X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Fstream_out%2Frtp.c;h=7de60c7564c5d0182fd97507da37b37db1c9d2d9;hb=1c3dea6380bb097a7cc43b040b25bf272321bedd;hp=52ca4fca30a8989dba37f31d8d0c38bb3f7297f6;hpb=764332810076c074bcb72e9dd0f8b78987543647;p=vlc diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c index 52ca4fca30..7de60c7564 100644 --- a/modules/stream_out/rtp.c +++ b/modules/stream_out/rtp.c @@ -39,7 +39,10 @@ #include #include #include -#include +#include +#ifdef HAVE_SRTP +# include +#endif #include "rtp.h" @@ -49,6 +52,9 @@ # include # include #endif +#ifdef HAVE_ARPA_INET_H +# include +#endif #ifdef HAVE_LINUX_DCCP_H # include #endif @@ -126,6 +132,11 @@ "This sends and receives RTCP packet multiplexed over the same port " \ "as RTP packets." ) +#define CACHING_TEXT N_("Caching value (ms)") +#define CACHING_LONGTEXT N_( \ + "Default caching value for outbound RTP streams. This " \ + "value should be set in milliseconds." ) + #define PROTO_TEXT N_("Transport protocol") #define PROTO_LONGTEXT N_( \ "This selects which transport protocol to use for RTP." ) @@ -166,47 +177,51 @@ vlc_module_begin () set_subcategory( SUBCAT_SOUT_STREAM ) add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT, - DEST_LONGTEXT, true ); + DEST_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT, - SDP_LONGTEXT, true ); + SDP_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT, - MUX_LONGTEXT, true ); + MUX_LONGTEXT, true ) add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT, - true ); + true ) add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT, - NAME_LONGTEXT, true ); + NAME_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT, - DESC_LONGTEXT, true ); + DESC_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT, - URL_LONGTEXT, true ); + URL_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT, - EMAIL_LONGTEXT, true ); + EMAIL_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT, - PHONE_LONGTEXT, true ); + PHONE_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT, - PROTO_LONGTEXT, false ); - change_string_list( ppsz_protos, ppsz_protocols, NULL ); + PROTO_LONGTEXT, false ) + change_string_list( ppsz_protos, ppsz_protocols, NULL ) add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT, - PORT_LONGTEXT, true ); + PORT_LONGTEXT, true ) add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT, - PORT_AUDIO_LONGTEXT, true ); + PORT_AUDIO_LONGTEXT, true ) add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT, - PORT_VIDEO_LONGTEXT, true ); + PORT_VIDEO_LONGTEXT, true ) add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT, - TTL_LONGTEXT, true ); + TTL_LONGTEXT, true ) add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL, - RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false ); + RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false ) + add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL, + CACHING_TEXT, CACHING_LONGTEXT, true ) +#ifdef HAVE_SRTP add_string( SOUT_CFG_PREFIX "key", "", NULL, - SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false ); + SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false ) add_string( SOUT_CFG_PREFIX "salt", "", NULL, - SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false ); + SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false ) +#endif - add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT, - RFC3016_LONGTEXT, false ); + add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, NULL, RFC3016_TEXT, + RFC3016_LONGTEXT, false ) set_callbacks( Open, Close ) vlc_module_end () @@ -217,7 +232,7 @@ vlc_module_end () static const char *const ppsz_sout_options[] = { "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux", "sap", "description", "url", "email", "phone", - "proto", "rtcp-mux", "key", "salt", + "proto", "rtcp-mux", "caching", "key", "salt", "mp4a-latm", NULL }; @@ -232,6 +247,7 @@ static int MuxSend( sout_stream_t *, sout_stream_id_t *, static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout ); static void* ThreadSend( vlc_object_t *p_this ); +static void *rtp_listen_thread( void * ); static void SDPHandleUrl( sout_stream_t *, const char * ); @@ -246,7 +262,6 @@ struct sout_stream_sys_t vlc_mutex_t lock_sdp; /* SDP to disk */ - bool b_export_sdp_file; char *psz_sdp_file; /* SDP via SAP */ @@ -260,6 +275,12 @@ struct sout_stream_sys_t /* RTSP */ rtsp_stream_t *rtsp; + /* RTSP NPT and timestamp computations */ + mtime_t i_npt_zero; /* when NPT=0 packet is sent */ + int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */ + int64_t i_pts_offset; /* matches actual PTS to prediction */ + vlc_mutex_t lock_ts; + /* */ char *psz_destination; uint32_t payload_bitmap; @@ -298,8 +319,13 @@ struct sout_stream_id_t /* rtp field */ uint16_t i_sequence; uint8_t i_payload_type; + bool b_ts_init; + uint32_t i_ts_offset; uint8_t ssrc[4]; + /* for rtsp */ + uint16_t i_seq_sent_next; + /* for sdp */ const char *psz_enc; char *psz_fmtp; @@ -311,7 +337,9 @@ struct sout_stream_id_t /* Packetizer specific fields */ int i_mtu; +#ifdef HAVE_SRTP srtp_session_t *srtp; +#endif pf_rtp_packetizer_t pf_packetize; /* Packets sinks */ @@ -319,7 +347,10 @@ struct sout_stream_id_t int sinkc; rtp_sink_t *sinkv; rtsp_stream_id_t *rtsp_id; - int *listen_fd; + struct { + int *fd; + vlc_thread_t thread; + } listen; block_fifo_t *p_fifo; int64_t i_caching; @@ -349,15 +380,14 @@ static int Open( vlc_object_t *p_this ) p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" ); p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" ); p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" ); - p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" ); + p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" ); - p_sys->psz_sdp_file = NULL; - - if( p_sys->i_port_audio == p_sys->i_port_video ) + if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio ) { - msg_Err( p_stream, "audio and video port cannot be the same" ); - p_sys->i_port_audio = 0; - p_sys->i_port_video = 0; + msg_Err( p_stream, "audio and video RTP port must be distinct" ); + free( p_sys->psz_destination ); + free( p_sys ); + return VLC_EGENERIC; } for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next ) @@ -437,6 +467,14 @@ static int Open( vlc_object_t *p_this ) p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" ); + /* NPT=0 time will be determined when we packetize the first packet + * (of any ES). But we want to be able to report rtptime in RTSP + * without waiting. So until then, we use an arbitrary reference + * PTS for timestamp computations, and then actual PTS will catch + * up using offsets. */ + p_sys->i_npt_zero = VLC_TS_INVALID; + p_sys->i_pts_zero = mdate(); /* arbitrary value, could probably be + * random */ p_sys->payload_bitmap = 0; p_sys->i_es = 0; p_sys->es = NULL; @@ -444,8 +482,8 @@ static int Open( vlc_object_t *p_this ) p_sys->psz_sdp = NULL; p_sys->b_export_sap = false; - p_sys->b_export_sdp_file = false; p_sys->p_session = NULL; + p_sys->psz_sdp_file = NULL; p_sys->p_httpd_host = NULL; p_sys->p_httpd_file = NULL; @@ -453,6 +491,7 @@ static int Open( vlc_object_t *p_this ) p_stream->p_sys = p_sys; vlc_mutex_init( &p_sys->lock_sdp ); + vlc_mutex_init( &p_sys->lock_ts ); vlc_mutex_init( &p_sys->lock_es ); psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" ); @@ -469,6 +508,7 @@ static int Open( vlc_object_t *p_this ) free( psz ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); + free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } @@ -483,6 +523,7 @@ static int Open( vlc_object_t *p_this ) sout_AccessOutDelete( p_sys->p_grab ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); + free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } @@ -494,6 +535,7 @@ static int Open( vlc_object_t *p_this ) sout_AccessOutDelete( p_sys->p_grab ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); + free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } @@ -561,10 +603,11 @@ static void Close( vlc_object_t * p_this ) if( p_sys->p_mux ) { assert( p_sys->i_es == 1 ); - Del( p_stream, p_sys->es[0] ); sout_MuxDelete( p_sys->p_mux ); + Del( p_stream, p_sys->es[0] ); sout_AccessOutDelete( p_sys->p_grab ); + if( p_sys->packet ) { block_Release( p_sys->packet ); @@ -580,6 +623,7 @@ static void Close( vlc_object_t * p_this ) RtspUnsetup( p_sys->rtsp ); vlc_mutex_destroy( &p_sys->lock_sdp ); + vlc_mutex_destroy( &p_sys->lock_ts ); vlc_mutex_destroy( &p_sys->lock_es ); if( p_sys->p_httpd_file ) @@ -590,7 +634,7 @@ static void Close( vlc_object_t * p_this ) free( p_sys->psz_sdp ); - if( p_sys->b_export_sdp_file ) + if( p_sys->psz_sdp_file != NULL ) { #ifdef HAVE_UNISTD_H unlink( p_sys->psz_sdp_file ); @@ -639,7 +683,7 @@ static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url ) if( p_sys->p_mux != NULL ) { sout_stream_id_t *id = p_sys->es[0]; - id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ), + id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ), p_sys->psz_destination, p_sys->i_ttl, id->i_port, id->i_port + 1 ); } @@ -652,16 +696,19 @@ static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url ) } else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) ) { - if( p_sys->b_export_sdp_file ) + if( p_sys->psz_sdp_file != NULL ) { msg_Err( p_stream, "you can use sdp=file:// only once" ); goto out; } - p_sys->b_export_sdp_file = true; psz_url = &psz_url[5]; if( psz_url[0] == '/' && psz_url[1] == '/' ) psz_url += 2; p_sys->psz_sdp_file = strdup( psz_url ); + if( p_sys->psz_sdp_file == NULL ) + goto out; + decode_URI( p_sys->psz_sdp_file ); /* FIXME? */ + FileSetup( p_stream ); } else { @@ -707,8 +754,8 @@ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url ) /* Oh boy, this is really ugly! (+ race condition on lock_es) */ dstlen = sizeof( dst ); - if( p_sys->es[0]->listen_fd != NULL ) - getsockname( p_sys->es[0]->listen_fd[0], + if( p_sys->es[0]->listen.fd != NULL ) + getsockname( p_sys->es[0]->listen.fd[0], (struct sockaddr *)&dst, &dstlen ); else getpeername( p_sys->es[0]->sinkv[0].rtp_fd, @@ -783,6 +830,9 @@ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url ) id->psz_enc, id->i_clock_rate, id->i_channels, id->psz_fmtp); + if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */ + sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 ); + if( rtsp_url != NULL ) { assert( strlen( rtsp_url ) > 0 ); @@ -793,7 +843,7 @@ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url ) } else { - if( id->listen_fd != NULL ) + if( id->listen.fd != NULL ) sdp_AddAttribute( &psz_sdp, "setup", "passive" ); if( p_sys->proto == IPPROTO_DCCP ) sdp_AddAttribute( &psz_sdp, "dccp-service-code", @@ -838,6 +888,13 @@ rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes) id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes); } +uint32_t rtp_compute_ts( const sout_stream_id_t *id, int64_t i_pts ) +{ + /* NOTE: this plays nice with offsets because the calculations are + * linear. */ + return i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ; +} + /** Add an ES as a new RTP stream */ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) { @@ -845,7 +902,6 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) * mux (TS/PS), then p_fmt is NULL. */ sout_stream_sys_t *p_sys = p_stream->p_sys; sout_stream_id_t *id; - int i_port, cscov = -1; char *psz_sdp; if (0xffffffff == p_sys->payload_bitmap) @@ -854,53 +910,53 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) return NULL; } - id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) ); - if( id == NULL ) - return NULL; - vlc_object_attach( id, p_stream ); - /* Choose the port */ - i_port = 0; + uint16_t i_port = 0; if( p_fmt == NULL ) ; else if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 ) - { i_port = p_sys->i_port_audio; - p_sys->i_port_audio = 0; - } else if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 ) - { i_port = p_sys->i_port_video; - p_sys->i_port_video = 0; - } - while( i_port == 0 ) + /* We do not need the ES lock (p_sys->lock_es) here, because this is the + * only one thread that can *modify* the ES table. The ES lock protects + * the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */ + for (int i = 0; i_port && (i < p_sys->i_es); i++) + if (i_port == p_sys->es[i]->i_port) + i_port = 0; /* Port already in use! */ + for (uint16_t p = p_sys->i_port; i_port == 0; p += 2) { - if( p_sys->i_port != p_sys->i_port_audio - && p_sys->i_port != p_sys->i_port_video ) + if (p == 0) { - i_port = p_sys->i_port; - p_sys->i_port += 2; - break; + msg_Err (p_stream, "too many RTP elementary streams"); + return NULL; } - p_sys->i_port += 2; + i_port = p; + for (int i = 0; i_port && (i < p_sys->i_es); i++) + if (p == p_sys->es[i]->i_port) + i_port = 0; } + id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) ); + if( id == NULL ) + return NULL; + vlc_object_attach( id, p_stream ); + id->p_stream = p_stream; - id->i_sequence = rand()&0xffff; /* Look for free dymanic payload type */ id->i_payload_type = 96; while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96))) id->i_payload_type++; assert (id->i_payload_type < 128); - id->ssrc[0] = rand()&0xff; - id->ssrc[1] = rand()&0xff; - id->ssrc[2] = rand()&0xff; - id->ssrc[3] = rand()&0xff; + vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence)); + vlc_rand_bytes (id->ssrc, sizeof (id->ssrc)); + + id->i_seq_sent_next = id->i_sequence; id->psz_enc = NULL; id->psz_fmtp = NULL; @@ -928,9 +984,11 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) id->i_mtu = 576 - 20 - 8; /* pessimistic */ msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu ); - id->srtp = NULL; id->pf_packetize = NULL; +#ifdef HAVE_SRTP + id->srtp = NULL; + char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key"); if (key) { @@ -953,13 +1011,14 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) } id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */ } +#endif vlc_mutex_init( &id->lock_sink ); id->sinkc = 0; id->sinkv = NULL; id->rtsp_id = NULL; id->p_fifo = NULL; - id->listen_fd = NULL; + id->listen.fd = NULL; id->i_caching = (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching"); @@ -980,14 +1039,21 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) var_SetString (p_stream, "dccp-service", code); } /* fall