diff --git a/src/audio/Makefile.am b/src/audio/Makefile.am
index 13bb710847217ab361407cd1306492841f5b7030..c65f297c5a4906131d65769d2e21be80c7d80783 100644
--- a/src/audio/Makefile.am
+++ b/src/audio/Makefile.am
@@ -23,15 +23,15 @@ endif
SUBDIRS = codecs
libaudio_la_SOURCES = audiofile.cpp tonelist.cpp \
-audiortp.cpp dtmf.cpp tone.cpp alsalayer.cpp pulselayer.cpp audiodevice.cpp dtmfgenerator.cpp \
-tonegenerator.cpp codecDescriptor.cpp \
+audiortp.cpp audiostream.cpp dtmf.cpp tone.cpp alsalayer.cpp pulselayer.cpp audiodevice.cpp dtmfgenerator.cpp \
+tonegenerator.cpp codecDescriptor.cpp samplecache.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)
-noinst_HEADERS = audioloop.h common.h ringbuffer.h audiofile.h \
+noinst_HEADERS = audioloop.h audiostream.h common.h ringbuffer.h audiofile.h \
tonelist.h audiortp.h audiolayer.h alsalayer.h pulselayer.h audiodevice.h \
- dtmfgenerator.h tonegenerator.h \
+ dtmfgenerator.h tonegenerator.h samplecache.h \
codecDescriptor.h dtmf.h tone.h
diff --git a/src/audio/audiostream.cpp b/src/audio/audiostream.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..02deda2559781a3a81756e31af42bf144c67010a
--- /dev/null
+++ b/src/audio/audiostream.cpp
@@ -0,0 +1,95 @@
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <audiostream.h>
+
+static pa_channel_map channel_map ;
+
+AudioStream::AudioStream( pa_context* context, int type, std::string desc )
+{
+ _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;
+
+ _audiostream = createStream( context );
+}
+
+AudioStream::~AudioStream()
+{
+}
+
+ void
+AudioStream::stream_state_callback( pa_stream* s, void* user_data )
+{
+ _debug("The state of the stream changed\n");
+ assert(s);
+ switch(pa_stream_get_state(s)){
+ case PA_STREAM_CREATING:
+ case PA_STREAM_TERMINATED:
+ _debug("Stream is creating...\n");
+ break;
+ case PA_STREAM_READY:
+ _debug("Stream successfully created\n");
+ break;
+ case PA_STREAM_FAILED:
+ default:
+ _debug("Stream error: %s\n" , pa_strerror(pa_context_errno(pa_stream_get_context(s))));
+ break;
+ }
+}
+
+
+ pa_stream*
+AudioStream::createStream( pa_context* c )
+{
+ _debug("Creating %s stream...\n" , _streamDescription.c_str());
+ pa_stream* s;
+
+ assert(pa_sample_spec_valid(&sample_spec));
+ assert(pa_channel_map_valid(&channel_map));
+
+ if( !( s = pa_stream_new( c, _streamDescription.c_str() , &sample_spec, &channel_map ) ) )
+ _debug("%s: pa_stream_new() failed : %s\n" , _streamDescription.c_str(), pa_strerror( pa_context_errno( c)));
+
+ assert( s );
+
+ if( _streamType == PLAYBACK_STREAM ){
+ pa_stream_connect_playback( s , NULL , NULL , flag , NULL, NULL );
+ //pa_stream_set_write_callback( s , audioCallback, this);
+ }
+ else if( _streamType == CAPTURE_STREAM ){
+ pa_stream_connect_record( s , NULL , NULL , flag );
+ //pa_stream_set_read_callback( s , audioCallback, this);
+ }
+ else if( _streamType == UPLOAD_STREAM ){
+ //pa_stream_connect_upload( s , 1024 );
+ }
+ else{
+ _debug( "Stream type unknown \n");
+ }
+
+ pa_stream_set_state_callback( s , stream_state_callback, NULL);
+
+ return s;
+}
+
diff --git a/src/audio/audiostream.h b/src/audio/audiostream.h
new file mode 100644
