add tolerance to download interval in tests and fix clean up percentage computing

src/voip/qosanalyzer.c

@@ -158,7 +158,7 @@ static bool_t simple_analyser_process_rtcp(MSQosAnalyser *objbase, mblk_t *rtcp)
 		cur->rt_prop=rtp_session_get_round_trip_propagation(obj->session);
 
 		obj->points[obj->curindex-1][0]=rtp_session_get_send_bandwidth(obj->session)/1000.0;
-		obj->points[obj->curindex-1][1]=cur->lost_percentage;
+		obj->points[obj->curindex-1][1]=cur->lost_percentage/100.0;
 		obj->points[obj->curindex-1][2]=cur->rt_prop;
 		ms_message("MSQosAnalyser: lost_percentage=%f, int_jitter=%f ms, rt_prop=%f sec, send_bw=%f",cur->lost_percentage,cur->int_jitter,cur->rt_prop,obj->points[obj->curindex][0]);
 
@@ -231,19 +231,20 @@ static void compute_available_bw(MSSimpleQosAnalyser *obj){
 	/*to compute estimated BW, we need a minimum x-axis interval size*/
 	if (diff > 0.05) {
 		double avail_bw = (fabs(m) > 0.0001f) ? -b/m : mean_bw;
-		printf("\tfor line is %f kbit/s:\ty=%f x + %f\n", avail_bw, m, b);
+		printf("\tEstimated BW by interpolation is %f kbit/s:\ty=%f x + %f\n", avail_bw, m, b);
 
-		lossy_network |= (m < .03f && b > 0.05);
+		lossy_network |= (m < .005f && b > 0.05);
 
-		if (!lossy_network && (m>.03f||b>0.05)){
+		if (!lossy_network && (m>.005f||b>0.05)){
 			obj->network_state = MS_QOS_ANALYSER_NETWORK_CONGESTED;
 		}
-	} else {
-		lossy_network |= (y_mean > 0.05);
+
 	}
 
+	lossy_network |= (y_mean > 0.05);
+
 	if (lossy_network) {
-		/*since congestion may loss a high count of packets, stay in congested network while
+		/*since congestion may loss a high number of packets, stay in congested network while
 		this is not a bit more stable*/
 		if (previous_state == MS_QOS_ANALYSER_NETWORK_CONGESTED) {
 			obj->network_state = MS_QOS_ANALYSER_NETWORK_CONGESTED;

tester/mediastreamer2_audio_stream_tester.c

@@ -254,7 +254,7 @@ static void adaptive_opus_audio_stream()  {
 		bw_usage = adaptive_audio_stream(OPUS_PAYLOAD_TYPE, 8000, EDGE_BW, 0, 14);
 		CU_ASSERT_IN_RANGE(bw_usage, 2.f, 3.f); // bad! since this codec cant change its ptime and it is the lower bitrate, no improvement can occur
 		bw_usage = adaptive_audio_stream(OPUS_PAYLOAD_TYPE, 48000, EDGE_BW, 0, 11);
-		CU_ASSERT_IN_RANGE(bw_usage, 1.f, 1.2f); // bad!
+		CU_ASSERT_IN_RANGE(bw_usage, 1.f, 1.4f); // bad!
 
 		// on 3G BW, both should be at max
 		bw_usage = adaptive_audio_stream(OPUS_PAYLOAD_TYPE, 8000, THIRDGENERATION_BW, 0, 5);
@@ -269,8 +269,9 @@ static void adaptive_speex16_audio_stream()  {
 	if( supported ) {
 		// at 16KHz -> 20 kb/s
 		// at 32KHz -> 30 kb/s
-
-		adaptive_audio_stream(SPEEX16_PAYLOAD_TYPE, 32000, EDGE_BW / 2., 0, 120);
+		float bw_usage;
+		bw_usage = adaptive_audio_stream(SPEEX16_PAYLOAD_TYPE, 32000, EDGE_BW / 2., 0, 10);
+		CU_ASSERT_IN_RANGE(bw_usage, 1.f, 5.f);
 	}
 }
 
@@ -288,24 +289,6 @@ static void adaptive_pcma_audio_stream() {
 	}
 }
 
-static void lossy_network_speex_audio_stream() {
-	bool_t supported = ms_filter_codec_supported("speex");
-	if( supported ) {
-		// at 16KHz -> 20 kb/s
-		// at 32KHz -> 30 kb/s
-		float bw_usage;
-		int loss_rate = getenv("GPP_LOSS") ? atoi(getenv("GPP_LOSS")) : 0;
-		int max_bw = getenv("GPP_MAXBW") ? atoi(getenv("GPP_MAXBW")) * 1000: 0;
-		printf("\nloss_rate=%d(GPP_LOSS) max_bw=%d(GPP_MAXBW)\n", loss_rate, max_bw);
-		bw_usage = adaptive_audio_stream(SPEEX16_PAYLOAD_TYPE, 32000, max_bw, loss_rate, 20);
-		CU_ASSERT_IN_RANGE(bw_usage, .9f, 1.f);
-		// bw_usage = adaptive_audio_stream(SPEEX16_PAYLOAD_TYPE, 16000, EDGE_BW, 8);
-		// CU_ASSERT_IN_RANGE(bw_usage, .9f, 1.f);
-		// bw_usage = adaptive_audio_stream(SPEEX16_PAYLOAD_TYPE, 32000, THIRDGENERATION_BW, 5);
-		// CU_ASSERT_IN_RANGE(bw_usage, .1f, .2f);
-	}
-}
-
 #if 0
 static void audio_stream_dtmf(int codec_payload, int initial_bitrate,int target_bw, int max_recv_rtcp_packet) {
 	audio_stream_manager_t * marielle = audio_stream_manager_new();
@@ -352,7 +335,6 @@ static test_t tests[] = {
 	{ "Adaptive audio stream [opus]", adaptive_opus_audio_stream },
 	{ "Adaptive audio stream [speex]", adaptive_speex16_audio_stream },
 	{ "Adaptive audio stream [pcma]", adaptive_pcma_audio_stream },
-	{ "Lossy network", lossy_network_speex_audio_stream },
 };
 
 test_suite_t audio_stream_test_suite = {

tester/mediastreamer2_video_stream_tester.c

@@ -133,7 +133,7 @@ static void handle_queue_events(video_stream_manager_t * stream_mgr, OrtpEvQueue
 				stream_mgr->latest_stats.rtt=rtp_session_get_round_trip_propagation(stream_mgr->stream->ms.sessions.rtp_session);
 				if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED)
 					stream_mgr->latest_stats.network_state=ms_qos_analyser_get_network_state(ms_bitrate_controller_get_qos_analyser(stream_mgr->stream->ms.rc));
-				ms_message("mediastreamer2_video_stream_tester: %s RTCP packet: loss=%f, RTT=%f, network_state=%d\n",
+				ms_message("mediastreamer2_video_stream_tester: %s RTCP packet: loss=%f, RTT=%f, network_state=%d",
 					(evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) ? "RECEIVED" : "EMITTED",
 					stream_mgr->latest_stats.loss,
 					stream_mgr->latest_stats.rtt,