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Add confcall management to SIP calls

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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
This commit is contained in:
Hung-ying Tyan
2010-08-18 14:37:59 +08:00
parent a83987d2cc
commit 3294d44b96
6 changed files with 142 additions and 94 deletions
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4
telephony/java/com/android/internal/telephony/CallManager.java
View File

@ -322,7 +322,9 @@ public final class CallManager {
}
break;
}
audioManager.setMode(mode);
// calling audioManager.setMode() multiple times in a short period of
// time seems to break the audio recorder in in-call mode
if (audioManager.getMode() != mode) audioManager.setMode(mode);
}
private Context getContext() {

6
telephony/java/com/android/internal/telephony/sip/SipConnectionBase.java
View File

@ -85,8 +85,10 @@ abstract class SipConnectionBase extends Connection {
protected void setState(Call.State state) {
switch (state) {
case ACTIVE:
connectTimeReal = SystemClock.elapsedRealtime();
connectTime = System.currentTimeMillis();
if (connectTime == 0) {
connectTimeReal = SystemClock.elapsedRealtime();
connectTime = System.currentTimeMillis();
}
break;
case DISCONNECTED:
duration = SystemClock.elapsedRealtime() - connectTimeReal;

159
telephony/java/com/android/internal/telephony/sip/SipPhone.java
View File

@ -100,7 +100,7 @@ public class SipPhone extends SipPhoneBase {
}
public String getPhoneName() {
return mProfile.getProfileName();
return "SIP:" + getUriString(mProfile);
}
public String getSipUri() {
@ -222,21 +222,25 @@ public class SipPhone extends SipPhoneBase {
}
public void conference() throws CallStateException {
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());
synchronized (SipPhone.class) {
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);
}
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());
synchronized (SipPhone.class) {
if (!(that instanceof SipCall)) {
throw new CallStateException("expect " + SipCall.class
+ ", cannot merge with " + that.getClass());
}
foregroundCall.merge((SipCall) that);
}
foregroundCall.merge((SipCall) that);
}
public boolean canTransfer() {
@ -248,12 +252,14 @@ public class SipPhone extends SipPhoneBase {
}
public void clearDisconnected() {
ringingCall.clearDisconnected();
foregroundCall.clearDisconnected();
backgroundCall.clearDisconnected();
synchronized (SipPhone.class) {
ringingCall.clearDisconnected();
foregroundCall.clearDisconnected();
backgroundCall.clearDisconnected();
updatePhoneState();
notifyPreciseCallStateChanged();
updatePhoneState();
notifyPreciseCallStateChanged();
}
}
public void sendDtmf(char c) {
@ -261,7 +267,9 @@ public class SipPhone extends SipPhoneBase {
Log.e(LOG_TAG,
"sendDtmf called with invalid character '" + c + "'");
} else if (foregroundCall.getState().isAlive()) {
foregroundCall.sendDtmf(c);
synchronized (SipPhone.class) {
foregroundCall.sendDtmf(c);
}
}
}
@ -307,7 +315,9 @@ public class SipPhone extends SipPhoneBase {
}
public void setMute(boolean muted) {
foregroundCall.setMute(muted);
synchronized (SipPhone.class) {
foregroundCall.setMute(muted);
}
}
public boolean getMute() {
@ -410,18 +420,20 @@ public class SipPhone extends SipPhoneBase {
@Override
public void hangup() throws CallStateException {
Log.v(LOG_TAG, "hang up call: " + getState() + ": " + this
+ " on phone " + getPhone());
CallStateException excp = null;
for (Connection c : connections) {
try {
c.hangup();
} catch (CallStateException e) {
excp = e;
synchronized (SipPhone.class) {
Log.v(LOG_TAG, "hang up call: " + getState() + ": " + this
+ " on phone " + getPhone());
CallStateException excp = null;
for (Connection c : connections) {
try {
c.hangup();
} catch (CallStateException e) {
excp = e;
}
}
if (excp != null) throw excp;
