2009-05-08 21:51:13 +00:00
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/*
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* routines (with C-linkage) that interface between MPlayer
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* and the "LIVE555 Streaming Media" libraries
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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2003-02-03 10:27:50 +00:00
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2002-08-05 00:39:07 +00:00
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extern "C" {
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2004-06-03 13:30:55 +00:00
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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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2002-08-05 00:39:07 +00:00
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#include "demux_rtp.h"
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#include "stheader.h"
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}
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2003-02-03 10:27:50 +00:00
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#include "demux_rtp_internal.h"
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2002-08-05 00:39:07 +00:00
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#include "BasicUsageEnvironment.hh"
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#include "liveMedia.hh"
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2003-03-11 19:08:31 +00:00
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#include "GroupsockHelper.hh"
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2002-08-05 00:39:07 +00:00
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#include <unistd.h>
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2003-03-11 19:08:31 +00:00
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// A data structure representing input data for each stream:
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2002-08-05 00:39:07 +00:00
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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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2002-09-28 18:51:44 +00:00
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RTPSource* rtpSource() const { return fRTPSource; }
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2002-08-05 00:39:07 +00:00
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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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2003-03-11 19:08:31 +00:00
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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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2007-04-01 03:03:05 +00:00
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// For H264 over rtsp using AVParser, the next packet has to be saved
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demux_packet_t* nextpacket;
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2002-08-05 00:39:07 +00:00
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private:
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2003-03-11 19:08:31 +00:00
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demux_packet_t* pendingDPHead;
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demux_packet_t* pendingDPTail;
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2002-08-05 00:39:07 +00:00
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FramedSource* fReadSource;
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2002-09-28 18:51:44 +00:00
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RTPSource* fRTPSource;
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2002-08-05 00:39:07 +00:00
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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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2003-05-03 06:13:11 +00:00
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SIPClient* sipClient;
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2002-08-05 00:39:07 +00:00
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MediaSession* mediaSession;
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ReadBufferQueue* audioBufferQueue;
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ReadBufferQueue* videoBufferQueue;
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2003-02-03 10:27:50 +00:00
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unsigned flags;
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2002-09-28 18:51:44 +00:00
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struct timeval firstSyncTime;
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2002-08-05 00:39:07 +00:00
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};
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2003-04-11 02:35:01 +00:00
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extern "C" char* network_username;
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extern "C" char* network_password;
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2003-05-03 06:13:11 +00:00
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static char* openURL_rtsp(RTSPClient* client, char const* url) {
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2009-05-13 02:58:57 +00:00
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// If we were given a user name (and optional password), then use them:
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2003-05-03 06:13:11 +00:00
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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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2009-05-13 02:58:57 +00:00
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// If we were given a user name (and optional password), then use them:
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2003-05-03 06:13:11 +00:00
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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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2009-05-25 20:28:35 +00:00
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#ifdef CONFIG_LIBNEMESI
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extern int rtsp_transport_tcp;
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#else
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int rtsp_transport_tcp = 0;
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#endif
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2006-06-26 22:00:44 +00:00
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extern int rtsp_port;
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2009-08-11 18:52:18 +00:00
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#ifdef CONFIG_LIBAVCODEC
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extern AVCodecContext *avcctx;
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#endif
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2002-10-08 05:46:23 +00:00
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2004-06-02 06:48:25 +00:00
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extern "C" int audio_id, video_id, dvdsub_id;
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2004-02-22 06:20:08 +00:00
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extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
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2004-03-19 10:15:41 +00:00
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Boolean success = False;
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2002-08-05 00:39:07 +00:00
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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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2003-05-03 06:13:11 +00:00
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SIPClient* sipClient = NULL;
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2002-08-05 00:39:07 +00:00
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2003-02-18 22:33:44 +00:00
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if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen
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2009-05-13 02:58:57 +00:00
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demuxer->stream->eof = 0; // just in case
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2003-02-18 22:33:44 +00:00
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2002-08-05 00:39:07 +00:00
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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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2003-05-03 06:13:11 +00:00
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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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2002-08-05 00:39:07 +00:00
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char const* url = demuxer->stream->streaming_ctrl->url->url;
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extern int verbose;
