From 1e58e1e6a5a7d08bdcf3c487bc900f75558e9eee Mon Sep 17 00:00:00 2001
From: houxiao <houxiao@454eff88-639b-444f-9e54-f578c98de674>
Date: 星期四, 22 十二月 2016 18:22:22 +0800
Subject: [PATCH] add rtsp server
---
RtspFace/PL_RTSPServer.cpp | 81 ++++++++
RtspFace/testOnDemandRTSPServer.cpp | 455 +++++++++++++++++++++++++++++++++++++++++++++
RtspFace/PL_RTSPServer.h | 24 ++
RtspFace/PL_RTSPClient.cpp | 2
RtspFace/testRTSPClient.hpp | 0
5 files changed, 561 insertions(+), 1 deletions(-)
diff --git a/RtspFace/PL_RTSPClient.cpp b/RtspFace/PL_RTSPClient.cpp
index 30c18fd..1ed98ee 100644
--- a/RtspFace/PL_RTSPClient.cpp
+++ b/RtspFace/PL_RTSPClient.cpp
@@ -5,7 +5,7 @@
void rtsp_client_fmtp_callback(void* arg, const char* val);
void rtsp_client_frame_callback(void* arg, uint8_t* buffer, size_t buffSize);
void rtsp_client_continue_callback(void* arg);
-#include "RTSPClient.hpp"
+#include "testRTSPClient.hpp"
struct RTSPClient_Internal
{
diff --git a/RtspFace/PL_RTSPServer.cpp b/RtspFace/PL_RTSPServer.cpp
new file mode 100644
index 0000000..be0807b
--- /dev/null
+++ b/RtspFace/PL_RTSPServer.cpp
@@ -0,0 +1,81 @@
+#include "PL_RTSPServer.h"
+
+struct PL_RTSPServer_Internal
+{
+ uint8_t buffer[1920*1080*4];
+ size_t buffSize;
+ size_t buffSizeMax;
+
+ bool payError;
+
+ PL_RTSPServer_Internal() :
+ buffSize(0), buffSizeMax(sizeof(buffer)),
+ payError(true)
+ {
+ }
+
+ ~PL_RTSPServer_Internal()
+ {
+ }
+
+ void reset()
+ {
+ buffSize = 0;
+ payError = true;
+ }
+};
+
+PipeLineElem* create_PL_RTSPServer()
+{
+ return new PL_RTSPServer;
+}
+
+PL_RTSPServer::PL_RTSPServer() : internal(new PL_RTSPServer_Internal)
+{
+}
+
+PL_RTSPServer::~PL_RTSPServer()
+{
+ delete (PL_RTSPServer_Internal*)internal;
+ internal= nullptr;
+}
+
+bool PL_RTSPServer::init(void* args)
+{
+ PL_RTSPServer_Internal* in = (PL_RTSPServer_Internal*)internal;
+ in->reset();
+
+ return true;
+}
+
+void PL_RTSPServer::finit()
+{
+ PL_RTSPServer_Internal* in = (PL_RTSPServer_Internal*)internal;
+
+}
+
+bool PL_RTSPServer::pay(const PipeMaterial& pm)
+{
+ PL_RTSPServer_Internal* in = (PL_RTSPServer_Internal*)internal;
+
+ //in->buffer readly
+
+ //static size_t f=0;
+ //char fname[50];
+ //sprintf(fname, "%u.bgra", ++f);
+ //FILE * pFile = fopen (fname,"wb");
+ //fwrite (in->buffer , sizeof(char), in->buffSize, pFile);
+ //fclose(pFile);
+
+ return true;
+}
+
+bool PL_RTSPServer::gain(PipeMaterial& pm)
+{
+ PL_RTSPServer_Internal* in = (PL_RTSPServer_Internal*)internal;
+
+ pm.buffer = in->buffer;
+ pm.buffSize = in->buffSize;
+ pm.former = this;
+ return true;
+}
diff --git a/RtspFace/PL_RTSPServer.h b/RtspFace/PL_RTSPServer.h
new file mode 100644
index 0000000..8b0afec
--- /dev/null
+++ b/RtspFace/PL_RTSPServer.h
@@ -0,0 +1,24 @@
+#ifndef _PL_RTSPSERVER_H_
+#define _PL_PL_RTSPSERVER_H_
+
+#include "PipeLine.h"
+
+class PL_RTSPServer : public PipeLineElem
+{
+public:
+ PL_RTSPServer();
+ virtual ~PL_RTSPServer();
+
+ virtual bool init(void* args);
+ virtual void finit();
+
+ virtual bool pay(const PipeMaterial& pm);
+ virtual bool gain(PipeMaterial& pm);
+
+private:
+ void* internal;
+};
+
+PipeLineElem* create_PL_RTSPServer();
+
+#endif
diff --git a/RtspFace/testOnDemandRTSPServer.cpp b/RtspFace/testOnDemandRTSPServer.cpp
new file mode 100644
index 0000000..2608308
--- /dev/null
+++ b/RtspFace/testOnDemandRTSPServer.cpp
@@ -0,0 +1,455 @@
+/**********
+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 3 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) 1996-2017, Live Networks, Inc. All rights reserved
+// A test program that demonstrates how to stream - via unicast RTP
+// - various kinds of file on demand, using a built-in RTSP server.
