mirror of
https://github.com/mpv-player/mpv
synced 2024-12-09 16:36:15 +00:00
a26dcc7b48
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12867 b3059339-0415-0410-9bf9-f77b7e298cf2
612 lines
21 KiB
C++
612 lines
21 KiB
C++
////////// Routines (with C-linkage) that interface between "MPlayer"
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////////// and the "LIVE.COM Streaming Media" libraries:
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extern "C" {
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// on MinGW, we must include windows.h before the things it conflicts
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#ifdef __MINGW32__ // with. they are each protected from
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#include <windows.h> // windows.h, but not the other way around.
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#endif
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#include "demux_rtp.h"
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#include "stheader.h"
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}
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#include "demux_rtp_internal.h"
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#include "BasicUsageEnvironment.hh"
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#include "liveMedia.hh"
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#include "GroupsockHelper.hh"
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#include <unistd.h>
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extern "C" stream_t* stream_open_sdp(int fd, off_t fileSize,
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int* file_format) {
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*file_format = DEMUXER_TYPE_RTP;
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stream_t* newStream = NULL;
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do {
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char* sdpDescription = (char*)malloc(fileSize+1);
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if (sdpDescription == NULL) break;
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ssize_t numBytesRead = read(fd, sdpDescription, fileSize);
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if (numBytesRead != fileSize) break;
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sdpDescription[fileSize] = '\0'; // to be safe
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newStream = (stream_t*)calloc(sizeof (stream_t), 1);
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if (newStream == NULL) break;
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// Store the SDP description in the 'priv' field, for later use:
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newStream->priv = sdpDescription;
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} while (0);
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return newStream;
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}
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extern "C" int _rtsp_streaming_seek(int /*fd*/, off_t /*pos*/,
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streaming_ctrl_t* /*streaming_ctrl*/) {
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return -1; // For now, we don't handle RTSP stream seeking
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}
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extern "C" int rtsp_streaming_start(stream_t* stream) {
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stream->streaming_ctrl->streaming_seek = _rtsp_streaming_seek;
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return 0;
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}
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// A data structure representing input data for each stream:
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class ReadBufferQueue {
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public:
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ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
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char const* tag);
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virtual ~ReadBufferQueue();
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FramedSource* readSource() const { return fReadSource; }
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RTPSource* rtpSource() const { return fRTPSource; }
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demuxer_t* ourDemuxer() const { return fOurDemuxer; }
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char const* tag() const { return fTag; }
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char blockingFlag; // used to implement synchronous reads
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// For A/V synchronization:
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Boolean prevPacketWasSynchronized;
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float prevPacketPTS;
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ReadBufferQueue** otherQueue;
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// The 'queue' actually consists of just a single "demux_packet_t"
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// (because the underlying OS does the actual queueing/buffering):
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demux_packet_t* dp;
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// However, we sometimes inspect buffers before delivering them.
