Merge "Move the sip related codes to framework." into gingerbread (8c503569) · Commits · e / os / android_frameworks_base

voip/java/android/net/rtp/AudioCodec.java

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		/*
		* Copyright (C) 2010 The Android Open Source Project
		*
		* Licensed under the Apache License, Version 2.0 (the "License");
		* you may not use this file except in compliance with the License.
		* You may obtain a copy of the License at
		*
		* http://www.apache.org/licenses/LICENSE-2.0
		*
		* Unless required by applicable law or agreed to in writing, software
		* distributed under the License is distributed on an "AS IS" BASIS,
		* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		* See the License for the specific language governing permissions and
		* limitations under the License.
		*/

		package android.net.rtp;

		/** @hide */
		public class AudioCodec {
		public static final AudioCodec ULAW = new AudioCodec("PCMU", 8000, 160, 0);
		public static final AudioCodec ALAW = new AudioCodec("PCMA", 8000, 160, 8);

		/**
		* Returns system supported codecs.
		*/
		public static AudioCodec[] getSystemSupportedCodecs() {
		return new AudioCodec[] {AudioCodec.ULAW, AudioCodec.ALAW};
		}

		/**
		* Returns the codec instance if it is supported by the system.
		*
		* @param name name of the codec
		* @return the matched codec or null if the codec name is not supported by
		* the system
		*/
		public static AudioCodec getSystemSupportedCodec(String name) {
		for (AudioCodec codec : getSystemSupportedCodecs()) {
		if (codec.name.equals(name)) return codec;
		}
		return null;
		}

		public final String name;
		public final int sampleRate;
		public final int sampleCount;
		public final int defaultType;

		private AudioCodec(String name, int sampleRate, int sampleCount, int defaultType) {
		this.name = name;
		this.sampleRate = sampleRate;
		this.sampleCount = sampleCount;
		this.defaultType = defaultType;
		}
		}

voip/java/android/net/rtp/AudioGroup.java

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		/*
		* Copyright (C) 2010 The Android Open Source Project
		*
		* Licensed under the Apache License, Version 2.0 (the "License");
		* you may not use this file except in compliance with the License.
		* You may obtain a copy of the License at
		*
		* http://www.apache.org/licenses/LICENSE-2.0
		*
		* Unless required by applicable law or agreed to in writing, software
		* distributed under the License is distributed on an "AS IS" BASIS,
		* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		* See the License for the specific language governing permissions and
		* limitations under the License.
		*/

		package android.net.rtp;

		import java.util.HashMap;
		import java.util.Map;

		/**
		*/
		/** @hide */
		public class AudioGroup {
		public static final int MODE_ON_HOLD = 0;
		public static final int MODE_MUTED = 1;
		public static final int MODE_NORMAL = 2;
		public static final int MODE_EC_ENABLED = 3;

		private final Map<AudioStream, Integer> mStreams;
		private int mMode = MODE_ON_HOLD;

		private int mNative;
		static {
		System.loadLibrary("rtp_jni");
		}

		public AudioGroup() {
		mStreams = new HashMap<AudioStream, Integer>();
		}

		public int getMode() {
		return mMode;
		}

		public synchronized native void setMode(int mode);

		synchronized void add(AudioStream stream, AudioCodec codec, int codecType, int dtmfType) {
		if (!mStreams.containsKey(stream)) {
		try {
		int id = add(stream.getMode(), stream.dup(),
		stream.getRemoteAddress().getHostAddress(), stream.getRemotePort(),
		codec.name, codec.sampleRate, codec.sampleCount, codecType, dtmfType);
		mStreams.put(stream, id);
		} catch (NullPointerException e) {
		throw new IllegalStateException(e);
		}
		}
		}

		private native int add(int mode, int socket, String remoteAddress, int remotePort,
		String codecName, int sampleRate, int sampleCount, int codecType, int dtmfType);

		synchronized void remove(AudioStream stream) {
		Integer id = mStreams.remove(stream);
		if (id != null) {
		remove(id);
		}
		}

		private native void remove(int id);

