static GF_Err RP_ServiceCommand(GF_InputService *plug, GF_NetworkCommand *com)
{
RTPStream *ch;
RTPClient *priv = (RTPClient *)plug->priv;
if (com->command_type==GF_NET_SERVICE_HAS_AUDIO) {
u32 i;
for (i=0; i<gf_list_count(priv->channels); i++) {
ch = gf_list_get(priv->channels, i);
if (ch->depacketizer->sl_map.StreamType==GF_STREAM_AUDIO)
return GF_OK;
}
return GF_NOT_SUPPORTED;
}
if (com->command_type==GF_NET_SERVICE_MIGRATION_INFO) {
RP_SaveSessionState(priv);
priv->session_migration=1;
if (priv->session_state_data) {
com->migrate.data = priv->session_state_data;
com->migrate.data_len = strlen(priv->session_state_data);
return GF_OK;
}
return GF_NOT_SUPPORTED;
}
/*ignore commands other than channels one*/
if (!com->base.on_channel) {
if (com->command_type==GF_NET_IS_CACHABLE) return GF_OK;
return GF_NOT_SUPPORTED;
}
ch = RP_FindChannel(priv, com->base.on_channel, 0, NULL, 0);
if (!ch) return GF_STREAM_NOT_FOUND;
switch (com->command_type) {
case GF_NET_CHAN_SET_PULL:
if (ch->rtp_ch || ch->rtsp || !ch->control) return GF_NOT_SUPPORTED;
/*embedded channels work in pull mode*/
if (strstr(ch->control, "data:application/")) return GF_OK;
return GF_NOT_SUPPORTED;
case GF_NET_CHAN_INTERACTIVE:
/*looks like pure RTP / multicast etc, not interactive*/
if (!ch->control) return GF_NOT_SUPPORTED;
/*emulated broadcast mode*/
else if (ch->flags & RTP_FORCE_BROADCAST) return GF_NOT_SUPPORTED;
/*regular rtsp mode*/
else if (ch->flags & RTP_HAS_RANGE) return GF_OK;
/*embedded data*/
else if (strstr(ch->control, "application")) return GF_OK;
return GF_NOT_SUPPORTED;
case GF_NET_CHAN_BUFFER:
if (!(ch->rtp_ch || ch->rtsp || !ch->control)) {
com->buffer.max = com->buffer.min = 0;
} else {
const char *opt;
/*amount of buffering in ms*/
opt = gf_modules_get_option((GF_BaseInterface *)plug, "Network", "BufferLength");
com->buffer.max = opt ? atoi(opt) : 1000;
/*rebuffer low limit in ms - if the amount of buffering is less than this, rebuffering will never occur*/
opt = gf_modules_get_option((GF_BaseInterface *)plug, "Network", "RebufferLength");
if (opt) com->buffer.min = atoi(opt);
else com->buffer.min = 500;
if (com->buffer.min >= com->buffer.max ) com->buffer.min = 0;
}
return GF_OK;
case GF_NET_CHAN_DURATION:
com->duration.duration = (ch->flags & RTP_HAS_RANGE) ? (ch->range_end - ch->range_start) : 0;
return GF_OK;
/*RTP channel config is done upon connection, once the complete SL mapping is known
however we must store some info not carried in SDP*/
case GF_NET_CHAN_CONFIG:
if (com->cfg.frame_duration) ch->depacketizer->sl_hdr.au_duration = com->cfg.frame_duration;
ch->ts_res = com->cfg.sl_config.timestampResolution;
return GF_OK;
case GF_NET_CHAN_PLAY:
GF_LOG(GF_LOG_DEBUG, GF_LOG_RTP, ("[RTP] Processing play on channel @%08x - %s\n", ch, ch->rtsp ? "RTSP control" : "No control (RTP)" ));
/*is this RTSP or direct RTP?*/
ch->flags &= ~RTP_EOS;
if (ch->rtsp) {
if (ch->status==RTP_SessionResume) {
const char *opt = gf_modules_get_option((GF_BaseInterface *) plug, "Streaming", "SessionMigrationPause");
if (opt && !strcmp(opt, "yes")) {
ch->status = RTP_Connected;
