Merge changes I7efff4f2,I1631ee20,I14d9886a,Ic2858920,Iff588130,I37ba9d83 into gingerbread (75c3853e) · Commits · e / os / android_frameworks_base

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

+116 −24

Original line number	Diff line number	Diff line
		@@ -16,41 +16,133 @@

		package android.net.rtp;

		/** @hide */
		import java.util.Arrays;

		/**
		* This class defines a collection of audio codecs to be used with
		* {@link AudioStream}s. Their parameters are designed to be exchanged using
		* Session Description Protocol (SDP). Most of the values listed here can be
		* found in RFC 3551, while others are described in separated standards.
		*
		* <p>Few simple configurations are defined as public static instances for the
		* convenience of direct uses. More complicated ones could be obtained using
		* {@link #getCodec(int, String, String)}. For example, one can use the
		* following snippet to create a mode-1-only AMR codec.</p>
		* <pre>
		* AudioCodec codec = AudioCodec.getCodec(100, "AMR/8000", "mode-set=1");
		* </pre>
		*
		* @see AudioStream
		* @hide
		*/
		public class AudioCodec {
		public static final AudioCodec ULAW = new AudioCodec("PCMU", 8000, 160, 0);
		public static final AudioCodec ALAW = new AudioCodec("PCMA", 8000, 160, 8);
		/**
		* The RTP payload type of the encoding.
		*/
		public final int type;

		/**
		* The encoding parameters to be used in the corresponding SDP attribute.
		*/
		public final String rtpmap;

		/**
		* The format parameters to be used in the corresponding SDP attribute.
		*/
		public final String fmtp;

		/**
		* G.711 u-law audio codec.
		*/
		public static final AudioCodec PCMU = new AudioCodec(0, "PCMU/8000", null);

		/**
		* G.711 a-law audio codec.
		*/
		public static final AudioCodec PCMA = new AudioCodec(8, "PCMA/8000", null);

		/**
		* GSM Full-Rate audio codec, also known as GSM-FR, GSM 06.10, GSM, or
		* simply FR.
		*/
		public static final AudioCodec GSM = new AudioCodec(3, "GSM/8000", null);

		/**
		* Returns system supported codecs.
		* GSM Enhanced Full-Rate audio codec, also known as GSM-EFR, GSM 06.60, or
		* simply EFR.
		*/
		public static AudioCodec[] getSystemSupportedCodecs() {
		return new AudioCodec[] {AudioCodec.ULAW, AudioCodec.ALAW};
		public static final AudioCodec GSM_EFR = new AudioCodec(96, "GSM-EFR/8000", null);

		/**
		* Adaptive Multi-Rate narrowband audio codec, also known as AMR or AMR-NB.
		* Currently CRC, robust sorting, and interleaving are not supported. See
		* more details about these features in RFC 4867.
		*/
		public static final AudioCodec AMR = new AudioCodec(97, "AMR/8000", null);

		// TODO: add rest of the codecs when the native part is done.
		private static final AudioCodec[] sCodecs = {PCMU, PCMA};

		private AudioCodec(int type, String rtpmap, String fmtp) {
		this.type = type;
		this.rtpmap = rtpmap;
		this.fmtp = fmtp;
		}

		/**
		* 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
		* Returns system supported audio codecs.
		*/
		public static AudioCodec getSystemSupportedCodec(String name) {
		for (AudioCodec codec : getSystemSupportedCodecs()) {
		if (codec.name.equals(name)) return codec;
		public static AudioCodec[] getCodecs() {
		return Arrays.copyOf(sCodecs, sCodecs.length);
		}

		/**
		* Creates an AudioCodec according to the given configuration.
		*
		* @param type The payload type of the encoding defined in RTP/AVP.
		* @param rtpmap The encoding parameters specified in the corresponding SDP
		* attribute, or null if it is not available.
		* @param fmtp The format parameters specified in the corresponding SDP
		* attribute, or null if it is not available.
		* @return The configured AudioCodec or {@code null} if it is not supported.
		*/
		public static AudioCodec getCodec(int type, String rtpmap, String fmtp) {
		if (type < 0 \|\| type > 127) {
		return null;
		}

		public final String name;
		public final int sampleRate;
		public final int sampleCount;
		public final int defaultType;
		AudioCodec hint = null;
		if (rtpmap != null) {
		String clue = rtpmap.trim().toUpperCase();
		for (AudioCodec codec : sCodecs) {
		if (clue.startsWith(codec.rtpmap)) {
		String channels = clue.substring(codec.rtpmap.length());
		if (channels.length() == 0 \|\| channels.equals("/1")) {
		hint = codec;
		}
		break;
		}
		}
		} else if (type < 96) {
		for (AudioCodec codec : sCodecs) {
		if (type == codec.type) {
		hint = codec;
		rtpmap = codec.rtpmap;
		break;
		}
		}
		}

		private AudioCodec(String name, int sampleRate, int sampleCount, int defaultType) {
		this.name = name;
		this.sampleRate = sampleRate;
		this.sampleCount = sampleCount;
		this.defaultType = defaultType;
		if (hint == null) {
		return null;
		}
		if (hint == AMR && fmtp != null) {
		String clue = fmtp.toLowerCase();
		if (clue.contains("crc=1") \|\| clue.contains("robust-sorting=1") \|\|
		clue.contains("interleaving=")) {
		return null;
		}
		}
		return new AudioCodec(type, rtpmap, fmtp);
		}
		}

