Mercurial > mplayer.hg
view libmpdemux/demux_rtp.cpp @ 22480:9837a8799315
Move command.c.tmp over now identical command.c
author | uau |
---|---|
date | Fri, 09 Mar 2007 12:43:26 +0000 |
parents | 979b2aa16e80 |
children | 5f3a0a712afb |
line wrap: on
line source
////////// Routines (with C-linkage) that interface between "MPlayer" ////////// and the "LIVE555 Streaming Media" libraries: extern "C" { // on MinGW, we must include windows.h before the things it conflicts #ifdef __MINGW32__ // with. they are each protected from #include <windows.h> // windows.h, but not the other way around. #endif #include "demux_rtp.h" #include "stheader.h" } #include "demux_rtp_internal.h" #include "BasicUsageEnvironment.hh" #include "liveMedia.hh" #include "GroupsockHelper.hh" #include <unistd.h> // A data structure representing input data for each stream: class ReadBufferQueue { public: ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer, char const* tag); virtual ~ReadBufferQueue(); FramedSource* readSource() const { return fReadSource; } RTPSource* rtpSource() const { return fRTPSource; } demuxer_t* ourDemuxer() const { return fOurDemuxer; } char const* tag() const { return fTag; } char blockingFlag; // used to implement synchronous reads // For A/V synchronization: Boolean prevPacketWasSynchronized; float prevPacketPTS; ReadBufferQueue** otherQueue; // The 'queue' actually consists of just a single "demux_packet_t" // (because the underlying OS does the actual queueing/buffering): demux_packet_t* dp; // However, we sometimes inspect buffers before delivering them. // For this, we maintain a queue of pending buffers: void savePendingBuffer(demux_packet_t* dp); demux_packet_t* getPendingBuffer(); private: demux_packet_t* pendingDPHead; demux_packet_t* pendingDPTail; FramedSource* fReadSource; RTPSource* fRTPSource; demuxer_t* fOurDemuxer; char const* fTag; // used for debugging }; // A structure of RTP-specific state, kept so that we can cleanly // reclaim it: typedef struct RTPState { char const* sdpDescription; RTSPClient* rtspClient; SIPClient* sipClient; MediaSession* mediaSession; ReadBufferQueue* audioBufferQueue; ReadBufferQueue* videoBufferQueue; unsigned flags; struct timeval firstSyncTime; }; extern "C" char* network_username; extern "C" char* network_password; static char* openURL_rtsp(RTSPClient* client, char const* url) { // If we were given a user name (and optional password), then use them: if (network_username != NULL) { char const* password = network_password == NULL ? "" : network_password; return client->describeWithPassword(url, network_username, password); } else { return client->describeURL(url); } } static char* openURL_sip(SIPClient* client, char const* url) { // If we were given a user name (and optional password), then use them: if (network_username != NULL) { char const* password = network_password == NULL ? "" : network_password; return client->inviteWithPassword(url, network_username, password); } else { return client->invite(url); } } int rtspStreamOverTCP = 0; extern int rtsp_port; extern "C" int audio_id, video_id, dvdsub_id; extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { Boolean success = False; do { TaskScheduler* scheduler = BasicTaskScheduler::createNew(); if (scheduler == NULL) break; UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); if (env == NULL) break; RTSPClient* rtspClient = NULL; SIPClient* sipClient = NULL; if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen demuxer->stream->eof = 0; // just in case // Look at the stream's 'priv' field to see if we were initiated // via a SDP description: char* sdpDescription = (char*)(demuxer->stream->priv); if (sdpDescription == NULL) { // We weren't given a SDP description directly, so assume that // we were given a RTSP or SIP URL: char const* protocol = demuxer->stream->streaming_ctrl->url->protocol; char const* url = demuxer->stream->streaming_ctrl->url->url; extern int verbose; if (strcmp(protocol, "rtsp") == 0) { rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer"); if (rtspClient == NULL) { fprintf(stderr, "Failed to create RTSP client: %s\n", env->getResultMsg()); break; } sdpDescription = openURL_rtsp(rtspClient, url); } else { // SIP unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM) sipClient = SIPClient::createNew(*env, desiredAudioType, NULL, verbose, "MPlayer"); if (sipClient == NULL) { fprintf(stderr, "Failed to create SIP client: %s\n", env->getResultMsg()); break; } sipClient->setClientStartPortNum(8000); sdpDescription = openURL_sip(sipClient, url); } if (sdpDescription == NULL) { fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n", url, env->getResultMsg()); break; } } // Now