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Diffstat (limited to 'voip/java/android/net/rtp/AudioStream.java')
| -rw-r--r-- | voip/java/android/net/rtp/AudioStream.java | 167 |
1 files changed, 0 insertions, 167 deletions
diff --git a/voip/java/android/net/rtp/AudioStream.java b/voip/java/android/net/rtp/AudioStream.java deleted file mode 100644 index 5cd1abc..0000000 --- a/voip/java/android/net/rtp/AudioStream.java +++ /dev/null @@ -1,167 +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; - -/** - * 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 - */ -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; - } -} |