void RP_SaveSessionState(RTPClient *rtp)
{
GF_Err e;
char *sdp_buf;
const char *opt;
GF_X_Attribute*att;
u32 i, j;
GF_SDPInfo *sdp;
RTSPSession *sess = NULL;
if (!rtp->session_state_data) return;
sdp_buf = rtp->session_state_data + strlen("data:application/sdp,");
sdp = gf_sdp_info_new();
e = gf_sdp_info_parse(sdp, sdp_buf, (u32) strlen(sdp_buf) );
for (i=0; i<gf_list_count(rtp->channels); i++) {
GF_SDPMedia *media = NULL;
RTPStream *ch = gf_list_get(rtp->channels, i);
if (!ch->control) continue;
for (j=0; j<gf_list_count(sdp->media_desc); j++) {
u32 k;
GF_SDPMedia *med = (GF_SDPMedia*)gf_list_get(sdp->media_desc, j);
for (k=0; k<gf_list_count(med->Attributes); k++) {
att = (GF_X_Attribute*)gf_list_get(med->Attributes, k);
if (!stricmp(att->Name, "control") && (strstr(att->Value, ch->control)!=NULL) ) {
media = med;
break;
}
}
if (media)
break;
}
if (!media) continue;
if (ch->rtp_ch->net_info.IsUnicast) {
char szPorts[4096];
u16 porta, portb;
media->PortNumber = ch->rtp_ch->net_info.client_port_first;
/*remove x-server-port extension*/
for (j=0; j<gf_list_count(media->Attributes); j++) {
att = (GF_X_Attribute*)gf_list_get(media->Attributes, j);
if (!stricmp(att->Name, "x-stream-state") ) {
gf_free(att->Name);
gf_free(att->Value);
gf_free(att);
gf_list_rem(media->Attributes, j);
}
}
ch->current_start += gf_rtp_get_current_time(ch->rtp_ch);
GF_SAFEALLOC(att, GF_X_Attribute);
att->Name = gf_strdup("x-stream-state");
porta = ch->rtp_ch->net_info.port_first ? ch->rtp_ch->net_info.port_first : ch->rtp_ch->net_info.client_port_first;
portb = ch->rtp_ch->net_info.port_last ? ch->rtp_ch->net_info.port_last : ch->rtp_ch->net_info.client_port_last;
sprintf(szPorts, "server-port=%d-%d;ssrc=%X;npt=%g;seq=%d;rtptime=%d",
porta,
portb,
ch->rtp_ch->SenderSSRC,
ch->current_start,
ch->rtp_ch->rtp_first_SN,
ch->rtp_ch->rtp_time
);
att->Value = gf_strdup(szPorts);
gf_list_add(media->Attributes, att);
if (ch->rtsp)
sess = ch->rtsp;
} else {
media->PortNumber = ch->rtp_ch->net_info.port_first;
}
}
/*remove x-server-port/x-session-id extension*/
for (j=0; j<gf_list_count(sdp->Attributes); j++) {
att = (GF_X_Attribute*)gf_list_get(sdp->Attributes, j);
if (!stricmp(att->Name, "x-session-id") || !stricmp(att->Name, "x-session-name")
) {
gf_free(att->Name);
gf_free(att->Value);
gf_free(att);
gf_list_rem(sdp->Attributes, j);
}
}
if (sess) {
char szURL[4096];
if (sess->session_id) {
GF_SAFEALLOC(att, GF_X_Attribute);
att->Name = gf_strdup("x-session-id");
att->Value = gf_strdup(sess->session_id);
gf_list_add(sdp->Attributes, att);
}
GF_SAFEALLOC(att, GF_X_Attribute);
att->Name = gf_strdup("x-session-name");
sprintf(szURL, "rtsp://%s:%d/%s", sess->session->Server, sess->session->Port, sess->session->Service);
att->Value = gf_strdup(szURL);
gf_list_add(sdp->Attributes, att);
}
gf_free(rtp->session_state_data);
sdp_buf = NULL;
gf_sdp_info_write(sdp, &sdp_buf);
if (sdp_buf) {
rtp->session_state_data = gf_malloc(sizeof(char) * (strlen("data:application/sdp,") + strlen(sdp_buf) + 1) );
strcpy(rtp->session_state_data, "data:application/sdp,");
strcat(rtp->session_state_data, sdp_buf);
gf_free(sdp_buf);
}
gf_sdp_info_del(sdp);
opt = (char *) gf_modules_get_option((GF_BaseInterface *) gf_term_get_service_interface(rtp->service), "Streaming", "SessionMigrationServer");
if (opt) {
if (rtp->dnload) gf_term_download_del(rtp->dnload);
rtp->dnload = NULL;
if (strnicmp(opt, "http://", 7)) {
rtp->dnload = gf_term_download_new(rtp->service, opt, GF_NETIO_SESSION_NOT_THREADED, MigrateSDP_NetIO, rtp);
while (1) {
char buffer[100];
u32 read;
e = gf_dm_sess_fetch_data(rtp->dnload, buffer, 100, &read);
if (e && (e!=GF_IP_NETWORK_EMPTY)) break;
}
