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Merge commit 'origin/master' into video

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sfl
2008-02-07 16:33:54 -05:00
parent 03c7a8f7ad d7a1c2d678
commit a8283d052e
16 changed files with 281 additions and 302 deletions
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3
doc/Dependencies.txt
View File

@ -44,5 +44,6 @@ Dependencies to use audio codecs
`--------------------`----------`-----------------------------------------------------
Program Version Notes
--------------------------------------------------------------------------------------
libgsm1 1.0.10 Standard package - Necessary to use GSM codec
libgsm1-dev 1.0.10 Standard package - Necessary to use GSM codec
libspeex1-dev 1.1.12 Standard package - Necessary to use SPEEX codec
--------------------------------------------------------------------------------------

2
src/Makefile.am
View File

@ -1,4 +1,4 @@
sflcodecdir = ($libdir)/sflphone/codecs
sflcodecdir = $(libdir)/sflphone/codecs
bin_PROGRAMS = sflphoned

111
src/audio/CodecSpeex.cpp
View File

@ -18,78 +18,99 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "CodecSpeex.h"
#include "audiocodec.h"
#include <speex/speex.h>
CodecSpeex::CodecSpeex(int payload)
: AudioCodec(payload, "speex")
{
_description = "Speex";
_clockRate = 8000;
_channel = 1;
initSpeex();
}
class CodecSpeex : public AudioCodec{
public:
CodecSpeex(int payload=0)
: AudioCodec(payload, "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;
}
int getFrameSize(){ return _speex_frame_size; }
speex_bits_init(&_speex_dec_bits);
_speex_dec_state = speex_decoder_init(_speexModePtr);
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_enc_bits);
_speex_enc_state = speex_encoder_init(_speexModePtr);
speex_bits_init(&_speex_dec_bits);
_speex_dec_state = speex_decoder_init(_speexModePtr);
speex_decoder_ctl(_speex_dec_state, SPEEX_GET_FRAME_SIZE, &_speex_frame_size);
}
speex_bits_init(&_speex_enc_bits);
_speex_enc_state = speex_encoder_init(_speexModePtr);
CodecSpeex::~CodecSpeex()
{
terminateSpeex();
}
speex_decoder_ctl(_speex_dec_state, SPEEX_GET_FRAME_SIZE, &_speex_frame_size);
}
void
CodecSpeex::terminateSpeex() {
speex_bits_destroy(&_speex_dec_bits);
speex_decoder_destroy(_speex_dec_state);
_speex_dec_state = 0;
~CodecSpeex()
{
terminateSpeex();
}
speex_bits_destroy(&_speex_enc_bits);
speex_encoder_destroy(_speex_enc_state);
_speex_enc_state = 0;
}
void terminateSpeex() {
speex_bits_destroy(&_speex_dec_bits);
speex_decoder_destroy(_speex_dec_state);
_speex_dec_state = 0;
int
CodecSpeex::codecDecode (short *dst, unsigned char *src, unsigned int size)
{
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

22
src/audio/Makefile.am
View File

@ -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)