		/**
		* Sends a DTMF digit to every {@link AudioStream} in this group. Currently
		* only event {@code 0} to {@code 15} are supported.
		*
		* @throws IllegalArgumentException if the event is invalid.
		*/
		public native synchronized void sendDtmf(int event);

		public synchronized void reset() {
		remove(-1);
		}

		@Override
		protected void finalize() throws Throwable {
		reset();
		super.finalize();
		}
		}

voip/java/android/net/rtp/AudioStream.java

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		/*
		* Copyright (C) 2010 The Android Open Source Project
		*
		* Licensed under the Apache License, Version 2.0 (the "License");
		* you may not use this file except in compliance with the License.
		* You may obtain a copy of the License at
		*
		* http://www.apache.org/licenses/LICENSE-2.0
		*
		* Unless required by applicable law or agreed to in writing, software
		* distributed under the License is distributed on an "AS IS" BASIS,
		* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		* See the License for the specific language governing permissions and
		* limitations under the License.
		*/

		package android.net.rtp;

		import java.net.InetAddress;
		import java.net.SocketException;

		/**
		* AudioStream represents a RTP stream carrying audio payloads.
		*/
		/** @hide */
		public class AudioStream extends RtpStream {
		private AudioCodec mCodec;
		private int mCodecType = -1;
		private int mDtmfType = -1;
		private AudioGroup mGroup;

		/**
		* Creates an AudioStream on the given local address. Note that the local
		* port is assigned automatically to conform with RFC 3550.
		*
		* @param address The network address of the local host to bind to.
		* @throws SocketException if the address cannot be bound or a problem
		* occurs during binding.
		*/
		public AudioStream(InetAddress address) throws SocketException {
		super(address);
		}

		/**
		* Returns {@code true} if the stream already joined an {@link AudioGroup}.
		*/
		@Override
		public final boolean isBusy() {
		return mGroup != null;
		}

		/**
		* Returns the joined {@link AudioGroup}.
		*/
		public AudioGroup getAudioGroup() {
		return mGroup;
		}

		/**
		* Joins an {@link AudioGroup}. Each stream can join only one group at a
		* time. The group can be changed by passing a different one or removed
		* by calling this method with {@code null}.
		*
		* @param group The AudioGroup to join or {@code null} to leave.
		* @throws IllegalStateException if the stream is not properly configured.
		* @see AudioGroup
		*/
		public void join(AudioGroup group) {
		if (mGroup == group) {
		return;
		}
		if (mGroup != null) {
		mGroup.remove(this);
		mGroup = null;
		}
		if (group != null) {
		group.add(this, mCodec, mCodecType, mDtmfType);
		mGroup = group;
		}
		}

		/**
		* Sets the {@link AudioCodec} and its RTP payload type. According to RFC
		* 3551, the type must be in the range of 0 and 127, where 96 and above are
		* dynamic types. For codecs with static mappings (non-negative
		* {@link AudioCodec#defaultType}), assigning a different non-dynamic type
		* is disallowed.
		*
		* @param codec The AudioCodec to be used.
		* @param type The RTP payload type.
		* @throws IllegalArgumentException if the type is invalid or used by DTMF.
		* @throws IllegalStateException if the stream is busy.
		*/
		public void setCodec(AudioCodec codec, int type) {
		if (isBusy()) {
		throw new IllegalStateException("Busy");
		}
		if (type < 0 \|\| type > 127 \|\| (type != codec.defaultType && type < 96)) {
		throw new IllegalArgumentException("Invalid type");
		}
		if (type == mDtmfType) {
		throw new IllegalArgumentException("The type is used by DTMF");
		}
		mCodec = codec;
		mCodecType = type;
		}

