From eface0d72362f6bf6060a0ec3faffca4194300ac Mon Sep 17 00:00:00 2001
From: Yun Liu <yun@yun.(none)>
Date: Thu, 2 Oct 2008 09:34:16 -0400
Subject: [PATCH] Remove warnings continued
---
src/account.cpp | 4 +-
src/account.h | 17 ++----
src/audio/alsalayer.cpp | 109 ++++++++++++++++++----------------
src/audio/alsalayer.h | 6 ++
src/audio/audiodevice.cpp | 2 +-
src/audio/audiofile.h | 17 ++----
src/audio/audiolayer.h | 37 ++++--------
src/audio/audioloop.h | 23 +++----
src/audio/audiortp.cpp | 12 ++--
src/audio/audiortp.h | 13 ++++
src/audio/audiostream.cpp | 8 +--
src/audio/audiostream.h | 7 +++
src/audio/codecDescriptor.cpp | 16 +++--
src/audio/dtmf.cpp | 4 +-
src/audio/dtmfgenerator.cpp | 32 +---------
src/audio/dtmfgenerator.h | 9 ++-
src/audio/pulselayer.cpp | 21 ++++---
src/audio/pulselayer.h | 6 ++
src/audio/ringbuffer.cpp | 13 ----
src/audio/ringbuffer.h | 10 ++--
src/audio/samplecache.h | 17 ++----
src/audio/tone.cpp | 3 +-
src/audio/tonegenerator.cpp | 4 +-
src/audio/tonegenerator.h | 6 ++
src/audio/tonelist.h | 35 ++++-------
src/call.cpp | 25 +++++---
src/dbus/contactmanager.cpp | 2 +-
src/eventthread.cpp | 14 -----
src/eventthread.h | 7 ++-
src/managerimpl.cpp | 75 ++++++++++++++---------
src/managerimpl.h | 8 +++
src/samplerateconverter.cpp | 32 ++++++----
src/samplerateconverter.h | 6 ++
src/sipaccount.cpp | 5 +-
src/sipaccount.h | 6 ++
src/sipcall.cpp | 13 ++--
src/sipcall.h | 7 +++
src/sipvoiplink.cpp | 22 ++++---
src/sipvoiplink.h | 6 ++
src/useragent.cpp | 2 +-
src/useragent.h | 7 +++
41 files changed, 352 insertions(+), 316 deletions(-)
diff --git a/src/account.cpp b/src/account.cpp
index 0fc25638d0..5eedf99bd4 100644
--- a/src/account.cpp
+++ b/src/account.cpp
@@ -23,10 +23,8 @@
#include <string>
-Account::Account(const AccountID& accountID) : _accountID(accountID)
+Account::Account(const AccountID& accountID) : _accountID(accountID), _link(NULL), _enabled(false)
{
- _link = NULL;
- _enabled = false;
}
Account::~Account()
diff --git a/src/account.h b/src/account.h
index b89baf42c6..fc722ebd07 100644
--- a/src/account.h
+++ b/src/account.h
@@ -76,17 +76,6 @@ class Account{
*/
virtual ~Account();
- // copy constructor
- Account(const Account& rh):_accountID(rh._accountID), _link(rh._link), _enabled(rh._enabled){
- _debug("Account copy constructor hasn't been implemented yet. Quit! ");
- exit(0);
- }
-
- // assignment operator
- Account& operator=(const Account& rh){
- _debug("Account assignment operator hasn't been implemented yet. Quit! ");
- exit(0);
- }
/**
* Load the settings for this account.
@@ -147,6 +136,12 @@ class Account{
private:
+ // copy constructor
+ Account(const Account& rh);
+
+ // assignment operator
+ Account& operator=(const Account& rh);
+
protected:
/**
* Account ID are assign in constructor and shall not changed
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index e0278d7b13..8df372658b 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -24,11 +24,17 @@
// Constructor
AlsaLayer::AlsaLayer( ManagerImpl* manager )
: AudioLayer( manager , ALSA )
- , _PlaybackHandle(NULL)
- , _CaptureHandle(NULL)
+ , _PlaybackHandle(NULL)
+ , _CaptureHandle(NULL)
+ , _periodSize()
+ , _AsyncHandler(NULL)
+ , _audioPlugin()
+ , _inChannel()
+ , _outChannel()
+ , _defaultVolume(100)
+ , IDSoundCards()
{
_debug(" Constructor of AlsaLayer called\n");
- _defaultVolume = 100;
}
// Destructor
@@ -99,7 +105,7 @@ AlsaLayer::startStream(void)
snd_pcm_start( _CaptureHandle ) ;
snd_pcm_prepare( _PlaybackHandle );
- if( err = snd_pcm_start( _PlaybackHandle) < 0 ) _debugAlsa(" Cannot start (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_start( _PlaybackHandle)) < 0 ) _debugAlsa(" Cannot start (%s)\n", snd_strerror(err));
}
}
@@ -132,7 +138,7 @@ AlsaLayer::fillHWBuffer( void)
short s1, s2;
int periodSize = 128 ;
int frames = periodSize >> 2 ;
- _debug("frames = %d\n");
+ _debug("frames = %d\n", frames);
data = (unsigned char*)malloc(periodSize);
for(l1 = 0; l1 < 100; l1++) {
@@ -286,7 +292,7 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
int dir = 0;
snd_pcm_uframes_t period_size_in = getFrameSize() * getSampleRate() / 1000 * 2 ;
snd_pcm_uframes_t buffer_size_in = period_size_in * 4 ;
- snd_pcm_uframes_t threshold = 1024 ;
+ //snd_pcm_uframes_t threshold = 1024 ;
snd_pcm_uframes_t period_size_out = getFrameSize() * getSampleRate() / 1000 * 2;//1024 ;
snd_pcm_uframes_t buffer_size_out = period_size_out * 4 ;
@@ -296,41 +302,41 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
if(flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE)
{
_debugAlsa("Opening capture device %s\n", pcm_c.c_str());
- if(err = snd_pcm_open(&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0) < 0){
+ if( (err = snd_pcm_open(&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0){
_debugAlsa("Error while opening capture device %s\n", pcm_c.c_str());
setErrorMessage( ALSA_CAPTURE_DEVICE );
return false;
}
- if( err = snd_pcm_hw_params_malloc( &hwParams ) < 0 ) {
+ if( (err = snd_pcm_hw_params_malloc( &hwParams )) < 0 ) {
_debugAlsa(" Cannot allocate hardware parameter structure (%s)\n", snd_strerror(err));
return false;
}
- if( err = snd_pcm_hw_params_any(_CaptureHandle, hwParams) < 0) _debugAlsa(" Cannot initialize hardware parameter structure (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_access( _CaptureHandle, hwParams, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_format( _CaptureHandle, hwParams, SND_PCM_FORMAT_S16_LE) < 0) _debugAlsa(" Cannot set sample format (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_rate_near( _CaptureHandle, hwParams, &rate_in, &dir) < 0) _debugAlsa(" Cannot set sample rate (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_channels( _CaptureHandle, hwParams, 1) < 0) _debugAlsa(" Cannot set channel count (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_period_time_near( _CaptureHandle, hwParams, &period_time , &dir) < 0) _debugAlsa(" Cannot set period time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_buffer_time_near( _CaptureHandle, hwParams, &buffer_time , &dir) < 0) _debugAlsa(" Cannot set buffer time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_period_size( hwParams, &period_size_in , &dir) < 0) _debugAlsa(" Cannot get period size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_buffer_size( hwParams, &buffer_size_in ) < 0) _debugAlsa(" Cannot get buffer size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params( _CaptureHandle, hwParams ) < 0) _debugAlsa(" Cannot set hw parameters (%s)\n", snd_strerror(err));
