/**********
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This library is free software; you can redistribute it and/or modify it under
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the terms of the GNU Lesser General Public License as published by the
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Free Software Foundation; either version 3 of the License, or (at your
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option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
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This library is distributed in the hope that it will be useful, but WITHOUT
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ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
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FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
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more details.
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You should have received a copy of the GNU Lesser General Public License
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along with this library; if not, write to the Free Software Foundation, Inc.,
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51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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**********/
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// Copyright (c) 1996-2017, Live Networks, Inc. All rights reserved
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// A demo application, showing how to create and run a RTSP client (that can potentially receive multiple streams concurrently).
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//
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// NOTE: This code - although it builds a running application - is intended only to illustrate how to develop your own RTSP
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// client application. For a full-featured RTSP client application - with much more functionality, and many options - see
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// "openRTSP": http://www.live555.com/openRTSP/
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#include <liveMedia.hh>
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#include <BasicUsageEnvironment.hh>
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#include <iostream>
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// Forward function definitions:
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// RTSP 'response handlers':
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void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString);
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void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString);
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void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString);
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// Other event handler functions:
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void subsessionAfterPlaying(void* clientData); // called when a stream's subsession (e.g., audio or video substream) ends
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void subsessionByeHandler(void* clientData); // called when a RTCP "BYE" is received for a subsession
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void streamTimerHandler(void* clientData);
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// called at the end of a stream's expected duration (if the stream has not already signaled its end using a RTCP "BYE")
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// The main streaming routine (for each "rtsp://" URL):
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void openURL(UsageEnvironment& env, void* args, char const* progName, char const* rtspURL);
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// Used to iterate through each stream's 'subsessions', setting up each one:
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void setupNextSubsession(RTSPClient* rtspClient);
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// Used to shut down and close a stream (including its "RTSPClient" object):
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void shutdownStream(RTSPClient* rtspClient, int exitCode = 1);
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// A function that outputs a string that identifies each stream (for debugging output). Modify this if you wish:
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UsageEnvironment& operator<<(UsageEnvironment& env, const RTSPClient& rtspClient) {
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return env << "[URL:\"" << rtspClient.url() << "\"]: ";
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}
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// A function that outputs a string that identifies each subsession (for debugging output). Modify this if you wish:
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UsageEnvironment& operator<<(UsageEnvironment& env, const MediaSubsession& subsession) {
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return env << subsession.mediumName() << "/" << subsession.codecName();
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}
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void usage(UsageEnvironment& env, char const* progName) {
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env << "Usage: " << progName << " <rtsp-url-1> ... <rtsp-url-N>\n";
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env << "\t(where each <rtsp-url-i> is a \"rtsp://\" URL)\n";
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}
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char eventLoopWatchVariable = 0;
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int test_main(int argc, char** argv) {
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// Begin by setting up our usage environment:
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TaskScheduler* scheduler = BasicTaskScheduler::createNew();
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UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
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// We need at least one "rtsp://" URL argument:
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if (argc < 2) {
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usage(*env, argv[0]);
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return 1;
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}
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// There are argc-1 URLs: argv[1] through argv[argc-1]. Open and start streaming each one:
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for (int i = 1; i <= argc-1; ++i) {
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openURL(*env, NULL, argv[0], argv[i]);
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}
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// All subsequent activity takes place within the event loop:
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env->taskScheduler().doEventLoop(&eventLoopWatchVariable);
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// This function call does not return, unless, at some point in time, "eventLoopWatchVariable" gets set to something non-zero.
