/***************************************************************************** * live555.cpp : LIVE555 Streaming Media support. ***************************************************************************** * Copyright (C) 2003-2007 VLC authors and VideoLAN * $Id$ * * Authors: Laurent Aimar * Derk-Jan Hartman * Derk-Jan Hartman for M2X * Sébastien Escudier * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU Lesser General Public License as published by * the Free Software Foundation; either version 2.1 of the License, or * (at your option) any later version. * * This program 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 Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public License * along with this program; if not, write to the Free Software Foundation, * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble *****************************************************************************/ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include #include #include #include #include #include #include #include #include #if defined( _WIN32 ) # include #endif #include #include #include #include #include #include extern "C" { #include "../access/mms/asf.h" /* Who said ugly ? */ #include "../codec/opus_header.h" /* Who said uglier ? */ #include "live555_dtsgen.h" } /***************************************************************************** * Module descriptor *****************************************************************************/ static int Open ( vlc_object_t * ); static void Close( vlc_object_t * ); #define KASENNA_TEXT N_( "Kasenna RTSP dialect") #define KASENNA_LONGTEXT N_( "Kasenna servers use an old and nonstandard " \ "dialect of RTSP. With this parameter VLC will try this dialect, but "\ "then it cannot connect to normal RTSP servers." ) #define WMSERVER_TEXT N_("WMServer RTSP dialect") #define WMSERVER_LONGTEXT N_("WMServer uses a nonstandard dialect " \ "of RTSP. Selecting this parameter will tell VLC to assume some " \ "options contrary to RFC 2326 guidelines.") #define USER_TEXT N_("Username") #define USER_LONGTEXT N_("Sets the username for the connection, " \ "if no username or password are set in the url.") #define PASS_TEXT N_("Password") #define PASS_LONGTEXT N_("Sets the password for the connection, " \ "if no username or password are set in the url.") #define FRAME_BUFFER_SIZE_TEXT N_("RTSP frame buffer size") #define FRAME_BUFFER_SIZE_LONGTEXT N_("RTSP start frame buffer size of the video " \ "track, can be increased in case of broken pictures due " \ "to too small buffer.") #define DEFAULT_FRAME_BUFFER_SIZE 250000 vlc_module_begin () set_description( N_("RTP/RTSP/SDP demuxer (using Live555)" ) ) set_capability( "demux", 50 ) set_shortname( "RTP/RTSP") set_callbacks( Open, Close ) add_shortcut( "live", "livedotcom" ) set_category( CAT_INPUT ) set_subcategory( SUBCAT_INPUT_DEMUX ) add_submodule () set_description( N_("RTSP/RTP access and demux") ) add_shortcut( "rtsp", "pnm", "live", "livedotcom" ) set_capability( "access_demux", 0 ) set_callbacks( Open, Close ) add_bool( "rtsp-tcp", false, N_("Use RTP over RTSP (TCP)"), N_("Use RTP over RTSP (TCP)"), true ) change_safe() add_integer( "rtp-client-port", -1, N_("Client port"), N_("Port to use for the RTP source of the session"), true ) add_bool( "rtsp-mcast", false, N_("Force multicast RTP via RTSP"), N_("Force multicast RTP via RTSP"), true ) change_safe() add_bool( "rtsp-http", false, N_("Tunnel RTSP and RTP over HTTP"), N_("Tunnel RTSP and RTP over HTTP"), true ) change_safe() add_integer( "rtsp-http-port", 80, N_("HTTP tunnel port"), N_("Port to use for tunneling the RTSP/RTP over HTTP."), true ) add_bool( "rtsp-kasenna", false, KASENNA_TEXT, KASENNA_LONGTEXT, true ) change_safe() add_bool( "rtsp-wmserver", false, WMSERVER_TEXT, WMSERVER_LONGTEXT, true) change_safe() add_string( "rtsp-user", NULL, USER_TEXT, USER_LONGTEXT, true ) change_safe() add_password( "rtsp-pwd", NULL, PASS_TEXT, PASS_LONGTEXT, true ) add_integer( "rtsp-frame-buffer-size", DEFAULT_FRAME_BUFFER_SIZE, FRAME_BUFFER_SIZE_TEXT, FRAME_BUFFER_SIZE_LONGTEXT, true ) vlc_module_end () /***************************************************************************** * Local prototypes *****************************************************************************/ typedef struct { demux_t *p_demux; MediaSubsession *sub; es_format_t fmt; es_out_id_t *p_es; enum { SINGLE_STREAM, MULTIPLEXED_STREAM, QUICKTIME_STREAM, ASF_STREAM } format; block_t *p_asf_block; bool b_discard_trunc; vlc_demux_chained_t *p_out_muxed; /* for muxed stream */ uint8_t *p_buffer; unsigned int i_buffer; bool b_rtcp_sync; bool b_flushing_discontinuity; int i_next_block_flags; char waiting; int64_t i_prevpts; int64_t i_pcr; double f_npt; struct dtsgen_t dtsgen; enum { STATE_NONE, STATE_SELECTED, STATE_IGNORED, STATE_TEARDOWN, } state; } live_track_t; class RTSPClientVlc; #define CAP_RATE_CONTROL (1 << 1) #define CAP_SUBSESSION_TEARDOWN (1 << 2) #define CAP_SUBSESSION_PAUSE (1 << 3) #define CAPS_DEFAULT CAP_RATE_CONTROL #if LIVEMEDIA_LIBRARY_VERSION_INT < 1732752000 // 2024.11.28 typedef char EventLoopWatchVariable; #endif struct demux_sys_t { char *p_sdp; /* XXX mallocated */ char *psz_pl_url; /* password-less URL */ vlc_url_t url; MediaSession *ms; TaskScheduler *scheduler; UsageEnvironment *env ; RTSPClientVlc *rtsp; int capabilities; /* Server capabilities workaround */ /* */ int i_track; live_track_t **track; /* Weird formats */ asf_header_t asfh; vlc_demux_chained_t *p_out_asf; bool b_real; /* */ int64_t i_pcr; /* The clock */ bool b_rtcp_sync; /* At least one track received sync */ double f_npt; double f_npt_length; double f_npt_start; /* timeout thread information */ vlc_timer_t timer; vlc_mutex_t timeout_mutex; /* Serialise calls to live555 in timeout thread w.r.t. Demux()/Control() */ /* */ bool b_force_mcast; bool b_multicast; /* if one of the tracks is multicasted */ bool b_no_data; /* if we never received any data */ int i_no_data_ti; /* consecutive number of TaskInterrupt */ EventLoopWatchVariable event_rtsp; EventLoopWatchVariable event_data; bool b_get_param; /* Does the server support GET_PARAMETER */ bool b_paused; /* Are we paused? */ bool b_error; int i_live555_ret; /* live555 callback return code */ float f_seek_request;/* In case we receive a seek request while paused*/ }; class RTSPClientVlc : public RTSPClient { public: RTSPClientVlc( UsageEnvironment& env, char const* rtspURL, int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum, demux_sys_t *p_sys) : RTSPClient( env, rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum #if LIVEMEDIA_LIBRARY_VERSION_INT >= 1373932800 , -1 #endif ) { this->p_sys = p_sys; } demux_sys_t *p_sys; }; static int Demux ( demux_t * ); static int Control( demux_t *, int, va_list ); static int Connect ( demux_t * ); static int SessionsSetup( demux_t * ); static int Play ( demux_t *); static int ParseASF ( demux_t * ); static int RollOverTcp ( demux_t * ); static void StreamRead ( void *, unsigned int, unsigned int, struct timeval, unsigned int ); static void StreamClose ( void * ); static void TaskInterruptData( void * ); static void TaskInterruptRTSP( void * ); static void TimeoutPrevention( void * ); static unsigned char* parseH264ConfigStr( char const* configStr, unsigned int& configSize ); static unsigned char* parseVorbisConfigStr( char const* configStr, unsigned int& configSize ); static char *passwordLessURL( vlc_url_t *url ); #define PCR_OBS (CLOCK_FREQ / 4) #define PCR_OFF PCR_OBS /***************************************************************************** * DemuxOpen: *****************************************************************************/ static int Open ( vlc_object_t *p_this ) { demux_t *p_demux = (demux_t*)p_this; demux_sys_t *p_sys = NULL; char *psz_url; int i_return; int i_error = VLC_EGENERIC; /* if the rtsp