Commit 93acd551 authored by Alexandre Savard's avatar Alexandre Savard

Debuging messages

parent 996021f1
......@@ -122,7 +122,7 @@ AlsaLayer::startStream(void)
void
AlsaLayer::stopStream(void)
{
_debug ("Stop ALSA streams\n");
_debug ("AlsaLayer::stopStream :: Stop ALSA streams\n");
stopCaptureStream ();
//stopPlaybackStream ();
......@@ -190,9 +190,13 @@ bool AlsaLayer::isCaptureActive(void) {
void AlsaLayer::stopCaptureStream (void)
{
int err;
if(_CaptureHandle){
snd_pcm_drop (_CaptureHandle);
err = snd_pcm_drop (_CaptureHandle);
_debug("AlsaLayer::stopCaptureStream snd_pcm_drop returned vaue : %i\n",err);
stop_capture ();
_debug("Wroking here !!!!!!!!!!!!!!!\n");
}
}
......@@ -373,7 +377,8 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
if(flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK)
{
if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
// if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
if((err = snd_pcm_open(&_PlaybackHandle, PCM_PLUGHW, SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
_debugAlsa("Error while opening playback device %s\n", pcm_p.c_str());
setErrorMessage( ALSA_PLAYBACK_DEVICE );
close_playback ();
......
......@@ -248,7 +248,8 @@ PulseLayer::processData( void )
// If the playback buffer is full, we don't overflow it; wait for it to have free space
if( pa_stream_writable_size(playback->pulseStream()) == 0 )
return;
_debug("PulseLayer::processData() \n");
writeToSpeaker();
}
}
......@@ -256,6 +257,7 @@ PulseLayer::processData( void )
void
PulseLayer::writeToSpeaker( void )
{
_debug("PulseLayer::writeToSpeaker() \n");
/** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
int urgentAvail;
/** Bytes available in the regular ringbuffer ( reserved for voice ) */
......
......@@ -114,6 +114,8 @@ Tone::genBuffer(const std::string& definition)
void
Tone::genSin(SFLDataFormat* buffer, int frequency1, int frequency2, int nb)
{
_debug("Tone::genSin :: called!!!!!!!!!!!!!!!!!! %i \n",nb);
double pi2 = 6.28318520;
double var1 = pi2 * (double)frequency1 / (double)_sampleRate;
double var2 = pi2 * (double)frequency2 / (double)_sampleRate;
......
......@@ -45,7 +45,9 @@ ToneGenerator::~ToneGenerator (void) {
void
ToneGenerator::generateSin (int lowerfreq, int higherfreq, int16* ptr, int len) const {
double var1, var2;
_debug("when is this called");
var1 = (double)2 * (double)M_PI * (double)higherfreq / (double)_sampleRate;
var2 = (double)2 * (double)M_PI * (double)lowerfreq / (double)_sampleRate;
......
......@@ -313,6 +313,7 @@ ManagerImpl::cancelCall (const CallID& id)
bool
ManagerImpl::onHoldCall(const CallID& id)
{
_debug("*************** ON HOLD ***********************************\n");
stopTone(true);
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
......@@ -335,6 +336,7 @@ ManagerImpl::onHoldCall(const CallID& id)
bool
ManagerImpl::offHoldCall(const CallID& id)
{
_debug("*************** OFF HOLD ***********************************\n");
stopTone(false);
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
......
......@@ -3,6 +3,4 @@ include $(top_srcdir)/globals.mak
noinst_LTLIBRARIES = libaudiorecorder.la
libaudiorecorder_la_SOURCES = \
audiorecord.cpp \
audiodsp.h \
audiodsp.cpp
\ No newline at end of file
audiorecord.cpp
\ No newline at end of file
......@@ -805,15 +805,7 @@ std::string
SIPVoIPLink::getCurrentCodecName()
{
_debug("SIPVoIPLink::getCurrentCodecName Ok 1 : \n");
printf("AAAAAAAAAAAAAHHHHHHH!!!!!!:: %s \n",Manager::instance().getCurrentCallId().c_str());
SIPCall *call = getSIPCall(Manager::instance().getCurrentCallId());
_debug("SIPVoIPLink::getCurrentCodecName Ok 2 : \n");
printf("OOOOOOOUUUUUUPPPPPPPPPPSSSSSSSSSSS!!!!!!:: %s \n",call->getCallId().c_str());
SIPCall *call = getSIPCall(Manager::instance().getCurrentCallId());
AudioCodec *ac = call->getCodecMap().getCodec(call->getAudioCodec());
......
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