gf_term_download_del(rtp->dnload);
rtp->dnload = NULL;
} else {
FILE *f = gf_f64_open(opt, "wt");
if (f) {
sdp_buf = rtp->session_state_data + strlen("data:application/sdp,");
gf_fwrite(sdp_buf, 1, strlen(sdp_buf), f);
fclose(f);
} else {
e = GF_IO_ERR;
}
}
if (e<0) {
gf_term_on_message(sess->owner->service, e, "Error saving session state");
}
}
}
void RP_ProcessRTP(RTPStream *ch, char *pck, u32 size)
{
GF_NetworkCommand com;
GF_Err e;
GF_RTPHeader hdr;
u32 PayloadStart;
ch->rtp_bytes += size;
/*first decode RTP*/
e = gf_rtp_decode_rtp(ch->rtp_ch, pck, size, &hdr, &PayloadStart);
/*corrupted or NULL data*/
if (e || (PayloadStart >= size)) {
//gf_term_on_sl_packet(ch->owner->service, ch->channel, NULL, 0, NULL, GF_CORRUPTED_DATA);
return;
}
/*if we must notify some timing, do it now. If the channel has no range, this should NEVER be called*/
if (ch->check_rtp_time /*&& gf_rtp_is_active(ch->rtp_ch)*/) {
Double ch_time;
/*it may happen that we still receive packets from a previous "play" request. If this is the case,
filter until we reach the indicated rtptime*/
if (ch->rtp_ch->rtp_time
&& (ch->rtp_ch->rtp_first_SN > hdr.SequenceNumber)
&& (ch->rtp_ch->rtp_time < hdr.TimeStamp)
) {
GF_LOG(GF_LOG_WARNING, GF_LOG_RTP, ("[RTP] Rejecting too early packet (TS %d vs signaled rtp time %d - diff %d ms)\n",
hdr.TimeStamp, ch->rtp_ch->rtp_time, ((hdr.TimeStamp - ch->rtp_ch->rtp_time)*1000) / ch->rtp_ch->TimeScale));
return;
}
ch_time = gf_rtp_get_current_time(ch->rtp_ch);
/*this is the first packet on the channel (no PAUSE)*/
if (ch->check_rtp_time == RTP_SET_TIME_RTP) {
/*Note: in a SEEK with RTSP, the rtp-info time given by the server is
the rtp time of the desired range. But the server may (and should) send from
the previous I frame on video, so the time of the first rtp packet after
a SEEK can actually be less than CurrentStart. We don't drop these
packets in order to see the maximum video. We could drop it, this would mean
wait for next RAP...*/
memset(&com, 0, sizeof(com));
com.command_type = GF_NET_CHAN_MAP_TIME;
com.base.on_channel = ch->channel;
if (ch->rtsp) {
com.map_time.media_time = ch->current_start + ch_time;
} else {
com.map_time.media_time = 0;
}
com.map_time.timestamp = hdr.TimeStamp;
com.map_time.reset_buffers = 0;
gf_term_on_command(ch->owner->service, &com, GF_OK);
GF_LOG(GF_LOG_INFO, GF_LOG_RTP, ("[RTP] Mapping RTP Time seq %d TS %d Media Time %g - rtp info seq %d TS %d\n",
hdr.SequenceNumber, hdr.TimeStamp, com.map_time.media_time, ch->rtp_ch->rtp_first_SN, ch->rtp_ch->rtp_time
));
/*skip RTCP clock init when RTSP is used*/
if (ch->rtsp) ch->rtcp_init = 1;
// if (ch->depacketizer->payt==GF_RTP_PAYT_H264_AVC) ch->depacketizer->flags |= GF_RTP_AVC_WAIT_RAP;
}
/*this is RESUME on channel, filter packet based on time (darwin seems to send
couple of packet before)
do not fetch if we're below 10 ms or <0, because this means we already have
this packet - as the PAUSE is issued with the RTP currentTime*/
else if (ch_time <= 0.021) {
return;
}
ch->check_rtp_time = RTP_SET_TIME_NONE;
}
gf_rtp_depacketizer_process(ch->depacketizer, &hdr, pck + PayloadStart, size - PayloadStart);
/*last check: signal EOS if we're close to end range in case the server do not send RTCP BYE*/
if ((ch->flags & RTP_HAS_RANGE) && !(ch->flags & RTP_EOS) ) {
/*also check last CTS*/
Double ts = (Double) ((u32) ch->depacketizer->sl_hdr.compositionTimeStamp - hdr.TimeStamp);
ts /= gf_rtp_get_clockrate(ch->rtp_ch);
if (ABSDIFF(ch->range_end, (ts + ch->current_start + gf_rtp_get_current_time(ch->rtp_ch)) ) < 0.2) {
ch->flags |= RTP_EOS;
ch->stat_stop_time = gf_sys_clock();
gf_term_on_sl_packet(ch->owner->service, ch->channel, NULL, 0, NULL, GF_EOS);
}
}
}
