SIP: implement conference call
Change-Id: Ifd420ed95e77e744c6aff28ac63e7363f97d9dc6
diff --git a/telephony/java/com/android/internal/telephony/CallManager.java b/telephony/java/com/android/internal/telephony/CallManager.java
index 784f022..511440e 100644
--- a/telephony/java/com/android/internal/telephony/CallManager.java
+++ b/telephony/java/com/android/internal/telephony/CallManager.java
@@ -16,6 +16,8 @@
package com.android.internal.telephony;
+import com.android.internal.telephony.sip.SipPhone;
+
import android.content.Context;
import android.os.AsyncResult;
import android.os.Handler;
@@ -453,7 +455,7 @@
heldPhone = heldCall.getPhone();
}
- return (heldPhone == activePhone);
+ return heldPhone.getClass().equals(activePhone.getClass());
}
/**
@@ -466,10 +468,14 @@
* In these cases, this operation may not be performed.
*/
public void conference(Call heldCall) throws CallStateException {
- if (canConference(heldCall))
+ Phone fgPhone = getFgPhone();
+ if (fgPhone instanceof SipPhone) {
+ ((SipPhone) fgPhone).conference(heldCall);
+ } else if (canConference(heldCall)) {
+ fgPhone.conference();
+ } else {
throw(new CallStateException("Can't conference foreground and selected background call"));
-
- heldCall.getPhone().conference();
+ }
}
/**
diff --git a/telephony/java/com/android/internal/telephony/sip/SipPhone.java b/telephony/java/com/android/internal/telephony/sip/SipPhone.java
index a94518f..a052db0 100755
--- a/telephony/java/com/android/internal/telephony/sip/SipPhone.java
+++ b/telephony/java/com/android/internal/telephony/sip/SipPhone.java
@@ -21,6 +21,7 @@
import android.content.SharedPreferences;
import android.net.Uri;
import android.net.rtp.AudioGroup;
+import android.net.rtp.AudioStream;
import android.net.sip.SipAudioCall;
import android.net.sip.SipManager;
import android.net.sip.SipProfile;
@@ -221,7 +222,21 @@
}
public void conference() throws CallStateException {
- // TODO
+ if ((foregroundCall.getState() != SipCall.State.ACTIVE)
+ || (foregroundCall.getState() != SipCall.State.ACTIVE)) {
+ throw new CallStateException("wrong state to merge calls: fg="
+ + foregroundCall.getState() + ", bg="
+ + backgroundCall.getState());
+ }
+ foregroundCall.merge(backgroundCall);
+ }
+
+ public void conference(Call that) throws CallStateException {
+ if (!(that instanceof SipCall)) {
+ throw new CallStateException("expect " + SipCall.class
+ + ", cannot merge with " + that.getClass());
+ }
+ foregroundCall.merge((SipCall) that);
}
public boolean canTransfer() {
@@ -467,6 +482,18 @@
return (audioGroup.getMode() == AudioGroup.MODE_MUTED);
}
+ void merge(SipCall that) throws CallStateException {
+ AudioGroup myGroup = getAudioGroup();
+ for (Connection c : that.connections) {
+ SipConnection conn = (SipConnection) c;
+ conn.mergeTo(myGroup);
+ connections.add(conn);
+ conn.changeOwner(this);
+ }
+ that.connections.clear();
+ that.setState(Call.State.IDLE);
+ }
+
void sendDtmf(char c) {
AudioGroup audioGroup = getAudioGroup();
if (audioGroup == null) return;
@@ -541,6 +568,7 @@
private class SipConnection extends SipConnectionBase {
private SipCall mOwner;
private SipAudioCall mSipAudioCall;
+ private AudioGroup mOriginalGroup;
private Call.State mState = Call.State.IDLE;
private SipProfile mPeer;
private boolean mIncoming = false;
@@ -660,6 +688,16 @@
}
}
+ void mergeTo(AudioGroup group) throws CallStateException {
+ AudioStream stream = mSipAudioCall.getAudioStream();
+ if (stream == null) {
+ throw new CallStateException("wrong state to merge: "
+ + mSipAudioCall.getState());
+ }
+ if (mOriginalGroup == null) mOriginalGroup = getAudioGroup();
+ stream.join(group);
+ }
+
@Override
protected void setState(Call.State state) {
if (state == mState) return;
@@ -694,6 +732,8 @@
@Override
public void hangup() throws CallStateException {
+ // TODO: need to pull AudioStream out of the AudioGroup in case
+ // this conn was part of a conf call
Log.v(LOG_TAG, "hangup conn: " + mPeer.getUriString() + ": "
+ ": on phone " + getPhone());
try {
