diff --git a/src/audio/CodecSpeex.cpp b/src/audio/CodecSpeex.cpp
index 48cbad39136e721ece409ad96ecb1fa20a2e6755..8200b3ee246bd53f192c83edfbcd722286534fa2 100644
--- a/src/audio/CodecSpeex.cpp
+++ b/src/audio/CodecSpeex.cpp
@@ -18,78 +18,99 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#include "CodecSpeex.h"
-
-CodecSpeex::CodecSpeex(int payload)
- : AudioCodec(payload, "speex")
-{
- _description = "Speex";
-
- _clockRate = 8000;
- _channel = 1;
- initSpeex();
-}
-
-void
-CodecSpeex::initSpeex() {
- if (_clockRate < 16000 ) {
- _speexModePtr = &speex_nb_mode;
- } else if (_clockRate < 32000) {
- _speexModePtr = &speex_wb_mode;
- } else {
- _speexModePtr = &speex_uwb_mode;
- }
-
- speex_bits_init(&_speex_dec_bits);
- _speex_dec_state = speex_decoder_init(_speexModePtr);
-
- speex_bits_init(&_speex_enc_bits);
- _speex_enc_state = speex_encoder_init(_speexModePtr);
-
- speex_decoder_ctl(_speex_dec_state, SPEEX_GET_FRAME_SIZE, &_speex_frame_size);
-}
-
-CodecSpeex::~CodecSpeex()
-{
- terminateSpeex();
-}
-
-void
-CodecSpeex::terminateSpeex() {
- speex_bits_destroy(&_speex_dec_bits);
- speex_decoder_destroy(_speex_dec_state);
- _speex_dec_state = 0;
-
- speex_bits_destroy(&_speex_enc_bits);
- speex_encoder_destroy(_speex_enc_state);
- _speex_enc_state = 0;
-}
-
-int
-CodecSpeex::codecDecode (short *dst, unsigned char *src, unsigned int size)
-{
+#include "audiocodec.h"
+#include <speex/speex.h>
+
+
+class CodecSpeex : public AudioCodec{
+public:
+ CodecSpeex(int payload=0)
+ : AudioCodec(payload, "speex")
+ {
+ _clockRate = 8000;
+ _channel = 1;
+ initSpeex();
+ }
+
+ int getFrameSize(){ return _speex_frame_size; }
+
+ void initSpeex() {
+ if (_clockRate < 16000 ) {
+ _speexModePtr = &speex_nb_mode;
+ } else if (_clockRate < 32000) {
+ _speexModePtr = &speex_wb_mode;
+ } else {
+ _speexModePtr = &speex_uwb_mode;
+ }
+
+ speex_bits_init(&_speex_dec_bits);
+ _speex_dec_state = speex_decoder_init(_speexModePtr);
+
+ speex_bits_init(&_speex_enc_bits);
+ _speex_enc_state = speex_encoder_init(_speexModePtr);
+
+ speex_decoder_ctl(_speex_dec_state, SPEEX_GET_FRAME_SIZE, &_speex_frame_size);
+ }
+
+ ~CodecSpeex()
+ {
+ terminateSpeex();
+ }
+
+ void terminateSpeex() {
+ speex_bits_destroy(&_speex_dec_bits);
+ speex_decoder_destroy(_speex_dec_state);
+ _speex_dec_state = 0;
+
+ speex_bits_destroy(&_speex_enc_bits);
+ speex_encoder_destroy(_speex_enc_state);
+ _speex_enc_state = 0;
+ }
+
+ virtual int codecDecode (short *dst, unsigned char *src, unsigned int size)
+ {
// void *enh; speex_decoder_ctl(dec_state, SPEEX_SET_ENH, &enh);
// decoding
speex_bits_read_from(&_speex_dec_bits, (char*)src, size);
- int return_status = speex_decode_int(_speex_dec_state, &_speex_dec_bits, dst);
+ //int return_status = speex_decode_int(_speex_dec_state, &_speex_dec_bits, dst);
// 0 = no error
// -1 = end of stream
// -2 = other
+ speex_decode_int(_speex_dec_state, &_speex_dec_bits, dst);
