#include <net/if.h>
#include <ifaddrs.h>
#endif
-
+#include <math.h>
#if !defined(WIN32)
return pt->normal_bitrate;
}
+/*
+ *((codec-birate*ptime/8) + RTP header + UDP header + IP header)*8/ptime;
+ *ptime=1/npacket
+ */
static double get_audio_payload_bandwidth(LinphoneCore *lc, const PayloadType *pt){
double npacket=50;
double packet_size;
int bitrate;
+ if (strcmp(payload_type_get_mime(&payload_type_aaceld_44k), payload_type_get_mime(pt))==0) {
+ /*special case of aac 44K because ptime= 10ms*/
+ npacket=100;
+ }
+
bitrate=get_codec_bitrate(lc,pt);
- packet_size= (((double)bitrate)/(50*8))+UDP_HDR_SZ+RTP_HDR_SZ+IP4_HDR_SZ;
+ packet_size= (((double)bitrate)/(npacket*8))+UDP_HDR_SZ+RTP_HDR_SZ+IP4_HDR_SZ;
return packet_size*8.0*npacket;
}
void linphone_core_update_allocated_audio_bandwidth_in_call(LinphoneCall *call, const PayloadType *pt){
- call->audio_bw=(int)(get_audio_payload_bandwidth(call->core,pt)/1000.0);
+ call->audio_bw=(int)(ceil(get_audio_payload_bandwidth(call->core,pt)/1000.0)); /*rounding codec bandwidth should be avoid, specially for AMR*/
ms_message("Audio bandwidth for this call is %i",call->audio_bw);
}
for (i = 0; i < md->n_total_streams; i++) {
const SalStreamDescription *stream = &md->streams[i];
IceCheckList *cl = ice_session_check_list(call->ice_session, i);
- if (cl == NULL) {
+ if ((cl == NULL) && (i < md->n_active_streams)) {
cl = ice_check_list_new();
ice_session_add_check_list(call->ice_session, cl);
switch (stream->type) {
return ret;
}
+bool_t linphone_core_tone_indications_enabled(LinphoneCore*lc){
+ return lp_config_get_int(lc->config,"sound","tone_indications",1);
+}
#ifdef HAVE_GETIFADDRS