URL may contain a sat.ip fake DNS, bail-out early and * let the SAT>IP module handle that */ if( !strncmp(p_demux->psz_location, "sat.ip", 6) ) { msg_Err( p_demux, "SAT>IP server, bailing out"); return VLC_EGENERIC; } /* If satip-host is set on the item, we shall assume it is a rtsp for the * SAT>IP module and bail-out early. */ char *psz_host = var_InheritString(p_demux, "satip-host"); if (psz_host != NULL) { msg_Err( p_demux, "URL is for SAT>IP, bailing out"); free(psz_host); return VLC_EGENERIC; } if( p_demux->s ) { /* See if it looks like a SDP v, o, s fields are mandatory and in this order */ const uint8_t *p_peek; if( vlc_stream_Peek( p_demux->s, &p_peek, 7 ) < 7 ) return VLC_EGENERIC; if( memcmp( p_peek, "v=0\r\n", 5 ) && memcmp( p_peek, "v=0\n", 4 ) && ( p_peek[0] < 'a' || p_peek[0] > 'z' || p_peek[1] != '=' ) ) { return VLC_EGENERIC; } } p_demux->pf_demux = Demux; p_demux->pf_control= Control; p_demux->p_sys = p_sys = (demux_sys_t*)calloc( 1, sizeof( demux_sys_t ) ); if( !p_sys ) return VLC_ENOMEM; if( vlc_timer_create(&p_sys->timer, TimeoutPrevention, p_demux) ) { free( p_sys ); return VLC_ENOMEM; } msg_Dbg( p_demux, "version " LIVEMEDIA_LIBRARY_VERSION_STRING ); p_sys->capabilities = CAPS_DEFAULT; if( var_GetBool( p_demux, "rtsp-kasenna" ) || var_GetBool( p_demux, "rtsp-wmserver" ) ) { p_sys->capabilities &= ~CAP_RATE_CONTROL; } TAB_INIT( p_sys->i_track, p_sys->track ); p_sys->b_no_data = true; p_sys->b_force_mcast = var_InheritBool( p_demux, "rtsp-mcast" ); p_sys->f_seek_request = -1; vlc_mutex_init(&p_sys->timeout_mutex); /* parse URL for rtsp://[user:[passwd]@]serverip:port/options */ if( asprintf( &psz_url, "%s://%s", p_demux->psz_access, p_demux->psz_location ) == -1 ) { i_error = VLC_ENOMEM; goto error; } vlc_UrlParse( &p_sys->url, psz_url ); free( psz_url ); if( ( p_sys->psz_pl_url = passwordLessURL( &p_sys->url ) ) == NULL ) { i_error = VLC_ENOMEM; goto error; } if( ( p_sys->scheduler = BasicTaskScheduler::createNew() ) == NULL ) { msg_Err( p_demux, "BasicTaskScheduler::createNew failed" ); goto error; } if( !( p_sys->env = BasicUsageEnvironment::createNew(*p_sys->scheduler) ) ) { msg_Err( p_demux, "BasicUsageEnvironment::createNew failed" ); goto error; } if( strcasecmp( p_demux->psz_access, "sdp" ) ) { char *p = p_sys->psz_pl_url; while( (p = strchr( p, ' ' )) != NULL ) *p = '+'; } if( p_demux->s != NULL ) { /* Gather the complete sdp file */ int i_sdp = 0; int i_sdp_max = 1000; uint8_t *p_sdp = (uint8_t*) malloc( i_sdp_max ); if( !p_sdp ) { i_error = VLC_ENOMEM; goto error; } for( ;; ) { int i_read = vlc_stream_Read( p_demux->s, &p_sdp[i_sdp], i_sdp_max - i_sdp - 1 ); if( i_read < 0 ) { msg_Err( p_demux, "failed to read SDP" ); free( p_sdp ); goto error; } i_sdp += i_read; if( i_read < i_sdp_max - i_sdp - 1 ) { p_sdp[i_sdp] = '\0'; break; } i_sdp_max += 1000; p_sdp = (uint8_t*)xrealloc( p_sdp, i_sdp_max ); } p_sys->p_sdp = (char*)p_sdp; } else if( ( i_return = Connect( p_demux ) ) != VLC_SUCCESS ) { msg_Err( p_demux, "Failed to connect with %s", p_sys->psz_pl_url ); goto error; } if( p_sys->p_sdp == NULL ) { msg_Err( p_demux, "Failed to retrieve the RTSP Session Description" ); i_error = VLC_ENOMEM; goto error; } if( ( i_return = SessionsSetup( p_demux ) ) != VLC_SUCCESS ) { msg_Err( p_demux, "Nothing to play for %s", p_sys->psz_pl_url ); goto error; } if( p_sys->b_real ) goto error; if( ( i_return = Play( p_demux ) ) != VLC_SUCCESS ) goto error; if( p_sys->p_out_asf && ParseASF( p_demux ) ) { msg_Err( p_demux, "cannot find a usable asf header" ); /* TODO Clean tracks */ goto error; } if( p_sys->i_track <= 0 ) goto error; return VLC_SUCCESS; error: Close( p_this ); return i_error; } /***************************************************************************** * DemuxClose: *****************************************************************************/ static void Close( vlc_object_t *p_this ) { demux_t *p_demux = (demux_t*)p_this; demux_sys_t *p_sys = p_demux->p_sys; vlc_timer_destroy(p_sys->timer); if( p_sys->rtsp && p_sys->ms ) p_sys->rtsp->sendTeardownCommand( *p_sys->ms, NULL ); if( p_sys->ms ) Medium::close( p_sys->ms ); if( p_sys->rtsp ) RTSPClient::close( p_sys->rtsp ); if( p_sys->env ) p_sys->env->reclaim(); for( int i = 0; i < p_sys->i_track; i++ ) { live_track_t *tk = p_sys->track[i]; if( tk->p_out_muxed ) vlc_demux_chained_Delete( tk->p_out_muxed ); es_format_Clean( &tk->fmt ); dtsgen_Clean( &tk->dtsgen ); free( tk->p_buffer ); free( tk ); } TAB_CLEAN( p_sys->i_track, p_sys->track ); if( p_sys->p_out_asf ) vlc_demux_chained_Delete( p_sys->p_out_asf ); delete p_sys->scheduler; free( p_sys->p_sdp ); free( p_sys->psz_pl_url ); vlc_UrlClean( &p_sys->url ); vlc_mutex_destroy(&p_sys->timeout_mutex); free( p_sys ); } static inline Boolean toBool( bool b ) { return b?True:False; } // silly, no? static void default_live555_callback( RTSPClient* client, int result_code, char* result_string ) { RTSPClientVlc *client_vlc = static_cast ( client ); demux_sys_t *p_sys = client_vlc->p_sys; delete []result_string; p_sys->i_live555_ret = result_code; p_sys->b_error = p_sys->i_live555_ret != 0; p_sys->event_rtsp = 1; } /* return true if the RTSP command succeeded */ static bool wait_Live555_response( demux_t *p_demux, int i_timeout = 0 /* ms */ ) { TaskToken task; demux_sys_t * p_sys = p_demux->p_sys; p_sys->event_rtsp = 0; if( i_timeout > 0 ) { /* Create a task that will be called if we wait more than timeout ms */ task = p_sys->scheduler->scheduleDelayedTask( i_timeout*1000, TaskInterruptRTSP, p_demux ); } p_sys->event_rtsp = 0; p_sys->b_error = true; p_sys->i_live555_ret = 0; p_sys->scheduler->doEventLoop( &p_sys->event_rtsp ); //here, if b_error is true and i_live555_ret = 0 we didn't receive a response if( i_timeout > 0 ) { /* remove the task */ p_sys->scheduler->unscheduleDelayedTask( task ); } return !p_sys->b_error; } static void continueAfterDESCRIBE( RTSPClient* client, int result_code, char* result_string ) { RTSPClientVlc *client_vlc = static_cast ( client ); demux_sys_t *p_sys = client_vlc->p_sys; p_sys->i_live555_ret = result_code; if ( result_code == 0 ) { char* sdpDescription = result_string; free( p_sys->p_sdp ); p_sys->p_sdp = NULL; if( sdpDescription ) { p_sys->p_sdp = strdup( sdpDescription ); p_sys->b_error = false; } } else p_sys->b_error = true; delete[] result_string; p_sys->event_rtsp = 1; #ifdef VLC_PATCH_RTSPCLIENT_SERVERSTRING if( client_vlc->serverString() ) { if( !strncmp(client_vlc->serverString(), "Kasenna", 7) || !strncmp(client_vlc->serverString(), "WMServer", 8) ) p_sys->capabilities &= ~CAP_RATE_CONTROL; if( !strncmp(client_vlc->serverString(), "VLC/", 4) ) p_sys->capabilities |= (CAP_SUBSESSION_TEARDOWN|CAP_SUBSESSION_PAUSE); } #endif } static void continueAfterOPTIONS( RTSPClient* client, int result_code, char* result_string ) { RTSPClientVlc *client_vlc = static_cast (client); demux_sys_t *p_sys = client_vlc->p_sys; p_sys->b_get_param = // If OPTIONS fails, assume GET_PARAMETER is not supported but // still continue on with the stream. Some servers (foscam) // return 501/not implemented for OPTIONS. result_code == 0 && result_string != NULL && strstr( result_string, "GET_PARAMETER" ) != NULL; client->sendDescribeCommand( continueAfterDESCRIBE ); delete[] result_string; } /***************************************************************************** * Connect: connects to the RTSP server to setup the session DESCRIBE *****************************************************************************/ static int Connect( demux_t *p_demux ) { demux_sys_t *p_sys = p_demux->p_sys; Authenticator authenticator; vlc_credential credential; const char *psz_user = NULL; const char *psz_pwd = NULL; int i_http_port = 0; int i_ret = VLC_SUCCESS; const int i_timeout = var_InheritInteger( p_demux, "ipv4-timeout" ); vlc_credential_init( &credential, &p_sys->url ); /* Credentials can be NULL since they may not be needed */ if( vlc_credential_get( &credential, p_demux, "rtsp-user", "rtsp-pwd", NULL, NULL) ) { psz_user = credential.psz_username; psz_pwd = credential.psz_password; } createnew: /* FIXME: This is naive and incorrect; it does not prevent the thread * getting stuck in blocking socket operations. */ if( vlc_killed() ) { i_ret = VLC_EGENERIC; goto bailout; } if( var_CreateGetBool( p_demux, "rtsp-http" ) ) i_http_port = var_InheritInteger( p_demux, "rtsp-http-port" ); p_sys->rtsp = new (std::nothrow) RTSPClientVlc( *p_sys->env, p_sys->psz_pl_url, var_InheritInteger( p_demux, "verbose" ) > 1 ? 