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return 0;
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// If you choose to continue the application past this point (i.e., if you comment out the "return 0;" statement above),
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// and if you don't intend to do anything more with the "TaskScheduler" and "UsageEnvironment" objects,
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// then you can also reclaim the (small) memory used by these objects by uncommenting the following code:
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/*
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env->reclaim(); env = NULL;
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delete scheduler; scheduler = NULL;
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*/
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}
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// Define a class to hold per-stream state that we maintain throughout each stream's lifetime:
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class StreamClientState {
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public:
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StreamClientState();
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virtual ~StreamClientState();
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public:
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MediaSubsessionIterator* iter;
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MediaSession* session;
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MediaSubsession* subsession;
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TaskToken streamTimerTask;
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double duration;
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};
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// If you're streaming just a single stream (i.e., just from a single URL, once), then you can define and use just a single
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// "StreamClientState" structure, as a global variable in your application. However, because - in this demo application - we're
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// showing how to play multiple streams, concurrently, we can't do that. Instead, we have to have a separate "StreamClientState"
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// structure for each "RTSPClient". To do this, we subclass "RTSPClient", and add a "StreamClientState" field to the subclass:
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class ourRTSPClient: public RTSPClient {
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public:
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static ourRTSPClient* createNew(UsageEnvironment& env, char const* rtspURL,
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int verbosityLevel = 0,
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char const* applicationName = NULL,
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portNumBits tunnelOverHTTPPortNum = 0);
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protected:
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ourRTSPClient(UsageEnvironment& env, char const* rtspURL,
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int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum);
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// called only by createNew();
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virtual ~ourRTSPClient();
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public:
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StreamClientState scs;
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void* args;
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};
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// Define a data sink (a subclass of "MediaSink") to receive the data for each subsession (i.e., each audio or video 'substream').
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// In practice, this might be a class (or a chain of classes) that decodes and then renders the incoming audio or video.
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// Or it might be a "FileSink", for outputting the received data into a file (as is done by the "openRTSP" application).
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// In this example code, however, we define a simple 'dummy' sink that receives incoming data, but does nothing with it.
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class DummySink: public MediaSink
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{
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public:
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static DummySink* createNew(UsageEnvironment& env,
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void* _args,
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MediaSubsession& subsession, // identifies the kind of data that's being received
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char const* streamId = NULL); // identifies the stream itself (optional)
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private:
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DummySink(UsageEnvironment& env, void* _args, MediaSubsession& subsession, char const* streamId);
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// called only by "createNew()"
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virtual ~DummySink();
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static void afterGettingFrame(void* clientData, unsigned frameSize,
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unsigned numTruncatedBytes,
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struct timeval presentationTime,
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unsigned durationInMicroseconds);
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void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
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struct timeval presentationTime, unsigned durationInMicroseconds);
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public:
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void* args;
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private:
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// redefined virtual functions:
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virtual Boolean continuePlaying();
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private:
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u_int8_t* fReceiveBuffer;
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MediaSubsession& fSubsession;
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char* fStreamId;
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};
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#define RTSP_CLIENT_VERBOSITY_LEVEL 1 // by default, print verbose output from each "RTSPClient"
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static unsigned rtspClientCount = 0; // Counts how many streams (i.e., "RTSPClient"s) are currently in use.
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void openURL(UsageEnvironment& env, void* args, char const* progName, char const* rtspURL) {
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// Begin by creating a "RTSPClient" object. Note that there is a separate "RTSPClient" object for each stream that we wish
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// to receive (even if more than stream uses the same "rtsp://" URL).
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RTSPClient* rtspClient = ourRTSPClient::createNew(env, rtspURL, RTSP_CLIENT_VERBOSITY_LEVEL, progName);
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if (rtspClient == NULL) {
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env << "Failed to create a RTSP client for URL \"" << rtspURL << "\": " << env.getResultMsg() << "\n";
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return;
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}
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((ourRTSPClient*)rtspClient)->args = args;
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++rtspClientCount;
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// Next, send a RTSP "DESCRIBE" command, to get a SDP description for the stream.
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// Note that this command - like all RTSP commands - is sent asynchronously; we do not block, waiting for a response.