168
src/audio/alaw.cpp
View File

@ -23,100 +23,98 @@
class Alaw : public AudioCodec {
public:
// 8 PCMA A 8000 1 [RFC3551]
Alaw(int payload=0)
: AudioCodec(payload, "PCMA")
{
//_description = "G711a";
_clockRate = 8000;
_channel = 1;
}
// 8 PCMA A 8000 1 [RFC3551]
Alaw(int payload=0)
: AudioCodec(payload, "PCMA")
{
_clockRate = 8000;
_channel = 1;
}
virtual int codecDecode (short *dst, unsigned char *src, unsigned int size)
{
int16* end = dst+size;
while(dst<end)
*dst++ = ALawDecode(*src++);
return size<<1;
}
virtual int codecDecode (short *dst, unsigned char *src, unsigned int size)
{
int16* end = dst+size;
while(dst<end)
*dst++ = ALawDecode(*src++);
return size<<1;
}
int ALawDecode(uint8 alaw)
{
alaw ^= 0x55; // A-law has alternate bits inverted for transmission
uint sign = alaw&0x80;
int linear = alaw&0x1f;
linear <<= 4;
linear += 8; // Add a 'half' bit (0x08) to place PCM value in middle of range
alaw &= 0x7f;
if(alaw>=0x20)
{
linear |= 0x100; // Put in MSB
uint shift = (alaw>>4)-1;
linear <<= shift;
}
if(!sign)
return -linear;
else
return linear;
}
virtual int codecEncode (unsigned char *dst, short *src, unsigned int size)
{
size >>= 1;
uint8* end = dst+size;
while(dst<end)
*dst++ = ALawEncode(*src++);
return size;
}
virtual int codecEncode (unsigned char *dst, short *src, unsigned int size)
{
size >>= 1;
uint8* end = dst+size;
while(dst<end)
*dst++ = ALawEncode(*src++);
return size;
}
uint8 ALawEncode (int16 pcm16)
{
int p = pcm16;
uint u; // u-law value we are forming
if(p<0)
{
p = ~p;
u = 0x80^0x10^0xff; // Sign bit = 1 (^0x10 because this will get inverted later) ^0xff ^0xff to invert final u-Law code
}
else
int ALawDecode(uint8 alaw)
{
alaw ^= 0x55; // A-law has alternate bits inverted for transmission
uint sign = alaw&0x80;
int linear = alaw&0x1f;
linear <<= 4;
linear += 8; // Add a 'half' bit (0x08) to place PCM value in middle of range
alaw &= 0x7f;
if(alaw>=0x20)
{
linear |= 0x100; // Put in MSB
uint shift = (alaw>>4)-1;
linear <<= shift;
}
if(!sign)
return -linear;
else
return linear;
}
uint8 ALawEncode (int16 pcm16)
{
int p = pcm16;
uint a; // u-law value we are forming
if(p<0)
{
p = ~p;
a = 0x00; // sign = 0
}
else
{
//+ve value
a = 0x80; //sign = 1
}
//calculate segment and interval numbers
p >>= 4;
if(p>=0x20)
{
//+ve value
u = 0x00^0x10^0xff;
}
p += 0x84; // Add uLaw bias
if(p>0x7f00)
p = 0x7f00; // Clip to 15 bits
p >>= 3; // Shift down to 13bit
if(p>=0x100)
{
p >>= 4;
u ^= 0x40;
if(p>=0x100)
{
p >>= 4;
a += 0x40;
}
if(p>=0x40)
{
p >>= 2;
a += 0x20;
}
if(p>=0x20)
{
p >>= 1;
a += 0x10;
}
}
if(p>=0x40)
{
p >>= 2;
u ^= 0x20;
}
if(p>=0x20)
{
p >>= 1;
u ^= 0x10;
}
u ^= p; // u now equal to encoded u-law value (with all bits inverted)
// a&0x70 now holds segment value and 'p' the interval number
a += p; // a now equal to encoded A-law value
return a^0x55; // A-law has alternate bits inverted for transmission
}
return u;
}
};
// the class factories

40
src/audio/alaw.h
View File

@ -1,40 +0,0 @@
/*
* Copyright (C) 2004-2005-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@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 __ALAW_H__
#define __ALAW_H__
#include "audiocodec.h"
/**
* Alaw audio codec (payload is 8)
*/
class Alaw : public AudioCodec {
public:
// payload should be 8
Alaw (int payload=8);
~Alaw (void);
int codecDecode (short *, unsigned char *, unsigned int);
int codecEncode (unsigned char *, short *, unsigned int);
void test () ;
};
#endif // __ULAW_H__

29
src/audio/audiortp.cpp
View File

@ -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) {

3
src/audio/audiortp.h
View File

@ -74,7 +74,8 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
float32* _floatBufferUp;
/** Debugging output file */
//std::ofstream _fstream;
std::ofstream _fstream;
FILE* fd;
/** libsamplerate converter for incoming voice */
SRC_STATE* _src_state_spkr;

24
src/audio/codecDescriptor.cpp
View File

@ -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&
@ -70,19 +68,15 @@ CodecDescriptor::getCodecName(CodecType payload)
}
bool
CodecDescriptor::isSupported(CodecType payload)
CodecDescriptor::isActive(CodecType payload)
{
CodecMap::iterator iter = _codecMap.begin();
while(iter!=_codecMap.end()) {
if (iter->first == payload) {
// codec is already in the map --> nothing to do
_debug("Codec with payload %i already in the map\n", payload);
//break;
return true;
}
iter++;
int i;
for(i=0 ; i < _codecOrder.size() ; i++)
{
if(_codecOrder[i] == payload)
return true;
}
return false;
return false;
}
void
@ -121,7 +115,7 @@ CodecDescriptor::getBitRate(CodecType payload)
return 15.2;
}
return -1;
return 0.0;
}
double
@ -137,7 +131,7 @@ CodecDescriptor::getBandwidthPerCall(CodecType payload)
case PAYLOAD_CODEC_ILBC_20:
return 30.8;
}
return -1;
return 0.0;
}

