android-34/android/net/rtp/AudioStream.java

/*
 * 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;

/**
 * An AudioStream is a {@link RtpStream} which carrys audio payloads over
 * Real-time Transport Protocol (RTP). Two different classes are developed in
 * order to support various usages such as audio conferencing. An AudioStream
 * represents a remote endpoint which consists of a network mapping and a
 * configured {@link AudioCodec}. On the other side, An {@link AudioGroup}
 * represents a local endpoint which mixes all the AudioStreams and optionally
 * interacts with the speaker and the microphone at the same time. The simplest
 * usage includes one for each endpoints. For other combinations, developers
 * should be aware of the limitations described in {@link AudioGroup}.
 *
 * <p>An AudioStream becomes busy when it joins an AudioGroup. In this case most
 * of the setter methods are disabled. This is designed to ease the task of
 * managing native resources. One can always make an AudioStream leave its
 * AudioGroup by calling {@link #join(AudioGroup)} with {@code null} and put it
 * back after the modification is done.</p>
 *
 * <p class="note">Using this class requires
 * {@link android.Manifest.permission#INTERNET} permission.</p>
 *
 * @see RtpStream
 * @see AudioGroup
 * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
 * should not be used as the basis of future VOIP apps.
 */
public class AudioStream extends RtpStream {
    private AudioCodec mCodec;
    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 has already joined an
     * {@link AudioGroup}.
     */
    @Override
    public final boolean isBusy() {
        return mGroup != null;
    }

    /**
     * Returns the joined {@link AudioGroup}.
     */
    public AudioGroup getGroup() {
        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) {
        synchronized (this) {
            if (mGroup == group) {
                return;
            }
            if (mGroup != null) {
                mGroup.remove(this);
                mGroup = null;
            }
            if (group != null) {
                group.add(this);
                mGroup = group;
            }
        }
    }

    /**
     * Returns the {@link AudioCodec}, or {@code null} if it is not set.
     *
     * @see #setCodec(AudioCodec)
     */
    public AudioCodec getCodec() {
        return mCodec;
    }

    /**
     * Sets the {@link AudioCodec}.
     *
     * @param codec The AudioCodec to be used.
     * @throws IllegalArgumentException if its type is used by DTMF.
     * @throws IllegalStateException if the stream is busy.
     */
    public void setCodec(AudioCodec codec) {
        if (isBusy()) {
            throw new IllegalStateException("Busy");
        }
        if (codec.type == mDtmfType) {
            throw new IllegalArgumentException("The type is used by DTMF");
        }
        mCodec = codec;
    }

    /**
     * Returns the RTP payload type for dual-tone multi-frequency (DTMF) digits,
     * or {@code -1} if it is not enabled.
     *
     * @see #setDtmfType(int)
     */
    public int getDtmfType() {
        return mDtmfType;
    }

    /**
     * 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 type. 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 (mCodec != null && type == mCodec.type) {
                throw new IllegalArgumentException("The type is used by codec");
            }
        }
        mDtmfType = type;
    }
}