| /* |
| * Copyright (C) 2008 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.media; |
| |
| import java.lang.annotation.Retention; |
| import java.lang.annotation.RetentionPolicy; |
| import java.lang.ref.WeakReference; |
| import java.nio.ByteBuffer; |
| import java.nio.NioUtils; |
| import java.util.Iterator; |
| import java.util.Set; |
| |
| import android.annotation.IntDef; |
| import android.app.ActivityThread; |
| import android.app.AppOpsManager; |
| import android.content.Context; |
| import android.os.Handler; |
| import android.os.IBinder; |
| import android.os.Looper; |
| import android.os.Message; |
| import android.os.Process; |
| import android.os.RemoteException; |
| import android.os.ServiceManager; |
| import android.util.Log; |
| |
| import com.android.internal.app.IAppOpsService; |
| |
| |
| /** |
| * The AudioTrack class manages and plays a single audio resource for Java applications. |
| * It allows streaming of PCM audio buffers to the audio sink for playback. This is |
| * achieved by "pushing" the data to the AudioTrack object using one of the |
| * {@link #write(byte[], int, int)}, {@link #write(short[], int, int)}, |
| * and {@link #write(float[], int, int, int)} methods. |
| * |
| * <p>An AudioTrack instance can operate under two modes: static or streaming.<br> |
| * In Streaming mode, the application writes a continuous stream of data to the AudioTrack, using |
| * one of the {@code write()} methods. These are blocking and return when the data has been |
| * transferred from the Java layer to the native layer and queued for playback. The streaming |
| * mode is most useful when playing blocks of audio data that for instance are: |
| * |
| * <ul> |
| * <li>too big to fit in memory because of the duration of the sound to play,</li> |
| * <li>too big to fit in memory because of the characteristics of the audio data |
| * (high sampling rate, bits per sample ...)</li> |
| * <li>received or generated while previously queued audio is playing.</li> |
| * </ul> |
| * |
| * The static mode should be chosen when dealing with short sounds that fit in memory and |
| * that need to be played with the smallest latency possible. The static mode will |
| * therefore be preferred for UI and game sounds that are played often, and with the |
| * smallest overhead possible. |
| * |
| * <p>Upon creation, an AudioTrack object initializes its associated audio buffer. |
| * The size of this buffer, specified during the construction, determines how long an AudioTrack |
| * can play before running out of data.<br> |
| * For an AudioTrack using the static mode, this size is the maximum size of the sound that can |
| * be played from it.<br> |
| * For the streaming mode, data will be written to the audio sink in chunks of |
| * sizes less than or equal to the total buffer size. |
| * |
| * AudioTrack is not final and thus permits subclasses, but such use is not recommended. |
| */ |
| public class AudioTrack |
| { |
| //--------------------------------------------------------- |
| // Constants |
| //-------------------- |
| /** Minimum value for a linear gain or auxiliary effect level. |
| * This value must be exactly equal to 0.0f; do not change it. |
| */ |
| private static final float GAIN_MIN = 0.0f; |
| /** Maximum value for a linear gain or auxiliary effect level. |
| * This value must be greater than or equal to 1.0f. |
| */ |
| private static final float GAIN_MAX = 1.0f; |
| |
| /** Minimum value for sample rate */ |
| private static final int SAMPLE_RATE_HZ_MIN = 4000; |
| /** Maximum value for sample rate */ |
| private static final int SAMPLE_RATE_HZ_MAX = 96000; |
| |
| /** Maximum value for AudioTrack channel count */ |
| private static final int CHANNEL_COUNT_MAX = 8; |
