Merge branch 'master' of git.savannah.nongnu.org:/srv/git/linphone

coreapi/linphonecall.c

@@ -669,6 +669,7 @@ void linphone_call_init_media_streams(LinphoneCall *call){
 		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);
 
@@ -853,6 +854,7 @@ void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_mut
 	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)
 	{
@@ -876,7 +878,7 @@ void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_mut
 			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) {
@@ -907,11 +909,17 @@ void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_mut
 					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,

coreapi/linphonecore.c

@@ -782,6 +782,26 @@ static void autoreplier_config_init(LinphoneCore *lc)
 }
 */
 
+/**
+ * Enable adaptive rate control (experimental feature, audio-only).
+ *
+ * Adaptive rate control consists in using RTCP feedback provided information to dynamically
+ * control the output bitrate of the encoders, so that we can adapt to the network conditions and
+ * available bandwidth.
+**/
+void linphone_core_enable_adaptive_rate_control(LinphoneCore *lc, bool_t enabled){
+	lp_config_set_int(lc->config,"net","adaptive_rate_control",(int)enabled);
+}
+
+/**
+ * Returns whether adaptive rate control is enabled.
+ *
+ * See linphone_core_enable_adaptive_rate_control().
+**/
+bool_t linphone_core_adaptive_rate_control_enabled(const LinphoneCore *lc){
+	return lp_config_get_int(lc->config,"net","adaptive_rate_control",FALSE);
+}
+
 /**
  * Sets maximum available download bandwidth
  *

coreapi/linphonecore.h

@@ -705,14 +705,17 @@ void linphone_core_set_upload_bandwidth(LinphoneCore *lc, int bw);
 
 int linphone_core_get_download_bandwidth(const LinphoneCore *lc);
 int linphone_core_get_upload_bandwidth(const LinphoneCore *lc);
+
+void linphone_core_enable_adaptive_rate_control(LinphoneCore *lc, bool_t enabled);
+bool_t linphone_core_adaptive_rate_control_enabled(const LinphoneCore *lc);
 /**
- * set audio packetization time linphone expect to received from peer
+ * set audio packetization time linphone expect to receive from peer
  * @ingroup media_parameters
  *
  */
 void linphone_core_set_download_ptime(LinphoneCore *lc, int ptime);
 /**
- * get audio packetization time linphone expect to received from peer, 0 means unspecified
+ * get audio packetization time linphone expect to receive from peer, 0 means unspecified
  * @ingroup media_parameters
  */
 int  linphone_core_get_download_ptime(LinphoneCore *lc);

coreapi/linphonecore_jni.cc

@@ -1023,6 +1023,18 @@ extern "C" jlong Java_org_linphone_core_LinphoneCallImpl_getReplacedCall(	JNIEnv
 	return (jlong)linphone_call_get_replaced_call((LinphoneCall*)ptr);
 }
 
+extern "C" jfloat Java_org_linphone_core_LinphoneCallImpl_getCurrentQuality(	JNIEnv*  env
+																		,jobject  thiz
+																		,jlong ptr) {
+	return (jfloat)linphone_call_get_current_quality((LinphoneCall*)ptr);
+}
+
+extern "C" jfloat Java_org_linphone_core_LinphoneCallImpl_getAverageQuality(	JNIEnv*  env
+																		,jobject  thiz
+																		,jlong ptr) {
+	return (jfloat)linphone_call_get_average_quality((LinphoneCall*)ptr);
+}
+
 
 //LinphoneFriend
 extern "C" long Java_org_linphone_core_LinphoneFriendImpl_newLinphoneFriend(JNIEnv*  env

java/common/org/linphone/core/LinphoneCall.java

@@ -190,4 +190,28 @@ public interface LinphoneCall {
 	 * @return call duration computed from media start
 	 */
 	int getDuration();
+	/**
+	 * 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 getCurrentQuality();
+	/**
+	 * Returns call quality averaged over all the duration of the call.
+	 *
+	 * See getCurrentQuality() for more details about quality measurement.
+	 */
+	float getAverageQuality();
 }

mediastreamer2

@@ -1 +1 @@
-Subproject commit 79cbc5277d3fae8a70ae42a4e958042484ca2725
+Subproject commit fd74240c297e33c7e96d14dc5f7ed5d3dbdb8dc0