return _speex_frame_size;
-}
+ }
-int
-CodecSpeex::codecEncode (unsigned char *dst, short *src, unsigned int size)
-{
+ virtual int codecEncode (unsigned char *dst, short *src, unsigned int size)
+ {
speex_bits_reset(&_speex_enc_bits);
speex_encoder_ctl(_speex_enc_state,SPEEX_SET_SAMPLING_RATE,&_clockRate);
speex_encode_int(_speex_enc_state, src, &_speex_enc_bits);
int nbBytes = speex_bits_write(&_speex_enc_bits, (char*)dst, size);
return nbBytes;
+ }
+
+private:
+ const SpeexMode* _speexModePtr;
+ SpeexBits _speex_dec_bits;
+ SpeexBits _speex_enc_bits;
+ void *_speex_dec_state;
+ void *_speex_enc_state;
+ int _speex_frame_size;
+
+
+
+};
+
+//the class factories
+extern "C" AudioCodec* create(){
+ return new CodecSpeex(110);
}
+extern "C" void destroy(AudioCodec* a){
+ delete a;
+}
/*
* Speex example
diff --git a/src/audio/Makefile.am b/src/audio/Makefile.am
index 5621175377d55490fe5bb48a6c6a13ad3fcc4e25..af2ee76b5841f10d1e4791fecf0f41bd4dc691d3 100644
--- a/src/audio/Makefile.am
+++ b/src/audio/Makefile.am
@@ -1,19 +1,23 @@
sflcodecdir = $(libdir)/sflphone/codecs/
noinst_LTLIBRARIES = libaudio.la
-noinst_PROGRAMS = libcodec_ulaw.so libcodec_alaw.so libcodec_gsm.so
+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
+libcodec_ulaw_so_LDFLAGS = -shared -lc
libcodec_alaw_so_SOURCES = alaw.cpp
libcodec_alaw_so_CFLAGS = -fPIC -g -Wall
-libcodec_alaw_so_LDFLAGS = -shared
+libcodec_alaw_so_LDFLAGS = -shared -lc
libcodec_gsm_so_SOURCES = gsmcodec.cpp
libcodec_gsm_so_CFLAGS = -fPIC -g -Wall
-libcodec_gsm_so_LDFLAGS = -shared -lgsm
+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
@@ -25,6 +29,8 @@ SPEEX_FLAG=
SPEEX_LIB=
endif
+SUBDIRS = ilbc
+
libaudio_la_SOURCES = audiofile.cpp tonelist.cpp \
audiortp.cpp dtmf.cpp tone.cpp audiolayer.cpp audiodevice.cpp dtmfgenerator.cpp \
tonegenerator.cpp codecDescriptor.cpp \
@@ -40,8 +46,8 @@ noinst_HEADERS = audioloop.h common.h ringbuffer.h audiofile.h
codecDescriptor.h dtmf.h tone.h \
CodecSpeex.h
-install-exec-local: install-libcodec_ulaw_so install-libcodec_alaw_so install-libcodec_gsm_so
-uninstall-local: uninstall-libcodec_ulaw_so uninstall-libcodec_alaw_so uninstall-libcodec_gsm_so
+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)
@@ -50,6 +56,8 @@ 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)
uninstall-libcodec_ulaw_so:
rm -f $(sflcodecdir)/libcodec_ulaw.so
@@ -57,4 +65,6 @@ 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)
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 13c787961de894b357fe694c2a480ee62e8f5c87..4bbc93508d04380e97f5d14e928da885020cce52 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -1,4 +1,5 @@
/*
+ *
* Copyright (C) 2004-2007 Savoir-Faire Linux inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
@@ -26,8 +27,6 @@
#include <assert.h>
#include <string>
#include <cstring>
-//#include <fstream> // fstream + iostream pour fstream debugging..