| |
| /** indicates AudioTrack state is stopped */ |
| public static final int PLAYSTATE_STOPPED = 1; // matches SL_PLAYSTATE_STOPPED |
| /** indicates AudioTrack state is paused */ |
| public static final int PLAYSTATE_PAUSED = 2; // matches SL_PLAYSTATE_PAUSED |
| /** indicates AudioTrack state is playing */ |
| public static final int PLAYSTATE_PLAYING = 3; // matches SL_PLAYSTATE_PLAYING |
| |
| // keep these values in sync with android_media_AudioTrack.cpp |
| /** |
| * Creation mode where audio data is transferred from Java to the native layer |
| * only once before the audio starts playing. |
| */ |
| public static final int MODE_STATIC = 0; |
| /** |
| * Creation mode where audio data is streamed from Java to the native layer |
| * as the audio is playing. |
| */ |
| public static final int MODE_STREAM = 1; |
| |
| /** |
| * State of an AudioTrack that was not successfully initialized upon creation. |
| */ |
| public static final int STATE_UNINITIALIZED = 0; |
| /** |
| * State of an AudioTrack that is ready to be used. |
| */ |
| public static final int STATE_INITIALIZED = 1; |
| /** |
| * State of a successfully initialized AudioTrack that uses static data, |
| * but that hasn't received that data yet. |
| */ |
| public static final int STATE_NO_STATIC_DATA = 2; |
| |
| /** |
| * Denotes a successful operation. |
| */ |
| public static final int SUCCESS = AudioSystem.SUCCESS; |
| /** |
| * Denotes a generic operation failure. |
| */ |
| public static final int ERROR = AudioSystem.ERROR; |
| /** |
| * Denotes a failure due to the use of an invalid value. |
| */ |
| public static final int ERROR_BAD_VALUE = AudioSystem.BAD_VALUE; |
| /** |
| * Denotes a failure due to the improper use of a method. |
| */ |
| public static final int ERROR_INVALID_OPERATION = AudioSystem.INVALID_OPERATION; |
| |
| // Error codes: |
| // to keep in sync with frameworks/base/core/jni/android_media_AudioTrack.cpp |
| private static final int ERROR_NATIVESETUP_AUDIOSYSTEM = -16; |
| private static final int ERROR_NATIVESETUP_INVALIDCHANNELMASK = -17; |
| private static final int ERROR_NATIVESETUP_INVALIDFORMAT = -18; |
| private static final int ERROR_NATIVESETUP_INVALIDSTREAMTYPE = -19; |
| private static final int ERROR_NATIVESETUP_NATIVEINITFAILED = -20; |
| |
| // Events: |
| // to keep in sync with frameworks/av/include/media/AudioTrack.h |
| /** |
| * Event id denotes when playback head has reached a previously set marker. |
| */ |
| private static final int NATIVE_EVENT_MARKER = 3; |
| /** |
| * Event id denotes when previously set update period has elapsed during playback. |
| */ |
| private static final int NATIVE_EVENT_NEW_POS = 4; |
| |
| private final static String TAG = "android.media.AudioTrack"; |
| |
| |
| /** @hide */ |
| @IntDef({ |
| WRITE_BLOCKING, |
| WRITE_NON_BLOCKING |
| }) |
| @Retention(RetentionPolicy.SOURCE) |
| public @interface WriteMode {} |
| |
| /** |
| * The write mode indicating the write operation will block until all data has been written, |
| * to be used in {@link #write(ByteBuffer, int, int)} |
| */ |
| public final static int WRITE_BLOCKING = 0; |
| /** |
| * The write mode indicating the write operation will return immediately after |
| * queuing as much audio data for playback as possible without blocking, to be used in |
| * {@link #write(ByteBuffer, int, int)}. |
| */ |
| public final static int WRITE_NON_BLOCKING = 1; |
| |
| //-------------------------------------------------------------------------- |
| // Member variables |
| //-------------------- |
| /** |
| * Indicates the state of the AudioTrack instance. |
| */ |
| private int mState = STATE_UNINITIALIZED; |
| /** |
| * Indicates the play state of the AudioTrack instance. |
| */ |