index 0000000000000000000000000000000000000000..a4464500f58759463daf1affaf9ef64b36169565
--- /dev/null
+++ b/src/audio/audiostream.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef _AUDIO_STREAM_H
+#define _AUDIO_STREAM_H
+
+#include <pulse/pulseaudio.h>
+#include <string>
+
+#include "../global.h"
+#include "ringbuffer.h"
+#include "audioloop.h"
+
+enum STREAM_TYPE {
+ PLAYBACK_STREAM,
+ CAPTURE_STREAM,
+ UPLOAD_STREAM
+};
+
+
+class AudioStream {
+ public:
+ AudioStream(pa_context* context , int type, std::string desc);
+ ~AudioStream();
+
+ int putMain( void* buffer , int toCopy );
+ int putUrgent( void* buffer , int toCopy );
+
+ pa_stream* pulseStream(){ return _audiostream; }
+
+ private:
+ pa_stream* createStream( pa_context* c );
+
+ static void stream_state_callback( pa_stream* s, void* user_data );
+ static void audioCallback ( pa_stream* s, size_t bytes, void* userdata );
+ void write( void );
+
+ int _streamType;
+ std::string _streamDescription;
+
+
+ pa_stream* _audiostream;
+ pa_stream_flags_t flag;
+ pa_sample_spec sample_spec ;
+ //pa_channel_map channel_map;
+
+};
+
+#endif // _AUDIO_STREAM_H
diff --git a/src/audio/dtmf.cpp b/src/audio/dtmf.cpp
index a5e9f7fa6b2e6cd796033b3234dbf7d90df2a8aa..faabf1d69aad7fe79369b669e157756066534d15 100644
--- a/src/audio/dtmf.cpp
+++ b/src/audio/dtmf.cpp
@@ -77,3 +77,4 @@ DTMF::generateDTMF (SFLDataFormat* buffer, size_t n)
return false;
}
}
+
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 6e1be72dc20e747842d1b9640b7d5723d51c069b..121c7f0048913e25b6d81a1fbfad51a69eef6430 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -19,23 +19,13 @@
#include "pulselayer.h"
-static pa_context *context = NULL;
-static pa_stream* playback = NULL;
-static pa_stream* record = NULL;
-static pa_mainloop_api *mainloop_api = NULL;
-static pa_threaded_mainloop *m = NULL;
-static pa_sample_spec sample_spec ;
+int framesPerBuffer = 1024;
-static pa_channel_map channel_map;
-
-static std::string pcm_p, pcm_r;
-
-int framesPerBuffer = 882;
-
- PulseLayer::PulseLayer(ManagerImpl* manager)
+PulseLayer::PulseLayer(ManagerImpl* manager)
: AudioLayer( manager , PULSEAUDIO )
- , _mainSndRingBuffer( SIZEBUF )
+ , _urgentRingBuffer( SIZEBUF)
+ ,_mainSndRingBuffer( SIZEBUF )
{
_debug("Pulse audio constructor: Create context\n");
}
@@ -43,11 +33,10 @@ int framesPerBuffer = 882;
// Destructor
PulseLayer::~PulseLayer (void)
{
- assert(mainloop_api);
- //mainloop_api->quit( mainloop_api, 0 );
- pa_stream_flush( playback , NULL, NULL);
- pa_stream_disconnect( playback );
- pa_context_disconnect(context);
+ delete playback;
+ delete record;
+ ////delete cache;
+ pa_context_disconnect(context);
}
void
@@ -75,33 +64,12 @@ PulseLayer::connectPulseServer( void )
_debug("Context creation done\n");
}
- void
-PulseLayer::stream_state_callback( pa_stream* s, void* user_data )
-{
- _debug("The state of the stream changed\n");
- assert(s);
- switch(pa_stream_get_state(s)){
- case PA_STREAM_CREATING:
- case PA_STREAM_TERMINATED:
- _debug("Stream is creating...\n");
- break;
- case PA_STREAM_READY:
- _debug("Stream successfully created\n");
- break;
- case PA_STREAM_FAILED:
