RTP: remove dead code. Now we are officially in the frameworks.

Change-Id: I772d47805b982ead28cec53dda7a8acdd308a407

21 files changed, 0 insertions, 3456 deletions

diff --git a/src/android/net/rtp/AudioCodec.java b/src/android/net/rtp/AudioCodec.java
deleted file mode 100644
index 242ad58..0000000
--- a/src/android/net/rtp/AudioCodec.java
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * 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;
-
-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;
-    }
-}
diff --git a/src/android/net/rtp/AudioGroup.java b/src/android/net/rtp/AudioGroup.java
deleted file mode 100644
index 0708613..0000000
--- a/src/android/net/rtp/AudioGroup.java
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * 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;
-
-/**
- */
-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();
-    }
-}
diff --git a/src/android/net/rtp/AudioStream.java b/src/android/net/rtp/AudioStream.java
deleted file mode 100644
index c1da7ba..0000000
--- a/src/android/net/rtp/AudioStream.java
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- * 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.
- */
-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;
-    }
-}
diff --git a/src/android/net/rtp/RtpStream.java b/src/android/net/rtp/RtpStream.java
deleted file mode 100644
index b542ca7..0000000
--- a/src/android/net/rtp/RtpStream.java
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- * 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).
- */
-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();
-    }
-}
diff --git a/src/com/android/sip/rtp/AudioCodec.java b/src/com/android/sip/rtp/AudioCodec.java
deleted file mode 100644
index 1e298b2..0000000
--- a/src/com/android/sip/rtp/AudioCodec.java
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * 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 com.android.sip.rtp;
-
-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;
-    }
-}
diff --git a/src/com/android/sip/rtp/AudioStream.java b/src/com/android/sip/rtp/AudioStream.java
deleted file mode 100644
index e4a98aa..0000000
--- a/src/com/android/sip/rtp/AudioStream.java
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * 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 com.android.sip.rtp;
-
-
-/**
- * The AudioStream class represents a RTP stream carrying audio payloads.
- */
-public class AudioStream {
-    private int mNative;
-    private final RtpSocket mRtpSocket;
-
-    private AudioCodec mCodec;
-    private int mCodecType = -1;
-    private int mDtmfType = -1;
-
-    static {
-        System.loadLibrary("siprtp");
-    }
-
-    /**
-     * Creates an AudioStream on a given {@link RtpSocket}.
-     *
-     * @param rtpSocket The socket to send and receive RTP packets.
-     * @throws IllegalStateException if the socket is not associated or is
-     *     already used by another stream.
-     */
-    public AudioStream(RtpSocket rtpSocket) {
-        rtpSocket.occupy();
-        mRtpSocket = rtpSocket;
-    }
-
-    /**
-     * Closes the stream. Once closed, the stream stops sending and receiving,
-     * and relinquishes the ownership of the {@link RtpSocket}.
-     */
-    public void close() {
-        stopSending();
-        stopReceiving();
-        mRtpSocket.vacate();
-    }
-
-    /**
-     * Returns the bound {@link RtpSocket}.
-     */
-    public RtpSocket getRtpSocket() {
-        return mRtpSocket;
-    }
-
-    /**
-     * Sets the {@link AudioCodec} and the RTP payload type. According to RFC
-     * 3551, the type must be in the range of 0 and 127, where 96 and above are
-     * used by dynamic types. For codecs with static mappings (non-negative
-     * {@link AudioCodec#defaultType}), assigning a different non-dynamic type
-     * is disallowed. This method can be called at any time but only takes
-     * effect after the next call to {@link #prepare()}.
-     *
-     * @param codec The audio codec to be used.
-     * @param type The RTP payload type.
-     * @throws IllegalArgumentException if the type is out of range or
-     *     disallowed for this codec.
-     * @throws IllegalStateException if the type is already used by DTMF.
-     */
-    public synchronized void setCodec(AudioCodec codec, int type) {
-        if (type < 0 || type > 127) {
-            throw new IllegalArgumentException("The type is out of range");
-        }
-        if (type != codec.defaultType && type < 96) {
-            throw new IllegalArgumentException(
-                    "Assigning a different non-dynamic type is not allowed");
-        }
-        if (type == mDtmfType) {
-            throw new IllegalStateException("Codec and DTMF cannot use the same type");
-        }
-        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 mapping. This method can be called at any time but
-     * only takes effect after the next call to {@link #prepare()}.
-     *
-     * @param type The RTP payload type or {@code -1} to disable DTMF.
-     * @throws IllegalArgumentException if the type is out of range.
-     * @throws IllegalStateException if the type is already used by the codec.
-     * @see #sendDtmf(int)
-     */
-    public synchronized void setDtmf(int type) {
-        if (type != -1) {
-            if (type < 96 || type > 127) {
-                throw new IllegalArgumentException("The type is out of range");
-            }
-            if (type == mCodecType) {
-                throw new IllegalStateException("Codec and DTMF cannot use the same type");
-            }
-        }
-        mDtmfType = type;
-    }
-
-    /**
-     * Allocates native resources for the current configuration.
-     *
-     * @throws IllegalStateException if the codec is not set or the stream is
-     *     already prepared.
-     */
-    public synchronized void prepare() {
-        if (mCodec == null) {
-            throw new IllegalStateException("Codec is not set");
-        }
-        prepare(mRtpSocket, mCodec.name, mCodec.sampleRate,
-                mCodec.sampleCount, mCodecType, mDtmfType);
-    }
-    private native void prepare(RtpSocket rtpSocket, String codecName,
-            int sampleRate, int sampleCount, int codecType, int dtmfType);
-
-    /**
-     * Returns {@code true} if the stream is already prepared.
-     */
-    public native synchronized boolean isPrepared();
-
-    /**
-     * Starts recording and sending encoded audio to the remote host.
-     *
-     * @throws IllegalStateException if the stream is not prepared.
-     */
-    public native synchronized void startSending();
-
-    /**
-     * Starts receiving and playing decoded audio from the remote host.
-     *
-     * @throws IllegalStateException if the stream is not prepared.
-     */
-    public native synchronized void startReceiving();
-
-    /**
-     * Sends a DTMF digit to the remote host. One must set the RTP payload type
-     * for DTMF before calling this method. Currently only events {@code 0}
-     * through {@code 15} are supported.
-     *
-     * @throws IllegalArgumentException if the event is out of range.
-     * @throws IllegalStateException if the stream is not sending or DTMF is
-     *     disabled.
-     * @see #setDtmf(int)
-     */
-    public native synchronized void sendDtmf(int event);
-
-    /**
-     * Stops sending.
-     */
-    public native synchronized void stopSending();
-
-    /**
-     * Stops receiving.
-     */
-    public native synchronized void stopReceiving();
-
-    /**
-     * Releases native resources. It also stops sending and receiving. After
-     * calling this method, one can call {@link #prepare()} again with or
-     * without changing the configuration.
-     */
-    public native synchronized void release();
-
-    @Override
-    protected void finalize() throws Throwable {
-        release();
-        super.finalize();
-    }
-}
diff --git a/src/com/android/sip/rtp/RtpSocket.java b/src/com/android/sip/rtp/RtpSocket.java
deleted file mode 100644
index 1f29f21..0000000
--- a/src/com/android/sip/rtp/RtpSocket.java
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * 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 com.android.sip.rtp;
-
-import java.net.InetAddress;
-import java.net.SocketException;
-
-/**
- * The RtpSocket class represents a UDP socket carrying RTP packets. The network
- * port of the local host is assigned automatically to conform with RFC 3550.
- */
-public class RtpSocket {
-    private int mNative;
-    private final InetAddress mLocalAddress;
-    private final int mLocalPort;
-
-    private InetAddress mRemoteAddress;
-    private int mRemotePort = -1;
-    private boolean mInUse = false;
-
-    static {
-        System.loadLibrary("siprtp");
-    }
-
-    /**
-     * Creates a RtpSocket by specifying the network address of the local host.
-     *
-     * @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 RtpSocket(InetAddress address) throws SocketException {
-        mLocalPort = create(address.getHostAddress());
-        mLocalAddress = address;
-    }
-    private native int create(String address) throws SocketException;
-
-    synchronized void occupy() {
-        if (mRemoteAddress == null) {
-            throw new IllegalStateException("RtpSocket is not associated");
-        }
-        if (mInUse) {
-            throw new IllegalStateException("RtpSocket is already in use");
-        }
-        mInUse = true;
-    }
-
-    synchronized void vacate() {
-        mInUse = false;
-    }
-
-    /**
-     * 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
-     * socket is not associated.
-     */
-    public InetAddress getRemoteAddress() {
-        return mRemoteAddress;
-    }
-
-    /**
-     * Returns the network port of the remote host or {@code -1} if the socket
-     * is not associated.
-     */
-    public int getRemotePort() {
-        return mRemotePort;
-    }
-
-    /**
-     * Associates with a remote host.
-     *
-     * @param address The network address of the remote host.
-     * @param port The network port of the remote host.
-     * @throws SocketException if the address family is different or the socket
-     *     is already associated.
-     */
-    public synchronized void associate(InetAddress address, int port)
-            throws SocketException {
-        associate(address.getHostAddress(), port);
-        mRemoteAddress = address;
-        mRemotePort = port;
-    }
-    private native void associate(String address, int port) throws SocketException;
-
-    @Override
-    protected void finalize() throws Throwable {
-        release();
-        super.finalize();
-    }
-    private native void release();
-}
diff --git a/src/jni/rtp/Android.mk b/src/jni/rtp/Android.mk
deleted file mode 100644
index fb1fdb8..0000000
--- a/src/jni/rtp/Android.mk
+++ /dev/null
@@ -1,43 +0,0 @@
-#
-# 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.
-#
-
-LOCAL_PATH := $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := libsiprtp
-
-LOCAL_SRC_FILES := \
-	RtpSocket.cpp \
-	AudioStream.cpp \
-	G711Codec.cpp \
-	libsiprtp.cpp
-
-LOCAL_SHARED_LIBRARIES := \
-	libnativehelper \
-	libutils \
-	libcutils \
-	libmedia
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_C_INCLUDES += \
-	$(JNI_H_INCLUDE)
-
-LOCAL_CFLAGS += -fvisibility=hidden
-
-LOCAL_PRELINK_MODULE := false
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/src/jni/rtp/AudioCodec.h b/src/jni/rtp/AudioCodec.h
deleted file mode 100644
index f9aee94..0000000
--- a/src/jni/rtp/AudioCodec.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyrightm (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.
- */
-
-#ifndef __AUDIO_CODEC_H__
-#define __AUDIO_CODEC_H__
-
-class AudioCodec
-{
-public:
-    virtual ~AudioCodec() {}
-    // Returns true if sampleCount is acceptable.
-    virtual bool set(int sampleCount) = 0;
-    // Returns the length of payload in bytes.
-    virtual int encode(void *payload, int16_t *samples) = 0;
-    // Returns the number of decoded samples.
