Mercurial > mplayer.hg
view libmpdemux/demux_rtp.cpp @ 7788:25bf4f45639e
#ifdef PIX_FMT_YUV411P...
author | arpi |
---|---|
date | Sat, 19 Oct 2002 00:23:50 +0000 |
parents | 3a2b4bf47dbc |
children | 099205debc97 |
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extern "C" { #include "demux_rtp.h" #include "stheader.h" } #include "BasicUsageEnvironment.hh" #include "liveMedia.hh" #include <unistd.h> ////////// Routines (with C-linkage) that interface between "mplayer" ////////// and the "LIVE.COM Streaming Media" libraries: extern "C" stream_t* stream_open_sdp(int fd, off_t fileSize, int* file_format) { *file_format = DEMUXER_TYPE_RTP; stream_t* newStream = NULL; do { char* sdpDescription = (char*)malloc(fileSize+1); if (sdpDescription == NULL) break; ssize_t numBytesRead = read(fd, sdpDescription, fileSize); if (numBytesRead != fileSize) break; sdpDescription[fileSize] = '\0'; // to be safe newStream = (stream_t*)calloc(sizeof (stream_t), 1); if (newStream == NULL) break; // Store the SDP description in the 'priv' field, for later use: newStream->priv = sdpDescription; } while (0); return newStream; } extern "C" int _rtsp_streaming_seek(int /*fd*/, off_t /*pos*/, streaming_ctrl_t* /*streaming_ctrl*/) { return -1; // For now, we don't handle RTSP stream seeking } extern "C" int rtsp_streaming_start(stream_t* stream) { stream->streaming_ctrl->streaming_seek = _rtsp_streaming_seek; return 0; } // A data structure representing a buffer being read: class ReadBufferQueue; // forward class ReadBuffer { public: ReadBuffer(ReadBufferQueue* ourQueue, demux_packet_t* dp); virtual ~ReadBuffer(); Boolean enqueue(); demux_packet_t* dp() const { return fDP; } ReadBufferQueue* ourQueue() { return fOurQueue; } ReadBuffer* next; private: demux_packet_t* fDP; ReadBufferQueue* fOurQueue; }; class ReadBufferQueue { public: ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer, char const* tag); virtual ~ReadBufferQueue(); ReadBuffer* dequeue(); FramedSource* readSource() const { return fReadSource; } RTPSource* rtpSource() const { return fRTPSource; } demuxer_t* ourDemuxer() const { return fOurDemuxer; } char const* tag() const { return fTag; } ReadBuffer* head; ReadBuffer* tail; char blockingFlag; // used to implement synchronous reads unsigned counter; // used for debugging private: 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; MediaSession* mediaSession; ReadBufferQueue* audioBufferQueue; ReadBufferQueue* videoBufferQueue; int isMPEG; // TRUE for MPEG audio, video, or transport streams struct timeval firstSyncTime; }; int rtspStreamOverTCP = 0; extern "C" void demux_open_rtp(demuxer_t* demuxer) { if (rtspStreamOverTCP && LIVEMEDIA_LIBRARY_VERSION_INT < 1033689600) { fprintf(stderr, "TCP streaming of RTP/RTCP requires \"LIVE.COM Streaming Media\" library version 2002.10.04 or later - ignoring the \"-rtsp-stream-over-tcp\" flag\n"); rtspStreamOverTCP = 0; } do { TaskScheduler* scheduler = BasicTaskScheduler::createNew(); if (scheduler == NULL) break; UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); if (env == NULL) break; RTSPClient* rtspClient = NULL; int isMPEG = 0; // 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 give a RTSP URL char const* url = demuxer->stream->streaming_ctrl->url->url; extern int verbose; rtspClient = RTSPClient::createNew(*env, verbose, "mplayer"); if (rtspClient == NULL) { fprintf(stderr, "Failed to create RTSP client: %s\n", env->getResultMsg()); break; } sdpDescription = rtspClient->describeURL(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 RTP receivers (sources) for each subsession: MediaSubsessionIterator iter(*mediaSession); MediaSubsession* subsession; MediaSubsession* audioSubsession = NULL; MediaSubsession* videoSubsession = NULL; while ((subsession = iter.next()) != NULL) { // Ignore any subsession that's not audio or video: if (strcmp(subsession->mediumName(), "audio") == 0) { audioSubsession = subsession; } else if (strcmp(subsession->mediumName(), "video") == 0) { videoSubsession = subsession; } else { continue; } 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\n", subsession->mediumName(), subsession->codecName()); if (rtspClient != NULL) { // Issue RTSP "SETUP" and "PLAY" commands on the chosen subsession: if (!rtspClient->setupMediaSubsession(*subsession, False, rtspStreamOverTCP)) break; if (!rtspClient->playMediaSubsession(*subsession)) break; } // Now that the subsession is ready to be read, do additional // mplayer-specific initialization on it: if (subsession == videoSubsession) { // Create a dummy video stream header // to make the main mplayer code happy: sh_video_t* sh_video = new_sh_video(demuxer,0); BITMAPINFOHEADER* bih = (BITMAPINFOHEADER*)calloc(1,sizeof(BITMAPINFOHEADER)); bih->biSize = sizeof(BITMAPINFOHEADER); sh_video->bih = bih; demux_stream_t* d_video = demuxer->video; d_video->sh = sh_video; sh_video->ds = d_video; // If we happen to know the subsession's video frame rate, set it, // so that the user doesn't have to give the "-fps" option instead. int fps = (int)(subsession->videoFPS()); if (fps != 0) sh_video->fps = fps; // Map known video MIME types to the BITMAPINFOHEADER parameters // that this program uses. (Note that not all types need all // of the parameters to be set.) if (strcmp(subsession->codecName(), "MPV") == 0 || strcmp(subsession->codecName(), "MP1S") == 0 || strcmp(subsession->codecName(), "MP2T") == 0) { isMPEG = 1; } else if (strcmp(subsession->codecName(), "H263") == 0 || strcmp(subsession->codecName(), "H263-1998") == 0) { bih->biCompression = sh_video->format = mmioFOURCC('H','2','6','3'); } else if (strcmp(subsession->codecName(), "H261") == 0) { bih->biCompression = sh_video->format = mmioFOURCC('H','2','6','1'); } else { fprintf(stderr, "Unknown mplayer format code for MIME type \"video/%s\"\n", subsession->codecName()); } } else if (subsession == audioSubsession) { // Create a dummy audio stream header // to make the main mplayer code happy: sh_audio_t* sh_audio = new_sh_audio(demuxer,0); WAVEFORMATEX* wf = (WAVEFORMATEX*)calloc(1,sizeof(WAVEFORMATEX)); sh_audio->wf = wf; demux_stream_t* d_audio = demuxer->audio; d_audio->sh = sh_audio; sh_audio->ds = d_audio; // Map known audio MIME types to the WAVEFORMATEX parameters // that this program uses. (Note that not all types need all // of the parameters to be set.) wf->nSamplesPerSec = subsession->rtpSource()->timestampFrequency(); // by default if (strcmp(subsession->codecName(), "MPA") == 0 || strcmp(subsession->codecName(), "MPA-ROBUST") == 0 || strcmp(subsession->codecName(), "X-MP3-DRAFT-00") == 0) { wf->wFormatTag = sh_audio->format = 0x55; // Note: 0x55 is for layer III, but should work for I,II also wf->nSamplesPerSec = 0; // sample rate is deduced from the data } else if (strcmp(subsession->codecName(), "AC3") == 0) { wf->wFormatTag = sh_audio->format = 0x2000; wf->nSamplesPerSec = 0; // sample rate is deduced from the data } else if (strcmp(subsession->codecName(), "PCMU") == 0) { wf->wFormatTag = sh_audio->format = 0x7; wf->nChannels = 1; wf->nAvgBytesPerSec = 8000; wf->nBlockAlign = 1; wf->wBitsPerSample = 8; wf->cbSize = 0; } else if (strcmp(subsession->codecName(), "PCMA") == 0) { wf->wFormatTag = sh_audio->format = 0x6; wf->nChannels = 1; wf->nAvgBytesPerSec = 8000; wf->nBlockAlign = 1; wf->wBitsPerSample = 8; wf->cbSize = 0; } else if (strcmp(subsession->codecName(), "GSM") == 0) { wf->wFormatTag = sh_audio->format = mmioFOURCC('a','g','s','m'); wf->nChannels = 1; wf->nAvgBytesPerSec = 1650; wf->nBlockAlign = 33; wf->wBitsPerSample = 16; wf->cbSize = 0; } else { fprintf(stderr, "Unknown mplayer format code for MIME type \"audio/%s\"\n", subsession->codecName()); } } } } // Hack: Create a 'RTPState' structure containing the state that // we just created, and store it in the demuxer's 'priv' field: RTPState* rtpState = new RTPState; rtpState->sdpDescription = sdpDescription; rtpState->rtspClient = rtspClient; rtpState->mediaSession = mediaSession; rtpState->audioBufferQueue = new ReadBufferQueue(audioSubsession, demuxer, "audio"); rtpState->videoBufferQueue = new ReadBufferQueue(videoSubsession, demuxer, "video"); rtpState->isMPEG = isMPEG; rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0; demuxer->priv = rtpState; } while (0); } 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->isMPEG; } static Boolean deliverBufferIfAvailable(ReadBufferQueue* bufferQueue, demux_stream_t* ds); // 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. // 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; if (ds == demuxer->video) { bufferQueue = rtpState->videoBufferQueue; } else if (ds == demuxer->audio) { bufferQueue = rtpState->audioBufferQueue; } else { fprintf(stderr, "demux_rtp_fill_buffer: internal error: unknown stream\n"); return 0; } if (bufferQueue == NULL || bufferQueue->readSource() == NULL) { fprintf(stderr, "demux_rtp_fill_buffer failed: no appropriate RTP subsession has been set up\n"); return 0; } // Check whether there's a full buffer to deliver to the client: