factorize code related to call quality indicator.

add accessors to loss and late rates.

include/mediastreamer2/mediastream.h

@@ -86,10 +86,15 @@ struct _MediaStream {
 	MSFilter *decoder;
 	MSFilter *voidsink;
 	MSBitrateController *rc;
+	MSQualityIndicator *qi;
 	IceCheckList *ice_check_list;
 	OrtpZrtpContext *zrtp_context;
 	srtp_t srtp_session;
+	time_t start_time;
+	time_t last_iterate_time;
 	bool_t use_rc;
+	bool_t is_beginning;
+	bool_t pad[2];
 };
 
 typedef struct _MediaStream MediaStream;
@@ -106,7 +111,14 @@ MS2_PUBLIC void media_stream_enable_adaptive_jittcomp(MediaStream *stream, bool_
 
 MS2_PUBLIC bool_t media_stream_enable_srtp(MediaStream* stream, enum ortp_srtp_crypto_suite_t suite, const char* snd_key, const char* rcv_key);
 
+MS2_PUBLIC const MSQualityIndicator *media_stream_get_quality_indicator(MediaStream *stream);
+
+MS2_PUBLIC float media_stream_get_quality_rating(MediaStream *stream);
 
+MS2_PUBLIC float media_stream_get_average_quality_rating(MediaStream *stream);
+
+/*shall only called internally*/
+void media_stream_iterate(MediaStream * stream);
 /**
  * @addtogroup audio_stream_api
  * @{
@@ -140,16 +152,12 @@ struct _AudioStream
 	uint64_t last_packet_count;
 	time_t last_packet_time;
 	EchoLimiterType el_type; /*use echo limiter: two MSVolume, measured input level controlling local output level*/
-	MSQualityIndicator *qi;
-	time_t start_time;
-	time_t last_iterate_time;
 	uint32_t features;
 	bool_t play_dtmfs;
 	bool_t use_gc;
 	bool_t use_agc;
 	bool_t eq_active;
 	bool_t use_ng;/*noise gate*/
-	bool_t is_beginning;
 	bool_t is_ec_delay_set;
 };
 

include/mediastreamer2/qualityindicator.h

@@ -30,17 +30,51 @@ typedef struct _MSQualityIndicator MSQualityIndicator;
 extern "C"{
 #endif
 
-MSQualityIndicator *ms_quality_indicator_new(RtpSession *session);
+/**
+ * Creates a quality indicator object.
+ * @param session the RtpSession being monitored.
+**/ 
+MS2_PUBLIC MSQualityIndicator *ms_quality_indicator_new(RtpSession *session);
+
+/**
+ * Updates quality indicator based on a received RTCP packet.
+**/
+MS2_PUBLIC void ms_quality_indicator_update_from_feedback(MSQualityIndicator *qi, mblk_t *rtcp);
+
+/**
+ * Updates quality indicator based on the local statistics directly computed by the RtpSession used when creating the indicator.
+ * This function must be called typically every second.
+**/
+MS2_PUBLIC void ms_quality_indicator_update_local(MSQualityIndicator *qi);
+
+/**
+ * Return the real time rating of the session. Its value is between 0 (worse) and 5.0 (best).
+**/
+MS2_PUBLIC float ms_quality_indicator_get_rating(MSQualityIndicator *qi);
+
+/**
+ * Returns the average rating of the session, that is the rating for all the duration of the session.
+**/
+MS2_PUBLIC float ms_quality_indicator_get_average_rating(MSQualityIndicator *qi);
+
+/**
+ * Returns the local loss rate, as computed internally by ms_quality_indicator_update_local().
+ * This method is for advanced usage.
+**/
+MS2_PUBLIC float ms_quality_indicator_get_local_loss_rate(const MSQualityIndicator *qi);
+
+/**
+ *  Returns the local late rate, as computed internally by ms_quality_indicator_update_local().
+ * This method is for advanced usage.
+**/
+MS2_PUBLIC float ms_quality_indicator_get_local_late_rate(const MSQualityIndicator *qi);
+
+/**
+ * Destroys the quality indicator object.
+**/
+MS2_PUBLIC void ms_quality_indicator_destroy(MSQualityIndicator *qi);
 
-float ms_quality_indicator_get_rating(MSQualityIndicator *qi);
 