- _debug("buffer size = %d\n" , period_size_in);
- _debug("period size = %d\n" , buffer_size_in);
+ if( (err = snd_pcm_hw_params_any(_CaptureHandle, hwParams)) < 0) _debugAlsa(" Cannot initialize hardware parameter structure (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_access( _CaptureHandle, hwParams, SND_PCM_ACCESS_MMAP_INTERLEAVED)) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_format( _CaptureHandle, hwParams, SND_PCM_FORMAT_S16_LE)) < 0) _debugAlsa(" Cannot set sample format (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_rate_near( _CaptureHandle, hwParams, &rate_in, &dir) < 0)) _debugAlsa(" Cannot set sample rate (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_channels( _CaptureHandle, hwParams, 1)) < 0) _debugAlsa(" Cannot set channel count (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_period_time_near( _CaptureHandle, hwParams, &period_time , &dir)) < 0) _debugAlsa(" Cannot set period time (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_buffer_time_near( _CaptureHandle, hwParams, &buffer_time , &dir)) < 0) _debugAlsa(" Cannot set buffer time (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_get_period_size( hwParams, &period_size_in , &dir) < 0)) _debugAlsa(" Cannot get period size (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_get_buffer_size( hwParams, &buffer_size_in ) < 0)) _debugAlsa(" Cannot get buffer size (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params( _CaptureHandle, hwParams ) < 0)) _debugAlsa(" Cannot set hw parameters (%s)\n", snd_strerror(err));
+ _debug("buffer size = %d\n" , (int) period_size_in);
+ _debug("period size = %d\n" , (int) buffer_size_in);
snd_pcm_hw_params_free( hwParams );
snd_pcm_uframes_t val ;
snd_pcm_sw_params_malloc( &swparams );
snd_pcm_sw_params_current( _CaptureHandle, swparams );
- if( err = snd_pcm_sw_params_set_start_threshold( _CaptureHandle, swparams, period_size_out) < 0 ) _debugAlsa(" Cannot set start threshold (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_sw_params_set_start_threshold( _CaptureHandle, swparams, period_size_out)) < 0 ) _debugAlsa(" Cannot set start threshold (%s)\n", snd_strerror(err));
snd_pcm_sw_params_get_start_threshold( swparams , &val);
- _debug("Start threshold = %d\n" ,val);
- if( err = snd_pcm_sw_params_set_avail_min( _CaptureHandle, swparams, period_size_out) < 0) _debugAlsa(" Cannot set min avail (%s)\n" , snd_strerror(err));
+ _debug("Start threshold = %d\n" ,(int)val);
+ if( (err = snd_pcm_sw_params_set_avail_min( _CaptureHandle, swparams, period_size_out)) < 0) _debugAlsa(" Cannot set min avail (%s)\n" , snd_strerror(err));
snd_pcm_sw_params_get_avail_min( swparams , &val);
- _debug("Min available = %d\n" ,val);
- if( err = snd_pcm_sw_params( _CaptureHandle, swparams ) < 0 ) _debugAlsa(" Cannot set sw parameters (%s)\n", snd_strerror(err));
+ _debug("Min available = %d\n" ,(int)val);
+ if( (err = snd_pcm_sw_params( _CaptureHandle, swparams )) < 0 ) _debugAlsa(" Cannot set sw parameters (%s)\n", snd_strerror(err));
snd_pcm_sw_params_free( swparams );
deviceClosed = false;
}
@@ -339,26 +345,26 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
{
_debugAlsa(" Opening playback device %s\n", pcm_p.c_str());
- if(err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) < 0){
+ if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK )) < 0){
_debugAlsa("Error while opening playback device %s\n", pcm_p.c_str());
setErrorMessage( ALSA_PLAYBACK_DEVICE );
return false;
}
- if( err = snd_pcm_hw_params_malloc( &hwParams ) < 0 ) {
+ if( (err = snd_pcm_hw_params_malloc( &hwParams )) < 0 ) {
_debugAlsa(" Cannot allocate hardware parameter structure (%s)\n", snd_strerror(err));
return false;
}
- if( err = snd_pcm_hw_params_any( _PlaybackHandle,hwParams) < 0) _debugAlsa(" Cannot initialize hardware parameter structure (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_access( _PlaybackHandle, hwParams, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
+ if(( err = snd_pcm_hw_params_any( _PlaybackHandle,hwParams)) < 0) _debugAlsa(" Cannot initialize hardware parameter structure (%s)\n", snd_strerror(err));
+ if(( err = snd_pcm_hw_params_set_access( _PlaybackHandle, hwParams, SND_PCM_ACCESS_MMAP_INTERLEAVED)) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
//if( err = snd_pcm_hw_params_set_access( _PlaybackHandle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_format( _PlaybackHandle, hwParams, SND_PCM_FORMAT_S16_LE) < 0) _debugAlsa(" Cannot set sample format (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_rate( _PlaybackHandle, hwParams, rate_out, dir) < 0) _debugAlsa(" Cannot set sample rate (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_channels( _PlaybackHandle, hwParams, 1) < 0) _debugAlsa(" Cannot set channel count (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_buffer_time_near( _PlaybackHandle, hwParams, &buffer_time , &dir) < 0) _debugAlsa(" Cannot set buffer time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_period_time_near( _PlaybackHandle, hwParams, &period_time , &dir) < 0) _debugAlsa(" Cannot set period time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_period_size( hwParams, &_periodSize , &dir) < 0) _debugAlsa(" Cannot get period size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_buffer_size( hwParams, &buffer_size_out ) < 0) _debugAlsa(" Cannot get buffer size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params( _PlaybackHandle, hwParams ) < 0) _debugAlsa(" Cannot set hw parameters (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_format( _PlaybackHandle, hwParams, SND_PCM_FORMAT_S16_LE)) < 0) _debugAlsa(" Cannot set sample format (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_rate( _PlaybackHandle, hwParams, rate_out, dir)) < 0) _debugAlsa(" Cannot set sample rate (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_channels( _PlaybackHandle, hwParams, 1)) < 0) _debugAlsa(" Cannot set channel count (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_buffer_time_near( _PlaybackHandle, hwParams, &buffer_time , &dir)) < 0) _debugAlsa(" Cannot set buffer time (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_set_period_time_near( _PlaybackHandle, hwParams, &period_time , &dir)) < 0) _debugAlsa(" Cannot set period time (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_get_period_size( hwParams, &_periodSize , &dir)) < 0) _debugAlsa(" Cannot get period size (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params_get_buffer_size( hwParams, &buffer_size_out )) < 0) _debugAlsa(" Cannot get buffer size (%s)\n", snd_strerror(err));