1 : 0, "LibVLC/" VERSION, i_http_port, p_sys ); if( !p_sys->rtsp ) { msg_Err( p_demux, "RTSPClient::createNew failed (%s)", p_sys->env->getResultMsg() ); i_ret = VLC_EGENERIC; goto bailout; } /* Kasenna enables KeepAlive by analysing the User-Agent string. * Appending _KA to the string should be enough to enable this feature, * however, there is a bug where the _KA doesn't get parsed from the * default User-Agent as created by VLC/Live555 code. This is probably due * to spaces in the string or the string being too long. Here we override * the default string with a more compact version. */ if( var_InheritBool( p_demux, "rtsp-kasenna" )) { p_sys->rtsp->setUserAgentString( "VLC_MEDIA_PLAYER_KA" ); } describe: authenticator.setUsernameAndPassword( psz_user, psz_pwd ); p_sys->rtsp->sendOptionsCommand( &continueAfterOPTIONS, &authenticator ); if( !wait_Live555_response( p_demux, i_timeout ) ) { int i_code = p_sys->i_live555_ret; if( i_code == 401 ) { msg_Dbg( p_demux, "authentication failed" ); if( vlc_credential_get( &credential, p_demux, "rtsp-user", "rtsp-pwd", _("RTSP authentication"), _("Please enter a valid login name and a password.") ) ) { psz_user = credential.psz_username; psz_pwd = credential.psz_password; msg_Dbg( p_demux, "retrying with user=%s", psz_user ); goto describe; } } else if( i_code > 0 && i_code != 404 && !var_GetBool( p_demux, "rtsp-http" ) ) { /* Perhaps a firewall is being annoying. Try HTTP tunneling mode */ msg_Dbg( p_demux, "we will now try HTTP tunneling mode" ); var_SetBool( p_demux, "rtsp-http", true ); if( p_sys->rtsp ) RTSPClient::close( p_sys->rtsp ); p_sys->rtsp = NULL; goto createnew; } else { if( i_code == 0 ) msg_Dbg( p_demux, "connection timeout" ); else { msg_Dbg( p_demux, "connection error %d", i_code ); if( i_code == 403 ) vlc_dialog_display_error( p_demux, _("RTSP connection failed"), _("Access to the stream is denied by the server configuration.") ); } if( p_sys->rtsp ) RTSPClient::close( p_sys->rtsp ); p_sys->rtsp = NULL; } i_ret = VLC_EGENERIC; } else vlc_credential_store( &credential, p_demux ); bailout: vlc_credential_clean( &credential ); return i_ret; } /***************************************************************************** * SessionsSetup: prepares the subsessions and does the SETUP *****************************************************************************/ static int SessionsSetup( demux_t *p_demux ) { demux_sys_t *p_sys = p_demux->p_sys; MediaSubsessionIterator *iter = NULL; MediaSubsession *sub = NULL; bool b_rtsp_tcp; int i_client_port; int i_return = VLC_SUCCESS; unsigned int i_receive_buffer = 0; int i_frame_buffer = DEFAULT_FRAME_BUFFER_SIZE; unsigned const thresh = 200000; /* RTP reorder threshold .2 second (default .1) */ const char *p_sess_lang = NULL; const char *p_lang; b_rtsp_tcp = var_CreateGetBool( p_demux, "rtsp-tcp" ) || var_GetBool( p_demux, "rtsp-http" ); i_client_port = var_InheritInteger( p_demux, "rtp-client-port" ); /* Create the session from the SDP */ if( !( p_sys->ms = MediaSession::createNew( *p_sys->env, p_sys->p_sdp ) ) ) { msg_Err( p_demux, "Could not create the RTSP Session: %s", p_sys->env->getResultMsg() ); return VLC_EGENERIC; } if( strcmp( p_sys->p_sdp, "m=" ) != 0 ) { const char *p_sess_attr_end; p_sess_attr_end = strstr( p_sys->p_sdp, "\nm=" ); if( !p_sess_attr_end ) p_sess_attr_end = strstr( p_sys->p_sdp, "\rm=" ); p_sess_lang = p_sess_attr_end ? strstr( p_sys->p_sdp, "a=lang:" ) : NULL; if( p_sess_lang && p_sess_lang - p_sys->p_sdp > p_sess_attr_end - p_sys->p_sdp ) p_sess_lang = NULL; } /* Initialise each media subsession */ iter = new MediaSubsessionIterator( *p_sys->ms ); while( ( sub = iter->next() ) != NULL ) { Boolean bInit; live_track_t *tk; /* Value taken from mplayer */ if( !strcmp( sub->mediumName(), "audio" ) ) i_receive_buffer = 100000; else if( !strcmp( sub->mediumName(), "video" ) ) { int i_var_buf_size = var_InheritInteger( p_demux, "rtsp-frame-buffer-size" ); if( i_var_buf_size > 0 ) i_frame_buffer = i_var_buf_size; i_receive_buffer = 2000000; } else if( !strcmp( sub->mediumName(), "text" ) ) ; else continue; if( strcasestr( sub->codecName(), "REAL" ) ) { msg_Info( p_demux, "real codec detected, using real-RTSP instead" ); p_sys->b_real = true; /* This is a problem, we'll handle it later */ continue; } if( p_sys->rtsp && i_client_port != -1 ) { sub->setClientPortNum( i_client_port ); i_client_port += 2; } if( !strcmp( sub->codecName(), "X-ASF-PF" ) ) bInit = sub->initiate( 0 ); else bInit = sub->initiate(); if( !bInit ) { msg_Warn( p_demux, "RTP subsession '%s/%s' failed (%s)", sub->mediumName(), sub->codecName(), p_sys->env->getResultMsg() ); } else { if( sub->rtpSource() != NULL ) { int fd = sub->rtpSource()->RTPgs()->socketNum(); /* Increase the buffer size */ if( i_receive_buffer > 0 ) increaseReceiveBufferTo( *p_sys->env, fd, i_receive_buffer ); /* Increase the RTP reorder timebuffer just a bit */ sub->rtpSource()->setPacketReorderingThresholdTime(thresh); } msg_Dbg( p_demux, "RTP subsession '%s/%s'", sub->mediumName(), sub->codecName() ); /* Issue the SETUP */ if( p_sys->rtsp ) { p_sys->rtsp->sendSetupCommand( *sub, default_live555_callback, False, toBool( b_rtsp_tcp ), toBool( p_sys->b_force_mcast && !b_rtsp_tcp ) ); if( !wait_Live555_response( p_demux ) ) { /* if we get an unsupported transport error, toggle TCP * use and try again */ if( p_sys->i_live555_ret == 461 ) p_sys->rtsp->sendSetupCommand( *sub, default_live555_callback, False, !toBool( b_rtsp_tcp ), False ); if( p_sys->i_live555_ret != 461 || !wait_Live555_response( p_demux ) ) { msg_Err( p_demux, "SETUP of'%s/%s' failed %s", sub->mediumName(), sub->codecName(), p_sys->env->getResultMsg() ); continue; } else { var_SetBool( p_demux, "rtsp-tcp", true ); b_rtsp_tcp = true; } } } /* Check if we will receive data from this subsession for * this track */ if( sub->readSource() == NULL ) continue; if( !p_sys->b_multicast ) { /* We need different rollover behaviour for multicast */ #if LIVEMEDIA_LIBRARY_VERSION_INT <= 1607558400 // 2020.12.10 netAddressBits addr = sub->connectionEndpointAddress(); #else struct sockaddr_storage addr; sub->getConnectionEndpointAddress(addr); #endif p_sys->b_multicast = IsMulticastAddress( addr ); } tk = (live_track_t*)malloc( sizeof( live_track_t ) ); if( !tk ) { delete iter; return VLC_ENOMEM; } tk->p_demux = p_demux; tk->sub = sub; tk->p_es = NULL; tk->format = live_track_t::SINGLE_STREAM; tk->p_asf_block = NULL; tk->b_discard_trunc = false; tk->p_out_muxed = NULL; tk->waiting = 0; tk->b_rtcp_sync = false; tk->b_flushing_discontinuity = false; tk->i_next_block_flags = 0; tk->i_prevpts = VLC_TICK_INVALID; tk->i_pcr = VLC_TICK_INVALID; tk->f_npt = 0.; dtsgen_Init( &tk->dtsgen ); tk->state = live_track_t::STATE_SELECTED; tk->i_buffer = i_frame_buffer; tk->p_buffer = (uint8_t *)malloc( i_frame_buffer ); if( !tk->p_buffer ) { free( tk ); delete iter; return VLC_ENOMEM; } /* Value taken from mplayer */ if( !strcmp( sub->mediumName(), "audio" ) ) { es_format_Init( &tk->fmt, AUDIO_ES, VLC_CODEC_UNKNOWN ); tk->fmt.audio.i_channels = sub->numChannels(); tk->fmt.audio.i_rate = sub->rtpTimestampFrequency(); if( !strcmp( sub->codecName(), "MPA" ) || !strcmp( sub->codecName(), "MPA-ROBUST" ) || !strcmp( sub->codecName(), "X-MP3-DRAFT-00" ) ) { tk->fmt.i_codec = VLC_CODEC_MPGA; tk->fmt.audio.i_rate = 0; } else if( !strcmp( sub->codecName(), "AC3" ) ) { tk->fmt.i_codec = VLC_CODEC_A52; tk->fmt.audio.i_rate = 0; } else if( !strcmp( sub->codecName(), "L16" ) ) { tk->fmt.i_codec = VLC_CODEC_S16B; tk->fmt.audio.i_bitspersample = 16; } else if( !strcmp( sub->codecName(), "L20" ) ) { tk->fmt.i_codec = VLC_CODEC_S20B; tk->fmt.audio.i_bitspersample = 20; } else if( !strcmp( sub->codecName(), "L24" ) ) { tk->fmt.i_codec = VLC_CODEC_S24B; tk->fmt.audio.i_bitspersample = 24; } else if( !strcmp( sub->codecName(), "L8" ) ) { tk->fmt.i_codec = VLC_CODEC_U8; tk->fmt.audio.i_bitspersample = 8; } else if( !strcmp( sub->codecName(), "DAT12" ) ) { tk->fmt.i_codec = VLC_CODEC_DAT12; tk->fmt.audio.i_bitspersample = 12; } else if( !strcmp( sub->codecName(), "PCMU" ) ) { tk->fmt.i_codec = VLC_CODEC_MULAW; tk->fmt.audio.i_bitspersample = 8; } else if( !strcmp( sub->codecName(), "PCMA" ) ) { tk->fmt.i_codec = VLC_CODEC_ALAW; tk->fmt.audio.i_bitspersample = 8; } else if( !strncmp( sub->codecName(), "G726", 4 ) ) { tk->fmt.i_codec = VLC_CODEC_ADPCM_G726; tk->fmt.audio.i_rate = 8000; tk->fmt.audio.i_channels = 1; if( !strcmp( sub->codecName()+5, "40" ) ) tk->fmt.i_bitrate = 40000; else if( !strcmp( sub->codecName()+5, "32" ) ) tk->fmt.i_bitrate = 32000; else if( !strcmp( sub->codecName()+5, "24" ) ) tk->fmt.i_bitrate = 24000; else if( !strcmp( sub->codecName()+5, "16" ) ) tk->fmt.i_bitrate = 16000; } else if( !strcmp( sub->codecName(), "AMR" ) ) { tk->fmt.i_codec = VLC_CODEC_AMR_NB; } else if( !strcmp( sub->codecName(), "AMR-WB" ) ) { tk->fmt.i_codec = VLC_CODEC_AMR_WB; } else if( !strcmp( sub->codecName(), "MP4A-LATM" ) ) { unsigned int i_extra; uint8_t *p_extra; tk->fmt.i_codec = VLC_CODEC_MP4A; if( ( p_extra = parseStreamMuxConfigStr( sub->fmtp_config(), i_extra ) ) ) { tk->fmt.i_extra = i_extra; tk->fmt.p_extra = xmalloc( i_extra ); memcpy( tk->fmt.p_extra, p_extra, i_extra ); delete[] p_extra; } /* Because the "faad" decoder does not handle the LATM * data length field at the start of each returned LATM * frame, tell the RTP source to omit. */ ((MPEG4LATMAudioRTPSource*)sub->rtpSource())->omitLATMDataLengthField(); } else if( !strcmp( sub->codecName(), "MPEG4-GENERIC" ) ) { unsigned int i_extra; uint8_t *p_extra; tk->fmt.i_codec = VLC_CODEC_MP4A; if( ( p_extra = parseGeneralConfigStr( sub->fmtp_config(), i_extra ) ) ) { tk->fmt.i_extra = i_extra; tk->fmt.p_extra = xmalloc( i_extra ); memcpy( tk->fmt.p_extra, p_extra, i_extra ); delete[] p_extra; } } else if( !strcmp( sub->codecName(), "X-ASF-PF" ) ) { tk->format = live_track_t::ASF_STREAM; if( p_sys->p_out_asf == NULL ) p_sys->p_out_asf = vlc_demux_chained_New( VLC_OBJECT(p_demux), "asf", p_demux->out ); } else if( !strcmp( sub->codecName(), "X-QT" ) || !strcmp( sub->codecName(), "X-QUICKTIME" ) ) { tk->format = live_track_t::QUICKTIME_STREAM; } else if( !strcmp( sub->codecName(), "SPEEX" ) ) { tk->fmt.i_codec = VLC_FOURCC( 's', 'p', 'x', 'r' ); } else if( !strcmp( sub->codecName(), "VORBIS" ) ) { tk->fmt.i_codec = VLC_CODEC_VORBIS; unsigned int i_extra; unsigned char *p_extra; if( ( p_extra=parseVorbisConfigStr( sub->fmtp_config(), i_extra ) ) ) { tk->fmt.i_extra = i_extra; tk->fmt.p_extra = p_extra; } else msg_Warn( p_demux,"Missing or unsupported vorbis header." ); } else if( !strcmp( sub->codecName(), "OPUS" ) ) { int i_extra; unsigned char *p_extra; tk->fmt.i_codec = VLC_CODEC_OPUS; OpusHeader header; opus_header_init(&header); // "The RTP clock rate in "a=rtpmap" MUST be 48000 and the number of channels MUST be 2." // See: https://datatracker.ietf.org/doc/html/draft-ietf-payload-rtp-opus-11#section-7 opus_prepare_header( 2, 48000, &header ); if( opus_write_header( &p_extra, &i_extra, &header, NULL ) ) return VLC_ENOMEM; opus_header_clean(&header); tk->fmt.i_extra = i_extra; tk->fmt.p_extra = p_extra; } } else if( !strcmp( sub->mediumName(), "video" ) ) { es_format_Init( &tk->fmt, VIDEO_ES, VLC_CODEC_UNKNOWN ); if( !strcmp( sub->codecName(), "MPV" ) ) { tk->fmt.i_codec = VLC_CODEC_MPGV; tk->fmt.b_packetized = false; } else if( !strcmp( sub->codecName(), "H263" ) || !strcmp( sub->codecName(), "H263-1998" ) || !strcmp( sub->codecName(), "H263-2000" ) ) { tk->fmt.i_codec = VLC_CODEC_H263; } else if( !strcmp( sub->codecName(), "H261" ) ) { tk->fmt.i_codec = VLC_CODEC_H261; } else if( !strcmp( sub->codecName(), "H264" ) ) { unsigned int i_extra = 0; uint8_t *p_extra = NULL; tk->fmt.i_codec = VLC_CODEC_H264; tk->fmt.b_packetized = false; if((p_extra=parseH264ConfigStr( sub->fmtp_spropparametersets(), i_extra ) ) ) { tk->fmt.i_extra = i_extra; tk->fmt.p_extra = xmalloc( i_extra ); memcpy( tk->fmt.p_extra, p_extra, i_extra ); delete[] p_extra; } } #if LIVEMEDIA_LIBRARY_VERSION_INT >= 1393372800 // 2014.02.26 else if( !strcmp( sub->codecName(), "H265" ) ) { unsigned int i_extra1 = 0, i_extra2 = 0, i_extra3 = 0, i_extraTot; uint8_t *p_extra1 = NULL, *p_extra2 = NULL, *p_extra3 = NULL; tk->fmt.i_codec = VLC_CODEC_HEVC; tk->fmt.b_packetized = false; p_extra1 = parseH264ConfigStr( sub->fmtp_spropvps(), i_extra1 ); p_extra2 = parseH264ConfigStr( sub->fmtp_spropsps(), i_extra2 ); p_extra3 = parseH264ConfigStr( sub->fmtp_sproppps(), i_extra3 ); i_extraTot = i_extra1 + i_extra2 + i_extra3; if( i_extraTot > 0 ) { tk->fmt.i_extra = i_extraTot; tk->fmt.p_extra = xmalloc( i_extraTot ); if( p_extra1 ) { memcpy( tk->fmt.p_extra, p_extra1, i_extra1 ); } if( p_extra2 ) { memcpy( ((char*)tk->fmt.p_extra)+i_extra1, p_extra2, i_extra2 ); } if( p_extra3 ) { memcpy( ((char*)tk->fmt.p_extra)+i_extra1+i_extra2, p_extra3, i_extra3 ); } delete[] p_extra1; delete[] p_extra2; delete[] p_extra3; } } #endif else if( !strcmp( sub->codecName(), "JPEG" ) ) { tk->fmt.i_codec = VLC_CODEC_MJPG; } else if( !strcmp( sub->codecName(), "MP4V-ES" ) ) { unsigned int i_extra; uint8_t *p_extra; tk->fmt.i_codec = VLC_CODEC_MP4V; if( ( p_extra = parseGeneralConfigStr( sub->fmtp_config(), i_extra ) ) ) { tk->fmt.i_extra = i_extra; tk->fmt.p_extra = xmalloc( i_extra ); memcpy( tk->fmt.p_extra, p_extra, i_extra ); delete[] p_extra; } } else if( !strcmp( sub->codecName(), "X-QT" ) || !strcmp( sub->codecName(), "X-QUICKTIME" ) || !strcmp( sub->codecName(), "X-QDM" ) || !strcmp( sub->codecName(), "X-SV3V-ES" ) || !strcmp( sub->codecName(), "X-SORENSONVIDEO" ) ) { tk->format = live_track_t::QUICKTIME_STREAM; } else if( !strcmp( sub->codecName(), "MP2T" ) ) { tk->format = live_track_t::MULTIPLEXED_STREAM; tk->p_out_muxed = vlc_demux_chained_New( VLC_OBJECT(p_demux), "ts", p_demux->out ); } else if( !strcmp( sub->codecName(), "MP2P" ) || !strcmp( sub->codecName(), "MP1S" ) ) { tk->format = live_track_t::MULTIPLEXED_STREAM; tk->p_out_muxed = vlc_demux_chained_New( VLC_OBJECT(p_demux), "ps", p_demux->out ); } else if( !strcmp( sub->codecName(), "X-ASF-PF" ) ) { tk->format = live_track_t::ASF_STREAM; if( p_sys->p_out_asf == NULL ) p_sys->p_out_asf = vlc_demux_chained_New( VLC_OBJECT(p_demux), "asf", p_demux->out ); } else if( !strcmp( sub->codecName(), "DV" ) ) { tk->format = live_track_t::MULTIPLEXED_STREAM; tk->b_discard_trunc = true; tk->p_out_muxed = vlc_demux_chained_New( VLC_OBJECT(p_demux), "rawdv", p_demux->out ); } else if( !strcmp( sub->codecName(), "VP8" ) ) { tk->fmt.i_codec = VLC_CODEC_VP8; } else if( !strcmp( sub->codecName(), "THEORA" ) ) { tk->fmt.i_codec = VLC_CODEC_THEORA; unsigned int i_extra; unsigned char *p_extra; if( ( p_extra=parseVorbisConfigStr( sub->fmtp_config(), i_extra ) ) ) { tk->fmt.i_extra = i_extra; tk->fmt.p_extra = p_extra; } else msg_Warn( p_demux,"Missing or unsupported theora header." ); } } else if( !strcmp( sub->mediumName(), "text" ) ) { es_format_Init( &tk->fmt, SPU_ES, VLC_CODEC_UNKNOWN ); if( !strcmp( sub->codecName(), "T140" ) ) { tk->fmt.i_codec = VLC_CODEC_ITU_T140; } } /* Try and parse a=lang: attribute */ p_lang = strstr( sub->savedSDPLines(), "a=lang:" ); if( !p_lang ) p_lang = p_sess_lang; if( p_lang ) { unsigned i_lang_len; p_lang += 7; i_lang_len = strcspn( p_lang, " \r\n" ); tk->fmt.psz_language = strndup( p_lang, i_lang_len ); } if( tk->format == live_track_t::SINGLE_STREAM ) { tk->p_es = es_out_Add( p_demux->out, &tk->fmt ); } if( sub->rtcpInstance() != NULL ) { sub->rtcpInstance()->setByeHandler( StreamClose, tk ); } if( tk->p_es || tk->format == live_track_t::QUICKTIME_STREAM || (tk->format == live_track_t::MULTIPLEXED_STREAM && tk->p_out_muxed ) || (tk->format == live_track_t::ASF_STREAM && p_sys->p_out_asf ) ) { TAB_APPEND_CAST( (live_track_t **), p_sys->i_track, p_sys->track, tk ); } else { /* BUG ??? */ msg_Err( p_demux, "unusable RTSP track. this should not happen" ); es_format_Clean( &tk->fmt ); free( tk ); } } } delete iter; if( p_sys->i_track <= 0 ) i_return = VLC_EGENERIC; /* Retrieve the starttime if possible */ p_sys->f_npt_start = p_sys->ms->playStartTime(); /* Retrieve the duration if possible */ p_sys->f_npt_length = p_sys->ms->playEndTime(); /* */ msg_Dbg( p_demux, "setup start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length ); /* */ p_sys->b_no_data = true; p_sys->i_no_data_ti = 0; p_sys->b_rtcp_sync = false; p_sys->i_pcr = VLC_TICK_INVALID; return i_return; } /***************************************************************************** * Play: starts the actual playback of the stream *****************************************************************************/ static int Play( demux_t *p_demux ) { demux_sys_t *p_sys = p_demux->p_sys; if( p_sys->rtsp ) { /* The PLAY */ p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, p_sys->f_npt_start, -1, 1 ); if( !wait_Live555_response(p_demux) ) { msg_Err( p_demux, "RTSP PLAY failed %s", p_sys->env->getResultMsg() ); return VLC_EGENERIC; } /* Retrieve the timeout value and set up a timeout prevention thread */ int timeout = p_sys->rtsp->sessionTimeoutParameter(); if( timeout <= 2 ) timeout = 60; /* default value from RFC2326 */ msg_Dbg( p_demux, "We have a timeout of %d seconds", timeout ); vlc_tick_t interval = (timeout - 2) * CLOCK_FREQ; vlc_timer_schedule( p_sys->timer, false, interval, interval); } p_sys->i_pcr = VLC_TICK_INVALID; /* Retrieve the starttime if possible */ p_sys->f_npt_start = p_sys->ms->playStartTime(); if( p_sys->ms->playEndTime() > 0 ) p_sys->f_npt_length = p_sys->ms->playEndTime(); msg_Dbg( p_demux, "play start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length ); return VLC_SUCCESS; } /***************************************************************************** * HasSharedSession: returns if the session is shared with another stream *****************************************************************************/ static bool HasSharedSession( MediaSubsession *session ) { if( session->sessionId() == NULL ) return false; MediaSubsessionIterator *it = new MediaSubsessionIterator( session->parentSession() ); MediaSubsession *subsession; bool b_shared = false; while( (subsession = it->next()) != NULL ) { if( session == subsession ) continue; if( subsession->sessionId() != NULL && !strcmp( session->sessionId(), subsession->sessionId() ) ) { b_shared = true; break; } } delete it; return b_shared; } /***************************************************************************** * ResumeTrack: setup or resume a silenced track *****************************************************************************/ static void ResumeTrack( demux_t *p_demux, live_track_t *tk ) { demux_sys_t *p_sys = p_demux->p_sys; bool b_rtsp_tcp = var_GetBool( p_demux, "rtsp-tcp" ) || var_GetBool( p_demux, "rtsp-http" ); p_sys->rtsp->sendSetupCommand( *tk->sub, default_live555_callback, False, toBool( b_rtsp_tcp ), toBool( p_sys->b_force_mcast && !b_rtsp_tcp ) ); if( !wait_Live555_response( p_demux ) ) { msg_Err( p_demux, "SETUP of'%s/%s' failed %s", tk->sub->mediumName(), tk->sub->codecName(), p_sys->env->getResultMsg() ); } else { p_sys->rtsp->sendPlayCommand( *tk->sub, default_live555_callback, -1, -1, p_sys->ms->scale() ); if( !wait_Live555_response(p_demux) ) { msg_Err( p_demux, "RTSP PLAY failed %s", p_sys->env->getResultMsg() ); if( (p_sys->capabilities & CAP_SUBSESSION_TEARDOWN) || !HasSharedSession( tk->sub ) ) { tk->state = live_track_t::STATE_TEARDOWN; p_sys->rtsp->sendTeardownCommand( *tk->sub, NULL ); } } else tk->state = live_track_t::STATE_SELECTED; } } /***************************************************************************** * Demux: *****************************************************************************/ static int Demux( demux_t *p_demux ) { demux_sys_t *p_sys = p_demux->p_sys; TaskToken task; bool b_send_pcr = true; int i; /* Protect Live555 from simultaneous calls in TimeoutPrevention() during pause */ vlc_mutex_locker locker(&p_sys->timeout_mutex); for( i = 0; i < p_sys->i_track; i++ ) { live_track_t *tk = p_sys->track[i]; if( tk->p_es ) { bool b; es_out_Control( p_demux->out, ES_OUT_GET_ES_STATE, tk->p_es, &b ); if( !b && (tk->state == live_track_t::STATE_SELECTED) && p_sys->rtsp ) { if( (p_sys->capabilities & CAP_SUBSESSION_TEARDOWN) || !HasSharedSession( tk->sub ) ) { tk->state = live_track_t::STATE_TEARDOWN; p_sys->rtsp->sendTeardownCommand( *tk->sub, NULL ); } else tk->state = live_track_t::STATE_IGNORED; } else if( b && tk->state != live_track_t::STATE_SELECTED ) { if( tk->state != live_track_t::STATE_IGNORED ) ResumeTrack( p_demux, tk ); else tk->state = live_track_t::STATE_SELECTED; if( tk->state != live_track_t::STATE_SELECTED ) es_out_Control( p_demux->out, ES_OUT_SET_ES_STATE, tk->p_es, false ); } } if( tk->format == live_track_t::ASF_STREAM || tk->format == live_track_t::MULTIPLEXED_STREAM ) { b_send_pcr = false; } } /* First warn we want to read data */ p_sys->event_data = 0; for( i = 0; i < p_sys->i_track; i++ ) { live_track_t *tk = p_sys->track[i]; if( tk->waiting == 0 ) { tk->waiting = 1; tk->sub->readSource()->getNextFrame( tk->p_buffer, tk->i_buffer, StreamRead, tk, StreamClose, tk ); } } /* Create a task that will be called if we wait more than 300ms */ task = p_sys->scheduler->scheduleDelayedTask( 300000, TaskInterruptData, p_demux ); /* Do the read */ p_sys->scheduler->doEventLoop( &p_sys->event_data ); /* remove the task */ p_sys->scheduler->unscheduleDelayedTask( task ); if( b_send_pcr ) { vlc_tick_t i_minpcr = VLC_TICK_INVALID; bool b_need_flush = false; /* Check for gap in pts value */ for( i = 0; i < p_sys->i_track; i++ ) { live_track_t *tk = p_sys->track[i]; if( tk->state != live_track_t::STATE_SELECTED || (p_sys->b_rtcp_sync && !tk->b_rtcp_sync) ) continue; /* Check for gap in pts value */ b_need_flush |= (tk->b_flushing_discontinuity); if( i_minpcr == VLC_TICK_INVALID || ( tk->i_pcr != VLC_TICK_INVALID && i_minpcr > tk->i_pcr ) ) i_minpcr = tk->i_pcr; } if( p_sys->i_pcr > VLC_TICK_INVALID && b_need_flush ) { es_out_Control( p_demux->out, ES_OUT_RESET_PCR ); p_sys->i_pcr = i_minpcr; p_sys->f_npt = 0.; for( i = 0; i < p_sys->i_track; i++ ) { live_track_t *tk = p_sys->track[i]; tk->i_prevpts = VLC_TICK_INVALID; tk->i_pcr = VLC_TICK_INVALID; tk->f_npt = 0.; tk->b_flushing_discontinuity = false; tk->i_next_block_flags |= BLOCK_FLAG_DISCONTINUITY; } if( p_sys->i_pcr != VLC_TICK_INVALID ) es_out_SetPCR( p_demux->out, VLC_TICK_0 + __MAX(0, p_sys->i_pcr - PCR_OFF) ); } else if( p_sys->i_pcr == VLC_TICK_INVALID || i_minpcr > p_sys->i_pcr + PCR_OBS ) { p_sys->i_pcr = __MAX(0, i_minpcr - PCR_OFF); if( p_sys->i_pcr != VLC_TICK_INVALID ) es_out_SetPCR( p_demux->out, VLC_TICK_0 + p_sys->i_pcr ); } } if( p_sys->b_multicast && p_sys->b_no_data && ( p_sys->i_no_data_ti > 120 ) ) { /* FIXME Make this configurable msg_Err( p_demux, "no multicast data received in 36s, aborting" ); return 0; */ } else if( !p_sys->b_multicast && !p_sys->b_paused && p_sys->b_no_data && ( p_sys->i_no_data_ti > 34 ) ) { bool b_rtsp_tcp = var_GetBool( p_demux, "rtsp-tcp" ) || var_GetBool( p_demux, "rtsp-http" ); if( !b_rtsp_tcp && p_sys->rtsp && p_sys->ms ) { msg_Warn( p_demux, "no data received in 10s. Switching to TCP" ); if( RollOverTcp( p_demux ) ) { msg_Err( p_demux, "TCP rollover failed, aborting" ); return 0; } return 1; } msg_Err( p_demux, "no data received in 10s, aborting" ); return 0; } else if( !p_sys->b_multicast && !p_sys->b_paused && ( p_sys->i_no_data_ti > 34 ) ) { /* EOF ? */ msg_Warn( p_demux, "no data received in 10s, eof ?" ); return 0; } return p_sys->b_error ? 