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playCommon.cpp
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playCommon.cpp
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/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
This library 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 Lesser General Public License for
more details.
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// Copyright (c) 1996-2010, Live Networks, Inc. All rights reserved
// A common framework, used for the "openRTSP" and "playSIP" applications
// Implementation
#include "playCommon.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
#ifdef SUPPORT_REAL_RTSP
#include "../RealRTSP/include/RealRTSP.hh"
#endif
#if defined(__WIN32__) || defined(_WIN32)
#define snprintf _snprintf
#else
#include <signal.h>
#define USE_SIGNALS 1
#endif
// Forward function definitions:
void setupStreams();
void startPlayingStreams();
void tearDownStreams();
void closeMediaSinks();
void subsessionAfterPlaying(void* clientData);
void subsessionByeHandler(void* clientData);
void sessionAfterPlaying(void* clientData = NULL);
void sessionTimerHandler(void* clientData);
void shutdown(int exitCode = 1);
void signalHandlerShutdown(int sig);
void checkForPacketArrival(void* clientData);
void checkInterPacketGaps(void* clientData);
void beginQOSMeasurement();
char const* progName;
UsageEnvironment* env;
Medium* ourClient = NULL;
MediaSession* session = NULL;
TaskToken sessionTimerTask = NULL;
TaskToken arrivalCheckTimerTask = NULL;
TaskToken interPacketGapCheckTimerTask = NULL;
TaskToken qosMeasurementTimerTask = NULL;
Boolean createReceivers = True;
Boolean outputQuickTimeFile = False;
Boolean generateMP4Format = False;
QuickTimeFileSink* qtOut = NULL;
Boolean outputAVIFile = False;
AVIFileSink* aviOut = NULL;
Boolean audioOnly = False;
Boolean videoOnly = False;
char const* singleMedium = NULL;
int verbosityLevel = 1; // by default, print verbose output
double duration = 0;
double durationSlop = -1.0; // extra seconds to play at the end
double initialSeekTime = 0.0f;
double scale = 1.0f;
unsigned interPacketGapMaxTime = 0;
unsigned totNumPacketsReceived = ~0; // used if checking inter-packet gaps
Boolean playContinuously = False;
int simpleRTPoffsetArg = -1;
Boolean sendOptionsRequest = True;
Boolean sendOptionsRequestOnly = False;
Boolean oneFilePerFrame = False;
Boolean notifyOnPacketArrival = False;
Boolean streamUsingTCP = False;
portNumBits tunnelOverHTTPPortNum = 0;
char* username = NULL;
char* password = NULL;
char* proxyServerName = NULL;
unsigned short proxyServerPortNum = 0;
unsigned char desiredAudioRTPPayloadFormat = 0;
char* mimeSubtype = NULL;
unsigned short movieWidth = 240; // default
Boolean movieWidthOptionSet = False;
unsigned short movieHeight = 180; // default
Boolean movieHeightOptionSet = False;
unsigned movieFPS = 15; // default
Boolean movieFPSOptionSet = False;
char const* fileNamePrefix = "";
unsigned fileSinkBufferSize = 2*100000; //yzhu: change 100000 into 2*100000,to adapt to Amba 720p encoding or bigger
unsigned socketInputBufferSize = 0;
Boolean packetLossCompensate = False;
Boolean syncStreams = False;
Boolean generateHintTracks = False;
unsigned qosMeasurementIntervalMS = 0; // 0 means: Don't output QOS data
unsigned statusCode = 0;
struct timeval startTime;
void usage() {
*env << "Usage: " << progName
<< " [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V] [-d <duration>] [-D <max-inter-packet-gap-time> [-c] [-S <offset>] [-n] [-O]"
<< (controlConnectionUsesTCP ? " [-t|-T <http-port>]" : "")
<< " [-u <username> <password>"
<< (allowProxyServers ? " [<proxy-server> [<proxy-server-port>]]" : "")
<< "]" << (supportCodecSelection ? " [-A <audio-codec-rtp-payload-format-code>|-M <mime-subtype-name>]" : "")
<< " [-s <initial-seek-time>] [-z <scale>]"