| private int mPlayState = PLAYSTATE_STOPPED; |
| /** |
| * Lock to make sure mPlayState updates are reflecting the actual state of the object. |
| */ |
| private final Object mPlayStateLock = new Object(); |
| /** |
| * Sizes of the native audio buffer. |
| */ |
| private int mNativeBufferSizeInBytes = 0; |
| private int mNativeBufferSizeInFrames = 0; |
| /** |
| * Handler for events coming from the native code. |
| */ |
| private NativeEventHandlerDelegate mEventHandlerDelegate; |
| /** |
| * Looper associated with the thread that creates the AudioTrack instance. |
| */ |
| private final Looper mInitializationLooper; |
| /** |
| * The audio data source sampling rate in Hz. |
| */ |
| private int mSampleRate; // initialized by all constructors |
| /** |
| * The number of audio output channels (1 is mono, 2 is stereo). |
| */ |
| private int mChannelCount = 1; |
| /** |
| * The audio channel mask. |
| */ |
| private int mChannels = AudioFormat.CHANNEL_OUT_MONO; |
| |
| /** |
| * The type of the audio stream to play. See |
| * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, |
| * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, |
| * {@link AudioManager#STREAM_ALARM}, {@link AudioManager#STREAM_NOTIFICATION}, and |
| * {@link AudioManager#STREAM_DTMF}. |
| */ |
| private int mStreamType = AudioManager.STREAM_MUSIC; |
| |
| private final AudioAttributes mAttributes; |
| /** |
| * The way audio is consumed by the audio sink, streaming or static. |
| */ |
| private int mDataLoadMode = MODE_STREAM; |
| /** |
| * The current audio channel configuration. |
| */ |
| private int mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO; |
| /** |
| * The encoding of the audio samples. |
| * @see AudioFormat#ENCODING_PCM_8BIT |
| * @see AudioFormat#ENCODING_PCM_16BIT |
| * @see AudioFormat#ENCODING_PCM_FLOAT |
| */ |
| private int mAudioFormat = AudioFormat.ENCODING_PCM_16BIT; |
| /** |
| * Audio session ID |
| */ |
| private int mSessionId = AudioSystem.AUDIO_SESSION_ALLOCATE; |
| /** |
| * Reference to the app-ops service. |
| */ |
| private final IAppOpsService mAppOps; |
| |
| //-------------------------------- |
| // Used exclusively by native code |
| //-------------------- |
| /** |
| * Accessed by native methods: provides access to C++ AudioTrack object. |
| */ |
| @SuppressWarnings("unused") |
| private long mNativeTrackInJavaObj; |
| /** |
| * Accessed by native methods: provides access to the JNI data (i.e. resources used by |
| * the native AudioTrack object, but not stored in it). |
| */ |
| @SuppressWarnings("unused") |
| private long mJniData; |
| |
| |
| //-------------------------------------------------------------------------- |
| // Constructor, Finalize |
| //-------------------- |
| /** |
| * Class constructor. |
| * @param streamType the type of the audio stream. See |
| * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, |
| * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, |
| * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. |
| * @param sampleRateInHz the initial source sample rate expressed in Hz. |
| * @param channelConfig describes the configuration of the audio channels. |
| * See {@link AudioFormat#CHANNEL_OUT_MONO} and |
| * {@link AudioFormat#CHANNEL_OUT_STEREO} |
| * @param audioFormat the format in which the audio data is represented. |
| * See {@link AudioFormat#ENCODING_PCM_16BIT}, |
| * {@link AudioFormat#ENCODING_PCM_8BIT}, |
| * and {@link AudioFormat#ENCODING_PCM_FLOAT}. |
| * @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is |
| * read from for playback. |
| * If track's creation mode is {@link #MODE_STREAM}, you can write data into |
| * this buffer in chunks less than or equal to this size, and it is typical to use |