-    virtual int decode(int16_t *samples, void *payload, int length) = 0;
-};
-
-class UlawCodec : public AudioCodec
-{
-public:
-    virtual bool set(int sampleCount) {
-        mSampleCount = sampleCount;
-        return true;
-    }
-    virtual int encode(void *payload, int16_t *samples);
-    virtual int decode(int16_t *samples, void *payload, int length);
-private:
-    int mSampleCount;
-};
-
-class AlawCodec : public AudioCodec
-{
-public:
-    virtual bool set(int sampleCount) {
-        mSampleCount = sampleCount;
-        return true;
-    }
-    virtual int encode(void *payload, int16_t *samples);
-    virtual int decode(int16_t *samples, void *payload, int length);
-private:
-    int mSampleCount;
-};
-
-#endif
diff --git a/src/jni/rtp/AudioStream.cpp b/src/jni/rtp/AudioStream.cpp
deleted file mode 100644
index 8093414..0000000
--- a/src/jni/rtp/AudioStream.cpp
+++ /dev/null
@@ -1,660 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#include <string.h>
-#include <unistd.h>
-#include <errno.h>
-#include <fcntl.h>
-#include <sys/stat.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <time.h>
-#include <arpa/inet.h>
-#include <pthread.h>
-
-#define LOG_TAG "AudioStream"
-#include <utils/Log.h>
-#include <utils/Errors.h>
-#include <utils/RefBase.h>
-#include <utils/threads.h>
-#include <media/AudioRecord.h>
-#include <media/AudioTrack.h>
-#include <media/mediarecorder.h>
-
-#include "jni.h"
-#include "JNIHelp.h"
-
-#include "RtpSocket.h"
-#include "AudioCodec.h"
-
-using namespace android;
-
-static int gRandom = -1;
-
-class JitterBuffer {
-    static const int SIZE = 6;
-    uint32_t mBufferSize;
-    uint8_t *mBuffer[SIZE];
-    uint16_t *mLength;
-    uint16_t mHead;
-    uint16_t mTail;
-
-    public:
-        JitterBuffer(int sampleCount) {
-            mHead = mTail = 0;
-            mBufferSize = 2048 + (sizeof(int16_t) * sampleCount);
-            for (int i = 0; i < SIZE ; ++i) {
-                mBuffer[i] = (uint8_t*) malloc(mBufferSize);
-            }
-            mLength = (uint16_t*) malloc(SIZE * sizeof(uint16_t));
-        }
-
-        ~JitterBuffer() {
-            for (int i = 0; i < SIZE ; ++i) {
-                free(mBuffer[i]);
-            }
-            free(mLength);
-        }
-
-        int getBufferSize() {
-            return mBufferSize;
-        }
-
-        uint8_t* obtainBuffer() {
-            // we need extra buffer to save one memcpy.
-            int reservedHead = ((mHead == 0) ? (SIZE - 1) : (mHead - 1));
-            if (mTail == reservedHead) return NULL;
-            return mBuffer[mTail];
-        }
-
-        void pushBack(int length) {
-            mLength[mTail] = length;
-            if (++mTail == SIZE) mTail = 0;
-        }
-
-        unsigned int popFront(uint8_t **packet) {
-            int length = mLength[mHead];
-            *packet = mBuffer[mHead];
-            if (++mHead == SIZE) mHead = 0;
-            return length;
-        }
-
-        bool empty() {
-            return (mHead == mTail);
-        }
-};
-
-class AudioStream
-{
-public:
-    AudioStream();
-    ~AudioStream();
-
-    bool set(RtpSocket *rtpSocket, const char *codecName,
-        int sampleRate, int sampleCount, int codecType, int dtmfType);
-
-    bool startSending();
-    bool startReceiving();
-    bool sendDtmf(int event);
-    void stopSending();
-    void stopReceiving();
-
-private:
-    RtpSocket *mSocket;
-    AudioCodec *mCodec;
-    AudioRecord mRecord;
-    AudioTrack mTrack;
-    pthread_mutex_t mDtmfLock;
-
-    uint16_t mLocalSequence;
-    uint32_t mLocalTimestamp;
-    uint32_t mLocalSsrc;
-    uint32_t mRemoteTimestamp;
-    uint32_t mRemoteSsrc;
-    uint32_t mCodecMagic;
-    uint32_t mDtmfMagic;
-
-    int mSampleRate;
-    int mSampleCount;
-    int mInterval;
-    int mTimer;
-
-    JitterBuffer *mJitterBuffer;
-
-    volatile int32_t mNextDtmfEvent;
-    int mDtmfEvent;
-    int mDtmfDuration;
-
-    bool encode();
-    bool decode();
-
-    void adjustMicGain(int16_t *buf, int len, int factor);
-
-    int getPacketFromJB(RtpSocket *rtpSocket, uint8_t **buffer, timeval *deadline);
-
-    class Sender : public Thread
-    {
-    public:
-        Sender(AudioStream *stream) : Thread(false), mStream(stream) {}
-    private:
-        virtual bool threadLoop()
-        {
-            if (!mStream->encode()) {
-                mStream->mRecord.stop();
-                return false;
-            }
-            return true;
-        }
-        AudioStream *mStream;
-    };
-    sp<Sender> mSender;
-
-    class Receiver : public Thread
-    {
-    public:
-        Receiver(AudioStream *stream) : Thread(false), mStream(stream) {}
-    private:
-        virtual bool threadLoop()
-        {
-            if (!mStream->decode()) {
-                mStream->mTrack.stop();
-                return false;
-            }
-            return true;
-        }
-        AudioStream *mStream;
-    };
-    sp<Receiver> mReceiver;
-};
-
-AudioStream::AudioStream()
-{
-    mSender = new Sender(this);
-    mReceiver = new Receiver(this);
-    mCodec = NULL;
-    mJitterBuffer = NULL;
-    pthread_mutex_init(&mDtmfLock, NULL);
-}
-
-AudioStream::~AudioStream()
-{
-    stopSending();
-    stopReceiving();
-    mSender.clear();
-    mReceiver.clear();
-    delete mCodec;
-    delete mJitterBuffer;
-}
-
-bool AudioStream::set(RtpSocket *rtpSocket, const char *codecName,
-    int sampleRate, int sampleCount, int codecType, int dtmfType)
-{
-    mSocket = rtpSocket;
-
-    // One frame per second is just not reasonable.
-    if (sampleRate <= 0 || sampleCount <= 0 || sampleRate <= sampleCount) {
-        return false;
-    }
-
-    // Find AudioCodec and configure it.
-    if (strcmp("PCMU", codecName) == 0) {
-        mCodec = new UlawCodec;
-    } else if (strcmp("PCMA", codecName) == 0) {
-        mCodec = new AlawCodec;
-    } else {
-        mCodec = NULL;
-    }
-    if (mCodec == NULL || !mCodec->set(sampleCount)) {
-        return false;
-    }
-
-    // Set AudioRecord with double buffer. Otherwise try system default.
-    if (mRecord.set(AUDIO_SOURCE_MIC, sampleRate, AudioSystem::PCM_16_BIT,
-        AudioSystem::CHANNEL_IN_MONO, sampleCount * 2) != NO_ERROR &&
-        mRecord.set(AUDIO_SOURCE_MIC, sampleRate, AudioSystem::PCM_16_BIT,
-        AudioSystem::CHANNEL_IN_MONO) != NO_ERROR) {
-        return false;
-    }
-
-    // Set AudioTrack with double buffer. Otherwise try system default.
-    if (mTrack.set(AudioSystem::VOICE_CALL, sampleRate, AudioSystem::PCM_16_BIT,
-        AudioSystem::CHANNEL_OUT_MONO, sampleCount * 2) != NO_ERROR &&
-        mTrack.set(AudioSystem::VOICE_CALL, sampleRate, AudioSystem::PCM_16_BIT,
-        AudioSystem::CHANNEL_OUT_MONO) != NO_ERROR) {
-        return false;
-    }
-
-    // Only initialize these random bits once for the maximum compatibility.
-    read(gRandom, &mLocalSequence, sizeof(mLocalSequence));
-    read(gRandom, &mLocalTimestamp, sizeof(mLocalTimestamp));
-    read(gRandom, &mLocalSsrc, sizeof(mLocalSsrc));
-
-    mCodecMagic = (0x8000 | codecType) << 16;
-    mDtmfMagic = (dtmfType == -1 ? -1 : (0x8000 | dtmfType) << 16);
-
-    mSampleRate = sampleRate;
-    mSampleCount = sampleCount;
-
-    if (mJitterBuffer == NULL) {
-        mJitterBuffer = new JitterBuffer(sampleCount);
-    }
-
-    // mInterval is a threshold for jitter control in microseconds. To avoid
-    // introducing more latencies, here we use 0.8 times of the real interval.
-    mInterval = 1000 * sampleCount / sampleRate * 800;
-
-    return true;
-}
-
-bool AudioStream::startSending()
-{
-    if (mRecord.stopped()) {
-        mTimer = 0;
-        mNextDtmfEvent = -1;
-        mDtmfEvent = -1;
-
-        if (mRecord.start() != NO_ERROR ||
-            mSender->run("Sender", ANDROID_PRIORITY_AUDIO) != NO_ERROR) {
-            mRecord.stop();
-            return false;
-        }
-    }
-    return true;
-}
-
-bool AudioStream::startReceiving()
-{
-    if (mTrack.stopped()) {
-        mRemoteTimestamp = 0;
-        mRemoteSsrc = 0;
-
-        mTrack.start();
-        if (mReceiver->run("Receiver", ANDROID_PRIORITY_AUDIO) != NO_ERROR) {
-            mTrack.stop();
-            return false;
-        }
-    }
-    return true;
-}
-
-bool AudioStream::sendDtmf(int event)
-{
-    if (mRecord.stopped() || ~mDtmfMagic == 0) {
-        return false;
-    }
-    if (pthread_mutex_trylock(&mDtmfLock) != 0) {
-        usleep(mInterval * 2);
-        if (pthread_mutex_trylock(&mDtmfLock) != 0) return false;
-    }
-    mNextDtmfEvent = event;
-    pthread_mutex_unlock(&mDtmfLock);
-    return true;
-}
-
-void AudioStream::stopSending()
-{
-    if (!mRecord.stopped()) {
-        mSender->requestExitAndWait();
-        mRecord.stop();
-    }
-}
-
-void AudioStream::stopReceiving()
-{
-    if (!mTrack.stopped()) {
-        mReceiver->requestExitAndWait();
-        mTrack.stop();
-    }
-}
-
-// TODO: remove this function after the mic level issue was fixed in driver.
-void AudioStream::adjustMicGain(int16_t *buf, int len, int factor)
-{
-    int i, j;
-    int bound = 32768/factor;
-    for (i = 0; i < len; i++) {
-        j = buf[i];
-        if (j > bound) {
-            buf[i] = 32767;
-        } else if (j < -bound) {
-            buf[i] = -32767;
-        } else {
-            buf[i] = (int16_t)(factor*j);
-        }
-    }
-}
-
-// -----------------------------------------------------------------------------
-
-bool AudioStream::encode()
-{
-    int16_t samples[mSampleCount];
-
-    // Read samples from AudioRecord. Since AudioRecord itself has fault
-    // recovery mechanism, we just return false if the length is wrong.
-    int length = mRecord.read(samples, sizeof(samples));
-    if (length - sizeof(samples) != 0) {
-        LOGD("read");
-        return false;
-    }
-
-    adjustMicGain(samples, length/sizeof(int16_t), 8);
-
-    mLocalSequence++;
-    mLocalTimestamp += mSampleCount;
-
-    // If we have a DTMF event to send, send it now.