+ if( (err = snd_pcm_hw_params( _PlaybackHandle, hwParams )) < 0) _debugAlsa(" Cannot set hw parameters (%s)\n", snd_strerror(err));
snd_pcm_hw_params_free( hwParams );
@@ -367,23 +373,23 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
snd_pcm_sw_params_malloc( &swparams );
snd_pcm_sw_params_current( _PlaybackHandle, swparams );
- if( err = snd_pcm_sw_params_set_start_threshold( _PlaybackHandle, swparams, period_size_out) < 0 ) _debugAlsa(" Cannot set start threshold (%s)\n", snd_strerror(err));
+ if((err = snd_pcm_sw_params_set_start_threshold( _PlaybackHandle, swparams, period_size_out)) < 0 ) _debugAlsa(" Cannot set start threshold (%s)\n", snd_strerror(err));
snd_pcm_sw_params_get_start_threshold( swparams , &val);
- _debug("Start threshold = %d\n" ,val);
+ _debug("Start threshold = %d\n" ,(int)val);
//if( err = snd_pcm_sw_params_set_stop_threshold( _PlaybackHandle, swparams, buffer_size_out ) < 0 ) _debugAlsa(" Cannot set stop threshold (%s)\n", snd_strerror(err));
//snd_pcm_sw_params_get_stop_threshold( swparams , &val);
//_debug("Stop threshold = %d\n" ,val);
- if( err = snd_pcm_sw_params_set_avail_min( _PlaybackHandle, swparams, period_size_out) < 0) _debugAlsa(" Cannot set min avail (%s)\n" , snd_strerror(err));
- if( err = snd_pcm_sw_params_set_xfer_align( _PlaybackHandle, swparams, 1) < 0) _debugAlsa(" Cannot set xfer align (%s)\n" , snd_strerror(err));
+ if((err = snd_pcm_sw_params_set_avail_min( _PlaybackHandle, swparams, period_size_out)) < 0) _debugAlsa(" Cannot set min avail (%s)\n" , snd_strerror(err));
+ if((err = snd_pcm_sw_params_set_xfer_align( _PlaybackHandle, swparams, 1)) < 0) _debugAlsa(" Cannot set xfer align (%s)\n" , snd_strerror(err));
snd_pcm_sw_params_get_avail_min( swparams , &val);
- _debug("Min available = %d\n" ,val);
+ _debug("Min available = %d\n" ,(int)val);
//if( err = snd_pcm_sw_params_set_silence_threshold( _PlaybackHandle, swparams, period_size_out) < 0) _debugAlsa(" Cannot set silence threshold (%s)\n" , snd_strerror(err));
//snd_pcm_sw_params_get_silence_threshold( swparams , &val);
// _debug("Silence threshold = %d\n" ,val);
- if( err = snd_pcm_sw_params( _PlaybackHandle, swparams ) < 0 ) _debugAlsa(" Cannot set sw parameters (%s)\n", snd_strerror(err));
+ if(( err = snd_pcm_sw_params( _PlaybackHandle, swparams )) < 0 ) _debugAlsa(" Cannot set sw parameters (%s)\n", snd_strerror(err));
snd_pcm_sw_params_free( swparams );
- if ( err = snd_async_add_pcm_handler( &_AsyncHandler, _PlaybackHandle , AlsaCallBack, this ) < 0) _debugAlsa(" Unable to install the async callback handler (%s)\n", snd_strerror(err));
+ if (( err = snd_async_add_pcm_handler( &_AsyncHandler, _PlaybackHandle , AlsaCallBack, this ) < 0)) _debugAlsa(" Unable to install the async callback handler (%s)\n", snd_strerror(err));
deviceClosed = false;
}
//fillHWBuffer();
@@ -421,8 +427,8 @@ AlsaLayer::write(void* buffer, int length)
break;
}
- if( err >=0 && err < frames )
- _debugAlsa("Short write : %d out of %d\n", err , frames);
+ if( ( err >=0 ) && ( err < (int)frames ) )
+ _debugAlsa("Short write : %d out of %d\n", err , (int)frames);
return ( err > 0 )? err : 0 ;
}
@@ -439,7 +445,7 @@ AlsaLayer::read( void* buffer, int toCopy)
snd_pcm_start( _CaptureHandle );
}
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _CaptureHandle, toCopy );
- if( err = snd_pcm_mmap_readi( _CaptureHandle, buffer, frames) < 0 ) {
+ if(( err = snd_pcm_mmap_readi( _CaptureHandle, buffer, frames)) < 0 ) {
switch(err){
case EPERM:
_debugAlsa(" Capture EPERM (%s)\n", snd_strerror(err));
@@ -470,7 +476,9 @@ AlsaLayer::read( void* buffer, int toCopy)
int
AlsaLayer::putInCache( char code, void *buffer, int toCopy )
-{}
+{
+ return 1;
+}
void
AlsaLayer::handle_xrun_capture( void )
@@ -497,7 +505,7 @@ AlsaLayer::handle_xrun_playback( void )
snd_pcm_status_t* status;
snd_pcm_status_alloca( &status );
- if( state = snd_pcm_status( _PlaybackHandle, status ) < 0 ) _debugAlsa(" Error: Cannot get playback handle status (%s)\n" , snd_strerror( state ) );
+ if( (state = snd_pcm_status( _PlaybackHandle, status )) < 0 ) _debugAlsa(" Error: Cannot get playback handle status (%s)\n" , snd_strerror( state ) );
else
{
state = snd_pcm_status_get_state( status );
@@ -541,7 +549,6 @@ AlsaLayer::getSoundCardsInfo( int stream )
snd_pcm_info_alloca( &pcminfo );
int numCard = -1 ;
- int err;
std::string description;
if(snd_card_next( &numCard ) < 0 || numCard < 0)
@@ -624,7 +631,7 @@ AlsaLayer::soundCardIndexExist( int card , int stream )
int
AlsaLayer::soundCardGetIndex( std::string description )
{
- int i;
+ unsigned int i;
for( i = 0 ; i < IDSoundCards.size() ; i++ )
{
HwIDPair p = IDSoundCards[i];
diff --git a/src/audio/alsalayer.h b/src/audio/alsalayer.h
index fb5def1fa6..78a51988e9 100644
--- a/src/audio/alsalayer.h
+++ b/src/audio/alsalayer.h
@@ -203,6 +203,12 @@ class AlsaLayer : public AudioLayer {
void setPlaybackVolume( double volume );
private:
+
+ // Copy Constructor
+ AlsaLayer(const AlsaLayer& rh);
+
+ // Assignment Operator
+ AlsaLayer& operator=( const AlsaLayer& rh);
/**
* Drop the pending frames and close the capture device
diff --git a/src/audio/audiodevice.cpp b/src/audio/audiodevice.cpp
index a4672da846..f82ce075a2 100644
--- a/src/audio/audiodevice.cpp
+++ b/src/audio/audiodevice.cpp
@@ -23,7 +23,7 @@
const double AudioDevice::DEFAULT_RATE = 8000.0;
AudioDevice::AudioDevice(int id, const std::string& name) :
- _id(id), _name(name)
+ _id(id), _name(name), _rate(DEFAULT_RATE)
{
_rate = DEFAULT_RATE;
}
diff --git a/src/audio/audiofile.h b/src/audio/audiofile.h
index 8911f41d5d..a016289ed6 100644
--- a/src/audio/audiofile.h
+++ b/src/audio/audiofile.h
@@ -45,17 +45,6 @@ public:
*/
~AudioFile();
- // Copy Constructor
- AudioFile(const AudioFile& rh):_filename(rh._filename), _codec(rh._codec), _start(rh._start) {
- _debug("AudioFilie copy constructor hasn't been implemented yet. Quit!");
- exit(0);
- }
-
- // Assignment Operator
- AudioFile& operator=( const AudioFile& rh){
