The very beginning of pulseaudio nativ interface

Doesn't conpile yet

src/audio/Makefile.am

@@ -23,7 +23,7 @@ endif
 SUBDIRS = codecs
 
 libaudio_la_SOURCES =    audiofile.cpp	 tonelist.cpp    \
-audiortp.cpp         dtmf.cpp   tone.cpp  audiolayer.cpp audiodevice.cpp  dtmfgenerator.cpp   \
+audiortp.cpp         dtmf.cpp   tone.cpp  audiolayer.cpp audiolayer-pulse.cpp audiodevice.cpp  dtmfgenerator.cpp   \
 tonegenerator.cpp   codecDescriptor.cpp \
 audioloop.cpp ringbuffer.cpp $(SPEEX_SOURCES_CPP)
 
@@ -31,7 +31,7 @@ AM_CXXFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/libs $(libccext2_CFLAGS) $
  	      -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 audiodevice.h \
+ tonelist.h         audiortp.h   audiolayer.h audiolayer.h audiodevice.h \
  dtmfgenerator.h         tonegenerator.h     \
  codecDescriptor.h    dtmf.h tone.h 
 

src/audio/audiolayer-pulse.cpp

+/*
+ *  Copyright (C) 2005 Savoir-Faire Linux inc.
+ *  Author: Yan Morin <yan.morin@savoirfairelinux.com>
+ *  Author: Jerome Oufella <jerome.oufella@savoirfairelinux.com> 
+ *  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 <stdio.h>
+#include <stdlib.h>
+
+#include <pulse/pulseaudio.h>
+
+#include "audiolayer-pulse.h"
+#include "../global.h"
+#include "../manager.h"
+
+static pa_context *context = NULL;
+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)
+  : _urgentRingBuffer(SIZEBUF)
+  , _mainSndRingBuffer(SIZEBUF)
+  , _micRingBuffer(SIZEBUF)
+  , _defaultVolume(PA_VOLUME_NORM)
+  , playback(NULL)  
+    , record(NULL)  
+  , _errorMessage("")
+    , _manager(manager)
+{
+  _sampleRate = 8000;
+
+  _inChannel  = 1; // don't put in stereo
+  _outChannel = 1; // don't put in stereo
+  _echoTesting = false;
+}
+
+// Destructor
+PulseLayer::~PulseLayer (void) 
+{ 
+  assert(mainloop_api);
+  mainloop_api->quit( mainloop_api, 0 );
+  // pa_stream_flush();
+  // pa_stream_disconnect();
+}
+
+
+  void
+PulseLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize) 
+{
+  _sampleRate = sampleRate;
+  _frameSize = frameSize;	
+
+  sample_spec.rate = sampleRate;
+  sample_spec.format = PA_SAMPLE_S16LE; 
+  sample_spec.channels = 1; 
+  channel_map.channels = 1; 
+
+
+  _debug(" Setting PulseLayer: device     in=%2d, out=%2d\n", indexIn, indexOut);
+  _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:
+      break;
+    case PA_CONTEXT_READY:
+      pa_cvolume cv;
+      assert(context && !playback && !record);
+      _debug("Connection to PulseAudio server established\n");	
+      playback = pa_stream_new( context, "playback stream" , &sample_spec, &channel_map);
+      record = pa_stream_new( context, "capture stream" , &sample_spec, &channel_map);
+
+      assert(playback);
+      assert(record);
+
+      // 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 , 0 , NULL, NULL );
+
+      break;
+    case PA_CONTEXT_TERMINATED:
+      quit(0);
+      break;
+    case PA_CONTEXT_FAILED:
+    default:
+      _debug(" Error : %s" , pa_strerror(pa_context_errno(c)));
+      quit(1);
+  }
+}
+
+PulseLayer::stream_state_callback( void )
+{
+  _debug("The state of the stream changed\n");
+}
+
+  int 
+PulseLayer::putMain(void* buffer, int toCopy)
+{
+}
+
+  void
+PulseLayer::flushMain()
+{
+}
+
+  int
+PulseLayer::putUrgent(void* buffer, int toCopy)
+{
+}
+
+  int
+PulseLayer::canGetMic()
+{
+}
+
+  int 
+PulseLayer::getMic(void *buffer, int toCopy)
+{
+}
+
+  void
+PulseLayer::flushMic()
+{
+}
+
+  bool
+PulseLayer::isStreamStopped (void) 
+{
+}
+
+void
+PulseLayer::toggleEchoTesting() {
+}
+
+  int 
+PulseLayer::audioCallback ()
+{ 
+  _debug("Audio callback: New data may be written to the stream\n");
+  // pa_stream_write
+  // pa_stream_peek ( to read the next fragment from the buffer ) / pa_stream_drop( to remove the data from the buffer )
+}
+