through */ case IPPROTO_TCP: - id->listen_fd = net_Listen( VLC_OBJECT(p_stream), + id->listen.fd = net_Listen( VLC_OBJECT(p_stream), p_sys->psz_destination, i_port, p_sys->proto ); - if( id->listen_fd == NULL ) + if( id->listen.fd == NULL ) { msg_Err( p_stream, "passive COMEDIA RTP socket failed" ); goto error; } + if( vlc_clone( &id->listen.thread, rtp_listen_thread, id, + VLC_THREAD_PRIORITY_LOW ) ) + { + net_ListenClose( id->listen.fd ); + id->listen.fd = NULL; + goto error; + } break; default: @@ -1000,7 +1066,11 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) msg_Err( p_stream, "cannot create RTP socket" ); goto error; } - rtp_add_sink( id, fd, p_sys->rtcp_mux ); + /* Ignore any unexpected incoming packet (including RTCP-RR + * packets in case of rtcp-mux) */ + setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 }, + sizeof (int)); + rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL ); } } @@ -1025,21 +1095,22 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) else switch( p_fmt->i_codec ) { - case VLC_FOURCC( 'u', 'l', 'a', 'w' ): + case VLC_CODEC_MULAW: if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 ) id->i_payload_type = 0; id->psz_enc = "PCMU"; id->pf_packetize = rtp_packetize_split; rtp_set_ptime (id, 20, 1); break; - case VLC_FOURCC( 'a', 'l', 'a', 'w' ): + case VLC_CODEC_ALAW: if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 ) id->i_payload_type = 8; id->psz_enc = "PCMA"; id->pf_packetize = rtp_packetize_split; rtp_set_ptime (id, 20, 1); break; - case VLC_FOURCC( 's', '1', '6', 'b' ): + case VLC_CODEC_S16B: + case VLC_CODEC_S16L: if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 ) { id->i_payload_type = 11; @@ -1050,28 +1121,29 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) id->i_payload_type = 10; } id->psz_enc = "L16"; - id->pf_packetize = rtp_packetize_split; + if( p_fmt->i_codec == VLC_CODEC_S16B ) + id->pf_packetize = rtp_packetize_split; + else + id->pf_packetize = rtp_packetize_swab; rtp_set_ptime (id, 20, 2); break; - case VLC_FOURCC( 'u', '8', ' ', ' ' ): + case VLC_CODEC_U8: id->psz_enc = "L8"; id->pf_packetize = rtp_packetize_split; rtp_set_ptime (id, 20, 1); break; - case VLC_FOURCC( 'm', 'p', 'g', 'a' ): - case VLC_FOURCC( 'm', 'p', '3', ' ' ): + case VLC_CODEC_MPGA: id->i_payload_type = 14; id->psz_enc = "MPA"; id->i_clock_rate = 90000; /* not 44100 */ id->pf_packetize = rtp_packetize_mpa; break; - case VLC_FOURCC( 'm', 'p', 'g', 'v' ): + case VLC_CODEC_MPGV: id->i_payload_type = 32; id->psz_enc = "MPV"; id->pf_packetize = rtp_packetize_mpv; break; - case VLC_FOURCC( 'G', '7', '2', '6' ): - case VLC_FOURCC( 'g', '7', '2', '6' ): + case VLC_CODEC_ADPCM_G726: switch( p_fmt->i_bitrate / 1000 ) { case 16: @@ -1090,17 +1162,21 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) id->psz_enc = "G726-40"; id->pf_packetize = rtp_packetize_g726_40; break; + default: + msg_Err( p_stream, "cannot add this stream (unsupported " + "G.726 bit rate: %u)", p_fmt->i_bitrate ); + goto error; } break; - case VLC_FOURCC( 'a', '5', '2', ' ' ): + case VLC_CODEC_A52: id->psz_enc = "ac3"; id->pf_packetize = rtp_packetize_ac3; break; - case VLC_FOURCC( 'H', '2', '6', '3' ): + case VLC_CODEC_H263: id->psz_enc = "H263-1998"; id->pf_packetize = rtp_packetize_h263; break; - case VLC_FOURCC( 'h', '2', '6', '4' ): + case VLC_CODEC_H264: id->psz_enc = "H264"; id->pf_packetize = rtp_packetize_h264; id->psz_fmtp = NULL; @@ -1159,14 +1235,13 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) id->psz_fmtp = strdup( "packetization-mode=1" ); break; - case