setState(State.DISCONNECTING);
}
if (excp != null) throw excp;
setState(State.DISCONNECTING);
}
void initIncomingCall(SipAudioCall sipAudioCall, boolean makeCallWait) {
@ -454,19 +466,20 @@ public class SipPhone extends SipPhoneBase {
}
void hold() throws CallStateException {
AudioGroup audioGroup = getAudioGroup();
if (audioGroup == null) return;
audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
setState(State.HOLDING);
AudioGroup audioGroup = getAudioGroup();
if (audioGroup != null) {
audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
}
for (Connection c : connections) ((SipConnection) c).hold();
}
void unhold() throws CallStateException {
AudioGroup audioGroup = getAudioGroup();
if (audioGroup == null) return;
audioGroup.setMode(AudioGroup.MODE_NORMAL);
setState(State.ACTIVE);
for (Connection c : connections) ((SipConnection) c).unhold();
AudioGroup audioGroup = new AudioGroup();
for (Connection c : connections) {
((SipConnection) c).unhold(audioGroup);
}
}
void setMute(boolean muted) {
@ -483,17 +496,26 @@ public class SipPhone extends SipPhoneBase {
}
void merge(SipCall that) throws CallStateException {
AudioGroup myGroup = getAudioGroup();
AudioGroup audioGroup = getAudioGroup();
for (Connection c : that.connections) {
SipConnection conn = (SipConnection) c;
conn.mergeTo(myGroup);
connections.add(conn);
conn.changeOwner(this);
add(conn);
if (conn.getState() == Call.State.HOLDING) {
conn.unhold(audioGroup);
}
}
that.connections.clear();
that.setState(Call.State.IDLE);
}
private void add(SipConnection conn) {
SipCall call = conn.getCall();
if (call == this) return;
if (call != null) call.connections.remove(conn);
connections.add(conn);
conn.changeOwner(this);
}
void sendDtmf(char c) {
AudioGroup audioGroup = getAudioGroup();
if (audioGroup == null) return;
@ -568,7 +590,6 @@ public class SipPhone extends SipPhoneBase {
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;
@ -673,6 +694,7 @@ public class SipPhone extends SipPhoneBase {
}
void hold() throws CallStateException {
setState(Call.State.HOLDING);
try {
mSipAudioCall.holdCall();
} catch (SipException e) {
@ -680,7 +702,9 @@ public class SipPhone extends SipPhoneBase {
}
}
void unhold() throws CallStateException {
void unhold(AudioGroup audioGroup) throws CallStateException {
mSipAudioCall.setAudioGroup(audioGroup);
setState(Call.State.ACTIVE);
try {
mSipAudioCall.continueCall();
} catch (SipException e) {
@ -688,16 +712,6 @@ public class SipPhone extends SipPhoneBase {
}
}
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;
@ -732,29 +746,36 @@ public class SipPhone extends SipPhoneBase {
@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 {
mSipAudioCall.endCall();
setState(Call.State.DISCONNECTING);
setDisconnectCause(DisconnectCause.LOCAL);
} catch (SipException e) {
throw new CallStateException("hangup(): " + e);
synchronized (SipPhone.class) {
Log.v(LOG_TAG, "hangup conn: " + mPeer.getUriString() + ": "
+ ": on phone " + getPhone().getPhoneName());
try {
mSipAudioCall.endCall();
setState(Call.State.DISCONNECTING);
setDisconnectCause(DisconnectCause.LOCAL);
} catch (SipException e) {
throw new CallStateException("hangup(): " + e);
}
}
}
@Override
public void separate() throws CallStateException {
// TODO: what's this for SIP?
/*
if (!disconnected) {
owner.separate(this);
} else {
throw new CallStateException ("disconnected");
synchronized (SipPhone.class) {
SipCall call = (SipCall) SipPhone.this.getBackgroundCall();
if (call.getState() != Call.State.IDLE) {
throw new CallStateException(
"cannot put conn back to a call in non-idle state: "
+ call.getState());
}
Log.v(LOG_TAG, "separate conn: " + mPeer.getUriString()
+ " from " + mOwner + " back to " + call);
AudioGroup audioGroup = call.getAudioGroup(); // may be null
call.add(this);
mSipAudioCall.setAudioGroup(audioGroup);
call.hold();
}
*/
}
@Override