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// For this, we maintain a queue of pending buffers:
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void savePendingBuffer(demux_packet_t* dp);
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demux_packet_t* getPendingBuffer();
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private:
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demux_packet_t* pendingDPHead;
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demux_packet_t* pendingDPTail;
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FramedSource* fReadSource;
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RTPSource* fRTPSource;
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demuxer_t* fOurDemuxer;
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char const* fTag; // used for debugging
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};
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// A structure of RTP-specific state, kept so that we can cleanly
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// reclaim it:
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typedef struct RTPState {
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char const* sdpDescription;
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RTSPClient* rtspClient;
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SIPClient* sipClient;
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MediaSession* mediaSession;
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ReadBufferQueue* audioBufferQueue;
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ReadBufferQueue* videoBufferQueue;
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unsigned flags;
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struct timeval firstSyncTime;
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};
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extern "C" char* network_username;
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extern "C" char* network_password;
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static char* openURL_rtsp(RTSPClient* client, char const* url) {
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// If we were given a user name (and optional password), then use them:
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if (network_username != NULL) {
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char const* password = network_password == NULL ? "" : network_password;
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return client->describeWithPassword(url, network_username, password);
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} else {
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return client->describeURL(url);
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}
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}
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static char* openURL_sip(SIPClient* client, char const* url) {
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// If we were given a user name (and optional password), then use them:
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if (network_username != NULL) {
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char const* password = network_password == NULL ? "" : network_password;
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return client->inviteWithPassword(url, network_username, password);
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} else {
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return client->invite(url);
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}
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}
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int rtspStreamOverTCP = 0;
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extern "C" int audio_id, video_id, dvdsub_id;
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extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
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Boolean success = False;
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do {
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TaskScheduler* scheduler = BasicTaskScheduler::createNew();
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if (scheduler == NULL) break;
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UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
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if (env == NULL) break;
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RTSPClient* rtspClient = NULL;
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SIPClient* sipClient = NULL;
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if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen
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demuxer->stream->eof = 0; // just in case
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// Look at the stream's 'priv' field to see if we were initiated
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// via a SDP description:
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char* sdpDescription = (char*)(demuxer->stream->priv);
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if (sdpDescription == NULL) {
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// We weren't given a SDP description directly, so assume that
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// we were given a RTSP or SIP URL:
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char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
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char const* url = demuxer->stream->streaming_ctrl->url->url;
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extern int verbose;
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if (strcmp(protocol, "rtsp") == 0) {
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rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
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if (rtspClient == NULL) {
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fprintf(stderr, "Failed to create RTSP client: %s\n",
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env->getResultMsg());
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break;
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}
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sdpDescription = openURL_rtsp(rtspClient, url);
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} else { // SIP
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unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
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sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
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verbose, "MPlayer");
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if (sipClient == NULL) {
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fprintf(stderr, "Failed to create SIP client: %s\n",
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env->getResultMsg());
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break;
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}
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sipClient->setClientStartPortNum(8000);
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sdpDescription = openURL_sip(sipClient, url);
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}
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if (sdpDescription == NULL) {
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fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
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url, env->getResultMsg());
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break;
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}
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}
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// Now that we have a SDP description, create a MediaSession from it:
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MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
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if (mediaSession == NULL) break;
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// Create a 'RTPState' structure containing the state that we just created,
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// and store it in the demuxer's 'priv' field, for future reference:
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RTPState* rtpState = new RTPState;
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rtpState->sdpDescription = sdpDescription;
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rtpState->rtspClient = rtspClient;
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rtpState->sipClient = sipClient;
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rtpState->mediaSession = mediaSession;
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rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
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rtpState->flags = 0;
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rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
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demuxer->priv = rtpState;
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// Create RTP receivers (sources) for each subsession:
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MediaSubsessionIterator iter(*mediaSession);
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MediaSubsession* subsession;
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unsigned desiredReceiveBufferSize;
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while ((subsession = iter.next()) != NULL) {
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// Ignore any subsession that's not audio or video:
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if (strcmp(subsession->mediumName(), "audio") == 0) {
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desiredReceiveBufferSize = 100000;
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} else if (strcmp(subsession->mediumName(), "video") == 0) {
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desiredReceiveBufferSize = 2000000;
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} else {
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continue;
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}
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if (!subsession->initiate()) {
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fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
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} else {
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fprintf(stderr, "Initiated \"%s/%s\" RTP subsession\n", subsession->mediumName(), subsession->codecName());
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// Set the OS's socket receive buffer sufficiently large to avoid
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// incoming packets getting dropped between successive reads from this
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// subsession's demuxer. Depending on the bitrate(s) that you expect,
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// you may wish to tweak the "desiredReceiveBufferSize" values above.
