Merge branch 'master' of git+ssh://sflphone_org@sflphone.org/git/sflphone

configure.ac

@@ -35,6 +35,8 @@ AC_CONFIG_FILES([libs/Makefile \
 AC_CONFIG_FILES([src/Makefile \
   src/sflphone \
   src/audio/Makefile \
+  src/audio/codecs/Makefile
+  src/audio/codecs/ilbc/Makefile \
   src/config/Makefile \
   src/dbus/Makefile \
   src/zeroconf/Makefile])

sflphone-gtk/src/actions.c

@@ -587,7 +587,7 @@ sflphone_fill_codec_list()
   gchar** order = (gchar**)dbus_get_active_codec_list();
   gchar** details;
   gchar** pl;
-  
+
   for(pl=order; *order; order++)
   {
     codec_t * c = g_new0(codec_t, 1);
@@ -601,7 +601,7 @@ sflphone_fill_codec_list()
     c->_bandwidth = atof(details[3]);
     codec_list_add(c);
   }
-  
+ 
   for(pl=codecs; *codecs; codecs++)
   {
     details = (gchar **)dbus_codec_details(atoi(*codecs));

src/audio/CodecSpeex.h

-/*
- *  Copyright (C) 2005 Savoir-Faire Linux inc.
- *  Author: Yan Morin <yan.morin@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 __CODEC_SPEEX_H_
-#define __CODEC_SPEEX_H_
-
-#include "audiocodec.h"
-#include <speex/speex.h>
-
-class CodecSpeex : public AudioCodec
-{
-public:
-  CodecSpeex(int payload);
-  ~CodecSpeex();
-  int codecDecode(short *, unsigned char *, unsigned int);
-  int codecEncode(unsigned char *, short *, unsigned int);
-
-  // only for speex
-  int getFrameSize() { return _speex_frame_size; }
-
-private:
-  unsigned int _clockRate;
-  unsigned int _channel;
-
-  void initSpeex();
-  void terminateSpeex();
-  const SpeexMode* _speexModePtr;
-  SpeexBits  _speex_dec_bits;
-  SpeexBits  _speex_enc_bits;
-  void *_speex_dec_state;
-  void *_speex_enc_state;
-  int _speex_frame_size;
-};
-
-#endif // __CODEC_SPEEX_H_
-

src/audio/Makefile.am

-sflcodecdir = $(libdir)/sflphone/codecs/
+sflcodecdir = $(libdir)/sflphone/codecs
 
 noinst_LTLIBRARIES = libaudio.la
-noinst_PROGRAMS = libcodec_ulaw.so libcodec_alaw.so libcodec_gsm.so libcodec_speex.so
 
-libcodec_ulaw_so_SOURCES = ulaw.cpp
-libcodec_ulaw_so_CFLAGS = -fPIC -g -Wall 
-libcodec_ulaw_so_LDFLAGS = -shared -lc
+SUBDIRS = codecs
 
-libcodec_alaw_so_SOURCES = alaw.cpp
-libcodec_alaw_so_CFLAGS = -fPIC -g -Wall 
-libcodec_alaw_so_LDFLAGS = -shared -lc
-
-libcodec_gsm_so_SOURCES = gsmcodec.cpp
-libcodec_gsm_so_CFLAGS = -fPIC -g -Wall 
-libcodec_gsm_so_LDFLAGS = -shared -lc -lgsm
-
-libcodec_speex_so_SOURCES = CodecSpeex.cpp
-libcodec_speex_so_CFLAGS = -fPIC -g -Wall 
-libcodec_speex_so_LDFLAGS = -shared -lc -lspeex
-
-if USE_SPEEX
-SPEEX_SOURCES_CPP=CodecSpeex.cpp
-SPEEX_FLAG=-DHAVE_SPEEX
-SPEEX_LIB=-lspeex
-else
-SPEEX_SOURCES_CPP=
-SPEEX_FLAG=
-SPEEX_LIB=
-endif
-
-#SUBDIRS = ilbc
- 
-libaudio_la_SOURCES =      audiofile.cpp	 tonelist.cpp     \
+libaudio_la_SOURCES =    audiofile.cpp	 tonelist.cpp    \
 audiortp.cpp         dtmf.cpp   tone.cpp  audiolayer.cpp audiodevice.cpp  dtmfgenerator.cpp   \
 tonegenerator.cpp   codecDescriptor.cpp \
 audioloop.cpp ringbuffer.cpp $(SPEEX_SOURCES_CPP)
@@ -40,31 +13,8 @@ AM_CXXFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/libs $(libccext2_CFLAGS) $
  	      -DCODECS_DIR=\""$(sflcodecdir)"\"	
 libaudio_la_CPPFLAGS = $(SPEEX_FLAG)
 