com->play.start_range = ch->current_start;
} else {
ch->status = RTP_Running;
return GF_OK;
}
}
RP_UserCommand(ch->rtsp, ch, com);
} else {
ch->status = RTP_Running;
if (ch->rtp_ch) {
/*technically we shouldn't attempt to synchronize streams based on RTP, we should use RTCP/ However it
may happen that the RTCP traffic is absent ...*/
ch->check_rtp_time = RTP_SET_TIME_RTP;
ch->rtcp_init = 0;
gf_mx_p(priv->mx);
RP_InitStream(ch, (ch->flags & RTP_CONNECTED) ? 1 : 0);
gf_mx_v(priv->mx);
gf_rtp_set_info_rtp(ch->rtp_ch, 0, 0, 0);
} else {
/*direct channel, store current start*/
ch->current_start = com->play.start_range;
ch->flags |= GF_RTP_NEW_AU;
gf_rtp_depacketizer_reset(ch->depacketizer, 0);
}
}
return GF_OK;
case GF_NET_CHAN_STOP:
/*is this RTSP or direct RTP?*/
if (ch->rtsp) {
if (! ch->owner->session_migration) {
RP_UserCommand(ch->rtsp, ch, com);
}
} else {
ch->status = RTP_Connected;
ch->owner->last_ntp = 0;
}
ch->rtcp_init = 0;
return GF_OK;
case GF_NET_CHAN_SET_SPEED:
case GF_NET_CHAN_PAUSE:
case GF_NET_CHAN_RESUME:
assert(ch->rtsp);
RP_UserCommand(ch->rtsp, ch, com);
return GF_OK;
case GF_NET_CHAN_GET_DSI:
if (ch->depacketizer && ch->depacketizer->sl_map.configSize) {
com->get_dsi.dsi_len = ch->depacketizer->sl_map.configSize;
com->get_dsi.dsi = (char*)gf_malloc(sizeof(char)*com->get_dsi.dsi_len);
memcpy(com->get_dsi.dsi, ch->depacketizer->sl_map.config, sizeof(char)*com->get_dsi.dsi_len);
} else {
com->get_dsi.dsi = NULL;
com->get_dsi.dsi_len = 0;
}
return GF_OK;
case GF_NET_GET_STATS:
memset(&com->net_stats, 0, sizeof(GF_NetComStats));
if (ch->rtp_ch) {
u32 time;
Float bps;
com->net_stats.pck_loss_percentage = gf_rtp_get_loss(ch->rtp_ch);
if (ch->flags & RTP_INTERLEAVED) {
com->net_stats.multiplex_port = gf_rtsp_get_session_port(ch->rtsp->session);
com->net_stats.port = gf_rtp_get_low_interleave_id(ch->rtp_ch);
com->net_stats.ctrl_port = gf_rtp_get_hight_interleave_id(ch->rtp_ch);
} else {
com->net_stats.multiplex_port = 0;
gf_rtp_get_ports(ch->rtp_ch, &com->net_stats.port, &com->net_stats.ctrl_port);
}
if (ch->stat_stop_time) {
time = ch->stat_stop_time - ch->stat_start_time;
} else {
time = gf_sys_clock() - ch->stat_start_time;
}
bps = 8.0f * ch->rtp_bytes; bps *= 1000; bps /= time; com->net_stats.bw_down = (u32) bps;
bps = 8.0f * ch->rtcp_bytes; bps *= 1000; bps /= time; com->net_stats.ctrl_bw_down = (u32) bps;
bps = 8.0f * gf_rtp_get_tcp_bytes_sent(ch->rtp_ch); bps *= 1000; bps /= time; com->net_stats.ctrl_bw_up = (u32) bps;
}
return GF_OK;
}
return GF_NOT_SUPPORTED;
}
RTPStream *RP_NewStream(RTPClient *rtp, GF_SDPMedia *media, GF_SDPInfo *sdp, RTPStream *input_stream)
{
GF_RTSPRange *range;
RTPStream *tmp;
GF_RTPMap *map;
u32 i, ESID, ODID, ssrc, rtp_seq, rtp_time;
Bool force_bcast = 0;
Double Start, End;
Float CurrentTime;
u16 rvc_predef = 0;
char *rvc_config_att = NULL;
u32 s_port_first, s_port_last;
GF_X_Attribute *att;
Bool is_migration = 0;
char *ctrl;
GF_SDPConnection *conn;
GF_RTSPTransport trans;
u32 mid, prev_stream, base_stream;
//extract all relevant info from the GF_SDPMedia
Start = 0.0;
End = -1.0;
