2 * routines (with C-linkage) that interface between MPlayer
3 * and the "LIVE555 Streaming Media" libraries
5 * This file is part of MPlayer.
7 * MPlayer is free software; you can redistribute it and/or modify
8 * it under the terms of the GNU General Public License as published by
9 * the Free Software Foundation; either version 2 of the License, or
10 * (at your option) any later version.
12 * MPlayer is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 * GNU General Public License for more details.
17 * You should have received a copy of the GNU General Public License along
18 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
19 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
23 // on MinGW, we must include windows.h before the things it conflicts
24 #ifdef __MINGW32__ // with. they are each protected from
25 #include <windows.h> // windows.h, but not the other way around.
27 #include "demux_rtp.h"
30 #include "demux_rtp_internal.h"
32 #include "BasicUsageEnvironment.hh"
33 #include "liveMedia.hh"
34 #include "GroupsockHelper.hh"
37 // A data structure representing input data for each stream:
38 class ReadBufferQueue
{
40 ReadBufferQueue(MediaSubsession
* subsession
, demuxer_t
* demuxer
,
42 virtual ~ReadBufferQueue();
44 FramedSource
* readSource() const { return fReadSource
; }
45 RTPSource
* rtpSource() const { return fRTPSource
; }
46 demuxer_t
* ourDemuxer() const { return fOurDemuxer
; }
47 char const* tag() const { return fTag
; }
49 char blockingFlag
; // used to implement synchronous reads
51 // For A/V synchronization:
52 Boolean prevPacketWasSynchronized
;
54 ReadBufferQueue
** otherQueue
;
56 // The 'queue' actually consists of just a single "demux_packet_t"
57 // (because the underlying OS does the actual queueing/buffering):
60 // However, we sometimes inspect buffers before delivering them.
61 // For this, we maintain a queue of pending buffers:
62 void savePendingBuffer(demux_packet_t
* dp
);
63 demux_packet_t
* getPendingBuffer();
65 // For H264 over rtsp using AVParser, the next packet has to be saved
66 demux_packet_t
* nextpacket
;
69 demux_packet_t
* pendingDPHead
;
70 demux_packet_t
* pendingDPTail
;
72 FramedSource
* fReadSource
;
73 RTPSource
* fRTPSource
;
74 demuxer_t
* fOurDemuxer
;
75 char const* fTag
; // used for debugging
78 // A structure of RTP-specific state, kept so that we can cleanly
80 typedef struct RTPState
{
81 char const* sdpDescription
;
82 RTSPClient
* rtspClient
;
84 MediaSession
* mediaSession
;
85 ReadBufferQueue
* audioBufferQueue
;
86 ReadBufferQueue
* videoBufferQueue
;
88 struct timeval firstSyncTime
;
91 extern "C" char* network_username
;
92 extern "C" char* network_password
;
93 static char* openURL_rtsp(RTSPClient
* client
, char const* url
) {
94 // If we were given a user name (and optional password), then use them:
95 if (network_username
!= NULL
) {
96 char const* password
= network_password
== NULL
? "" : network_password
;
97 return client
->describeWithPassword(url
, network_username
, password
);
99 return client
->describeURL(url
);
103 static char* openURL_sip(SIPClient
* client
, char const* url
) {
104 // If we were given a user name (and optional password), then use them:
105 if (network_username
!= NULL
) {
106 char const* password
= network_password
== NULL
? "" : network_password
;
107 return client
->inviteWithPassword(url
, network_username
, password
);
109 return client
->invite(url
);
113 #ifdef CONFIG_LIBNEMESI
114 extern int rtsp_transport_tcp
;
116 int rtsp_transport_tcp
= 0;
119 extern int rtsp_port
;
120 #ifdef CONFIG_LIBAVCODEC
121 extern AVCodecContext
*avcctx
;
124 extern "C" int audio_id
, video_id
, dvdsub_id
;
125 extern "C" demuxer_t
* demux_open_rtp(demuxer_t
* demuxer
) {
126 Boolean success
= False
;