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int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
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int receiveBufferSize
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= increaseReceiveBufferTo(*env, rtpSocketNum,
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desiredReceiveBufferSize);
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if (verbose > 0) {
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fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
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subsession->mediumName(), receiveBufferSize);
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}
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if (rtspClient != NULL) {
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// Issue a RTSP "SETUP" command on the chosen subsession:
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if (!rtspClient->setupMediaSubsession(*subsession, False,
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rtspStreamOverTCP)) break;
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}
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}
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}
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if (rtspClient != NULL) {
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// Issue a RTSP aggregate "PLAY" command on the whole session:
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if (!rtspClient->playMediaSession(*mediaSession)) break;
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} else if (sipClient != NULL) {
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sipClient->sendACK(); // to start the stream flowing
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}
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// Now that the session is ready to be read, do additional
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// MPlayer codec-specific initialization on each subsession:
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iter.reset();
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while ((subsession = iter.next()) != NULL) {
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if (subsession->readSource() == NULL) continue; // not reading this
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unsigned flags = 0;
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if (strcmp(subsession->mediumName(), "audio") == 0) {
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rtpState->audioBufferQueue
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= new ReadBufferQueue(subsession, demuxer, "audio");
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rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
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rtpCodecInitialize_audio(demuxer, subsession, flags);
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} else if (strcmp(subsession->mediumName(), "video") == 0) {
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rtpState->videoBufferQueue
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= new ReadBufferQueue(subsession, demuxer, "video");
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rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
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rtpCodecInitialize_video(demuxer, subsession, flags);
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}
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rtpState->flags |= flags;
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}
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success = True;
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} while (0);
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if (!success) return NULL; // an error occurred
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// Hack: If audio and video are demuxed together on a single RTP stream,
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// then create a new "demuxer_t" structure to allow the higher-level
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// code to recognize this:
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if (demux_is_multiplexed_rtp_stream(demuxer)) {
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stream_t* s = new_ds_stream(demuxer->video);
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demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN,
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audio_id, video_id, dvdsub_id, NULL);
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demuxer = new_demuxers_demuxer(od, od, od);
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}
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return demuxer;
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}
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extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
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// Get the RTP state that was stored in the demuxer's 'priv' field:
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
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}
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extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) {
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// Get the RTP state that was stored in the demuxer's 'priv' field:
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0;
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}
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static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
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Boolean mustGetNewData,
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float& ptsBehind); // forward
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extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
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// Get a filled-in "demux_packet" from the RTP source, and deliver it.
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// Note that this is called as a synchronous read operation, so it needs
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// to block in the (hopefully infrequent) case where no packet is
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// immediately available.
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while (1) {
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float ptsBehind;
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demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
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if (dp == NULL) return 0;
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if (demuxer->stream->eof) return 0; // source stream has closed down
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// Before using this packet, check to make sure that its presentation
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// time is not far behind the other stream (if any). If it is,
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// then we discard this packet, and get another instead. (The rest of
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// MPlayer doesn't always do a good job of synchronizing when the
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// audio and video streams get this far apart.)
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// (We don't do this when streaming over TCP, because then the audio and
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// video streams are interleaved.)
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// (Also, if the stream is *excessively* far behind, then we allow
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// the packet, because in this case it probably means that there was
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// an error in the source's timestamp synchronization.)
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const float ptsBehindThreshold = 1.0; // seconds
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const float ptsBehindLimit = 60.0; // seconds
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if (ptsBehind < ptsBehindThreshold ||
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ptsBehind > ptsBehindLimit ||
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rtspStreamOverTCP) { // packet's OK
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ds_add_packet(ds, dp);
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break;
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}
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#ifdef DEBUG_PRINT_DISCARDED_PACKETS
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue;
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fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind);
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#endif
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free_demux_packet(dp); // give back this packet, and get another one
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}
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return 1;
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}
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Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds,
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unsigned char*& packetData, unsigned& packetDataLen,
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float& pts) {
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// Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
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// is not delivered to the "demux_stream".
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float ptsBehind;
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demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking
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if (dp == NULL) return False;
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packetData = dp->buffer;
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packetDataLen = dp->len;
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pts = dp->pts;
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return True;
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}
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Boolean insertRTPData(demuxer_t* demuxer, demux_stream_t* ds,
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unsigned char* data, unsigned dataLen) {
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// Begin by finding the buffer queue that we want to add data to.
