/*process describe reply*/
void RP_ProcessDescribe(RTPSession *sess, RTSPCommand *com, M4Err e)
{
RTPStream *ch;
ChannelDescribe *ch_desc;
ch = NULL;
ch_desc = com->user_data;
if (e) goto exit;
switch (sess->rtsp_rsp->ResponseCode) {
//TODO handle all 3xx codes (redirections)
case NC_RTSP_Multiple_Choice:
e = ch_desc ? M4ChannelNotFound : M4URLNotFound;
goto exit;
case NC_RTSP_Not_Found:
e = M4URLNotFound;
goto exit;
case NC_RTSP_OK:
break;
default:
//we should have a basic error code mapping here
e = M4ServiceError;
goto exit;
}
ch = NULL;
if (ch_desc) {
ch = RP_FindChannel(sess->owner, ch_desc->channel, ch_desc->ES_ID, ch_desc->esd_url, 0);
} else {
NM_OnMessage(sess->owner->service, M4OK, "Connected");
}
/*error on loading SDP is done internally*/
RP_LoadSDP(sess->owner, sess->rtsp_rsp->body, sess->rtsp_rsp->Content_Length, ch);
if (!ch_desc) goto exit;
if (!ch) {
e = M4ChannelNotFound;
goto exit;
}
e = RP_SetupChannel(ch, ch_desc);
exit:
if (e) {
if (!ch_desc) {
NM_OnConnect(sess->owner->service, NULL, e);
} else if (ch) {
RP_ConfirmChannelConnect(ch, e);
} else {
NM_OnConnect(sess->owner->service, ch_desc->channel, e);
}
}
if (ch_desc) free(ch_desc);
com->user_data = NULL;
}
static GF_Err RP_ChannelReleaseSLP(GF_InputService *plug, LPNETCHANNEL channel)
{
RTPStream *ch;
RTPClient *priv = (RTPClient *)plug->priv;
ch = RP_FindChannel(priv, channel, 0, NULL, 0);
if (!ch) return GF_STREAM_NOT_FOUND;
if (ch->rtp_ch || ch->rtsp || !ch->control) return GF_SERVICE_ERROR;
if (ch->status != RTP_Running) return GF_SERVICE_ERROR;
/*this will trigger EOS at next fetch*/
ch->current_start = -1.0;
return GF_OK;
}
/*send describe*/
void RP_Describe(RTPSession *sess, char *esd_url, LPNETCHANNEL channel)
{
RTPStream *ch;
ChannelDescribe *ch_desc;
RTSPCommand *com;
/*locate the channel by URL - if we have one, this means the channel is already described
this happens when 2 ESD with URL use the same RTSP service - skip describe and send setup*/
if (esd_url || channel) {
ch = RP_FindChannel(sess->owner, channel, 0, esd_url, 0);
if (ch) {
if (!ch->channel) ch->channel = channel;
ch_desc = malloc(sizeof(ChannelDescribe));
ch_desc->esd_url = esd_url ? strdup(esd_url) : NULL;
ch_desc->channel = channel;
RP_SetupChannel(ch, ch_desc);
if (esd_url) free(ch_desc->esd_url);
free(ch_desc);
return;
}
/*channel not found, send describe on service*/
}
/*send describe*/
com = RTSP_NewCommand();
com->method = strdup(RTSP_DESCRIBE);
if (channel || esd_url) {
com->Accept = strdup("application/sdp");
com->ControlString = esd_url ? strdup(esd_url) : NULL;
ch_desc = malloc(sizeof(ChannelDescribe));
ch_desc->esd_url = esd_url ? strdup(esd_url) : NULL;
ch_desc->channel = channel;
com->user_data = ch_desc;
} else {
//always accept both SDP and IOD
com->Accept = strdup("application/sdp, application/mpeg4-iod");
// com->Accept = strdup("application/sdp");
}
com->Bandwidth = sess->owner->bandwidth;
MX_P(sess->owner->mx);
ChainAddEntry(sess->rtsp_commands, com);
MX_V(sess->owner->mx);
}
static GF_Err RP_DisconnectChannel(GF_InputService *plug, LPNETCHANNEL channel)
{
RTPStream *ch;
RTPClient *priv = (RTPClient *)plug->priv;
GF_LOG(GF_LOG_DEBUG, GF_LOG_RTP, ("[RTP] Disconnecting channel @%08x\n", channel));
ch = RP_FindChannel(priv, channel, 0, NULL, 0);
if (!ch) return GF_STREAM_NOT_FOUND;
gf_mx_p(priv->mx);
/*disconnect stream BUT DO NOT DELETE IT since we don't store SDP*/
ch->flags &= ~RTP_CONNECTED;
ch->channel = NULL;
gf_mx_v(priv->mx);
gf_term_on_disconnect(priv->service, channel, GF_OK);
return GF_OK;
}
void RP_DeleteStream(RTPStream *ch)
{
if (ch->rtsp) {
if (ch->status == RTP_Running) {
RP_Teardown(ch->rtsp, ch);
ch->status = RTP_Disconnected;