that we have a SDP description, create a MediaSession from it: MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription); if (mediaSession == NULL) break; // Create a 'RTPState' structure containing the state that we just created, // and store it in the demuxer's 'priv' field, for future reference: RTPState* rtpState = new RTPState; rtpState->sdpDescription = sdpDescription; rtpState->rtspClient = rtspClient; rtpState->sipClient = sipClient; rtpState->mediaSession = mediaSession; rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL; rtpState->flags = 0; rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0; demuxer->priv = rtpState; // Create RTP receivers (sources) for each subsession: MediaSubsessionIterator iter(*mediaSession); MediaSubsession* subsession; unsigned desiredReceiveBufferSize; while ((subsession = iter.next()) != NULL) { // Ignore any subsession that's not audio or video: if (strcmp(subsession->mediumName(), "audio") == 0) { desiredReceiveBufferSize = 100000; } else if (strcmp(subsession->mediumName(), "video") == 0) { desiredReceiveBufferSize = 2000000; } else { continue; } if (rtsp_port) subsession->setClientPortNum (rtsp_port); if (!subsession->initiate()) { fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg()); } else { fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum()); // Set the OS's socket receive buffer sufficiently large to avoid // incoming packets getting dropped between successive reads from this // subsession's demuxer. Depending on the bitrate(s) that you expect, // you may wish to tweak the "desiredReceiveBufferSize" values above. int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum(); int receiveBufferSize = increaseReceiveBufferTo(*env, rtpSocketNum, desiredReceiveBufferSize); if (verbose > 0) { fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n", subsession->mediumName(), receiveBufferSize); } if (rtspClient != NULL) { // Issue a RTSP "SETUP" command on the chosen subsession: if (!rtspClient->setupMediaSubsession(*subsession, False, rtspStreamOverTCP)) break; } } } if (rtspClient != NULL) { // Issue a RTSP aggregate "PLAY" command on the whole session: if (!rtspClient->playMediaSession(*mediaSession)) break; } else if (sipClient != NULL) { sipClient->sendACK(); // to start the stream flowing } // Now that the session is ready to be read, do additional // MPlayer codec-specific initialization on each subsession: iter.reset(); while ((subsession = iter.next()) != NULL) { if (subsession->readSource() == NULL) continue; // not reading this unsigned flags = 0; if (strcmp(subsession->mediumName(), "audio") == 0) { rtpState->audioBufferQueue = new ReadBufferQueue(subsession, demuxer, "audio"); rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue); rtpCodecInitialize_audio(demuxer, subsession, flags); } else if (strcmp(subsession->mediumName(), "video") == 0) { rtpState->videoBufferQueue = new ReadBufferQueue(subsession, demuxer, "video"); rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue); rtpCodecInitialize_video(demuxer, subsession, flags); } rtpState->flags |= flags; } success = True; } while (0); if (!success) return NULL; // an error occurred // Hack: If audio and video are demuxed together on a single RTP stream, // then create a new "demuxer_t" structure to allow the higher-level // code to recognize this: if (demux_is_multiplexed_rtp_stream(demuxer)) { stream_t* s = new_ds_stream(demuxer->video); demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN, audio_id, video_id, dvdsub_id, NULL); demuxer = new_demuxers_demuxer(od, od, od); } return demuxer; } extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) { // Get the RTP state that was stored in the demuxer's 'priv' field: RTPState* rtpState = (RTPState*)(demuxer->priv); return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0; } extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) { // Get the RTP state that was stored in the demuxer's 'priv' field: RTPState* rtpState = (RTPState*)(demuxer->priv); return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0; } static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, Boolean mustGetNewData, float& ptsBehind); // forward extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) { // Get a filled-in "demux_packet" from the RTP source, and deliver it. // Note that this is called as a synchronous read operation, so it needs // to block in the (hopefully infrequent) case where no packet is // immediately available. while (1) { float ptsBehind; demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking if (dp == NULL) return 0; if (demuxer->stream->eof) return 