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2003-05-03 06:13:11 +00:00
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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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2002-08-05 00:39:07 +00:00
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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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2003-02-03 10:27:50 +00:00
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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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2003-05-03 06:13:11 +00:00
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rtpState->sipClient = sipClient;
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2003-02-03 10:27:50 +00:00
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rtpState->mediaSession = mediaSession;
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2003-02-09 17:06:38 +00:00
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rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
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2003-03-11 19:08:31 +00:00
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rtpState->flags = 0;
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2003-02-03 10:27:50 +00:00
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rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
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demuxer->priv = rtpState;
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2007-12-02 18:04:14 +00:00
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int audiofound = 0, videofound = 0;
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2002-08-05 00:39:07 +00:00
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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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2003-03-11 19:08:31 +00:00
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unsigned desiredReceiveBufferSize;
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2002-08-05 00:39:07 +00:00
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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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2007-09-25 22:34:01 +00:00
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if (audiofound) {
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fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName());
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continue;
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}
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2003-03-11 19:08:31 +00:00
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desiredReceiveBufferSize = 100000;
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2002-08-05 00:39:07 +00:00
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} else if (strcmp(subsession->mediumName(), "video") == 0) {
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2007-12-02 18:04:14 +00:00
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if (videofound) {
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fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName());
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continue;
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}
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2003-03-11 19:08:31 +00:00
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desiredReceiveBufferSize = 2000000;
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2002-08-05 00:39:07 +00:00
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} else {
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continue;
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}
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2006-02-19 13:27:27 +00:00
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if (rtsp_port)
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subsession->setClientPortNum (rtsp_port);
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2009-05-13 02:58:57 +00:00
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2002-08-05 00:39:07 +00:00
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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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2006-02-19 13:27:27 +00:00
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fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum());
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2002-08-05 00:39:07 +00:00
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2003-03-11 19:08:31 +00:00
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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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2002-08-05 00:39:07 +00:00
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if (rtspClient != NULL) {
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2003-03-11 19:08:31 +00:00
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// Issue a RTSP "SETUP" command on the chosen subsession:
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2002-10-08 05:46:23 +00:00
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if (!rtspClient->setupMediaSubsession(*subsession, False,
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2009-05-25 20:28:35 +00:00
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rtsp_transport_tcp)) break;
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2007-09-25 22:34:01 +00:00
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if (!strcmp(subsession->mediumName(), "audio"))
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audiofound = 1;
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2007-12-02 18:04:14 +00:00
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if (!strcmp(subsession->mediumName(), "video"))
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videofound = 1;
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2002-08-05 00:39:07 +00:00
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}
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2003-03-11 19:08:31 +00:00
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}
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}
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2002-08-05 00:39:07 +00:00
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2003-03-11 19:08:31 +00:00
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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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2003-05-03 06:13:11 +00:00
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} else if (sipClient != NULL) {
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sipClient->sendACK(); // to start the stream flowing
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2003-03-11 19:08:31 +00:00
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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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2002-08-05 00:39:07 +00:00
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}
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2003-03-11 19:08:31 +00:00
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rtpState->flags |= flags;
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2002-08-05 00:39:07 +00:00
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}
|
2004-03-19 10:15:41 +00:00
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success = True;
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2002-08-05 00:39:07 +00:00
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} while (0);
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2004-03-19 10:15:41 +00:00
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if (!success) return NULL; // an error occurred
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2004-02-22 06:20:08 +00:00
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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);
|
2004-06-02 06:48:25 +00:00
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demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN,
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audio_id, video_id, dvdsub_id, NULL);
|
2004-02-22 06:20:08 +00:00
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|
|
demuxer = new_demuxers_demuxer(od, od, od);
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}
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|
|
|
|
|
|
return demuxer;
|
2002-08-05 00:39:07 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
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);
|
2003-02-03 10:27:50 +00:00
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
|
2002-08-05 00:39:07 +00:00
|
|
|
}
|
|
|
|
|
2003-07-27 10:15:10 +00:00
|
|
|
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;
|
|
|
|
}
|
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
|
|
|
|
Boolean mustGetNewData,
|
|
|
|
float& ptsBehind); // forward
|
2002-08-05 00:39:07 +00:00
|
|
|
|
|
|
|
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.