CurrentTime = 0.0f;
ODID = 0;
ESID = 0;
ctrl = NULL;
range = NULL;
s_port_first = s_port_last = 0;
ssrc = rtp_seq = rtp_time = 0;
mid = prev_stream = base_stream = 0;
i=0;
while ((att = (GF_X_Attribute*)gf_list_enum(media->Attributes, &i))) {
if (!stricmp(att->Name, "control")) ctrl = att->Value;
else if (!stricmp(att->Name, "gpac-broadcast")) force_bcast = 1;
else if (!stricmp(att->Name, "mpeg4-esid") && att->Value) ESID = atoi(att->Value);
else if (!stricmp(att->Name, "mpeg4-odid") && att->Value) ODID = atoi(att->Value);
else if (!stricmp(att->Name, "range") && !range) range = gf_rtsp_range_parse(att->Value);
else if (!stricmp(att->Name, "x-stream-state") ) {
sscanf(att->Value, "server-port=%u-%u;ssrc=%X;npt=%g;seq=%u;rtptime=%u",
&s_port_first, &s_port_last, &ssrc, &CurrentTime, &rtp_seq, &rtp_time);
is_migration = 1;
}
else if (!stricmp(att->Name, "x-server-port") ) {
sscanf(att->Value, "%u-%u", &s_port_first, &s_port_last);
} else if (!stricmp(att->Name, "rvc-config-predef")) {
rvc_predef = atoi(att->Value);
} else if (!stricmp(att->Name, "rvc-config")) {
rvc_config_att = att->Value;
} else if (!stricmp(att->Name, "mid")) {
sscanf(att->Value, "L%d", &mid);
} else if (!stricmp(att->Name, "depend")) {
char buf[3000];
memset(buf, 0, 3000);
sscanf(att->Value, "%*d lay L%d %*s %s", &base_stream, buf);
if (!strlen(buf))
sscanf(att->Value, "%*d lay %s", buf);
sscanf(buf, "L%d", &prev_stream);
}
}
if (range) {
Start = range->start;
End = range->end;
gf_rtsp_range_del(range);
}
/*check connection*/
conn = sdp->c_connection;
if (conn && (!conn->host || !strcmp(conn->host, "0.0.0.0"))) conn = NULL;
if (!conn) conn = (GF_SDPConnection*)gf_list_get(media->Connections, 0);
if (conn && (!conn->host || !strcmp(conn->host, "0.0.0.0"))) conn = NULL;
if (!conn) {
/*RTSP RFC recommends an empty "c= " line but some server don't send it. Use session info (o=)*/
if (!sdp->o_net_type || !sdp->o_add_type || strcmp(sdp->o_net_type, "IN")) return NULL;
if (strcmp(sdp->o_add_type, "IP4") && strcmp(sdp->o_add_type, "IP6")) return NULL;
} else {
if (strcmp(conn->net_type, "IN")) return NULL;
if (strcmp(conn->add_type, "IP4") && strcmp(conn->add_type, "IP6")) return NULL;
}
/*do we support transport*/
if (strcmp(media->Profile, "RTP/AVP") && strcmp(media->Profile, "RTP/AVP/TCP")
&& strcmp(media->Profile, "RTP/SAVP") && strcmp(media->Profile, "RTP/SAVP/TCP")
) return NULL;
/*check RTP map. For now we only support 1 RTPMap*/
if (media->fmt_list || (gf_list_count(media->RTPMaps) > 1)) return NULL;
/*check payload type*/
map = (GF_RTPMap*)gf_list_get(media->RTPMaps, 0);
/*this is an ESD-URL setup, we likely have namespace conflicts so overwrite given ES_ID
by the app one (client side), but keep control (server side) if provided*/
if (input_stream) {
ESID = input_stream->ES_ID;
if (!ctrl) ctrl = input_stream->control;
tmp = input_stream;
} else {
tmp = RP_FindChannel(rtp, NULL, ESID, NULL, 0);
if (tmp) return NULL;
GF_SAFEALLOC(tmp, RTPStream);
tmp->owner = rtp;
}
/*create an RTP channel*/
tmp->rtp_ch = gf_rtp_new();
if (ctrl) tmp->control = gf_strdup(ctrl);