		/**
		* Sets the RTP payload type for dual-tone multi-frequency (DTMF) digits.
		* The primary usage is to send digits to the remote gateway to perform
		* certain tasks, such as second-stage dialing. According to RFC 2833, the
		* RTP payload type for DTMF is assigned dynamically, so it must be in the
		* range of 96 and 127. One can use {@code -1} to disable DTMF and free up
		* the previous assigned value. This method cannot be called when the stream
		* already joined an {@link AudioGroup}.
		*
		* @param type The RTP payload type to be used or {@code -1} to disable it.
		* @throws IllegalArgumentException if the type is invalid or used by codec.
		* @throws IllegalStateException if the stream is busy.
		* @see AudioGroup#sendDtmf(int)
		*/
		public void setDtmfType(int type) {
		if (isBusy()) {
		throw new IllegalStateException("Busy");
		}
		if (type != -1) {
		if (type < 96 \|\| type > 127) {
		throw new IllegalArgumentException("Invalid type");
		}
		if (type == mCodecType) {
		throw new IllegalArgumentException("The type is used by codec");
		}
		}
		mDtmfType = type;
		}
		}

voip/java/android/net/rtp/RtpStream.java

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		/*
		* Copyright (C) 2010 The Android Open Source Project
		*
		* Licensed under the Apache License, Version 2.0 (the "License");
		* you may not use this file except in compliance with the License.
		* You may obtain a copy of the License at
		*
		* http://www.apache.org/licenses/LICENSE-2.0
		*
		* Unless required by applicable law or agreed to in writing, software
		* distributed under the License is distributed on an "AS IS" BASIS,
		* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		* See the License for the specific language governing permissions and
		* limitations under the License.
		*/

		package android.net.rtp;

		import java.net.InetAddress;
		import java.net.Inet4Address;
		import java.net.Inet6Address;
		import java.net.SocketException;

		/**
		* RtpStream represents a base class of media streams running over
		* Real-time Transport Protocol (RTP).
		*/
		/** @hide */
		public class RtpStream {
		public static final int MODE_NORMAL = 0;
		public static final int MODE_SEND_ONLY = 1;
		public static final int MODE_RECEIVE_ONLY = 2;

		private final InetAddress mLocalAddress;
		private final int mLocalPort;

		private InetAddress mRemoteAddress;
		private int mRemotePort = -1;
		private int mMode = MODE_NORMAL;

		private int mNative;
		static {
		System.loadLibrary("rtp_jni");
		}

		/**
		* Creates a RtpStream on the given local address. Note that the local
		* port is assigned automatically to conform with RFC 3550.
		*
		* @param address The network address of the local host to bind to.
		* @throws SocketException if the address cannot be bound or a problem
		* occurs during binding.
		*/
		RtpStream(InetAddress address) throws SocketException {
		mLocalPort = create(address.getHostAddress());
		mLocalAddress = address;
		}

		private native int create(String address) throws SocketException;

		/**
		* Returns the network address of the local host.
		*/
		public InetAddress getLocalAddress() {
		return mLocalAddress;
		}

		/**
		* Returns the network port of the local host.
		*/
		public int getLocalPort() {
		return mLocalPort;
		}

		/**
		* Returns the network address of the remote host or {@code null} if the
		* stream is not associated.
		*/
		public InetAddress getRemoteAddress() {
		return mRemoteAddress;
		}

		/**
		* Returns the network port of the remote host or {@code -1} if the stream
		* is not associated.
		*/
		public int getRemotePort() {
		return mRemotePort;
		}

		/**
		* Returns {@code true} if the stream is busy. This method is intended to be
		* overridden by subclasses.
		*/
		public boolean isBusy() {
		return false;
		}

		/**
		* Returns the current mode. The initial mode is {@link #MODE_NORMAL}.
		*/
		public int getMode() {
		return mMode;
		}