4
telephony/java/com/android/internal/telephony/sip/SipPhoneBase.java
View File

@ -175,10 +175,6 @@ abstract class SipPhoneBase extends PhoneBase {
return state;
}
public String getPhoneName() {
return "SIP";
}
public int getPhoneType() {
// FIXME: add SIP phone type
return Phone.PHONE_TYPE_GSM;

12
voip/java/android/net/sip/SipAudioCall.java
View File

@ -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
@ -252,6 +253,15 @@ 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.
*

51
voip/java/android/net/sip/SipAudioCallImpl.java
View File

@ -70,7 +70,8 @@ public class SipAudioCallImpl extends SipSessionAdapter
private ISipSession mSipSession;
private SdpSessionDescription mPeerSd;
private AudioStream mRtpSession;
private AudioStream mAudioStream;
private AudioGroup mAudioGroup;
private SdpSessionDescription.AudioCodec mCodec;
private long mSessionId = -1L; // SDP session ID
private boolean mInCall = false;
@ -505,11 +506,19 @@ public class SipAudioCallImpl extends SipSessionAdapter
}
public synchronized AudioStream getAudioStream() {
return mRtpSession;
return mAudioStream;
}
public synchronized AudioGroup getAudioGroup() {
return ((mRtpSession == null) ? null : mRtpSession.getAudioGroup());
if (mAudioGroup != null) return mAudioGroup;
return ((mAudioStream == null) ? null : mAudioStream.getAudioGroup());
}
public synchronized void setAudioGroup(AudioGroup group) {
if ((mAudioStream != null) && (mAudioStream.getAudioGroup() != null)) {
mAudioStream.join(group);
}
mAudioGroup = group;
}
private SdpSessionDescription.AudioCodec getCodec(SdpSessionDescription sd) {
@ -561,7 +570,7 @@ public class SipAudioCallImpl extends SipSessionAdapter
// TODO: get sample rate from sdp
mCodec = getCodec(peerSd);
AudioStream audioStream = mRtpSession;
AudioStream audioStream = mAudioStream;
audioStream.associate(InetAddress.getByName(peerMediaAddress),
peerMediaPort);
audioStream.setCodec(convert(mCodec), mCodec.payloadType);
@ -580,7 +589,7 @@ public class SipAudioCallImpl extends SipSessionAdapter
Log.d(TAG, " not sending");
audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
}
} else {
/* The recorder volume will be very low if the device is in
* IN_CALL mode. Therefore, we have to set the mode to NORMAL
* in order to have the normal microphone level.
@ -590,14 +599,22 @@ public class SipAudioCallImpl extends SipSessionAdapter
.setMode(AudioManager.MODE_NORMAL);
}
AudioGroup audioGroup = new AudioGroup();
audioStream.join(audioGroup);
// AudioGroup logic:
AudioGroup audioGroup = getAudioGroup();
if (mHold) {
audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
} else if (mMuted) {
audioGroup.setMode(AudioGroup.MODE_MUTED);
if (audioGroup != null) {
audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
}
// don't create an AudioGroup here; doing so will fail if
// there's another AudioGroup out there that's active
} else {
audioGroup.setMode(AudioGroup.MODE_NORMAL);
if (audioGroup == null) audioGroup = new AudioGroup();
audioStream.join(audioGroup);
if (mMuted) {
audioGroup.setMode(AudioGroup.MODE_MUTED);
} else {
audioGroup.setMode(AudioGroup.MODE_NORMAL);
}
}
} catch (Exception e) {
Log.e(TAG, "call()", e);
@ -606,20 +623,20 @@ public class SipAudioCallImpl extends SipSessionAdapter
private void stopCall(boolean releaseSocket) {
Log.d(TAG, "stop audiocall");
if (mRtpSession != null) {
mRtpSession.join(null);
if (mAudioStream != null) {
mAudioStream.join(null);
if (releaseSocket) {
mRtpSession.release();
mRtpSession = null;
mAudioStream.release();
mAudioStream = null;
}
}
}
private int getLocalMediaPort() {
if (mRtpSession != null) return mRtpSession.getLocalPort();
if (mAudioStream != null) return mAudioStream.getLocalPort();
try {
AudioStream s = mRtpSession =
AudioStream s = mAudioStream =
new AudioStream(InetAddress.getByName(getLocalIp()));
return s.getLocalPort();
} catch (IOException e) {