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

+80 −11

Original line number	Diff line number	Diff line
		@@ -20,13 +20,63 @@ import java.util.HashMap;
		import java.util.Map;

		/**
		* An AudioGroup acts as a router connected to the speaker, the microphone, and
		* {@link AudioStream}s. Its pipeline has four steps. First, for each
		* AudioStream not in {@link RtpStream#MODE_SEND_ONLY}, decodes its incoming
		* packets and stores in its buffer. Then, if the microphone is enabled,
		* processes the recorded audio and stores in its buffer. Third, if the speaker
		* is enabled, mixes and playbacks buffers of all AudioStreams. Finally, for
		* each AudioStream not in {@link RtpStream#MODE_RECEIVE_ONLY}, mixes all other
		* buffers and sends back the encoded packets. An AudioGroup does nothing if
		* there is no AudioStream in it.
		*
		* <p>Few things must be noticed before using these classes. The performance is
		* highly related to the system load and the network bandwidth. Usually a
		* simpler {@link AudioCodec} costs fewer CPU cycles but requires more network
		* bandwidth, and vise versa. Using two AudioStreams at the same time not only
		* doubles the load but also the bandwidth. The condition varies from one device
		* to another, and developers must choose the right combination in order to get
		* the best result.
		*
		* <p>It is sometimes useful to keep multiple AudioGroups at the same time. For
		* example, a Voice over IP (VoIP) application might want to put a conference
		* call on hold in order to make a new call but still allow people in the
		* previous call to talk to each other. This can be done easily using two
		* AudioGroups, but there are some limitations. Since the speaker and the
		* microphone are shared globally, only one AudioGroup is allowed to run in
		* modes other than {@link #MODE_ON_HOLD}. In addition, before adding an
		* AudioStream into an AudioGroup, one should always put all other AudioGroups
		* into {@link #MODE_ON_HOLD}. That will make sure the audio driver correctly
		* initialized.
		* @hide
		*/
		/** @hide */
		public class AudioGroup {
		/**
		* This mode is similar to {@link #MODE_NORMAL} except the speaker and
		* the microphone are disabled.
		*/
		public static final int MODE_ON_HOLD = 0;

		/**
		* This mode is similar to {@link #MODE_NORMAL} except the microphone is
		* muted.
		*/
		public static final int MODE_MUTED = 1;

		/**
		* This mode indicates that the speaker, the microphone, and all
		* {@link AudioStream}s in the group are enabled. First, the packets
		* received from the streams are decoded and mixed with the audio recorded
		* from the microphone. Then, the results are played back to the speaker,
		* encoded and sent back to each stream.
		*/
		public static final int MODE_NORMAL = 2;
		public static final int MODE_EC_ENABLED = 3;

		/**
		* This mode is similar to {@link #MODE_NORMAL} except the echo suppression
		* is enabled. It should be only used when the speaker phone is on.
		*/
		public static final int MODE_ECHO_SUPPRESSION = 3;

		private final Map<AudioStream, Integer> mStreams;
		private int mMode = MODE_ON_HOLD;
		@@ -36,23 +86,42 @@ public class AudioGroup {
		System.loadLibrary("rtp_jni");
		}

		/**
		* Creates an empty AudioGroup.
		*/
		public AudioGroup() {
		mStreams = new HashMap<AudioStream, Integer>();
		}

		/**
		* Returns the current mode.
		*/
		public int getMode() {
		return mMode;
		}

		/**
		* Changes the current mode. It must be one of {@link #MODE_ON_HOLD},
		* {@link #MODE_MUTED}, {@link #MODE_NORMAL}, and
		* {@link #MODE_ECHO_SUPPRESSION}.
		*
		* @param mode The mode to change to.
		* @throws IllegalArgumentException if the mode is invalid.
		*/
		public synchronized native void setMode(int mode);

		synchronized void add(AudioStream stream, AudioCodec codec, int codecType, int dtmfType) {
		private native void add(int mode, int socket, String remoteAddress,
		int remotePort, String codecSpec, int dtmfType);

		synchronized void add(AudioStream stream, AudioCodec codec, int dtmfType) {
		if (!mStreams.containsKey(stream)) {
		try {
		int socket = stream.dup();
		String codecSpec = String.format("%d %s %s", codec.type,
		codec.rtpmap, codec.fmtp);
		add(stream.getMode(), socket,
		stream.getRemoteAddress().getHostAddress(), stream.getRemotePort(),
		codec.name, codec.sampleRate, codec.sampleCount, codecType, dtmfType);
		stream.getRemoteAddress().getHostAddress(),
		stream.getRemotePort(), codecSpec, dtmfType);
		mStreams.put(stream, socket);
		} catch (NullPointerException e) {
		throw new IllegalStateException(e);
		@@ -60,8 +129,7 @@ public class AudioGroup {
		}
		}

