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mpv/libmpdemux/demux_rtp.cpp

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/*
* routines (with C-linkage) that interface between MPlayer
* and the "LIVE555 Streaming Media" libraries
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#define RTSPCLIENT_SYNCHRONOUS_INTERFACE 1
extern "C" {
// on MinGW, we must include windows.h before the things it conflicts
#ifdef __MINGW32__ // with. they are each protected from
#include <windows.h> // windows.h, but not the other way around.
#endif
#include "demux_rtp.h"
#include "stream/stream.h"
#include "stheader.h"
#include "options.h"
#include "config.h"
}
#include "demux_rtp_internal.h"
#include "BasicUsageEnvironment.hh"
#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include <unistd.h>
// A data structure representing input data for each stream:
class ReadBufferQueue {
public:
ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
char const* tag);
virtual ~ReadBufferQueue();
FramedSource* readSource() const { return fReadSource; }
RTPSource* rtpSource() const { return fRTPSource; }
demuxer_t* ourDemuxer() const { return fOurDemuxer; }
char const* tag() const { return fTag; }
char blockingFlag; // used to implement synchronous reads
// For A/V synchronization:
Boolean prevPacketWasSynchronized;
float prevPacketPTS;
ReadBufferQueue** otherQueue;
// The 'queue' actually consists of just a single "demux_packet_t"
// (because the underlying OS does the actual queueing/buffering):
demux_packet_t* dp;
// However, we sometimes inspect buffers before delivering them.
// For this, we maintain a queue of pending buffers:
void savePendingBuffer(demux_packet_t* dp);
demux_packet_t* getPendingBuffer();
// For H264 over rtsp using AVParser, the next packet has to be saved
demux_packet_t* nextpacket;
private:
demux_packet_t* pendingDPHead;
demux_packet_t* pendingDPTail;
FramedSource* fReadSource;
RTPSource* fRTPSource;
demuxer_t* fOurDemuxer;
char const* fTag; // used for debugging
};
// A structure of RTP-specific state, kept so that we can cleanly
// reclaim it:
struct RTPState {
char const* sdpDescription;
RTSPClient* rtspClient;
SIPClient* sipClient;
MediaSession* mediaSession;
ReadBufferQueue* audioBufferQueue;
ReadBufferQueue* videoBufferQueue;
unsigned flags;
struct timeval firstSyncTime;
};
extern "C" char* network_username;
extern "C" char* network_password;
static char* openURL_rtsp(RTSPClient* client, char const* url) {
// If we were given a user name (and optional password), then use them:
if (network_username != NULL) {
char const* password = network_password == NULL ? "" : network_password;
return client->describeWithPassword(url, network_username, password);
} else {
return client->describeURL(url);
}
}
static char* openURL_sip(SIPClient* client, char const* url) {
// If we were given a user name (and optional password), then use them:
if (network_username != NULL) {
char const* password = network_password == NULL ? "" : network_password;
return client->inviteWithPassword(url, network_username, password);
} else {
return client->invite(url);
}
}
#ifdef CONFIG_LIBNEMESI
extern int rtsp_transport_tcp;
extern int rtsp_transport_http;
#else
int rtsp_transport_tcp = 0;
int rtsp_transport_http = 0;
#endif
extern int rtsp_port;
extern AVCodecContext *avcctx;
extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
struct MPOpts *opts = demuxer->opts;
Boolean success = False;
do {
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
if (scheduler == NULL) break;
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
if (env == NULL) break;
RTSPClient* rtspClient = NULL;
SIPClient* sipClient = NULL;
if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen
demuxer->stream->eof = 0; // just in case
// Look at the stream's 'priv' field to see if we were initiated
// via a SDP description:
char* sdpDescription = (char*)(demuxer->stream->priv);
if (sdpDescription == NULL) {
// We weren't given a SDP description directly, so assume that
// we were given a RTSP or SIP URL:
char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
char const* url = demuxer->stream->streaming_ctrl->url->url;
extern int verbose;
if (strcmp(protocol, "rtsp") == 0) {
if (rtsp_transport_http == 1) {
rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port;
rtsp_transport_tcp = 1;
}
rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http);
if (rtspClient == NULL) {
fprintf(stderr, "Failed to create RTSP client: %s\n",
env->getResultMsg());
break;
}
sdpDescription = openURL_rtsp(rtspClient, url);
} else { // SIP
unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
verbose, "MPlayer");
if (sipClient == NULL) {
fprintf(stderr, "Failed to create SIP client: %s\n",
env->getResultMsg());
break;
}
sipClient->setClientStartPortNum(8000);