128 TaskScheduler
* scheduler
= BasicTaskScheduler::createNew();
129 if (scheduler
== NULL
) break;
130 UsageEnvironment
* env
= BasicUsageEnvironment::createNew(*scheduler
);
131 if (env
== NULL
) break;
133 RTSPClient
* rtspClient
= NULL
;
134 SIPClient
* sipClient
= NULL
;
136 if (demuxer
== NULL
|| demuxer
->stream
== NULL
) break; // shouldn't happen
137 demuxer
->stream
->eof
= 0; // just in case
139 // Look at the stream's 'priv' field to see if we were initiated
140 // via a SDP description:
141 char* sdpDescription
= (char*)(demuxer
->stream
->priv
);
142 if (sdpDescription
== NULL
) {
143 // We weren't given a SDP description directly, so assume that
144 // we were given a RTSP or SIP URL:
145 char const* protocol
= demuxer
->stream
->streaming_ctrl
->url
->protocol
;
146 char const* url
= demuxer
->stream
->streaming_ctrl
->url
->url
;
148 if (strcmp(protocol
, "rtsp") == 0) {
149 rtspClient
= RTSPClient::createNew(*env
, verbose
, "MPlayer");
150 if (rtspClient
== NULL
) {
151 fprintf(stderr
, "Failed to create RTSP client: %s\n",
152 env
->getResultMsg());
155 sdpDescription
= openURL_rtsp(rtspClient
, url
);
157 unsigned char desiredAudioType
= 0; // PCMU (use 3 for GSM)
158 sipClient
= SIPClient::createNew(*env
, desiredAudioType
, NULL
,
160 if (sipClient
== NULL
) {
161 fprintf(stderr
, "Failed to create SIP client: %s\n",
162 env
->getResultMsg());
165 sipClient
->setClientStartPortNum(8000);
166 sdpDescription
= openURL_sip(sipClient
, url
);
169 if (sdpDescription
== NULL
) {
170 fprintf(stderr
, "Failed to get a SDP description from URL \"%s\": %s\n",
171 url
, env
->getResultMsg());
176 // Now that we have a SDP description, create a MediaSession from it:
177 MediaSession
* mediaSession
= MediaSession::createNew(*env
, sdpDescription
);
178 if (mediaSession
== NULL
) break;
181 // Create a 'RTPState' structure containing the state that we just created,
182 // and store it in the demuxer's 'priv' field, for future reference:
183 RTPState
* rtpState
= new RTPState
;
184 rtpState
->sdpDescription
= sdpDescription
;
185 rtpState
->rtspClient
= rtspClient
;
186 rtpState
->sipClient
= sipClient
;
187 rtpState
->mediaSession
= mediaSession
;
188 rtpState
->audioBufferQueue
= rtpState
->videoBufferQueue
= NULL
;
190 rtpState
->firstSyncTime
.tv_sec
= rtpState
->firstSyncTime
.tv_usec
= 0;
191 demuxer
->priv
= rtpState
;
193 int audiofound
= 0, videofound
= 0;
194 // Create RTP receivers (sources) for each subsession:
195 MediaSubsessionIterator
iter(*mediaSession
);
196 MediaSubsession
* subsession
;
197 unsigned desiredReceiveBufferSize
;
198 while ((subsession
= iter
.next()) != NULL
) {
199 // Ignore any subsession that's not audio or video:
200 if (strcmp(subsession
->mediumName(), "audio") == 0) {
202 fprintf(stderr
, "Additional subsession \"audio/%s\" skipped\n", subsession
->codecName());
205 desiredReceiveBufferSize
= 100000;
206 } else if (strcmp(subsession
->mediumName(), "video") == 0) {
208 fprintf(stderr
, "Additional subsession \"video/%s\" skipped\n", subsession
->codecName());
211 desiredReceiveBufferSize
= 2000000;
217 subsession
->setClientPortNum (rtsp_port
);
219 if (!subsession
->initiate()) {
220 fprintf(stderr
, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession
->mediumName(), subsession
->codecName(), env
->getResultMsg());
222 fprintf(stderr
, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession
->mediumName(), subsession
->codecName(), subsession
->clientPortNum());
224 // Set the OS's socket receive buffer sufficiently large to avoid
225 // incoming packets getting dropped between successive reads from this
226 // subsession's demuxer. Depending on the bitrate(s) that you expect,
227 // you may wish to tweak the "desiredReceiveBufferSize" values above.
