audiortp.cpp 11.4 KB
Newer Older
savoirfairelinux's avatar
savoirfairelinux committed
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92
/**
 *  Copyright (C) 2004 Savoir-Faire Linux inc.
 *  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 2 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 <cstdio>
#include <cstdlib>
#include <ccrtp/rtp.h>

#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <qhostaddress.h>
#include <qstring.h>

#include "audiocodec.h"
#include "configuration.h"
#include "manager.h"
#include "audiortp.h"
#include "sip.h"
#include "../stund/stun.h"

#ifdef  CCXX_NAMESPACES
using namespace ost;
#endif

////////////////////////////////////////////////////////////////////////////////
// AudioRtp                                                          
////////////////////////////////////////////////////////////////////////////////
AudioRtp::AudioRtp (SIP *sip, Manager *manager) {
	QString svr;
	this->sip = sip;
	this->manager = manager;
	RTXThread = NULL;
	
	if (!manager->useStun()) {
		if (Config::gets("Signalisations/SIP.sipproxy")) {
			svr = Config::gets("Signalisations/SIP.sipproxy");
		}
	} else {
		svr = Config::gets("Signalisations/SIP.hostPart");
	}
}

AudioRtp::~AudioRtp (void) {
}

int 
AudioRtp::createNewSession (SipCall *ca) {
	// Start RTP Send/Receive threads
	ca->enable_audio = 1;
	if (!manager->useStun()) { 
		symetric = false;
	} else {
		symetric = true;
	}

	RTXThread = new AudioRtpRTX (ca, manager->audiodriver, manager, symetric);
	RTXThread->start();
	
/*	if (!manager->useStun()) {
		RTXThread = new AudioRtpRTX (ca, manager->audiodriver, manager);
		qDebug("new RTXThread = 0x%X", (int)RTXThread);
		RTXThread->start();
	} else {
		symThread = new AudioRtpSymmetric (ca, manager->audiodriver, manager);
		symThread->start();
	}
*/	
	return 0;
}

	
void
AudioRtp::closeRtpSession (SipCall *ca) {
	// This will make RTP threads finish.
	ca->enable_audio = -1;

savoirfairelinux's avatar
savoirfairelinux committed
93
	if (RTXThread != NULL) {
savoirfairelinux's avatar
savoirfairelinux committed
94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452
	// Wait for them...and delete.
	RTXThread->join();
	
		delete RTXThread;
		qDebug ("RTXThread deleted!");
		RTXThread = NULL;
	}

/*	if (!manager->useStun()) {
		RTXThread->join();
		if (RTXThread != NULL) {
			delete RTXThread;
			qDebug ("RTXThread deleted!");
			RTXThread = NULL;
		}
	} else {
		symThread->join();
		if (symThread != NULL) {
			delete symThread;
			symThread = NULL;
		}
	}
*/	
	// Flush audio read buffer
	manager->audiodriver->resetDevice();
}

void
AudioRtp::rtpexit (void) {
}

////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class                                                          //
////////////////////////////////////////////////////////////////////////////////
AudioRtpRTX::AudioRtpRTX (SipCall *sipcall, AudioDrivers *driver, 
						Manager *mngr, bool sym) {
	this->ca = sipcall;
	this->audioDevice = driver;
	this->manager = mngr;
	this->sym =sym;

	// TODO: Change bind address according to user settings.
	InetHostAddress local_ip("0.0.0.0");

	if (!sym) {
		sessionRecv = new RTPSession (local_ip, ca->getLocalAudioPort());
		sessionSend = new RTPSession (local_ip);
	} else {
		int forcedPort = manager->getFirewallPort();
		qDebug("Forced port %d", forcedPort);
		session = new SymmetricRTPSession (local_ip, forcedPort);
	}
}

AudioRtpRTX::~AudioRtpRTX () {
	if (!sym) {
		if (sessionRecv != NULL) {
			delete sessionRecv;	
			sessionRecv = NULL;
		}
		if (sessionSend != NULL) {
			delete sessionSend;	
			sessionSend = NULL;
		}
	} else {
		if (session != NULL) {
			delete session;
			session = NULL;
		}
	}
}

void
AudioRtpRTX::run (void) {
	AudioCodec 		 ac;
	unsigned char	*data_to_send;
	short			*data_mute;
	short			*data_from_mic;
	int				 i,
					 compSize, 
					 timestamp;
	int				 expandedSize;
	short			*data_for_speakers = NULL;
	
	data_for_speakers = new short[2048];
	data_from_mic = new short[1024];
	data_to_send = new unsigned char[1024];
	data_mute = new short[1024];