| * chunks of 1/2 of the total size to permit double-buffering. |
| * If the track's creation mode is {@link #MODE_STATIC}, |
| * this is the maximum length sample, or audio clip, that can be played by this instance. |
| * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size |
| * for the successful creation of an AudioTrack instance in streaming mode. Using values |
| * smaller than getMinBufferSize() will result in an initialization failure. |
| * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} |
| * @throws java.lang.IllegalArgumentException |
| */ |
| public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, |
| int bufferSizeInBytes, int mode) |
| throws IllegalArgumentException { |
| this(streamType, sampleRateInHz, channelConfig, audioFormat, |
| bufferSizeInBytes, mode, AudioSystem.AUDIO_SESSION_ALLOCATE); |
| } |
| |
| /** |
| * Class constructor with audio session. Use this constructor when the AudioTrack must be |
| * attached to a particular audio session. The primary use of the audio session ID is to |
| * associate audio effects to a particular instance of AudioTrack: if an audio session ID |
| * is provided when creating an AudioEffect, this effect will be applied only to audio tracks |
| * and media players in the same session and not to the output mix. |
| * When an AudioTrack is created without specifying a session, it will create its own session |
| * which can be retrieved by calling the {@link #getAudioSessionId()} method. |
| * If a non-zero session ID is provided, this AudioTrack will share effects attached to this |
| * session |
| * with all other media players or audio tracks in the same session, otherwise a new session |
| * will be created for this track if none is supplied. |
| * @param streamType the type of the audio stream. See |
| * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, |
| * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, |
| * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. |
| * @param sampleRateInHz the initial source sample rate expressed in Hz. |
| * @param channelConfig describes the configuration of the audio channels. |
| * See {@link AudioFormat#CHANNEL_OUT_MONO} and |
| * {@link AudioFormat#CHANNEL_OUT_STEREO} |
| * @param audioFormat the format in which the audio data is represented. |
| * See {@link AudioFormat#ENCODING_PCM_16BIT} and |
| * {@link AudioFormat#ENCODING_PCM_8BIT}, |
| * and {@link AudioFormat#ENCODING_PCM_FLOAT}. |
| * @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read |
| * from for playback. If using the AudioTrack in streaming mode, you can write data into |
| * this buffer in smaller chunks than this size. If using the AudioTrack in static mode, |
| * this is the maximum size of the sound that will be played for this instance. |
| * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size |
| * for the successful creation of an AudioTrack instance in streaming mode. Using values |
| * smaller than getMinBufferSize() will result in an initialization failure. |
| * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} |
| * @param sessionId Id of audio session the AudioTrack must be attached to |
| * @throws java.lang.IllegalArgumentException |
| */ |
| public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, |
| int bufferSizeInBytes, int mode, int sessionId) |
| throws IllegalArgumentException { |
| // mState already == STATE_UNINITIALIZED |
| this((new AudioAttributes.Builder()) |
| .setLegacyStreamType(streamType) |
| .build(), |
| (new AudioFormat.Builder()) |
| .setChannelMask(channelConfig) |
| .setEncoding(audioFormat) |
| .setSampleRate(sampleRateInHz) |
| .build(), |
| bufferSizeInBytes, |
| mode, sessionId); |
| } |
| |
| /** |
| * Class constructor with {@link AudioAttributes} and {@link AudioFormat}. |