228 int rtpSocketNum
= subsession
->rtpSource()->RTPgs()->socketNum();
229 int receiveBufferSize
230 = increaseReceiveBufferTo(*env
, rtpSocketNum
,
231 desiredReceiveBufferSize
);
233 fprintf(stderr
, "Increased %s socket receive buffer to %d bytes \n",
234 subsession
->mediumName(), receiveBufferSize
);
237 if (rtspClient
!= NULL
) {
238 // Issue a RTSP "SETUP" command on the chosen subsession:
239 if (!rtspClient
->setupMediaSubsession(*subsession
, False
,
240 rtsp_transport_tcp
)) break;
241 if (!strcmp(subsession
->mediumName(), "audio"))
243 if (!strcmp(subsession
->mediumName(), "video"))
249 if (rtspClient
!= NULL
) {
250 // Issue a RTSP aggregate "PLAY" command on the whole session:
251 if (!rtspClient
->playMediaSession(*mediaSession
)) break;
252 } else if (sipClient
!= NULL
) {
253 sipClient
->sendACK(); // to start the stream flowing
256 // Now that the session is ready to be read, do additional
257 // MPlayer codec-specific initialization on each subsession:
259 while ((subsession
= iter
.next()) != NULL
) {
260 if (subsession
->readSource() == NULL
) continue; // not reading this
263 if (strcmp(subsession
->mediumName(), "audio") == 0) {
264 rtpState
->audioBufferQueue
265 = new ReadBufferQueue(subsession
, demuxer
, "audio");
266 rtpState
->audioBufferQueue
->otherQueue
= &(rtpState
->videoBufferQueue
);
267 rtpCodecInitialize_audio(demuxer
, subsession
, flags
);
268 } else if (strcmp(subsession
->mediumName(), "video") == 0) {
269 rtpState
->videoBufferQueue
270 = new ReadBufferQueue(subsession
, demuxer
, "video");
271 rtpState
->videoBufferQueue
->otherQueue
= &(rtpState
->audioBufferQueue
);
272 rtpCodecInitialize_video(demuxer
, subsession
, flags
);
274 rtpState
->flags
|= flags
;
278 if (!success
) return NULL
; // an error occurred
280 // Hack: If audio and video are demuxed together on a single RTP stream,
281 // then create a new "demuxer_t" structure to allow the higher-level
282 // code to recognize this:
283 if (demux_is_multiplexed_rtp_stream(demuxer
)) {
284 stream_t
* s
= new_ds_stream(demuxer
->video
);
285 demuxer_t
* od
= demux_open(s
, DEMUXER_TYPE_UNKNOWN
,
286 audio_id
, video_id
, dvdsub_id
, NULL
);
287 demuxer
= new_demuxers_demuxer(od
, od
, od
);
293 extern "C" int demux_is_mpeg_rtp_stream(demuxer_t
* demuxer
) {
294 // Get the RTP state that was stored in the demuxer's 'priv' field:
295 RTPState
* rtpState
= (RTPState
*)(demuxer
->priv
);
297 return (rtpState
->flags
&RTPSTATE_IS_MPEG12_VIDEO
) != 0;
300 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t
* demuxer
) {
301 // Get the RTP state that was stored in the demuxer's 'priv' field:
302 RTPState
* rtpState
= (RTPState
*)(demuxer
->priv
);
304 return (rtpState
->flags
&RTPSTATE_IS_MULTIPLEXED
) != 0;
307 static demux_packet_t
* getBuffer(demuxer_t
* demuxer
, demux_stream_t
* ds
,
308 Boolean mustGetNewData
,
309 float& ptsBehind
); // forward
311 extern "C" int demux_rtp_fill_buffer(demuxer_t
* demuxer
, demux_stream_t
* ds
) {
312 // Get a filled-in "demux_packet" from the RTP source, and deliver it.