sdpDescription = openURL_sip(sipClient, url);
}
if (sdpDescription == NULL) {
fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
url, env->getResultMsg());
break;
}
}
// Now that we have a SDP description, create a MediaSession from it:
MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
if (mediaSession == NULL) break;
// Create a 'RTPState' structure containing the state that we just created,
// and store it in the demuxer's 'priv' field, for future reference:
RTPState* rtpState = new RTPState;
rtpState->sdpDescription = sdpDescription;
rtpState->rtspClient = rtspClient;
rtpState->sipClient = sipClient;
rtpState->mediaSession = mediaSession;
rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
rtpState->flags = 0;
rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
demuxer->priv = rtpState;
int audiofound = 0, videofound = 0;
// Create RTP receivers (sources) for each subsession:
MediaSubsessionIterator iter(*mediaSession);
MediaSubsession* subsession;
unsigned desiredReceiveBufferSize;
while ((subsession = iter.next()) != NULL) {
// Ignore any subsession that's not audio or video:
if (strcmp(subsession->mediumName(), "audio") == 0) {
if (audiofound) {
fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName());
continue;
}
desiredReceiveBufferSize = 100000;
} else if (strcmp(subsession->mediumName(), "video") == 0) {
if (videofound) {
fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName());
continue;
}
desiredReceiveBufferSize = 2000000;
} else {
continue;
}
if (rtsp_port)
subsession->setClientPortNum (rtsp_port);
if (!subsession->initiate()) {
fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
} else {
fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum());
// Set the OS's socket receive buffer sufficiently large to avoid
// incoming packets getting dropped between successive reads from this
// subsession's demuxer. Depending on the bitrate(s) that you expect,
// you may wish to tweak the "desiredReceiveBufferSize" values above.
int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
int receiveBufferSize
= increaseReceiveBufferTo(*env, rtpSocketNum,
desiredReceiveBufferSize);
if (verbose > 0) {
fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
subsession->mediumName(), receiveBufferSize);
}
if (rtspClient != NULL) {
// Issue a RTSP "SETUP" command on the chosen subsession:
if (!rtspClient->setupMediaSubsession(*subsession, False,
rtsp_transport_tcp)) break;
if (!strcmp(subsession->mediumName(), "audio"))
audiofound = 1;
if (!strcmp(subsession->mediumName(), "video"))
videofound = 1;
}
}
}
if (rtspClient != NULL) {
// Issue a RTSP aggregate "PLAY" command on the whole session:
if (!rtspClient->playMediaSession(*mediaSession)) break;
} else if (sipClient != NULL) {
sipClient->sendACK(); // to start the stream flowing
}
// Now that the session is ready to be read, do additional
// MPlayer codec-specific initialization on each subsession:
iter.reset();
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // not reading this
unsigned flags = 0;
if (strcmp(subsession->mediumName(), "audio") == 0) {
rtpState->audioBufferQueue
= new ReadBufferQueue(subsession, demuxer, "audio");
rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
rtpCodecInitialize_audio(demuxer, subsession, flags);
} else if (strcmp(subsession->mediumName(), "video") == 0) {
rtpState->videoBufferQueue
= new ReadBufferQueue(subsession, demuxer, "video");
rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
rtpCodecInitialize_video(demuxer, subsession, flags);
}
rtpState->flags |= flags;
}
success = True;
} while (0);
if (!success) return NULL; // an error occurred
// Hack: If audio and video are demuxed together on a single RTP stream,
// then create a new "demuxer_t" structure to allow the higher-level
// code to recognize this:
if (demux_is_multiplexed_rtp_stream(demuxer)) {
stream_t* s = new_ds_stream(demuxer->video);
demuxer_t* od = demux_open(opts, s, DEMUXER_TYPE_UNKNOWN,
opts->audio_id, opts->video_id, opts->sub_id,
NULL);
demuxer = new_demuxers_demuxer(od, od, od);
}
return demuxer;
}
extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
// Get the RTP state that was stored in the demuxer's 'priv' field:
RTPState* rtpState = (RTPState*)(demuxer->priv);
return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
}
extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) {
// Get the RTP state that was stored in the demuxer's 'priv' field:
RTPState* rtpState = (RTPState*)(demuxer->priv);
return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0;
}
static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
Boolean mustGetNewData,
float& ptsBehind); // forward
extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
// Get a filled-in "demux_packet" from the RTP source, and deliver it.