-    pthread_mutex_lock(&mDtmfLock);
-    int32_t event = mNextDtmfEvent;
-    mNextDtmfEvent = -1;
-    pthread_mutex_unlock(&mDtmfLock);
-    if (event != -1) {
-        mDtmfEvent = event << 24;
-        mDtmfDuration = 0;
-    }
-    if (mDtmfEvent != -1) {
-        mDtmfDuration += mSampleCount;
-        uint32_t packet[4] = {
-            htonl(mDtmfMagic | mLocalSequence),
-            htonl(mLocalTimestamp - mDtmfDuration),
-            mLocalSsrc,
-            htonl(mDtmfEvent | mDtmfDuration),
-        };
-        // Make the DTMF event roughly 200 millisecond long.
-        if (mDtmfDuration * 5 >= mSampleRate) {
-            packet[3] |= 1 << 24;
-            mDtmfEvent = -1;
-        }
-        send(mSocket, packet, sizeof(packet));
-        return true;
-    }
-
-    // Otherwise encode the samples and prepare the packet.
-    __attribute__((aligned(4))) uint8_t packet[12 + sizeof(samples)];
-
-    uint32_t *header = (uint32_t *)packet;
-    header[0] = htonl(mCodecMagic | mLocalSequence);
-    header[1] = htonl(mLocalTimestamp);
-    header[2] = mLocalSsrc;
-
-    length = mCodec->encode(&packet[12], samples);
-    if (length <= 0) {
-        LOGD("encode");
-        return false;
-    }
-    length += 12;
-
-    // Here we implement a simple jitter control for the outgoing packets.
-    // Ideally we should send out packets at a constant rate, but in practice
-    // every component in the pipeline might delay or speed up a little. To
-    // avoid making things worse, we only delay the packet which comes early.
-    // Note that interval is less than the real one, so the delay will be
-    // converged.
-    timeval now;
-    if (gettimeofday(&now, NULL) != 0) {
-        LOGD("gettimeofday");
-        return false;
-    }
-    int interval = now.tv_sec * 1000000 + now.tv_usec - mTimer;
-    if (interval > 0 && interval < mInterval) {
-        usleep(mInterval - interval);
-        interval = mInterval;
-    }
-    mTimer += interval;
-
-    send(mSocket, packet, length);
-    return true;
-}
-
-bool AudioStream::decode()
-{
-    timeval deadline;
-
-    if (gettimeofday(&deadline, NULL) != 0) {
-        LOGD("gettimeofday");
-        return false;
-    }
-
-    // mInterval is always less than 1000000.
-    deadline.tv_usec += mInterval;
-    if (deadline.tv_usec > 1000000) {
-        deadline.tv_usec -= 1000000;
-        deadline.tv_sec++;
-    }
-
-    int16_t samples[mSampleCount];
-    uint8_t *packet;
-
-    while (1) {
-        int length = getPacketFromJB(mSocket, &packet, &deadline);
-        if (length <= 0) {
-            return true;
-        }
-        if (length < 12) {
-            continue;
-        }
-
-        // Here we check all the fields in the standard RTP header. Some
-        // restrictions might be too tight and could be removed in the future.
-        int offset = 12 + (packet[0] & 0x0F) * 4;
-        if ((packet[0] & 0x10) != 0) {
-            offset += 4 + (packet[offset + 2] << 8 | packet[offset + 3]) * 4;
-        }
-        if ((packet[0] & 0x20) != 0 && length - sizeof(packet) <= 0) {
-            length -= packet[length - 1];
-        }
-        length -= offset;
-        if (length < 0) {
-            continue;
-        }
-
-        uint32_t *header = (uint32_t *)packet;
-        header[0] = ntohl(header[0]);
-        header[1] = ntohl(header[1]);
-
-        if ((header[0] & 0xC07F0000) != mCodecMagic) {
-            LOGD("wrong magic (%X != %X)", mCodecMagic, header[0] & 0xC07F0000);
-            continue;
-        }
-
-        mRemoteTimestamp = header[1];
-        mRemoteSsrc = header[2];
-
-        length = mCodec->decode(samples, &packet[offset], length) * 2;
-        if (length <= 0) {
-            LOGD("decode");
-            continue;
-        }
-
-        // Write samples to AudioTrack. Again, since AudioTrack itself has fault
-        // recovery mechanism, we just return false if the length is wrong.
-        return mTrack.write(samples, length) == length;
-    }
-}
-
-int AudioStream::getPacketFromJB(RtpSocket *rtpSocket, uint8_t **buffer,
-    timeval *deadline)
-{
-    // Here we implement a simple jitter control for the incoming packets.
-    // Ideally there should be only one packet every time we try to read
-    // from the socket. If any packets are late, we must drop incoming packets
-    // if the jitter buffer is full already.
-
-    int result, count = 0;
-    if (mJitterBuffer->empty()) {
-        *buffer = mJitterBuffer->obtainBuffer();
-        result = receive(rtpSocket, *buffer, mJitterBuffer->getBufferSize(),
-                         deadline);
-        if (result <= 0) return result;
-        mJitterBuffer->pushBack(result);
-    }
-    result = mJitterBuffer->popFront(buffer);
-    while (1) {
-        void *fillBuffer = (void*) mJitterBuffer->obtainBuffer();
-        int length = receive(mSocket, fillBuffer, ((fillBuffer == NULL) ?
-                             0 : mJitterBuffer->getBufferSize()), NULL);
-        if (length <= 0) break;
-        if (fillBuffer != NULL) {
-            mJitterBuffer->pushBack(length);
-        } else {
-            count++;
-        }
-    }
-    if (count > 0) {
-        LOGD("Drop %d packet(s), jitter buffer is full!", count);
-    }
-    return result;
-}
-// -----------------------------------------------------------------------------
-
-static jfieldID gNative;
-
-static void throwIllegalStateException(JNIEnv *env, const char *message)
-{
-    jniThrowException(env, "java/lang/IllegalStateException", message);
-}
-
-// All these JNI methods are synchronized in java class, so we implement them
-// without using any mutex locks. Simple is best!
-
-static void prepare(JNIEnv *env, jobject thiz, jobject jRtpSocket,
-    jstring jCodecName, jint sampleRate, jint sampleCount, jint codecType,
-    jint dtmfType)
-{
-    AudioStream *stream = (AudioStream *)env->GetIntField(thiz, gNative);
-    if (stream != NULL) {
-        throwIllegalStateException(env, "Already prepared");
-        return;
-    }
-
-    RtpSocket *rtpSocket = getRtpSocket(env, jRtpSocket, true);
-    if (rtpSocket == NULL) {
-        // Exception already thrown.
-        return;
-    }
-
-    if (jCodecName == NULL) {
-        jniThrowNullPointerException(env, "codecName");
-        return;
-    }
-
-    const char *codecName = env->GetStringUTFChars(jCodecName, NULL);
-    stream = new AudioStream;
-    if (!stream->set(rtpSocket, codecName, sampleRate, sampleCount, codecType,
-        dtmfType)) {
-        delete stream;
-        stream = NULL;
-    }
-    env->ReleaseStringUTFChars(jCodecName, codecName);
-
-    if (stream == NULL) {
-         throwIllegalStateException(env, "Failed to create native AudioStream");
-    }
-    env->SetIntField(thiz, gNative, (int)stream);
-}
-
-static jboolean isPrepared(JNIEnv *env, jobject thiz)
-{
-    return env->GetIntField(thiz, gNative) != 0;
-}
-
-static void startSending(JNIEnv *env, jobject thiz)
-{
-    AudioStream *stream = (AudioStream *)env->GetIntField(thiz, gNative);
-    if (stream == NULL) {
-        throwIllegalStateException(env, "Not prepared");
-    } else if (!stream->startSending()) {
-        throwIllegalStateException(env, "Failed to start native AudioRecord");
-    }
-}
-
-static void startReceiving(JNIEnv *env, jobject thiz)
-{
-    AudioStream *stream = (AudioStream *)env->GetIntField(thiz, gNative);
-    if (stream == NULL) {
-        throwIllegalStateException(env, "Not prepared");
-    } else if (!stream->startReceiving()) {
-        throwIllegalStateException(env, "Failed to start native AudioTrack");
-    }
-}
-
-static void sendDtmf(JNIEnv *env, jobject thiz, jint event)
-{
-    AudioStream *stream = (AudioStream *)env->GetIntField(thiz, gNative);
-    if (stream == NULL) {
-        throwIllegalStateException(env, "Not prepared");
-    } else if (event < 0 || event > 15) {
-        jniThrowException(env, "java/lang/IllegalArgumentException", "event");
-    } else if (!stream->sendDtmf(event)) {
-        throwIllegalStateException(env, "Failed to send DTMF");
-    }
-}
-
-static void stopSending(JNIEnv *env, jobject thiz)
-{
-    AudioStream *stream = (AudioStream *)env->GetIntField(thiz, gNative);
-    if (stream != NULL) {
-        stream->stopSending();
-    }
-}
-
-static void stopReceiving(JNIEnv *env, jobject thiz)
-{
-    AudioStream *stream = (AudioStream *)env->GetIntField(thiz, gNative);
-    if (stream != NULL) {
-        stream->stopReceiving();
-    }
-}
-
-static void release(JNIEnv *env, jobject thiz)
-{
-    delete (AudioStream *)env->GetIntField(thiz, gNative);
-    env->SetIntField(thiz, gNative, NULL);
-}
-
-//------------------------------------------------------------------------------
-
-static JNINativeMethod gMethods[] = {
-    {"prepare", "(Lcom/android/sip/rtp/RtpSocket;Ljava/lang/String;IIII)V",
-        (void *)prepare},
-    {"isPrepared", "()Z", (void *)isPrepared},
-    {"startSending", "()V", (void *)startSending},
-    {"startReceiving", "()V", (void *)startReceiving},
-    {"sendDtmf", "(I)V", (void *)sendDtmf},
-    {"stopSending", "()V", (void *)stopSending},
-    {"stopReceiving", "()V", (void *)stopReceiving},
-    {"release", "()V", (void *)release},
-};
-
-int registerAudioStream(JNIEnv *env)
-{
-    gRandom = open("/dev/urandom", O_RDONLY);
-    if (gRandom == -1) {
-        LOGE("urandom: %s", strerror(errno));
-        return -1;
-    }
-
-    jclass clazz;
-    if ((clazz = env->FindClass("com/android/sip/rtp/AudioStream")) == NULL ||
-        (gNative = env->GetFieldID(clazz, "mNative", "I")) == NULL ||
-        env->RegisterNatives(clazz, gMethods, NELEM(gMethods)) < 0) {
-        LOGE("JNI registration failed");
-        return -1;
-    }
-    return 0;
-}
diff --git a/src/jni/rtp/G711Codec.cpp b/src/jni/rtp/G711Codec.cpp
deleted file mode 100644
index 2226850..0000000
--- a/src/jni/rtp/G711Codec.cpp
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyrightm (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.
