/* this function is called internally to get rid of a call.
It performs the following tasks:
- remove the call from the internal list of calls
- - unref the LinphoneCall object
- update the call logs accordingly
*/
cp->audio_bw=bandwidth;
}
+#ifdef VIDEO_ENABLED
+/**
+ * Request remote side to send us a Video Fast Update.
+**/
+void linphone_call_send_vfu_request(LinphoneCall *call)
+{
+ if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
+ sal_call_send_vfu_request(call->op);
+}
+#endif
+
/**
*
**/
int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
audio_stream_enable_noise_gate(audiostream,enabled);
}
+
if (lc->a_rtp)
rtp_session_set_transports(audiostream->session,lc->a_rtp,lc->a_rtcp);
}
}
-
-
-
static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
int bw;
const MSList *elem;
bool_t first=TRUE;
int remote_bw=0;
LinphoneCore *lc=call->core;
+ int up_ptime=0;
*used_pt=-1;
for(elem=desc->payloads;elem!=NULL;elem=elem->next){
PayloadType *pt=(PayloadType*)elem->data;
int number;
- if (first) {
+ if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
if (desc->type==SalAudio){
linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
+ up_ptime=linphone_core_get_upload_ptime(lc);
}
*used_pt=payload_type_get_number(pt);
first=FALSE;
pt->normal_bitrate=-1;
}
if (desc->ptime>0){
+ up_ptime=desc->ptime;
+ }
+ if (up_ptime>0){
char tmp[40];
- snprintf(tmp,sizeof(tmp),"ptime=%i",desc->ptime);
+ snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
payload_type_append_send_fmtp(pt,tmp);
}
number=payload_type_get_number(pt);
const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpAvp,SalVideo);
#endif
+ bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
if(call->audiostream == NULL)
{
const char *playfile=lc->play_file;
const char *recfile=lc->rec_file;
call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
- bool_t use_ec;
+ bool_t use_ec,use_arc_audio=use_arc;
if (used_pt!=-1){
if (playcard==NULL) {
playcard=NULL;
}
use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
-#if defined(VIDEO_ENABLED) && defined(ANDROID)
- /*On android we have to disable the echo canceller to preserve CPU for video codecs */
- if (vstream && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL)
+#if defined(VIDEO_ENABLED)
+ if (vstream && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
+ /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
+ use_arc_audio=FALSE;
+ #if defined(ANDROID)
+ /*On android we have to disable the echo canceller to preserve CPU for video codecs */
use_ec=FALSE;
+ #endif
+ }
#endif
+ audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc_audio);
audio_stream_start_full(
call->audiostream,
call->audio_profile,
static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
audio_stream_get_local_rtp_stats (st,&log->local_stats);
+ log->quality=audio_stream_get_average_quality_rating(st);
}
void linphone_call_stop_media_streams(LinphoneCall *call){
}
}
-#ifdef VIDEO_ENABLED
-/**
- * Request remote side to send us VFU.
-**/
-void linphone_call_send_vfu_request(LinphoneCall *call)
-{
- if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
- sal_call_send_vfu_request(call->op);
-}
-#endif
+
void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
}
}
+/**
+ * @addtogroup call_misc
+ * @{
+**/
+
/**
* Returns the measured sound volume played locally (received from remote)
* It is expressed in dbm0.
return LINPHONE_VOLUME_DB_LOWEST;
}
+/**
+ * Obtain real-time quality rating of the call
+ *
+ * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
+ * during all the duration of the call. This function returns its value at the time of the function call.
+ * It is expected that the rating is updated at least every 5 seconds or so.
+ * The rating is a floating point number comprised between 0 and 5.
+ *
+ * 4-5 = good quality <br>
+ * 3-4 = average quality <br>
+ * 2-3 = poor quality <br>
+ * 1-2 = very poor quality <br>
+ * 0-1 = can't be worse, mostly unusable <br>
+ *
+ * @returns The function returns -1 if no quality measurement is available, for example if no
+ * active audio stream exist. Otherwise it returns the quality rating.
+**/
+float linphone_call_get_current_quality(LinphoneCall *call){
+ if (call->audiostream){
+ return audio_stream_get_quality_rating(call->audiostream);
+ }
+ return -1;
+}
+
+/**
+ * Returns call quality averaged over all the duration of the call.
+ *
+ * See linphone_call_get_current_quality() for more details about quality measurement.
+**/
+float linphone_call_get_average_quality(LinphoneCall *call){
+ if (call->audiostream){
+ return audio_stream_get_average_quality_rating(call->audiostream);
+ }
+ return -1;
+}
+
+/**
+ * @}
+**/
static void display_bandwidth(RtpSession *as, RtpSession *vs){
ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
if (call->videostream!=NULL)
video_stream_iterate(call->videostream);
#endif
+ if (call->audiostream!=NULL)
+ audio_stream_iterate(call->audiostream);
if (one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
if (disconnected)
projects / linphone / blobdiff