}
RP_RemoveStream(ch->owner, ch);
} else {
RP_FindChannel(ch->owner, ch->channel, 0, NULL, 1);
}
if (ch->depacketizer) gf_rtp_depacketizer_del(ch->depacketizer);
if (ch->rtp_ch) gf_rtp_del(ch->rtp_ch);
if (ch->control) gf_free(ch->control);
if (ch->session_id) gf_free(ch->session_id);
gf_free(ch);
}
/*filter describe commands in case of ESD URLs*/
Bool RP_PreprocessDescribe(RTPSession *sess, RTSPCommand *com)
{
RTPStream *ch;
ChannelDescribe *ch_desc;
/*not a channel describe*/
if (!com->user_data) {
NM_OnMessage(sess->owner->service, M4OK, "Connecting...");
return 1;
}
ch_desc = (ChannelDescribe *)com->user_data;
ch = RP_FindChannel(sess->owner, NULL, ch_desc->ES_ID, ch_desc->esd_url, 0);
if (!ch) return 1;
/*channel has been described already, skip describe and send setup directly*/
RP_SetupChannel(ch, ch_desc);
if (ch_desc->esd_url) free(ch_desc->esd_url);
free(ch_desc);
return 0;
}
static GF_Err RP_ChannelGetSLP(GF_InputService *plug, LPNETCHANNEL channel, char **out_data_ptr, u32 *out_data_size, GF_SLHeader *out_sl_hdr, Bool *sl_compressed, GF_Err *out_reception_status, Bool *is_new_data)
{
char *data;
RTPStream *ch;
RTPClient *priv = (RTPClient *)plug->priv;
ch = RP_FindChannel(priv, channel, 0, NULL, 0);
if (!ch) return GF_STREAM_NOT_FOUND;
if (ch->rtp_ch || ch->rtsp || !ch->control) return GF_SERVICE_ERROR;
if (ch->status != RTP_Running) return GF_SERVICE_ERROR;
data = strstr(ch->control, ";base64");
if (!data) return GF_SERVICE_ERROR;
if (ch->current_start>=0) {
*sl_compressed = 0;
memset(out_sl_hdr, 0, sizeof(GF_SLHeader));
out_sl_hdr->accessUnitEndFlag = 1;
out_sl_hdr->accessUnitStartFlag = 1;
out_sl_hdr->compositionTimeStamp = (u64) (ch->current_start * ch->ts_res);
out_sl_hdr->compositionTimeStampFlag = 1;
out_sl_hdr->randomAccessPointFlag = 1;
*out_reception_status = GF_OK;
*is_new_data = (ch->flags & GF_RTP_NEW_AU) ? 1 : 0;
/*decode data*/
data = strstr(data, ",");
data += 1;
*out_data_size = gf_base64_decode(data, strlen(data), ch->buffer, RTP_BUFFER_SIZE);
/*FIXME - currently only support for empty SL header*/
*out_data_ptr = ch->buffer;
ch->flags &= ~GF_RTP_NEW_AU;
} else {
*out_data_ptr = NULL;
*out_data_size = 0;
*out_reception_status = GF_EOS;
ch->flags |= RTP_EOS;
}
return GF_OK;
}
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;
}
static GF_Err RP_ConnectChannel(GF_InputService *plug, LPNETCHANNEL channel, const char *url, Bool upstream)
{
u32 ESID;
RTPStream *ch;
RTSPSession *sess;
char *es_url;
RTPClient *priv = (RTPClient *)plug->priv;
if (upstream) return GF_NOT_SUPPORTED;
GF_LOG(GF_LOG_DEBUG, GF_LOG_RTP, ("[RTP] Connecting channel @%08x - %s\n", channel, url));
ch = RP_FindChannel(priv, channel, 0, (char *) url, 0);
if (ch && (ch->status != RTP_Disconnected) ) return GF_SERVICE_ERROR;
es_url = NULL;
sess = NULL;
if (strstr(url, "ES_ID=")) {
sscanf(url, "ES_ID=%ud", &ESID);
/*first case: simple URL (same namespace)*/
ch = RP_FindChannel(priv, NULL, ESID, NULL, 0);
/*this should not happen, the sdp must describe all streams in the service*/
if (!ch) return GF_STREAM_NOT_FOUND;
/*assign app channel*/
ch->channel = channel;
sess = ch->rtsp;
}
/*rtsp url - create a session if needed*/
else if (!strnicmp(url, "rtsp://", 7) || !strnicmp(url, "rtspu://", 8)) {
sess = RP_CheckSession(priv, (char *) url);
if (!sess) sess = RP_NewSession(priv, (char *) url);
es_url = (char *) url;
}
/*data: url*/
else if (strstr(url, "data:application/mpeg4-od-au;base64")
|| strstr(url, "data:application/mpeg4-bifs-au;base64")
|| strstr(url, "data:application/mpeg4-es-au;base64")
) {
GF_SAFEALLOC(ch, RTPStream);
ch->control = gf_strdup(url);
ch->owner = priv;
ch->channel = channel;