|
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
while (1) {
|
|
|
|
float ptsBehind;
|
|
|
|
demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
|
|
|
|
if (dp == NULL) return 0;
|
2002-08-05 00:39:07 +00:00
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
if (demuxer->stream->eof) return 0; // source stream has closed down
|
2009-05-13 02:58:57 +00:00
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
// 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.)
|
2004-03-02 08:52:59 +00:00
|
|
|
// (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.)
|
2003-03-11 19:08:31 +00:00
|
|
|
const float ptsBehindThreshold = 1.0; // seconds
|
2004-03-02 08:52:59 +00:00
|
|
|
const float ptsBehindLimit = 60.0; // seconds
|
|
|
|
if (ptsBehind < ptsBehindThreshold ||
|
|
|
|
ptsBehind > ptsBehindLimit ||
|
2009-05-25 20:28:35 +00:00
|
|
|
rtsp_transport_tcp) { // packet's OK
|
2003-03-11 19:08:31 +00:00
|
|
|
ds_add_packet(ds, dp);
|
|
|
|
break;
|
|
|
|
}
|
2009-05-13 02:58:57 +00:00
|
|
|
|
2003-04-02 01:38:07 +00:00
|
|
|
#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
|
2003-03-11 19:08:31 +00:00
|
|
|
free_demux_packet(dp); // give back this packet, and get another one
|
|
|
|
}
|
2003-02-03 10:27:50 +00:00
|
|
|
|
2002-08-05 00:39:07 +00:00
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
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;
|
|
|
|
}
|
|
|
|
|
2003-05-03 06:13:11 +00:00
|
|
|
static void teardownRTSPorSIPSession(RTPState* rtpState); // forward
|
2003-03-11 19:08:31 +00:00
|
|
|
|
2002-08-05 00:39:07 +00:00
|
|
|
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);
|
2002-08-29 20:19:33 +00:00
|
|
|
if (rtpState == NULL) return;
|
2003-03-11 19:08:31 +00:00
|
|
|
|
2003-05-03 06:13:11 +00:00
|
|
|
teardownRTSPorSIPSession(rtpState);
|
2003-03-11 19:08:31 +00:00
|
|
|
|
2002-08-05 00:39:07 +00:00
|
|
|
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);
|
2003-05-03 06:13:11 +00:00
|
|
|
Medium::close(rtpState->sipClient);
|
2002-08-05 00:39:07 +00:00
|
|
|
delete rtpState->audioBufferQueue;
|
|
|
|
delete rtpState->videoBufferQueue;
|
|
|
|
delete rtpState->sdpDescription;
|
|
|
|
delete rtpState;
|
2009-08-11 18:28:26 +00:00
|
|
|
#ifdef CONFIG_LIBAVCODEC
|
|
|
|
av_free(avcctx);
|
|
|
|
#endif
|
2002-08-05 00:39:07 +00:00
|
|
|
|
2004-07-20 02:12:08 +00:00
|
|
|
env->reclaim(); delete scheduler;
|
2002-08-05 00:39:07 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
////////// Extra routines that help implement the above interface functions:
|
|
|
|
|
2008-10-22 16:08:26 +00:00
|
|
|
#define MAX_RTP_FRAME_SIZE 5000000
|
2003-03-11 19:08:31 +00:00
|
|
|
// >= the largest conceivable frame composed from one or more RTP packets
|
2002-08-05 00:39:07 +00:00
|
|
|
|
|
|
|
static void afterReading(void* clientData, unsigned frameSize,
|
2004-01-06 09:30:27 +00:00
|
|
|
unsigned /*numTruncatedBytes*/,
|
|
|
|
struct timeval presentationTime,
|
|
|
|
unsigned /*durationInMicroseconds*/) {
|
2007-03-06 22:53:52 +00:00
|
|
|
int headersize = 0;
|
2003-03-11 19:08:31 +00:00
|
|
|
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;
|
2002-08-05 00:39:07 +00:00
|
|
|
demuxer_t* demuxer = bufferQueue->ourDemuxer();
|
2002-09-28 18:51:44 +00:00
|
|
|
RTPState* rtpState = (RTPState*)(demuxer->priv);
|
2002-08-05 00:39:07 +00:00
|
|
|
|
|
|
|
if (frameSize > 0) demuxer->stream->eof = 0;
|
|
|
|
|
2007-04-01 03:00:47 +00:00
|
|
|
demux_packet_t* dp = bufferQueue->dp;
|
|
|
|
|
2007-03-06 22:53:52 +00:00
|
|
|
if (bufferQueue->readSource()->isAMRAudioSource())
|
|
|
|
headersize = 1;
|
2007-04-01 03:03:05 +00:00
|
|
|
else if (bufferQueue == rtpState->videoBufferQueue &&
|
|
|
|
((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) {
|
|
|
|
dp->buffer[0]=0x00;
|
|
|
|
dp->buffer[1]=0x00;
|
|
|
|
dp->buffer[2]=0x01;
|
|
|
|
headersize = 3;
|
|
|
|
}
|
2007-03-06 22:53:52 +00:00
|
|
|
|
|
|
|
resize_demux_packet(dp, frameSize + headersize);
|
2002-09-28 18:51:44 +00:00
|
|
|
|
|
|
|
// Set the packet's presentation time stamp, depending on whether or
|
2009-05-13 02:58:57 +00:00
|
|
|
// not our RTP source's timestamps have been synchronized yet:
|
2003-03-11 19:08:31 +00:00
|
|
|
Boolean hasBeenSynchronized
|
|
|
|
= bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
|
|
|
|
if (hasBeenSynchronized) {
|
|
|
|