313 // Note that this is called as a synchronous read operation, so it needs
314 // to block in the (hopefully infrequent) case where no packet is
315 // immediately available.
319 demux_packet_t
* dp
= getBuffer(demuxer
, ds
, False
, ptsBehind
); // blocking
320 if (dp
== NULL
) return 0;
322 if (demuxer
->stream
->eof
) return 0; // source stream has closed down
324 // Before using this packet, check to make sure that its presentation
325 // time is not far behind the other stream (if any). If it is,
326 // then we discard this packet, and get another instead. (The rest of
327 // MPlayer doesn't always do a good job of synchronizing when the
328 // audio and video streams get this far apart.)
329 // (We don't do this when streaming over TCP, because then the audio and
330 // video streams are interleaved.)
331 // (Also, if the stream is *excessively* far behind, then we allow
332 // the packet, because in this case it probably means that there was
333 // an error in the source's timestamp synchronization.)
334 const float ptsBehindThreshold
= 1.0; // seconds
335 const float ptsBehindLimit
= 60.0; // seconds
336 if (ptsBehind
< ptsBehindThreshold
||
337 ptsBehind
> ptsBehindLimit
||
338 rtsp_transport_tcp
) { // packet's OK
339 ds_add_packet(ds
, dp
);
343 #ifdef DEBUG_PRINT_DISCARDED_PACKETS
344 RTPState
* rtpState
= (RTPState
*)(demuxer
->priv
);
345 ReadBufferQueue
* bufferQueue
= ds
== demuxer
->video
? rtpState
->videoBufferQueue
: rtpState
->audioBufferQueue
;
346 fprintf(stderr
, "Discarding %s packet (%fs behind)\n", bufferQueue
->tag(), ptsBehind
);
348 free_demux_packet(dp
); // give back this packet, and get another one
354 Boolean
awaitRTPPacket(demuxer_t
* demuxer
, demux_stream_t
* ds
,
355 unsigned char*& packetData
, unsigned& packetDataLen
,
357 // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
358 // is not delivered to the "demux_stream".
360 demux_packet_t
* dp
= getBuffer(demuxer
, ds
, True
, ptsBehind
); // blocking
361 if (dp
== NULL
) return False
;
363 packetData
= dp
->buffer
;
364 packetDataLen
= dp
->len
;
370 static void teardownRTSPorSIPSession(RTPState
* rtpState
); // forward
372 extern "C" void demux_close_rtp(demuxer_t
* demuxer
) {
373 // Reclaim all RTP-related state:
375 // Get the RTP state that was stored in the demuxer's 'priv' field:
376 RTPState
* rtpState
= (RTPState
*)(demuxer
->priv
);
377 if (rtpState
== NULL
) return;
379 teardownRTSPorSIPSession(rtpState
);
381 UsageEnvironment
* env
= NULL
;
382 TaskScheduler
* scheduler
= NULL
;
383 if (rtpState
->mediaSession
!= NULL
) {
384 env
= &(rtpState
->mediaSession
->envir());
385 scheduler
= &(env
->taskScheduler());
387 Medium::close(rtpState
->mediaSession
);
388 Medium::close(rtpState
->rtspClient
);
389 Medium::close(rtpState
->sipClient
);
390 delete rtpState
->audioBufferQueue
;
391 delete rtpState
->videoBufferQueue
;