- _debug("AudioFilie assignment operator hasn't been implemented yet. Quit!");
- exit(0);
- }
/**
* Load a sound file in memory
@@ -84,6 +73,12 @@ public:
bool isStarted() { return _start; }
private:
+ // Copy Constructor
+ AudioFile(const AudioFile& rh);
+
+ // Assignment Operator
+ AudioFile& operator=( const AudioFile& rh);
+
/** The absolute path to the sound file */
std::string _filename;
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index eb8025e3da..b0b765e446 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -46,6 +46,15 @@
*/
class AudioLayer {
+
+ private:
+
+ //copy constructor
+ AudioLayer(const AudioLayer& rh);
+
+ // assignment operator
+ AudioLayer& operator=(const AudioLayer& rh);
+
public:
/**
* Constructor
@@ -69,37 +78,11 @@ class AudioLayer {
}
- //copy constructor
- AudioLayer(const AudioLayer& rh)
- : _layerType( rh._layerType )
- , _manager(rh._manager)
- , _urgentBuffer( rh._urgentBuffer )
- , _talk ( rh._talk )
- , deviceClosed ( rh.deviceClosed )
- , _indexIn ( rh._indexIn )
- , _indexOut ( rh._indexOut )
- , _sampleRate ( rh._sampleRate )
- , _frameSize ( rh._frameSize )
- , _inChannel( rh._inChannel )
- , _outChannel ( rh._outChannel )
- , _errorMessage ( rh._errorMessage )
- , _mutex (rh._mutex)
- {
- _debug("AudioLayer copy constructor hasn't been implemented yet. Quit!");
- exit(0);
- }
-
- // assignment operator
- AudioLayer& operator=(const AudioLayer& rh)
- {
- _debug("AudioLayer assignment operator hasn't been implemented yet. Quit!");
- exit(0);
- }
/**
* Destructor
*/
- ~AudioLayer(void){}
+ virtual ~AudioLayer(void){}
virtual void closeLayer( void ) = 0;
diff --git a/src/audio/audioloop.h b/src/audio/audioloop.h
index 98114e9002..e5f7984100 100644
--- a/src/audio/audioloop.h
+++ b/src/audio/audioloop.h
@@ -42,18 +42,6 @@ public:
*/
virtual ~AudioLoop();
- // Copy Constructor
- AudioLoop(const AudioLoop& rh):_buffer(rh._buffer),_size(rh._size), _pos(rh._pos), _sampleRate(rh._sampleRate) {
- _debug("AudioLoop copy constructor hasn't been implemented yet. Quit!");
- exit(0);
- }
-
- // Assignment Operator
- AudioLoop& operator=( const AudioLoop& rh){
- _debug("AudioLoop assignment operator hasn't been implemented yet. Quit!");
- exit(0);
- }
-
/**
* Get the next fragment of the tone
* the function change the intern position, and will loop
@@ -76,7 +64,6 @@ public:
unsigned int getSize() { return _size; }
-
protected:
/** The data buffer */
SFLDataFormat* _buffer;
@@ -88,7 +75,15 @@ protected:
int _pos;
/** Sample rate */
- int _sampleRate;
+ int _sampleRate;
+
+private:
+
+ // Copy Constructor
+ AudioLoop(const AudioLoop& rh);
+
+ // Assignment Operator
+ AudioLoop& operator=( const AudioLoop& rh);
};
#endif // __AUDIOLOOP_H__
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 2c7da7d7e2..c69828e918 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -42,9 +42,8 @@
////////////////////////////////////////////////////////////////////////////////
// AudioRtp
////////////////////////////////////////////////////////////////////////////////
-AudioRtp::AudioRtp ()
+AudioRtp::AudioRtp() :_RTXThread(0), _symmetric(), _threadMutex()
{
- _RTXThread = 0;
}
AudioRtp::~AudioRtp (void) {
@@ -94,12 +93,11 @@ AudioRtp::closeRtpSession () {
////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
-AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
+AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time(new ost::Time()), _ca(sipcall), _sessionSend(NULL), _sessionRecv(NULL), _session(NULL), _start(),
+ _sym(sym), micData(NULL), micDataConverted(NULL), micDataEncoded(NULL), spkrDataDecoded(NULL), spkrDataConverted(NULL),
+ converter(NULL), _layerSampleRate(),_codecSampleRate(), _layerFrameSize(), _audiocodec(NULL)
{
setCancel(cancelDeferred);
- time = new ost::Time();
- _ca = sipcall;
- _sym = sym;
// AudioRtpRTX should be close if we change sample rate
// TODO: Change bind address according to user settings.
// TODO: this should be the local ip not the external (router) IP
@@ -324,7 +322,7 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
//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) {
+ if ((unsigned int)nbInt16 > max) {
_debug("We have decoded an RTP packet larger than expected: %d VS %d. Cropping.\n", nbInt16, max);
nbInt16=max;
}
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index a0cb003bf3..030840f94e 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -65,6 +65,13 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
virtual void run ();
private:
+
+ // copy constructor
+ AudioRtpRTX(const AudioRtpRTX& rh);
+
+ // assignment operator
+ AudioRtpRTX& operator=(const AudioRtpRTX& rh);
+
/** A SIP call */
SIPCall* _ca;
@@ -169,6 +176,12 @@ class AudioRtp {
void closeRtpSession( void );
private:
+ // copy constructor
+ AudioRtp(const AudioRtp& rh);
+
+ // assignment operator
+ AudioRtp& operator=(const AudioRtp& rh);
+
/** The RTP thread */
AudioRtpRTX* _RTXThread;
diff --git a/src/audio/audiostream.cpp b/src/audio/audiostream.cpp
index 55478be095..9c8638e4a8 100644
--- a/src/audio/audiostream.cpp
+++ b/src/audio/audiostream.cpp
@@ -22,17 +22,15 @@
static pa_channel_map channel_map ;
AudioStream::AudioStream( pa_context* context, int type, std::string desc, double vol )
+ : _audiostream (createStream( context )), _streamType(type), _streamDescription(desc), flag(PA_STREAM_AUTO_TIMING_UPDATE), sample_spec(), _volume()
{
- _streamType = type;
- _streamDescription = desc;
sample_spec.format = PA_SAMPLE_S16LE;
sample_spec.rate = 44100;
sample_spec.channels = 1;
channel_map.channels = 1;
- flag = PA_STREAM_AUTO_TIMING_UPDATE ;
pa_cvolume_set( &_volume , 1 , PA_VOLUME_MUTED ) ; // * vol / 100 ;
- _audiostream = createStream( context );
+ //_audiostream = createStream( context );
}
AudioStream::~AudioStream()
@@ -76,7 +74,7 @@ AudioStream::stream_state_callback( pa_stream* s, void* user_data )
AudioStream::createStream( pa_context* c )
{
pa_stream* s;
- pa_cvolume cv;
+ //pa_cvolume cv;
assert(pa_sample_spec_valid(&sample_spec));
assert(pa_channel_map_valid(&channel_map));
diff --git a/src/audio/audiostream.h b/src/audio/audiostream.h
index 7645f2bd51..121b87ad9c 100644
--- a/src/audio/audiostream.h
+++ b/src/audio/audiostream.h
@@ -95,6 +95,13 @@ class AudioStream {
pa_cvolume getVolume( void ) { return _volume; }
private:
+
+ // Copy Constructor
+ AudioStream(const AudioStream& rh);
+
+ // Assignment Operator