src/audio/audiolayer-pulse.h

+/*
+ *  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_LAYER_PULSE_H
+#define _AUDIO_LAYER_PULSE_H
+
+#include <cc++/thread.h> // for ost::Mutex
+
+
+#include "../global.h"
+#include "ringbuffer.h"
+#include "audiodevice.h"
+
+#define FRAME_PER_BUFFER	160
+
+class RingBuffer;
+class ManagerImpl;
+
+class PulseLayer {
+	public:
+		PulseLayer(ManagerImpl* manager);
+		~PulseLayer(void);
+
+		/*
+		 * @param indexIn
+		 * @param indexOut
+		 * @param sampleRate
+		 * @param frameSize
+		 */
+		void openDevice(int, int, int, int);
+		void startStream(void);
+		void stopStream(void);
+		void sleep(int);
+		bool hasStream(void);
+		bool isStreamActive(void);
+		bool isStreamStopped(void);
+
+		void flushMain();
+		int putMain(void* buffer, int toCopy);
+		int putUrgent(void* buffer, int toCopy);
+		int canGetMic();
+		int getMic(void *, int);
+		void flushMic();
+
+		int audioCallback ();
+
+
+
+		void setErrorMessage(const std::string& error) { _errorMessage = error; }
+		std::string getErrorMessage() { return _errorMessage; }
+
+		/**
+		 * Get the sample rate of PulseLayer
+		 * accessor only
+		 */
+		unsigned int getSampleRate() { return _sampleRate; }
+		unsigned int getFrameSize() { return _frameSize; }
+
+		/**
+		 * Toggle echo testing on/off
+		 */
+		void toggleEchoTesting();
+
+	private:
+		void closeStream (void);
+		RingBuffer _urgentRingBuffer;
+		RingBuffer _mainSndRingBuffer;
+		RingBuffer _micRingBuffer;
+		ManagerImpl* _manager; // augment coupling, reduce indirect access
+		// a PulseLayer can't live without manager
+
+		pa_stream* playback;
+		pa_stream* record;
+
+		/**
+		 * Sample Rate SFLphone should send sound data to the sound card 
+		 * The value can be set in the user config file- now: 44100HZ
+		 */
+		unsigned int _sampleRate;
+		
+		/**
+ 		 * Length of the sound frame we capture or read in ms
+ 		 * The value can be set in the user config file - now: 20ms
+ 		 */	 		
+		unsigned int _frameSize;
+		
+		/**
+		 * Input channel (mic) should be 1 mono
+		 */
+		unsigned int _inChannel; // mic
+
+		/**
+		 * Output channel (stereo) should be 1 mono
+		 */
+		unsigned int _outChannel; // speaker
+
+		/**
+		 * Default volume for incoming RTP and Urgent sounds.
+		 */
+		//unsigned short _defaultVolume; // 100
+		pa_volume_t _defaultVolume;
+
+		/**
+		 * Echo testing or not
+		 */
+		bool _echoTesting;
+
+		std::string _errorMessage;
+		ost::Mutex _mutex;
+
+		float *table_;
+		int tableSize_;
+		int leftPhase_;
+};
+
+#endif // _AUDIO_LAYER_PULSE_H_
+