-noinst_HEADERS = audioloop.h       common.h ringbuffer.h           audiofile.h  \
- tonelist.h         audiortp.h audiocodec.h    audiolayer.h audiodevice.h \
+noinst_HEADERS = audioloop.h    common.h ringbuffer.h    audiofile.h  \
+ tonelist.h         audiortp.h   audiolayer.h audiodevice.h \
  dtmfgenerator.h         tonegenerator.h     \
- codecDescriptor.h    dtmf.h tone.h \
- CodecSpeex.h
-
-install-exec-local:	install-libcodec_ulaw_so install-libcodec_alaw_so install-libcodec_gsm_so install-libcodec_speex_so
-uninstall-local:  uninstall-libcodec_ulaw_so uninstall-libcodec_alaw_so uninstall-libcodec_gsm_so uninstall-libcodec_speex_so
-
-install-libcodec_ulaw_so: libcodec_ulaw.so
-	mkdir -p $(sflcodecdir)
-	$(INSTALL_PROGRAM)  libcodec_ulaw.so $(sflcodecdir)
-install-libcodec_alaw_so: libcodec_alaw.so
-	$(INSTALL_PROGRAM)  libcodec_alaw.so $(sflcodecdir)
-install-libcodec_gsm_so: libcodec_gsm.so
-	$(INSTALL_PROGRAM)  libcodec_gsm.so $(sflcodecdir)
-install-libcodec_speex_so: libcodec_speex.so
-	$(INSTALL_PROGRAM)  libcodec_speex.so $(sflcodecdir)
+ codecDescriptor.h    dtmf.h tone.h 
 
-uninstall-libcodec_ulaw_so:
-	rm -f $(sflcodecdir)/libcodec_ulaw.so
-uninstall-libcodec_alaw_so:
-	rm -f $(sflcodecdir)/libcodec_alaw.so
-uninstall-libcodec_gsm_so:
-	rm -f $(sflcodecdir)/libcodec_gsm.so  
-uninstall-libcodec_speex_so:
-	rm -f $(sflcodecdir)/libcodec_speex.so  
-	rm -rf $(sflcodecdir) 

src/audio/audiofile.h

@@ -24,7 +24,7 @@
 #define __AUDIOFILE_H__
 
 #include "audioloop.h"
-#include "audiocodec.h"
+#include "codecs/audiocodec.h"
 
 /**
 	@author Yan Morin <yan.morin@savoirfairelinux.com>

src/audio/audiortp.cpp

@@ -97,13 +97,12 @@ AudioRtp::closeRtpSession () {
 // AudioRtpRTX Class                                                          //
 ////////////////////////////////////////////////////////////////////////////////
 AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
-	 : _fstream("/tmp/audio.dat", std::ofstream::binary|std::ios::out|std::ios::app)
+	 : _fstream("/tmp/audio.gsm", std::ofstream::binary|std::ios::out|std::ios::app)
 {
 	setCancel(cancelDeferred);
 	time = new ost::Time();
 	_ca = sipcall;
 	_sym = sym;
-	//std::string s = "snd.dat";
 	// AudioRtpRTX should be close if we change sample rate
 
 	//_codecSampleRate = _ca->getAudioCodec()->getClockRate();
@@ -141,7 +140,6 @@ AudioRtpRTX::~AudioRtpRTX () {
 	}
 	//_debug("terminate audiortprtx ended...\n");
 	_ca = 0;
-	//fd = fopen("snd_data", "wa");
 	if (!_sym) {
 		delete _sessionRecv; _sessionRecv = NULL;
 		delete _sessionSend; _sessionSend = NULL;
@@ -184,8 +182,7 @@ AudioRtpRTX::initAudioRtpSession (void)
 
 	try {
 		if (_ca == 0) { return; }
-		AudioCodec* audiocodec = loadCodec(_ca->getAudioCodec());
-		_codecSampleRate = audiocodec->getClockRate();	
+		_codecSampleRate = _audiocodec->getClockRate();	
 	