-float ms_quality_indicator_get_average_rating(MSQualityIndicator *qi);
-
-void ms_quality_indicator_update_from_feedback(MSQualityIndicator *qi, mblk_t *rtcp);
-
-void ms_quality_indicator_update_local(MSQualityIndicator *qi);
-
-void ms_quality_indicator_destroy(MSQualityIndicator *qi);
 
 #ifdef __cplusplus
 }

src/voip/audiostream.c

@@ -63,7 +63,6 @@ static void audio_stream_free(AudioStream *stream) {
 	if (stream->recorder) ms_filter_destroy(stream->recorder);
 	if (stream->recorder_mixer) ms_filter_destroy(stream->recorder_mixer);
 	if (stream->recorder_file) ms_free(stream->recorder_file);
-	if (stream->qi) ms_quality_indicator_destroy(stream->qi);
 	ms_free(stream);
 }
 
@@ -122,17 +121,12 @@ static void audio_stream_process_rtcp(AudioStream *stream, mblk_t *m){
 			ms_message("audio_stream_iterate(): remote statistics available\n\tremote's interarrival jitter=%u\n"
 			           "\tremote's lost packets percentage since last report=%f\n\tround trip time=%f seconds",ij,flost,rt);
 			if (stream->ms.rc) ms_bitrate_controller_process_rtcp(stream->ms.rc,m);
-			if (stream->qi) ms_quality_indicator_update_from_feedback(stream->qi,m);
+			if (stream->ms.qi) ms_quality_indicator_update_from_feedback(stream->ms.qi,m);
 		}
 	}while(rtcp_next_packet(m));
 }
 
 void audio_stream_iterate(AudioStream *stream){
-	time_t curtime=time(NULL);
-	if (stream->is_beginning && ms_time(NULL)-stream->start_time>15){
-		rtp_session_set_rtcp_report_interval(stream->ms.session,5000);
-		stream->is_beginning=FALSE;
-	}
 	if (stream->ms.evq){
 		OrtpEvent *ev=ortp_ev_queue_get(stream->ms.evq);
 		if (ev!=NULL){
@@ -148,10 +142,7 @@ void audio_stream_iterate(AudioStream *stream){
 			ortp_event_destroy(ev);
 		}
 	}
-	if (stream->ms.ice_check_list) ice_check_list_process(stream->ms.ice_check_list,stream->ms.session);
-	/*we choose to update the quality indicator as much as possible, since local statistics can be computed realtime. */
-	if (stream->qi && curtime>stream->last_iterate_time) ms_quality_indicator_update_local(stream->qi);
-	stream->last_iterate_time=curtime;
+	media_stream_iterate(&stream->ms);
 }
 
 bool_t audio_stream_alive(AudioStream * stream, int timeout){
@@ -191,7 +182,7 @@ static void stop_preload_graph(AudioStream *stream){
 }
 
 bool_t audio_stream_started(AudioStream *stream){
-	return stream->start_time!=0;
+	return stream->ms.start_time!=0;
 }
 
 /* This function is used either on IOS to workaround the long time to initialize the Audio Unit or for ICE candidates gathering. */
@@ -261,7 +252,7 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	else
 		stream->dtmfgen=NULL;
 	rtp_session_signal_connect(rtps,"telephone-event",(RtpCallback)on_dtmf_received,(unsigned long)stream);
-	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)payload_type_changed,(unsigned long)&stream->ms);
+	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)mediastream_payload_type_changed,(unsigned long)&stream->ms);
 	/* creates the local part */
 	if (captcard!=NULL){
 		if (stream->soundread==NULL)
@@ -440,7 +431,6 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	if (stream->ms.use_rc){
 		stream->ms.rc=ms_audio_bitrate_controller_new(stream->ms.session,stream->ms.encoder,0);
 	}
-	stream->qi=ms_quality_indicator_new(stream->ms.session);
 	
 	/* Create PLC */
 	if ((stream->features & AUDIO_STREAM_FEATURE_PLC) != 0) {
@@ -523,8 +513,8 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 				,stream->ms.rtprecv
 				,NULL);
 