0 : 1; } /***************************************************************************** * Control: *****************************************************************************/ static int Control( demux_t *p_demux, int i_query, va_list args ) { demux_sys_t *p_sys = p_demux->p_sys; int64_t *pi64, i64; double *pf, f; bool *pb; vlc_mutex_locker locker(&p_sys->timeout_mutex); /* (see same in Demux) */ switch( i_query ) { case DEMUX_GET_TIME: pi64 = va_arg( args, int64_t * ); if( p_sys->f_npt > 0 ) { *pi64 = (int64_t)(p_sys->f_npt * 1000000.); return VLC_SUCCESS; } return VLC_EGENERIC; case DEMUX_GET_LENGTH: pi64 = va_arg( args, int64_t * ); if( p_sys->f_npt_length > 0 ) { double d_length = p_sys->f_npt_length * 1000000.0; if( d_length >= INT64_MAX ) *pi64 = INT64_MAX; else *pi64 = (int64_t)d_length; return VLC_SUCCESS; } return VLC_EGENERIC; case DEMUX_GET_POSITION: pf = va_arg( args, double * ); if( (p_sys->f_npt_length > 0) && (p_sys->f_npt > 0) ) { *pf = p_sys->f_npt / p_sys->f_npt_length; return VLC_SUCCESS; } return VLC_EGENERIC; case DEMUX_SET_POSITION: case DEMUX_SET_TIME: if( p_sys->rtsp && (p_sys->f_npt_length > 0) ) { float time; if( (i_query == DEMUX_SET_TIME) && (p_sys->f_npt > 0) ) { i64 = va_arg( args, int64_t ); time = (float)(i64 / 1000000.0); /* in second */ } else if( i_query == DEMUX_SET_TIME ) return VLC_EGENERIC; else { f = va_arg( args, double ); time = f * p_sys->f_npt_length; /* in second */ } if( p_sys->b_paused ) { p_sys->f_seek_request = time; return VLC_SUCCESS; } p_sys->rtsp->sendPauseCommand( *p_sys->ms, default_live555_callback ); if( !wait_Live555_response( p_demux ) ) { msg_Err( p_demux, "PAUSE before seek failed %s", p_sys->env->getResultMsg() ); return VLC_EGENERIC; } p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, time, -1, 1 ); if( !wait_Live555_response( p_demux ) ) { msg_Err( p_demux, "seek PLAY failed %s", p_sys->env->getResultMsg() ); return VLC_EGENERIC; } p_sys->i_pcr = VLC_TICK_INVALID; for( int i = 0; i < p_sys->i_track; i++ ) { p_sys->track[i]->b_rtcp_sync = false; p_sys->track[i]->i_prevpts = VLC_TICK_INVALID; p_sys->track[i]->i_pcr = VLC_TICK_INVALID; dtsgen_Resync( &p_sys->track[i]->dtsgen ); } /* Retrieve the starttime if possible */ p_sys->f_npt = p_sys->f_npt_start = p_sys->ms->playStartTime(); /* Retrieve the duration if possible */ if( p_sys->ms->playEndTime() > 0 ) p_sys->f_npt_length = p_sys->ms->playEndTime(); msg_Dbg( p_demux, "seek start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length ); return VLC_SUCCESS; } return VLC_EGENERIC; /* Special for access_demux */ case DEMUX_CAN_PAUSE: case DEMUX_CAN_SEEK: pb = va_arg( args, bool * ); if( p_sys->rtsp && p_sys->f_npt_length > 0 ) /* Not always true, but will be handled in SET_PAUSE_STATE */ *pb = true; else *pb = false; return VLC_SUCCESS; case DEMUX_CAN_CONTROL_PACE: pb = va_arg( args, bool * ); #if 1 /* Disable for now until we have a clock synchro algo * which works with something else than MPEG over UDP */ *pb = false; #else *pb = true; #endif return VLC_SUCCESS; case DEMUX_CAN_CONTROL_RATE: pb = va_arg( args, bool * ); *pb = (p_sys->rtsp != NULL) && (p_sys->f_npt_length > 0) && (p_sys->capabilities & CAP_RATE_CONTROL); return VLC_SUCCESS; case DEMUX_SET_RATE: { int *pi_int; double f_scale, f_old_scale; if( !p_sys->rtsp || (p_sys->f_npt_length <= 0) || !(p_sys->capabilities & CAP_RATE_CONTROL) ) return VLC_EGENERIC; /* According to RFC 2326 p56 chapter 12.35 a RTSP server that * supports Scale: * * "[...] should try to approximate the viewing rate, but * may restrict the range of scale values that it supports. * The response MUST contain the actual scale value chosen * by the server." * * Scale = 1 indicates normal play * Scale > 1 indicates fast forward * Scale < 1 && Scale > 0 indicates slow motion * Scale < 0 value indicates rewind */ pi_int = va_arg( args, int * ); f_scale = (double)INPUT_RATE_DEFAULT / (*pi_int); f_old_scale = p_sys->ms->scale(); /* Passing -1 for the start and end time will mean liveMedia won't * create a Range: section for the RTSP message. The server should * pick up from the current position */ p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, -1, -1, f_scale ); if( !wait_Live555_response( p_demux ) ) { msg_Err( p_demux, "PLAY with Scale %0.2f failed %s", f_scale, p_sys->env->getResultMsg() ); return VLC_EGENERIC; } if( p_sys->ms->scale() == f_old_scale ) { msg_Err( p_demux, "no scale change using old Scale %0.2f", p_sys->ms->scale() ); return VLC_EGENERIC; } /* ReSync the stream */ p_sys->f_npt_start = 0; p_sys->i_pcr = VLC_TICK_INVALID; p_sys->f_npt = 0.0; *pi_int = (int)( INPUT_RATE_DEFAULT / p_sys->ms->scale() ); msg_Dbg( p_demux, "PLAY with new Scale %0.2f (%d)", p_sys->ms->scale(), (*pi_int) ); return VLC_SUCCESS; } case DEMUX_SET_PAUSE_STATE: { bool b_pause = (bool)va_arg( args, int ); if( p_sys->rtsp == NULL ) return VLC_EGENERIC; if( b_pause == p_sys->b_paused ) return VLC_SUCCESS; if( b_pause ) p_sys->rtsp->sendPauseCommand( *p_sys->ms, default_live555_callback ); else p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, p_sys->f_seek_request, -1.0f, p_sys->ms->scale() ); if( !wait_Live555_response( p_demux ) ) { msg_Err( p_demux, "PLAY or PAUSE failed %s", p_sys->env->getResultMsg() ); return VLC_EGENERIC; } p_sys->f_seek_request = -1; p_sys->b_paused = b_pause; if( !p_sys->b_paused ) { for( int i = 0; i < p_sys->i_track; i++ ) { live_track_t *tk = p_sys->track[i]; tk->b_rtcp_sync = false; tk->b_flushing_discontinuity = false; tk->i_next_block_flags |= BLOCK_FLAG_DISCONTINUITY; tk->i_prevpts = VLC_TICK_INVALID; tk->i_pcr = VLC_TICK_INVALID; } p_sys->i_pcr = VLC_TICK_INVALID; es_out_Control( p_demux->out, ES_OUT_RESET_PCR ); } /* Reset data received counter */ p_sys->i_no_data_ti = 0; /* Retrieve the starttime if possible */ p_sys->f_npt_start = p_sys->ms->playStartTime(); /* Retrieve the duration if possible */ if( p_sys->ms->playEndTime() ) p_sys->f_npt_length = p_sys->ms->playEndTime(); msg_Dbg( p_demux, "pause start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length ); return VLC_SUCCESS; } case DEMUX_GET_TITLE_INFO: case DEMUX_SET_TITLE: case DEMUX_SET_SEEKPOINT: return VLC_EGENERIC; case DEMUX_GET_PTS_DELAY: pi64 = va_arg( args, int64_t * ); *pi64 = INT64_C(1000) * var_InheritInteger( p_demux, "network-caching" ); return VLC_SUCCESS; default: return VLC_EGENERIC; } } /***************************************************************************** * RollOverTcp: reopen the rtsp into TCP mode * XXX: ugly, a lot of code are duplicated from Open() * This should REALLY be fixed *****************************************************************************/ static int RollOverTcp( demux_t *p_demux ) { demux_sys_t *p_sys = p_demux->p_sys; int i, i_return; var_SetBool( p_demux, "rtsp-tcp", true ); /* We close the old RTSP session */ vlc_timer_schedule(p_sys->timer, false, 0, 0); p_sys->rtsp->sendTeardownCommand( *p_sys->ms, NULL ); Medium::close( p_sys->ms ); RTSPClient::close( p_sys->rtsp ); for( i = 0; i < p_sys->i_track; i++ ) { live_track_t *tk = p_sys->track[i]; if( tk->p_out_muxed ) vlc_demux_chained_Delete( tk->p_out_muxed ); if( tk->p_es ) es_out_Del( p_demux->out, tk->p_es ); if( tk->p_asf_block ) block_Release( tk->p_asf_block ); es_format_Clean( &tk->fmt ); free( tk->p_buffer ); free( tk ); } TAB_CLEAN( p_sys->i_track, p_sys->track ); if( p_sys->p_out_asf ) vlc_demux_chained_Delete( p_sys->p_out_asf ); p_sys->ms = NULL; p_sys->rtsp = NULL; p_sys->b_no_data = true; p_sys->i_no_data_ti = 0; p_sys->p_out_asf = NULL; /* Reopen rtsp client */ if( ( i_return = Connect( p_demux ) ) != VLC_SUCCESS ) { msg_Err( p_demux, "Failed to connect with %s", p_sys->psz_pl_url ); goto error; } if( p_sys->p_sdp == NULL ) { msg_Err( p_demux, "Failed to retrieve the RTSP Session Description" ); goto error; } if( ( i_return = SessionsSetup( p_demux ) ) != VLC_SUCCESS ) { msg_Err( p_demux, "Nothing to play for %s", p_sys->psz_pl_url ); goto error; } if( ( i_return = Play( p_demux ) ) != VLC_SUCCESS ) goto error; return VLC_SUCCESS; error: return VLC_EGENERIC; } /***************************************************************************** * *****************************************************************************/ static block_t *StreamParseAsf( demux_t *p_demux, live_track_t *tk, bool b_marker, const uint8_t *p_data, unsigned i_size ) { const unsigned i_packet_size = p_demux->p_sys->asfh.i_min_data_packet_size; block_t *p_list = NULL; while( i_size >= 4 ) { unsigned i_flags = p_data[0]; unsigned i_length_offset = (p_data[1] << 16) | (p_data[2] << 8) | (p_data[3] ); bool b_length = i_flags & 0x40; bool b_relative_ts = i_flags & 0x20; bool b_duration = i_flags & 0x10; bool b_location_id = i_flags & 0x08; //msg_Dbg( p_demux, "ASF: marker=%d size=%d : %c=%d id=%d", // b_marker, i_size, b_length ? 'L' : 'O', i_length_offset ); unsigned i_header_size = 4; if( b_relative_ts ) i_header_size += 4; if( b_duration ) i_header_size += 4; if( b_location_id ) i_header_size += 4; if( i_header_size > i_size ) { msg_Warn( p_demux, "Invalid header size" ); break; } /* XXX * When b_length is true, the streams I found do not seems to respect * the documentation. * From them, I have failed to find which choice between '__MIN()' or * 'i_length_offset - i_header_size' is the right one. */ unsigned i_payload; if( b_length ) i_payload = __MIN( i_length_offset, i_size - i_header_size); else i_payload = i_size - i_header_size; if( !tk->p_asf_block ) { tk->p_asf_block = block_Alloc( i_packet_size ); if( !tk->p_asf_block ) break; tk->p_asf_block->i_buffer = 0; } unsigned i_offset = b_length ? 0 : i_length_offset; if( i_offset == tk->p_asf_block->i_buffer && i_offset + i_payload <= i_packet_size ) { memcpy( &tk->p_asf_block->p_buffer[i_offset], &p_data[i_header_size], i_payload ); tk->p_asf_block->i_buffer += i_payload; if( b_marker ) { /* We have a complete packet */ tk->p_asf_block->i_buffer = i_packet_size; block_ChainAppend( &p_list, tk->p_asf_block ); tk->p_asf_block = NULL; } } else { /* Reset on broken stream */ msg_Err( p_demux, "Broken packet detected (%d vs %zu or %d + %d vs %d)", i_offset, tk->p_asf_block->i_buffer, i_offset, i_payload, i_packet_size); tk->p_asf_block->i_buffer = 0; } /* */ p_data += i_header_size + i_payload; i_size -= i_header_size + i_payload; } return p_list; } /***************************************************************************** * *****************************************************************************/ static void StreamRead( void *p_private, unsigned int i_size, unsigned int i_truncated_bytes, struct timeval pts, unsigned int duration ) { VLC_UNUSED( duration ); live_track_t *tk = (live_track_t*)p_private; demux_t *p_demux = tk->p_demux; demux_sys_t *p_sys = p_demux->p_sys; block_t *p_block; //msg_Dbg( p_demux, "pts: %d", pts.tv_sec ); int64_t i_pts = (int64_t)pts.tv_sec * INT64_C(1000000) + (int64_t)pts.tv_usec; /* XXX Beurk beurk beurk Avoid having negative value XXX */ i_pts &= INT64_C(0x00ffffffffffffff); /* Retrieve NPT for this pts */ tk->f_npt = tk->sub->getNormalPlayTime(pts); if( tk->format == live_track_t::QUICKTIME_STREAM && tk->p_es == NULL ) { QuickTimeGenericRTPSource *qtRTPSource = (QuickTimeGenericRTPSource*)tk->sub->rtpSource(); QuickTimeGenericRTPSource::QTState &qtState = qtRTPSource->qtState; uint8_t *sdAtom = (uint8_t*)&qtState.sdAtom[4]; /* Get codec information from the quicktime atoms : * http://developer.apple.com/quicktime/icefloe/dispatch026.html */ if( tk->fmt.i_cat == VIDEO_ES ) { if( qtState.sdAtomSize < 16 + 32 ) { /* invalid */ p_sys->event_data = 0xff; tk->waiting = 0; return; } tk->fmt.i_codec = VLC_FOURCC(sdAtom[0],sdAtom[1],sdAtom[2],sdAtom[3]); tk->fmt.video.i_width = (sdAtom[28] << 8) | sdAtom[29]; tk->fmt.video.i_height = (sdAtom[30] << 8) | sdAtom[31]; if( tk->fmt.i_codec == VLC_FOURCC('a', 'v', 'c', '1') ) { uint8_t *pos = (uint8_t*)qtRTPSource->qtState.sdAtom + 86; uint8_t *endpos = (uint8_t*)qtRTPSource->qtState.sdAtom + qtRTPSource->qtState.sdAtomSize; while (pos+8 < endpos) { unsigned int atomLength = pos[0]<<24 | pos[1]<<16 | pos[2]<<8 | pos[3]; if( atomLength == 0 || atomLength > (unsigned int)(endpos-pos)) break; if( memcmp(pos+4, "avcC", 4) == 0 && atomLength > 8 && atomLength <= INT_MAX ) { tk->fmt.i_extra = atomLength-8; tk->fmt.p_extra = xmalloc( tk->fmt.i_extra ); memcpy(tk->fmt.p_extra, pos+8, atomLength-8); break; } pos += atomLength; } } else { tk->fmt.i_extra = qtState.sdAtomSize - 16; tk->fmt.p_extra = xmalloc( tk->fmt.i_extra ); memcpy( tk->fmt.p_extra, &sdAtom[12], tk->fmt.i_extra ); } } else { if( qtState.sdAtomSize < 24 ) { /* invalid */ p_sys->event_data = 0xff; tk->waiting = 0; return; } tk->fmt.i_codec = VLC_FOURCC(sdAtom[0],sdAtom[1],sdAtom[2],sdAtom[3]); tk->fmt.audio.i_bitspersample = (sdAtom[22] << 8) | sdAtom[23]; } tk->p_es = es_out_Add( p_demux->out, &tk->fmt ); } #if 0 fprintf( stderr, "StreamRead size=%d pts=%lld\n", i_size, pts.tv_sec * 1000000LL + pts.tv_usec ); #endif /* grow buffer if it looks like buffer is too small, but don't eat * up all the memory on strange streams */ if( i_truncated_bytes > 0 ) { if( tk->i_buffer < 2000000 ) { void *p_tmp; msg_Dbg( p_demux, "lost %d bytes", i_truncated_bytes ); msg_Dbg( p_demux, "increasing buffer size to %d", tk->i_buffer * 2 ); p_tmp = realloc( tk->p_buffer, tk->i_buffer * 2 ); if( p_tmp == NULL ) { msg_Warn( p_demux, "realloc failed" ); } else { tk->p_buffer = (uint8_t*)p_tmp; tk->i_buffer *= 2; } } if( tk->b_discard_trunc ) { p_sys->event_data = 0xff; tk->waiting = 0; return; } } assert( i_size <= tk->i_buffer ); if( tk->fmt.i_codec == VLC_CODEC_AMR_NB || tk->fmt.i_codec == VLC_CODEC_AMR_WB ) { AMRAudioSource *amrSource = (AMRAudioSource*)tk->sub->readSource(); if( (p_block = block_Alloc( i_size + 1 )) ) { p_block->p_buffer[0] = amrSource->lastFrameHeader(); memcpy( p_block->p_buffer + 1, tk->p_buffer, i_size ); } } else if( tk->fmt.i_codec == VLC_CODEC_H261 ) { H261VideoRTPSource *h261Source = (H261VideoRTPSource*)tk->sub->rtpSource(); uint32_t header = h261Source->lastSpecialHeader(); if( (p_block = block_Alloc( i_size + 4 )) ) { memcpy( p_block->p_buffer, &header, 4 ); memcpy( p_block->p_buffer + 4, tk->p_buffer, i_size ); } } else if( tk->fmt.i_codec == VLC_CODEC_H264 || tk->fmt.i_codec == VLC_CODEC_HEVC ) { if( tk->fmt.i_codec == VLC_CODEC_H264 && (tk->p_buffer[0] & 0x1f) >= 24 ) msg_Warn( p_demux, "unsupported NAL type for H264" ); else if( tk->fmt.i_codec == VLC_CODEC_HEVC && ((tk->p_buffer[0] & 0x7e)>>1) >= 48 ) msg_Warn( p_demux, "unsupported NAL type for H265" ); /* Normal NAL type */ if( (p_block = block_Alloc( i_size + 4 )) ) { p_block->p_buffer[0] = 0x00; p_block->p_buffer[1] = 0x00; p_block->p_buffer[2] = 0x00; p_block->p_buffer[3] = 0x01; memcpy( &p_block->p_buffer[4], tk->p_buffer, i_size ); } } else if( tk->format == live_track_t::ASF_STREAM ) { p_block = StreamParseAsf( p_demux, tk, tk->sub->rtpSource()->curPacketMarkerBit(), tk->p_buffer, i_size ); } else { if( (p_block = block_Alloc( i_size )) ) memcpy( p_block->p_buffer, tk->p_buffer, i_size ); } /* No data sent. Always in sync then */ if( !tk->b_rtcp_sync && tk->sub->rtpSource() && tk->sub->rtpSource()->hasBeenSynchronizedUsingRTCP() ) { msg_Dbg( p_demux, "tk->rtpSource->hasBeenSynchronizedUsingRTCP()" ); p_sys->b_rtcp_sync = tk->b_rtcp_sync = true; if( tk->i_pcr != VLC_TICK_INVALID ) { tk->i_next_block_flags |= BLOCK_FLAG_DISCONTINUITY; const int64_t i_max_diff = CLOCK_FREQ * (( tk->fmt.i_cat == SPU_ES ) ? 