if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
|
|
|
|
fprintf(stderr, "%s stream has been synchronized using RTCP \n",
|
|
|
|
bufferQueue->tag());
|
|
|
|
}
|
2002-09-28 18:51:44 +00:00
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
|
|
|
|
if (fst->tv_sec == 0 && fst->tv_usec == 0) {
|
|
|
|
*fst = presentationTime;
|
2002-09-28 18:51:44 +00:00
|
|
|
}
|
2009-05-13 02:58:57 +00:00
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
// 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;
|
2002-09-28 18:51:44 +00:00
|
|
|
}
|
2003-03-11 19:08:31 +00:00
|
|
|
bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized;
|
2002-09-28 18:51:44 +00:00
|
|
|
|
2002-08-05 00:39:07 +00:00
|
|
|
dp->pos = demuxer->filepos;
|
2007-03-22 10:14:32 +00:00
|
|
|
demuxer->filepos += frameSize + headersize;
|
2002-08-05 00:39:07 +00:00
|
|
|
|
2003-02-03 10:27:50 +00:00
|
|
|
// Signal any pending 'doEventLoop()' call on this queue:
|
2002-08-05 00:39:07 +00:00
|
|
|
bufferQueue->blockingFlag = ~0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void onSourceClosure(void* clientData) {
|
2003-03-11 19:08:31 +00:00
|
|
|
ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
|
2002-08-05 00:39:07 +00:00
|
|
|
demuxer_t* demuxer = bufferQueue->ourDemuxer();
|
|
|
|
|
|
|
|
demuxer->stream->eof = 1;
|
|
|
|
|
2003-02-03 10:27:50 +00:00
|
|
|
// Signal any pending 'doEventLoop()' call on this queue:
|
2002-08-05 00:39:07 +00:00
|
|
|
bufferQueue->blockingFlag = ~0;
|
|
|
|
}
|
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
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;
|
2007-03-12 22:13:48 +00:00
|
|
|
int headersize = 0;
|
2007-06-18 08:53:53 +00:00
|
|
|
TaskToken task;
|
|
|
|
|
|
|
|
if (demuxer->stream->eof) return NULL;
|
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
if (ds == demuxer->video) {
|
|
|
|
bufferQueue = rtpState->videoBufferQueue;
|
2007-04-01 03:03:05 +00:00
|
|
|
if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4'))
|
|
|
|
headersize = 3;
|
2003-03-11 19:08:31 +00:00
|
|
|
} else if (ds == demuxer->audio) {
|
|
|
|
bufferQueue = rtpState->audioBufferQueue;
|
2007-03-06 22:53:52 +00:00
|
|
|
if (bufferQueue->readSource()->isAMRAudioSource())
|
2007-03-12 22:13:48 +00:00
|
|
|
headersize = 1;
|
2003-03-11 19:08:31 +00:00
|
|
|
} else {
|
|
|
|
fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n");
|
|
|
|
return NULL;
|
|
|
|
}
|
2003-02-03 10:27:50 +00:00
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
|
|
|
|
fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
|
|
|
|
return NULL;
|
|
|
|
}
|
2009-05-13 02:58:57 +00:00
|
|
|
|
2007-06-18 09:15:32 +00:00
|
|
|
demux_packet_t* dp = NULL;
|
2003-03-11 19:08:31 +00:00
|
|
|
if (!mustGetNewData) {
|
|
|
|
// Check whether we have a previously-saved buffer that we can use:
|
|
|
|
dp = bufferQueue->getPendingBuffer();
|
2003-04-12 09:30:19 +00:00
|
|
|
if (dp != NULL) {
|
|
|
|
ptsBehind = 0.0; // so that we always accept this data
|
|
|
|
return dp;
|
|
|
|
}
|
2003-03-11 19:08:31 +00:00
|
|
|
}
|
2003-02-03 10:27:50 +00:00
|
|
|
|
2003-03-11 19:08:31 +00:00
|
|
|
// Allocate a new packet buffer, and arrange to read into it:
|
2007-04-01 03:03:05 +00:00
|
|
|
if (!bufferQueue->nextpacket) {
|
2003-03-11 19:08:31 +00:00
|
|
|
dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
|
|
|
|
bufferQueue->dp = dp;
|
|
|
|
if (dp == NULL) return NULL;
|
2007-04-01 03:03:05 +00:00
|
|
|
}
|
2003-02-03 10:27:50 +00:00
|
|
|
|
2008-07-30 12:01:30 +00:00
|
|
|
#ifdef CONFIG_LIBAVCODEC
|
2007-04-01 03:03:05 +00:00
|
|
|
extern AVCodecParserContext * h264parserctx;
|
|
|
|
int consumed, poutbuf_size = 1;
|
2007-05-07 08:13:22 +00:00
|
|
|
const uint8_t *poutbuf = NULL;
|
2007-06-18 09:15:32 +00:00
|
|
|
float lastpts = 0.0;
|
2007-04-01 03:03:05 +00:00
|
|
|
|
|
|
|
do {
|
|
|
|
if (!bufferQueue->nextpacket) {
|
|
|
|
#endif
|
2003-03-11 19:08:31 +00:00
|
|
|
// Schedule the read operation:
|
2003-02-03 10:27:50 +00:00
|
|
|
bufferQueue->blockingFlag = 0;
|
2007-03-12 22:13:48 +00:00
|
|
|
bufferQueue->readSource()->getNextFrame(&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize,
|
2003-03-11 19:08:31 +00:00
|
|
|
afterReading, bufferQueue,
|
|
|
|
onSourceClosure, bufferQueue);
|
|
|
|
// Block ourselves until data becomes available:
|
|
|
|
TaskScheduler& scheduler
