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-rw-r--r--Makefile3
-rw-r--r--cfg-mplayer.h7
-rwxr-xr-xconfigure57
-rw-r--r--libmpdemux/demux_rtp.cpp733
-rw-r--r--libmpdemux/demux_rtp.h43
-rw-r--r--libmpdemux/demux_rtp_codec.cpp426
-rw-r--r--libmpdemux/demux_rtp_internal.h54
-rw-r--r--libmpdemux/demuxer.c4
-rw-r--r--libmpdemux/demuxer.h1
-rw-r--r--libmpdemux/video.c4
-rw-r--r--stream/cache2.c4
-rw-r--r--stream/stream.c6
-rw-r--r--stream/stream.h4
-rw-r--r--stream/stream_live555.c132
14 files changed, 2 insertions, 1476 deletions
diff --git a/Makefile b/Makefile
index 2a8b4a536f..91795fb4fa 100644
--- a/Makefile
+++ b/Makefile
@@ -65,9 +65,6 @@ SRCS_COMMON-$(LIBNEMESI) += libmpdemux/demux_nemesi.c \
SRCS_COMMON-$(LIBPOSTPROC) += libmpcodecs/vf_pp.c
SRCS_COMMON-$(LIBSMBCLIENT) += stream/stream_smb.c
-SRCS_COMMON-$(LIVE555) += libmpdemux/demux_rtp.cpp \
- libmpdemux/demux_rtp_codec.cpp \
- stream/stream_live555.c
SRCS_COMMON-$(MACOSX_FINDER) += osdep/macosx_finder_args.m
SRCS_COMMON-$(COCOA) += libvo/osx_common.c \
libvo/cocoa_common.m \
diff --git a/cfg-mplayer.h b/cfg-mplayer.h
index 7d26316ac5..ed788aab9a 100644
--- a/cfg-mplayer.h
+++ b/cfg-mplayer.h
@@ -382,13 +382,10 @@ const m_option_t common_opts[] = {
#endif /* HAVE_AF_INET6 */
#endif /* CONFIG_NETWORKING */
-#ifdef CONFIG_LIVE555
- {"rtsp-stream-over-http", &rtsp_transport_http, CONF_TYPE_FLAG, 0, 0, 1, NULL},
-#endif /* CONFIG_LIVE555 */
-#if defined(CONFIG_LIBNEMESI) || defined(CONFIG_LIVE555)
+#if defined(CONFIG_LIBNEMESI)
// -rtsp-stream-over-tcp option, specifying TCP streaming of RTP/RTCP
{"rtsp-stream-over-tcp", &rtsp_transport_tcp, CONF_TYPE_FLAG, 0, 0, 1, NULL},
-#endif /* defined(CONFIG_LIBNEMESI) || defined(CONFIG_LIVE555) */
+#endif /* defined(CONFIG_LIBNEMESI) */
#ifdef CONFIG_LIBNEMESI
{"rtsp-stream-over-sctp", &rtsp_transport_sctp, CONF_TYPE_FLAG, 0, 0, 1, NULL},
#endif /* CONFIG_LIBNEMESI */
diff --git a/configure b/configure
index 0effc81147..3ebf7be99d 100755
--- a/configure
+++ b/configure
@@ -319,7 +319,6 @@ Optional features:
--disable-networking disable networking [enable]
--enable-winsock2_h enable winsock2_h [autodetect]
--enable-smb enable Samba (SMB) input [autodetect]
- --enable-live enable LIVE555 Streaming Media [disable]
--enable-libquvi enable libquvi [autodetect]
--enable-nemesi enable Nemesi Streaming Media [autodetect]
--enable-lcms2 enable LCMS2 support [autodetect]
@@ -474,7 +473,6 @@ _libbs2b=auto
_vcd=auto
_bluray=auto
_dvdread=auto
-_live=no
_nemesi=auto
_lcms2=auto
_xinerama=auto
@@ -694,8 +692,6 @@ for ac_option do
--disable-bluray) _bluray=no ;;
--enable-dvdread) _dvdread=yes ;;
--disable-dvdread) _dvdread=no ;;
- --enable-live) _live=yes ;;
- --disable-live) _live=no ;;
--enable-nemesi) _nemesi=yes ;;
--disable-nemesi) _nemesi=no ;;
--enable-lcms2) _lcms2=yes ;;
@@ -3034,57 +3030,6 @@ else
fi
echores "$_nemesi"
-echocheck "LIVE555 Streaming Media libraries"
-if test "$_live" != no && test "$networking" = yes ; then
- cat > $TMPCPP << EOF
-#include <liveMedia.hh>
-#if (LIVEMEDIA_LIBRARY_VERSION_INT < 1141257600)