+ AudioStream& operator=( const AudioStream& rh);
+
/**
* Create the audio stream into the given context
* @param c The pulseaudio context
diff --git a/src/audio/codecDescriptor.cpp b/src/audio/codecDescriptor.cpp
index ba6ca1e681..04b35d547a 100644
--- a/src/audio/codecDescriptor.cpp
+++ b/src/audio/codecDescriptor.cpp
@@ -24,7 +24,7 @@
#include "codecDescriptor.h"
-CodecDescriptor::CodecDescriptor()
+CodecDescriptor::CodecDescriptor() : _CodecsMap(), _codecOrder(), _Cache(), _nbCodecs(), _CodecInMemory()
{
}
@@ -36,7 +36,7 @@ CodecDescriptor::~CodecDescriptor()
CodecDescriptor::deleteHandlePointer( void )
{
int i;
- for( i = 0 ; i < _CodecInMemory.size() ; i++)
+ for( i = 0 ; (unsigned int)i < _CodecInMemory.size() ; i++)
{
unloadCodec( _CodecInMemory[i] );
}
@@ -97,7 +97,7 @@ CodecDescriptor::getCodec(AudioCodecType payload)
CodecDescriptor::isActive(AudioCodecType payload)
{
int i;
- for(i=0 ; i < _codecOrder.size() ; i++)
+ for(i=0 ; (unsigned int)i < _codecOrder.size() ; i++)
{
if(_codecOrder[i] == payload)
return true;
@@ -164,7 +164,7 @@ CodecDescriptor::saveActiveCodecs(const std::vector<std::string>& list)
int i=0;
int payload;
size_t size = list.size();
- while( i < size )
+ while( (unsigned int)i < size )
{
payload = std::atoi(list[i].data());
if( isCodecLoaded( payload ) ) {
@@ -188,7 +188,7 @@ CodecDescriptor::scanCodecDirectory( void )
dirToScan.push_back(homeDir);
dirToScan.push_back(libDir);
- for( i = 0 ; i < dirToScan.size() ; i++ )
+ for( i = 0 ; (unsigned int)i < dirToScan.size() ; i++ )
{
std::string dirStr = dirToScan[i];
_debug("Scanning %s to find audio codecs....\n", dirStr.c_str());
@@ -196,7 +196,7 @@ CodecDescriptor::scanCodecDirectory( void )
AudioCodec* audioCodec;
if( dir ){
dirent *dirStruct;
- while( dirStruct = readdir( dir )) {
+ while( (dirStruct = readdir( dir )) ) {
tmp = dirStruct -> d_name ;
if( tmp == CURRENT_DIR || tmp == PARENT_DIR){}
else{
@@ -241,7 +241,6 @@ CodecDescriptor::unloadCodec( CodecHandlePointer p )
{
// _debug("Unload codec %s\n", p.first->getCodecName().c_str());
using std::cerr;
- int i;
destroy_t* destroyCodec = (destroy_t*)dlsym( p.second , "destroy");
if(dlerror())
cerr << dlerror() << '\n';
@@ -311,7 +310,7 @@ CodecDescriptor::seemsValid( std::string lib)
CodecDescriptor::alreadyInCache( std::string lib )
{
int i;
- for( i = 0 ; i < _Cache.size() ; i++ )
+ for( i = 0 ; (unsigned int)i < _Cache.size() ; i++ )
{
if( _Cache[i] == lib ){
return true;}
@@ -322,7 +321,6 @@ CodecDescriptor::alreadyInCache( std::string lib )
bool
CodecDescriptor::isCodecLoaded( int payload )
{
- int i;
CodecsMap::iterator iter = _CodecsMap.begin();
while( iter != _CodecsMap.end())
{
diff --git a/src/audio/dtmf.cpp b/src/audio/dtmf.cpp
index faabf1d69a..c6ba907aae 100644
--- a/src/audio/dtmf.cpp
+++ b/src/audio/dtmf.cpp
@@ -24,10 +24,8 @@
#include "dtmf.h"
DTMF::DTMF (unsigned int sampleRate)
-: dtmfgenerator(sampleRate)
+: currentTone(0), newTone(0), dtmfgenerator(sampleRate)
{
- currentTone = 0;
- newTone = 0;
}
DTMF::~DTMF (void)
diff --git a/src/audio/dtmfgenerator.cpp b/src/audio/dtmfgenerator.cpp
index 24b00942ac..f35f1e50ac 100644
--- a/src/audio/dtmfgenerator.cpp
+++ b/src/audio/dtmfgenerator.cpp
@@ -51,9 +51,8 @@ const DTMFGenerator::DTMFTone DTMFGenerator::tones[NUM_TONES] = {
};
-DTMFException::DTMFException(const char* _reason) throw()
+DTMFException::DTMFException(const char* _reason) throw() : reason(_reason)
{
- reason = _reason;
}
@@ -61,20 +60,6 @@ DTMFException::~DTMFException() throw()
{
}
-
-DTMFException::DTMFException(const DTMFException& rh) throw()
-{
- _debug("DTMFException copy constructor hasn't been implemented yet. Quit!");
- exit(0);
-}
-
-DTMFException& DTMFException::operator=(const DTMFException& rh) throw()
-{
- _debug("DTMFException assignment operator hasn't been implemented yet. Quit!");
- exit(0);
-}
-
-
const char* DTMFException::what() const throw()
{
return reason;
@@ -85,9 +70,8 @@ const char* DTMFException::what() const throw()
/*
* Initialize the generator
*/
-DTMFGenerator::DTMFGenerator(unsigned int sampleRate) : tone("", sampleRate)
+DTMFGenerator::DTMFGenerator(unsigned int sampleRate) : state(), _sampleRate(sampleRate), tone("", sampleRate)
{
- _sampleRate = sampleRate;
state.offset = 0;
state.sample = 0;
@@ -103,18 +87,6 @@ DTMFGenerator::~DTMFGenerator() {
}
}
-DTMFGenerator::DTMFGenerator(const DTMFGenerator& rh) : tone("", rh._sampleRate)
-{
- _debug("DTMFGenerator copy constructor hasn't been implemented yet. Quit!");
- exit(0);
-}
-
-DTMFGenerator& DTMFGenerator::operator=(const DTMFGenerator& rh)
-{
- _debug("DTMFGenerator assignment operator hasn't been implemented yet. Quit!");
- exit(0);
-}
-
/*
* Get n samples of the signal of code code
*/
diff --git a/src/audio/dtmfgenerator.h b/src/audio/dtmfgenerator.h
index 0f6250c2ae..a4bed74983 100644
--- a/src/audio/dtmfgenerator.h
+++ b/src/audio/dtmfgenerator.h
@@ -39,6 +39,7 @@
class DTMFException : public std::exception
{
private:
+
/** Message */
const char* reason;
public:
@@ -52,13 +53,13 @@ class DTMFException : public std::exception
* Destructor
*/
virtual ~DTMFException() throw();
-
+/*
// Copy Constructor
DTMFException(const DTMFException& rh) throw();
// Assignment Operator
- DTMFException& operator=( const DTMFException& rh) throw();
-
+ DTMFException& operator=( const DTMFException& rh) throw();
+*/
/**
* @return const char* The error
*/
@@ -113,6 +114,7 @@ class DTMFGenerator
*/
~DTMFGenerator();
+
// Copy Constructor
DTMFGenerator(const DTMFGenerator& rh);
@@ -136,6 +138,7 @@ class DTMFGenerator
void getNextSamples(SFLDataFormat* buffer, size_t n) throw (DTMFException);
private:
+
/**
* Generate samples for a specific dtmf code
* @param code The code
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 41af80b5e1..a9942eacea 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -22,10 +22,15 @@
int framesPerBuffer = 2048;
PulseLayer::PulseLayer(ManagerImpl* manager)
- : AudioLayer( manager , PULSEAUDIO )
+ : AudioLayer( manager , PULSEAUDIO )
+ , _mainSndRingBuffer( SIZEBUF )
, _urgentRingBuffer( SIZEBUF)
- ,_mainSndRingBuffer( SIZEBUF )
- ,_micRingBuffer( SIZEBUF )
+ , _micRingBuffer( SIZEBUF )
+ , context(NULL)
+ , m(NULL)
+ , playback()
+ , record()
+ , cache()
{
_debug("Pulse audio constructor: Create context\n");
}
@@ -53,7 +58,6 @@ PulseLayer::closeLayer( void )
PulseLayer::connectPulseAudioServer( void )
{