<< " [-w <width> -h <height>] [-f <frames-per-second>] [-y] [-H] [-Q [<measurement-interval>]] [-F <filename-prefix>] [-b <file-sink-buffer-size>] [-B <input-socket-buffer-size>] [-I <input-interface-ip-address>] [-m] <url> (or " << progName << " -o [-V] <url>)\n";
//##### Add "-R <dest-rtsp-url>" #####
shutdown();
}
int main(int argc, char** argv)
{
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
progName = argv[0];
gettimeofday(&startTime, NULL);
#ifdef USE_SIGNALS
// Allow ourselves to be shut down gracefully by a SIGHUP or a SIGUSR1:
signal(SIGHUP, signalHandlerShutdown);
signal(SIGUSR1, signalHandlerShutdown);
#endif
unsigned short desiredPortNum = 0;
// unfortunately we can't use getopt() here, as Windoze doesn't have it
while (argc > 2) {
char* const opt = argv[1];
if (opt[0] != '-') usage();
switch (opt[1]) {
case 'p': { // specify start port number
int portArg;
if (sscanf(argv[2], "%d", &portArg) != 1) {
usage();
}
if ((portArg <= 0) || (portArg >= 65536) || (portArg&1)) {
*env << "bad port number: " << portArg
<< " (must be even, and in the range (0,65536))\n";
usage();
}
desiredPortNum = (unsigned short)portArg;
++argv; --argc;
break;
}
case 'r': { // do not receive data (instead, just 'play' the stream(s))
createReceivers = False;
break;
}
case 'q': { // output a QuickTime file (to stdout)
outputQuickTimeFile = True;
break;
}
case '4': { // output a 'mp4'-format file (to stdout)
outputQuickTimeFile = True;
generateMP4Format = True;
break;
}
case 'i': { // output an AVI file (to stdout)
outputAVIFile = True;
break;
}
case 'I': { // specify input interface...
NetAddressList addresses(argv[2]);
if (addresses.numAddresses() == 0) {
*env << "Failed to find network address for \"" << argv[2] << "\"";
break;
}
ReceivingInterfaceAddr = *(unsigned*)(addresses.firstAddress()->data());
++argv; --argc;
break;
}
case 'a': { // receive/record an audio stream only
audioOnly = True;
singleMedium = "audio";
break;
}
case 'v': { // receive/record a video stream only
videoOnly = True;
singleMedium = "video";
break;
}
case 'V': { // disable verbose output
verbosityLevel = 0;
break;
}
case 'd': { // specify duration, or how much to delay after end time
float arg;
if (sscanf(argv[2], "%g", &arg) != 1) {
usage();
}
if (argv[2][0] == '-') { // not "arg<0", in case argv[2] was "-0"
// a 'negative' argument was specified; use this for "durationSlop":
duration = 0; // use whatever's in the SDP
durationSlop = -arg;
} else {
duration = arg;
durationSlop = 0;
}
++argv; --argc;
break;
}
case 'D': { // specify maximum number of seconds to wait for packets:
if (sscanf(argv[2], "%u", &interPacketGapMaxTime) != 1) {
usage();
}
++argv; --argc;
break;
}
case 'c': { // play continuously
playContinuously = True;
break;
}
case 'S': { // specify an offset to use with "SimpleRTPSource"s
if (sscanf(argv[2], "%d", &simpleRTPoffsetArg) != 1) {
usage();
}
if (simpleRTPoffsetArg < 0) {
*env << "offset argument to \"-S\" must be >= 0\n";
usage();
}
++argv; --argc;
break;
}
case 'O': { // Don't send an "OPTIONS" request before "DESCRIBE"
sendOptionsRequest = False;
break;
}
case 'o': { // Send only the "OPTIONS" request to the server
sendOptionsRequestOnly = True;
break;
}
case 'm': { // output multiple files - one for each frame
oneFilePerFrame = True;
break;
}
case 'n': { // notify the user when the first data packet arrives
notifyOnPacketArrival = True;
break;
}
case 't': {
// stream RTP and RTCP over the TCP 'control' connection
if (controlConnectionUsesTCP) {
streamUsingTCP = True;
} else {
usage();
}
break;
}
case 'T': {
// stream RTP and RTCP over a HTTP connection
if (controlConnectionUsesTCP) {
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the HTTP server port number:
if (sscanf(argv[2], "%hu", &tunnelOverHTTPPortNum) == 1