| * @param attributes a non-null {@link AudioAttributes} instance. |
| * @param format a non-null {@link AudioFormat} instance describing the format of the data |
| * that will be played through this AudioTrack. See {@link AudioFormat.Builder} for |
| * configuring the audio format parameters such as encoding, channel mask and sample rate. |
| * @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read |
| * from for playback. If using the AudioTrack in streaming mode, you can write data into |
| * this buffer in smaller chunks than this size. If using the AudioTrack in static mode, |
| * this is the maximum size of the sound that will be played for this instance. |
| * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size |
| * for the successful creation of an AudioTrack instance in streaming mode. Using values |
| * smaller than getMinBufferSize() will result in an initialization failure. |
| * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}. |
| * @param sessionId ID of audio session the AudioTrack must be attached to, or |
| * {@link AudioManager#AUDIO_SESSION_ID_GENERATE} if the session isn't known at construction |
| * time. See also {@link AudioManager#generateAudioSessionId()} to obtain a session ID before |
| * construction. |
| * @throws IllegalArgumentException |
| */ |
| public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes, |
| int mode, int sessionId) |
| throws IllegalArgumentException { |
| // mState already == STATE_UNINITIALIZED |
| |
| if (attributes == null) { |
| throw new IllegalArgumentException("Illegal null AudioAttributes"); |
| } |
| if (format == null) { |
| throw new IllegalArgumentException("Illegal null AudioFormat"); |
| } |
| |
| // remember which looper is associated with the AudioTrack instantiation |
| Looper looper; |
| if ((looper = Looper.myLooper()) == null) { |
| looper = Looper.getMainLooper(); |
| } |
| |
| int rate = 0; |
| if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE) != 0) |
| { |
| rate = format.getSampleRate(); |
| } else { |
| rate = AudioSystem.getPrimaryOutputSamplingRate(); |
| if (rate <= 0) { |
| rate = 44100; |
| } |
| } |
| int channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT; |
| if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) |
| { |
| channelMask = format.getChannelMask(); |
| } |
| int encoding = AudioFormat.ENCODING_DEFAULT; |
| if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) { |
| encoding = format.getEncoding(); |
| } |
| audioParamCheck(rate, channelMask, encoding, mode); |
| mStreamType = AudioSystem.STREAM_DEFAULT; |
| |
| audioBuffSizeCheck(bufferSizeInBytes); |
| |
| mInitializationLooper = looper; |
| IBinder b = ServiceManager.getService(Context.APP_OPS_SERVICE); |
| mAppOps = IAppOpsService.Stub.asInterface(b); |
| |
| mAttributes = (new AudioAttributes.Builder(attributes).build()); |
| |
| if (sessionId < 0) { |
| throw new IllegalArgumentException("Invalid audio session ID: "+sessionId); |
| } |
| |
| int[] session = new int[1]; |
| session[0] = sessionId; |
| // native initialization |
| int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes, |
| mSampleRate, mChannels, mAudioFormat, |
| mNativeBufferSizeInBytes, mDataLoadMode, session); |
| if (initResult != SUCCESS) { |
| loge("Error code "+initResult+" when initializing AudioTrack."); |
| return; // with mState == STATE_UNINITIALIZED |
| } |
| |
| mSessionId = session[0]; |
| |
| if (mDataLoadMode == MODE_STATIC) { |
| mState = STATE_NO_STATIC_DATA; |
| } else { |
| mState = STATE_INITIALIZED; |
| } |
| } |
| |
| // mask of all the channels supported by this implementation |
| private static final int SUPPORTED_OUT_CHANNELS = |
| AudioFormat.CHANNEL_OUT_FRONT_LEFT | |