pa_context_flags_t flag = PA_CONTEXT_NOAUTOSPAWN ;
- int ret = 1;
pa_threaded_mainloop_lock( m );
@@ -87,7 +91,6 @@ void PulseLayer::context_state_callback( pa_context* c, void* user_data )
_debug("Waiting....\n");
break;
case PA_CONTEXT_READY:
- pa_cvolume cv;
pulse->createStreams( c );
_debug("Connection to PulseAudio server established\n");
break;
@@ -148,6 +151,7 @@ PulseLayer::openDevice(int indexIn, int indexOut, int sampleRate, int frameSize
connectPulseAudioServer();
_debug("Connection Done!! \n");
+ return true;
}
void
@@ -254,7 +258,7 @@ PulseLayer::underflow ( pa_stream* s, void* userdata )
void
PulseLayer::overflow ( pa_stream* s, void* userdata )
{
- PulseLayer* pulse = (PulseLayer*) userdata;
+ //PulseLayer* pulse = (PulseLayer*) userdata;
pa_stream_drop( s );
pa_stream_trigger( s, NULL, NULL);
}
@@ -355,6 +359,7 @@ PulseLayer::putInCache( char code, void *buffer, int toCopy )
_debug("Put the DTMF in cache\n");
//pa_stream_write( cache->pulseStream() , buffer , toCopy , pa_xfree, 0 , PA_SEEK_RELATIVE);
//pa_stream_finish_upload( cache->pulseStream() );
+ return 1;
}
static void retrieve_server_info(pa_context *c, const pa_server_info *i, void *userdata)
@@ -365,6 +370,7 @@ static void retrieve_server_info(pa_context *c, const pa_server_info *i, void *u
_debug("\t\tDefault source name : %s\n" , i->default_source_name);
}
+/*
static void retrieve_client_list(pa_context *c, const pa_client_info *i, int eol, void *userdata)
{
_debug("end of list = %i\n", eol);
@@ -373,6 +379,7 @@ static void retrieve_client_list(pa_context *c, const pa_client_info *i, int eol
_debug("\t\tOwner module : %i\n" , i->owner_module);
_debug("\t\tDriver : %s\n" , i->driver);
}
+*/
static void reduce_sink_list(pa_context *c, const pa_sink_input_info *i, int eol, void *userdata)
{
@@ -392,7 +399,7 @@ static void reduce_sink_list(pa_context *c, const pa_sink_input_info *i, int eol
static void restore_sink_list(pa_context *c, const pa_sink_input_info *i, int eol, void *userdata)
{
PulseLayer* pulse = (PulseLayer*) userdata;
- AudioStream* s = pulse->getPlaybackStream();
+ //AudioStream* s = pulse->getPlaybackStream();
if( !eol ){
_debug("Sink Info: index : %i\n" , i->index);
_debug("\t\tSink name : -%s-\n" , i->name);
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index 122b267753..08db1de9d3 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -173,6 +173,12 @@ class PulseLayer : public AudioLayer {
void setPlaybackVolume( double volume );
private:
+ // Copy Constructor
+ PulseLayer(const PulseLayer& rh);
+
+ // Assignment Operator
+ PulseLayer& operator=( const PulseLayer& rh);
+
/**
* Drop the pending frames and close the capture device
*/
diff --git a/src/audio/ringbuffer.cpp b/src/audio/ringbuffer.cpp
index 2d474a1bc6..f56e8ec0e9 100644
--- a/src/audio/ringbuffer.cpp
+++ b/src/audio/ringbuffer.cpp
@@ -42,19 +42,6 @@ RingBuffer::~RingBuffer() {
delete[] mBuffer; mBuffer = NULL;
}
-//copy constructor
-RingBuffer::RingBuffer(const RingBuffer& rh): mStart(rh.mStart), mEnd(rh.mEnd), mBufferSize( rh.mBufferSize ), mBuffer ( rh.mBuffer )
-{
- _debug("RingBuffer copy constructor hasn't been implemented yet. Quit!");
- exit(0);
-}
-
-// Assignment operator
-RingBuffer& RingBuffer::operator=(const RingBuffer& rh) {
- _debug("RingBuffer assignment operator hasn't been implemented yet. Quit!");
- exit(0);
-}
-
void
RingBuffer::flush (void) {
mStart = 0;
diff --git a/src/audio/ringbuffer.h b/src/audio/ringbuffer.h
index 8c991bd133..27db3e31b3 100644
--- a/src/audio/ringbuffer.h
+++ b/src/audio/ringbuffer.h
@@ -38,11 +38,6 @@ class RingBuffer {
*/
~RingBuffer();
- // Copy Constructor
- RingBuffer(const RingBuffer& rh);
-
- // Assignment operator
- RingBuffer& operator=(const RingBuffer& rh);
/**
* Reset the counters to 0
@@ -98,6 +93,11 @@ class RingBuffer {
void debug();
private:
+ // Copy Constructor
+ RingBuffer(const RingBuffer& rh);
+
+ // Assignment operator
+ RingBuffer& operator=(const RingBuffer& rh);
/** Pointer on the first data */
int mStart;
diff --git a/src/audio/samplecache.h b/src/audio/samplecache.h
index c5335d8b52..4c07d558b4 100644
--- a/src/audio/samplecache.h
+++ b/src/audio/samplecache.h
@@ -29,23 +29,18 @@ class SampleCache {
SampleCache( pa_stream* stream );
~SampleCache();
- // Copy Constructor
- SampleCache(const SampleCache& rh):_stream(rh._stream) {
- _debug("SampleCache copy constructor hasn't been implemented yet. Quit!");
- exit(0);
- }
-
- // Assignment Operator
- SampleCache& operator=( const SampleCache& rh){
- _debug("SampleCache assignment operator hasn't been implemented yet. Quit!");
- exit(0);
- }
bool uploadSample( SFLDataFormat* buffer, size_t size );
bool removeSample( );
bool isSampleCached( );
private:
+ // Copy Constructor
+ SampleCache(const SampleCache& rh);
+
+ // Assignment Operator
+ SampleCache& operator=( const SampleCache& rh);
+
pa_stream* _stream;
};
diff --git a/src/audio/tone.cpp b/src/audio/tone.cpp
index d015abed7f..646dae56a5 100644
--- a/src/audio/tone.cpp
+++ b/src/audio/tone.cpp
@@ -26,9 +26,8 @@
#include "tone.h"
#include <math.h>
-Tone::Tone(const std::string& definition, unsigned int sampleRate) : AudioLoop()
+Tone::Tone(const std::string& definition, unsigned int sampleRate) : AudioLoop(), _sampleRate(sampleRate)
{
- _sampleRate = sampleRate;
genBuffer(definition); // allocate memory with definition parameter
}
diff --git a/src/audio/tonegenerator.cpp b/src/audio/tonegenerator.cpp
index 37cfd1eaaa..e5a29cd078 100644
--- a/src/audio/tonegenerator.cpp
+++ b/src/audio/tonegenerator.cpp
@@ -31,8 +31,8 @@ int AMPLITUDE = 32767;
// ToneGenerator implementation
///////////////////////////////////////////////////////////////////////////////
-ToneGenerator::ToneGenerator (unsigned int sampleRate) {
- _sampleRate = sampleRate;
+ToneGenerator::ToneGenerator (unsigned int sampleRate): sample(NULL), freq1(), freq2(), time(), totalbytes(), _sampleRate(sampleRate)
+{
}
ToneGenerator::~ToneGenerator (void) {
diff --git a/src/audio/tonegenerator.h b/src/audio/tonegenerator.h
index 062caebd51..8fae9ea29f 100644
--- a/src/audio/tonegenerator.h
+++ b/src/audio/tonegenerator.h
@@ -63,6 +63,12 @@ class ToneGenerator {
int totalbytes;
private:
+ // Copy Constructor
+ ToneGenerator(const ToneGenerator& rh);
+
+ // Assignment Operator
+ ToneGenerator& operator=( const ToneGenerator& rh);
+
/*
* Initialisation of the supported tones according to the countries.