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// (Get this from the RTP state, which we stored in
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// the demuxer's 'priv' field)
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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ReadBufferQueue* bufferQueue = NULL;
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if (ds == demuxer->video) {
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bufferQueue = rtpState->videoBufferQueue;
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} else if (ds == demuxer->audio) {
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bufferQueue = rtpState->audioBufferQueue;
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} else {
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fprintf(stderr, "(demux_rtp)insertRTPData: internal error: unknown stream\n");
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return False;
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}
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if (data == NULL || dataLen == 0) return False;
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demux_packet_t* dp = new_demux_packet(dataLen);
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if (dp == NULL) return False;
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// Copy our data into the buffer, and save it:
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memmove(dp->buffer, data, dataLen);
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dp->len = dataLen;
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dp->pts = 0;
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bufferQueue->savePendingBuffer(dp);
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return True;
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}
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static void teardownRTSPorSIPSession(RTPState* rtpState); // forward
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extern "C" void demux_close_rtp(demuxer_t* demuxer) {
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// Reclaim all RTP-related state:
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// Get the RTP state that was stored in the demuxer's 'priv' field:
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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if (rtpState == NULL) return;
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teardownRTSPorSIPSession(rtpState);
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UsageEnvironment* env = NULL;
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TaskScheduler* scheduler = NULL;
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if (rtpState->mediaSession != NULL) {
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env = &(rtpState->mediaSession->envir());
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scheduler = &(env->taskScheduler());
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}
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Medium::close(rtpState->mediaSession);
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Medium::close(rtpState->rtspClient);
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Medium::close(rtpState->sipClient);
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delete rtpState->audioBufferQueue;
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delete rtpState->videoBufferQueue;
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delete rtpState->sdpDescription;
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delete rtpState;
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env->reclaim(); delete scheduler;
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}
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////////// Extra routines that help implement the above interface functions:
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#define MAX_RTP_FRAME_SIZE 50000
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// >= the largest conceivable frame composed from one or more RTP packets
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static void afterReading(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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if (frameSize >= MAX_RTP_FRAME_SIZE) {
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fprintf(stderr, "Saw an input frame too large (>=%d). Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
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MAX_RTP_FRAME_SIZE);
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}
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ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
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demuxer_t* demuxer = bufferQueue->ourDemuxer();
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RTPState* rtpState = (RTPState*)(demuxer->priv);
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if (frameSize > 0) demuxer->stream->eof = 0;
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demux_packet_t* dp = bufferQueue->dp;
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dp->len = frameSize;
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// Set the packet's presentation time stamp, depending on whether or
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// not our RTP source's timestamps have been synchronized yet:
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Boolean hasBeenSynchronized
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= bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
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if (hasBeenSynchronized) {
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if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
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fprintf(stderr, "%s stream has been synchronized using RTCP \n",
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bufferQueue->tag());
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}
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struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
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if (fst->tv_sec == 0 && fst->tv_usec == 0) {
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*fst = presentationTime;
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}
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// For the "pts" field, use the time differential from the first
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// synchronized time, rather than absolute time, in order to avoid
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// round-off errors when converting to a float:
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dp->pts = presentationTime.tv_sec - fst->tv_sec
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+ (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
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bufferQueue->prevPacketPTS = dp->pts;
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} else {
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if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) {
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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;
|
|
if (ds == demuxer->video) {
|
|
bufferQueue = rtpState->videoBufferQueue;
|
|
} else if (ds == demuxer->audio) {
|
|
bufferQueue = rtpState->audioBufferQueue;
|
|
} 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, MAX_RTP_FRAME_SIZE,
|
|
afterReading, bufferQueue,
|
|
onSourceClosure, bufferQueue);
|
|
// Block ourselves until data becomes available:
|
|
TaskScheduler& scheduler
|
|
= bufferQueue->readSource()->envir().taskScheduler();
|
|
scheduler.doEventLoop(&bufferQueue->blockingFlag);
|
|
|
|
// 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;
|
|
}
|