&& tunnelOverHTTPPortNum > 0) {
++argv; --argc;
break;
}
}
}
// If we get here, the option was specified incorrectly:
usage();
break;
}
case 'u': { // specify a username and password
username = argv[2];
password = argv[3];
argv+=2; argc-=2;
if (allowProxyServers && argc > 3 && argv[2][0] != '-') {
// The next argument is the name of a proxy server:
proxyServerName = argv[2];
++argv; --argc;
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the proxy server port number:
if (sscanf(argv[2], "%hu", &proxyServerPortNum) != 1) {
usage();
}
++argv; --argc;
}
}
break;
}
case 'A': { // specify a desired audio RTP payload format
unsigned formatArg;
if (sscanf(argv[2], "%u", &formatArg) != 1 || formatArg >= 96) {
usage();
}
desiredAudioRTPPayloadFormat = (unsigned char)formatArg;
++argv; --argc;
break;
}
case 'M': { // specify a MIME subtype for a dynamic RTP payload type
mimeSubtype = argv[2];
if (desiredAudioRTPPayloadFormat==0) desiredAudioRTPPayloadFormat = 96;
++argv; --argc;
break;
}
case 'w': {
// specify a width (pixels) for an output QuickTime or AVI movie
if (sscanf(argv[2], "%hu", &movieWidth) != 1) {
usage();
}
movieWidthOptionSet = True;
++argv; --argc;
break;
}
case 'h': {
// specify a height (pixels) for an output QuickTime or AVI movie
if (sscanf(argv[2], "%hu", &movieHeight) != 1) {
usage();
}
movieHeightOptionSet = True;
++argv; --argc;
break;
}
case 'f': {
// specify a frame rate (per second) for an output QT or AVI movie
if (sscanf(argv[2], "%u", &movieFPS) != 1) {
usage();
}
movieFPSOptionSet = True;
++argv; --argc;
break;
}
case 'F': { // specify a prefix for the audio and video output files
fileNamePrefix = argv[2];
++argv; --argc;
break;
}
case 'b': { // specify the size of buffers for "FileSink"s
if (sscanf(argv[2], "%u", &fileSinkBufferSize) != 1) {
usage();
}
++argv; --argc;
break;
}
case 'B': { // specify the size of input socket buffers
if (sscanf(argv[2], "%u", &socketInputBufferSize) != 1) {
usage();
}
++argv; --argc;
break;
}
// Note: The following option is deprecated, and may someday be removed:
case 'l': { // try to compensate for packet loss by repeating frames
packetLossCompensate = True;
break;
}
case 'y': { // synchronize audio and video streams
syncStreams = True;
break;
}
case 'H': { // generate hint tracks (as well as the regular data tracks)
generateHintTracks = True;
break;
}
case 'Q': { // output QOS measurements
qosMeasurementIntervalMS = 1000; // default: 1 second
if (argc > 3 && argv[2][0] != '-') {
// The next argument is the measurement interval,
// in multiples of 100 ms
if (sscanf(argv[2], "%u", &qosMeasurementIntervalMS) != 1) {
usage();
}
qosMeasurementIntervalMS *= 100;
++argv; --argc;
}
break;
}
case 's': { // specify initial seek time (trick play)
double arg;
if (sscanf(argv[2], "%lg", &arg) != 1 || arg < 0) {
usage();
}
initialSeekTime = arg;
++argv; --argc;
break;
}
case 'z': { // scale (trick play)
float arg;
if (sscanf(argv[2], "%g", &arg) != 1 || arg == 0.0f) {
usage();
}
scale = arg;
++argv; --argc;
break;
}
default: {
usage();
break;
}
}
++argv; --argc;
}
if (argc != 2) usage();
if (outputQuickTimeFile && outputAVIFile) {
*env << "The -i and -q (or -4) flags cannot both be used!\n";
usage();
}
Boolean outputCompositeFile = outputQuickTimeFile || outputAVIFile;
if (!createReceivers && outputCompositeFile) {
*env << "The -r and -q (or -4 or -i) flags cannot both be used!\n";
usage();
}
if (outputCompositeFile && !movieWidthOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -w. Assuming a video width of "
<< movieWidth << " pixels\n";
}
if (outputCompositeFile && !movieHeightOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -h. Assuming a video height of "
<< movieHeight << " pixels\n";
}
if (outputCompositeFile && !movieFPSOptionSet) {
*env << "Warning: The -q, -4 or -i option was used, but not -f. Assuming a video frame rate of "