	InetHostAddress remote_ip(ca->remote_sdp_audio_ip);
	
	if (!remote_ip) {
	   qDebug("RTX: IP address is not correct!");
	   exit();
	} else {
		qDebug("RTX: Connected to %s:%d",
				ca->remote_sdp_audio_ip, ca->remote_sdp_audio_port);
	}
	
	// Initialization
	if (!sym) {
		sessionRecv->setSchedulingTimeout (100000);
		sessionRecv->setExpireTimeout(1000000);
		
		sessionSend->setSchedulingTimeout(10000);
		sessionSend->setExpireTimeout(1000000);
	} else {
		session->setSchedulingTimeout(10000);
		session->setExpireTimeout(1000000);
	}

#if 0 // Necessaire ?
    if (!sessionRecv->addDestination(remote_ip,
				(unsigned short) ca->remote_sdp_audio_port)) {
		qDebug("RTX recv: could not connect to port %d", 
				ca->remote_sdp_audio_port);
		this->exit();
	} else {
		qDebug("RTP(Recv): Added destination %s:%d",
				remote_ip.getHostname(),
				(unsigned short) ca->remote_sdp_audio_port)
	}
#endif

	if (!sym) {
		if (!sessionSend->addDestination (remote_ip, 
					(unsigned short) ca->remote_sdp_audio_port)) {
			qDebug("RTX send: could not connect to port %d", 
					ca->remote_sdp_audio_port);
			this->exit();
		} else {
			qDebug("RTP(Send): Added destination %s:%d",
					remote_ip.getHostname(),
					(unsigned short) ca->remote_sdp_audio_port);
		}

		sessionRecv->setPayloadFormat(StaticPayloadFormat(
				(StaticPayloadType) ca->payload));
		sessionSend->setPayloadFormat(StaticPayloadFormat(
				(StaticPayloadType) ca->payload));
		setCancel(cancelImmediate);
		sessionSend->setMark(true);

	} else {
		if (!session->addDestination (remote_ip, 
					(unsigned short) ca->remote_sdp_audio_port)) {
			qDebug("Symmetric: could not connect to port %d", 
					ca->remote_sdp_audio_port);
			this->exit();
		} else {
			qDebug("Symmetric: Connected to %s:%d",
					remote_ip.getHostname(),
					(unsigned short) ca->remote_sdp_audio_port);

			session->setPayloadFormat(StaticPayloadFormat(
				(StaticPayloadType) ca->payload));
			setCancel(cancelImmediate);
		}
	}
	
	timestamp = 0;

	// TODO: get frameSize from user config 
	int frameSize = 20; // 20ms frames
	TimerPort::setTimer(frameSize);
	
	// start running the packet queue scheduler.
	if (!sym) {
		sessionRecv->startRunning();
		sessionSend->startRunning();	
	} else {
		session->startRunning();
	}
	
	while (ca->enable_audio != -1) {
		////////////////////////////
		// Send session
		////////////////////////////
		if (!manager->mute) {
			i = audioDevice->readBuffer (data_from_mic, 320);
		} else {
			// When IP-phone user click on mute button, we read buffer of a
			// temp buffer to avoid delay in sound.
			i = audioDevice->readBuffer (data_mute, 320);
		}

		// Encode acquired audio sample
		compSize = AudioCodec::codecEncode (
				ac.handleCodecs[0], 
				data_to_send,
				data_from_mic, i);

		// Send encoded audio sample
		if (!sym) {
			sessionSend->putData(timestamp, data_to_send, compSize);
		} else {
			session->putData(timestamp, data_to_send, compSize);
		}
		timestamp += compSize;

		////////////////////////////
		// Recv session
		////////////////////////////
		const AppDataUnit* adu = NULL;

		do {
			Thread::sleep(5); // in msec.
			if (!sym) {
				adu = sessionRecv->getData(sessionRecv->getFirstTimestamp());
			} else {
				adu = session->getData(session->getFirstTimestamp());
			}
		} while (adu == NULL);