*/
diff --git a/src/audio/tonelist.h b/src/audio/tonelist.h
index 88c91824da..849af15477 100644
--- a/src/audio/tonelist.h
+++ b/src/audio/tonelist.h
@@ -40,17 +40,6 @@ public:
*/
~ToneList();
- // Copy Constructor
- ToneList(const ToneList& rh):_nbTone(rh._nbTone),_nbCountry(rh._nbCountry),_defaultCountryId(rh._defaultCountryId) {
- _debug("ToneList copy constructor hasn't been implemented yet. Quit!");
- exit(0);
- }
-
- // Assignment Operator
- ToneList& operator=( const ToneList& rh){
- _debug("ToneList assignment operator hasn't been implemented yet. Quit!");
- exit(0);
- }
/** Countries */
enum COUNTRYID {
@@ -85,6 +74,13 @@ public:
int getNbTone() { return _nbTone; }
private:
+
+ // Copy Constructor
+ ToneList(const ToneList& rh);
+
+ // Assignment Operator
+ ToneList& operator=( const ToneList& rh);
+
void initToneDefinition();
std::string _toneZone[TONE_NBCOUNTRY][TONE_NBTONE];
int _nbTone;
@@ -101,17 +97,6 @@ public:
TelephoneTone(const std::string& countryName, unsigned int sampleRate);
~TelephoneTone();
- // Copy Constructor
- TelephoneTone(const TelephoneTone& rh):_currentTone(rh._currentTone), _toneList(rh._toneList) {
- _debug("TelephoneTone copy constructor hasn't been implemented yet. Quit!");
- exit(0);
- };
-
- // Assignment Operator
- TelephoneTone& operator=( const TelephoneTone& rh){
- _debug("TelephoneTone assignment operator hasn't been implemented yet. Quit!");
- exit(0);
- }
/** send TONE::ZT_TONE_NULL to stop the playing */
void setCurrentTone(Tone::TONEID toneId);
@@ -126,6 +111,12 @@ public:
bool shouldPlay();
private:
+ // Copy Constructor
+ TelephoneTone(const TelephoneTone& rh);
+
+ // Assignment Operator
+ TelephoneTone& operator=( const TelephoneTone& rh);
+
Tone* _tone[TONE_NBTONE];
Tone::TONEID _currentTone;
ToneList _toneList;
diff --git a/src/call.cpp b/src/call.cpp
index 7d27f026c8..a29c60adfa 100644
--- a/src/call.cpp
+++ b/src/call.cpp
@@ -19,16 +19,23 @@
*/
#include "call.h"
-Call::Call(const CallID& id, Call::CallType type) : _id(id), _type(type),
- _localIPAddress(""),
- _remoteIPAddress("")
+Call::Call(const CallID& id, Call::CallType type)
+ : _callMutex()
+ , _codecMap()
+ , _audioCodec()
+ , _audioStarted(false)
+ , _localIPAddress("")
+ , _localAudioPort(0)
+ , _localExternalAudioPort(0)
+ , _remoteIPAddress("")
+ , _remoteAudioPort(0)
+ , _id(id)
+ , _type(type)
+ , _connectionState(Call::Disconnected)
+ , _callState(Call::Inactive)
+ , _peerName()
+ , _peerNumber()
{
- _connectionState = Call::Disconnected;
- _callState = Call::Inactive;
- //_audioCodec = 0;
- _localAudioPort = 0;
- _localExternalAudioPort = 0;
- _remoteAudioPort = 0;
}
diff --git a/src/dbus/contactmanager.cpp b/src/dbus/contactmanager.cpp
index 27e9c8cc8c..52dcfa1b93 100644
--- a/src/dbus/contactmanager.cpp
+++ b/src/dbus/contactmanager.cpp
@@ -25,7 +25,7 @@ const char* ContactManager::SERVER_PATH = "/org/sflphone/SFLphone/ContactManager
std::map< ::DBus::String, ::DBus::String >
ContactManager::getContacts( const ::DBus::String& accountID )
{
- // TODO
+ // TODO
}
void
diff --git a/src/eventthread.cpp b/src/eventthread.cpp
index 49d361d50f..1ec4363470 100644
--- a/src/eventthread.cpp
+++ b/src/eventthread.cpp
@@ -27,20 +27,6 @@ EventThread::EventThread (VoIPLink* link) : Thread (), _linkthread(link), stopI
setCancel(cancelDeferred);
}
-// Copy Constructor
-EventThread::EventThread(const EventThread& rh): _linkthread(rh._linkthread), stopIt(rh.stopIt)
-{
- _debug("EventThread copy constructor hasn't been implemented yet. Quit! ");
- exit();
-}
-
-// Assignment Operator overloading
-EventThread& EventThread::operator=(const EventThread& rh)
-{
- _debug("EventThread assignment operator hasn't been implemented yet. Quit! ");
- exit();
-}
-
EventThread::~EventThread (void)
{
terminate();
diff --git a/src/eventthread.h b/src/eventthread.h
index 96b70f7e1c..8772c72043 100644
--- a/src/eventthread.h
+++ b/src/eventthread.h
@@ -37,15 +37,16 @@ public:
*/
EventThread (VoIPLink*);
~EventThread (void);
- EventThread(const EventThread& rh); // copy constructor
- EventThread& operator=(const EventThread& rh); // assignment operator
virtual void run ();
virtual void stop();
virtual void startLoop();
bool isStopped();
private:
- /** VoIPLink is the object being called by getEvents() method */
+ EventThread(const EventThread& rh); // copy constructor
+ EventThread& operator=(const EventThread& rh); // assignment operator
+
+ /** VoIPLink is the object being called by getEvents() method */
VoIPLink* _linkthread;
bool stopIt;
};
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 76c5587747..082551eac0 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -61,35 +61,52 @@
#define fill_config_int(name, value) \
(_config.addConfigTreeItem(section, Conf::ConfigTreeItem(std::string(name), std::string(value), type_int)))
-ManagerImpl::ManagerImpl (void)
+ManagerImpl::ManagerImpl (void)
+ : _hasTriedToRegister(false)
+ , _config()
+ , _currentCallId2()
+ , _currentCallMutex()
+ , _codecBuilder(NULL)
+ , _audiodriver(NULL)
+ , _dtmfKey(NULL)
+ , _codecDescriptorMap()
+ , _toneMutex()
+ , _telephoneTone(NULL)
+ , _audiofile()
+ , _spkr_volume(0)
+ , _mic_volume(0)
+ , _mutex()
+ , _dbus(NULL)
+ , _waitingCall()
+ , _waitingCallMutex()
+ , _nbIncomingWaitingCall(0)
+ , _path("")
+ , _exist(0)
+ , _setupLoaded(false)
+ , _firewallPort()
+ , _firewallAddr("")
+ , _hasZeroconf(false)
+ , _callAccountMap()
+ , _callAccountMapMutex()
+ , _accountMap()
+ , _userAgent(NULL)
+ , _userAgentInitlized(false)
+ , _sipThreadStop()
+
{
- // Init private variables
- _hasZeroconf = false;
+ /* Init private variables
+ setup: _path, _exist, _setupLoaded , _dbus
+ sound: _audiodriver, _dtmfKey,
+ _spkr_volume,_mic_volume = 0; // Initialize after by init() -> initVolume()
+ Call: _nbIncomingWaitingCall, _hasTriedToRegister
+ SIP Link: _userAgent, _userAgentInitlized
+ */
+
#ifdef USE_ZEROCONF
_hasZeroconf = true;
_DNSService = new DNSService();
#endif
- // setup
- _path = "";
- _exist = 0;
- _setupLoaded = false;
- _dbus = NULL;
-
- // sound
- _audiodriver = NULL;
- _dtmfKey = 0;
- _spkr_volume = 0; // Initialize after by init() -> initVolume()
- _mic_volume = 0; // Initialize after by init() -> initVolume()
-
- // Call
- _nbIncomingWaitingCall=0;
- _hasTriedToRegister = false;
-
- // SIP Link
- _userAgent = NULL;
- _userAgentInitlized = false;
-
// initialize random generator for call id
srand (time(NULL));
@@ -809,6 +826,7 @@ ManagerImpl::stopTone(bool stopAudio=true) {
ManagerImpl::playTone()
{
playATone(Tone::TONE_DIALTONE);
+ return true;
}
/**
@@ -818,6 +836,7 @@ ManagerImpl::playTone()
ManagerImpl::playToneWithMessage()
{
playATone(Tone::TONE_CONGESTION);
+ return true;
}
/**
@@ -1111,7 +1130,7 @@ ManagerImpl::setActiveCodecList(const std::vector< ::DBus::String >& list)
std::string
ManagerImpl::serialize(std::vector<std::string> v)
{
- int i;
+ unsigned int i;
std::string res;
for(i=0;i<v.size();i++)
{
@@ -1127,7 +1146,7 @@ ManagerImpl::getActiveCodecList( void )
_debug("Get Active codecs list\n");
std::vector< std::string > v;
CodecOrder active = _codecDescriptorMap.getActiveCodecs();
- int i=0;
+ unsigned int i=0;
size_t size = active.size();
while(i<size)
{
@@ -1247,7 +1266,7 @@ ManagerImpl::setInputAudioPlugin(const std::string& audioPlugin)
void
ManagerImpl::setOutputAudioPlugin(const std::string& audioPlugin)
{
- int layer = _audiodriver -> getLayerType();
+ //int layer = _audiodriver -> getLayerType();
_debug("Set output audio plugin\n");
_audiodriver -> setErrorMessage( -1 );
_audiodriver -> openDevice( _audiodriver -> getIndexIn(),
@@ -1278,7 +1297,7 @@ ManagerImpl::getAudioOutputDeviceList(void)
void
ManagerImpl::setAudioOutputDevice(const int index)
{
- int layer = _audiodriver -> getLayerType();
+ //int layer = _audiodriver -> getLayerType();
_debug("Set audio output device: %i\n", index);
_audiodriver -> setErrorMessage( -1 );
_audiodriver->openDevice(_audiodriver->getIndexIn(),
@@ -1309,7 +1328,7 @@ ManagerImpl::getAudioInputDeviceList(void)
void
ManagerImpl::setAudioInputDevice(const int index)
{
- int layer = _audiodriver -> getLayerType();
+ //int layer = _audiodriver -> getLayerType();
_debug("Set audio input device %i\n", index);
_audiodriver -> setErrorMessage( -1 );
_audiodriver->openDevice(index,
diff --git a/src/managerimpl.h b/src/managerimpl.h
index f3c1e71478..f43800cb5a 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -819,6 +819,7 @@ class ManagerImpl {
//pjsip_regc *getSipRegcFromID(const AccountID& id);
private:
+
/**
* Create .PROGNAME directory in home user and create
* configuration tree from the settings file if this file exists.