ch->status = RTP_Connected;
/*register*/
gf_list_add(priv->channels, ch);
RP_ConfirmChannelConnect(ch, GF_OK);
return GF_OK;
}
/*session migration resume - don't send data to the server*/
if (ch->status==RTP_SessionResume) {
ch->flags |= RTP_CONNECTED;
RP_InitStream(ch, 0);
RP_ConfirmChannelConnect(ch, GF_OK);
return GF_OK;
}
/*send a DESCRIBE (not a setup) on the channel. If the channel is already created then the
describe is skipped and a SETUP is sent directly, otherwise the channel is first described then setup*/
if (sess) RP_Describe(sess, es_url, channel);
/*otherwise confirm channel connection*/
else RP_ConfirmChannelConnect(ch, GF_OK);
return GF_OK;
}
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;
}
void RP_ProcessUserCommand(RTPSession *sess, RTSPCommand *com, M4Err e)
{
ChannelControl *ch_ctrl;
RTPStream *ch, *agg_ch;
u32 i;
RTP_Info *info;
ch_ctrl = com->user_data;
ch = ch_ctrl->ch;
if (!channel_is_valid(sess->owner, ch)) {
free(ch_ctrl);
com->user_data = NULL;
return;
}
assert(ch->channel==ch_ctrl->com.base.on_channel);
/*some consistency checking: on interleaved sessions, some servers do NOT reply to the
teardown. If our command is STOP just skip the error notif*/
if (e) {
if (!strcmp(com->method, RTSP_TEARDOWN)) {
goto process_reply;
} else {
goto err_exit;
}
}
switch (sess->rtsp_rsp->ResponseCode) {
//handle all 3xx codes (redirections)
case NC_RTSP_Method_Not_Allowed:
e = M4NotSupported;
goto err_exit;
case NC_RTSP_OK:
break;
default:
//we should have a basic error code mapping here
e = M4ServiceError;
goto err_exit;
}
process_reply:
NM_OnCommand(sess->owner->service, &ch_ctrl->com, M4OK);
if ( (ch_ctrl->com.command_type==CHAN_PLAY)
|| (ch_ctrl->com.command_type==CHAN_SET_SPEED)
|| (ch_ctrl->com.command_type==CHAN_RESUME) ) {
//auto-detect any aggregated control if not done yet
if (ChainGetCount(sess->rtsp_rsp->RTP_Infos) > 1) {
sess->has_aggregated_control = 1;
}
//process all RTP infos
for (i=0;i<ChainGetCount(sess->rtsp_rsp->RTP_Infos); i++) {
info = ChainGetEntry(sess->rtsp_rsp->RTP_Infos, i);
agg_ch = RP_FindChannel(sess->owner, NULL, 0, info->url, 0);
if (!agg_ch || (agg_ch->rtsp != sess) ) continue;
/*if play/seeking we must send update RTP/NPT link*/
if (ch_ctrl->com.command_type != CHAN_RESUME) {
agg_ch->check_rtp_time = 1;
}
/*this is used to discard RTP packets re-sent on resume*/
else {
agg_ch->check_rtp_time = 2;
}
/* reset the buffers */
RP_InitStream(agg_ch, 1);
RTP_SetInfo(agg_ch->rtp_ch, info->seq, info->rtp_time, info->ssrc);
agg_ch->status = RTP_Running;
/*skip next play command on this channel if aggregated control*/
if (ch!=agg_ch && ch->rtsp->has_aggregated_control) agg_ch->skip_next_command = 1;
if (RTP_IsInterleaved(agg_ch->rtp_ch)) {
RTSP_RegisterTCPChannel(sess->session,
agg_ch,
RTP_GetLowInterleavedID(agg_ch->rtp_ch),
RTP_GetHighInterleavedID(agg_ch->rtp_ch));
}
}
/*no rtp info (just in case), no time mapped - set to 0 and specify we're not interactive*/
if (!i) {
ch->current_start = 0.0;
ch->check_rtp_time = 1;
RP_InitStream(ch, 1);
ch->status = RTP_Running;
if (RTP_IsInterleaved(ch->rtp_ch)) {
RTSP_RegisterTCPChannel(sess->session,
ch, RTP_GetLowInterleavedID(ch->rtp_ch), RTP_GetHighInterleavedID(ch->rtp_ch));
}
}
ch->skip_next_command = 0;
} else if (ch_ctrl->com.command_type == CHAN_PAUSE) {
SkipCommandOnSession(ch);
ch->skip_next_command = 0;
} else if (ch_ctrl->com.command_type == CHAN_STOP) {
assert(0);
}
free(ch_ctrl);
com->user_data = NULL;
return;
err_exit:
ch->status = RTP_Disconnected;
NM_OnCommand(sess->owner->service, &ch_ctrl->com, e);
RTSP_ResetAggregation(ch->rtsp->session);
ch->check_rtp_time = 0;
free(ch_ctrl);
com->user_data = NULL;
}