		// Decode data with relevant codec
		expandedSize = AudioCodec::codecDecode (
				adu->getType(),
				data_for_speakers,
				(unsigned char*) adu->getData(),
				adu->getSize());
		
		// Write decoded data to sound device
		i = audioDevice->writeBuffer (data_for_speakers, expandedSize);
		delete adu;


		// Let's wait for the next transmit cycle
		Thread::sleep(TimerPort::getTimer());
		TimerPort::incTimer(frameSize); // 'frameSize' ms
	}
		 
	delete[] data_for_speakers;
	delete[] data_from_mic;
	delete[] data_mute;
	delete[] data_to_send;
	this->exit();
}

#if 0
////////////////////////////////////////////////////////////////////////////////
// AudioRtpSymmetric Class                                                    //
////////////////////////////////////////////////////////////////////////////////
AudioRtpSymmetric::AudioRtpSymmetric (SipCall *sipcall, AudioDrivers *driver,
										Manager *mngr) {
	this->ca = sipcall;
	this->audioDevice = driver;
	this->manager = mngr;

	InetHostAddress local_ip("192.168.1.172");
	int forcedPort = manager->getFirewallPort();
	qDebug("port firewall = %d", forcedPort);

	session = new SymmetricRTPSession (local_ip, forcedPort);
}

AudioRtpSymmetric::~AudioRtpSymmetric () {
	delete session;	
	terminate();
}

void
AudioRtpSymmetric::run (void) {
	AudioCodec 		 ac;
	unsigned char	*data_to_send;
	short			*data_from_mic;
	int				 i,
					 compSize, 
					 timestamp;
	int				 expandedSize;
	short			*data_for_speakers = NULL;

	data_for_speakers = new short[2048];
	data_from_mic = new short[1024];
	data_to_send = new unsigned char[1024];

	InetHostAddress remote_ip;
	remote_ip = ca->remote_sdp_audio_ip;
	int remote_port = ca->remote_sdp_audio_port;
	
	if (!remote_ip) {
	   qDebug("Symmetric: IP address is not correct!");
	   exit();
	} 
	
	// Initialization
	session->setSchedulingTimeout(10000);
	session->setExpireTimeout(1000000);

	if (!session->addDestination (remote_ip, (unsigned short) remote_port)) {
		qDebug("Symmetric: could not connect to port %d", remote_port);
		this->exit();
	} else {
		qDebug("Symmetric: Connected to %s:%d",
				ca->remote_sdp_audio_ip, remote_port);
	}
	
    session->setPayloadFormat(StaticPayloadFormat(
				(enum StaticPayloadType) ca->payload));
	
	setCancel(cancelImmediate);
	
	timestamp = 0;

	// TODO: get frameSize from user config 
	int frameSize = 20; // 20ms frames
	TimerPort::setTimer(frameSize);
	
	// start running the packet queue scheduler.
	session->startRunning();	
 
	while (ca->enable_audio != -1) {
		////////////////////////////
		// Send session
		////////////////////////////
		i = audioDevice->readBuffer (data_from_mic, 320);
		// Encode acquired audio sample
		compSize = AudioCodec::codecEncode (
				ac.handleCodecs[0], 
				data_to_send,
				data_from_mic, i);

		// Send encoded audio sample
		session->putData(timestamp, data_to_send, compSize);
		timestamp += compSize;

		////////////////////////////
		// Recv session
		////////////////////////////
		const AppDataUnit* adu = NULL;
		
		do {
			Thread::sleep(10);
			adu = session->getData(session->getFirstTimestamp());	
		} while (adu == NULL);

		// Decode data with relevant codec
		expandedSize = AudioCodec::codecDecode (
				adu->getType(),
				data_for_speakers,
				(unsigned char*) adu->getData(),
				adu->getSize());

		// Write decoded data to sound device
		audioDevice->writeBuffer (data_for_speakers, expandedSize);
		delete adu;

		// Let's wait for the next cycle
		Thread::sleep(TimerPort::getTimer());
		TimerPort::incTimer(frameSize); // 'frameSize' ms
	}
		 
	delete[] data_for_speakers;
	delete[] data_from_mic;
	delete[] data_to_send;
	this->exit();
}
#endif

// EOF