- */
-
-#include <stdint.h>
-
-#include "AudioCodec.h"
-
-static int8_t gExponents[128] = {
-    0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4,
-    5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
-    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-};
-
-int UlawCodec::encode(void *payload, int16_t *samples)
-{
-    int8_t *ulaws = (int8_t *)payload;
-    for (int i = 0; i < mSampleCount; ++i) {
-        int sample = samples[i];
-        int sign = (sample >> 8) & 0x80;
-        if (sample < 0) {
-            sample = -sample;
-        }
-        sample += 132;
-        if (sample > 32767) {
-            sample = 32767;
-        }
-        int exponent = gExponents[sample >> 8];
-        int mantissa = (sample >> (exponent + 3)) & 0x0F;
-        ulaws[i] = ~(sign | (exponent << 4) | mantissa);
-    }
-    return mSampleCount;
-}
-
-int UlawCodec::decode(int16_t *samples, void *payload, int length)
-{
-    int8_t *ulaws = (int8_t *)payload;
-    for (int i = 0; i < length; ++i) {
-        int ulaw = ~ulaws[i];
-        int exponent = (ulaw >> 4) & 0x07;
-        int mantissa = ulaw & 0x0F;
-        int sample = (((mantissa << 3) + 132) << exponent) - 132;
-        samples[i] = (ulaw < 0 ? -sample : sample);
-    }
-    return length;
-}
-
-int AlawCodec::encode(void *payload, int16_t *samples)
-{
-    int8_t *alaws = (int8_t *)payload;
-    for (int i = 0; i < mSampleCount; ++i) {
-        int sample = samples[i];
-        int sign = (sample >> 8) & 0x80;
-        if (sample < 0) {
-            sample = -sample;
-        }
-        if (sample > 32767) {
-            sample = 32767;
-        }
-        int exponent = gExponents[sample >> 8];
-        int mantissa = (sample >> (exponent == 0 ? 4 : exponent + 3)) & 0x0F;
-        alaws[i] = (sign | (exponent << 4) | mantissa) ^ 0xD5;
-    }
-    return mSampleCount;
-}
-
-int AlawCodec::decode(int16_t *samples, void *payload, int length)
-{
-    int8_t *alaws = (int8_t *)payload;
-    for (int i = 0; i < length; ++i) {
-        int alaw = alaws[i] ^ 0x55;
-        int exponent = (alaw >> 4) & 0x07;
-        int mantissa = alaw & 0x0F;
-        int sample = (exponent == 0 ? (mantissa << 4) + 8 :
-            ((mantissa << 3) + 132) << exponent);
-        samples[i] = (alaw < 0 ? sample : -sample);
-    }
-    return length;
-}
diff --git a/src/jni/rtp/RtpSocket.cpp b/src/jni/rtp/RtpSocket.cpp
deleted file mode 100644
index cc63e3f..0000000
--- a/src/jni/rtp/RtpSocket.cpp
+++ /dev/null
@@ -1,254 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#include <string.h>
-#include <errno.h>
-#include <sys/socket.h>
-#include <sys/types.h>
-#include <arpa/inet.h>
-#include <netinet/in.h>
-#include <sys/time.h>
-#include <time.h>
-
-#define LOG_TAG "RtpSocket"
-#include <utils/Log.h>
-
-#include "jni.h"
-#include "JNIHelp.h"
-
-#include "RtpSocket.h"
-
-static jfieldID gNative;
-
-struct RtpSocket
-{
-    int mFd;
-    int mFamily;
-    sockaddr_storage mRemote;
-
-    RtpSocket(int fd, sockaddr_storage *local)
-    {
-        mFd = fd;
-        mFamily = local->ss_family;
-        mRemote.ss_family = ~mFamily;
-    }
-
-    ~RtpSocket() { close(mFd); }
-};
-
-//------------------------------------------------------------------------------
-
-RtpSocket *getRtpSocket(JNIEnv *env, jobject jRtpSocket, bool associated)
-{
-    if (jRtpSocket == NULL) {
-        jniThrowNullPointerException(env, "rtpSocket");
-        return NULL;
-    }
-    RtpSocket *rtpSocket = (RtpSocket *)env->GetIntField(jRtpSocket, gNative);
-    if (rtpSocket == NULL) {
-        jniThrowException(env, "java/lang/IllegalStateException", "native");
-        LOGE("native is NULL");
-        return NULL;
-    }
-    if (associated && (rtpSocket->mRemote.ss_family != rtpSocket->mFamily)) {
-        jniThrowException(env, "java/lang/IllegalStateException",
-                strerror(ENOTCONN));
-        return NULL;
-    }
-    return rtpSocket;
-}
-
-int send(RtpSocket *rtpSocket, void *buffer, int length)
-{
-    return sendto(rtpSocket->mFd, buffer, length, MSG_NOSIGNAL,
-        (sockaddr *)&rtpSocket->mRemote, sizeof(rtpSocket->mRemote));
-}
-
-int receive(RtpSocket *rtpSocket, void *buffer, int length, timeval *deadline)
-{
-    int flag = MSG_TRUNC | MSG_DONTWAIT;
-    if (deadline != NULL) {
-        timeval timeout;
-        if (gettimeofday(&timeout, NULL) != 0) {
-            return -1;
-        }
-
-        int remain = (deadline->tv_sec - timeout.tv_sec) * 1000000 +
-            deadline->tv_usec - timeout.tv_usec;
-        if (remain <= 0) {
-            return 0;
-        }
-
-        if (remain < 1000000) {
-            timeout.tv_sec = 0;
-            timeout.tv_usec = remain;
-        } else {
-            timeout.tv_sec = remain / 1000000;
-            timeout.tv_usec = remain - timeout.tv_sec * 1000000;
-        }
-        if (setsockopt(rtpSocket->mFd, SOL_SOCKET, SO_RCVTIMEO, &timeout,
-            sizeof(timeout)) != 0) {
-            return -1;
-        }
-        flag ^= MSG_DONTWAIT;
-    }
-
-    length = recv(rtpSocket->mFd, buffer, length, flag);
-    if (length == -1 && (errno == EAGAIN || errno == EINTR)) {
-        return 0;
-    }
-    return length;
-}
-
-//------------------------------------------------------------------------------
-
-static void throwSocketException(JNIEnv *env, int error)
-{
-    jniThrowException(env, "java/net/SocketException", strerror(error));
-}
-
-static int parse(JNIEnv *env, jstring jAddress, jint port, sockaddr_storage *ss)
-{
-    if (jAddress == NULL) {
-        jniThrowNullPointerException(env, "address");
-        return -1;
-    }
-    if (port < 0 || port > 65535) {
-        jniThrowException(env, "java/lang/IllegalArgumentException", "port");
-        return -1;
-    }
-    const char *address = env->GetStringUTFChars(jAddress, NULL);
-    if (address == NULL) {
-        // Exception already thrown.
-        return -1;
-    }
-    memset(ss, 0, sizeof(*ss));
-
-    sockaddr_in *sin = (sockaddr_in *)ss;
-    if (inet_pton(AF_INET, address, &(sin->sin_addr)) > 0) {
-        sin->sin_family = AF_INET;
-        sin->sin_port = htons(port);
-        env->ReleaseStringUTFChars(jAddress, address);
-        return 0;
-    }
-
-    sockaddr_in6 *sin6 = (sockaddr_in6 *)ss;
-    if (inet_pton(AF_INET6, address, &(sin6->sin6_addr)) > 0) {
-        sin6->sin6_family = AF_INET6;
-        sin6->sin6_port = htons(port);
-        env->ReleaseStringUTFChars(jAddress, address);
-        return 0;
-    }
-
-    env->ReleaseStringUTFChars(jAddress, address);
-    jniThrowException(env, "java/lang/IllegalArgumentException", "address");
-    return -1;
-}
-
-static jint create(JNIEnv *env, jobject thiz, jstring jAddress)
-{
-    sockaddr_storage ss;
-    if (parse(env, jAddress, 0, &ss) < 0) {
-        // Exception already thrown.
-        return -1;
-    }
-
-    int fd = socket(ss.ss_family, SOCK_DGRAM, 0);
-    socklen_t len = sizeof(ss);
-    if (fd == -1 || bind(fd, (sockaddr *)&ss, sizeof(ss)) != 0 ||
-        getsockname(fd, (sockaddr *)&ss, &len) != 0) {
-        throwSocketException(env, errno);
-        close(fd);
-        return -1;
-    }
-
-    uint16_t *p = (ss.ss_family == AF_INET ?
-        &((sockaddr_in *)&ss)->sin_port : &((sockaddr_in6 *)&ss)->sin6_port);
-    uint16_t port = ntohs(*p);
-    if ((port & 1) == 0) {
-        env->SetIntField(thiz, gNative, (int)new RtpSocket(fd, &ss));
-        return port;
-    }
-    close(fd);
-
-    fd = socket(ss.ss_family, SOCK_DGRAM, 0);
-    if (fd != -1) {
-        uint16_t delta = port << 1;
-        ++port;
-
-        for (int i = 0; i < 1000; ++i) {
-            do {
-                port += delta;
-            } while (port < 1024);
-            *p = htons(port);
-
-            if (bind(fd, (sockaddr *)&ss, sizeof(ss)) == 0) {
-                env->SetIntField(thiz, gNative, (int)new RtpSocket(fd, &ss));
-                return port;
-            }
-        }
-    }
-
-    throwSocketException(env, errno);
-    close(fd);
-    return -1;
-}
-
-static void associate(JNIEnv *env, jobject thiz, jstring jAddress, jint port)
-{
-    RtpSocket *rtpSocket = getRtpSocket(env, thiz, false);
-    if (rtpSocket == NULL) {
-        // Exception already thrown.
-        return;
-    }
-    sockaddr_storage ss;
-    if (parse(env, jAddress, port, &ss) < 0) {
-        // Exception already thrown.
-        return;
-    }
-    if (rtpSocket->mFamily != ss.ss_family) {
-        throwSocketException(env, EAFNOSUPPORT);
-        return;
-    }
-    rtpSocket->mRemote = ss;
-}
-
-static void release(JNIEnv *env, jobject thiz)
-{
-    delete (RtpSocket *)env->GetIntField(thiz, gNative);
-}
-
-//------------------------------------------------------------------------------
-
-static JNINativeMethod gMethods[] = {
-    {"create", "(Ljava/lang/String;)I", (void *)create},
-    {"associate", "(Ljava/lang/String;I)V", (void *)associate},
-    {"release", "()V", (void *)release},
-};
-
-int registerRtpSocket(JNIEnv *env)
-{
-    jclass clazz;
-    if ((clazz = env->FindClass("com/android/sip/rtp/RtpSocket")) == NULL ||
-        (gNative = env->GetFieldID(clazz, "mNative", "I")) == NULL ||
-        env->RegisterNatives(clazz, gMethods, NELEM(gMethods)) < 0) {
-        LOGE("JNI registration failed");
-        return -1;
-    }
-    return 0;
-}
diff --git a/src/jni/rtp/RtpSocket.h b/src/jni/rtp/RtpSocket.h
deleted file mode 100644
index af1e558..0000000
--- a/src/jni/rtp/RtpSocket.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef __RTP_SOCKET_H__
-#define __RTP_SOCKET_H__
-
-#include "jni.h"
-
-struct RtpSocket;
-
-// Returns NULL and throws an exception if an error occurs.
-RtpSocket *getRtpSocket(JNIEnv *env, jobject jRtpSocket, bool associated);
-
-// Returns the number of bytes sent or -1 if an error occurs. The error code
-// can be found in errno.
-int send(RtpSocket *rtpSocket, void *buffer, int length);
-
-// Returns the REAL LENGTH of the packet received, 0 if deadline is reached,
-// or -1 if an error occurs. The error code can be found in errno.