<< movieFPS << " frames-per-second\n";
}
if (audioOnly && videoOnly) {
*env << "The -a and -v flags cannot both be used!\n";
usage();
}
if (sendOptionsRequestOnly && !sendOptionsRequest) {
*env << "The -o and -O flags cannot both be used!\n";
usage();
}
if (tunnelOverHTTPPortNum > 0) {
if (streamUsingTCP) {
*env << "The -t and -T flags cannot both be used!\n";
usage();
} else {
streamUsingTCP = True;
}
}
if (!createReceivers && notifyOnPacketArrival) {
*env << "Warning: Because we're not receiving stream data, the -n flag has no effect\n";
}
if (durationSlop < 0) {
// This parameter wasn't set, so use a default value.
// If we're measuring QOS stats, then don't add any slop, to avoid
// having 'empty' measurement intervals at the end.
durationSlop = qosMeasurementIntervalMS > 0 ? 0.0 : 5.0;
}
char* url = argv[1];
// Create our client object:
ourClient = createClient(*env, verbosityLevel, progName);
if (ourClient == NULL) {
*env << "Failed to create " << clientProtocolName
<< " client: " << env->getResultMsg() << "\n";
shutdown();
}
if (sendOptionsRequest) {
// Begin by sending an "OPTIONS" command:
char* optionsResponse
= getOptionsResponse(ourClient, url, username, password);
if (sendOptionsRequestOnly) {
if (optionsResponse == NULL) {
*env << clientProtocolName << " \"OPTIONS\" request failed: "
<< env->getResultMsg() << "\n";
} else {
*env << clientProtocolName << " \"OPTIONS\" request returned: "
<< optionsResponse << "\n";
}
shutdown();
}
delete[] optionsResponse;
}
// Open the URL, to get a SDP description:
char* sdpDescription
= getSDPDescriptionFromURL(ourClient, url, username, password,
proxyServerName, proxyServerPortNum,
desiredPortNum);
if (sdpDescription == NULL) {
*env << "Failed to get a SDP description from URL \"" << url
<< "\": " << env->getResultMsg() << "\n";
shutdown();
}
*env << "Opened URL \"" << url
<< "\", returning a SDP description:\n" << sdpDescription << "\n";
// Create a media session object from this SDP description:
session = MediaSession::createNew(*env, sdpDescription);
delete[] sdpDescription;
if (session == NULL) {
*env << "Failed to create a MediaSession object from the SDP description: "
<< env->getResultMsg() << "\n";
shutdown();
} else if (!session->hasSubsessions()) {
*env << "This session has no media subsessions (i.e., \"m=\" lines)\n";
shutdown();
}
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*session);
MediaSubsession *subsession;
Boolean madeProgress = False;
char const* singleMediumToTest = singleMedium;
while ((subsession = iter.next()) != NULL) {
// If we've asked to receive only a single medium, then check this now:
if (singleMediumToTest != NULL) {
if (strcmp(subsession->mediumName(), singleMediumToTest) != 0) {
*env << "Ignoring \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession, because we've asked to receive a single "
<< singleMedium
<< " session only\n";
continue;
} else {
// Receive this subsession only
singleMediumToTest = "xxxxx";
// this hack ensures that we get only 1 subsession of this type
}
}
if (desiredPortNum != 0) {
subsession->setClientPortNum(desiredPortNum);
desiredPortNum += 2;
}
if (createReceivers) {
if (!subsession->initiate(simpleRTPoffsetArg)) {
*env << "Unable to create receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession: " << env->getResultMsg() << "\n";
} else {
*env << "Created receiver for \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession (client ports " << subsession->clientPortNum()
<< "-" << subsession->clientPortNum()+1 << ")\n";
madeProgress = True;
if (subsession->rtpSource() != NULL) {
// Because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
// Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),
// or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.