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// Instead, the following function call returns immediately, and we handle the RTSP response later, from within the event loop:
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rtspClient->sendDescribeCommand(continueAfterDESCRIBE);
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}
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// Implementation of the RTSP 'response handlers':
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void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString) {
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do {
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UsageEnvironment& env = rtspClient->envir(); // alias
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StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
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if (resultCode != 0) {
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env << *rtspClient << "Failed to get a SDP description: " << resultString << "\n";
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delete[] resultString;
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break;
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}
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char* const sdpDescription = resultString;
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env << *rtspClient << "Got a SDP description:\n" << sdpDescription << "\n";
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// Create a media session object from this SDP description:
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scs.session = MediaSession::createNew(env, sdpDescription);
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delete[] sdpDescription; // because we don't need it anymore
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if (scs.session == NULL) {
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env << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg() << "\n";
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break;
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} else if (!scs.session->hasSubsessions()) {
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env << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)\n";
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break;
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}
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// Then, create and set up our data source objects for the session. We do this by iterating over the session's 'subsessions',
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// calling "MediaSubsession::initiate()", and then sending a RTSP "SETUP" command, on each one.
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// (Each 'subsession' will have its own data source.)
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scs.iter = new MediaSubsessionIterator(*scs.session);
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setupNextSubsession(rtspClient);
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return;
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} while (0);
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// An unrecoverable error occurred with this stream.
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shutdownStream(rtspClient);
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}
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// By default, we request that the server stream its data using RTP/UDP.
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// If, instead, you want to request that the server stream via RTP-over-TCP, change the following to True:
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#define REQUEST_STREAMING_OVER_TCP False
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void setupNextSubsession(RTSPClient* rtspClient) {
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UsageEnvironment& env = rtspClient->envir(); // alias
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StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
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scs.subsession = scs.iter->next();
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if (scs.subsession != NULL) {
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if (!scs.subsession->initiate()) {
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env << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << "\n";
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setupNextSubsession(rtspClient); // give up on this subsession; go to the next one
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} else {
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env << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (";
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if (scs.subsession->rtcpIsMuxed()) {
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env << "client port " << scs.subsession->clientPortNum();
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} else {
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env << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
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}
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env << ")\n";
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// Continue setting up this subsession, by sending a RTSP "SETUP" command:
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rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP);
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}
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return;
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}
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// We've finished setting up all of the subsessions. Now, send a RTSP "PLAY" command to start the streaming:
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if (scs.session->absStartTime() != NULL) {
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// Special case: The stream is indexed by 'absolute' time, so send an appropriate "PLAY" command:
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rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY, scs.session->absStartTime(), scs.session->absEndTime());
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} else {
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scs.duration = scs.session->playEndTime() - scs.session->playStartTime();
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rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY);
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}
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}
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void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString) {
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do {
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UsageEnvironment& env = rtspClient->envir(); // alias
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StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
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if (resultCode != 0) {
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env << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << "\n";
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break;
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}
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env << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (";
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if (scs.subsession->rtcpIsMuxed()) {
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env << "client port " << scs.subsession->clientPortNum();
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} else {
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env << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
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}
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env << ")\n";
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// Having successfully setup the subsession, create a data sink for it, and call "startPlaying()" on it.
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// (This will prepare the data sink to receive data; the actual flow of data from the client won't start happening until later,
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// after we've sent a RTSP "PLAY" command.)
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DummySink* mySink;
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scs.subsession->sink = mySink = DummySink::createNew(env, ((ourRTSPClient*)rtspClient)->args,
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*scs.subsession, rtspClient->url());
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// perhaps use your own custom "MediaSink" subclass instead
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if (scs.subsession->sink == NULL) {
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env << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession
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<< "\" subsession: " << env.getResultMsg() << "\n";
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break;
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}
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env << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession\n";
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scs.subsession->miscPtr = rtspClient; // a hack to let subsession handler functions get the "RTSPClient" from the subsession
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scs.subsession->sink->startPlaying(*(scs.subsession->readSource()),
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subsessionAfterPlaying, scs.subsession);
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// Also set a handler to be called if a RTCP "BYE" arrives for this subsession:
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if (scs.subsession->rtcpInstance() != NULL) {
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scs.subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, scs.subsession);
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}
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} while (0);
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delete[] resultString;
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// Set up the next subsession, if any:
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setupNextSubsession(rtspClient);
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}
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void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString) {
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Boolean success = False;
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do {
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UsageEnvironment& env = rtspClient->envir(); // alias
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StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
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if (resultCode != 0) {
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env << *rtspClient << "Failed to start playing session: " << resultString << "\n";
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break;
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}
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// Set a timer to be handled at the end of the stream's expected duration (if the stream does not already signal its end
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// using a RTCP "BYE"). This is optional. If, instead, you want to keep the stream active - e.g., so you can later
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// 'seek' back within it and do another RTSP "PLAY" - then you can omit this code.