-int receive(RtpSocket *rtpSocket, void *buffer, int length, timeval *deadline);
-
-#endif
diff --git a/src/jni/rtp/libsiprtp.cpp b/src/jni/rtp/libsiprtp.cpp
deleted file mode 100644
index 46b7f98..0000000
--- a/src/jni/rtp/libsiprtp.cpp
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-
-#include "jni.h"
-
-extern int registerRtpSocket(JNIEnv *env);
-extern int registerAudioStream(JNIEnv *env);
-
-__attribute__((visibility("default")))
-jint JNI_OnLoad(JavaVM *vm, void *reserved)
-{
-    JNIEnv *env = NULL;
-    if (vm->GetEnv((void **)&env, JNI_VERSION_1_4) != JNI_OK ||
-        registerRtpSocket(env) < 0 || registerAudioStream(env) < 0) {
-        return -1;
-    }
-    return JNI_VERSION_1_4;
-}
diff --git a/src/jni/rtp_jni/Android.mk b/src/jni/rtp_jni/Android.mk
deleted file mode 100644
index 46ac19d..0000000
--- a/src/jni/rtp_jni/Android.mk
+++ /dev/null
@@ -1,44 +0,0 @@
-#
-# 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.
-#
-
-LOCAL_PATH := $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := librtp_jni
-
-LOCAL_SRC_FILES := \
-	AudioCodec.cpp \
-	AudioGroup.cpp \
-	RtpStream.cpp \
-	util.cpp \
-	rtp_jni.cpp
-
-LOCAL_SHARED_LIBRARIES := \
-	libnativehelper \
-	libcutils \
-	libutils \
-	libmedia
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_C_INCLUDES += \
-	$(JNI_H_INCLUDE)
-
-LOCAL_CFLAGS += -fvisibility=hidden
-
-LOCAL_PRELINK_MODULE := false
-
-# include $(BUILD_SHARED_LIBRARY)
diff --git a/src/jni/rtp_jni/AudioCodec.cpp b/src/jni/rtp_jni/AudioCodec.cpp
deleted file mode 100644
index ddd07fc..0000000
--- a/src/jni/rtp_jni/AudioCodec.cpp
+++ /dev/null
@@ -1,161 +0,0 @@
-/*
- * Copyrightm (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.
- */
-
-#include <string.h>
-
-#include "AudioCodec.h"
-
-namespace {
-
-int8_t gExponents[128] = {
-    0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4,
-    5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
-    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-};
-
-//------------------------------------------------------------------------------
-
-class UlawCodec : public AudioCodec
-{
-public:
-    bool set(int sampleRate, int sampleCount) {
-        mSampleCount = sampleCount;
-        return sampleCount > 0;
-    }
-    int encode(void *payload, int16_t *samples);
-    int decode(int16_t *samples, void *payload, int length);
-private:
-    int mSampleCount;
-};
-
-int UlawCodec::encode(void *payload, int16_t *samples)
-{
-    int8_t *ulaws = (int8_t *)payload;
-    for (int i = 0; i < mSampleCount; ++i) {
-        int sample = samples[i];
-        int sign = (sample >> 8) & 0x80;
-        if (sample < 0) {
-            sample = -sample;
-        }
-        sample += 132;
-        if (sample > 32767) {
-            sample = 32767;
-        }
-        int exponent = gExponents[sample >> 8];
-        int mantissa = (sample >> (exponent + 3)) & 0x0F;
-        ulaws[i] = ~(sign | (exponent << 4) | mantissa);
-    }
-    return mSampleCount;
-}
-
-int UlawCodec::decode(int16_t *samples, void *payload, int length)
-{
-    int8_t *ulaws = (int8_t *)payload;
-    for (int i = 0; i < length; ++i) {
-        int ulaw = ~ulaws[i];
-        int exponent = (ulaw >> 4) & 0x07;
-        int mantissa = ulaw & 0x0F;
-        int sample = (((mantissa << 3) + 132) << exponent) - 132;
-        samples[i] = (ulaw < 0 ? -sample : sample);
-    }
-    return length;
-}
-
-AudioCodec *newUlawCodec()
-{
-    return new UlawCodec;
-}
-
-//------------------------------------------------------------------------------
-
-class AlawCodec : public AudioCodec
-{
-public:
-    bool set(int sampleRate, int sampleCount) {
-        mSampleCount = sampleCount;
-        return sampleCount > 0;
-    }
-    int encode(void *payload, int16_t *samples);
-    int decode(int16_t *samples, void *payload, int length);
-private:
-    int mSampleCount;
-};
-
-int AlawCodec::encode(void *payload, int16_t *samples)
-{
-    int8_t *alaws = (int8_t *)payload;
-    for (int i = 0; i < mSampleCount; ++i) {
-        int sample = samples[i];
-        int sign = (sample >> 8) & 0x80;
-        if (sample < 0) {
-            sample = -sample;
-        }
-        if (sample > 32767) {
-            sample = 32767;
-        }
-        int exponent = gExponents[sample >> 8];
-        int mantissa = (sample >> (exponent == 0 ? 4 : exponent + 3)) & 0x0F;
-        alaws[i] = (sign | (exponent << 4) | mantissa) ^ 0xD5;
-    }
-    return mSampleCount;
-}
-
-int AlawCodec::decode(int16_t *samples, void *payload, int length)
-{
-    int8_t *alaws = (int8_t *)payload;
-    for (int i = 0; i < length; ++i) {
-        int alaw = alaws[i] ^ 0x55;
-        int exponent = (alaw >> 4) & 0x07;
-        int mantissa = alaw & 0x0F;
-        int sample = (exponent == 0 ? (mantissa << 4) + 8 :
-            ((mantissa << 3) + 132) << exponent);
-        samples[i] = (alaw < 0 ? sample : -sample);
-    }
-    return length;
-}
-
-AudioCodec *newAlawCodec()
-{
-    return new AlawCodec;
-}
-
-struct AudioCodecType {
-    const char *name;
-    AudioCodec *(*create)();
-} gAudioCodecTypes[] = {
-    {"PCMA", newAlawCodec},
-    {"PCMU", newUlawCodec},
-    {NULL, NULL},
-};
-
-} // namespace
-
-AudioCodec *newAudioCodec(const char *codecName)
-{
-    AudioCodecType *type = gAudioCodecTypes;
-    while (type->name != NULL) {
-        if (strcmp(codecName, type->name) == 0) {
-            return type->create();
-        }
-        ++type;
-    }
-    return NULL;
-}
diff --git a/src/jni/rtp_jni/AudioCodec.h b/src/jni/rtp_jni/AudioCodec.h
deleted file mode 100644
index 797494c..0000000
--- a/src/jni/rtp_jni/AudioCodec.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyrightm (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.
- */
-
-#include <stdint.h>
-
-#ifndef __AUDIO_CODEC_H__
-#define __AUDIO_CODEC_H__
-
-class AudioCodec
-{
-public:
-    virtual ~AudioCodec() {}
-    // Returns true if initialization succeeds.
-    virtual bool set(int sampleRate, int sampleCount) = 0;
-    // Returns the length of payload in bytes.
-    virtual int encode(void *payload, int16_t *samples) = 0;
-    // Returns the number of decoded samples.
-    virtual int decode(int16_t *samples, void *payload, int length) = 0;
-};
-
-AudioCodec *newAudioCodec(const char *codecName);
-
-#endif
diff --git a/src/jni/rtp_jni/AudioGroup.cpp b/src/jni/rtp_jni/AudioGroup.cpp
deleted file mode 100644
index fc1ed9b..0000000
--- a/src/jni/rtp_jni/AudioGroup.cpp
+++ /dev/null
@@ -1,1004 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#include <string.h>
-#include <errno.h>
-#include <fcntl.h>
-#include <sys/epoll.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <sys/stat.h>
-#include <sys/time.h>
-#include <time.h>
-#include <arpa/inet.h>
-#include <netinet/in.h>
-
-#define LOG_TAG "AudioGroup"
-#include <cutils/atomic.h>
-#include <utils/Log.h>
-#include <utils/Errors.h>
-#include <utils/RefBase.h>
-#include <utils/threads.h>
-#include <utils/SystemClock.h>
-#include <media/AudioSystem.h>
-#include <media/AudioRecord.h>
-#include <media/AudioTrack.h>
-#include <media/mediarecorder.h>
-
-#include "jni.h"
-#include "JNIHelp.h"
-
-#include "AudioCodec.h"
-
-extern int parse(JNIEnv *env, jstring jAddress, int port, sockaddr_storage *ss);
-
-namespace {
-
-using namespace android;
-
-int gRandom = -1;
-
-// We use a circular array to implement jitter buffer. The simplest way is doing
-// a modulo operation on the index while accessing the array. However modulo can
-// be expensive on some platforms, such as ARM. Thus we round up the size of the
-// array to the nearest power of 2 and then use bitwise-and instead of modulo.
-// Currently we make it 256ms long and assume packet interval is 32ms or less.
-// The first 64ms is the place where samples get mixed. The rest 192ms is the
-// real jitter buffer. For a stream at 8000Hz it takes 4096 bytes. These numbers
-// are chosen by experiments and each of them can be adjusted as needed.
-
-// Other notes:
-// + We use elapsedRealtime() to get the time. Since we use 32bit variables
-//   instead of 64bit ones, comparison must be done by subtraction.
-// + Sampling rate must be multiple of 1000Hz, and packet length must be in
-//   milliseconds. No floating points.
-// + If we cannot get enough CPU, we drop samples and simulate packet loss.
-// + Resampling is not done yet, so streams in one group must use the same rate.
-//   For the first release we might only support 8kHz and 16kHz.
-
-class AudioStream
-{
-public:
-    AudioStream();
-    ~AudioStream();
-    bool set(int mode, int socket, sockaddr_storage *remote,
-        const char *codecName, int sampleRate, int sampleCount,
-        int codecType, int dtmfType);
-
-    void sendDtmf(int event);
-    bool mix(int32_t *output, int head, int tail, int sampleRate);
-    void encode(int tick, AudioStream *chain);
-    void decode(int tick);
-
-private:
-    enum {
-        NORMAL = 0,
-        SEND_ONLY = 1,
-        RECEIVE_ONLY = 2,
-        LAST_MODE = 2,
-    };
-
-    int mMode;
-    int mSocket;
-    sockaddr_storage mRemote;
-    AudioCodec *mCodec;
-    uint32_t mCodecMagic;
-    uint32_t mDtmfMagic;
-
-    int mTick;
-    int mSampleRate;
-    int mSampleCount;
-    int mInterval;
-
-    int16_t *mBuffer;
-    int mBufferMask;
-    int mBufferHead;
-    int mBufferTail;
-    int mLatencyScore;
-
-    uint16_t mSequence;
-    uint32_t mTimestamp;
-    uint32_t mSsrc;
-
-    int mDtmfEvent;
-    int mDtmfStart;
-
-    AudioStream *mNext;
-
-    friend class AudioGroup;
-};
-
-AudioStream::AudioStream()
-{
-    mSocket = -1;
-    mCodec = NULL;
-    mBuffer = NULL;
-    mNext = NULL;
-}
-
-AudioStream::~AudioStream()
-{
-    close(mSocket);
-    delete mCodec;
-    delete [] mBuffer;
-    LOGD("stream[%d] is dead", mSocket);
-}
-
-bool AudioStream::set(int mode, int socket, sockaddr_storage *remote,
-    const char *codecName, int sampleRate, int sampleCount,
-    int codecType, int dtmfType)
-{
-    if (mode < 0 || mode > LAST_MODE) {
-        return false;
-    }
-    mMode = mode;
-
-    if (codecName) {
-        mRemote = *remote;
-        mCodec = newAudioCodec(codecName);
-        if (!mCodec || !mCodec->set(sampleRate, sampleCount)) {
-            return false;
-        }
-    }
-
-    mCodecMagic = (0x8000 | codecType) << 16;
-    mDtmfMagic = (dtmfType == -1) ? 0 : (0x8000 | dtmfType) << 16;
-
-    mTick = elapsedRealtime();
-    mSampleRate = sampleRate / 1000;
-    mSampleCount = sampleCount;
-    mInterval = mSampleCount / mSampleRate;
-
-    // Allocate jitter buffer.
