From 35a4aa424b3b6cbff1126329c2c25f0d10065dba Mon Sep 17 00:00:00 2001
From: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
Date: Wed, 14 May 2008 15:43:58 -0400
Subject: [PATCH] pulseaudio addons
---
src/Makefile.am | 3 +-
src/audio/Makefile.am | 2 +-
src/audio/alsalayer.cpp | 54 ++++++++++--------
src/audio/audiolayer.cpp | 50 ++++++++--------
src/audio/audiolayer.h | 9 ++-
src/audio/pulselayer.cpp | 119 +++++++++++++++++++++++++++++++++++----
src/audio/pulselayer.h | 45 ++++++++++++---
src/global.h | 5 ++
src/managerimpl.cpp | 17 +++++-
9 files changed, 232 insertions(+), 72 deletions(-)
diff --git a/src/Makefile.am b/src/Makefile.am
index 2110acbfc9..ca7dc83f1b 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -29,6 +29,7 @@ IAXHEADERS =
endif
ALSAFLAG= -lasound
+PULSEAUDIO_LIBS=-lpulse
SUBDIRS = audio config dbus $(ZEROCONFDIR) contact memmanager video mixer
@@ -41,7 +42,7 @@ sflphoned_SOURCES = eventthread.cpp main.cpp voiplink.cpp \
sflphoned_CXXFLAGS = -DPREFIX=\"$(prefix)\" -DPROGSHAREDIR=\"${datadir}/sflphone\" $(ZEROCONFFLAGS) $(IAX_FLAGS) $(SFLPHONE_CFLAGS) $(SIP_CFLAGS)
#sflphoned_LDFLAGS = -static
-sflphoned_LDADD = ./libsflphone.la $(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) $(IAX_LIBS) $(EXOSIP_LIBS) $(ALSAFLAG)
+sflphoned_LDADD = ./libsflphone.la $(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) $(IAX_LIBS) $(EXOSIP_LIBS) $(ALSAFLAG) $(PULSEAUDIO_LIBS)
AM_CPPFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/libs $(libccext2_CFLAGS) $(libccgnu2_CFLAGS) $(IAX_CFLAGS) $(USER_INCLUDES) $(libdbuscpp_CFLAGS) \
-DCODECS_DIR=\""$(sflcodecdir)"\"
-DENABLE_TRACE
diff --git a/src/audio/Makefile.am b/src/audio/Makefile.am
index 22c0c99e45..13bb710847 100644
--- a/src/audio/Makefile.am
+++ b/src/audio/Makefile.am
@@ -28,7 +28,7 @@ tonegenerator.cpp codecDescriptor.cpp \
audioloop.cpp ringbuffer.cpp $(SPEEX_SOURCES_CPP)
AM_CXXFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/libs $(libccext2_CFLAGS) $(libdbuscpp_CFLAGS) $(libccrtp1_CFLAGS) $(USER_INCLUDES) \
- -DCODECS_DIR=\""$(sflcodecdir)"\" $(SPEEX_FLAG) $(GSM_FLAG) $(ILBC_FLAG)
+ -DCODECS_DIR=\""$(sflcodecdir)"\" $(SPEEX_FLAG) $(GSM_FLAG) $(ILBC_FLAG)
noinst_HEADERS = audioloop.h common.h ringbuffer.h audiofile.h \
tonelist.h audiortp.h audiolayer.h alsalayer.h pulselayer.h audiodevice.h \
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index 44c11dcba4..1003709b01 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -21,10 +21,9 @@
#include "alsalayer.h"
-
// Constructor
AlsaLayer::AlsaLayer( ManagerImpl* manager )
- : AudioLayer( manager )
+ : AudioLayer( manager , ALSA )
, _PlaybackHandle(NULL)
, _CaptureHandle(NULL)
{
@@ -135,7 +134,7 @@ AlsaLayer::fillHWBuffer( void)
data[4*l2+2] = (unsigned char)s2;
data[4*l2+3] = s2 >> 8;
}
- while ((pcmreturn = snd_pcm_writei(_PlaybackHandle, data, frames)) < 0) {
+ while ((pcmreturn = snd_pcm_mmap_writei(_PlaybackHandle, data, frames)) < 0) {
snd_pcm_prepare(_PlaybackHandle);
//_debugAlsa("< Buffer Underrun >\n");
}
@@ -213,7 +212,6 @@ AlsaLayer::isStreamStopped (void)
return !(isStreamActive());
}
-
//////////////////////////////////////////////////////////////////////////////////////////////
///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////////////////////
@@ -272,8 +270,8 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
unsigned int rate_in = getSampleRate();
unsigned int rate_out = getSampleRate();
int dir = 0;
- snd_pcm_uframes_t period_size = 2048;
- snd_pcm_uframes_t buffer_size = period_size * 4 ;
+ 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 period_size_out = getFrameSize() * getSampleRate() / 1000 * 2;//1024 ;
snd_pcm_uframes_t buffer_size_out = period_size_out * 4 ;