|
|
|
|
= bufferQueue->readSource()->envir().taskScheduler();
|
2007-06-20 10:39:57 +00:00
|
|
|
int delay = 10000000;
|
|
|
|
if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime())
|
|
|
|
delay /= 10;
|
2007-06-18 08:53:53 +00:00
|
|
|
task = scheduler.scheduleDelayedTask(delay, onSourceClosure, bufferQueue);
|
2003-03-11 19:08:31 +00:00
|
|
|
scheduler.doEventLoop(&bufferQueue->blockingFlag);
|
2007-06-18 08:53:53 +00:00
|
|
|
scheduler.unscheduleDelayedTask(task);
|
|
|
|
if (demuxer->stream->eof) {
|
|
|
|
free_demux_packet(dp);
|
|
|
|
return NULL;
|
|
|
|
}
|
2003-03-11 19:08:31 +00:00
|
|
|
|
2007-03-12 22:13:48 +00:00
|
|
|
if (headersize == 1) // amr
|
2007-03-06 22:53:52 +00:00
|
|
|
dp->buffer[0] =
|
|
|
|
((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader();
|
2008-07-30 12:01:30 +00:00
|
|
|
#ifdef CONFIG_LIBAVCODEC
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2007-04-01 03:03:05 +00:00
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} else {
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bufferQueue->dp = dp = bufferQueue->nextpacket;
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bufferQueue->nextpacket = NULL;
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}
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if (headersize == 3 && h264parserctx) { // h264
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consumed = h264parserctx->parser->parser_parse(h264parserctx,
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2009-08-11 18:52:18 +00:00
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avcctx,
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2007-04-01 03:03:05 +00:00
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&poutbuf, &poutbuf_size,
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dp->buffer, dp->len);
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if (!consumed && !poutbuf_size)
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return NULL;
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if (!poutbuf_size) {
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lastpts=dp->pts;
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free_demux_packet(dp);
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bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
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} else {
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bufferQueue->nextpacket = dp;
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bufferQueue->dp = dp = new_demux_packet(poutbuf_size);
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memcpy(dp->buffer, poutbuf, poutbuf_size);
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dp->pts=lastpts;
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}
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}
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} while (!poutbuf_size);
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#endif
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2007-03-06 22:53:52 +00:00
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2003-03-11 19:08:31 +00:00
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// Set the "ptsBehind" result parameter:
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2003-09-24 08:41:57 +00:00
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if (bufferQueue->prevPacketPTS != 0.0
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&& bufferQueue->prevPacketWasSynchronized
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&& *(bufferQueue->otherQueue) != NULL
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&& (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0
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&& (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) {
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2003-03-11 19:08:31 +00:00
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ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS
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- bufferQueue->prevPacketPTS;
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} else {
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ptsBehind = 0.0;
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2003-02-03 10:27:50 +00:00
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}