 		_debug("Init audio RTP session\n");
 		ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
@@ -217,12 +214,12 @@ AudioRtpRTX::initAudioRtpSession (void)
 			}
 
 			bool payloadIsSet = false;
-			if (audiocodec) {
-				if (audiocodec->hasDynamicPayload()) {
-					payloadIsSet = _sessionRecv->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
+			if (_audiocodec) {
+				if (_audiocodec->hasDynamicPayload()) {
+					payloadIsSet = _sessionRecv->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
 				} else {
-					payloadIsSet= _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
-					payloadIsSet = _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
+					payloadIsSet= _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload()));
+					payloadIsSet = _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload()));
 				}
 			}
 			_sessionSend->setMark(true);
@@ -235,22 +232,21 @@ AudioRtpRTX::initAudioRtpSession (void)
 			}
 
 			bool payloadIsSet = false;
-			if (audiocodec) {
-				if (audiocodec->hasDynamicPayload()) {
-					payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
+			if (_audiocodec) {
+				if (_audiocodec->hasDynamicPayload()) {
+					payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
 				} else {
-					payloadIsSet = _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
+					payloadIsSet = _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) _audiocodec->getPayload()));
 				}
 			}
 		}
-		unloadCodec(audiocodec);
 	} catch(...) {
 		_debugException("! ARTP Failure: initialisation failed");
 		throw;
 	}
 }
 
-AudioCodec*
+void
 AudioRtpRTX::loadCodec(int payload)
 {
         using std::cerr;
@@ -282,11 +278,12 @@ AudioRtpRTX::loadCodec(int payload)
         if(dlsym_error){
                 cerr << "Cannot load symbol create: " << dlsym_error << '\n';
         }
-        return create_codec();
+
+        _audiocodec = create_codec();
 }
 
 void
-AudioRtpRTX::unloadCodec(AudioCodec* audiocodec)
+AudioRtpRTX::unloadCodec()
 {
 	using std::cerr;
         destroy_t* destroy_codec = (destroy_t*)dlsym(handle_codec, "destroy");
@@ -294,7 +291,7 @@ AudioRtpRTX::unloadCodec(AudioCodec* audiocodec)
 	if(dlsym_error){
                 cerr << "Cannot load symbol destroy" << dlsym_error << '\n';
         }
-	destroy_codec(audiocodec);
+	destroy_codec(_audiocodec);
 	dlclose(handle_codec);
 }
 
@@ -302,7 +299,6 @@ AudioRtpRTX::unloadCodec(AudioCodec* audiocodec)
 void
 AudioRtpRTX::sendSessionFromMic(int timestamp)
 {
-	AudioCodec* audiocodec = loadCodec(_ca->getAudioCodec());
 // STEP:
 //   1. get data from mic
 //   2. convert it to int16 - good sample, good rate
@@ -316,25 +312,21 @@ try {
 	AudioLayer* audiolayer = Manager::instance().getAudioDriver();
 	if (!audiolayer) { _debug(" !ARTP: No audiolayer available for mic\n"); return; }
 
-	//AudioCodec* audiocodec = _ca->getAudioCodec();
-	if (!audiocodec) { _debug(" !ARTP: No audiocodec available for mic\n"); return; }
+	if (!_audiocodec) { _debug(" !ARTP: No audiocodec available for mic\n"); return; }
 
 	// we have to get 20ms of data from the mic *20/1000 = /50
 	int maxBytesToGet = _layerSampleRate * _layerFrameSize * sizeof(SFLDataFormat) / 1000;
-
 	// available bytes inside ringbuffer
 	int availBytesFromMic = audiolayer->canGetMic();
 
 	// take the lowest
 	int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
-        //printf("clock rate = %i\n", audiocodec->getClockRate());
 	// Get bytes from micRingBuffer to data_from_mic
 	int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
 	int nb_sample_up = nbSample;
-	int nbSamplesMax = _layerFrameSize * audiocodec->getClockRate() / 1000;
-	//_fstream.write((char*) _dataAudioLayer, nbSample);
+	int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000;
 	