		/**
		* Changes the current mode. It must be one of {@link #MODE_NORMAL},
		* {@link #MODE_SEND_ONLY}, and {@link #MODE_RECEIVE_ONLY}.
		*
		* @param mode The mode to change to.
		* @throws IllegalArgumentException if the mode is invalid.
		* @throws IllegalStateException if the stream is busy.
		* @see #isBusy()
		*/
		public void setMode(int mode) {
		if (isBusy()) {
		throw new IllegalStateException("Busy");
		}
		if (mode != MODE_NORMAL && mode != MODE_SEND_ONLY && mode != MODE_RECEIVE_ONLY) {
		throw new IllegalArgumentException("Invalid mode");
		}
		mMode = mode;
		}

		/**
		* Associates with a remote host.
		*
		* @param address The network address of the remote host.
		* @param port The network port of the remote host.
		* @throws IllegalArgumentException if the address is not supported or the
		* port is invalid.
		* @throws IllegalStateException if the stream is busy.
		* @see #isBusy()
		*/
		public void associate(InetAddress address, int port) {
		if (isBusy()) {
		throw new IllegalStateException("Busy");
		}
		if (!(address instanceof Inet4Address && mLocalAddress instanceof Inet4Address) &&
		!(address instanceof Inet6Address && mLocalAddress instanceof Inet6Address)) {
		throw new IllegalArgumentException("Unsupported address");
		}
		if (port < 0 \|\| port > 65535) {
		throw new IllegalArgumentException("Invalid port");
		}
		mRemoteAddress = address;
		mRemotePort = port;
		}

		synchronized native int dup();

		/**
		* Releases allocated resources. The stream becomes inoperable after calling
		* this method.
		*
		* @throws IllegalStateException if the stream is busy.
		* @see #isBusy()
		*/
		public void release() {
		if (isBusy()) {
		throw new IllegalStateException("Busy");
		}
		close();
		}

		private synchronized native void close();

		@Override
		protected void finalize() throws Throwable {
		close();
		super.finalize();
		}
		}

voip/java/android/net/sip/BinderHelper.java

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		/*
		* Copyright (C) 2010 The Android Open Source Project
		*
		* Licensed under the Apache License, Version 2.0 (the "License");
		* you may not use this file except in compliance with the License.
		* You may obtain a copy of the License at
		*
		* http://www.apache.org/licenses/LICENSE-2.0
		*
		* Unless required by applicable law or agreed to in writing, software
		* distributed under the License is distributed on an "AS IS" BASIS,
		* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		* See the License for the specific language governing permissions and
		* limitations under the License.
		*/

		package android.net.sip;

		import android.content.ComponentName;
		import android.content.Context;
		import android.content.Intent;
		import android.content.ServiceConnection;
		import android.os.ConditionVariable;
		import android.os.IBinder;
		import android.os.IInterface;
		import android.os.Looper;
		import android.util.Log;

		// TODO: throw away this class after moving SIP classes to framework
		// This class helps to get IBinder instance of a service in a blocking call.
		// The method cannot be called in app's main thread as the ServiceConnection
		// callback will.
		class BinderHelper<T extends IInterface> {
		private Context mContext;
		private IBinder mBinder;
		private Class<T> mClass;

		BinderHelper(Context context, Class<T> klass) {
		mContext = context;
		mClass = klass;
		}

		void startService() {
		mContext.startService(new Intent(mClass.getName()));
		}

		void stopService() {
		mContext.stopService(new Intent(mClass.getName()));
		}

		IBinder getBinder() {
		// cannot call this method in app's main thread
		if (Looper.getMainLooper().getThread() == Thread.currentThread()) {
		throw new RuntimeException(
		"This method cannot be called in app's main thread");
		}

		final ConditionVariable cv = new ConditionVariable();
		cv.close();
		ServiceConnection c = new ServiceConnection() {
		public synchronized void onServiceConnected(
		ComponentName className, IBinder binder) {
		Log.v("BinderHelper", "service connected!");
		mBinder = binder;
		cv.open();
		mContext.unbindService(this);
		}

		public void onServiceDisconnected(ComponentName className) {
		cv.open();
		mContext.unbindService(this);
		}
		};
		if (mContext.bindService(new Intent(mClass.getName()), c, 0)) {
		cv.block(4500);
		}
		return mBinder;
		}
		}