60 : 1); tk->b_flushing_discontinuity = (llabs(i_pts - tk->i_pcr) > i_max_diff); tk->i_pcr = i_pts; tk->dtsgen.count = 0; } } /* Update our global npt value */ if( tk->f_npt > 0 && ( tk->f_npt < p_sys->f_npt_length || p_sys->f_npt_length <= 0 ) ) p_sys->f_npt = tk->f_npt; if( p_block ) { switch( tk->format ) { case live_track_t::ASF_STREAM: vlc_demux_chained_Send( p_sys->p_out_asf, p_block ); break; case live_track_t::MULTIPLEXED_STREAM: vlc_demux_chained_Send( tk->p_out_muxed, p_block ); break; default: if( i_pts != tk->i_prevpts ) { p_block->i_pts = VLC_TICK_0 + i_pts; tk->i_prevpts = i_pts; dtsgen_AddNextPTS( &tk->dtsgen, i_pts ); } /*FIXME: for h264 you should check that packetization-mode=1 in sdp-file */ switch( tk->fmt.i_codec ) { case VLC_CODEC_MPGV: case VLC_CODEC_H264: case VLC_CODEC_HEVC: p_block->i_dts = dtsgen_GetDTS( &tk->dtsgen ); dtsgen_Debug( VLC_OBJECT(p_demux), &tk->dtsgen, p_block->i_dts, p_block->i_pts ); break; case VLC_CODEC_VP8: default: p_block->i_dts = VLC_TICK_0 + i_pts; break; } if( i_truncated_bytes ) p_block->i_flags |= BLOCK_FLAG_CORRUPTED; if( unlikely(tk->i_next_block_flags) ) { p_block->i_flags |= tk->i_next_block_flags; tk->i_next_block_flags = 0; } vlc_tick_t i_pcr = p_block->i_dts > VLC_TICK_INVALID ? p_block->i_dts : p_block->i_pts; es_out_Send( p_demux->out, tk->p_es, p_block ); if( i_pcr > VLC_TICK_INVALID ) { if( tk->i_pcr < i_pcr ) tk->i_pcr = i_pcr; } break; } } /* warn that's ok */ p_sys->event_data = 0xff; /* we have read data */ tk->waiting = 0; p_demux->p_sys->b_no_data = false; p_demux->p_sys->i_no_data_ti = 0; } /***************************************************************************** * *****************************************************************************/ static void StreamClose( void *p_private ) { live_track_t *tk = (live_track_t*)p_private; demux_t *p_demux = tk->p_demux; demux_sys_t *p_sys = p_demux->p_sys; tk->state = live_track_t::STATE_IGNORED; p_sys->event_rtsp = 0xff; p_sys->event_data = 0xff; if( tk->p_es ) es_out_Control( p_demux->out, ES_OUT_SET_ES_STATE, tk->p_es, false ); int nb_tracks = 0; for( int i = 0; i < p_sys->i_track; i++ ) { if( p_sys->track[i]->state == live_track_t::STATE_SELECTED ) nb_tracks++; } msg_Dbg( p_demux, "RTSP track Close, %d track remaining", nb_tracks ); if( !nb_tracks ) p_sys->b_error = true; } /***************************************************************************** * *****************************************************************************/ static void TaskInterruptRTSP( void *p_private ) { demux_t *p_demux = (demux_t*)p_private; /* Avoid lock */ p_demux->p_sys->event_rtsp = 0xff; } static void TaskInterruptData( void *p_private ) { demux_t *p_demux = (demux_t*)p_private; p_demux->p_sys->i_no_data_ti++; /* Avoid lock */ p_demux->p_sys->event_data = 0xff; } /***************************************************************************** * *****************************************************************************/ static void TimeoutPrevention( void *p_data ) { demux_t *p_demux = (demux_t *) p_data; demux_sys_t *p_sys = p_demux->p_sys; char *bye = NULL; if( var_GetBool( p_demux, "rtsp-tcp" ) ) return; /* Protect Live555 from us calling their functions simultaneously with Demux() or Control() */ vlc_mutex_locker locker(&p_sys->timeout_mutex); /* If the timer fires while the demuxer owns the lock, and the demuxer * then torns the session down, the pointers will become NULL. By the time * this timer callback obtains the callback, either a new session was * created and the timer is rescheduled, or the pointers are still NULL * and the timer is descheduled. In the second case, bail out (then wait * for the timer to be rescheduled or destroyed). In the first case, this * might send an early refresh - that´s harmless but suboptimal (FIXME). */ if( p_sys->rtsp == NULL || p_sys->ms == NULL ) return; bool use_get_param = p_sys->b_get_param; /* Use GET_PARAMETERS if supported. wmserver dialect supports * it, but does not report this properly. */ if( var_GetBool( p_demux, "rtsp-wmserver" ) ) use_get_param = true; if( use_get_param ) p_sys->rtsp->sendGetParameterCommand( *p_sys->ms, default_live555_callback, bye ); else p_sys->rtsp->sendOptionsCommand( default_live555_callback, NULL ); if( !wait_Live555_response( p_demux ) ) { msg_Err( p_demux, "keep-alive failed: %s", p_sys->env->getResultMsg() ); /* Just continue, worst case is we get timed out later */ } } /***************************************************************************** * *****************************************************************************/ static int ParseASF( demux_t *p_demux ) { demux_sys_t *p_sys = p_demux->p_sys; const char *psz_marker = "a=pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,"; char *psz_asf = strcasestr( p_sys->p_sdp, psz_marker ); char *psz_end; block_t *p_header; /* Parse the asf header */ if( psz_asf == NULL ) return VLC_EGENERIC; psz_asf += strlen( psz_marker ); psz_asf = strdup( psz_asf ); /* Duplicate it */ psz_end = strchr( psz_asf, '\n' ); while( psz_end > psz_asf && ( *psz_end == '\n' || *psz_end == '\r' ) ) *psz_end-- = '\0'; if( psz_asf >= psz_end ) { free( psz_asf ); return VLC_EGENERIC; } /* Always smaller */ p_header = block_Alloc( psz_end - psz_asf ); p_header->i_buffer = vlc_b64_decode_binary_to_buffer( p_header->p_buffer, p_header->i_buffer, psz_asf ); //msg_Dbg( p_demux, "Size=%d Hdrb64=%s", p_header->i_buffer, psz_asf ); if( p_header->i_buffer <= 0 ) { free( psz_asf ); return VLC_EGENERIC; } /* Parse it to get packet size */ asf_HeaderParse( &p_sys->asfh, p_header->p_buffer, p_header->i_buffer ); /* Send it to demuxer */ vlc_demux_chained_Send( p_sys->p_out_asf, p_header ); free( psz_asf ); return VLC_SUCCESS; } static unsigned char* parseH264ConfigStr( char const* configStr, unsigned int& configSize ) { char *dup, *psz; size_t i_records = 1; configSize = 0; if( configStr == NULL || *configStr == '\0' ) return NULL; psz = dup = strdup( configStr ); /* Count the number of commas */ for( psz = dup; *psz != '\0'; ++psz ) { if( *psz == ',') { ++i_records; *psz = '\0'; } } size_t configMax = 4*i_records+strlen(configStr); unsigned char *cfg = new unsigned char[configMax]; psz = dup; for( size_t i = 0; i < i_records; ++i ) { cfg[configSize++] = 0x00; cfg[configSize++] = 0x00; cfg[configSize++] = 0x00; cfg[configSize++] = 0x01; configSize += vlc_b64_decode_binary_to_buffer( cfg+configSize, configMax-configSize, psz ); psz += strlen(psz)+1; } free( dup ); return cfg; } static uint8_t *parseVorbisConfigStr( char const* configStr, unsigned int& configSize ) { configSize = 0; if( configStr == NULL || *configStr == '\0' ) return NULL; #if LIVEMEDIA_LIBRARY_VERSION_INT >= 1332115200 // 2012.03.20 unsigned char *p_cfg = base64Decode( configStr, configSize ); #else char* configStr_dup = strdup( configStr ); unsigned char *p_cfg = base64Decode( configStr_dup, configSize ); free( configStr_dup ); #endif uint8_t *p_extra = NULL; /* skip header count, ident number and length (cf. RFC 5215) */ const unsigned int headerSkip = 9; if( configSize > headerSkip && ((uint8_t*)p_cfg)[3] == 1 ) { configSize -= headerSkip; p_extra = (uint8_t*)xmalloc( configSize ); memcpy( p_extra, p_cfg+headerSkip, configSize ); } delete[] p_cfg; return p_extra; } static char *passwordLessURL( vlc_url_t *p_url ) { vlc_url_t url; memcpy( &url, p_url, sizeof( vlc_url_t ) ); url.psz_username = NULL; url.psz_password = NULL; if( url.i_port == 0 ) url.i_port = 554; return vlc_uri_compose( &url ); }