-//#include <iostream> // removeable...
#include <math.h>
#include <dlfcn.h>
#include <iostream>
@@ -98,17 +97,17 @@ AudioRtp::closeRtpSession () {
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
- // : _fstream("/tmp/audio.dat", std::ofstream::binary)
+ : //_fstream("/tmp/audio.dat", 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();
-
// TODO: Change bind address according to user settings.
// TODO: this should be the local ip not the external (router) IP
std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
@@ -142,7 +141,7 @@ AudioRtpRTX::~AudioRtpRTX () {
}
//_debug("terminate audiortprtx ended...\n");
_ca = 0;
-
+ //fd = fopen("snd_data", "wa");
if (!_sym) {
delete _sessionRecv; _sessionRecv = NULL;
delete _sessionSend; _sessionSend = NULL;
@@ -185,7 +184,6 @@ AudioRtpRTX::initAudioRtpSession (void)
try {
if (_ca == 0) { return; }
- _debug("AUDIOCODEC=%i\n", _ca->getAudioCodec());
AudioCodec* audiocodec = loadCodec(_ca->getAudioCodec());
_codecSampleRate = audiocodec->getClockRate();
@@ -270,6 +268,9 @@ AudioRtpRTX::loadCodec(int payload)
case 97:
handle_codec = dlopen(CODECS_DIR "/libcodec_ilbc.so", RTLD_LAZY);
break;
+ case 110:
+ handle_codec = dlopen(CODECS_DIR "/libcodec_speex.so", RTLD_LAZY);
+ break;
}
if(!handle_codec){
@@ -302,7 +303,6 @@ 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
@@ -327,11 +327,12 @@ try {
// 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);
nbSample = reSampleData(audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
@@ -343,11 +344,12 @@ try {
memset(toSIP + nbSample, 0, (nbSamplesMax-nbSample)*sizeof(int16));
nbSample = nbSamplesMax;
}
- //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toSIP[0], toSIP[1], toSIP[2]);
-
+ // debug - dump sound in a file
+//_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));
+ //printf("jusqu'ici tout vas bien\n");
// encode divise by two
// Send encoded audio sample over the network
@@ -393,8 +395,9 @@ 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
+ unsigned int size = adu->getSize(); // size in char
+ //_fstream.write((char*) data, size);
audiocodec = loadCodec(payload);
// Decode data with relevant codec
_codecSampleRate = audiocodec->getClockRate();
@@ -405,7 +408,8 @@ try {
_debug("The packet size has been cropped\n");
size=max;
}
-
+
+ //printf("size = %i\n", size);
if (audiocodec != NULL) {
int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
@@ -565,6 +569,7 @@ try {
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
}
+ //_fstream.close();
//_debug("stop stream for audiortp loop\n");
audiolayer->stopStream();
} catch(std::exception &e) {
diff --git a/src/audio/codecDescriptor.cpp b/src/audio/codecDescriptor.cpp
index fdb4dd39d2d7e78a22c9943ea3e69a6ecd9938cd..a00720cda87dead0037694b4c5e42c8c521197f7 100644
--- a/src/audio/codecDescriptor.cpp
+++ b/src/audio/codecDescriptor.cpp
@@ -52,11 +52,9 @@ void
CodecDescriptor::setDefaultOrder()
{
_codecOrder.clear();