| AudioFormat.CHANNEL_OUT_FRONT_RIGHT | |
| AudioFormat.CHANNEL_OUT_FRONT_CENTER | |
| AudioFormat.CHANNEL_OUT_LOW_FREQUENCY | |
| AudioFormat.CHANNEL_OUT_BACK_LEFT | |
| AudioFormat.CHANNEL_OUT_BACK_RIGHT | |
| AudioFormat.CHANNEL_OUT_BACK_CENTER | |
| AudioFormat.CHANNEL_OUT_SIDE_LEFT | |
| AudioFormat.CHANNEL_OUT_SIDE_RIGHT; |
| |
| // Convenience method for the constructor's parameter checks. |
| // This is where constructor IllegalArgumentException-s are thrown |
| // postconditions: |
| // mChannelCount is valid |
| // mChannels is valid |
| // mAudioFormat is valid |
| // mSampleRate is valid |
| // mDataLoadMode is valid |
| private void audioParamCheck(int sampleRateInHz, |
| int channelConfig, int audioFormat, int mode) { |
| //-------------- |
| // sample rate, note these values are subject to change |
| if (sampleRateInHz < SAMPLE_RATE_HZ_MIN || sampleRateInHz > SAMPLE_RATE_HZ_MAX) { |
| throw new IllegalArgumentException(sampleRateInHz |
| + "Hz is not a supported sample rate."); |
| } |
| mSampleRate = sampleRateInHz; |
| |
| //-------------- |
| // channel config |
| mChannelConfiguration = channelConfig; |
| |
| switch (channelConfig) { |
| case AudioFormat.CHANNEL_OUT_DEFAULT: //AudioFormat.CHANNEL_CONFIGURATION_DEFAULT |
| case AudioFormat.CHANNEL_OUT_MONO: |
| case AudioFormat.CHANNEL_CONFIGURATION_MONO: |
| mChannelCount = 1; |
| mChannels = AudioFormat.CHANNEL_OUT_MONO; |
| break; |
| case AudioFormat.CHANNEL_OUT_STEREO: |
| case AudioFormat.CHANNEL_CONFIGURATION_STEREO: |
| mChannelCount = 2; |
| mChannels = AudioFormat.CHANNEL_OUT_STEREO; |
| break; |
| default: |
| if (!isMultichannelConfigSupported(channelConfig)) { |
| // input channel configuration features unsupported channels |
| throw new IllegalArgumentException("Unsupported channel configuration."); |
| } |
| mChannels = channelConfig; |
| mChannelCount = Integer.bitCount(channelConfig); |
| } |
| |
| //-------------- |
| // audio format |
| if (audioFormat == AudioFormat.ENCODING_DEFAULT) { |
| audioFormat = AudioFormat.ENCODING_PCM_16BIT; |
| } |
| |
| if (!AudioFormat.isValidEncoding(audioFormat)) { |
| throw new IllegalArgumentException("Unsupported audio encoding."); |
| } |
| mAudioFormat = audioFormat; |
| |
| //-------------- |
| // audio load mode |
| if (((mode != MODE_STREAM) && (mode != MODE_STATIC)) || |
| ((mode != MODE_STREAM) && !AudioFormat.isEncodingLinearPcm(mAudioFormat))) { |
| throw new IllegalArgumentException("Invalid mode."); |
| } |
| mDataLoadMode = mode; |
| } |
| |
| /** |
| * Convenience method to check that the channel configuration (a.k.a channel mask) is supported |
| * @param channelConfig the mask to validate |
| * @return false if the AudioTrack can't be used with such a mask |
| */ |
| private static boolean isMultichannelConfigSupported(int channelConfig) { |
| // check for unsupported channels |
| if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { |
| loge("Channel configuration features unsupported channels"); |
| return false; |
| } |
| final int channelCount = Integer.bitCount(channelConfig); |
| if (channelCount > CHANNEL_COUNT_MAX) { |
| loge("Channel configuration contains too many channels " + |
| channelCount + ">" + CHANNEL_COUNT_MAX); |
| return false; |
| } |
| // check for unsupported multichannel combinations: |
| // - FL/FR must be present |
| // - L/R channels must be paired (e.g. no single L channel) |
| final int frontPair = |
| AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT; |
| if ((channelConfig & frontPair) != frontPair) { |
| loge("Front channels must be present in multichannel configurations"); |
| return false; |
| } |
| final int backPair = |
| AudioFormat.CHANNEL_OUT_BACK_LEFT | AudioFormat.CHANNEL_OUT_BACK_RIGHT; |
| if ((channelConfig & backPair) != 0) { |