-	stream->start_time=ms_time(NULL);
-	stream->is_beginning=TRUE;
+	stream->ms.start_time=ms_time(NULL);
+	stream->ms.is_beginning=TRUE;
 
 	return 0;
 }
@@ -664,6 +654,7 @@ AudioStream *audio_stream_new(int loc_rtp_port, int loc_rtcp_port, bool_t ipv6){
 	/*some filters are created right now to allow configuration by the application before start() */
 	stream->ms.rtpsend=ms_filter_new(MS_RTP_SEND_ID);
 	stream->ms.ice_check_list=NULL;
+	stream->ms.qi=ms_quality_indicator_new(stream->ms.session);
 
 	if (ec_desc!=NULL)
 		stream->ec=ms_filter_new_from_desc(ec_desc);
@@ -770,7 +761,7 @@ void audio_stream_stop(AudioStream * stream)
 		
 		if (stream->dummy){
 			stop_preload_graph(stream);
-		}else if (stream->start_time!=0){
+		}else if (stream->ms.start_time!=0){
 		
 			ms_ticker_detach(stream->ms.ticker,stream->soundread);
 			ms_ticker_detach(stream->ms.ticker,stream->ms.rtprecv);
@@ -850,17 +841,11 @@ void audio_stream_mute_rtp(AudioStream *stream, bool_t val)
 }
 
 float audio_stream_get_quality_rating(AudioStream *stream){
-	if (stream->qi){
-		return ms_quality_indicator_get_rating(stream->qi);
-	}
-	return 0;
+	return media_stream_get_quality_rating(&stream->ms);
 }
 
 float audio_stream_get_average_quality_rating(AudioStream *stream){
-	if (stream->qi){
-		return ms_quality_indicator_get_average_rating(stream->qi);
-	}
-	return 0;
+	return media_stream_get_average_quality_rating(&stream->ms);
 }
 
 void audio_stream_enable_zrtp(AudioStream *stream, OrtpZrtpParams *params){

src/voip/mediastream.c

@@ -173,6 +173,7 @@ void media_stream_free(MediaStream *stream) {
 	if (stream->decoder != NULL) ms_filter_destroy(stream->decoder);
 	if (stream->voidsink != NULL) ms_filter_destroy(stream->voidsink);
 	if (stream->ticker != NULL) ms_ticker_destroy(stream->ticker);
+	if (stream->qi) ms_quality_indicator_destroy(stream->qi);
 }
 
 void media_stream_set_rtcp_information(MediaStream *stream, const char *cname, const char *tool) {
@@ -220,6 +221,10 @@ bool_t media_stream_enable_srtp(MediaStream *stream, enum ortp_srtp_crypto_suite
 	return TRUE;
 }
 
+const MSQualityIndicator *media_stream_get_quality_indicator(MediaStream *stream){
+	return stream->qi;
+}
+
 bool_t ms_is_ipv6(const char *remote) {
 	bool_t ret = FALSE;
 #ifdef INET6
@@ -240,8 +245,37 @@ bool_t ms_is_ipv6(const char *remote) {
 	return ret;
 }
 
-void payload_type_changed(RtpSession *session, unsigned long data) {
+void mediastream_payload_type_changed(RtpSession *session, unsigned long data) {
 	MediaStream *stream = (MediaStream *)data;
 	int pt = rtp_session_get_recv_payload_type(stream->session);
 	media_stream_change_decoder(stream, pt);
 }
+
+void media_stream_iterate(MediaStream *stream){
+	time_t curtime=ms_time(NULL);
+	
+	if (stream->is_beginning && (curtime-stream->start_time>15)){
+		rtp_session_set_rtcp_report_interval(stream->session,5000);
+		stream->is_beginning=FALSE;
+	}
+	if (stream->ice_check_list) ice_check_list_process(stream->ice_check_list,stream->session);
+	/*we choose to update the quality indicator as much as possible, since local statistics can be computed realtime. */
+	if (stream->qi && curtime>stream->last_iterate_time) ms_quality_indicator_update_local(stream->qi);
+	stream->last_iterate_time=curtime;
+}
+
+float media_stream_get_quality_rating(MediaStream *stream){
+	if (stream->qi){
+		return ms_quality_indicator_get_rating(stream->qi);
+	}
+	return -1;
+}
+
+float media_stream_get_average_quality_rating(MediaStream *stream){
+	if (stream->qi){
+		return ms_quality_indicator_get_average_rating(stream->qi);
+	}
+	return -1;
+}
+
+

src/voip/private.h

@@ -47,7 +47,7 @@ MEDIASTREAMER2_INTERNAL_EXPORT RtpSession * create_duplex_rtpsession(int loc_rtp
 
 void start_ticker(MediaStream *stream);
 
-void payload_type_changed(RtpSession *session, unsigned long data);
+void mediastream_payload_type_changed(RtpSession *session, unsigned long data);
 
 const char * media_stream_type_str(MediaStream *stream);
 

src/voip/qualityindicator.c

@@ -39,6 +39,8 @@ struct _MSQualityIndicator{
 	uint32_t last_ext_seq;
 	uint32_t last_late;
 	int count;
+	float cur_late_rate;
+	float cur_loss_rate;
 };
 