-	nbSample = reSampleData(audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);	
+	nbSample = reSampleData(_audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);	
 	
 	toSIP = _intBufferDown;
 	
@@ -348,7 +340,7 @@ try {
 //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toSIP[0], toSIP[1], toSIP[2]);
 	// for the mono: range = 0 to RTP_FRAME2SEND * sizeof(int16)
 	// codecEncode(char *dest, int16* src, size in bytes of the src)
-	int compSize = audiocodec->codecEncode(_sendDataEncoded, toSIP, nbSample*sizeof(int16));
+	int compSize = _audiocodec->codecEncode(_sendDataEncoded, toSIP, nbSample*sizeof(int16));
 	//printf("jusqu'ici tout vas bien\n");
 
 	// encode divise by two
@@ -364,7 +356,6 @@ try {
 	_debugException("! ARTP: sending failed");
 	throw;
 }
-	unloadCodec(audiocodec);
 }
 
 
@@ -373,7 +364,6 @@ void
 AudioRtpRTX::receiveSessionForSpkr (int& countTime)
 {
 
-AudioCodec* audiocodec;
 
 if (_ca == 0) { return; }
 try {
@@ -396,12 +386,11 @@ try {
 	int payload = adu->getType(); // codec type
 	unsigned char* data  = (unsigned char*)adu->getData(); // data in char
 	unsigned int size = adu->getSize(); // size in char
+
 	
 	//_fstream.write((char*) data, size);
-	audiocodec = loadCodec(payload);
 	// Decode data with relevant codec
-	_codecSampleRate = audiocodec->getClockRate();
-	int max = (int)(_codecSampleRate * _layerFrameSize);
+	int max = (int)(_codecSampleRate * _layerFrameSize / 1000);
 
 	if ( size > max ) {
 		_debug("We have received from RTP a packet larger than expected: %s VS %s\n", size, max);
@@ -411,8 +400,11 @@ try {
 	
 	//printf("size = %i\n", size);
 
-	if (audiocodec != NULL) {
-		int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
+	if (_audiocodec != NULL) {
+		
+		int expandedSize = _audiocodec->codecDecode(_receiveDataDecoded, data, size);
+	//	printf("%i\n", expandedSize);
+	        //_fstream.write((char*) _receiveDataDecoded, );
 		//buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
 		int nbInt16 = expandedSize / sizeof(int16);
 		//nbInt16 represents the number of samples we just decoded
@@ -455,7 +447,6 @@ try {
 	throw;
 }
 
-  unloadCodec(audiocodec);
 
 }
 
@@ -527,6 +518,7 @@ AudioRtpRTX::run () {
 //mic, we receive from soundcard in stereo, and we send encoded
 //encoding before sending
 AudioLayer *audiolayer = Manager::instance().getAudioDriver();
+loadCodec(_ca->getAudioCodec());
 
 _layerFrameSize = audiolayer->getFrameSize(); // en ms
 _layerSampleRate = audiolayer->getSampleRate();	
@@ -569,7 +561,8 @@ try {
 		Thread::sleep(TimerPort::getTimer());
 		TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
 	}
-	//_fstream.close();
+	_fstream.close();
+ 	unloadCodec();
 	//_debug("stop stream for audiortp loop\n");
 	audiolayer->stopStream();
 } catch(std::exception &e) {

src/audio/audiortp.h

@@ -123,19 +123,22 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
 
 		/** Pointer on function to handle codecs **/
 		void* handle_codec;
-		
+
+		AudioCodec* _audiocodec;	
+
+	
 		/**
  		 * Load dynamically a codec (.so library) 
  		 * @param payload The payload of the codec you want to load
  		 * @return AudioCodec* A pointer on a audio codec object
  		 */
-		AudioCodec* loadCodec(int payload);
+		void loadCodec(int payload);
 		
 		/**
  		 * Destroy and close dynamically a codec (.so library) 
  		 * @param audiocodec The audio codec you want to unload
  		 */
-		void unloadCodec(AudioCodec* audiocodec);
+		void unloadCodec(void);
 };
 
 ///////////////////////////////////////////////////////////////////////////////

src/audio/codecDescriptor.cpp

@@ -22,11 +22,7 @@
 #include <iostream>
 #include <cstdlib>
 
-#include "audiocodec.h"
 #include "codecDescriptor.h"
-/*#ifdef HAVE_SPEEX
- #include "CodecSpeex.h"
-#endif*/
 