@@ -301,7 +299,11 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
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);
snd_pcm_hw_params_free( hwParams );
snd_pcm_uframes_t val ;
@@ -323,8 +325,8 @@ 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, 0 ) < 0){
- _debugAlsa("Error while 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){
+ _debugAlsa("Error while opening playback device %s\n", pcm_c.c_str());
setErrorMessage( ALSA_PLAYBACK_DEVICE );
return false;
}
@@ -333,14 +335,15 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
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_RW_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_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_size( _PlaybackHandle, hwParams, buffer_size ) < 0) _debugAlsa(" Cannot set buffer size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_period_size_near( _PlaybackHandle, hwParams, &period_size , &dir) < 0) _debugAlsa(" Cannot set period size (%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));
@@ -366,7 +369,7 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
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();
@@ -379,26 +382,28 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
int
AlsaLayer::write(void* buffer, int length)
{
+ //if(snd_pcm_state( _PlaybackHandle ) == SND_PCM_STATE_XRUN)
+ //handle_xrun_playback();
+ //_debugAlsa("avail = %d - toWrite = %d\n" , snd_pcm_avail_update( _PlaybackHandle ) , length / 2);
+
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _PlaybackHandle, length);
- //int err = snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
- int err = snd_pcm_writei( _PlaybackHandle , buffer , frames );
+ int err = snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
switch(err) {
case -EAGAIN:
- _debugAlsa("EAGAIN (%s)\n", snd_strerror( err ));
- //snd_pcm_resume( _PlaybackHandle );
+ _debugAlsa(" (%s)\n", snd_strerror( err ));
+ snd_pcm_resume( _PlaybackHandle );
break;
case -EPIPE:
_debugAlsa(" UNDERRUN (%s)\n", snd_strerror(err));
handle_xrun_playback();
- //snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
- //snd_pcm_writei( _PlaybackHandle , buffer , frames );
+ snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
break;
case -ESTRPIPE:
- _debugAlsa("ESTRPIPE (%s)\n", snd_strerror(err));
- //snd_pcm_resume( _PlaybackHandle );
+ _debugAlsa(" (%s)\n", snd_strerror(err));
+ snd_pcm_resume( _PlaybackHandle );
break;
case -EBADFD:
- _debugAlsa("EBADFD (%s)\n", snd_strerror( err ));
+ _debugAlsa(" (%s)\n", snd_strerror( err ));
break;
}
@@ -480,8 +485,7 @@ AlsaLayer::handle_xrun_playback( void )
state = snd_pcm_status_get_state( status );
if( state == SND_PCM_STATE_XRUN )
{
- //snd_pcm_drop( _PlaybackHandle );
- _debug("Underrun\n");
+ snd_pcm_drop( _PlaybackHandle );
snd_pcm_prepare( _PlaybackHandle );
//snd_pcm_start( _PlaybackHandle );
}
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 0b0c54edb2..d799939335 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -152,7 +152,7 @@ AudioLayer::fillHWBuffer( void)
data[4*l2+2] = (unsigned char)s2;
data[4*l2+3] = s2 >> 8;
}
- while ((pcmreturn = snd_pcm_writei(_PlaybackHandle, data, frames)) < 0) {
+ while ((pcmreturn = snd_pcm_mmap_writei(_PlaybackHandle, data, frames)) < 0) {
snd_pcm_prepare(_PlaybackHandle);
//_debugAlsa("< Buffer Underrun >\n");
}
@@ -295,8 +295,8 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
unsigned int rate_in = getSampleRate();
unsigned int rate_out = getSampleRate();
int dir = 0;
- snd_pcm_uframes_t period_size = 2048;
- snd_pcm_uframes_t buffer_size = period_size * 4 ;
+ 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 period_size_out = getFrameSize() * getSampleRate() / 1000 * 2;//1024 ;