-#error Please upgrade to version 2006.03.03 or later of the "LIVE555 Streaming Media" libraries - available from <www.live555.com/liveMedia/>
-#endif
-int main(void) { return 0; }
-EOF
-
- _live=no
- for I in $extra_cflags "-I$_libdir/live" "-I/usr/lib/live" "-I/usr/lib64/live" "-I/usr/local/live" "-I/usr/local/lib/live" ; do
- cxx_check $I/liveMedia/include $I/UsageEnvironment/include $I/groupsock/include &&
- _livelibdir=$(echo $I| sed s/-I//) &&
- extra_ldflags="$_livelibdir/liveMedia/libliveMedia.a \
- $_livelibdir/groupsock/libgroupsock.a \
- $_livelibdir/UsageEnvironment/libUsageEnvironment.a \
- $_livelibdir/BasicUsageEnvironment/libBasicUsageEnvironment.a \
- $extra_ldflags -lstdc++" \
- extra_cxxflags="-I$_livelibdir/liveMedia/include \
- -I$_livelibdir/UsageEnvironment/include \
- -I$_livelibdir/BasicUsageEnvironment/include \
- -I$_livelibdir/groupsock/include" &&
- _live=yes && break
- done
- if test "$_live" != yes ; then
- ld_tmp="-lliveMedia -lgroupsock -lUsageEnvironment -lBasicUsageEnvironment -lstdc++"
- if cxx_check -I/usr/include/liveMedia -I/usr/include/UsageEnvironment -I/usr/include/groupsock $ld_tmp; then
- _live_dist=yes
- fi
- fi
-fi
-if test "$_live" = yes && test "$networking" = yes; then
- test $_livelibdir && res_comment="using $_livelibdir"
- def_live='#define CONFIG_LIVE555 1'
- inputmodules="live555 $inputmodules"
-elif test "$_live_dist" = yes && test "$networking" = yes; then
- res_comment="using distribution version"
- _live="yes"
- def_live='#define CONFIG_LIVE555 1'
- extra_ldflags="$extra_ldflags $ld_tmp"
- extra_cxxflags="-I/usr/include/liveMedia -I/usr/include/UsageEnvironment -I/usr/include/BasicUsageEnvironment -I/usr/include/groupsock"
- inputmodules="live555 $inputmodules"
-else
- _live=no
- def_live='#undef CONFIG_LIVE555'
- noinputmodules="live555 $noinputmodules"
-fi
-echores "$_live"
-
-
# Test with > against Libav 0.8 versions which will NOT work rather than
# specify minimum version, to allow (future) point releases to possibly work.
@@ -3551,7 +3496,6 @@ LIBSMBCLIENT = $_smb
LIBQUVI = $_libquvi
LIBTHEORA = $_theora
LIRC = $_lirc
-LIVE555 = $_live
MACOSX_FINDER = $_macosx_finder
MNG = $_mng
MPG123 = $_mpg123
@@ -3772,7 +3716,6 @@ $def_ftp
$def_inet6
$def_inet_aton
$def_inet_pton
-$def_live
$def_nemesi
$def_networking
$def_smb
diff --git a/libmpdemux/demux_rtp.cpp b/libmpdemux/demux_rtp.cpp
deleted file mode 100644
index df81d6d89c..0000000000
--- a/libmpdemux/demux_rtp.cpp
+++ /dev/null
@@ -1,733 +0,0 @@
-/*
- * 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
-};
diff --git a/libmpdemux/demux_rtp.h b/libmpdemux/demux_rtp.h
deleted file mode 100644
index 6d3462d856..0000000000
--- a/libmpdemux/demux_rtp.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef MPLAYER_DEMUX_RTP_H
-#define MPLAYER_DEMUX_RTP_H
-
-#include <stdlib.h>
-#include <stdio.h>
-#include "demuxer.h"
-
-// Open a RTP demuxer (which was initiated either from a SDP file,
-// or from a RTSP URL):
-demuxer_t* demux_open_rtp(demuxer_t* demuxer);
-
-// Test whether a