// (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,
// then the input data rate may be large enough to justify increasing the OS socket buffer size also.)
int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
unsigned curBufferSize = getReceiveBufferSize(*env, socketNum);
if (socketInputBufferSize > 0 ||
fileSinkBufferSize > curBufferSize) {
unsigned newBufferSize = ((socketInputBufferSize) > 0 ?
socketInputBufferSize :
fileSinkBufferSize);
newBufferSize = setReceiveBufferTo(*env, socketNum, newBufferSize);
if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it:
*env << "Changed socket receive buffer size for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession from "
<< curBufferSize << " to "
<< newBufferSize << " bytes\n";
}
}
}
}
} else {
if (subsession->clientPortNum() == 0) {
*env << "No client port was specified for the \""
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession. (Try adding the \"-p <portNum>\" option.)\n";
} else {
madeProgress = True;
}
}
}
if (!madeProgress) shutdown();
// Perform additional 'setup' on each subsession, before playing them:
setupStreams();
// Create output files:
if (createReceivers) {
if (outputQuickTimeFile) {
// Create a "QuickTimeFileSink", to write to 'stdout':
qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate,
syncStreams,
generateHintTracks,
generateMP4Format);
if (qtOut == NULL) {
*env << "Failed to create QuickTime file sink for stdout: "
<< env->getResultMsg();
shutdown();
}
qtOut->startPlaying(sessionAfterPlaying, NULL);
} else if (outputAVIFile) {
// Create an "AVIFileSink", to write to 'stdout':
aviOut = AVIFileSink::createNew(*env, *session, "stdout",
fileSinkBufferSize,
movieWidth, movieHeight,
movieFPS,
packetLossCompensate);
if (aviOut == NULL) {
*env << "Failed to create AVI file sink for stdout: "
<< env->getResultMsg();
shutdown();
}
aviOut->startPlaying(sessionAfterPlaying, NULL);
#ifdef SUPPORT_REAL_RTSP
} else if (session->isRealNetworksRDT) {
// For RealNetworks' sessions, we create a single output file,
// named "output.rm".
char outFileName[1000];
if (singleMedium == NULL) {
snprintf(outFileName, sizeof(outFileName),
"%soutput.rm", fileNamePrefix);
} else {
// output to 'stdout' as normal, even though we actually output all media
sprintf(outFileName, "stdout");
}
FileSink* fileSink = FileSink::createNew(*env, outFileName,
fileSinkBufferSize,
oneFilePerFrame);
// The output file needs to begin with a special 'RMFF' header,
// in order for it to be usable. Write this header first:
unsigned headerSize;
unsigned char* headerData = RealGenerateRMFFHeader(session, headerSize);
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
fileSink->addData(headerData, headerSize, timeNow);
delete[] headerData;
// Start playing the output file from the first subsession.
// (Hack: Because all subsessions' data is actually multiplexed on the
// single RTSP TCP connection, playing from one subsession is sufficient.)