-    for (mBufferMask = 8192; mBufferMask < sampleRate; mBufferMask <<= 1);
-    mBufferMask >>= 2;
-    mBuffer = new int16_t[mBufferMask];
-    --mBufferMask;
-    mBufferHead = 0;
-    mBufferTail = 0;
-    mLatencyScore = 0;
-
-    // Initialize random bits.
-    read(gRandom, &mSequence, sizeof(mSequence));
-    read(gRandom, &mTimestamp, sizeof(mTimestamp));
-    read(gRandom, &mSsrc, sizeof(mSsrc));
-
-    mDtmfEvent = -1;
-    mDtmfStart = 0;
-
-    // Only take over the socket when succeeded.
-    mSocket = socket;
-
-    LOGD("stream[%d] is configured as %s %dkHz %dms", mSocket,
-        (codecName ? codecName : "RAW"), mSampleRate, mInterval);
-    return true;
-}
-
-void AudioStream::sendDtmf(int event)
-{
-    if (mDtmfMagic != 0) {
-        mDtmfEvent = event << 24;
-        mDtmfStart = mTimestamp + mSampleCount;
-    }
-}
-
-bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate)
-{
-    if (mMode == SEND_ONLY) {
-        return false;
-    }
-
-    if (head - mBufferHead < 0) {
-        head = mBufferHead;
-    }
-    if (tail - mBufferTail > 0) {
-        tail = mBufferTail;
-    }
-    if (tail - head <= 0) {
-        return false;
-    }
-
-    head *= mSampleRate;
-    tail *= mSampleRate;
-
-    if (sampleRate == mSampleRate) {
-        for (int i = head; i - tail < 0; ++i) {
-            output[i - head] += mBuffer[i & mBufferMask];
-        }
-    } else {
-        // TODO: implement resampling.
-        return false;
-    }
-    return true;
-}
-
-void AudioStream::encode(int tick, AudioStream *chain)
-{
-    if (tick - mTick >= mInterval) {
-        // We just missed the train. Pretend that packets in between are lost.
-        int skipped = (tick - mTick) / mInterval;
-        mTick += skipped * mInterval;
-        mSequence += skipped;
-        mTimestamp += skipped * mSampleCount;
-        LOGD("stream[%d] skips %d packets", mSocket, skipped);
-    }
-
-    tick = mTick;
-    mTick += mInterval;
-    ++mSequence;
-    mTimestamp += mSampleCount;
-
-    if (mMode == RECEIVE_ONLY) {
-        return;
-    }
-
-    // If there is an ongoing DTMF event, send it now.
-    if (mDtmfEvent != -1) {
-        int duration = mTimestamp - mDtmfStart;
-        // Make sure duration is reasonable.
-        if (duration >= 0 && duration < mSampleRate * 100) {
-            duration += mSampleCount;
-            int32_t buffer[4] = {
-                htonl(mDtmfMagic | mSequence),
-                htonl(mDtmfStart),
-                mSsrc,
-                htonl(mDtmfEvent | duration),
-            };
-            if (duration >= mSampleRate * 100) {
-                buffer[3] |= htonl(1 << 23);
-                mDtmfEvent = -1;
-            }
-            sendto(mSocket, buffer, sizeof(buffer), MSG_DONTWAIT,
-                (sockaddr *)&mRemote, sizeof(mRemote));
-            return;
-        }
-        mDtmfEvent = -1;
-    }
-
-    // It is time to mix streams.
-    bool mixed = false;
-    int32_t buffer[mSampleCount + 3];
-    memset(buffer, 0, sizeof(buffer));
-    while (chain) {
-        if (chain != this &&
-            chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
-            mixed = true;
-        }
-        chain = chain->mNext;
-    }
-    if (!mixed) {
-        LOGD("stream[%d] no data", mSocket);
-        return;
-    }
-
-    // Cook the packet and send it out.
-    int16_t samples[mSampleCount];
-    for (int i = 0; i < mSampleCount; ++i) {
-        int32_t sample = buffer[i];
-        if (sample < -32768) {
-            sample = -32768;
-        }
-        if (sample > 32767) {
-            sample = 32767;
-        }
-        samples[i] = sample;
-    }
-    if (!mCodec) {
-        // Special case for device stream.
-        send(mSocket, samples, sizeof(samples), MSG_DONTWAIT);
-        return;
-    }
-
-    buffer[0] = htonl(mCodecMagic | mSequence);
-    buffer[1] = htonl(mTimestamp);
-    buffer[2] = mSsrc;
-    int length = mCodec->encode(&buffer[3], samples);
-    if (length <= 0) {
-        LOGD("stream[%d] encoder error", mSocket);
-        return;
-    }
-    sendto(mSocket, buffer, length + 12, MSG_DONTWAIT, (sockaddr *)&mRemote,
-        sizeof(mRemote));
-}
-
-void AudioStream::decode(int tick)
-{
-    char c;
-    if (mMode == SEND_ONLY) {
-        recv(mSocket, &c, 1, MSG_DONTWAIT);
-        return;
-    }
-
-    // Make sure mBufferHead and mBufferTail are reasonable.
-    if ((unsigned int)(tick + 256 - mBufferHead) > 1024) {
-        mBufferHead = tick - 64;
-        mBufferTail = mBufferHead;
-    }
-
-    if (tick - mBufferHead > 64) {
-        // Throw away outdated samples.
-        mBufferHead = tick - 64;
-        if (mBufferTail - mBufferHead < 0) {
-            mBufferTail = mBufferHead;
-        }
-    }
-
-    if (mBufferTail - tick <= 80) {
-        mLatencyScore = tick;
-    } else if (tick - mLatencyScore >= 5000) {
-        // Reset the jitter buffer to 40ms if the latency keeps larger than 80ms
-        // in the past 5s. This rarely happens, so let us just keep it simple.
-        LOGD("stream[%d] latency control", mSocket);
-        mBufferTail = tick + 40;
-    }
-
-    if (mBufferTail - mBufferHead > 256 - mInterval) {
-        // Buffer overflow. Drop the packet.
-        LOGD("stream[%d] buffer overflow", mSocket);
-        recv(mSocket, &c, 1, MSG_DONTWAIT);
-        return;
-    }
-
-    // Receive the packet and decode it.
-    int16_t samples[mSampleCount];
-    int length = 0;
-    if (!mCodec) {
-        // Special case for device stream.
-        length = recv(mSocket, samples, sizeof(samples),
-            MSG_TRUNC | MSG_DONTWAIT) >> 1;
-    } else {
-        __attribute__((aligned(4))) uint8_t buffer[2048];
-        length = recv(mSocket, buffer, sizeof(buffer),
-            MSG_TRUNC | MSG_DONTWAIT);
-
-        // Do we need to check SSRC, sequence, and timestamp? They are not
-        // reliable but at least they can be used to identity duplicates?
-        if (length < 12 || length > (int)sizeof(buffer) ||
-            (ntohl(*(uint32_t *)buffer) & 0xC07F0000) != mCodecMagic) {
-            LOGD("stream[%d] malformed packet", mSocket);
-            return;
-        }
-        int offset = 12 + ((buffer[0] & 0x0F) << 2);
-        if ((buffer[0] & 0x10) != 0) {
-            offset += 4 + (ntohs(*(uint16_t *)&buffer[offset + 2]) << 2);
-        }
-        if ((buffer[0] & 0x20) != 0) {
-            length -= buffer[length - 1];
-        }
-        length -= offset;
-        if (length >= 0) {
-            length = mCodec->decode(samples, &buffer[offset], length);
-        }
-    }
-    if (length != mSampleCount) {
-        LOGD("stream[%d] decoder error", mSocket);
-        return;
-    }
-
-    if (tick - mBufferTail > 0) {
-        // Buffer underrun. Reset the jitter buffer to 40ms.
-        LOGD("stream[%d] buffer underrun", mSocket);
-        if (mBufferTail - mBufferHead <= 0) {
-            mBufferHead = tick + 40;
-            mBufferTail = mBufferHead;
-        } else {
-            int tail = (tick + 40) * mSampleRate;
-            for (int i = mBufferTail * mSampleRate; i - tail < 0; ++i) {
-                mBuffer[i & mBufferMask] = 0;
-            }
-            mBufferTail = tick + 40;
-        }
-    }
-
-    // Append to the jitter buffer.
-    int tail = mBufferTail * mSampleRate;
-    for (int i = 0; i < mSampleCount; ++i) {
-        mBuffer[tail & mBufferMask] = samples[i];
-        ++tail;
-    }
-    mBufferTail += mInterval;
-}
-
-//------------------------------------------------------------------------------
-
-class AudioGroup
-{
-public:
-    AudioGroup();
-    ~AudioGroup();
-    bool set(int sampleRate, int sampleCount);
-
-    bool setMode(int mode);
-    bool sendDtmf(int event);
-    bool add(AudioStream *stream);
-    bool remove(int socket);
-
-private:
-    enum {
-        ON_HOLD = 0,
-        MUTED = 1,
-        NORMAL = 2,
-        EC_ENABLED = 3,
-        LAST_MODE = 3,
-    };
-    int mMode;
-    AudioStream *mChain;
-    int mEventQueue;
-    volatile int mDtmfEvent;
-
-    int mSampleCount;
-    int mDeviceSocket;
-    AudioTrack mTrack;
-    AudioRecord mRecord;
-
-    bool networkLoop();
-    bool deviceLoop();
-
-    class NetworkThread : public Thread
-    {
-    public:
-        NetworkThread(AudioGroup *group) : Thread(false), mGroup(group) {}
-
-        bool start()
-        {
-            if (run("Network", ANDROID_PRIORITY_AUDIO) != NO_ERROR) {
-                LOGE("cannot start network thread");
-                return false;
-            }
-            return true;
-        }
-
-    private:
-        AudioGroup *mGroup;
-        bool threadLoop()
-        {
-            return mGroup->networkLoop();
-        }
-    };
-    sp<NetworkThread> mNetworkThread;
-
-    class DeviceThread : public Thread
-    {
-    public:
-        DeviceThread(AudioGroup *group) : Thread(false), mGroup(group) {}
-
-        bool start()
-        {
-            char c;
-            while (recv(mGroup->mDeviceSocket, &c, 1, MSG_DONTWAIT) == 1);
-
-            if (run("Device", ANDROID_PRIORITY_AUDIO) != NO_ERROR) {
-                LOGE("cannot start device thread");
-                return false;
-            }
-            return true;
-        }
-
-    private:
-        AudioGroup *mGroup;
-        bool threadLoop()
-        {
-            return mGroup->deviceLoop();
-        }
-    };
-    sp<DeviceThread> mDeviceThread;
-};
-
-AudioGroup::AudioGroup()
-{
-    mMode = ON_HOLD;
-    mChain = NULL;
-    mEventQueue = -1;
-    mDtmfEvent = -1;
-    mDeviceSocket = -1;
-    mNetworkThread = new NetworkThread(this);
-    mDeviceThread = new DeviceThread(this);
-}
-
-AudioGroup::~AudioGroup()
-{
-    mNetworkThread->requestExitAndWait();
-    mDeviceThread->requestExitAndWait();
-    mTrack.stop();
-    mRecord.stop();
-    close(mEventQueue);
-    close(mDeviceSocket);
-    while (mChain) {
-        AudioStream *next = mChain->mNext;
-        delete mChain;
-        mChain = next;
-    }
-    LOGD("group[%d] is dead", mDeviceSocket);
-}
-
-#define FROYO_COMPATIBLE
-#ifdef FROYO_COMPATIBLE
-
-// Copied from AudioRecord.cpp.