6
src/audio/codecDescriptor.h
View File

@ -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"
@ -89,7 +91,7 @@ public:
* @return true if the codec specified is supported
* false otherwise
*/
bool isSupported(CodecType payload);
bool isActive(CodecType payload);
/**
* Remove the codec with payload payload from the list

23
src/audio/ulaw.cpp
View File

@ -1,3 +1,25 @@
/**
* Copyright (C) 2004-2005 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@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 "common.h"
#include "audiocodec.h"
@ -11,7 +33,6 @@ public:
_channel = 1;
}
virtual int codecDecode (short *dst, unsigned char *src, unsigned int size) {
int16* end = dst+size;
while(dst<end)

40
src/audio/ulaw.h
View File

@ -1,40 +0,0 @@
/*
* Copyright (C) 2004-2005-2006 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@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 __ULAW_H__
#define __ULAW_H__
#include "audiocodec.h"
/**
* Ulaw audio codec (payload = 0)
*/
class Ulaw : public AudioCodec {
public:
// payload should be 0
Ulaw(int payload=0);
~Ulaw(void);
int codecDecode(short *, unsigned char *, unsigned int);
int codecEncode(unsigned char *, short *, unsigned int);
void test();
};
#endif // __ULAW_H__

46
src/iaxcall.cpp
View File

@ -35,16 +35,6 @@ IAXCall::setFormat(int format)
{
_format = format;
switch(format) {
/*case AST_FORMAT_ULAW:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_ULAW)); break;
case AST_FORMAT_GSM:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_GSM)); break;
case AST_FORMAT_ALAW:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_ALAW)); break;
case AST_FORMAT_ILBC:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_ILBC_20)); break;
case AST_FORMAT_SPEEX:
setAudioCodec(_codecMap.getCodec(PAYLOAD_CODEC_SPEEX)); break;*/
case AST_FORMAT_ULAW:
setAudioCodec(PAYLOAD_CODEC_ULAW); break;
case AST_FORMAT_GSM:
@ -54,7 +44,7 @@ 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;
@ -65,12 +55,12 @@ IAXCall::setFormat(int format)
int
IAXCall::getSupportedFormat()
{
CodecMap map = getCodecMap().getCodecMap();
int format = 0;
CodecOrder map = getCodecMap().getActiveCodecs();
int format = 0;
int iter;
CodecMap::iterator iter = map.begin();
while(iter != map.end()) {
switch(iter->first) {
for(iter=0 ; iter < map.size() ; iter++){
switch(map[iter]) {
case PAYLOAD_CODEC_ULAW:
format |= AST_FORMAT_ULAW; break;
case PAYLOAD_CODEC_GSM:
@ -79,12 +69,12 @@ IAXCall::getSupportedFormat()
format |= AST_FORMAT_ALAW; break;
case PAYLOAD_CODEC_ILBC_20:
format |= AST_FORMAT_ILBC; break;
case PAYLOAD_CODEC_SPEEX:
format |= AST_FORMAT_SPEEX; break;
case PAYLOAD_CODEC_SPEEX_8000:
format |= AST_FORMAT_SPEEX;
break;
default:
break;
}
iter++;
}
return format;
@ -93,30 +83,30 @@ IAXCall::getSupportedFormat()
int
IAXCall::getFirstMatchingFormat(int needles)
{
CodecMap map = getCodecMap().getCodecMap();
int format = 0;
CodecOrder map = getCodecMap().getActiveCodecs();
int format = 0;
int iter;
CodecMap::iterator iter = map.begin();
while(iter != map.end()) {
switch(iter->first) {
for(iter=0 ; iter < map.size() ; iter++) {
switch(map[iter]) {
case PAYLOAD_CODEC_ULAW:
format = AST_FORMAT_ULAW; break;
case PAYLOAD_CODEC_GSM:
format = AST_FORMAT_GSM; break;
case PAYLOAD_CODEC_ALAW:
format = AST_FORMAT_ALAW; break;
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;
}
// Return the first that matches
if (format & needles)
return format;
iter++;
}
return 0;
}