- //_codecOrder.push_back(PAYLOAD_CODEC_ILBC_20);
_codecOrder.push_back(PAYLOAD_CODEC_ULAW);
_codecOrder.push_back(PAYLOAD_CODEC_ALAW);
_codecOrder.push_back(PAYLOAD_CODEC_GSM);
- //_codecOrder.push_back(PAYLOAD_CODEC_SPEEX_8000);
}
std::string&
@@ -121,7 +119,7 @@ CodecDescriptor::getBitRate(CodecType payload)
return 15.2;
}
- return -1;
+ return 0.0;
}
double
@@ -137,7 +135,7 @@ CodecDescriptor::getBandwidthPerCall(CodecType payload)
case PAYLOAD_CODEC_ILBC_20:
return 30.8;
}
- return -1;
+ return 0.0;
}
diff --git a/src/audio/codecDescriptor.h b/src/audio/codecDescriptor.h
index 4cb9be9a5c03c88ef58b6c653ce77958b7aa7053..77da6e81dc02b74eaa253bd82b4a020ff05ee42d 100644
--- a/src/audio/codecDescriptor.h
+++ b/src/audio/codecDescriptor.h
@@ -45,7 +45,9 @@ typedef enum {
// 97 speex/8000
// http://support.xten.com/viewtopic.php?p=8684&sid=3367a83d01fdcad16c7459a79859b08e
// 100 speex/16000
- PAYLOAD_CODEC_SPEEX = 110
+ PAYLOAD_CODEC_SPEEX_8000 = 110,
+ PAYLOAD_CODEC_SPEEX_16000 = 111,
+ PAYLOAD_CODEC_SPEEX_32000 = 112
} CodecType;
#include "audiocodec.h"
diff --git a/src/iaxcall.cpp b/src/iaxcall.cpp
index e522e028a1466e9c785e3d04f8d51a38a238e1ef..9ffc3808568ca1aef9ca6856f5665fa73a899b91 100644
--- a/src/iaxcall.cpp
+++ b/src/iaxcall.cpp
@@ -44,7 +44,7 @@ IAXCall::setFormat(int format)
case AST_FORMAT_ILBC:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_ILBC_20)); break;
case AST_FORMAT_SPEEX:
- setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_SPEEX)); break;*/
+ setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_SPEEX_8000)); break;*/
case AST_FORMAT_ULAW:
setAudioCodec(PAYLOAD_CODEC_ULAW); break;
case AST_FORMAT_GSM:
@@ -54,7 +54,7 @@ IAXCall::setFormat(int format)
case AST_FORMAT_ILBC:
setAudioCodec(PAYLOAD_CODEC_ILBC_20); break;
case AST_FORMAT_SPEEX:
- setAudioCodec(PAYLOAD_CODEC_SPEEX); break;
+ setAudioCodec(PAYLOAD_CODEC_SPEEX_8000); break;
default:
setAudioCodec((CodecType) -1);
break;
@@ -79,7 +79,7 @@ IAXCall::getSupportedFormat()
format |= AST_FORMAT_ALAW; break;
case PAYLOAD_CODEC_ILBC_20:
format |= AST_FORMAT_ILBC; break;
- case PAYLOAD_CODEC_SPEEX:
+ case PAYLOAD_CODEC_SPEEX_8000:
format |= AST_FORMAT_SPEEX; break;
default:
break;
@@ -107,7 +107,7 @@ IAXCall::getFirstMatchingFormat(int needles)
format = AST_FORMAT_ALAW; break;
case PAYLOAD_CODEC_ILBC_20:
format = AST_FORMAT_ILBC; break;
- case PAYLOAD_CODEC_SPEEX:
+ case PAYLOAD_CODEC_SPEEX_8000:
format = AST_FORMAT_SPEEX; break;
default:
break;
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 4e4738deb1d51cee566c0b46c6161c1e03d86c2b..43970a6a830c772714f9d0e8860fec8048d5a9f2 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -247,6 +247,9 @@ IAXVoIPLink::loadCodec(int payload)
case 97:
handle_codec = dlopen(CODECS_DIR "/libcodec_ilbc.so", RTLD_LAZY);
break;
+ case 110:
+ handle_codec = dlopen(CODECS_DIR "/libcodec_speex.so", RTLD_LAZY);
+ break;
}
if(!handle_codec){
cerr<<"cannot load library: "<< dlerror() <<'\n';