- default:
- _debug("Stream error: %s\n" , pa_strerror(pa_context_errno(pa_stream_get_context(s))));
- break;
- }
-}
-
void
PulseLayer::context_state_callback( pa_context* c, void* user_data )
{
_debug("The state of the context changed\n");
PulseLayer* pulse = (PulseLayer*)user_data;
- //pa_stream_flags_t flag = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE ;
- assert(c && m);
+ assert(c && pulse->m);
switch(pa_context_get_state(c)){
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
@@ -110,7 +78,6 @@ PulseLayer::context_state_callback( pa_context* c, void* user_data )
break;
case PA_CONTEXT_READY:
pa_cvolume cv;
- assert(c && !playback && !record);
pulse->createStreams( c );
_debug("Connection to PulseAudio server established\n");
break;
@@ -119,7 +86,7 @@ PulseLayer::context_state_callback( pa_context* c, void* user_data )
break;
case PA_CONTEXT_FAILED:
default:
- _debug(" Error : %s" , pa_strerror(pa_context_errno(context)));
+ _debug(" Error : %s" , pa_strerror(pa_context_errno(c)));
exit(1);
}
}
@@ -127,47 +94,13 @@ PulseLayer::context_state_callback( pa_context* c, void* user_data )
void
PulseLayer::createStreams( pa_context* c )
{
- _debug( " Create Audio Streams \n ");
- //pa_stream_flags_t flag = PA_STREAM_AUTO_TIMING_UPDATE;
- pa_stream_flags_t flag = PA_STREAM_FIX_RATE;
- sample_spec.format = PA_SAMPLE_S16LE;
- sample_spec.rate = 44100;
- sample_spec.channels = 1;
- channel_map.channels = 1;
-
- if( !pa_sample_spec_valid( &sample_spec ) ){
- _debug("Invalid sample specifications\n");
- exit(0);
- }
-
- if(!(playback = pa_stream_new( c, "SFLphone out" , &sample_spec, &channel_map))){
- _debug("Playback: pa_stream_new() failed : %s\n" , pa_strerror( pa_context_errno( c)));
- exit(0);
- }
-
- if(!(record = pa_stream_new( c, "SFLphone Mic" , &sample_spec, &channel_map))){
- _debug("Capture: pa_stream_new() failed : %s\n" , pa_strerror( pa_context_errno( c)));
- exit(0);
- }
-
- assert(playback);
- assert(record);
- assert(m);
-
- // Set up the parameters required to open a (Callback)Stream:
-
- pa_stream_connect_playback( playback, NULL , NULL , flag , NULL, NULL );
- pa_stream_set_state_callback(playback, stream_state_callback, NULL);
- // Transferring Data - Asynchronous Mode
- pa_stream_set_write_callback(playback, audioCallback, this);
+ playback = new AudioStream(c, PLAYBACK_STREAM, "SFLphone out");
+ pa_stream_set_write_callback( playback->pulseStream() , audioCallback, this);
+ record = new AudioStream(c, CAPTURE_STREAM, "SFLphone in");
+ pa_stream_set_read_callback( record->pulseStream() , audioCallback, this);
+ //cache = new AudioStream(c, UPLOAD_STREAM, "Cache samples");
pa_threaded_mainloop_signal(m , 0);
-
- pa_stream_set_state_callback(record, stream_state_callback, NULL);
- // Transferring Data - Asynchronous Mode
- pa_stream_set_read_callback(record, audioCallback, this);
- pa_stream_connect_record( record, NULL , NULL , flag );
-
}
bool
@@ -223,7 +156,6 @@ PulseLayer::putMain(void* buffer, int toCopy)
_debug("Chopping sound, Ouch! RingBuffer full ?\n");
return _mainSndRingBuffer.Put(buffer, a, 100);
}
-
return 0;
}
@@ -235,6 +167,13 @@ PulseLayer::flushMain()
