
+ unsigned char *data;
+ unsigned int size, max, nbInt16;
+ int expandedSize, nbSample;
+ AudioCodec *ac;
+
// If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
// were triggered
if (!event->datalen) {
@@ -691,28 +706,28 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
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() );
+ ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
- unsigned char* data = (unsigned char*)event->data;
- unsigned int size = event->datalen;
+ data = (unsigned char*)event->data;
+ size = event->datalen;
// Decode data with relevant codec
- int max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
+ max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
if (size > max) {
_debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, max);
size = max;
}
- int expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
- int nbInt16 = expandedSize/sizeof(int16);
+ expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
+ nbInt16 = expandedSize/sizeof(int16);
if (nbInt16 > max) {
- _debug("We have decoded an IAX VOICE packet larger than expected: %s VS %s. Cropping.\n", nbInt16, max);
+ _debug("We have decoded an IAX VOICE packet larger than expected: %i VS %i. Cropping.\n", nbInt16, max);
nbInt16 = max;
}
- int nbSample = nbInt16;
+ nbSample = nbInt16;
// resample
nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 9f1525afe2..770bd82d5d 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -124,7 +124,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- bool cancel(const CallID& id) { return false; }
+ bool cancel(const CallID& id UNUSED ) { return false; }
/**
* Put a call on hold
@@ -168,7 +168,7 @@ class IAXVoIPLink : public VoIPLink
*/
bool carryingDTMFdigits(const CallID& id, char code);
- bool sendMessage(const std::string& to, const std::string& body) { return false; }
+ bool sendMessage(const std::string& to UNUSED, const std::string& body UNUSED) { return false; }
bool isContactPresenceSupported() { return false; }
--
GitLab