snd_pcm_uframes_t buffer_size_out = period_size_out * 4 ;
@@ -324,7 +324,11 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
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);
snd_pcm_hw_params_free( hwParams );
snd_pcm_uframes_t val ;
@@ -346,8 +350,8 @@ AudioLayer::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, 0 ) < 0){
- _debugAlsa("Error while 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){
+ _debugAlsa("Error while opening playback device %s\n", pcm_c.c_str());
setErrorMessage( ALSA_PLAYBACK_DEVICE );
return false;
}
@@ -356,14 +360,15 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
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_RW_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_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_size( _PlaybackHandle, hwParams, buffer_size ) < 0) _debugAlsa(" Cannot set buffer size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_period_size_near( _PlaybackHandle, hwParams, &period_size , &dir) < 0) _debugAlsa(" Cannot set period size (%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));
@@ -389,7 +394,7 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
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();
@@ -402,26 +407,28 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
int
AudioLayer::write(void* buffer, int length)
{
+ //if(snd_pcm_state( _PlaybackHandle ) == SND_PCM_STATE_XRUN)
+ //handle_xrun_playback();
+ //_debugAlsa("avail = %d - toWrite = %d\n" , snd_pcm_avail_update( _PlaybackHandle ) , length / 2);
+
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _PlaybackHandle, length);
- //int err = snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
- int err = snd_pcm_writei( _PlaybackHandle , buffer , frames );
+ int err = snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
switch(err) {
case -EAGAIN:
- _debugAlsa("EAGAIN (%s)\n", snd_strerror( err ));
- //snd_pcm_resume( _PlaybackHandle );
+ _debugAlsa(" (%s)\n", snd_strerror( err ));
+ snd_pcm_resume( _PlaybackHandle );
break;
case -EPIPE:
_debugAlsa(" UNDERRUN (%s)\n", snd_strerror(err));
handle_xrun_playback();
- //snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
- //snd_pcm_writei( _PlaybackHandle , buffer , frames );
+ snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
break;
case -ESTRPIPE:
- _debugAlsa("ESTRPIPE (%s)\n", snd_strerror(err));
- //snd_pcm_resume( _PlaybackHandle );
+ _debugAlsa(" (%s)\n", snd_strerror(err));
+ snd_pcm_resume( _PlaybackHandle );
break;
case -EBADFD:
- _debugAlsa("EBADFD (%s)\n", snd_strerror( err ));
+ _debugAlsa(" (%s)\n", snd_strerror( err ));
break;
}
@@ -503,8 +510,7 @@ AudioLayer::handle_xrun_playback( void )
state = snd_pcm_status_get_state( status );
if( state == SND_PCM_STATE_XRUN )
{
- //snd_pcm_drop( _PlaybackHandle );
- _debug("Underrun\n");
+ snd_pcm_drop( _PlaybackHandle );
snd_pcm_prepare( _PlaybackHandle );
//snd_pcm_start( _PlaybackHandle );
}
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 3a0f549936..af585c00af 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -51,9 +51,10 @@ class AudioLayer {
* Constructor
* @param manager An instance of managerimpl
*/
- AudioLayer( ManagerImpl* manager )
+ AudioLayer( ManagerImpl* manager , int type )
: _manager(manager)
, _urgentBuffer( SIZEBUF )
+ , _layerType( type )
{
_inChannel = 1; // don't put in stereo
_outChannel = 1; // don't put in stereo
@@ -80,7 +81,7 @@ class AudioLayer {
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
- virtual bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize, int stream, std::string plugin) = 0;
+ virtual bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin) = 0;
/**
* Start the capture stream and prepare the playback stream.