+// main program
+
+#include "liveMedia.hh"
+#include "BasicUsageEnvironment.hh"
+
+UsageEnvironment* env;
+
+// To make the second and subsequent client for each stream reuse the same
+// input stream as the first client (rather than playing the file from the
+// start for each client), change the following "False" to "True":
+Boolean reuseFirstSource = False;
+
+// To stream *only* MPEG-1 or 2 video "I" frames
+// (e.g., to reduce network bandwidth),
+// change the following "False" to "True":
+Boolean iFramesOnly = False;
+
+static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
+ char const* streamName, char const* inputFileName); // fwd
+
+static char newDemuxWatchVariable;
+
+static MatroskaFileServerDemux* matroskaDemux;
+static void onMatroskaDemuxCreation(MatroskaFileServerDemux* newDemux, void* /*clientData*/) {
+ matroskaDemux = newDemux;
+ newDemuxWatchVariable = 1;
+}
+
+static OggFileServerDemux* oggDemux;
+static void onOggDemuxCreation(OggFileServerDemux* newDemux, void* /*clientData*/) {
+ oggDemux = newDemux;
+ newDemuxWatchVariable = 1;
+}
+
+int main(int argc, char** argv) {
+ // Begin by setting up our usage environment:
+ TaskScheduler* scheduler = BasicTaskScheduler::createNew();
+ env = BasicUsageEnvironment::createNew(*scheduler);
+
+ UserAuthenticationDatabase* authDB = NULL;
+#ifdef ACCESS_CONTROL
+ // To implement client access control to the RTSP server, do the following:
+ authDB = new UserAuthenticationDatabase;
+ authDB->addUserRecord("username1", "password1"); // replace these with real strings
+ // Repeat the above with each <username>, <password> that you wish to allow
+ // access to the server.
+#endif
+
+ // Create the RTSP server:
+ RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
+ if (rtspServer == NULL) {
+ *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
+ exit(1);
+ }
+
+ char const* descriptionString
+ = "Session streamed by \"testOnDemandRTSPServer\"";
+
+ // Set up each of the possible streams that can be served by the
+ // RTSP server. Each such stream is implemented using a
+ // "ServerMediaSession" object, plus one or more
+ // "ServerMediaSubsession" objects for each audio/video substream.
+
+ // A MPEG-4 video elementary stream:
+ {
+ char const* streamName = "mpeg4ESVideoTest";
+ char const* inputFileName = "test.m4e";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(MPEG4VideoFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A H.264 video elementary stream:
+ {
+ char const* streamName = "h264ESVideoTest";
+ char const* inputFileName = "test.264";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(H264VideoFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A H.265 video elementary stream:
+ {
+ char const* streamName = "h265ESVideoTest";
+ char const* inputFileName = "test.265";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(H265VideoFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A MPEG-1 or 2 audio+video program stream:
+ {
+ char const* streamName = "mpeg1or2AudioVideoTest";
+ char const* inputFileName = "test.mpg";
+ // NOTE: This *must* be a Program Stream; not an Elementary Stream
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ MPEG1or2FileServerDemux* demux
+ = MPEG1or2FileServerDemux::createNew(*env, inputFileName, reuseFirstSource);
+ sms->addSubsession(demux->newVideoServerMediaSubsession(iFramesOnly));
+ sms->addSubsession(demux->newAudioServerMediaSubsession());
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A MPEG-1 or 2 video elementary stream:
+ {
+ char const* streamName = "mpeg1or2ESVideoTest";
+ char const* inputFileName = "testv.mpg";
+ // NOTE: This *must* be a Video Elementary Stream; not a Program Stream
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(MPEG1or2VideoFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource, iFramesOnly));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A MP3 audio stream (actually, any MPEG-1 or 2 audio file will work):
+ // To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
+//#define STREAM_USING_ADUS 1
+ // To also reorder ADUs before streaming, uncomment the following:
+//#define INTERLEAVE_ADUS 1
+ // (For more information about ADUs and interleaving,
+ // see <http://www.live555.com/rtp-mp3/>)
+ {