int
PulseLayer::putUrgent(void* buffer, int toCopy)
{
+ int a = _urgentRingBuffer.AvailForPut();
+ if ( a >= toCopy ) {
+ return _urgentRingBuffer.Put(buffer, toCopy, 100 );
+ } else {
+ return _urgentRingBuffer.Put(buffer, a, 100 );
+ }
+ return 0;
}
int
@@ -254,23 +193,22 @@ PulseLayer::flushMic()
{
}
- bool
+/* bool
PulseLayer::isStreamStopped (void)
{
}
-
+*/
void
PulseLayer::startStream (void)
{
_debug("Start stream\n");
- //pa_stream_cork( playback , 0, NULL, NULL);
}
void
PulseLayer::stopStream (void)
{
_debug("Stop stream\n");
- pa_stream_drop( playback );
+ //pa_stream_drop( playback );
}
bool
@@ -281,67 +219,47 @@ PulseLayer::isStreamActive (void)
void
PulseLayer::audioCallback ( pa_stream* s, size_t bytes, void* userdata )
{
- pa_threaded_mainloop_signal( m , 0);
-
- assert( s && bytes );
-
PulseLayer* pulse = (PulseLayer*) userdata;
+ assert( s && bytes );
pulse->write();
- //if(pa_stream_get_state(s) == PA_STREAM_READY )
- //{
- //if( bytes > 0 ){
- // pa_stream_write( s, user_data, bytes, pa_xfree, 0, PA_SEEK_RELATIVE );
- //}
- //}
- // pa_stream_write
- // // pa_stream_peek ( to read the next fragment from the buffer ) / pa_stream_drop( to remove the data from the buffer )
- //
- // int toPut;
- // int urgentAvail; // number of data right and data left
- // int micAvailPut;
- //
- // // AvailForGet tell the number of chars inside the buffer
- // // framePerBuffer are the number of data for one channel (left)
- // urgentAvail = _urgentRingBuffer.AvailForGet();
- // if (urgentAvail > 0) {
- // // Urgent data (dtmf, incoming call signal) come first.
- // toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
- // _urgentRingBuffer.Get(out, toGet, spkrVolume);
- // // Consume the regular one as well (same amount of bytes)
- // _mainSndRingBuffer.Discard(toGet);
- // } else {
- // AudioLoop* tone = _manager->getTelephoneTone();
- // if ( tone != 0) {
- // tone->getNext(out, framesPerBuffer, spkrVolume);
- // } else if ( (tone=_manager->getTelephoneFile()) != 0 ) {
- // tone->getNext(out, framesPerBuffer, spkrVolume);
- // } else {
-
- // }
- // }
- //
- // // Additionally handle the mic's audio stream
- // micAvailPut = _micRingBuffer.AvailForPut();
- // toPut = (micAvailPut <= (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? micAvailPut : framesPerBuffer * sizeof(SFLDataFormat);
- // //_debug("AL: Nb sample: %d char, [0]=%f [1]=%f [2]=%f\n", toPut, in[0], in[1], in[2]);
- // _micRingBuffer.Put(in, toPut, micVolume);
-
}
void
PulseLayer::write( void )
{
int toGet;
+ int urgentAvail; // number of data right and data left
int normalAvail; // number of data right and data left
SFLDataFormat* out = (SFLDataFormat*)malloc(framesPerBuffer * sizeof(SFLDataFormat));
- normalAvail = _mainSndRingBuffer.AvailForGet();
- toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
- if (toGet) {
- _mainSndRingBuffer.Get(out, toGet, 100);
- _debug("Write %i bytes\n" , toGet);
- pa_stream_write( playback , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
+ urgentAvail = _urgentRingBuffer.AvailForGet();
+ if (urgentAvail > 0) {
+ // Urgent data (dtmf, incoming call signal) come first.
