diff --git a/src/audio/Makefile.am b/src/audio/Makefile.am
index 3dfd56bdffac16cc42b6a46e508ad30fbc11deef..81564003259883cb3a623bceed515a11a3d26b8c 100644
--- a/src/audio/Makefile.am
+++ b/src/audio/Makefile.am
@@ -29,9 +29,9 @@ SPEEX_FLAG=
SPEEX_LIB=
endif
-#SUBDIRS = ilbc
+#SUBDIRS = gsm
-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 $(SPEEX_SOURCES_CPP)
@@ -40,7 +40,7 @@ AM_CXXFLAGS = -I$(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 \
+noinst_HEADERS = audioloop.h common.h ringbuffer.h audiofile.h \
tonelist.h audiortp.h audiocodec.h audiolayer.h audiodevice.h \
dtmfgenerator.h tonegenerator.h \
codecDescriptor.h dtmf.h tone.h \
@@ -63,8 +63,8 @@ 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_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/alaw.cpp b/src/audio/alaw.cpp
index 71490076edca91712cc98a97f59215aac8061d0a..331e5ddd1c7d9b51a6dfe0335ca01990d714e606 100644
--- a/src/audio/alaw.cpp
+++ b/src/audio/alaw.cpp
@@ -31,7 +31,7 @@ public:
_channel = 1;
}
-
+ virtual ~Alaw(){}
virtual int codecDecode (short *dst, unsigned char *src, unsigned int size)
{
int16* end = dst+size;
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index d92255c4b63abd7c9f101b617aa2b2c0d15a4d99..0f0f2e21ec88a99081009126d11b4db7d094ee5d 100644
--- a/src/audio/audiortp.cpp
+++ b/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;
@@ -321,7 +319,6 @@ try {
// 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();
@@ -332,7 +329,6 @@ try {
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);
@@ -348,6 +344,7 @@ 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");
@@ -401,7 +398,7 @@ try {
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);
@@ -413,6 +410,7 @@ try {
if (audiocodec != NULL) {
int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
+ _fstream.write((char*) _receiveDataDecoded, 160);
//buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
int nbInt16 = expandedSize / sizeof(int16);
//nbInt16 represents the number of samples we just decoded
@@ -569,7 +567,7 @@ try {
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
}
- //_fstream.close();
+ _fstream.close();
//_debug("stop stream for audiortp loop\n");
audiolayer->stopStream();
} catch(std::exception &e) {
diff --git a/src/audio/gsmcodec.cpp b/src/audio/gsmcodec.cpp
index a340882103b1a281b1fb4091d2b0989ac8f596af..30b9ed3bbb5ff9866edddd0682d84c93aa5a2b42 100644
--- a/src/audio/gsmcodec.cpp
+++ b/src/audio/gsmcodec.cpp
@@ -19,54 +19,94 @@
*/
#include <iostream>
+extern "C" {
#include <gsm/gsm.h>
+}
#include "audiocodec.h"
#include "../global.h"
+#define GSM_PACKED_FRAME_SIZE_IN_BYTES 33
+#define GSM_UNPACKED_FRAME_SIZE_IN_BYTES 320
+#define GSM_UNPACKED_FRAME_SIZE_IN_SHORTS 160
+/*
+void * GSMCodecHandle = 0;
+void* (*gsm_session_create)() = 0;
+void (*gsm_session_destroy)(void *) = 0;
+void (*gsm_session_encode)(void*, unsigned char*, short*) = 0 ;
+int (*gsm_session_decode)(void*, short* , unsigned char*) = 0 ;
+*/
class Gsm : public AudioCodec {
public:
// 3 GSM A 8000 1 [RFC3551]
Gsm(int payload = 0)
: AudioCodec(payload, "GSM")
{
- //_description = "GSM";
_clockRate = 8000;
_channel = 1;
-
+
+ //initGSMStruct();
+
if (!(_decode_gsmhandle = gsm_create() ))
_debug("ERROR: decode_gsm_create\n");
if (!(_encode_gsmhandle = gsm_create() ))
_debug("AudioCodec: ERROR: encode_gsm_create\n");
+
+ //_encode_state = gsm_session_create();
+ //_decode_state = gsm_session_create();
+
}
~Gsm (void)
{
+ //gsm_session_destroy(_decode_state);
+ //gsm_session_destroy(_encode_state);
gsm_destroy(_decode_gsmhandle);
gsm_destroy(_encode_gsmhandle);
}
+/*
+bool initGSMStruct()
+{
+ if(GSMCodecHandle) return true;
+
+ GSMCodecHandle = dlopen("libgsm.so", RTLD_NOW | RTLD_GLOBAL);
+ if(!GSMCodecHandle) return false;
+ gsm_session_create = (void * (*)()) dlsym(GSMCodecHandle, "gsm_create");
+ gsm_session_destroy = (void (*) (void *)) dlsym(GSMCodecHandle, "gsm_destroy");
+ gsm_session_encode = ( void (*) (void*, unsigned char*, short*)) dlsym(GSMCodecHandle, "gsm_encode");
+ gsm_session_decode = (int (*) (void*, short*, unsigned char*)) dlsym(GSMCodecHandle, "gsm_decode");
+
+ if(!(gsm_session_create && gsm_session_destroy && gsm_session_encode && gsm_session_decode)){
+ dlclose(GSMCodecHandle);
+ GSMCodecHandle = 0 ;
+ return false;}
+ return true;
+}*/
+
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) {
- _debug("ERROR: gsm_decode\n");
- }
- return 320;
+ if(gsm_decode(_decode_gsmhandle, (gsm_byte *)src, (gsm_signal*)dst) < 0)
+ return 0;
+
+ return GSM_UNPACKED_FRAME_SIZE_IN_BYTES;
}
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;
+ if(size < GSM_UNPACKED_FRAME_SIZE_IN_BYTES) return 0;
+ //if (gsm_session_encode( gsm_signal*)src, (gsm_byte*)dst);
+ gsm_encode( _encode_gsmhandle, (gsm_signal*)src, (gsm_byte*) dst);
+ return GSM_PACKED_FRAME_SIZE_IN_BYTES;
}
private:
gsm _decode_gsmhandle;
gsm _encode_gsmhandle;
+ //void * _encode_state;
+ //void * _decode_state;
};
@@ -78,3 +118,4 @@ extern "C" AudioCodec* create() {
extern "C" void destroy(AudioCodec* a) {
delete a;
}
+