-status_t AudioRecord_getMinFrameCount(
-        int* frameCount,
-        uint32_t sampleRate,
-        int format,
-        int channelCount)
-{
-    size_t size = 0;
-    if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size)
-            != NO_ERROR) {
-        LOGE("AudioSystem could not query the input buffer size.");
-        return NO_INIT;
-    }
-
-    if (size == 0) {
-        LOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d",
-            sampleRate, format, channelCount);
-        return BAD_VALUE;
-    }
-
-    // We double the size of input buffer for ping pong use of record buffer.
-    size <<= 1;
-
-    if (AudioSystem::isLinearPCM(format)) {
-        size /= channelCount * (format == AudioSystem::PCM_16_BIT ? 2 : 1);
-    }
-
-    *frameCount = size;
-    return NO_ERROR;
-}
-
-// Copied from AudioTrack.cpp.
-status_t AudioTrack_getMinFrameCount(
-        int* frameCount,
-        int streamType,
-        uint32_t sampleRate)
-{
-    int afSampleRate;
-    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
-        return NO_INIT;
-    }
-    int afFrameCount;
-    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
-        return NO_INIT;
-    }
-    uint32_t afLatency;
-    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
-        return NO_INIT;
-    }
-
-    // Ensure that buffer depth covers at least audio hardware latency
-    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
-    if (minBufCount < 2) minBufCount = 2;
-
-    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
-              afFrameCount * minBufCount * sampleRate / afSampleRate;
-    return NO_ERROR;
-}
-
-#endif
-
-bool AudioGroup::set(int sampleRate, int sampleCount)
-{
-    mEventQueue = epoll_create(2);
-    if (mEventQueue == -1) {
-        LOGE("epoll_create: %s", strerror(errno));
-        return false;
-    }
-
-    mSampleCount = sampleCount;
-
-    // Find out the frame count for AudioTrack and AudioRecord.
-    int output = 0;
-    int input = 0;
-#ifdef FROYO_COMPATIBLE
-    if (AudioTrack_getMinFrameCount(&output, AudioSystem::VOICE_CALL,
-        sampleRate) != NO_ERROR || output <= 0 ||
-        AudioRecord_getMinFrameCount(&input, sampleRate,
-        AudioSystem::PCM_16_BIT, 1) != NO_ERROR || input <= 0) {
-        LOGE("cannot compute frame count");
-        return false;
-    }
-#else
-    if (AudioTrack::getMinFrameCount(&output, AudioSystem::VOICE_CALL,
-        sampleRate) != NO_ERROR || output <= 0 ||
-        AudioRecord::getMinFrameCount(&input, sampleRate,
-        AudioSystem::PCM_16_BIT, 1) != NO_ERROR || input <= 0) {
-        LOGE("cannot compute frame count");
-        return false;
-    }
-#endif
-    LOGD("reported frame count: output %d, input %d", output, input);
-
-    output = (output + sampleCount - 1) / sampleCount * sampleCount;
-    input = (input + sampleCount - 1) / sampleCount * sampleCount;
-    if (input < output * 2) {
-        input = output * 2;
-    }
-    LOGD("adjusted frame count: output %d, input %d", output, input);
-
-    // Initialize AudioTrack and AudioRecord.
-    if (mTrack.set(AudioSystem::VOICE_CALL, sampleRate, AudioSystem::PCM_16_BIT,
-        AudioSystem::CHANNEL_OUT_MONO, output) != NO_ERROR ||
-        mRecord.set(AUDIO_SOURCE_MIC, sampleRate, AudioSystem::PCM_16_BIT,
-        AudioSystem::CHANNEL_IN_MONO, input) != NO_ERROR) {
-        LOGE("cannot initialize audio device");
-        return false;
-    }
-    LOGD("latency: output %d, input %d", mTrack.latency(), mRecord.latency());
-
-    // TODO: initialize echo canceler here.
-
-    // Create device socket.
-    int pair[2];
-    if (socketpair(AF_UNIX, SOCK_DGRAM, 0, pair)) {
-        LOGE("socketpair: %s", strerror(errno));
-        return false;
-    }
-    mDeviceSocket = pair[0];
-
-    // Create device stream.
-    mChain = new AudioStream;
-    if (!mChain->set(AudioStream::NORMAL, pair[1], NULL, NULL,
-        sampleRate, sampleCount, -1, -1)) {
-        close(pair[1]);
-        LOGE("cannot initialize device stream");
-        return false;
-    }
-
-    // Give device socket a reasonable timeout and buffer size.
-    timeval tv;
-    tv.tv_sec = 0;
-    tv.tv_usec = 1000 * sampleCount / sampleRate * 500;
-    if (setsockopt(pair[0], SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv)) ||
-        setsockopt(pair[0], SOL_SOCKET, SO_RCVBUF, &output, sizeof(output)) ||
-        setsockopt(pair[1], SOL_SOCKET, SO_SNDBUF, &output, sizeof(output))) {
-        LOGE("setsockopt: %s", strerror(errno));
-        return false;
-    }
-
-    // Add device stream into event queue.
-    epoll_event event;
-    event.events = EPOLLIN;
-    event.data.ptr = mChain;
-    if (epoll_ctl(mEventQueue, EPOLL_CTL_ADD, pair[1], &event)) {
-        LOGE("epoll_ctl: %s", strerror(errno));
-        return false;
-    }
-
-    // Anything else?
-    LOGD("stream[%d] joins group[%d]", pair[1], pair[0]);
-    return true;
-}
-
-bool AudioGroup::setMode(int mode)
-{
-    if (mode < 0 || mode > LAST_MODE) {
-        return false;
-    }
-    if (mMode == mode) {
-        return true;
-    }
-
-    LOGD("group[%d] switches from mode %d to %d", mDeviceSocket, mMode, mode);
-    mMode = mode;
-
-    mDeviceThread->requestExitAndWait();
-    if (mode == ON_HOLD) {
-        mTrack.stop();
-        mRecord.stop();
-        return true;
-    }
-
-    mTrack.start();
-    if (mode == MUTED) {
-        mRecord.stop();
-    } else {
-        mRecord.start();
-    }
-
-    if (!mDeviceThread->start()) {
-        mTrack.stop();
-        mRecord.stop();
-        return false;
-    }
-    return true;
-}
-
-bool AudioGroup::sendDtmf(int event)
-{
-    if (event < 0 || event > 15) {
-        return false;
-    }
-
-    // DTMF is rarely used, so we try to make it as lightweight as possible.
-    // Using volatile might be dodgy, but using a pipe or pthread primitives
-    // or stop-set-restart threads seems too heavy. Will investigate later.
-    timespec ts;
-    ts.tv_sec = 0;
-    ts.tv_nsec = 100000000;
-    for (int i = 0; mDtmfEvent != -1 && i < 20; ++i) {
-        nanosleep(&ts, NULL);
-    }
-    if (mDtmfEvent != -1) {
-        return false;
-    }
-    mDtmfEvent = event;
-    nanosleep(&ts, NULL);
-    return true;
-}
-
-bool AudioGroup::add(AudioStream *stream)
-{
-    mNetworkThread->requestExitAndWait();
-
-    epoll_event event;
-    event.events = EPOLLIN;
-    event.data.ptr = stream;
-    if (epoll_ctl(mEventQueue, EPOLL_CTL_ADD, stream->mSocket, &event)) {
-        LOGE("epoll_ctl: %s", strerror(errno));
-        return false;
-    }
-
-    stream->mNext = mChain->mNext;
-    mChain->mNext = stream;
-    if (!mNetworkThread->start()) {
-        // Only take over the stream when succeeded.
-        mChain->mNext = stream->mNext;
-        return false;
-    }
-
-    LOGD("stream[%d] joins group[%d]", stream->mSocket, mDeviceSocket);
-    return true;
-}
-
-bool AudioGroup::remove(int socket)
-{
-    mNetworkThread->requestExitAndWait();
-
-    for (AudioStream *stream = mChain; stream->mNext; stream = stream->mNext) {
-        AudioStream *target = stream->mNext;
-        if (target->mSocket == socket) {
-            stream->mNext = target->mNext;
-            LOGD("stream[%d] leaves group[%d]", socket, mDeviceSocket);
-            delete target;
-            break;
-        }
-    }
-
-    // Do not start network thread if there is only one stream.
-    if (!mChain->mNext || !mNetworkThread->start()) {
-        return false;
-    }
-    return true;
-}
-
-bool AudioGroup::networkLoop()
-{
-    int tick = elapsedRealtime();
-    int deadline = tick + 10;
-    int count = 0;
-
-    for (AudioStream *stream = mChain; stream; stream = stream->mNext) {
-        if (!stream->mTick || tick - stream->mTick >= 0) {
-            stream->encode(tick, mChain);
-        }
-        if (deadline - stream->mTick > 0) {
-            deadline = stream->mTick;
-        }
-        ++count;
-    }
-
-    if (mDtmfEvent != -1) {
-        int event = mDtmfEvent;
-        for (AudioStream *stream = mChain; stream; stream = stream->mNext) {
-            stream->sendDtmf(event);
-        }
-        mDtmfEvent = -1;
-    }
-
-    deadline -= tick;
-    if (deadline < 1) {
-        deadline = 1;
-    }
-
-    epoll_event events[count];
-    count = epoll_wait(mEventQueue, events, count, deadline);
-    if (count == -1) {
-        LOGE("epoll_wait: %s", strerror(errno));
-        return false;
-    }
-    for (int i = 0; i < count; ++i) {
-        ((AudioStream *)events[i].data.ptr)->decode(tick);
-    }
-
-    return true;
-}
-
-bool AudioGroup::deviceLoop()
-{
-    int16_t output[mSampleCount];
-
-    if (recv(mDeviceSocket, output, sizeof(output), 0) <= 0) {
-        memset(output, 0, sizeof(output));
-    }
-    if (mTrack.write(output, sizeof(output)) != (int)sizeof(output)) {
-        LOGE("cannot write to AudioTrack");
-        return false;
-    }
-
-    if (mMode != MUTED) {
-        uint32_t frameCount = mRecord.frameCount();
-        AudioRecord::Buffer input;
-        input.frameCount = frameCount;
-
-        if (mRecord.obtainBuffer(&input, -1) != NO_ERROR) {
-            LOGE("cannot read from AudioRecord");
-            return false;
-        }
-
-        if (input.frameCount < (uint32_t)mSampleCount) {
-            input.frameCount = 0;
-        } else {
-            if (mMode == NORMAL) {
-                send(mDeviceSocket, input.i8, sizeof(output), MSG_DONTWAIT);
-            } else {
-                // TODO: Echo canceller runs here.