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2003-03-11 19:08:31 +00:00
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if (mustGetNewData) {
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// Save this buffer for future reads:
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bufferQueue->savePendingBuffer(dp);
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}
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2002-08-05 00:39:07 +00:00
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2003-03-11 19:08:31 +00:00
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return dp;
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2002-08-05 00:39:07 +00:00
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}
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2003-05-03 06:13:11 +00:00
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static void teardownRTSPorSIPSession(RTPState* rtpState) {
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2003-03-11 19:08:31 +00:00
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MediaSession* mediaSession = rtpState->mediaSession;
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2003-05-03 06:13:11 +00:00
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if (mediaSession == NULL) return;
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if (rtpState->rtspClient != NULL) {
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2007-04-03 17:00:27 +00:00
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rtpState->rtspClient->teardownMediaSession(*mediaSession);
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2003-05-03 06:13:11 +00:00
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} else if (rtpState->sipClient != NULL) {
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rtpState->sipClient->sendBYE();
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2003-03-11 19:08:31 +00:00
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}
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2002-08-05 00:39:07 +00:00
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}
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2003-03-11 19:08:31 +00:00
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////////// "ReadBuffer" and "ReadBufferQueue" implementation:
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2002-08-05 00:39:07 +00:00
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ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession,
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demuxer_t* demuxer, char const* tag)
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2003-03-11 19:08:31 +00:00
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: prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL),
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2007-06-18 09:15:32 +00:00
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dp(NULL), nextpacket(NULL),
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pendingDPHead(NULL), pendingDPTail(NULL),
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2002-08-05 00:39:07 +00:00
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fReadSource(subsession == NULL ? NULL : subsession->readSource()),
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2002-09-28 18:51:44 +00:00
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fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
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2002-08-05 00:39:07 +00:00
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fOurDemuxer(demuxer), fTag(strdup(tag)) {
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2009-05-13 02:58:57 +00:00
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}
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2002-08-05 00:39:07 +00:00
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ReadBufferQueue::~ReadBufferQueue() {
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delete fTag;
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2003-03-11 19:08:31 +00:00
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// Free any pending buffers (that never got delivered):
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demux_packet_t* dp = pendingDPHead;
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while (dp != NULL) {
