/*
* Copyright (C) 2002-2003 Fhg Fokus
*
* This file is part of SEMS, a free SIP media server.
*
* SEMS 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 2 of the License, or
* (at your option) any later version. This program is released under
* the GPL with the additional exemption that compiling, linking,
* and/or using OpenSSL is allowed.
*
* For a license to use the SEMS software under conditions
* other than those described here, or to purchase support for this
* software, please contact iptel.org by e-mail at the following addresses:
* info@iptel.org
*
* SEMS 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "AmRtpAudio.h"
#include <sys/time.h>
#include <assert.h>
#include "AmSession.h"
#include "AmPlayoutBuffer.h"
AmAudioRtpFormat::AmAudioRtpFormat()
: AmAudioFormat(-1),
advertized_rate(-1)
{
}
AmAudioRtpFormat::~AmAudioRtpFormat()
{
}
int AmAudioRtpFormat::setCurrentPayload(Payload pl)
{
if (this->codec_id != pl.codec_id) {
this->codec_id = pl.codec_id;
DBG("fmt.codec_id = %d", this->codec_id);
this->channels = 1;
this->rate = pl.clock_rate;
DBG("fmt.rate = %d", this->rate);
this->advertized_rate = pl.advertised_clock_rate;
DBG("fmt.advertized_rate = %d", this->advertized_rate);
this->frame_size = 20*this->rate/1000;
DBG("fmt.sdp_format_parameters = %s", this->sdp_format_parameters.c_str());
if (this->codec != NULL) {
destroyCodec();
}
}
return 0;
}
void AmAudioRtpFormat::initCodec()
{
amci_codec_fmt_info_t fmt_i[4];
fmt_i[0].id=0;
if( codec && codec->init ) {
if ((h_codec = (*codec->init)(sdp_format_parameters.c_str(), fmt_i)) == -1) {
ERROR("could not initialize codec %i\n",codec->id);
} else {
string s;
int i=0;
while (fmt_i[i].id) {
switch (fmt_i[i].id) {
case AMCI_FMT_FRAME_LENGTH : {
//frame_length=fmt_i[i].value;
} break;
case AMCI_FMT_FRAME_SIZE: {
frame_size=fmt_i[i].value;
} break;
case AMCI_FMT_ENCODED_FRAME_SIZE: {
// frame_encoded_size=fmt_i[i].value;
} break;
default: {
DBG("Unknown codec format descriptor: %d\n", fmt_i[i].id);
} break;
}
i++;
}
}
}
}
AmRtpAudio::AmRtpAudio(AmSession* _s, int _if)
: AmRtpStream(_s,_if), AmAudio(0),
/*last_ts_i(false),*/ use_default_plc(true),
playout_buffer(NULL),
m_playout_type(SIMPLE_PLAYOUT),
last_check(0),last_check_i(false),send_int(false),
last_send_ts_i(false)
{
#ifdef USE_SPANDSP_PLC
plc_state = plc_init(NULL);
#endif // USE_SPANDSP_PLC
}
AmRtpAudio::~AmRtpAudio() {
#ifdef USE_SPANDSP_PLC
plc_release(plc_state);
#endif // USE_SPANDSP_PLC
}
bool AmRtpAudio::checkInterval(unsigned long long ts)
{
if(!last_check_i){
send_int = true;
last_check_i = true;
last_check = ts;
}
else {
if(scaleSystemTS(ts - last_check) >= getFrameSize()){
send_int = true;
last_check = ts;
}
else {
send_int = false;
}
}
return send_int;
}
bool AmRtpAudio::sendIntReached()
{
return send_int;
}
bool AmRtpAudio::sendIntReached(unsigned long long ts)
{
if (!last_send_ts_i) return true;
else return (scaleSystemTS(ts - last_send_ts) >= getFrameSize());
}
unsigned int AmRtpAudio::bytes2samples(unsigned int bytes) const
{
return AmAudio::bytes2samples(bytes);
}
/*
@param wallclock_ts [in] the current ts in the audio buffer
*/
int AmRtpAudio::receive(unsigned long long system_ts)
{
int size;
unsigned int rtp_ts;
int new_payload = -1;
if(!fmt.get() || (!playout_buffer.get())) {
DBG("audio format not initialized\n");
return RTP_ERROR;
}
unsigned int wallclock_ts = scaleSystemTS(system_ts);
while(true) {
size = AmRtpStream::receive((unsigned char*)samples,
(unsigned int)AUDIO_BUFFER_SIZE, rtp_ts,
new_payload);
if(size <= 0) {
switch(size){
case 0: break;
case RTP_DTMF:
case RTP_UNKNOWN_PL:
case RTP_PARSE_ERROR:
continue;
case RTP_TIMEOUT:
//FIXME: postRequest(new SchedRequest(AmMediaProcessor::RemoveSession,s));
// post to the session (FIXME: is session always set? seems to be...)