// Note that this is called as a synchronous read operation, so it needs
// to block in the (hopefully infrequent) case where no packet is
// immediately available.
while (1) {
float ptsBehind;
demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
if (dp == NULL) return 0;
if (demuxer->stream->eof) return 0; // source stream has closed down
// Before using this packet, check to make sure that its presentation
// time is not far behind the other stream (if any). If it is,
// then we discard this packet, and get another instead. (The rest of
// MPlayer doesn't always do a good job of synchronizing when the
// audio and video streams get this far apart.)
// (We don't do this when streaming over TCP, because then the audio and
// video streams are interleaved.)
// (Also, if the stream is *excessively* far behind, then we allow
// the packet, because in this case it probably means that there was
// an error in the source's timestamp synchronization.)
const float ptsBehindThreshold = 1.0; // seconds
const float ptsBehindLimit = 60.0; // seconds
if (ptsBehind < ptsBehindThreshold ||
ptsBehind > ptsBehindLimit ||
rtsp_transport_tcp) { // packet's OK
ds_add_packet(ds, dp);
break;
}
#ifdef DEBUG_PRINT_DISCARDED_PACKETS
RTPState* rtpState = (RTPState*)(demuxer->priv);
ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue;
fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind);
#endif
free_demux_packet(dp); // give back this packet, and get another one
}
return 1;
}
Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds,
unsigned char*& packetData, unsigned& packetDataLen,
float& pts) {
// Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
// is not delivered to the "demux_stream".
float ptsBehind;
demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking
if (dp == NULL) return False;
packetData = dp->buffer;
packetDataLen = dp->len;
pts = dp->pts;
return True;
}
static void teardownRTSPorSIPSession(RTPState* rtpState); // forward
extern "C" void demux_close_rtp(demuxer_t* demuxer) {
// Reclaim all RTP-related state:
// Get the RTP state that was stored in the demuxer's 'priv' field:
RTPState* rtpState = (RTPState*)(demuxer->priv);
if (rtpState == NULL) return;
teardownRTSPorSIPSession(rtpState);
UsageEnvironment* env = NULL;
TaskScheduler* scheduler = NULL;
if (rtpState->mediaSession != NULL) {
env = &(rtpState->mediaSession->envir());
scheduler = &(env->taskScheduler());
}
Medium::close(rtpState->mediaSession);
Medium::close(rtpState->rtspClient);
Medium::close(rtpState->sipClient);
delete rtpState->audioBufferQueue;
delete rtpState->videoBufferQueue;
delete[] rtpState->sdpDescription;
delete rtpState;
av_freep(&avcctx);
env->reclaim(); delete scheduler;
}
////////// Extra routines that help implement the above interface functions:
#define MAX_RTP_FRAME_SIZE 5000000
// >= the largest conceivable frame composed from one or more RTP packets
static void afterReading(void* clientData, unsigned frameSize,
unsigned /*numTruncatedBytes*/,
struct timeval presentationTime,
unsigned /*durationInMicroseconds*/) {
int headersize = 0;
if (frameSize >= MAX_RTP_FRAME_SIZE) {
fprintf(stderr, "Saw an input frame too large (>=%d). Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