@@ -226,8 +227,12 @@ class AudioLayer {
*/
bool getCurrentState( void ) { return _talk; }
+ int getLayerType( void ) { return _layerType; }
+
protected:
+ int _layerType;
+
/**
* Drop the pending frames and close the capture device
*/
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 9755ced1e3..5cb4dd9d2f 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -17,7 +17,6 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-
#include "pulselayer.h"
static pa_context *context = NULL;
@@ -25,15 +24,13 @@ static pa_mainloop_api *mainloop_api = NULL;
static pa_sample_spec sample_spec;
static pa_channel_map channel_map;
-static std::string stream_p = NULL;
-static std::string stream_r = NULL;
-
-
PulseLayer::PulseLayer(ManagerImpl* manager)
- : AudioLayer( manager )
+ : AudioLayer( manager , PULSEAUDIO )
+ , playback( NULL )
+ , record( NULL )
{
- playback = NULL;
- record = NULL;
+ _debug("Pulse audio constructor: Create context\n");
+ create_context();
}
// Destructor
@@ -45,15 +42,81 @@ PulseLayer::~PulseLayer (void)
// pa_stream_disconnect();
}
+void
+PulseLayer::create_context( void )
+{
+ pa_context_flags_t flag = PA_CONTEXT_NOAUTOSPAWN ;
+ int ret = 1;
+
+ // Instanciate a mainloop object
+ pa_mainloop *m;
+ if(!( m=pa_mainloop_new()))
+ _debug("Error while creating the mainloop\n");
+ mainloop_api = pa_mainloop_get_api( m );
+
+ // Instanciate a context
+ if( !(context = pa_context_new( mainloop_api , "SFLphone" )))
+ _debug("Error while creating the context\n");
+ pa_context_ref( context );
+
+ pa_context_set_state_callback(context, context_state_callback, NULL);
+
+ pa_context_connect( context, NULL , flag , NULL );
+
+ // Run the main loop
+ pa_mainloop_run(m , &ret);
+
+ _debug("Context creation done\n");
+}
+
void
PulseLayer::stream_state_callback( pa_stream* s, void* user_data )
{
_debug("The state of the stream changed\n");
}
+void
+PulseLayer::context_state_callback( pa_context* c, void* user_data )
+{
+ _debug("The state of the context changed\n");
+ assert(c);
+ switch(pa_context_get_state(c)){
+ case PA_CONTEXT_CONNECTING:
+ case PA_CONTEXT_AUTHORIZING:
+ case PA_CONTEXT_SETTING_NAME:
+ _debug("Waiting....\n");
+ break;
+ case PA_CONTEXT_READY:
+ pa_cvolume cv;
+ assert(c && !playback && !record);
+ _debug("Connection to PulseAudio server established\n");
+ playback = pa_stream_new( c, "playback stream" , &sample_spec, &channel_map);
+ record = pa_stream_new( c, "capture stream" , &sample_spec, &channel_map);
+
+ assert(playback);
+ assert(record);
- void
-PulseLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize)
+ // Set up the parameters required to open a (Callback)Stream:
+
+ pa_stream_set_state_callback(playback, stream_state_callback, NULL);
+ // Transferring Data - Asynchronous Mode
+ pa_stream_set_write_callback(playback, audioCallback, NULL);
+ pa_stream_connect_playback( playback, NULL , NULL , flag , NULL, NULL );
+
+ break;
+ case PA_CONTEXT_TERMINATED:
+ _debug("Context terminated\n");
+ break;
+ case PA_CONTEXT_FAILED:
+ default:
+ _debug(" Error : %s" , pa_strerror(pa_context_errno(context)));
+ exit(1);
+ }
+}
+
+
+ bool
+PulseLayer::openDevice(int indexIn, int indexOut, int sampleRate, int frameSize , int stream, std::string plugin)
{
_sampleRate = sampleRate;
_frameSize = frameSize;
@@ -68,11 +131,13 @@ PulseLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize
_debug(" : nb channel in=%2d, out=%2d\n", _inChannel, _outChannel);
_debug(" : sample rate=%5d\n", _sampleRate );
+ /*
assert(context);
switch(pa_context_get_state(context)){
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
+ _debug("Waiting....\n");
break;
case PA_CONTEXT_READY:
pa_cvolume cv;
@@ -99,7 +164,22 @@ PulseLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize
default:
_debug(" Error : %s" , pa_strerror(pa_context_errno(context)));
exit(1);
- }
+ }*/
+}
+
+void
+PulseLayer::closeCaptureStream( void )
+{
+}
+
+void
+PulseLayer::closePlaybackStream( void )
+{
+}
+
+int
+PulseLayer::playSamples(void* buffer, int toCopy, bool isTalking)
+{
}
int
@@ -137,6 +217,23 @@ PulseLayer::isStreamStopped (void)
{
}
+ void
+PulseLayer::startStream (void)
+{
+ _debug("Start stream\n");
+}
+
+ void
+PulseLayer::stopStream (void)
+{
+ _debug("Stop stream\n");
+}
+
+ bool
+PulseLayer::isStreamActive (void)
+{
+}
+
void
PulseLayer::audioCallback ( pa_stream* s, size_t bytes, void* user_data )
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index 9fc0d6a3d5..6076711f88 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -30,13 +30,22 @@ class PulseLayer : public AudioLayer {
PulseLayer(ManagerImpl* manager);
~PulseLayer(void);
- /*
- * @param indexIn
- * @param indexOut
- * @param sampleRate
- * @param frameSize
+ /**
+ * Check if no devices are opened, otherwise close them.