+ _debug("Play urgent!\n");
+ toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ _urgentRingBuffer.Get(out, toGet, 100);
+ // Consume the regular one as well (same amount of bytes)
_mainSndRingBuffer.Discard(toGet);
- } else {
- bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
}
-}
+ else
+ {
+ AudioLoop* tone = _manager->getTelephoneTone();
+ if ( tone != 0) {
+ tone->getNext(out, framesPerBuffer, 100);
+ toGet = framesPerBuffer;
+ } else if ( (tone=_manager->getTelephoneFile()) != 0 ) {
+ tone->getNext(out, framesPerBuffer, 100);
+ toGet = framesPerBuffer;
+ } else {
+ normalAvail = _mainSndRingBuffer.AvailForGet();
+ toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ if (toGet) {
+ _mainSndRingBuffer.Get(out, toGet, 100);
+ _debug("Write %i bytes\n" , toGet);
+ _mainSndRingBuffer.Discard(toGet);
+ } else {
+ bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
+ }
+ }
+ }
+ pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
+}
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index a63ce2319b635c524061e1722e5c4a3c658dd173..1593862c1eded324eef11c55d6e1d5b1d30e69fe 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -21,6 +21,7 @@
#define _PULSE_LAYER_H
#include "audiolayer.h"
+#include "audiostream.h"
class RingBuffer;
class ManagerImpl;
@@ -48,8 +49,6 @@ class PulseLayer : public AudioLayer {
void startStream(void);
void stopStream(void);
- bool isStreamActive(void);
- bool isStreamStopped(void);
/**
* Check if the capture is running
@@ -57,6 +56,7 @@ class PulseLayer : public AudioLayer {
* false otherwise
*/
bool isCaptureActive( void ) { return true; }
+ bool isStreamActive (void);
void flushMain();
int putMain(void* buffer, int toCopy);
@@ -76,7 +76,6 @@ class PulseLayer : public AudioLayer {
static void audioCallback ( pa_stream* s, size_t bytes, void* userdata );
- static void stream_state_callback( pa_stream* s, void* user_data );
static void context_state_callback( pa_context* c, void* user_data );
/**
@@ -117,6 +116,8 @@ class PulseLayer : public AudioLayer {
*/
std::string getAudioPlugin( void ) { return "default"; }
+ //pa_stream* getCacheStream( void ) { return caching; }
+
private:
/**
* Drop the pending frames and close the capture device
@@ -132,10 +133,17 @@ class PulseLayer : public AudioLayer {
void connectPulseServer( void );
+ /** Ringbuffers for data */
RingBuffer _mainSndRingBuffer;
+ RingBuffer _urgentRingBuffer;
+
+ /** PulseAudio streams and context */
+ pa_context* context;
+ pa_threaded_mainloop* m;
- //pa_stream* playback;
- //pa_stream* record;
+ AudioStream* playback;
+ AudioStream* record;
+ AudioStream* cache;
};
diff --git a/src/audio/samplecache.cpp b/src/audio/samplecache.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..2d15d82c64a682c892ddf003039bff5926a64fb8
--- /dev/null
+++ b/src/audio/samplecache.cpp
@@ -0,0 +1,37 @@
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <samplecache.h>
+
+SampleCache::SampleCache( pa_stream* s )
+{
+ _stream = s ;
+}
+
+SampleCache::~SampleCache()
+{
+ //delete _pulse;
+}
+
+bool
+SampleCache::uploadSample( SFLDataFormat* buffer , size_t size )
+{
+ //pa_stream_write( pulse->caching , buffer , size , pa_xfree, 0 , PA_SEEK_RELATIVE);
+ //pa_stream_finish_upload( pulse->caching );