MAX_RTP_FRAME_SIZE);
}
ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
demuxer_t* demuxer = bufferQueue->ourDemuxer();
RTPState* rtpState = (RTPState*)(demuxer->priv);
if (frameSize > 0) demuxer->stream->eof = 0;
demux_packet_t* dp = bufferQueue->dp;
if (bufferQueue->readSource()->isAMRAudioSource())
headersize = 1;
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;
}
resize_demux_packet(dp, frameSize + headersize);
// Set the packet's presentation time stamp, depending on whether or
// not our RTP source's timestamps have been synchronized yet:
Boolean hasBeenSynchronized
= bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
if (hasBeenSynchronized) {
if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
fprintf(stderr, "%s stream has been synchronized using RTCP \n",
bufferQueue->tag());
}
struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
if (fst->tv_sec == 0 && fst->tv_usec == 0) {
*fst = presentationTime;
}
// For the "pts" field, use the time differential from the first
// synchronized time, rather than absolute time, in order to avoid
// round-off errors when converting to a float:
dp->pts = presentationTime.tv_sec - fst->tv_sec
+ (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
bufferQueue->prevPacketPTS = dp->pts;
} else {
if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) {
fprintf(stderr, "%s stream is no longer RTCP-synchronized \n",
bufferQueue->tag());
}
// use the previous packet's "pts" once again:
dp->pts = bufferQueue->prevPacketPTS;
}
bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized;
dp->pos = demuxer->filepos;
demuxer->filepos += frameSize + headersize;
// Signal any pending 'doEventLoop()' call on this queue:
bufferQueue->blockingFlag = ~0;
}
static void onSourceClosure(void* clientData) {
ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
demuxer_t* demuxer = bufferQueue->ourDemuxer();
demuxer->stream->eof = 1;
// Signal any pending 'doEventLoop()' call on this queue:
bufferQueue->blockingFlag = ~0;
}
static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
Boolean mustGetNewData,
float& ptsBehind) {
// Begin by finding the buffer queue that we want to read from:
// (Get this from the RTP state, which we stored in
// the demuxer's 'priv' field)
RTPState* rtpState = (RTPState*)(demuxer->priv);
ReadBufferQueue* bufferQueue = NULL;
int headersize = 0;
int waitboth = 0;
TaskToken task, task2;
if (demuxer->stream->eof) return NULL;
if (ds == demuxer->video) {
bufferQueue = rtpState->audioBufferQueue;
// HACK: for the latest versions we must also receive audio
// when probing for video FPS, otherwise the stream just hangs
// and times out
if (mustGetNewData &&
bufferQueue &&
bufferQueue->readSource() &&
!bufferQueue->nextpacket) {
headersize = bufferQueue->readSource()->isAMRAudioSource() ? 1 : 0;
demux_packet_t *dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
bufferQueue->dp = dp;
bufferQueue->blockingFlag = 0;
bufferQueue->readSource()->getNextFrame(
&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize,
afterReading, bufferQueue,
onSourceClosure, bufferQueue);
task2 = bufferQueue->readSource()->envir().taskScheduler().