+ char const* streamName = "mp3AudioTest";
+ char const* inputFileName = "test.mp3";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ Boolean useADUs = False;
+ Interleaving* interleaving = NULL;
+#ifdef STREAM_USING_ADUS
+ useADUs = True;
+#ifdef INTERLEAVE_ADUS
+ unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
+ unsigned const interleaveCycleSize
+ = (sizeof interleaveCycle)/(sizeof (unsigned char));
+ interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
+#endif
+#endif
+ sms->addSubsession(MP3AudioFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource,
+ useADUs, interleaving));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A WAV audio stream:
+ {
+ char const* streamName = "wavAudioTest";
+ char const* inputFileName = "test.wav";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ // To convert 16-bit PCM data to 8-bit u-law, prior to streaming,
+ // change the following to True:
+ Boolean convertToULaw = False;
+ sms->addSubsession(WAVAudioFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource, convertToULaw));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // An AMR audio stream:
+ {
+ char const* streamName = "amrAudioTest";
+ char const* inputFileName = "test.amr";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(AMRAudioFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A 'VOB' file (e.g., from an unencrypted DVD):
+ {
+ char const* streamName = "vobTest";
+ char const* inputFileName = "test.vob";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ // Note: VOB files are MPEG-2 Program Stream files, but using AC-3 audio
+ MPEG1or2FileServerDemux* demux
+ = MPEG1or2FileServerDemux::createNew(*env, inputFileName, reuseFirstSource);
+ sms->addSubsession(demux->newVideoServerMediaSubsession(iFramesOnly));
+ sms->addSubsession(demux->newAC3AudioServerMediaSubsession());
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A MPEG-2 Transport Stream:
+ {
+ char const* streamName = "mpeg2TransportStreamTest";
+ char const* inputFileName = "test.ts";
+ char const* indexFileName = "test.tsx";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(MPEG2TransportFileServerMediaSubsession
+ ::createNew(*env, inputFileName, indexFileName, reuseFirstSource));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // An AAC audio stream (ADTS-format file):
+ {
+ char const* streamName = "aacAudioTest";
+ char const* inputFileName = "test.aac";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(ADTSAudioFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A DV video stream:
+ {
+ // First, make sure that the RTPSinks' buffers will be large enough to handle the huge size of DV frames (as big as 288000).
+ OutPacketBuffer::maxSize = 300000;
+
+ char const* streamName = "dvVideoTest";
+ char const* inputFileName = "test.dv";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(DVVideoFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource));
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A AC3 video elementary stream:
+ {
+ char const* streamName = "ac3AudioTest";
+ char const* inputFileName = "test.ac3";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+
+ sms->addSubsession(AC3AudioFileServerMediaSubsession
+ ::createNew(*env, inputFileName, reuseFirstSource));
+
+ rtspServer->addServerMediaSession(sms);
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A Matroska ('.mkv') file, with video+audio+subtitle streams:
+ {
+ char const* streamName = "matroskaFileTest";
+ char const* inputFileName = "test.mkv";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+
+ newDemuxWatchVariable = 0;
+ MatroskaFileServerDemux::createNew(*env, inputFileName, onMatroskaDemuxCreation, NULL);
+ env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
+
+ Boolean sessionHasTracks = False;
+ ServerMediaSubsession* smss;
+ while ((smss = matroskaDemux->newServerMediaSubsession()) != NULL) {
+ sms->addSubsession(smss);
+ sessionHasTracks = True;
+ }
+ if (sessionHasTracks) {
+ rtspServer->addServerMediaSession(sms);
+ }
+ // otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A WebM ('.webm') file, with video(VP8)+audio(Vorbis) streams:
+ // (Note: ".webm' files are special types of Matroska files, so we use the same code as the Matroska ('.mkv') file code above.)