 MSQualityIndicator *ms_quality_indicator_new(RtpSession *session){
@@ -140,8 +142,8 @@ void ms_quality_indicator_update_local(MSQualityIndicator *qi){
 	if (lost<0) lost=0; /* will be the case at least the first time, because we don't know the initial sequence number*/
 	if (late<0) late=0;
 
-	loss_rate=(float)lost/(float)recvcnt;
-	late_rate=(float)late/(float)recvcnt;
+	qi->cur_loss_rate=loss_rate=(float)lost/(float)recvcnt;
+	qi->cur_late_rate=late_rate=(float)late/(float)recvcnt;
 	
 	qi->local_rating=compute_rating(loss_rate,0,late_rate,rtp_session_get_round_trip_propagation(qi->session));
 	update_global_rating(qi);
@@ -152,6 +154,14 @@ float ms_quality_indicator_get_average_rating(MSQualityIndicator *qi){
 	return (float)(qi->sum_ratings/(double)qi->count);
 }
 
+float ms_quality_indicator_get_local_loss_rate(const MSQualityIndicator *qi){
+	return qi->cur_loss_rate;
+}
+
+float ms_quality_indicator_get_local_late_rate(const MSQualityIndicator *qi){
+	return qi->cur_late_rate;
+}
+
 void ms_quality_indicator_destroy(MSQualityIndicator *qi){
 	ms_free(qi);
 }

src/voip/videostream.c

@@ -113,7 +113,7 @@ void video_stream_iterate(VideoStream *stream){
 			ortp_event_destroy(ev);
 		}
 	}
-	if (stream->ms.ice_check_list) ice_check_list_process(stream->ms.ice_check_list,stream->ms.session);
+	media_stream_iterate(&stream->ms);
 }
 
 static void choose_display_name(VideoStream *stream){
@@ -138,6 +138,7 @@ VideoStream *video_stream_new(int loc_rtp_port, int loc_rtcp_port, bool_t use_ip
 	VideoStream *stream = (VideoStream *)ms_new0 (VideoStream, 1);
 	stream->ms.type = VideoStreamType;
 	stream->ms.session=create_duplex_rtpsession(loc_rtp_port,loc_rtcp_port,use_ipv6);
+	stream->ms.qi=ms_quality_indicator_new(stream->ms.session);
 	stream->ms.evq=ortp_ev_queue_new();
 	stream->ms.rtpsend=ms_filter_new(MS_RTP_SEND_ID);
 	stream->ms.ice_check_list=NULL;
@@ -316,7 +317,7 @@ int video_stream_start (VideoStream *stream, RtpProfile *profile, const char *re
 	rtp_session_set_jitter_compensation(rtps,jitt_comp);
 
 	rtp_session_signal_connect(stream->ms.session,"payload_type_changed",
-			(RtpCallback)payload_type_changed,(unsigned long)&stream->ms);
+			(RtpCallback)mediastream_payload_type_changed,(unsigned long)&stream->ms);
 
 	rtp_session_get_jitter_buffer_params(stream->ms.session,&jbp);
 	jbp.max_packets=1000;//needed for high resolution video
@@ -454,6 +455,9 @@ int video_stream_start (VideoStream *stream, RtpProfile *profile, const char *re
 
 	/* create the ticker */
 	if (stream->ms.ticker==NULL) start_ticker(&stream->ms);
+	
+	stream->ms.start_time=ms_time(NULL);
+	stream->ms.is_beginning=TRUE;
 
 	/* attach the graphs */
 	if (stream->source)

tools/mediastream.c

@@ -766,7 +766,7 @@ void setup_media_streams(MediastreamDatas* args) {
 }
 
 
-static void mediastream_iterate(MediastreamDatas* args) {
+static void mediastream_tool_iterate(MediastreamDatas* args) {
 #ifndef WIN32
 	struct pollfd pfd;
 	int err;
@@ -844,7 +844,7 @@ void mediastream_run_loop(MediastreamDatas* args) {
 	{
 		int n;
 		for(n=0;n<100;++n){
-			mediastream_iterate(args);
+			mediastream_tool_iterate(args);
 #if defined(VIDEO_ENABLED)
 			if (args->video) video_stream_iterate(args->video);
 #endif