+}
diff --git a/src/audio/samplecache.h b/src/audio/samplecache.h
new file mode 100644
index 0000000000000000000000000000000000000000..655f791197134d2dddc60736e40d6ebd266de4d1
--- /dev/null
+++ b/src/audio/samplecache.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef _SAMPLE_CACHE_H
+#define _SAMPLE_CACHE_H
+
+#include <pulse/pulseaudio.h>
+#include <audiolayer.h>
+
+class SampleCache {
+
+ public:
+ SampleCache( pa_stream* stream );
+ ~SampleCache();
+
+ bool uploadSample( SFLDataFormat* buffer, size_t size );
+ bool removeSample( );
+ bool isSampleCached( );
+
+ private:
+ pa_stream* _stream;
+
+};
+
+#endif // _SAMPLE_CACHE_H
diff --git a/src/global.h b/src/global.h
index f6e9cd00f09f1a9476535cf569452198d965b7bb..a89b070f3e31b868b4fb8f48e21787a4b0b84ec0 100644
--- a/src/global.h
+++ b/src/global.h
@@ -124,7 +124,6 @@ typedef short int16;
#define ALSA 0
#define PULSEAUDIO 1
-//#define AUDIODRIVER ALSA
-#define CHECK_INTERFACE( layer , api ) (layer == api)
+#define CHECK_INTERFACE( layer , api ) (layer != api)
#endif // __GLOBAL_H__
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 08f0360d4f35879c86b950b657e165a5d9042c39..7c176428fd3dfb19cacf4f1aae08b30c9243334c 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -473,6 +473,7 @@ ManagerImpl::playDtmf(char code, bool isTalking)
// numbers of int = length in milliseconds / 1000 (number of seconds)
// = number of seconds * SAMPLING_RATE by SECONDS
AudioLayer* audiolayer = getAudioDriver();
+ int layer = audiolayer->getLayerType();
// fast return, no sound, so no dtmf
if (audiolayer==0 || _dtmfKey == 0) { return false; }
@@ -496,7 +497,13 @@ ManagerImpl::playDtmf(char code, bool isTalking)
// Put buffer to urgentRingBuffer
// put the size in bytes...
// so size * 1 channel (mono) * sizeof (bytes for the data)
+#if CHECK_INTERFACE( layer , ALSA )
+ _debug("%i No good\n", layer);
audiolayer->playSamples(_buf, size * sizeof(SFLDataFormat), isTalking);
+#else
+ _debug("%i Good\n" , layer);
+ audiolayer->putUrgent( _buf, size * sizeof(SFLDataFormat) );
+#endif
}
returnValue = true;
@@ -822,13 +829,14 @@ ManagerImpl::ringtone()
_audiofile.start();
_toneMutex.leaveMutex();
#if CHECK_INTERFACE( layer, ALSA )
+ _debug()
int size = _audiofile.getSize();
SFLDataFormat output[ size ];
_audiofile.getNext(output, size , 100);
audiolayer->putUrgent( output , size );
#else
// pulseaudio code
- startStream();
+ audiolayer->startStream();
#endif
} else {
ringback();
@@ -1491,7 +1499,6 @@ 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 );
@@ -1514,13 +1521,15 @@ ManagerImpl::selectAudioDriver (void)
setConfig( AUDIO , ALSA_CARD_ID_OUT , ALSA_DFT_CARD_ID );
}
+#if CHECK_INTERFACE( layer , ALSA )
+ _debug("No good\n");
_debugInit(" AudioLayer Opening Device");
_audiodriver->setErrorMessage(-1);
_audiodriver->openDevice( numCardIn , numCardOut, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin );
if( _audiodriver -> getErrorMessage() != -1 )
notifyErrClient( _audiodriver -> getErrorMessage());
#else
-
+ _debug("Good\n");
_debug(" Pulse audio driver \n");
_audiodriver->openDevice( numCardIn , numCardOut, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin );
if( _audiodriver -> getErrorMessage() != -1 )