0; // source stream has closed down // Before using this packet, check to make sure that its presentation // time is not far behind the other stream (if any). If it is, // then we discard this packet, and get another instead. (The rest of // MPlayer doesn't always do a good job of synchronizing when the // audio and video streams get this far apart.) // (We don't do this when streaming over TCP, because then the audio and // video streams are interleaved.) // (Also, if the stream is *excessively* far behind, then we allow // the packet, because in this case it probably means that there was // an error in the source's timestamp synchronization.) const float ptsBehindThreshold = 1.0; // seconds const float ptsBehindLimit = 60.0; // seconds if (ptsBehind < ptsBehindThreshold || ptsBehind > ptsBehindLimit || rtspStreamOverTCP) { // packet's OK ds_add_packet(ds, dp); break; } #ifdef DEBUG_PRINT_DISCARDED_PACKETS RTPState* rtpState = (RTPState*)(demuxer->priv); ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue; fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind); #endif free_demux_packet(dp); // give back this packet, and get another one } return 1; } Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds, unsigned char*& packetData, unsigned& packetDataLen, float& pts) { // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet" // is not delivered to the "demux_stream". float ptsBehind; demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking if (dp == NULL) return False; packetData = dp->buffer; packetDataLen = dp->len; pts = dp->pts; return True; } static void teardownRTSPorSIPSession(RTPState* rtpState); // forward extern "C" void demux_close_rtp(demuxer_t* demuxer) { // Reclaim all RTP-related state: // Get the RTP state that was stored in the demuxer's 'priv' field: RTPState* rtpState = (RTPState*)(demuxer->priv); if (rtpState == NULL) return; teardownRTSPorSIPSession(rtpState); UsageEnvironment* env = NULL; TaskScheduler* scheduler = NULL; if (rtpState->mediaSession != NULL) { env = &(rtpState->mediaSession->envir()); scheduler = &(env->taskScheduler()); } Medium::close(rtpState->mediaSession); Medium::close(rtpState->rtspClient); Medium::close(rtpState->sipClient); delete rtpState->audioBufferQueue; delete rtpState->videoBufferQueue; delete rtpState->sdpDescription; delete rtpState; env->reclaim(); delete scheduler; } ////////// Extra routines that help implement the above interface functions: #define MAX_RTP_FRAME_SIZE 50000 // >= the largest conceivable frame composed from one or more RTP packets static void afterReading(void* clientData, unsigned frameSize, unsigned /*numTruncatedBytes*/, struct timeval presentationTime, unsigned /*durationInMicroseconds*/) { int headersize = 0; if (frameSize >= MAX_RTP_FRAME_SIZE) { fprintf(stderr, "Saw an input frame too large (>=%d). Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n", MAX_RTP_FRAME_SIZE); } ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData; demuxer_t* demuxer = bufferQueue->ourDemuxer(); RTPState* rtpState = (RTPState*)(demuxer->priv); if (frameSize > 0) demuxer->stream->eof = 0; if (bufferQueue->readSource()->isAMRAudioSource()) headersize = 1; demux_packet_t* dp = bufferQueue->dp; resize_demux_packet(dp, frameSize + headersize); // Set the packet's presentation time stamp, depending on whether or // not our RTP source's timestamps have been synchronized yet: Boolean hasBeenSynchronized = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP(); if (hasBeenSynchronized) { if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) { fprintf(stderr, "%s stream has been synchronized using RTCP \n", bufferQueue->tag()); } struct timeval* fst = &(rtpState->firstSyncTime); // abbrev if (fst->tv_sec == 0 && fst->tv_usec == 0) { *fst = presentationTime; } // For the "pts" field, use the time differential from the first // synchronized time, rather than absolute time, in order to avoid // round-off errors when converting to a float: dp->pts = presentationTime.tv_sec - fst->tv_sec + (presentationTime.tv_usec - fst->tv_usec)/1000000.0; bufferQueue->prevPacketPTS = dp->pts; } else { if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) { fprintf(stderr, "%s stream is no longer RTCP-synchronized \n", bufferQueue->tag()); } // use the previous packet's "pts" once again: dp->pts = bufferQueue->prevPacketPTS; } bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized; dp->pos = demuxer->filepos; demuxer->filepos += frameSize; // Signal any pending 'doEventLoop()' call on this queue: bufferQueue->blockingFlag = ~0; } static void onSourceClosure(void* clientData) { ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData; demuxer_t* demuxer = bufferQueue->ourDemuxer(); demuxer->stream->eof = 1; // Signal any pending 'doEventLoop()' call on this queue: bufferQueue->blockingFlag = ~0; } static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, Boolean mustGetNewData, float& ptsBehind) { // Begin by finding the buffer queue that we want to read from: // (Get this from the RTP state, which we stored in // the demuxer's 'priv' field) RTPState* rtpState = (RTPState*)(demuxer->priv); ReadBufferQueue* bufferQueue = NULL; int amr = 0; if (ds == demuxer->video) { bufferQueue = rtpState->videoBufferQueue; } else if (ds == demuxer->audio) { bufferQueue = rtpState->audioBufferQueue; if (bufferQueue->readSource()->isAMRAudioSource()) amr = 1; } else { fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n"); return NULL; } if (bufferQueue == NULL || bufferQueue->readSource() == NULL) { fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n"); return NULL; } demux_packet_t* dp; if (!mustGetNewData) { // Check whether we have a previously-saved buffer that we can use: dp = bufferQueue->getPendingBuffer(); if (dp != NULL) { ptsBehind = 0.0; // so that we always accept this data return dp; } } // Allocate a new packet buffer, and arrange to read into it: dp = new_demux_packet(MAX_RTP_FRAME_SIZE); bufferQueue->dp = dp; if (dp == NULL) return NULL; // Schedule the read operation: bufferQueue->blockingFlag = 0; bufferQueue->readSource()->getNextFrame(&dp->buffer[amr], MAX_RTP_FRAME_SIZE - amr, afterReading, bufferQueue, onSourceClosure, bufferQueue); // Block ourselves until data becomes available: TaskScheduler& scheduler = bufferQueue->readSource()->envir().taskScheduler(); scheduler.doEventLoop(&bufferQueue->blockingFlag); if (amr) dp->buffer[0] = ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader(); // Set the "ptsBehind" result parameter: if (bufferQueue->prevPacketPTS != 0.0 && bufferQueue->prevPacketWasSynchronized && *(bufferQueue->otherQueue) != NULL && (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0 && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) { ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS - bufferQueue->prevPacketPTS; } else { ptsBehind = 0.0; } if (mustGetNewData) { // Save this buffer for future reads: bufferQueue->savePendingBuffer(dp); } return dp; } static void teardownRTSPorSIPSession(RTPState* rtpState) { MediaSession* mediaSession = rtpState->mediaSession; if (mediaSession == NULL) return; if (rtpState->rtspClient != NULL) { MediaSubsessionIterator iter(*mediaSession); MediaSubsession* subsession; while ((subsession = iter.next()) != NULL) { rtpState->rtspClient->teardownMediaSubsession(*subsession); } } else if (rtpState->sipClient != NULL) { rtpState->sipClient->sendBYE(); } } ////////// "ReadBuffer" and "ReadBufferQueue" implementation: ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer, char const* tag) : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL), dp(NULL), pendingDPHead(NULL), pendingDPTail(NULL), fReadSource(subsession == NULL ? NULL : subsession->readSource()), fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()), fOurDemuxer(demuxer), fTag(strdup(tag)) { } ReadBufferQueue::~ReadBufferQueue() { delete fTag; // Free any pending buffers (that never got delivered): demux_packet_t* dp = pendingDPHead; while (dp != NULL) { demux_packet_t* dpNext = dp->next; dp->next = NULL; free_demux_packet(dp); dp = dpNext; } } void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) { // Keep this buffer around, until MPlayer asks for it later: if (pendingDPTail == NULL) { pendingDPHead = pendingDPTail = dp; } else { pendingDPTail->next = dp; pendingDPTail = dp; } dp->next = NULL; } demux_packet_t* ReadBufferQueue::getPendingBuffer() { demux_packet_t* dp = pendingDPHead; if (dp != NULL) { pendingDPHead = dp->next; if (pendingDPHead == NULL) pendingDPTail = NULL; dp->next = NULL; } return dp; } static int demux_rtp_control(struct demuxer_st *demuxer, int cmd, void *arg) { double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime(); switch(cmd) { case DEMUXER_CTRL_GET_TIME_LENGTH: if (endpts <= 0) return DEMUXER_CTRL_DONTKNOW; *((double *)arg) = endpts; return DEMUXER_CTRL_OK; case DEMUXER_CTRL_GET_PERCENT_POS: if (endpts <= 0) return DEMUXER_CTRL_DONTKNOW; *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts); return DEMUXER_CTRL_OK; default: return DEMUXER_CTRL_NOTIMPL; } } demuxer_desc_t demuxer_desc_rtp = { "LIVE555 RTP demuxer", "rtp", "", "Ross Finlayson", "requires LIVE555 Streaming Media library", DEMUXER_TYPE_RTP, 0, // no autodetect NULL, demux_rtp_fill_buffer, demux_open_rtp, demux_close_rtp, NULL, demux_rtp_control };