392 delete[] rtpState
->sdpDescription
;
394 #ifdef CONFIG_LIBAVCODEC
398 env
->reclaim(); delete scheduler
;
401 ////////// Extra routines that help implement the above interface functions:
403 #define MAX_RTP_FRAME_SIZE 5000000
404 // >= the largest conceivable frame composed from one or more RTP packets
406 static void afterReading(void* clientData
, unsigned frameSize
,
407 unsigned /*numTruncatedBytes*/,
408 struct timeval presentationTime
,
409 unsigned /*durationInMicroseconds*/) {
411 if (frameSize
>= MAX_RTP_FRAME_SIZE
) {
412 fprintf(stderr
, "Saw an input frame too large (>=%d). Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
415 ReadBufferQueue
* bufferQueue
= (ReadBufferQueue
*)clientData
;
416 demuxer_t
* demuxer
= bufferQueue
->ourDemuxer();
417 RTPState
* rtpState
= (RTPState
*)(demuxer
->priv
);
419 if (frameSize
> 0) demuxer
->stream
->eof
= 0;
421 demux_packet_t
* dp
= bufferQueue
->dp
;
423 if (bufferQueue
->readSource()->isAMRAudioSource())
425 else if (bufferQueue
== rtpState
->videoBufferQueue
&&
426 ((sh_video_t
*)demuxer
->video
->sh
)->format
== mmioFOURCC('H','2','6','4')) {
433 resize_demux_packet(dp
, frameSize
+ headersize
);
435 // Set the packet's presentation time stamp, depending on whether or
436 // not our RTP source's timestamps have been synchronized yet:
437 Boolean hasBeenSynchronized
438 = bufferQueue
->rtpSource()->hasBeenSynchronizedUsingRTCP();
439 if (hasBeenSynchronized
) {
440 if (verbose
> 0 && !bufferQueue
->prevPacketWasSynchronized
) {
441 fprintf(stderr
, "%s stream has been synchronized using RTCP \n",
445 struct timeval
* fst
= &(rtpState
->firstSyncTime
); // abbrev
446 if (fst
->tv_sec
== 0 && fst
->tv_usec
== 0) {
447 *fst
= presentationTime
;
450 // For the "pts" field, use the time differential from the first
451 // synchronized time, rather than absolute time, in order to avoid
452 // round-off errors when converting to a float:
453 dp
->pts
= presentationTime
.tv_sec
- fst
->tv_sec
454 + (presentationTime
.tv_usec
- fst
->tv_usec
)/1000000.0;
455 bufferQueue
->prevPacketPTS
= dp
->pts
;
457 if (verbose
> 0 && bufferQueue
->prevPacketWasSynchronized
) {
458 fprintf(stderr
, "%s stream is no longer RTCP-synchronized \n",
462 // use the previous packet's "pts" once again:
463 dp
->pts
= bufferQueue
->prevPacketPTS
;
465 bufferQueue
->prevPacketWasSynchronized
= hasBeenSynchronized
;
467 dp
->pos
= demuxer
->filepos
;
468 demuxer
->filepos
+= frameSize
+ headersize
;
470 // Signal any pending 'doEventLoop()' call on this queue:
471 bufferQueue
->blockingFlag
= ~0;
474 static void onSourceClosure(void* clientData
) {
475 ReadBufferQueue
* bufferQueue
= (ReadBufferQueue
*)clientData
;
476 demuxer_t
* demuxer
= bufferQueue
->ourDemuxer();
478 demuxer
->stream
->eof
= 1;
480 // Signal any pending 'doEventLoop()' call on this queue:
481 bufferQueue
->blockingFlag
= ~0;
484 static demux_packet_t