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demux_packet_t* dpNext = dp->next;
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dp->next = NULL;
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free_demux_packet(dp);
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dp = dpNext;
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}
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2002-08-05 00:39:07 +00:00
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}
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2003-03-11 19:08:31 +00:00
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void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) {
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// Keep this buffer around, until MPlayer asks for it later:
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if (pendingDPTail == NULL) {
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pendingDPHead = pendingDPTail = dp;
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} else {
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pendingDPTail->next = dp;
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pendingDPTail = dp;
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2002-08-05 00:39:07 +00:00
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}
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2003-03-11 19:08:31 +00:00
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dp->next = NULL;
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}
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demux_packet_t* ReadBufferQueue::getPendingBuffer() {
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demux_packet_t* dp = pendingDPHead;
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if (dp != NULL) {
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pendingDPHead = dp->next;
|
2009-05-13 02:58:57 +00:00
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if (pendingDPHead == NULL) pendingDPTail = NULL;
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2003-03-11 19:08:31 +00:00
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dp->next = NULL;
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}
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return dp;
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2002-08-05 00:39:07 +00:00
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}
|
2005-08-05 19:57:47 +00:00
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2007-02-27 22:25:00 +00:00
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static int demux_rtp_control(struct demuxer_st *demuxer, int cmd, void *arg) {
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double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime();
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switch(cmd) {
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case DEMUXER_CTRL_GET_TIME_LENGTH:
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if (endpts <= 0)
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return DEMUXER_CTRL_DONTKNOW;
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*((double *)arg) = endpts;
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return DEMUXER_CTRL_OK;
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case DEMUXER_CTRL_GET_PERCENT_POS:
|
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|
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if (endpts <= 0)
|
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return DEMUXER_CTRL_DONTKNOW;
|
|
|
|
*((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts);
|
|
|
|
return DEMUXER_CTRL_OK;
|
|
|
|
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|
|
|
default:
|
|
|
|
return DEMUXER_CTRL_NOTIMPL;
|
|
|
|
}
|
|
|
|
}
|
2005-08-05 19:57:47 +00:00
|
|
|
|
|
|
|
demuxer_desc_t demuxer_desc_rtp = {
|
2005-09-23 22:35:04 +00:00
|
|
|
"LIVE555 RTP demuxer",
|
2007-12-04 15:26:56 +00:00
|
|
|
"live555",
|
2005-08-05 19:57:47 +00:00
|
|
|
"",
|
|
|
|
"Ross Finlayson",
|
2005-09-23 22:35:04 +00:00
|
|
|
"requires LIVE555 Streaming Media library",
|
2005-08-05 19:57:47 +00:00
|
|
|
DEMUXER_TYPE_RTP,
|
|
|
|
0, // no autodetect
|
|
|
|
NULL,
|
|
|
|
demux_rtp_fill_buffer,
|
|
|
|
demux_open_rtp,
|
|
|
|
demux_close_rtp,
|
|
|
|
NULL,
|
2007-02-27 22:25:00 +00:00
|
|
|
demux_rtp_control
|
2005-08-05 19:57:47 +00:00
|
|
|
};
|