iter.reset();
madeProgress = False;
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
fileSink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying, subsession);
madeProgress = True;
break; // play from one subsession only
}
if (!madeProgress) shutdown();
#endif
} else {
// Create and start "FileSink"s for each subsession:
madeProgress = False;
iter.reset();
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
// Create an output file for each desired stream:
char outFileName[1000];
if (singleMedium == NULL) {
// Output file name is
// "<filename-prefix><medium_name>-<codec_name>-<counter>"
static unsigned streamCounter = 0;
snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",
fileNamePrefix, subsession->mediumName(),
subsession->codecName(), ++streamCounter);
} else {
sprintf(outFileName, "stdout");
}
FileSink* fileSink;
if (strcmp(subsession->mediumName(), "audio") == 0 &&
(strcmp(subsession->codecName(), "AMR") == 0 ||
strcmp(subsession->codecName(), "AMR-WB") == 0)) {
// For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
fileSink = AMRAudioFileSink::createNew(*env, outFileName,
fileSinkBufferSize,
oneFilePerFrame);
} else if (strcmp(subsession->mediumName(), "video") == 0 &&
(strcmp(subsession->codecName(), "H264") == 0)) {
// For H.264 video stream, we use a special sink that insert start_codes:
fileSink = H264VideoFileSink::createNew(*env, outFileName,
fileSinkBufferSize,
oneFilePerFrame);
} else {
// Normal case:
fileSink = FileSink::createNew(*env, outFileName,
fileSinkBufferSize,
oneFilePerFrame);
}
subsession->sink = fileSink;
if (subsession->sink == NULL) {
*env << "Failed to create FileSink for \"" << outFileName
<< "\": " << env->getResultMsg() << "\n";
} else {
if (singleMedium == NULL) {
*env << "Created output file: \"" << outFileName << "\"\n";
} else {
*env << "Outputting data from the \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession to 'stdout'\n";
}
if (strcmp(subsession->mediumName(), "video") == 0 &&
strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
subsession->fmtp_config() != NULL) {
// For MPEG-4 video RTP streams, the 'config' information
// from the SDP description contains useful VOL etc. headers.
// Insert this data at the front of the output file:
unsigned configLen;
unsigned char* configData
= parseGeneralConfigStr(subsession->fmtp_config(), configLen);
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
fileSink->addData(configData, configLen, timeNow);
delete[] configData;
}
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL) {
subsession->rtcpInstance()->setByeHandler(subsessionByeHandler,
subsession);
}
madeProgress = True;
}
}
if (!madeProgress) shutdown();
}
}
// Finally, start playing each subsession, to start the data flow:
startPlayingStreams();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
void setupStreams() {
MediaSubsessionIterator iter(*session);
MediaSubsession *subsession;
Boolean madeProgress = False;
while ((subsession = iter.next()) != NULL) {
if (subsession->clientPortNum() == 0) continue; // port # was not set
if (!clientSetupSubsession(ourClient, subsession, streamUsingTCP)) {
*env << "Failed to setup \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession: " << env->getResultMsg() << "\n";
} else {
*env << "Setup \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession (client ports " << subsession->clientPortNum()
<< "-" << subsession->clientPortNum()+1 << ")\n";
madeProgress = True;
}
}
if (!madeProgress) shutdown();
}
void startPlayingStreams() {
if (duration == 0) {
if (scale > 0) duration = session->playEndTime() - initialSeekTime; // use SDP end time
else if (scale < 0) duration = initialSeekTime;
}
if (duration < 0) duration = 0.0;
if (!clientStartPlayingSession(ourClient, session)) {
*env << "Failed to start playing session: " << env->getResultMsg() << "\n";
shutdown();
} else {
*env << "Started playing session\n";
}
if (qosMeasurementIntervalMS > 0) {
// Begin periodic QOS measurements:
beginQOSMeasurement();
}
// Figure out how long to delay (if at all) before shutting down, or
// repeating the playing
Boolean timerIsBeingUsed = False;
double secondsToDelay = duration;
if (duration > 0) {
double const maxDelayTime
= (double)( ((unsigned)0x7FFFFFFF)/1000000.0 );
if (duration > maxDelayTime) {
*env << "Warning: specified end time " << duration