| if ((channelConfig & backPair) != backPair) { |
| loge("Rear channels can't be used independently"); |
| return false; |
| } |
| } |
| final int sidePair = |
| AudioFormat.CHANNEL_OUT_SIDE_LEFT | AudioFormat.CHANNEL_OUT_SIDE_RIGHT; |
| if ((channelConfig & sidePair) != 0 |
| && (channelConfig & sidePair) != sidePair) { |
| loge("Side channels can't be used independently"); |
| return false; |
| } |
| return true; |
| } |
| |
| |
| // Convenience method for the constructor's audio buffer size check. |
| // preconditions: |
| // mChannelCount is valid |
| // mAudioFormat is valid |
| // postcondition: |
| // mNativeBufferSizeInBytes is valid (multiple of frame size, positive) |
| private void audioBuffSizeCheck(int audioBufferSize) { |
| // NB: this section is only valid with PCM data. |
| // To update when supporting compressed formats |
| int frameSizeInBytes; |
| if (AudioFormat.isEncodingLinearPcm(mAudioFormat)) { |
| frameSizeInBytes = mChannelCount |
| * (AudioFormat.getBytesPerSample(mAudioFormat)); |
| } else { |
| frameSizeInBytes = 1; |
| } |
| if ((audioBufferSize % frameSizeInBytes != 0) || (audioBufferSize < 1)) { |
| throw new IllegalArgumentException("Invalid audio buffer size."); |
| } |
| |
| mNativeBufferSizeInBytes = audioBufferSize; |
| mNativeBufferSizeInFrames = audioBufferSize / frameSizeInBytes; |
| } |
| |
| |
| /** |
| * Releases the native AudioTrack resources. |
| */ |
| public void release() { |
| // even though native_release() stops the native AudioTrack, we need to stop |
| // AudioTrack subclasses too. |
| try { |
| stop(); |
| } catch(IllegalStateException ise) { |
| // don't raise an exception, we're releasing the resources. |
| } |
| native_release(); |
| mState = STATE_UNINITIALIZED; |
| } |
| |
| @Override |
| protected void finalize() { |
| native_finalize(); |
| } |
| |
| //-------------------------------------------------------------------------- |
| // Getters |
| //-------------------- |
| /** |
| * Returns the minimum gain value, which is the constant 0.0. |
| * Gain values less than 0.0 will be clamped to 0.0. |
| * <p>The word "volume" in the API name is historical; this is actually a linear gain. |
| * @return the minimum value, which is the constant 0.0. |
| */ |
| static public float getMinVolume() { |
| return GAIN_MIN; |
| } |
| |
| /** |
| * Returns the maximum gain value, which is greater than or equal to 1.0. |
| * Gain values greater than the maximum will be clamped to the maximum. |
| * <p>The word "volume" in the API name is historical; this is actually a gain. |
| * expressed as a linear multiplier on sample values, where a maximum value of 1.0 |
| * corresponds to a gain of 0 dB (sample values left unmodified). |
| * @return the maximum value, which is greater than or equal to 1.0. |
| */ |
| static public float getMaxVolume() { |
| return GAIN_MAX; |
| } |
| |
| /** |
| * Returns the configured audio data sample rate in Hz |
| */ |
| public int getSampleRate() { |
| return mSampleRate; |
| } |
| |
| /** |
| * Returns the current playback rate in Hz. |
| */ |
| public int getPlaybackRate() { |
| return native_get_playback_rate(); |
| } |
| |
| /** |
| * Returns the configured audio data format. See {@link AudioFormat#ENCODING_PCM_16BIT} |
| * and {@link AudioFormat#ENCODING_PCM_8BIT}. |
| */ |
| public int getAudioFormat() { |
| return mAudioFormat; |
| } |
| |
| /** |
| * Returns the type of audio stream this AudioTrack is configured for. |
| * Compare the result against {@link AudioManager#STREAM_VOICE_CALL}, |
| * {@link AudioManager#STREAM_SYSTEM}, {@link AudioManager#STREAM_RING}, |
| * {@link AudioManager#STREAM_MUSIC}, {@link AudioManager#STREAM_ALARM}, |
| * {@link AudioManager#STREAM_NOTIFICATION}, or {@link AudioManager#STREAM_DTMF}. |