@@ -1026,6 +1027,13 @@ public:
AccountID getAccountIdFromNameAndServer(const std::string& userName, const std::string& server);
private:
+
+ // Copy Constructor
+ ManagerImpl(const ManagerImpl& rh);
+
+ // Assignment Operator
+ ManagerImpl& operator=( const ManagerImpl& rh);
+
/**
* The UserAgent provides sip operation facilities for all sip accounts
*/
diff --git a/src/samplerateconverter.cpp b/src/samplerateconverter.cpp
index 126c5d957c..3c6d0ed90d 100644
--- a/src/samplerateconverter.cpp
+++ b/src/samplerateconverter.cpp
@@ -18,19 +18,31 @@
#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 );
-
+SamplerateConverter::SamplerateConverter( void )
+ : _frequence(Manager::instance().getConfigInt( AUDIO , ALSA_SAMPLE_RATE )) //44100
+ , _framesize(Manager::instance().getConfigInt( AUDIO , ALSA_FRAME_SIZE ))
+ , _floatBufferDownMic(NULL)
+ , _floatBufferUpMic(NULL)
+ , _src_state_mic(NULL)
+ , _floatBufferDownSpkr(NULL)
+ , _floatBufferUpSpkr(NULL)
+ , _src_state_spkr(NULL)
+ , _src_err(0)
+{
init();
}
-SamplerateConverter::SamplerateConverter( int freq , int fs ) {
-
- _frequence = freq ;
- _framesize = fs ;
-
+SamplerateConverter::SamplerateConverter( int freq , int fs )
+ : _frequence(freq)
+ , _framesize(fs)
+ , _floatBufferDownMic(NULL)
+ , _floatBufferUpMic(NULL)
+ , _src_state_mic(NULL)
+ , _floatBufferDownSpkr(NULL)
+ , _floatBufferUpSpkr(NULL)
+ , _src_state_spkr(NULL)
+ , _src_err(0)
+{
init();
}
diff --git a/src/samplerateconverter.h b/src/samplerateconverter.h
index 86ea175319..f6423197fa 100644
--- a/src/samplerateconverter.h
+++ b/src/samplerateconverter.h
@@ -58,6 +58,12 @@ class SamplerateConverter {
int getFramesize( void ) { return _framesize; }
private:
+ // Copy Constructor
+ SamplerateConverter(const SamplerateConverter& rh);
+
+ // Assignment Operator
+ SamplerateConverter& operator=( const SamplerateConverter& rh);
+
void init( void );
/** Audio layer caracteristics */
diff --git a/src/sipaccount.cpp b/src/sipaccount.cpp
index 9d3aff2f6b..399b7fc2cc 100644
--- a/src/sipaccount.cpp
+++ b/src/sipaccount.cpp
@@ -28,8 +28,11 @@
SIPAccount::SIPAccount(const AccountID& accountID)
: Account(accountID)
+ , _userName("")
+ , _server("")
+ , _cred(NULL)
+ , _contact("")
{
- _cred = NULL;
_link = new SIPVoIPLink(accountID);
}
diff --git a/src/sipaccount.h b/src/sipaccount.h
index 48eb62c7a8..5c32f1e50a 100644
--- a/src/sipaccount.h
+++ b/src/sipaccount.h
@@ -40,6 +40,12 @@ public:
*/
SIPAccount(const AccountID& accountID);
+ /* Copy Constructor */
+ SIPAccount(const SIPAccount& rh);
+
+ /* Assignment Operator */
+ SIPAccount& operator=( const SIPAccount& rh);
+
/**
* Virtual destructor
*/
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 4512839603..c5e33a7cb3 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -30,12 +30,15 @@
#define _RECVONLY 2
SIPCall::SIPCall(const CallID& id, Call::CallType type) : Call(id, type)
+ , _cid(0)
+ , _did(0)
+ , _tid(0)
+ , _localSDP(NULL)
+ , _negociator(NULL)
+ , _ipAddr("")
+ , _xferSub(NULL)
+ , _invSession(NULL)
{
- _cid = 0;
- _did = 0;
- _tid = 0;
- _xferSub = NULL;
- _invSession = NULL;
}
SIPCall::~SIPCall()
diff --git a/src/sipcall.h b/src/sipcall.h
index 1e0dcb5f51..fe23553d5d 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -130,6 +130,13 @@ class SIPCall : public Call
pjsip_inv_session *getInvSession() {return _invSession;}
private:
+
+ // Copy Constructor
+ SIPCall(const SIPCall& rh);
+
+ // Assignment Operator
+ SIPCall& operator=( const SIPCall& rh);
+
/**
* Get a valid remote SDP or return a 400 bad request response if invalid
* @param
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 26cc9f91b4..129ad5ee3a 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -44,13 +44,21 @@
#define INVITE_METHOD "INVITE"
SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
- : VoIPLink(accountID), _localExternAddress("")
+ : VoIPLink(accountID)
+ , _initDone(false)
+ , _nbTryListenAddr(2) // number of times to try to start SIP listener
+ , _useStun(false)
+ , _stunServer("")
+ , _localExternAddress("")
+ , _localExternPort(0)
+ , _proxy("")
+ , _authname("")
+ , _password("")
+ , _audiortp(new AudioRtp())
+ , _regc()
+ , _server("")
+ , _bRegister(false)
{
-
- _nbTryListenAddr = 2; // number of times to try to start SIP listener
- _localExternPort = 0;
-
- _audiortp = new AudioRtp();
// to get random number for RANDOM_PORT
srand (time(NULL));
}
@@ -352,7 +360,7 @@ SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
SIPCall* call = getSIPCall(id);
if (call==0) { _debug("Call doesn't exist\n"); return false; }
- int duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
+ //int duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
// TODO Add DTMF with pjsip - INFO method
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 265c6366e7..35d82e6c1e 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -60,6 +60,12 @@ class SIPVoIPLink : public VoIPLink
*/
~SIPVoIPLink();
+ /* Copy Constructor */
+ SIPVoIPLink(const SIPVoIPLink& rh);
+
+ /* Assignment Operator */
+ SIPVoIPLink& operator=( const SIPVoIPLink& rh);
+
/**
* Try to initiate the pjsip engine/thread and set config
* @return bool True if OK
diff --git a/src/useragent.cpp b/src/useragent.cpp
index 4f4370208a..0ba6aefd4c 100644
--- a/src/useragent.cpp
+++ b/src/useragent.cpp
@@ -670,7 +670,7 @@ pj_bool_t UserAgent::mod_on_rx_request(pjsip_rx_data *rdata) {
// Get the account id of callee from username and server
account_id = Manager::instance().getAccountIdFromNameAndServer(userName, server);
if(account_id == AccountNULL) {
- _debug("UserAgent: Username %s doesn't match any account!\n",userName);
+ _debug("UserAgent: Username %s doesn't match any account!\n",userName.c_str());
return PJ_FALSE;
}
_debug("UserAgent: The receiver is : %s@%s\n", userName.data(), server.data());
diff --git a/src/useragent.h b/src/useragent.h
index 8251e15742..655ba60f51 100644
--- a/src/useragent.h
+++ b/src/useragent.h
@@ -137,6 +137,13 @@ public:
static UserAgent* getInstance() {return _current;}
static void set_voicemail_info( AccountID account, pjsip_msg_body *body );
+private:
+
+ // Copy Constructor
+ UserAgent(const UserAgent& rh);
+
+ // Assignment Operator
+ UserAgent& operator=( const UserAgent& rh);
};
--
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