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// (Alternatively, if you don't want to receive the entire stream, you could set this timer for some shorter value.)
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if (scs.duration > 0) {
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unsigned const delaySlop = 2; // number of seconds extra to delay, after the stream's expected duration. (This is optional.)
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scs.duration += delaySlop;
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unsigned uSecsToDelay = (unsigned)(scs.duration*1000000);
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scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtspClient);
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}
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env << *rtspClient << "Started playing session";
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if (scs.duration > 0) {
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env << " (for up to " << scs.duration << " seconds)";
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}
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env << "...\n";
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success = True;
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} while (0);
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delete[] resultString;
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if (!success) {
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// An unrecoverable error occurred with this stream.
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shutdownStream(rtspClient);
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}
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}
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// Implementation of the other event handlers:
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void subsessionAfterPlaying(void* clientData) {
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MediaSubsession* subsession = (MediaSubsession*)clientData;
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RTSPClient* rtspClient = (RTSPClient*)(subsession->miscPtr);
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// Begin by closing this subsession's stream:
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Medium::close(subsession->sink);
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subsession->sink = NULL;
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// Next, check whether *all* subsessions' streams have now been closed:
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MediaSession& session = subsession->parentSession();
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MediaSubsessionIterator iter(session);
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while ((subsession = iter.next()) != NULL) {
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if (subsession->sink != NULL) return; // this subsession is still active
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}
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// All subsessions' streams have now been closed, so shutdown the client:
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shutdownStream(rtspClient);
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}
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void subsessionByeHandler(void* clientData) {
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MediaSubsession* subsession = (MediaSubsession*)clientData;
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RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr;
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UsageEnvironment& env = rtspClient->envir(); // alias
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env << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession\n";
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// Now act as if the subsession had closed:
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subsessionAfterPlaying(subsession);
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}
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void streamTimerHandler(void* clientData) {
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ourRTSPClient* rtspClient = (ourRTSPClient*)clientData;
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StreamClientState& scs = rtspClient->scs; // alias
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scs.streamTimerTask = NULL;
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// Shut down the stream:
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shutdownStream(rtspClient);
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}
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void shutdownStream(RTSPClient* rtspClient, int exitCode) {
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UsageEnvironment& env = rtspClient->envir(); // alias
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StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
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// First, check whether any subsessions have still to be closed:
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if (scs.session != NULL) {
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Boolean someSubsessionsWereActive = False;
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MediaSubsessionIterator iter(*scs.session);
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MediaSubsession* subsession;
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while ((subsession = iter.next()) != NULL) {
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if (subsession->sink != NULL) {
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Medium::close(subsession->sink);
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subsession->sink = NULL;
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if (subsession->rtcpInstance() != NULL) {
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subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
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}
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someSubsessionsWereActive = True;
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}
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}
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if (someSubsessionsWereActive) {
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// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
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// Don't bother handling the response to the "TEARDOWN".
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rtspClient->sendTeardownCommand(*scs.session, NULL);
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}
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}
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env << *rtspClient << "Closing the stream.\n";
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Medium::close(rtspClient);
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// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
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if (--rtspClientCount == 0) {
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// The final stream has ended, so exit the application now.
