am afa583e6: SipAudioCall: expose startAudio()
Merge commit 'afa583e6557557577188c3e40146ac8d6f2aa7c7' into gingerbread-plus-aosp * commit 'afa583e6557557577188c3e40146ac8d6f2aa7c7': SipAudioCall: expose startAudio()
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@@ -626,6 +626,7 @@ public class SipPhone extends SipPhoneBase {
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if (newState == Call.State.INCOMING) {
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setState(mOwner.getState()); // INCOMING or WAITING
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} else {
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if (newState == Call.State.ACTIVE) call.startAudio();
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setState(newState);
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}
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mOwner.onConnectionStateChanged(SipConnection.this);
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@@ -153,7 +153,7 @@ public interface SipAudioCall {
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void setListener(Listener listener, boolean callbackImmediately);
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/**
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* Closes this object. The object is not usable after being closed.
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* Closes this object. This object is not usable after being closed.
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*/
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void close();
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@@ -171,6 +171,12 @@ public interface SipAudioCall {
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void makeCall(SipProfile callee, SipManager sipManager, int timeout)
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throws SipException;
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/**
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* Starts the audio for the established call. This method should be called
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* after {@link Listener#onCallEstablished} is called.
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*/
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void startAudio();
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/**
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* Attaches an incoming call to this call object.
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*
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@@ -37,6 +37,7 @@ import android.util.Log;
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import java.io.IOException;
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import java.net.InetAddress;
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import java.net.UnknownHostException;
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import java.text.ParseException;
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import java.util.ArrayList;
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import java.util.HashMap;
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@@ -239,24 +240,18 @@ public class SipAudioCallImpl extends SipSessionAdapter
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}
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}
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private synchronized void establishCall(String sessionDescription) {
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stopRingbackTone();
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stopRinging();
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try {
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SdpSessionDescription sd =
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new SdpSessionDescription(sessionDescription);
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Log.d(TAG, "sip call established: " + sd);
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startCall(sd);
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mInCall = true;
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} catch (SdpException e) {
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Log.e(TAG, "createSessionDescription()", e);
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}
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}
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@Override
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public void onCallEstablished(ISipSession session,
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String sessionDescription) {
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establishCall(sessionDescription);
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stopRingbackTone();
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stopRinging();
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try {
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mPeerSd = new SdpSessionDescription(sessionDescription);
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Log.d(TAG, "sip call established: " + mPeerSd);
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} catch (SdpException e) {
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Log.e(TAG, "createSessionDescription()", e);
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}
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Listener listener = mListener;
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if (listener != null) {
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try {
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@@ -609,10 +604,23 @@ public class SipAudioCallImpl extends SipSessionAdapter
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return copies;
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}
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private void startCall(SdpSessionDescription peerSd) {
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public void startAudio() {
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try {
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startAudioInternal();
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} catch (UnknownHostException e) {
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onError(mSipSession, SipErrorCode.PEER_NOT_REACHABLE.toString(),
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e.getMessage());
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} catch (Throwable e) {
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onError(mSipSession, SipErrorCode.CLIENT_ERROR.toString(),
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e.getMessage());
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}
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}
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private synchronized void startAudioInternal() throws UnknownHostException {
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stopCall(DONT_RELEASE_SOCKET);
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mInCall = true;
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SdpSessionDescription peerSd = mPeerSd;
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if (isWifiOn()) grabWifiHighPerfLock();
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mPeerSd = peerSd;
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String peerMediaAddress = peerSd.getPeerMediaAddress(AUDIO);
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// TODO: handle multiple media fields
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int peerMediaPort = peerSd.getPeerMediaPort(AUDIO);
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@@ -621,58 +629,55 @@ public class SipAudioCallImpl extends SipSessionAdapter
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int localPort = getLocalMediaPort();
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int sampleRate = 8000;
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int frameSize = sampleRate / 50; // 160
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try {
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// TODO: get sample rate from sdp
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mCodec = getCodec(peerSd);
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AudioStream audioStream = mAudioStream;
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audioStream.associate(InetAddress.getByName(peerMediaAddress),
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peerMediaPort);
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audioStream.setCodec(convert(mCodec), mCodec.payloadType);
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audioStream.setDtmfType(DTMF);
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Log.d(TAG, "start media: localPort=" + localPort + ", peer="
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+ peerMediaAddress + ":" + peerMediaPort);
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// TODO: get sample rate from sdp
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mCodec = getCodec(peerSd);
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audioStream.setMode(RtpStream.MODE_NORMAL);
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if (!mHold) {
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// FIXME: won't work if peer is not sending nor receiving
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if (!peerSd.isSending(AUDIO)) {
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Log.d(TAG, " not receiving");
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audioStream.setMode(RtpStream.MODE_SEND_ONLY);
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}
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if (!peerSd.isReceiving(AUDIO)) {
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Log.d(TAG, " not sending");
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audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
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}
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AudioStream audioStream = mAudioStream;
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audioStream.associate(InetAddress.getByName(peerMediaAddress),
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peerMediaPort);
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audioStream.setCodec(convert(mCodec), mCodec.payloadType);
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audioStream.setDtmfType(DTMF);
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Log.d(TAG, "start media: localPort=" + localPort + ", peer="
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+ peerMediaAddress + ":" + peerMediaPort);
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/* The recorder volume will be very low if the device is in
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* IN_CALL mode. Therefore, we have to set the mode to NORMAL
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* in order to have the normal microphone level.
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*/
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((AudioManager) mContext.getSystemService
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(Context.AUDIO_SERVICE))
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.setMode(AudioManager.MODE_NORMAL);
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audioStream.setMode(RtpStream.MODE_NORMAL);
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if (!mHold) {
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// FIXME: won't work if peer is not sending nor receiving
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if (!peerSd.isSending(AUDIO)) {
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Log.d(TAG, " not receiving");
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audioStream.setMode(RtpStream.MODE_SEND_ONLY);
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}
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if (!peerSd.isReceiving(AUDIO)) {
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Log.d(TAG, " not sending");
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audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
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}
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// AudioGroup logic:
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AudioGroup audioGroup = getAudioGroup();
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if (mHold) {
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if (audioGroup != null) {
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audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
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}
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// don't create an AudioGroup here; doing so will fail if
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// there's another AudioGroup out there that's active
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/* The recorder volume will be very low if the device is in
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* IN_CALL mode. Therefore, we have to set the mode to NORMAL
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* in order to have the normal microphone level.
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*/
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((AudioManager) mContext.getSystemService
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(Context.AUDIO_SERVICE))
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.setMode(AudioManager.MODE_NORMAL);
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}
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// AudioGroup logic:
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AudioGroup audioGroup = getAudioGroup();
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if (mHold) {
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if (audioGroup != null) {
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audioGroup.setMode(AudioGroup.MODE_ON_HOLD);
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}
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// don't create an AudioGroup here; doing so will fail if
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// there's another AudioGroup out there that's active
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} else {
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if (audioGroup == null) audioGroup = new AudioGroup();
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audioStream.join(audioGroup);
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if (mMuted) {
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audioGroup.setMode(AudioGroup.MODE_MUTED);
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} else {
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if (audioGroup == null) audioGroup = new AudioGroup();
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audioStream.join(audioGroup);
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if (mMuted) {
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audioGroup.setMode(AudioGroup.MODE_MUTED);
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} else {
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audioGroup.setMode(AudioGroup.MODE_NORMAL);
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}
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audioGroup.setMode(AudioGroup.MODE_NORMAL);
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}
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} catch (Exception e) {
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Log.e(TAG, "call()", e);
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}
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}
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