<< " exceeds maximum " << maxDelayTime
<< "; will not do a delayed shutdown\n";
} else {
timerIsBeingUsed = True;
double absScale = scale > 0 ? scale : -scale; // ASSERT: scale != 0
secondsToDelay = duration/absScale + durationSlop;
int uSecsToDelay = (int)(secondsToDelay*1000000.0);
sessionTimerTask = env->taskScheduler().scheduleDelayedTask(
uSecsToDelay, (TaskFunc*)sessionTimerHandler, (void*)NULL);
}
}
char const* actionString
= createReceivers? "Receiving streamed data":"Data is being streamed";
if (timerIsBeingUsed) {
*env << actionString
<< " (for up to " << secondsToDelay
<< " seconds)...\n";
} else {
#ifdef USE_SIGNALS
pid_t ourPid = getpid();
*env << actionString
<< " (signal with \"kill -HUP " << (int)ourPid
<< "\" or \"kill -USR1 " << (int)ourPid
<< "\" to terminate)...\n";
#else
*env << actionString << "...\n";
#endif
}
// Watch for incoming packets (if desired):
checkForPacketArrival(NULL);
checkInterPacketGaps(NULL);
}
void tearDownStreams() {
if (session == NULL) return;
clientTearDownSession(ourClient, session);
}
void closeMediaSinks() {
Medium::close(qtOut);
Medium::close(aviOut);
if (session == NULL) return;
MediaSubsessionIterator iter(*session);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
Medium::close(subsession->sink);
subsession->sink = NULL;
}
}
void subsessionAfterPlaying(void* clientData) {
// Begin by closing this media subsession's stream:
MediaSubsession* subsession = (MediaSubsession*)clientData;
Medium::close(subsession->sink);
subsession->sink = NULL;
// Next, check whether *all* subsessions' streams have now been closed:
MediaSession& session = subsession->parentSession();
MediaSubsessionIterator iter(session);
while ((subsession = iter.next()) != NULL) {
if (subsession->sink != NULL) return; // this subsession is still active
}
// All subsessions' streams have now been closed
sessionAfterPlaying();
}
void subsessionByeHandler(void* clientData) {
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
unsigned secsDiff = timeNow.tv_sec - startTime.tv_sec;
MediaSubsession* subsession = (MediaSubsession*)clientData;
*env << "Received RTCP \"BYE\" on \"" << subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession (after " << secsDiff
<< " seconds)\n";
// Act now as if the subsession had closed:
subsessionAfterPlaying(subsession);
}
void sessionAfterPlaying(void* /*clientData*/) {
if (!playContinuously) {
shutdown(0);
} else {
// We've been asked to play the stream(s) over again:
startPlayingStreams();
}
}
void sessionTimerHandler(void* /*clientData*/) {
sessionTimerTask = NULL;
sessionAfterPlaying();
}
class qosMeasurementRecord {
public:
qosMeasurementRecord(struct timeval const& startTime, RTPSource* src)
: fSource(src), fNext(NULL),
kbits_per_second_min(1e20), kbits_per_second_max(0),
kBytesTotal(0.0),
packet_loss_fraction_min(1.0), packet_loss_fraction_max(0.0),
totNumPacketsReceived(0), totNumPacketsExpected(0) {
measurementEndTime = measurementStartTime = startTime;
#ifdef SUPPORT_REAL_RTSP
if (session->isRealNetworksRDT) { // hack for RealMedia sessions (RDT, not RTP)
RealRDTSource* rdt = (RealRDTSource*)src;
kBytesTotal = rdt->totNumKBytesReceived();
totNumPacketsReceived = rdt->totNumPacketsReceived();
totNumPacketsExpected = totNumPacketsReceived; // because we use TCP
return;
}
#endif
RTPReceptionStatsDB::Iterator statsIter(src->receptionStatsDB());
// Assume that there's only one SSRC source (usually the case):
RTPReceptionStats* stats = statsIter.next(True);
if (stats != NULL) {
kBytesTotal = stats->totNumKBytesReceived();
totNumPacketsReceived = stats->totNumPacketsReceived();
totNumPacketsExpected = stats->totNumPacketsExpected();
}
}
virtual ~qosMeasurementRecord() { delete fNext; }
void periodicQOSMeasurement(struct timeval const& timeNow);
public:
RTPSource* fSource;
qosMeasurementRecord* fNext;
public:
struct timeval measurementStartTime, measurementEndTime;
double kbits_per_second_min, kbits_per_second_max;
double kBytesTotal;
double packet_loss_fraction_min, packet_loss_fraction_max;
unsigned totNumPacketsReceived, totNumPacketsExpected;
};
static qosMeasurementRecord* qosRecordHead = NULL;
static void periodicQOSMeasurement(void* clientData); // forward
static unsigned nextQOSMeasurementUSecs;
static void scheduleNextQOSMeasurement() {
nextQOSMeasurementUSecs += qosMeasurementIntervalMS*1000;