bufferQueue->blockingFlag = 0; while (!deliverBufferIfAvailable(bufferQueue, ds)) { // Because we weren't able to deliver a buffer to the client immediately, // block myself until one comes available: TaskScheduler& scheduler = bufferQueue->readSource()->envir().taskScheduler(); scheduler.blockMyself(&bufferQueue->blockingFlag); } if (demuxer->stream->eof) return 0; // source stream has closed down return 1; } 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; 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); delete rtpState->audioBufferQueue; delete rtpState->videoBufferQueue; delete rtpState->sdpDescription; delete rtpState; delete env; delete scheduler; } ////////// Extra routines that help implement the above interface functions: static void scheduleNewBufferRead(ReadBufferQueue* bufferQueue); // forward static Boolean deliverBufferIfAvailable(ReadBufferQueue* bufferQueue, demux_stream_t* ds) { Boolean deliveredBuffer = False; ReadBuffer* readBuffer = bufferQueue->dequeue(); if (readBuffer != NULL) { // Append the packet to the reader's DS stream: ds_add_packet(ds, readBuffer->dp()); deliveredBuffer = True; } // Arrange to read a new packet into this queue: scheduleNewBufferRead(bufferQueue); return deliveredBuffer; } static void afterReading(void* clientData, unsigned frameSize, struct timeval presentationTime); // forward static void onSourceClosure(void* clientData); // forward static void scheduleNewBufferRead(ReadBufferQueue* bufferQueue) { if (bufferQueue->readSource()->isCurrentlyAwaitingData()) return; // a read from this source is already in progress // Allocate a new packet buffer, and arrange to read into it: unsigned const bufferSize = 30000; // >= the largest conceivable RTP packet demux_packet_t* dp = new_demux_packet(bufferSize); if (dp == NULL) return; ReadBuffer* readBuffer = new ReadBuffer(bufferQueue, dp); // Schedule the read operation: bufferQueue->readSource()->getNextFrame(dp->buffer, bufferSize, afterReading, readBuffer, onSourceClosure, readBuffer); } static void afterReading(void* clientData, unsigned frameSize, struct timeval presentationTime) { ReadBuffer* readBuffer = (ReadBuffer*)clientData; ReadBufferQueue* bufferQueue = readBuffer->ourQueue(); demuxer_t* demuxer = bufferQueue->ourDemuxer(); RTPState* rtpState = (RTPState*)(demuxer->priv); if (frameSize > 0) demuxer->stream->eof = 0; demux_packet_t* dp = readBuffer->dp(); dp->len = frameSize; // 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) { 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; } else { dp->pts = 0.0; } } dp->pos = demuxer->filepos; demuxer->filepos += frameSize; if (!readBuffer->enqueue()) { // The queue is full, so discard the buffer: delete readBuffer; } // Signal any pending 'blockMyself()' call on this queue: bufferQueue->blockingFlag = ~0; // Finally, arrange to do another read, if appropriate scheduleNewBufferRead(bufferQueue); } static void onSourceClosure(void* clientData) { ReadBuffer* readBuffer = (ReadBuffer*)clientData; ReadBufferQueue* bufferQueue = readBuffer->ourQueue(); demuxer_t* demuxer = bufferQueue->ourDemuxer(); demuxer->stream->eof = 1; // Signal any pending 'blockMyself()' call on this queue: bufferQueue->blockingFlag = ~0; } ////////// "ReadBuffer" and "ReadBufferQueue" implementation: #define MAX_QUEUE_SIZE 5 ReadBuffer::ReadBuffer(ReadBufferQueue* ourQueue, demux_packet_t* dp) : next(NULL), fDP(dp), fOurQueue(ourQueue) { } Boolean ReadBuffer::enqueue() { if (fOurQueue->counter >= MAX_QUEUE_SIZE) { // This queue is full. Clear out an old entry from it, so that // this new one will fit: while (fOurQueue->counter >= MAX_QUEUE_SIZE) { delete fOurQueue->dequeue(); } } // Add ourselves to the tail of our queue: if (fOurQueue->tail == NULL) { fOurQueue->head = this; } else { fOurQueue->tail->next = this; } fOurQueue->tail = this; ++fOurQueue->counter; return True; } ReadBuffer::~ReadBuffer() { free_demux_packet(fDP); delete next; } ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer, char const* tag) : head(NULL), tail(NULL), counter(0), fReadSource(subsession == NULL ? NULL : subsession->readSource()), fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()), fOurDemuxer(demuxer), fTag(strdup(tag)) { } ReadBufferQueue::~ReadBufferQueue() { delete head; delete fTag; } ReadBuffer* ReadBufferQueue::dequeue() { ReadBuffer* readBuffer = head; if (readBuffer != NULL) { head = readBuffer->next; if (head == NULL) tail = NULL; --counter; readBuffer->next = NULL; } return readBuffer; }