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

		synchronized void remove(AudioStream stream) {
		Integer socket = mStreams.remove(stream);
		@@ -70,8 +138,6 @@ public class AudioGroup {
		}
		}

		private native void remove(int socket);

		/**
		* Sends a DTMF digit to every {@link AudioStream} in this group. Currently
		* only event {@code 0} to {@code 15} are supported.
		@@ -80,13 +146,16 @@ public class AudioGroup {
		*/
		public native synchronized void sendDtmf(int event);

		public synchronized void reset() {
		/**
		* Removes every {@link AudioStream} in this group.
		*/
		public synchronized void clear() {
		remove(-1);
		}

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

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

+47 −21

Original line number	Diff line number	Diff line
		@@ -20,12 +20,27 @@ import java.net.InetAddress;
		import java.net.SocketException;

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

		@@ -42,7 +57,8 @@ public class AudioStream extends RtpStream {
		}

		/**
		* Returns {@code true} if the stream already joined an {@link AudioGroup}.
		* Returns {@code true} if the stream has already joined an
		* {@link AudioGroup}.
		*/
		@Override
		public final boolean isBusy() {
		@@ -52,7 +68,7 @@ public class AudioStream extends RtpStream {
		/**
		* Returns the joined {@link AudioGroup}.
		*/
		public AudioGroup getAudioGroup() {
		public AudioGroup getGroup() {
		return mGroup;
		}

		@@ -74,35 +90,45 @@ public class AudioStream extends RtpStream {
		mGroup = null;
		}
		if (group != null) {
		group.add(this, mCodec, mCodecType, mDtmfType);
		group.add(this, mCodec, 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.
		* 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.
		* @param type The RTP payload type.
		* @throws IllegalArgumentException if the type is invalid or used by DTMF.
		* @throws IllegalArgumentException if its type is used by DTMF.
		* @throws IllegalStateException if the stream is busy.
		*/
		public void setCodec(AudioCodec codec, int type) {
		public void setCodec(AudioCodec codec) {
		if (isBusy()) {
		throw new IllegalStateException("Busy");
		}
		if (type < 0 \|\| type > 127 \|\| (type != codec.defaultType && type < 96)) {
		throw new IllegalArgumentException("Invalid type");
		}
		if (type == mDtmfType) {
		if (codec.type == mDtmfType) {
		throw new IllegalArgumentException("The type is used by DTMF");
		}
		mCodec = codec;
		mCodecType = type;
		}

		/**
		* 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;
		}

		/**
		@@ -111,7 +137,7 @@ public class AudioStream extends RtpStream {
		* 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
		* 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.
		@@ -127,7 +153,7 @@ public class AudioStream extends RtpStream {
		if (type < 96 \|\| type > 127) {
		throw new IllegalArgumentException("Invalid type");
		}
		if (type == mCodecType) {
		if (type == mCodec.type) {
		throw new IllegalArgumentException("The type is used by codec");
		}
		}

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

+21 −7

Original line number	Diff line number	Diff line
		@@ -22,13 +22,25 @@ import java.net.Inet6Address;
		import java.net.SocketException;

		/**
		* RtpStream represents a base class of media streams running over
		* Real-time Transport Protocol (RTP).
		* RtpStream represents the base class of streams which send and receive network
		* packets with media payloads over Real-time Transport Protocol (RTP).
		* @hide
		*/
		/** @hide */
		public class RtpStream {
		/**
		* This mode indicates that the stream sends and receives packets at the
		* same time. This is the initial mode for new streams.
		*/
		public static final int MODE_NORMAL = 0;

		/**
		* This mode indicates that the stream only sends packets.
		*/
		public static final int MODE_SEND_ONLY = 1;

		/**
		* This mode indicates that the stream only receives packets.
		*/
		public static final int MODE_RECEIVE_ONLY = 2;

		private final InetAddress mLocalAddress;
		@@ -89,15 +101,16 @@ public class RtpStream {
		}

		/**
		* Returns {@code true} if the stream is busy. This method is intended to be
		* overridden by subclasses.
		* Returns {@code true} if the stream is busy. In this case most of the
		* setter methods are disabled. 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}.
		* Returns the current mode.
		*/
		public int getMode() {
		return mMode;
		@@ -123,7 +136,8 @@ public class RtpStream {
		}

		/**
		* Associates with a remote host.
		* Associates with a remote host. This defines the destination of the
		* outgoing packets.
		*
		* @param address The network address of the remote host.
		* @param port The network port of the remote host.

voip/java/android/net/sip/SimpleSessionDescription.java

0 → 100644

+612 −0

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