+ * Then open the specified devices by calling the private functions open_device
+ * @param indexIn The number of the card choosen for capture
+ * @param indexOut The number of the card choosen for playback
+ * @param sampleRate The sample rate
+ * @param frameSize The frame size
+ * @param stream To indicate which kind of stream you want to open
+ * SFL_PCM_CAPTURE
+ * SFL_PCM_PLAYBACK
+ * SFL_PCM_BOTH
+ * @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
- void openDevice(int, int, int, int);
+ bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize , int stream, std::string plugin) ;
+
+
void startStream(void);
void stopStream(void);
bool isStreamActive(void);
@@ -56,9 +65,19 @@ class PulseLayer : public AudioLayer {
int getMic(void *, int);
void flushMic();
+ /**
+ * Send samples to the audio device.
+ * @param buffer The buffer containing the data to be played ( voice and DTMF )
+ * @param toCopy The number of samples, in bytes
+ * @param isTalking If whether or not the conversation is running
+ * @return int The number of bytes played
+ */
+ int playSamples(void* buffer, int toCopy, bool isTalking) ;
+
static void audioCallback ( pa_stream* s, size_t bytes, void* user_data );
static void stream_state_callback( pa_stream* s, void* user_data );
+ static void context_state_callback( pa_context* c, void* user_data );
/**
* Scan the sound card available on the system
@@ -96,10 +115,20 @@ class PulseLayer : public AudioLayer {
* Get the current audio plugin.
* @return std::string The name of the audio plugin
*/
- std::string getAudioPlugin( void ) { return ""; }
+ std::string getAudioPlugin( void ) { return "default"; }
private:
- void closeStream (void);
+ /**
+ * Drop the pending frames and close the capture device
+ */
+ void closeCaptureStream( void );
+
+ /**
+ * Drop the pending frames and close the playback device
+ */
+ void closePlaybackStream( void );
+
+ void create_context( void );
pa_stream* playback;
pa_stream* record;
diff --git a/src/global.h b/src/global.h
index af95fbc6dd..f6e9cd00f0 100644
--- a/src/global.h
+++ b/src/global.h
@@ -122,4 +122,9 @@ typedef short int16;
#define ALSA_PLAYBACK_DEVICE 0x0010 /** Error while opening playback device */
#define NETWORK_UNREACHABLE 0x0011 /** Network unreachable */
+#define ALSA 0
+#define PULSEAUDIO 1
+//#define AUDIODRIVER ALSA
+#define CHECK_INTERFACE( layer , api ) (layer == api)
+
#endif // __GLOBAL_H__
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 6b9f06e12f..b100deefd9 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -1433,8 +1433,8 @@ ManagerImpl::getCurrentAudioOutputPlugin( void )
ManagerImpl::initAudioDriver(void)
{
_debugInit("AudioLayer Creation");
- //_audiodriver = new AudioLayer(this);
- _audiodriver = new AlsaLayer( this );
+ //_audiodriver = new AlsaLayer( this );
+ _audiodriver = new PulseLayer( this );
if (_audiodriver == 0) {
_debug("Init audio driver error\n");
} else {
@@ -1451,6 +1451,10 @@ ManagerImpl::initAudioDriver(void)
void
ManagerImpl::selectAudioDriver (void)
{
+ int layer = _audiodriver->getLayerType();
+ _debug("Audio layer type: %i\n" , layer);
+
+#if CHECK_INTERFACE( layer , ALSA )
std::string alsaPlugin = getConfigString( AUDIO , ALSA_PLUGIN );
int numCardIn = getConfigInt( AUDIO , ALSA_CARD_ID_IN );
int numCardOut = getConfigInt( AUDIO , ALSA_CARD_ID_OUT );
@@ -1478,6 +1482,15 @@ ManagerImpl::selectAudioDriver (void)
_audiodriver->openDevice( numCardIn , numCardOut, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin );
if( _audiodriver -> getErrorMessage() != -1 )
notifyErrClient( _audiodriver -> getErrorMessage());
+#else
+
+ _debug(" Pulse audio driver \n");
+ _audiodriver->openDevice( numCardIn , numCardOut, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin );
+ if( _audiodriver -> getErrorMessage() != -1 )
+ notifyErrClient( _audiodriver -> getErrorMessage());
+
+#endif
+
}
/**
--
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