 CodecDescriptor::CodecDescriptor() 
 {
@@ -112,7 +108,7 @@ CodecDescriptor::getBitRate(CodecType payload)
     case PAYLOAD_CODEC_ILBC_20:
       return 15.2;
     case PAYLOAD_CODEC_ILBC_30:
-      return 15.2;
+      return 13.3;
 
   }
   return 0.0;

src/audio/codecDescriptor.h

@@ -50,7 +50,7 @@ typedef enum {
   PAYLOAD_CODEC_SPEEX_32000 = 112
 } CodecType;
 
-#include "audiocodec.h"
+#include "codecs/audiocodec.h"
 
 /* A codec is identified by its payload. A payload is associated with a name. */ 
 typedef std::map<CodecType, std::string> CodecMap;

src/audio/codecs/Makefile.am

+sflcodecdir = $(libdir)/sflphone/codecs
+
+noinst_PROGRAMS = libcodec_ulaw.so libcodec_alaw.so libcodec_gsm.so libcodec_speex.so libcodec_ilbc.so
+
+noinst_HEADERS = audiocodec.h
+
+libcodec_ulaw_so_SOURCES = ulaw.cpp
+libcodec_ulaw_so_CFLAGS = -fPIC -g -Wall
+libcodec_ulaw_so_LDFLAGS = -shared -lc
+
+libcodec_alaw_so_SOURCES = alaw.cpp
+libcodec_alaw_so_CFLAGS = -fPIC -g -Wall
+libcodec_alaw_so_LDFLAGS = -shared -lc
+
+libcodec_gsm_so_SOURCES = gsmcodec.cpp
+libcodec_gsm_so_CFLAGS = -fPIC -g -Wall
+libcodec_gsm_so_LDFLAGS = -shared -lc -lgsm
+
+libcodec_speex_so_SOURCES = speexcodec.cpp
+libcodec_speex_so_CFLAGS = -fPIC -g -Wall
+libcodec_speex_so_LDFLAGS = -shared -lc -lspeex
+
+libcodec_ilbc_so_SOURCES = ilbc.cpp
+libcodec_ilbc_so_CFLAGS = -fPIC -g -Wall
+libcodec_ilbc_so_LDFLAGS = -shared ilbc/*.o -lc
+
+SUBDIRS = ilbc
+
+install-exec-local:     install-libcodec_ulaw_so install-libcodec_alaw_so install-libcodec_gsm_so install-libcodec_speex_so install-libcodec_ilbc_so
+uninstall-local:  uninstall-libcodec_ulaw_so uninstall-libcodec_alaw_so uninstall-libcodec_gsm_so uninstall-libcodec_speex_so uninstall-libcodec_ilbc_so
+
+install-libcodec_ulaw_so: libcodec_ulaw.so
+	mkdir -p $(sflcodecdir)
+	$(INSTALL_PROGRAM)  libcodec_ulaw.so $(sflcodecdir)
+install-libcodec_alaw_so: libcodec_alaw.so
+	$(INSTALL_PROGRAM)  libcodec_alaw.so $(sflcodecdir)
+install-libcodec_gsm_so: libcodec_gsm.so
+	$(INSTALL_PROGRAM)  libcodec_gsm.so $(sflcodecdir)
+install-libcodec_speex_so: libcodec_speex.so
+	$(INSTALL_PROGRAM)  libcodec_speex.so $(sflcodecdir)
+install-libcodec_ilbc_so: libcodec_ilbc.so
+	$(INSTALL_PROGRAM)  libcodec_ilbc.so $(sflcodecdir)
+
+
+uninstall-libcodec_ulaw_so:
+	rm -f $(sflcodecdir)/libcodec_ulaw.so
+uninstall-libcodec_alaw_so:
+	rm -f $(sflcodecdir)/libcodec_alaw.so
+uninstall-libcodec_gsm_so:
+	rm -f $(sflcodecdir)/libcodec_gsm.so
+uninstall-libcodec_speex_so:
+	rm -f $(sflcodecdir)/libcodec_speex.so
+uninstall-libcodec_ilbc_so:
+	rm -f $(sflcodecdir)/libcodec_ilbc.so
+	rm -rf $(sflcodecdir)
+

src/audio/alaw.cpp → src/audio/codecs/alaw.cpp

@@ -18,7 +18,7 @@
  *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#include "common.h"
+#include "../common.h"
 #include "audiocodec.h"
 
 class Alaw : public AudioCodec {
@@ -31,7 +31,7 @@ public:
   		_channel   = 1;
 	}
 