scheduleDelayedTask(10000000, onSourceClosure, bufferQueue);
waitboth = 1;
}
bufferQueue = rtpState->videoBufferQueue;
if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4'))
headersize = 3;
} else if (ds == demuxer->audio) {
bufferQueue = rtpState->audioBufferQueue;
if (bufferQueue->readSource()->isAMRAudioSource())
headersize = 1;
} else {
fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n");
return NULL;
}
if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
return NULL;
}
demux_packet_t* dp = NULL;
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:
if (!bufferQueue->nextpacket) {
dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
bufferQueue->dp = dp;
if (dp == NULL) return NULL;
}
extern AVCodecParserContext * h264parserctx;
int consumed, poutbuf_size = 1;
const uint8_t *poutbuf = NULL;
float lastpts = 0.0;
do {
if (!bufferQueue->nextpacket) {
// Schedule the read operation:
bufferQueue->blockingFlag = 0;
bufferQueue->readSource()->getNextFrame(&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize,
afterReading, bufferQueue,
onSourceClosure, bufferQueue);
// Block ourselves until data becomes available:
TaskScheduler& scheduler
= bufferQueue->readSource()->envir().taskScheduler();
int delay = 10000000;
if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime())
delay /= 10;
task = scheduler.scheduleDelayedTask(delay, onSourceClosure, bufferQueue);
scheduler.doEventLoop(&bufferQueue->blockingFlag);
scheduler.unscheduleDelayedTask(task);
if (waitboth) {
scheduler.doEventLoop(&rtpState->audioBufferQueue->blockingFlag);
scheduler.unscheduleDelayedTask(task2);
}
if (demuxer->stream->eof) {
free_demux_packet(dp);
return NULL;
}
if (headersize == 1) // amr
dp->buffer[0] =
((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader();
} else {
bufferQueue->dp = dp = bufferQueue->nextpacket;
bufferQueue->nextpacket = NULL;
}
if (headersize == 3 && h264parserctx) { // h264
consumed = h264parserctx->parser->parser_parse(h264parserctx,
avcctx,
&poutbuf, &poutbuf_size,
dp->buffer, dp->len);
if (!consumed && !poutbuf_size)
return NULL;
if (!poutbuf_size) {
lastpts=dp->pts;
free_demux_packet(dp);
bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
} else {
bufferQueue->nextpacket = dp;
bufferQueue->dp = dp = new_demux_packet(poutbuf_size);
memcpy(dp->buffer, poutbuf, poutbuf_size);
dp->pts=lastpts;
}
}
} while (!poutbuf_size);
// 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) {
rtpState->rtspClient->teardownMediaSession(*mediaSession);
} 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), nextpacket(NULL),
pendingDPHead(NULL), pendingDPTail(NULL),
fReadSource(subsession == NULL ? NULL : subsession->readSource()),
fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
fOurDemuxer(demuxer), fTag(strdup(tag)) {
}
ReadBufferQueue::~ReadBufferQueue() {
free((void *)fTag);
// Free any pending buffers (that never got delivered):
demux_packet_t* dp = pendingDPHead;
while (dp != NULL) {
demux_packet_t* dpNext = dp->next;
dp->next = NULL;
free_demux_packet(dp);
dp = dpNext;
}
}
void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) {
// Keep this buffer around, until MPlayer asks for it later:
if (pendingDPTail == NULL) {
pendingDPHead = pendingDPTail = dp;
} else {
pendingDPTail->next = dp;
pendingDPTail = dp;
}
dp->next = NULL;
}
demux_packet_t* ReadBufferQueue::getPendingBuffer() {
demux_packet_t* dp = pendingDPHead;
if (dp != NULL) {
pendingDPHead = dp->next;
if (pendingDPHead == NULL) pendingDPTail = NULL;
dp->next = NULL;
}
return dp;
}
static int demux_rtp_control(struct demuxer *demuxer, int cmd, void *arg) {
double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime();
switch(cmd) {
case DEMUXER_CTRL_GET_TIME_LENGTH:
if (endpts <= 0)
return DEMUXER_CTRL_DONTKNOW;
*((double *)arg) = endpts;
return DEMUXER_CTRL_OK;
case DEMUXER_CTRL_GET_PERCENT_POS:
if (endpts <= 0)
return DEMUXER_CTRL_DONTKNOW;
*((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts);
return DEMUXER_CTRL_OK;
default:
return DEMUXER_CTRL_NOTIMPL;
}
}
demuxer_desc_t demuxer_desc_rtp = {
"LIVE555 RTP demuxer",
"live555",
"",
"Ross Finlayson",
"requires LIVE555 Streaming Media library",
DEMUXER_TYPE_RTP,
0, // no autodetect
NULL,
demux_rtp_fill_buffer,
demux_open_rtp,
demux_close_rtp,
NULL,
demux_rtp_control
};
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