+ {
+ char const* streamName = "webmFileTest";
+ char const* inputFileName = "test.webm";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+
+ newDemuxWatchVariable = 0;
+ MatroskaFileServerDemux::createNew(*env, inputFileName, onMatroskaDemuxCreation, NULL);
+ env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
+
+ Boolean sessionHasTracks = False;
+ ServerMediaSubsession* smss;
+ while ((smss = matroskaDemux->newServerMediaSubsession()) != NULL) {
+ sms->addSubsession(smss);
+ sessionHasTracks = True;
+ }
+ if (sessionHasTracks) {
+ rtspServer->addServerMediaSession(sms);
+ }
+ // otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // An Ogg ('.ogg') file, with video and/or audio streams:
+ {
+ char const* streamName = "oggFileTest";
+ char const* inputFileName = "test.ogg";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+
+ newDemuxWatchVariable = 0;
+ OggFileServerDemux::createNew(*env, inputFileName, onOggDemuxCreation, NULL);
+ env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
+
+ Boolean sessionHasTracks = False;
+ ServerMediaSubsession* smss;
+ while ((smss = oggDemux->newServerMediaSubsession()) != NULL) {
+ sms->addSubsession(smss);
+ sessionHasTracks = True;
+ }
+ if (sessionHasTracks) {
+ rtspServer->addServerMediaSession(sms);
+ }
+ // otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // An Opus ('.opus') audio file:
+ // (Note: ".opus' files are special types of Ogg files, so we use the same code as the Ogg ('.ogg') file code above.)
+ {
+ char const* streamName = "opusFileTest";
+ char const* inputFileName = "test.opus";
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+
+ newDemuxWatchVariable = 0;
+ OggFileServerDemux::createNew(*env, inputFileName, onOggDemuxCreation, NULL);
+ env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
+
+ Boolean sessionHasTracks = False;
+ ServerMediaSubsession* smss;
+ while ((smss = oggDemux->newServerMediaSubsession()) != NULL) {
+ sms->addSubsession(smss);
+ sessionHasTracks = True;
+ }
+ if (sessionHasTracks) {
+ rtspServer->addServerMediaSession(sms);
+ }
+ // otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
+
+ announceStream(rtspServer, sms, streamName, inputFileName);
+ }
+
+ // A MPEG-2 Transport Stream, coming from a live UDP (raw-UDP or RTP/UDP) source:
+ {
+ char const* streamName = "mpeg2TransportStreamFromUDPSourceTest";
+ char const* inputAddressStr = "239.255.42.42";
+ // This causes the server to take its input from the stream sent by the "testMPEG2TransportStreamer" demo application.
+ // (Note: If the input UDP source is unicast rather than multicast, then change this to NULL.)
+ portNumBits const inputPortNum = 1234;
+ // This causes the server to take its input from the stream sent by the "testMPEG2TransportStreamer" demo application.
+ Boolean const inputStreamIsRawUDP = False;
+ ServerMediaSession* sms
+ = ServerMediaSession::createNew(*env, streamName, streamName,
+ descriptionString);
+ sms->addSubsession(MPEG2TransportUDPServerMediaSubsession
+ ::createNew(*env, inputAddressStr, inputPortNum, inputStreamIsRawUDP));
+ rtspServer->addServerMediaSession(sms);
+
+ char* url = rtspServer->rtspURL(sms);
+ *env << "\n\"" << streamName << "\" stream, from a UDP Transport Stream input source \n\t(";
+ if (inputAddressStr != NULL) {
+ *env << "IP multicast address " << inputAddressStr << ",";
+ } else {
+ *env << "unicast;";
+ }
+ *env << " port " << inputPortNum << ")\n";
+ *env << "Play this stream using the URL \"" << url << "\"\n";
+ delete[] url;
+ }
+
+ // Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling.
+ // Try first with the default HTTP port (80), and then with the alternative HTTP
+ // port numbers (8000 and 8080).
+
+ if (rtspServer->setUpTunnelingOverHTTP(80) || rtspServer->setUpTunnelingOverHTTP(8000) || rtspServer->setUpTunnelingOverHTTP(8080)) {
+ *env << "\n(We use port " << rtspServer->httpServerPortNum() << " for optional RTSP-over-HTTP tunneling.)\n";
+ } else {
+ *env << "\n(RTSP-over-HTTP tunneling is not available.)\n";
+ }
+
+ env->taskScheduler().doEventLoop(); // does not return
+
+ return 0; // only to prevent compiler warning
+}
+
+static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
+ char const* streamName, char const* inputFileName) {
+ char* url = rtspServer->rtspURL(sms);
+ UsageEnvironment& env = rtspServer->envir();
+ env << "\n\"" << streamName << "\" stream, from the file \""
+ << inputFileName << "\"\n";
+ env << "Play this stream using the URL \"" << url << "\"\n";
+ delete[] url;
+}
diff --git a/RtspFace/RTSPClient.hpp b/RtspFace/testRTSPClient.hpp
similarity index 100%
rename from RtspFace/RTSPClient.hpp
rename to RtspFace/testRTSPClient.hpp
--
Gitblit v1.8.0