tmp->ES_ID = ESID;
tmp->OD_ID = ODID;
tmp->mid = mid;
tmp->prev_stream = prev_stream;
tmp->base_stream = base_stream;
memset(&trans, 0, sizeof(GF_RTSPTransport));
trans.Profile = media->Profile;
trans.source = conn ? conn->host : sdp->o_address;
trans.IsUnicast = gf_sk_is_multicast_address(trans.source) ? 0 : 1;
if (!trans.IsUnicast) {
trans.port_first = media->PortNumber;
trans.port_last = media->PortNumber + 1;
trans.TTL = conn ? conn->TTL : 0;
} else {
trans.client_port_first = media->PortNumber;
trans.client_port_last = media->PortNumber + 1;
trans.port_first = s_port_first ? s_port_first : trans.client_port_first;
trans.port_last = s_port_last ? s_port_last : trans.client_port_last;
}
if (gf_rtp_setup_transport(tmp->rtp_ch, &trans, NULL) != GF_OK) {
RP_DeleteStream(tmp);
return NULL;
}
/*setup depacketizer*/
tmp->depacketizer = gf_rtp_depacketizer_new(media, rtp_sl_packet_cbk, tmp);
if (!tmp->depacketizer) {
RP_DeleteStream(tmp);
return NULL;
}
/*setup channel*/
gf_rtp_setup_payload(tmp->rtp_ch, map);
// tmp->status = NM_Disconnected;
ctrl = (char *) gf_modules_get_option((GF_BaseInterface *) gf_term_get_service_interface(rtp->service), "Streaming", "DisableRTCP");
if (!ctrl || stricmp(ctrl, "yes")) tmp->flags |= RTP_ENABLE_RTCP;
/*setup NAT keep-alive*/
ctrl = (char *) gf_modules_get_option((GF_BaseInterface *) gf_term_get_service_interface(rtp->service), "Streaming", "NATKeepAlive");
if (ctrl) gf_rtp_enable_nat_keepalive(tmp->rtp_ch, atoi(ctrl));
tmp->range_start = Start;
tmp->range_end = End;
if (End != -1.0) tmp->flags |= RTP_HAS_RANGE;
if (force_bcast) tmp->flags |= RTP_FORCE_BROADCAST;
if (is_migration) {
tmp->current_start = (Double) CurrentTime;
tmp->check_rtp_time = RTP_SET_TIME_RTP;
gf_rtp_set_info_rtp(tmp->rtp_ch, rtp_seq, rtp_time, ssrc);
tmp->status = RTP_SessionResume;
}
if (rvc_predef) {
tmp->depacketizer->sl_map.rvc_predef = rvc_predef ;
} else if (rvc_config_att) {
char *rvc_data=NULL;
u32 rvc_size;
Bool is_gz = 0;
if (!strncmp(rvc_config_att, "data:application/rvc-config+xml", 32) && strstr(rvc_config_att, "base64") ) {
char *data = strchr(rvc_config_att, ',');
if (data) {
rvc_size = (u32) strlen(data) * 3 / 4 + 1;
rvc_data = gf_malloc(sizeof(char) * rvc_size);
rvc_size = gf_base64_decode(data, (u32) strlen(data), rvc_data, rvc_size);
rvc_data[rvc_size] = 0;
}
if (!strncmp(rvc_config_att, "data:application/rvc-config+xml+gz", 35)) is_gz = 1;
} else if (!strnicmp(rvc_config_att, "http://", 7) || !strnicmp(rvc_config_att, "https://", 8) ) {
char *mime;
if (gf_dm_get_file_memory(rvc_config_att, &rvc_data, &rvc_size, &mime) == GF_OK) {
if (mime && strstr(mime, "+gz")) is_gz = 1;
if (mime) gf_free(mime);
}
}
if (rvc_data) {
if (is_gz) {
#ifdef GPAC_DISABLE_ZLIB
fprintf(stderr, "Error: no zlib support - RVC not supported in RTP\n");
return NULL;
#endif
gf_gz_decompress_payload(rvc_data, rvc_size, &tmp->depacketizer->sl_map.rvc_config, &tmp->depacketizer->sl_map.rvc_config_size);
gf_free(rvc_data);
} else {
tmp->depacketizer->sl_map.rvc_config = rvc_data;
tmp->depacketizer->sl_map.rvc_config_size = rvc_size;
}
}
}
return tmp;
}