* getBuffer(demuxer_t
* demuxer
, demux_stream_t
* ds
,
485 Boolean mustGetNewData
,
487 // Begin by finding the buffer queue that we want to read from:
488 // (Get this from the RTP state, which we stored in
489 // the demuxer's 'priv' field)
490 RTPState
* rtpState
= (RTPState
*)(demuxer
->priv
);
491 ReadBufferQueue
* bufferQueue
= NULL
;
495 if (demuxer
->stream
->eof
) return NULL
;
497 if (ds
== demuxer
->video
) {
498 bufferQueue
= rtpState
->videoBufferQueue
;
499 if (((sh_video_t
*)ds
->sh
)->format
== mmioFOURCC('H','2','6','4'))
501 } else if (ds
== demuxer
->audio
) {
502 bufferQueue
= rtpState
->audioBufferQueue
;
503 if (bufferQueue
->readSource()->isAMRAudioSource())
506 fprintf(stderr
, "(demux_rtp)getBuffer: internal error: unknown stream\n");
510 if (bufferQueue
== NULL
|| bufferQueue
->readSource() == NULL
) {
511 fprintf(stderr
, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
515 demux_packet_t
* dp
= NULL
;
516 if (!mustGetNewData
) {
517 // Check whether we have a previously-saved buffer that we can use:
518 dp
= bufferQueue
->getPendingBuffer();
520 ptsBehind
= 0.0; // so that we always accept this data
525 // Allocate a new packet buffer, and arrange to read into it:
526 if (!bufferQueue
->nextpacket
) {
527 dp
= new_demux_packet(MAX_RTP_FRAME_SIZE
);
528 bufferQueue
->dp
= dp
;
529 if (dp
== NULL
) return NULL
;
532 #ifdef CONFIG_LIBAVCODEC
533 extern AVCodecParserContext
* h264parserctx
;
534 int consumed
, poutbuf_size
= 1;
535 const uint8_t *poutbuf
= NULL
;
539 if (!bufferQueue
->nextpacket
) {
541 // Schedule the read operation:
542 bufferQueue
->blockingFlag
= 0;
543 bufferQueue
->readSource()->getNextFrame(&dp
->buffer
[headersize
], MAX_RTP_FRAME_SIZE
- headersize
,
544 afterReading
, bufferQueue
,
545 onSourceClosure
, bufferQueue
);
546 // Block ourselves until data becomes available:
547 TaskScheduler
& scheduler
548 = bufferQueue
->readSource()->envir().taskScheduler();
549 int delay
= 10000000;
550 if (bufferQueue
->prevPacketPTS
* 1.05 > rtpState
->mediaSession
->playEndTime())
552 task
= scheduler
.scheduleDelayedTask(delay
, onSourceClosure
, bufferQueue
);
553 scheduler
.doEventLoop(&bufferQueue
->blockingFlag
);
554 scheduler
.unscheduleDelayedTask(task
);
555 if (demuxer
->stream
->eof
) {
556 free_demux_packet(dp
);
560 if (headersize
== 1) // amr
562 ((AMRAudioSource
*)bufferQueue
->readSource())->lastFrameHeader();
563 #ifdef CONFIG_LIBAVCODEC
565 bufferQueue
->dp
= dp
= bufferQueue
->nextpacket
;
566 bufferQueue
->nextpacket
= NULL
;
568 if (headersize
== 3 && h264parserctx
) { // h264
569 consumed
= h264parserctx
->parser
->parser_parse(h264parserctx
,
571 &poutbuf
, &poutbuf_size
,
572 dp
->buffer
, dp
->len
);
574 if (!consumed
&& !poutbuf_size
)
579 free_demux_packet(dp
);
580 bufferQueue
->dp
= dp
= new_demux_packet(MAX_RTP_FRAME_SIZE
);
582 bufferQueue
->nextpacket