VLC_FOURCC( 'm', 'p', '4', 'v' ): + case VLC_CODEC_MP4V: { - char hexa[2*p_fmt->i_extra +1]; - id->psz_enc = "MP4V-ES"; id->pf_packetize = rtp_packetize_split; if( p_fmt->i_extra > 0 ) { + char hexa[2*p_fmt->i_extra +1]; sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra ); if( asprintf( &id->psz_fmtp, "profile-level-id=3; config=%s;", hexa ) == -1 ) @@ -1174,7 +1249,7 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) } break; } - case VLC_FOURCC( 'm', 'p', '4', 'a' ): + case VLC_CODEC_MP4A: { if(!p_sys->b_latm) { @@ -1219,21 +1294,21 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) } break; } - case VLC_FOURCC( 's', 'a', 'm', 'r' ): + case VLC_CODEC_AMR_NB: id->psz_enc = "AMR"; id->psz_fmtp = strdup( "octet-align=1" ); id->pf_packetize = rtp_packetize_amr; break; - case VLC_FOURCC( 's', 'a', 'w', 'b' ): + case VLC_CODEC_AMR_WB: id->psz_enc = "AMR-WB"; id->psz_fmtp = strdup( "octet-align=1" ); id->pf_packetize = rtp_packetize_amr; break; - case VLC_FOURCC( 's', 'p', 'x', ' ' ): + case VLC_CODEC_SPEEX: id->psz_enc = "SPEEX"; id->pf_packetize = rtp_packetize_spx; break; - case VLC_FOURCC( 't', '1', '4', '0' ): + case VLC_CODEC_ITU_T140: id->psz_enc = "t140" ; id->i_clock_rate = 1000; id->pf_packetize = rtp_packetize_t140; @@ -1241,7 +1316,7 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) default: msg_Err( p_stream, "cannot add this stream (unsupported " - "codec:%4.4s)", (char*)&p_fmt->i_codec ); + "codec: %4.4s)", (char*)&p_fmt->i_codec ); goto error; } if (id->i_payload_type >= 96) @@ -1249,21 +1324,28 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96); #if 0 /* No payload formats sets this at the moment */ + int cscov = -1; if( cscov != -1 ) cscov += 8 /* UDP */ + 12 /* RTP */; if( id->sinkc > 0 ) net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 ); #endif + vlc_mutex_lock( &p_sys->lock_ts ); + id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID ); + vlc_mutex_unlock( &p_sys->lock_ts ); + if( id->b_ts_init ) + id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset ); + if( p_sys->rtsp != NULL ) - id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es, + id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ), p_sys->psz_destination, p_sys->i_ttl, id->i_port, id->i_port + 1 ); id->p_fifo = block_FifoNew(); if( vlc_thread_create( id, "RTP send thread", ThreadSend, - VLC_THREAD_PRIORITY_HIGHEST, false ) ) + VLC_THREAD_PRIORITY_HIGHEST ) ) goto error; /* Update p_sys context */ @@ -1282,7 +1364,7 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) /* Update SDP (sap/file) */ if( p_sys->b_export_sap ) SapSetup( p_stream ); - if( p_sys->b_export_sdp_file ) FileSetup( p_stream ); + if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream ); return id; @@ -1306,11 +1388,6 @@ static int Del( sout_stream_t *p_stream, sout_stream_id_t *id ) TAB_REMOVE( p_sys->i_es, p_sys->es, id ); vlc_mutex_unlock( &p_sys->lock_es ); - /* Release port */ - if( id->i_port == var_GetInteger( p_stream, "port-audio" ) ) - p_sys->i_port_audio = id->i_port; - if( id->i_port == var_GetInteger( p_stream, "port-video" ) ) - p_sys->i_port_video = id->i_port; /* Release dynamic payload type */ if (id->i_payload_type >= 96) p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96)); @@ -1321,16 +1398,22 @@ static int Del( sout_stream_t *p_stream, sout_stream_id_t *id ) RtspDelId( p_sys->rtsp, id->rtsp_id ); if( id->sinkc > 0 ) rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */ - if( id->listen_fd != NULL ) - net_ListenClose( id->listen_fd ); + if( id->listen.fd != NULL ) + { + vlc_cancel( id->listen.thread ); + vlc_join( id->listen.thread, NULL ); + net_ListenClose( id->listen.fd ); + } +#ifdef HAVE_SRTP if( id->srtp != NULL ) srtp_destroy( id->srtp ); +#endif vlc_mutex_destroy( &id->lock_sink ); /* Update SDP (sap/file) */ if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream ); - if( p_sys->b_export_sdp_file ) FileSetup( p_stream ); + if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream ); vlc_object_detach( id ); vlc_object_release( id ); @@ -1393,6 +1476,9 @@ static int FileSetup( sout_stream_t *p_stream ) sout_stream_sys_t *p_sys = p_stream->p_sys; FILE *f; + if( p_sys->psz_sdp == NULL ) + return VLC_EGENERIC; /* too early */ + if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL ) { msg_Err( p_stream, "cannot open file '%s' (%m)", @@ -1463,6 +1549,16 @@ static int HttpCallback( httpd_file_sys_t *p_args, ****************************************************************************/ static void* ThreadSend( vlc_object_t *p_this ) { +#ifdef WIN32 +# define ECONNREFUSED WSAECONNREFUSED +# define ENOPROTOOPT WSAENOPROTOOPT +# define EHOSTUNREACH WSAEHOSTUNREACH +# define ENETUNREACH WSAENETUNREACH +# define ENETDOWN WSAENETDOWN +# define ENOBUFS WSAENOBUFS +# define EAGAIN WSAEWOULDBLOCK +# define EWOULDBLOCK WSAEWOULDBLOCK +#endif sout_stream_id_t *id = (sout_stream_id_t *)p_this; unsigned i_caching = id->i_caching; @@ -1471,6 +1567,7 @@ static void* ThreadSend( vlc_object_t *p_this ) block_t *out = block_FifoGet( id->p_fifo ); block_cleanup_push (out); +#ifdef HAVE_SRTP if( id->srtp ) { /* FIXME: this is awfully inefficient */ size_t len = out->i_buffer; @@ -1490,8 +1587,8 @@ static void* ThreadSend( vlc_object_t *p_this ) else out->i_buffer = len; } - if (out) +#endif mwait (out->i_dts + i_caching); vlc_cleanup_pop (); if (out == NULL) @@ -1506,17 +1603,38 @@ static void* ThreadSend( vlc_object_t *p_this ) for( int i = 0; i < id->sinkc; i++ ) { +#ifdef HAVE_SRTP if( !id->srtp ) /* FIXME: SRTCP support */ +#endif SendRTCP( id->sinkv[i].rtcp, out ); if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 ) continue; - /* Retry sending to root out soft-errors */ - if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 ) - continue; + switch( net_errno ) + { + /* Soft errors (e.g. ICMP): */ + case ECONNREFUSED: /* Port unreachable */ + case ENOPROTOOPT: +#ifdef EPROTO + case EPROTO: /* Protocol unreachable */ +#endif + case EHOSTUNREACH: /* Host unreachable */ + case ENETUNREACH: /* Network unreachable */ + case ENETDOWN: /* Entire network down */ + send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ); + /* Transient congestion: */ + case ENOMEM: /* out of socket buffers */ + case ENOBUFS: + case EAGAIN: +#if (EAGAIN != EWOULDBLOCK) + case EWOULDBLOCK: +#endif + continue; + } deadv[deadc++] = id->sinkv[i].rtp_fd; } + id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1; vlc_mutex_unlock( &id->lock_sink ); block_Release( out ); @@ -1525,22 +1643,34 @@ static void* ThreadSend( vlc_object_t *p_this ) msg_Dbg( id, "removing socket %d", deadv[i] ); rtp_del_sink( id, deadv[i] ); } - - /* Hopefully we won't overflow the SO_MAXCONN accept queue */ - while( id->listen_fd != NULL ) - { - int fd = net_Accept( id, id->listen_fd, 0 ); - if( fd == -1 ) - break; - msg_Dbg( id, "adding socket %d", fd ); - rtp_add_sink( id, fd, true ); - } vlc_restorecancel (canc); } return NULL; } -int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux ) + +/* This thread dequeues incoming connections (DCCP streaming) */ +static void *rtp_listen_thread( void *data ) +{ + sout_stream_id_t *id = data; + + assert( id->listen.fd != NULL ); + + for( ;; ) + { + int fd = net_Accept( id, id->listen.fd ); + if( fd == -1 ) + continue; + int canc = vlc_savecancel( ); + rtp_add_sink( id, fd, true, NULL ); + vlc_restorecancel( canc ); + } + + assert( 0 ); +} + + +int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq ) { rtp_sink_t sink = { fd, NULL }; sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP, @@ -1550,6 +1680,8 @@ int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux ) vlc_mutex_lock( &id->lock_sink ); INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink ); + if( seq != NULL ) + *seq = id->i_seq_sent_next; vlc_mutex_unlock( &id->lock_sink ); return VLC_SUCCESS; } @@ -1575,36 +1707,62 @@ void rtp_del_sink( sout_stream_id_t *id, int fd ) net_Close( sink.rtp_fd ); } -uint16_t rtp_get_seq( const sout_stream_id_t *id ) +uint16_t rtp_get_seq( sout_stream_id_t *id ) { - /* This will return values for the next packet. - * Accounting for caching would not be totally trivial. */ - return id->i_sequence; + /* This will return values for the next packet. */ + uint16_t seq; + + vlc_mutex_lock( &id->lock_sink ); + seq = id->i_seq_sent_next; + vlc_mutex_unlock( &id->lock_sink ); + + return seq; } -/* FIXME: this is pretty bad - if we remove and then insert an ES - * the number will get unsynched from inside RTSP */ -unsigned rtp_get_num( const sout_stream_id_t *id ) +/* Return a timestamp corresponding to packets being sent now, and that + * can be passed to rtp_compute_ts() to get rtptime values for each ES. */ +int64_t rtp_get_ts( const sout_stream_t *p_stream ) { - sout_stream_sys_t *p_sys = id->p_stream->p_sys; - int i; + sout_stream_sys_t *p_sys = p_stream->p_sys; + mtime_t i_npt_zero; + vlc_mutex_lock( &p_sys->lock_ts ); + i_npt_zero = p_sys->i_npt_zero; + vlc_mutex_unlock( &p_sys->lock_ts ); - vlc_mutex_lock( &p_sys->lock_es ); - for( i = 0; i < p_sys->i_es; i++ ) - { - if( id == p_sys->es[i] ) - break; - } - vlc_mutex_unlock( &p_sys->lock_es ); + if( i_npt_zero == VLC_TS_INVALID ) + return p_sys->i_pts_zero; - return i; -} + mtime_t now = mdate(); + if( now < i_npt_zero ) + return p_sys->i_pts_zero; + return p_sys->i_pts_zero + (now - i_npt_zero); +} void rtp_packetize_common( sout_stream_id_t *id, block_t *out, int b_marker, int64_t i_pts ) { - uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / INT64_C(1000000); + if( !id->b_ts_init ) + { + sout_stream_sys_t *p_sys = id->p_stream->p_sys; + vlc_mutex_lock( &p_sys->lock_ts ); + if( p_sys->i_npt_zero == VLC_TS_INVALID ) + { + /* This is the first packet of any ES. We initialize the + * NPT=0 time reference, and the offset to match the + * arbitrary PTS reference. */ + p_sys->i_npt_zero = i_pts + id->i_caching; + p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts; + } + vlc_mutex_unlock( &p_sys->lock_ts ); + + /* And in any case this is the first packet of this ES, so we + * initialize the offset for this ES. */ + id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset ); + id->b_ts_init = true; + } + + uint32_t i_timestamp = rtp_compute_ts( id, i_pts ) + id->i_ts_offset; out->p_buffer[0] = 0x80; out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type; @@ -1623,14 +1781,7 @@ void rtp_packetize_common( sout_stream_id_t *id, block_t *out, void rtp_packetize_send( sout_stream_id_t *id, block_t *out ) { - static block_t *dummy = NULL; - if (!dummy) - { - dummy = out; - return; - } block_FifoPut( id->p_fifo, out ); - block_FifoPut( id->p_fifo, dummy ); } /**