Adds a property to enable rtpTranslator for conferences.

Account property that enables rtpTranslator instead of mixing in conferences.

4 files changed, 70 insertions, 5 deletions

diff --git a/src/net/java/sip/communicator/service/protocol/Call.java b/src/net/java/sip/communicator/service/protocol/Call.java
index d342373..a952cfd 100644
--- a/src/net/java/sip/communicator/service/protocol/Call.java
+++ b/src/net/java/sip/communicator/service/protocol/Call.java
@@ -111,6 +111,13 @@ public abstract class Call
     private boolean isAutoAnswer = false;
 
     /**
+     * The indicator which determines whether any telephony conference
+     * represented by this instance is mixing or relaying.
+     * By default what can be mixed is mixed (audio) and rest is relayed.
+     */
+    protected final boolean useTranslator;
+
+    /**
      * Creates a new Call instance.
      *
      * @param sourceProvider the proto provider that created us.
@@ -133,6 +140,11 @@ public abstract class Call
             = accountID.getAccountPropertyBoolean(
                     ProtocolProviderFactory.DEFAULT_SIPZRTP_ATTRIBUTE,
                     true);
+
+        useTranslator
+            = accountID.getAccountPropertyBoolean(
+                    ProtocolProviderFactory.USE_TRANSLATOR_IN_CONFERENCE,
+                    false);
     }
 
     /**
diff --git a/src/net/java/sip/communicator/service/protocol/ProtocolProviderFactory.java b/src/net/java/sip/communicator/service/protocol/ProtocolProviderFactory.java
index 2efdc29..561a092 100644
--- a/src/net/java/sip/communicator/service/protocol/ProtocolProviderFactory.java
+++ b/src/net/java/sip/communicator/service/protocol/ProtocolProviderFactory.java
@@ -376,6 +376,15 @@ public abstract class ProtocolProviderFactory
         = "USE_JITSI_VIDEO_BRIDGE";
 
     /**
+     * The name of the boolean account property which indicates whether Jitsi
+     * will use translator for media, instead of mixing, for conference
+     * calls.
+     * By default if supported mixing is used (audio mixed, video relayed).
+     */
+    public static final String USE_TRANSLATOR_IN_CONFERENCE
+        = "USE_TRANSLATOR_IN_CONFERENCE";
+
+    /**
      * The property name prefix for all stun server properties. We generally use
      * this prefix in conjunction with an index which is how we store multiple
      * servers.
diff --git a/src/net/java/sip/communicator/service/protocol/media/MediaAwareCall.java b/src/net/java/sip/communicator/service/protocol/media/MediaAwareCall.java
index f9b6c2f..2bf802b 100644
--- a/src/net/java/sip/communicator/service/protocol/media/MediaAwareCall.java
+++ b/src/net/java/sip/communicator/service/protocol/media/MediaAwareCall.java
@@ -906,7 +906,7 @@ public abstract class MediaAwareCall<
     @Override
     protected CallConference createConference()
     {
-        return new MediaAwareCallConference();
+        return new MediaAwareCallConference(false, this.useTranslator);
     }
 
     /**
diff --git a/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java b/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java
index da34f2c..81e917c 100644
--- a/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java
+++ b/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java
@@ -93,6 +93,21 @@ public class MediaAwareCallConference
     private RTPTranslator videoRTPTranslator;
 
     /**
+     * The <tt>RTPTranslator</tt> which forwards autio RTP and RTCP traffic
+     * between the <tt>CallPeer</tt>s of the <tt>Call</tt>s participating in
+     * this telephony conference when the local peer is acting as a conference
+     * focus.
+     */
+    private RTPTranslator audioRTPTranslator;
+
+    /**
+     * The indicator which determines whether the telephony conference
+     * represented by this instance is mixing or relaying.
+     * By default what can be mixed is mixed (audio) and rest is relayed.
+     */
+    private boolean translator = false;
+
+    /**
      * Initializes a new <tt>MediaAwareCallConference</tt> instance.
      */
     public MediaAwareCallConference()
@@ -111,8 +126,25 @@ public class MediaAwareCallConference
      */
     public MediaAwareCallConference(boolean jitsiVideobridge)
     {
+        this(jitsiVideobridge, false);
+    }
+
+    /**
+     * Initializes a new <tt>MediaAwareCallConference</tt> instance which is to
+     * optionally utilize the Jitsi Videobridge server-side telephony
+     * conferencing technology.
+     *
+     * @param jitsiVideobridge <tt>true</tt> if the telephony conference
+     * represented by the new instance is to utilize the Jitsi Videobridge
+     * server-side telephony conferencing technology; otherwise, <tt>false</tt>
+     */
+    public MediaAwareCallConference(boolean jitsiVideobridge,
+                                    boolean translator)
+    {
         super(jitsiVideobridge);
 
+        this.translator = translator;
+
         int mediaTypeCount = MediaType.values().length;
 
         devices = new MediaDevice[mediaTypeCount];
@@ -269,6 +301,7 @@ public class MediaAwareCallConference
                      * necessary.
                      */
                     if ((!OSUtils.IS_ANDROID || isConferenceFocus())
+                            && !this.translator
                             /*
                              * We can use the AudioMixer only if the device is
                              * able to capture (because the AudioMixer will push
@@ -325,8 +358,6 @@ public class MediaAwareCallConference
      */
     public RTPTranslator getRTPTranslator(MediaType mediaType)
     {
-        RTPTranslator rtpTranslator = null;
-
         /*
          * XXX A mixer is created for audio even when the local peer is not a
          * conference focus in order to enable additional functionality.
@@ -348,9 +379,22 @@ public class MediaAwareCallConference
                         .getMediaService()
                             .createRTPTranslator();
             }
-            rtpTranslator = videoRTPTranslator;
+            return videoRTPTranslator;
         }
-        return rtpTranslator;
+
+        if (this.translator)
+        {
+            if(audioRTPTranslator == null)
+            {
+                audioRTPTranslator
+                    = ProtocolMediaActivator
+                    .getMediaService()
+                    .createRTPTranslator();
+            }
+            return audioRTPTranslator;
+        }
+
+        return null;
     }
 
     /**