session->postEvent(new AmRtpTimeoutEvent());
return -1;
case RTP_BUFFER_SIZE:
default:
ERROR("AmRtpStream::receive() returned %i\n",size);
//FIXME: postRequest(new SchedRequest(AmMediaProcessor::ClearSession,s));
// or AmMediaProcessor::instance()->clearSession(session);
return -1;
break;
}
break;
}
if (// don't process if we don't need to
// ignore CN
COMFORT_NOISE_PAYLOAD_TYPE == new_payload ||
// ignore packet if payload not found
setCurrentPayload(new_payload)
){
playout_buffer->clearLastTs();
continue;
}
size = decode(size);
if(size <= 0){
ERROR("decode() returned %i\n",size);
return -1;
}
// This only works because the possible ratio (Rate/TSRate)
// is 2. Rate and TSRate are only different in case of g722.
// For g722, TSRate=8000 and Rate=16000
//
AmAudioRtpFormat* rtp_fmt = (AmAudioRtpFormat*)fmt.get();
unsigned long long adjusted_rtp_ts = rtp_ts;
if(rtp_fmt->getRate() != rtp_fmt->getTSRate()) {
adjusted_rtp_ts =
(unsigned long long)rtp_ts *
(unsigned long long)rtp_fmt->getRate()
/ (unsigned long long)rtp_fmt->getTSRate();
}
playout_buffer->write(wallclock_ts, adjusted_rtp_ts,
(ShortSample*)((unsigned char *)samples),
PCM16_B2S(size), begin_talk);
if(!active) {
DBG("switching to active-mode\t(ts=%u;stream=%p)\n",
rtp_ts,this);
active = true;
}
}
return size;
}
int AmRtpAudio::get(unsigned long long system_ts, unsigned char* buffer,
int output_sample_rate, unsigned int nb_samples)
{
if (!(receiving || getPassiveMode())) return 0; // like nothing received
int ret = receive(system_ts);
if(ret < 0)
return ret; // like nothing received?
if (!active) return 0;
unsigned int user_ts = scaleSystemTS(system_ts);
nb_samples = (unsigned int)((float)nb_samples * (float)getSampleRate()
/ (float)output_sample_rate);
u_int32_t size =
PCM16_S2B(playout_buffer->read(user_ts,
(ShortSample*)((unsigned char*)samples),
nb_samples));
if(output_sample_rate != getSampleRate()) {
size = resampleOutput((unsigned char*)samples, size,
getSampleRate(), output_sample_rate);
}
memcpy(buffer,(unsigned char*)samples,size);
return size;
}
int AmRtpAudio::put(unsigned long long system_ts, unsigned char* buffer,
int input_sample_rate, unsigned int size)
{
last_send_ts_i = true;
last_send_ts = system_ts;
if(!size){
return 0;
}
if (mute) return 0;
memcpy((unsigned char*)samples,buffer,size);
size = resampleInput((unsigned char*)samples, size,
input_sample_rate, getSampleRate());
int s = encode(size);
if(s<=0){
return s;
}
AmAudioRtpFormat* rtp_fmt = (AmAudioRtpFormat*)fmt.get();
// pre-division by 100 is important
// so that the first multiplication
// does not overflow the 64bit int
unsigned long long user_ts =
system_ts * ((unsigned long long)rtp_fmt->getTSRate() / 100)
/ (WALLCLOCK_RATE/100);
return send((unsigned int)user_ts,(unsigned char*)samples,s);
}
void AmRtpAudio::getSdpOffer(unsigned int index, SdpMedia& offer)
{
if (offer.type != MT_AUDIO) return;
AmRtpStream::getSdpOffer(index,offer);
}
void AmRtpAudio::getSdpAnswer(unsigned int index,
const SdpMedia& offer,
SdpMedia& answer)
{
answer.type = MT_AUDIO;
AmRtpStream::getSdpAnswer(index,offer,answer);
}
int AmRtpAudio::init(const AmSdp& local,
const AmSdp& remote, bool force_symmetric_rtp)
{
DBG("AmRtpAudio::init(...)\n");
if(AmRtpStream::init(local,remote,force_symmetric_rtp)){
return -1;
}
AmAudioRtpFormat* fmt_p = new AmAudioRtpFormat();
PayloadMappingTable::iterator pl_it = pl_map.find(payload);
if ((pl_it == pl_map.end()) || (pl_it->second.remote_pt < 0)) {
DBG("no default payload has been set\n");
return -1;
}