-
+        virtual ~Alaw(){}
 	virtual int codecDecode (short *dst, unsigned char *src, unsigned int size) 
 	{
 		int16* end = dst+size;

src/audio/codecs/audiocodec.h

+#ifndef _CODEC_AUDIO_H
+#define _CODEC_AUDIO_H
+
+#include <string> 
+#include <iostream>
+#include <dlfcn.h>
+
+class AudioCodec {
+protected:
+  /** Holds SDP-compliant codec name */
+  std::string _codecName; // what we put inside sdp
+  
+  /** Clock rate or sample rate of the codec, in Hz */
+  unsigned int _clockRate;
+
+  /** Number of channel 1 = mono, 2 = stereo */
+  unsigned int _channel;
+
+private:
+  int _payload;
+  bool _hasDynamicPayload;
+
+public:
+    AudioCodec(int payload, const std::string &codecName)
+        : _codecName(codecName) {
+	_payload = payload;
+  	_clockRate = 8000; // default
+  	_channel   = 1; // default
+
+  	_hasDynamicPayload = (_payload >= 96 && _payload <= 127) ? true : false;
+
+}
+
+    virtual ~AudioCodec() {
+	}
+    /**
+     * @return the number of bytes decoded
+     */
+    virtual int codecDecode(short *, unsigned char *, unsigned int) = 0;
+    virtual int codecEncode(unsigned char *, short *, unsigned int) = 0;   
+
+
+  /** Value used for SDP negotiation */
+  std::string getCodecName() { return _codecName; }
+  int getPayload() { return _payload; }
+  bool hasDynamicPayload() { return _hasDynamicPayload; }
+  unsigned int getClockRate() { return _clockRate; }
+  unsigned int getChannel() { return _channel; }
+  
+
+};
+
+// the types of the class factories
+typedef AudioCodec* create_t();
+typedef void destroy_t(AudioCodec*);
+
+#endif

src/audio/gsmcodec.h → src/audio/codecs/gsmcodec.cpp

@@ -18,15 +18,11 @@
  *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#ifndef __GSM_H__
-#define __GSM_H__
-
-#include <gsm.h>
 
 #include "audiocodec.h"
-/*extern "C" {
-#include "gsm/gsm.h"
-}*/
+extern "C"{
+#include <gsm.h>
+}
 
 /**
  * GSM audio codec C++ class (over gsm/gsm.h)
@@ -34,16 +30,43 @@
 class Gsm : public AudioCodec {
 public:
   // _payload should be 3
-  Gsm (int payload=3);
-  ~Gsm (void);
+  Gsm (int payload=3): AudioCodec(payload, "GSM"){
+    _clockRate = 8000;
+    _channel = 1;
+    
+    if (!(_decode_gsmhandle = gsm_create() ))
+    printf("ERROR: decode_gsm_create\n");
+    if (!(_encode_gsmhandle = gsm_create() ))
+    printf("AudioCodec: ERROR: encode_gsm_create\n");
+  }
+  
+  virtual ~Gsm (void){
+    gsm_destroy(_decode_gsmhandle);
+    gsm_destroy(_encode_gsmhandle);
+  }
 
-  int	codecDecode	(short *, unsigned char *, unsigned int);
-  int	codecEncode	(unsigned char *, short *, unsigned int);
-  void test();
+  virtual int	codecDecode	(short * dst, unsigned char * src, unsigned int size){
+    (void)size;
+    if(gsm_decode(_decode_gsmhandle, (gsm_byte*)src, (gsm_signal*)dst) < 0)
+      printf("ERROR: gsm_decode\n");
+    return 320;
+  }
+  
+  virtual int	codecEncode	(unsigned char * dst, short * src, unsigned int size){
+	(void)size;
+	gsm_encode(_encode_gsmhandle, (gsm_signal*)src, (gsm_byte*) dst);
+    	return 33;
+  }