= dp
;
583 bufferQueue
->dp
= dp
= new_demux_packet(poutbuf_size
);
584 memcpy(dp
->buffer
, poutbuf
, poutbuf_size
);
588 } while (!poutbuf_size
);
591 // Set the "ptsBehind" result parameter:
592 if (bufferQueue
->prevPacketPTS
!= 0.0
593 && bufferQueue
->prevPacketWasSynchronized
594 && *(bufferQueue
->otherQueue
) != NULL
595 && (*(bufferQueue
->otherQueue
))->prevPacketPTS
!= 0.0
596 && (*(bufferQueue
->otherQueue
))->prevPacketWasSynchronized
) {
597 ptsBehind
= (*(bufferQueue
->otherQueue
))->prevPacketPTS
598 - bufferQueue
->prevPacketPTS
;
603 if (mustGetNewData
) {
604 // Save this buffer for future reads:
605 bufferQueue
->savePendingBuffer(dp
);
611 static void teardownRTSPorSIPSession(RTPState
* rtpState
) {
612 MediaSession
* mediaSession
= rtpState
->mediaSession
;
613 if (mediaSession
== NULL
) return;
614 if (rtpState
->rtspClient
!= NULL
) {
615 rtpState
->rtspClient
->teardownMediaSession(*mediaSession
);
616 } else if (rtpState
->sipClient
!= NULL
) {
617 rtpState
->sipClient
->sendBYE();
621 ////////// "ReadBuffer" and "ReadBufferQueue" implementation:
623 ReadBufferQueue::ReadBufferQueue(MediaSubsession
* subsession
,
624 demuxer_t
* demuxer
, char const* tag
)
625 : prevPacketWasSynchronized(False
), prevPacketPTS(0.0), otherQueue(NULL
),
626 dp(NULL
), nextpacket(NULL
),
627 pendingDPHead(NULL
), pendingDPTail(NULL
),
628 fReadSource(subsession
== NULL
? NULL
: subsession
->readSource()),
629 fRTPSource(subsession
== NULL
? NULL
: subsession
->rtpSource()),
630 fOurDemuxer(demuxer
), fTag(strdup(tag
)) {
633 ReadBufferQueue::~ReadBufferQueue() {
636 // Free any pending buffers (that never got delivered):
637 demux_packet_t
* dp
= pendingDPHead
;
639 demux_packet_t
* dpNext
= dp
->next
;
641 free_demux_packet(dp
);
646 void ReadBufferQueue::savePendingBuffer(demux_packet_t
* dp
) {
647 // Keep this buffer around, until MPlayer asks for it later:
648 if (pendingDPTail
== NULL
) {
649 pendingDPHead
= pendingDPTail
= dp
;
651 pendingDPTail
->next
= dp
;
657 demux_packet_t
* ReadBufferQueue::getPendingBuffer() {
658 demux_packet_t
* dp
= pendingDPHead
;
660 pendingDPHead
= dp
->next
;
661 if (pendingDPHead
== NULL
) pendingDPTail
= NULL
;
669 static int demux_rtp_control(struct demuxer_st
*demuxer
, int cmd
, void *arg
) {
670 double endpts
= ((RTPState
*)demuxer
->priv
)->mediaSession
->playEndTime();
673 case DEMUXER_CTRL_GET_TIME_LENGTH
:
675 return DEMUXER_CTRL_DONTKNOW
;
676 *((double *)arg
) = endpts
;
677 return DEMUXER_CTRL_OK
;
679 case DEMUXER_CTRL_GET_PERCENT_POS
:
681 return DEMUXER_CTRL_DONTKNOW
;
682 *((int *)arg
) = (int)(((RTPState
*)demuxer
->priv
)->videoBufferQueue
->prevPacketPTS
*100/endpts
);
683 return DEMUXER_CTRL_OK
;
686 return DEMUXER_CTRL_NOTIMPL
;
690 demuxer_desc_t demuxer_desc_rtp
= {
691 "LIVE555 RTP demuxer",
695 "requires LIVE555 Streaming Media library",
699 demux_rtp_fill_buffer
,