fmt_p->setCurrentPayload(payloads[pl_it->second.index]);
fmt.reset(fmt_p);
amci_codec_t* codec = fmt->getCodec();
use_default_plc = ((codec==NULL) || (codec->plc == NULL));
fec.reset(new LowcFE(getSampleRate()));
if (m_playout_type == SIMPLE_PLAYOUT) {
playout_buffer.reset(new AmPlayoutBuffer(this,getSampleRate()));
} else if (m_playout_type == ADAPTIVE_PLAYOUT) {
playout_buffer.reset(new AmAdaptivePlayout(this,getSampleRate()));
} else {
playout_buffer.reset(new AmJbPlayout(this,getSampleRate()));
}
return 0;
}
unsigned int AmRtpAudio::getFrameSize()
{
if (!fmt.get())
return 0;
return ((AmAudioRtpFormat*)fmt.get())->getFrameSize();
}
int AmRtpAudio::setCurrentPayload(int payload)
{
if(payload != this->payload){
PayloadMappingTable::iterator pmt_it = pl_map.find(payload);
if(pmt_it == pl_map.end()){
ERROR("Could not set current payload: payload %i unknown to this stream\n",payload);
return -1;
}
unsigned char index = pmt_it->second.index;
if(index >= payloads.size()){
ERROR("Could not set current payload: payload %i maps to invalid index %i\n",
payload, index);
return -1;
}
this->payload = payload;
int res = ((AmAudioRtpFormat*)fmt.get())->setCurrentPayload(payloads[index]);
amci_codec_t* codec = fmt->getCodec();
use_default_plc = ((codec==NULL) || (codec->plc == NULL));
return res;
}
else {
return 0;
}
}
//int AmRtpAudio::getCurrentPayload() {
//return ((AmAudioRtpFormat *) fmt.get())->getCurrentPayload();
//}
unsigned int AmRtpAudio::conceal_loss(unsigned int ts_diff, unsigned char *buffer)
{
int s=0;
if(!use_default_plc){
amci_codec_t* codec = fmt->getCodec();
long h_codec = fmt->getHCodec();
assert(codec && codec->plc);
s = (*codec->plc)(buffer, PCM16_S2B(ts_diff),
fmt->channels,getSampleRate(),h_codec);
DBG("codec specific PLC (ts_diff = %i; s = %i)\n",ts_diff,s);
}
else {
s = default_plc(buffer, PCM16_S2B(ts_diff),
fmt->channels,getSampleRate());
DBG("default PLC (ts_diff = %i; s = %i)\n",ts_diff,s);
}
return s;
}
unsigned int AmRtpAudio::default_plc(unsigned char* out_buf,
unsigned int size,
unsigned int channels,
unsigned int sample_rate)
{
short* buf_offset = (short*)out_buf;
#ifdef USE_SPANDSP_PLC
plc_fillin(plc_state, buf_offset, PCM16_B2S(size));
#else
for(unsigned int i=0; i<(PCM16_B2S(size)/FRAMESZ); i++){
fec->dofe(buf_offset);
buf_offset += FRAMESZ;
}
#endif // USE_SPANDSP_PLC
return PCM16_S2B(buf_offset - (short*)out_buf);
}
void AmRtpAudio::add_to_history(int16_t *buffer, unsigned int size)
{
if (!use_default_plc)
return;
#ifdef USE_SPANDSP_PLC
plc_rx(plc_state, buffer, PCM16_B2S(size));
#else // USE_SPANDSP_PLC
int16_t* buf_offset = buffer;
unsigned int sample_rate = getSampleRate();
for(unsigned int i=0; i<(PCM16_B2S(size)/FRAMESZ); i++){
fec->addtohistory(buf_offset);
buf_offset += FRAMESZ;
}
#endif // USE_SPANDSP_PLC
}
void AmRtpAudio::setPlayoutType(PlayoutType type)
{
if (m_playout_type != type)
{
if (type == ADAPTIVE_PLAYOUT) {
session->lockAudio();
m_playout_type = type;
if (fmt.get())
playout_buffer.reset(new AmAdaptivePlayout(this,getSampleRate()));
session->unlockAudio();
DBG("Adaptive playout buffer activated\n");
}
else if (type == JB_PLAYOUT) {
session->lockAudio();
m_playout_type = type;
if (fmt.get())
playout_buffer.reset(new AmJbPlayout(this,getSampleRate()));
session->unlockAudio();
DBG("Adaptive jitter buffer activated\n");
}
else {
session->lockAudio();
m_playout_type = type;
if (fmt.get())
playout_buffer.reset(new AmPlayoutBuffer(this,getSampleRate()));
session->unlockAudio();
DBG("Simple playout buffer activated\n");
}
}
}