Add confcall management to SIP calls

and fix the bug of re-assigning connectTime's in SipConnection,
and adding synchronization for SipPhone to be thread-safe,
and set normal audio mode when call not on hold instead of on hold in SipAudioCallImpl,
and fix re-entrance problem in CallManager.setAudioMode() for in-call mode.

Change-Id: I54f39dab052062de1ce141e5358d892d30453a3a

1 files changed, 11 insertions, 1 deletions

diff --git a/voip/java/android/net/sip/SipAudioCall.java b/voip/java/android/net/sip/SipAudioCall.java
index f4be839da72b..3cdd11489d7e 100644
--- a/voip/java/android/net/sip/SipAudioCall.java
+++ b/voip/java/android/net/sip/SipAudioCall.java
@@ -244,7 +244,8 @@ public interface SipAudioCall {
      * Also, the {@code AudioStream} may change its group during a call (e.g.,
      * after the call is held/un-held). Finally, the {@code AudioGroup} object
      * returned by this method is undefined after the call ends or the
-     * {@link #close} method is called.
+     * {@link #close} method is called. If a group object is set by
+     * {@link #setAudioGroup(AudioGroup)}, then this method returns that object.
      *
      * @return the {@link AudioGroup} object or null if the RTP stream has not
      *      yet been set up
@@ -253,6 +254,15 @@ public interface SipAudioCall {
     AudioGroup getAudioGroup();
 
     /**
+     * Sets the {@link AudioGroup} object which the {@link AudioStream} object
+     * joins. If {@code audioGroup} is null, then the {@code AudioGroup} object
+     * will be dynamically created when needed.
+     *
+     * @see #getAudioStream
+     */
+    void setAudioGroup(AudioGroup audioGroup);
+
+    /**
      * Checks if the call is established.
      *
      * @return true if the call is established