15
src/iaxvoiplink.cpp
View File

@ -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';
@ -512,6 +515,8 @@ Call*
IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
{
IAXCall* call = new IAXCall(id, Call::Outgoing);
call->setCodecMap(Manager::instance().getCodecDescriptorMap());
if (call) {
call->setPeerNumber(toUrl);
@ -954,6 +959,7 @@ IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
call->setCodecMap(Manager::instance().getCodecDescriptorMap());
call->setConnectionState(Call::Progressing);
if (event->ies.calling_number)
call->setPeerNumber(std::string(event->ies.calling_number));
if (event->ies.calling_name)
@ -1005,3 +1011,12 @@ IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
}
int
IAXVoIPLink::iaxCodecMapToFormat(IAXCall* call)
{
CodecOrder map = call->getCodecMap().getActiveCodecs();
printf("taytciatcia = %i\n", map.size());
return 0;
}

11
src/sipcall.cpp
View File

@ -353,17 +353,18 @@ SIPCall::sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg)
remote_med_m_payloads = &(remote_med->m_payloads);
#endif
while (!osip_list_eol(remote_med_m_payloads, iPayload) && iPayload < 2) {
//while (!osip_list_eol(remote_med_m_payloads, iPayload) && iPayload < 2) {
while (!osip_list_eol(remote_med_m_payloads, iPayload)) {
tmp = (char *)osip_list_get(remote_med_m_payloads, iPayload);
if (tmp!=NULL) {
int payload = atoi(tmp);
_debug("remote payload = %s\n", tmp);
CodecType audiocodec = (CodecType)payload;
//if (audiocodec != NULL && audiocodec->isActive()) {
if (audiocodec != (CodecType)-1 && _codecMap.isSupported(audiocodec)) {
if (audiocodec != (CodecType)-1 && _codecMap.isActive(audiocodec)) {
_debug("PAYLOAD = %i", payload);
listCodec << payload << " ";
//listRtpMap << "a=rtpmap:" << payload << " " << audiocodec->getCodecName() << "/" << audiocodec->getClockRate();
listRtpMap << "a=rtpmap:" << payload << " " << _codecMap.getCodecName(audiocodec) << "/" << 8000;
listRtpMap << "a=rtpmap:" << payload << " " << _codecMap.getCodecName(audiocodec) << "/" << _codecMap.getSampleRate(audiocodec);
// TODO: manage a way to get the channel infos
/*if ( audiocodec->getChannel() != 1) {
listRtpMap << "/" << audiocodec->getChannel();
@ -586,7 +587,7 @@ SIPCall::setAudioCodecFromSDP(sdp_media_t* remote_med, int tid)
if (tmp != NULL ) {
int payload = atoi(tmp);
// stop if we find a correct codec
if (_codecMap.isSupported((CodecType)payload)){
if (_codecMap.isActive((CodecType)payload)){
break;
}
}

40
src/sipvoiplink.cpp
View File

@ -943,32 +943,32 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
std::ostringstream media_audio;
std::ostringstream rtpmap_attr;
int payload;
CodecType payload;
int nbChannel;
int iter;
// Set rtpmap according to the supported codec order
CodecMap map = call->getCodecMap().getCodecMap();
//CodecMap map = Manager::instance().getCodecDescriptorMap();
CodecMap::iterator iter = map.begin();
//CodecMap map = call->getCodecMap().getCodecMap();
CodecOrder map = call->getCodecMap().getActiveCodecs();
for(iter=0 ; iter < map.size() ; iter++){
if(map[iter] != -1){
payload = map[iter];
// add each payload in the list of payload
media_audio << payload << " ";
while(iter != map.end()) {
//if (iter->second!=0 && iter->second->isActive()) {
if(iter->first != -1){
payload = iter->first;
// add each payload in the list of payload
media_audio << payload << " ";
rtpmap_attr << "a=rtpmap:" << payload << " " <<
call->getCodecMap().getCodecName(payload) << "/" << call->getCodecMap().getSampleRate(payload);
rtpmap_attr << "a=rtpmap:" << payload << " " <<
iter->second.data() << "/" << 8000; //iter->second->getClockRate();
/*nbChannel = iter->second->getChannel();
if (nbChannel!=1) {
rtpmap_attr << "/" << nbChannel;
}*/
rtpmap_attr << "\r\n";
}
//TODO add channel infos
/*nbChannel = iter->second->getChannel();
if (nbChannel!=1) {
rtpmap_attr << "/" << nbChannel;
}*/
rtpmap_attr << "\r\n";
}
// go to next codec
iter++;
//*iter++;
}
// http://www.antisip.com/documentation/eXosip2/group__howto1__initialize.html