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// (Of course, if you're embedding this code into your own application, you might want to comment this out,
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// and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
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exit(exitCode);
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}
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}
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|
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// Implementation of "ourRTSPClient":
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ourRTSPClient* ourRTSPClient::createNew(UsageEnvironment& env, char const* rtspURL,
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int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum) {
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return new ourRTSPClient(env, rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum);
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}
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ourRTSPClient::ourRTSPClient(UsageEnvironment& env, char const* rtspURL,
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int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum)
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: RTSPClient(env,rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum, -1),
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args(nullptr)
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{
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}
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ourRTSPClient::~ourRTSPClient() {
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}
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|
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// Implementation of "StreamClientState":
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StreamClientState::StreamClientState()
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: iter(NULL), session(NULL), subsession(NULL), streamTimerTask(NULL), duration(0.0) {
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}
|
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StreamClientState::~StreamClientState() {
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delete iter;
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if (session != NULL) {
|
// We also need to delete "session", and unschedule "streamTimerTask" (if set)
|
UsageEnvironment& env = session->envir(); // alias
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|
env.taskScheduler().unscheduleDelayedTask(streamTimerTask);
|
Medium::close(session);
|
}
|
}
|
|
// Implementation of "DummySink":
|
|
// Even though we're not going to be doing anything with the incoming data, we still need to receive it.
|
// Define the size of the buffer that we'll use:
|
#define DUMMY_SINK_RECEIVE_BUFFER_SIZE 1920*1080*3
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|
DummySink* DummySink::createNew(UsageEnvironment& env, void* _args, MediaSubsession& subsession, char const* streamId)
|
{
|
return new DummySink(env, _args, subsession, streamId);
|
}
|
|
DummySink::DummySink(UsageEnvironment& env, void* _args, MediaSubsession& subsession, char const* streamId)
|
: MediaSink(env), args(_args), fSubsession(subsession)
|
{
|
fStreamId = strDup(streamId);
|
fReceiveBuffer = new u_int8_t[DUMMY_SINK_RECEIVE_BUFFER_SIZE];
|
|
// ffmpeg need AUX header
|
fReceiveBuffer[0]=0x00; fReceiveBuffer[1]=0x00; fReceiveBuffer[2]=0x00; fReceiveBuffer[3]=0x01;
|
|
//parse sdp
|
const char* strSDP = fSubsession.savedSDPLines();
|
rtsp_client_sdp_callback(args, strSDP);
|
|
const char* strFmtp = fSubsession.fmtp_spropparametersets();
|
rtsp_client_fmtp_callback(args, strFmtp);
|
//std::cout << strFmtp << std::endl;
|
}
|
|
DummySink::~DummySink() {
|
delete[] fReceiveBuffer;
|
delete[] fStreamId;
|
}
|
|
void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
|
struct timeval presentationTime, unsigned durationInMicroseconds) {
|
DummySink* sink = (DummySink*)clientData;
|
|
if (frameSize > 0)
|
rtsp_client_frame_callback(sink->args, sink->fReceiveBuffer, frameSize + 4);
|
|
sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);
|
}
|
|
// If you don't want to see debugging output for each received frame, then comment out the following line:
|
#define DEBUG_PRINT_EACH_RECEIVED_FRAME 1
|
|
void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
|
struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
|
// We've just received a frame of data. (Optionally) print out information about it:
|
#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME
|
if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";
|
envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
|
if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";
|
char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time
|
sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
|
envir() << ".\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;
|
if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
|
envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
|
}
|
#ifdef DEBUG_PRINT_NPT
|
envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
|
#endif
|
envir() << "\n";
|
#endif
|
|
// Then continue, to request the next frame of data:
|
continuePlaying();
|
}
|
|
Boolean DummySink::continuePlaying() {
|
if (fSource == NULL) return False; // sanity check (should not happen)
|
|
rtsp_client_continue_callback(args);
|
|
// Request the next frame of data from our input source. "afterGettingFrame()" will get called later, when it arrives:
|
fSource->getNextFrame(fReceiveBuffer + 4, DUMMY_SINK_RECEIVE_BUFFER_SIZE,
|
afterGettingFrame, this,
|
onSourceClosure, this);
|
return True;
|
}
|