-                send(mDeviceSocket, input.i8, sizeof(output), MSG_DONTWAIT);
-            }
-            if (input.frameCount < frameCount) {
-                input.frameCount = mSampleCount;
-            }
-        }
-
-        mRecord.releaseBuffer(&input);
-    }
-
-    return true;
-}
-
-//------------------------------------------------------------------------------
-
-static jfieldID gNative;
-static jfieldID gMode;
-
-jint add(JNIEnv *env, jobject thiz, jint mode,
-    jint socket, jstring jRemoteAddress, jint remotePort,
-    jstring jCodecName, jint sampleRate, jint sampleCount,
-    jint codecType, jint dtmfType)
-{
-    const char *codecName = NULL;
-    AudioStream *stream = NULL;
-    AudioGroup *group = NULL;
-
-    // Sanity check.
-    sockaddr_storage remote;
-    if (parse(env, jRemoteAddress, remotePort, &remote) < 0) {
-        // Exception already thrown.
-        return -1;
-    }
-    if (sampleRate < 0 || sampleCount < 0 || codecType < 0 || codecType > 127) {
-        jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
-        goto error;
-    }
-    if (!jCodecName) {
-        jniThrowNullPointerException(env, "codecName");
-        return -1;
-    }
-    codecName = env->GetStringUTFChars(jCodecName, NULL);
-    if (!codecName) {
-        // Exception already thrown.
-        return -1;
-    }
-
-    // Create audio stream.
-    stream = new AudioStream;
-    if (!stream->set(mode, socket, &remote, codecName, sampleRate, sampleCount,
-        codecType, dtmfType)) {
-        jniThrowException(env, "java/lang/IllegalStateException",
-            "cannot initialize audio stream");
-        goto error;
-    }
-    socket = -1;
-
-    // Create audio group.
-    group = (AudioGroup *)env->GetIntField(thiz, gNative);
-    if (!group) {
-        int mode = env->GetIntField(thiz, gMode);
-        group = new AudioGroup;
-        if (!group->set(8000, 256) || !group->setMode(mode)) {
-            jniThrowException(env, "java/lang/IllegalStateException",
-                "cannot initialize audio group");
-            goto error;
-        }
-    }
-
-    // Add audio stream into audio group.
-    if (!group->add(stream)) {
-        jniThrowException(env, "java/lang/IllegalStateException",
-            "cannot add audio stream");
-        goto error;
-    }
-
-    // Succeed.
-    env->SetIntField(thiz, gNative, (int)group);
-    env->ReleaseStringUTFChars(jCodecName, codecName);
-    return socket;
-
-error:
-    delete group;
-    delete stream;
-    close(socket);
-    env->SetIntField(thiz, gNative, NULL);
-    env->ReleaseStringUTFChars(jCodecName, codecName);
-    return -1;
-}
-
-void remove(JNIEnv *env, jobject thiz, jint socket)
-{
-    AudioGroup *group = (AudioGroup *)env->GetIntField(thiz, gNative);
-    if (group) {
-        if (socket == -1 || !group->remove(socket)) {
-            delete group;
-            env->SetIntField(thiz, gNative, NULL);
-        }
-    }
-}
-
-void setMode(JNIEnv *env, jobject thiz, jint mode)
-{
-    AudioGroup *group = (AudioGroup *)env->GetIntField(thiz, gNative);
-    if (group && !group->setMode(mode)) {
-        jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
-        return;
-    }
-    env->SetIntField(thiz, gMode, mode);
-}
-
-void sendDtmf(JNIEnv *env, jobject thiz, jint event)
-{
-    AudioGroup *group = (AudioGroup *)env->GetIntField(thiz, gNative);
-    if (group && !group->sendDtmf(event)) {
-        jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
-    }
-}
-
-JNINativeMethod gMethods[] = {
-    {"add", "(IILjava/lang/String;ILjava/lang/String;IIII)I", (void *)add},
-    {"remove", "(I)V", (void *)remove},
-    {"setMode", "(I)V", (void *)setMode},
-    {"sendDtmf", "(I)V", (void *)sendDtmf},
-};
-
-} // namespace
-
-int registerAudioGroup(JNIEnv *env)
-{
-    gRandom = open("/dev/urandom", O_RDONLY);
-    if (gRandom == -1) {
-        LOGE("urandom: %s", strerror(errno));
-        return -1;
-    }
-
-    jclass clazz;
-    if ((clazz = env->FindClass("android/net/rtp/AudioGroup")) == NULL ||
-        (gNative = env->GetFieldID(clazz, "mNative", "I")) == NULL ||
-        (gMode = env->GetFieldID(clazz, "mMode", "I")) == NULL ||
-        env->RegisterNatives(clazz, gMethods, NELEM(gMethods)) < 0) {
-        LOGE("JNI registration failed");
-        return -1;
-    }
-    return 0;
-}
diff --git a/src/jni/rtp_jni/RtpStream.cpp b/src/jni/rtp_jni/RtpStream.cpp
deleted file mode 100644
index 33b88e4..0000000
--- a/src/jni/rtp_jni/RtpStream.cpp
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#include <string.h>
-#include <errno.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <arpa/inet.h>
-#include <netinet/in.h>
-
-#define LOG_TAG "RtpStream"
-#include <utils/Log.h>
-
-#include "jni.h"
-#include "JNIHelp.h"
-
-extern int parse(JNIEnv *env, jstring jAddress, int port, sockaddr_storage *ss);
-
-namespace {
-
-jfieldID gNative;
-
-jint create(JNIEnv *env, jobject thiz, jstring jAddress)
-{
-    env->SetIntField(thiz, gNative, -1);
-
-    sockaddr_storage ss;
-    if (parse(env, jAddress, 0, &ss) < 0) {
-        // Exception already thrown.
-        return -1;
-    }
-
-    int socket = ::socket(ss.ss_family, SOCK_DGRAM, 0);
-    socklen_t len = sizeof(ss);
-    if (socket == -1 || bind(socket, (sockaddr *)&ss, sizeof(ss)) != 0 ||
-        getsockname(socket, (sockaddr *)&ss, &len) != 0) {
-        jniThrowException(env, "java/net/SocketException", strerror(errno));
-        ::close(socket);
-        return -1;
-    }
-
-    uint16_t *p = (ss.ss_family == AF_INET) ?
-        &((sockaddr_in *)&ss)->sin_port : &((sockaddr_in6 *)&ss)->sin6_port;
-    uint16_t port = ntohs(*p);
-    if ((port & 1) == 0) {
-        env->SetIntField(thiz, gNative, socket);
-        return port;
-    }
-    ::close(socket);
-
-    socket = ::socket(ss.ss_family, SOCK_DGRAM, 0);
-    if (socket != -1) {
-        uint16_t delta = port << 1;
-        ++port;
-
-        for (int i = 0; i < 1000; ++i) {
-            do {
-                port += delta;
-            } while (port < 1024);
-            *p = htons(port);
-
-            if (bind(socket, (sockaddr *)&ss, sizeof(ss)) == 0) {
-                env->SetIntField(thiz, gNative, socket);
-                return port;
-            }
-        }
-    }
-
-    jniThrowException(env, "java/net/SocketException", strerror(errno));
-    ::close(socket);
-    return -1;
-}
-
-jint dup(JNIEnv *env, jobject thiz)
-{
-    int socket1 = env->GetIntField(thiz, gNative);
-    int socket2 = ::dup(socket1);
-    if (socket2 == -1) {
-        jniThrowException(env, "java/lang/IllegalStateException", strerror(errno));
-    }
-    LOGD("dup %d to %d", socket1, socket2);
-    return socket2;
-}
-
-void close(JNIEnv *env, jobject thiz)
-{
-    int socket = env->GetIntField(thiz, gNative);
-    ::close(socket);
-    env->SetIntField(thiz, gNative, -1);
-    LOGD("close %d", socket);
-}
-
-JNINativeMethod gMethods[] = {
-    {"create", "(Ljava/lang/String;)I", (void *)create},
-    {"dup", "()I", (void *)dup},
-    {"close", "()V", (void *)close},
-};
-
-} // namespace
-
-int registerRtpStream(JNIEnv *env)
-{
-    jclass clazz;
-    if ((clazz = env->FindClass("android/net/rtp/RtpStream")) == NULL ||
-        (gNative = env->GetFieldID(clazz, "mNative", "I")) == NULL ||
-        env->RegisterNatives(clazz, gMethods, NELEM(gMethods)) < 0) {
-        LOGE("JNI registration failed");
-        return -1;
-    }
-    return 0;
-}
diff --git a/src/jni/rtp_jni/rtp_jni.cpp b/src/jni/rtp_jni/rtp_jni.cpp
deleted file mode 100644
index 9f4bff9..0000000
--- a/src/jni/rtp_jni/rtp_jni.cpp
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-
-#include "jni.h"
-
-extern int registerRtpStream(JNIEnv *env);
-extern int registerAudioGroup(JNIEnv *env);
-
-__attribute__((visibility("default"))) jint JNI_OnLoad(JavaVM *vm, void *unused)
-{
-    JNIEnv *env = NULL;
-    if (vm->GetEnv((void **)&env, JNI_VERSION_1_4) != JNI_OK ||
-        registerRtpStream(env) < 0 || registerAudioGroup(env) < 0) {
-        return -1;
-    }
-    return JNI_VERSION_1_4;
-}
diff --git a/src/jni/rtp_jni/util.cpp b/src/jni/rtp_jni/util.cpp
deleted file mode 100644
index 1d702fc..0000000
--- a/src/jni/rtp_jni/util.cpp
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <arpa/inet.h>
-#include <netinet/in.h>
-
-#include "jni.h"
-#include "JNIHelp.h"
-
-int parse(JNIEnv *env, jstring jAddress, int port, sockaddr_storage *ss)
-{
-    if (!jAddress) {
-        jniThrowNullPointerException(env, "address");
-        return -1;
-    }
-    if (port < 0 || port > 65535) {
-        jniThrowException(env, "java/lang/IllegalArgumentException", "port");
-        return -1;
-    }
-    const char *address = env->GetStringUTFChars(jAddress, NULL);
-    if (!address) {
-        // Exception already thrown.
-        return -1;
-    }
-    memset(ss, 0, sizeof(*ss));
-
-    sockaddr_in *sin = (sockaddr_in *)ss;
-    if (inet_pton(AF_INET, address, &(sin->sin_addr)) > 0) {
-        sin->sin_family = AF_INET;
-        sin->sin_port = htons(port);
-        env->ReleaseStringUTFChars(jAddress, address);
-        return 0;
-    }
-
-    sockaddr_in6 *sin6 = (sockaddr_in6 *)ss;
-    if (inet_pton(AF_INET6, address, &(sin6->sin6_addr)) > 0) {
-        sin6->sin6_family = AF_INET6;
-        sin6->sin6_port = htons(port);
-        env->ReleaseStringUTFChars(jAddress, address);
-        return 0;
-    }
-
-    env->ReleaseStringUTFChars(jAddress, address);
-    jniThrowException(env, "java/lang/IllegalArgumentException", "address");
-    return -1;
-}