| */ |
| public int getStreamType() { |
| return mStreamType; |
| } |
| |
| /** |
| * Returns the configured channel configuration. |
| * See {@link AudioFormat#CHANNEL_OUT_MONO} |
| * and {@link AudioFormat#CHANNEL_OUT_STEREO}. |
| */ |
| public int getChannelConfiguration() { |
| return mChannelConfiguration; |
| } |
| |
| /** |
| * Returns the configured number of channels. |
| */ |
| public int getChannelCount() { |
| return mChannelCount; |
| } |
| |
| /** |
| * Returns the state of the AudioTrack instance. This is useful after the |
| * AudioTrack instance has been created to check if it was initialized |
| * properly. This ensures that the appropriate resources have been acquired. |
| * @see #STATE_INITIALIZED |
| * @see #STATE_NO_STATIC_DATA |
| * @see #STATE_UNINITIALIZED |
| */ |
| public int getState() { |
| return mState; |
| } |
| |
| /** |
| * Returns the playback state of the AudioTrack instance. |
| * @see #PLAYSTATE_STOPPED |
| * @see #PLAYSTATE_PAUSED |
| * @see #PLAYSTATE_PLAYING |
| */ |
| public int getPlayState() { |
| synchronized (mPlayStateLock) { |
| return mPlayState; |
| } |
| } |
| |
| /** |
| * Returns the "native frame count", derived from the bufferSizeInBytes specified at |
| * creation time and converted to frame units. |
| * If track's creation mode is {@link #MODE_STATIC}, |
| * it is equal to the specified bufferSizeInBytes converted to frame units. |
| * If track's creation mode is {@link #MODE_STREAM}, |
| * it is typically greater than or equal to the specified bufferSizeInBytes converted to frame |
| * units; it may be rounded up to a larger value if needed by the target device implementation. |
| * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. |
| * See {@link AudioManager#getProperty(String)} for key |
| * {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}. |
| */ |
| @Deprecated |
| protected int getNativeFrameCount() { |
| return native_get_native_frame_count(); |
| } |
| |
| /** |
| * Returns marker position expressed in frames. |
| * @return marker position in wrapping frame units similar to {@link #getPlaybackHeadPosition}, |
| * or zero if marker is disabled. |
| */ |
| public int getNotificationMarkerPosition() { |
| return native_get_marker_pos(); |
| } |
| |
| /** |
| * Returns the notification update period expressed in frames. |
| * Zero means that no position update notifications are being delivered. |
| */ |
| public int getPositionNotificationPeriod() { |
| return native_get_pos_update_period(); |
| } |
| |
| /** |
| * Returns the playback head position expressed in frames. |
| * Though the "int" type is signed 32-bits, the value should be reinterpreted as if it is |
| * unsigned 32-bits. That is, the next position after 0x7FFFFFFF is (int) 0x80000000. |
| * This is a continuously advancing counter. It will wrap (overflow) periodically, |
| * for example approximately once every 27:03:11 hours:minutes:seconds at 44.1 kHz. |
| * It is reset to zero by flush(), reload(), and stop(). |
| */ |
| public int getPlaybackHeadPosition() { |
| return native_get_position(); |
| } |
| |
| /** |
| * Returns this track's estimated latency in milliseconds. This includes the latency due |
| * to AudioTrack buffer size, AudioMixer (if any) and audio hardware driver. |
| * |
| * DO NOT UNHIDE. The existing approach for doing A/V sync has too many problems. We need |
| * a better solution. |
| * @hide |
| */ |
| public int getLatency() { |
| return native_get_latency(); |
| } |
| |
| /** |
| * Returns the output sample rate in Hz for the specified stream type. |
| */ |
| static public int getNativeOutputSampleRate(int streamType) { |
| return native_get_output_sample_rate(streamType); |
| } |
| |
| /** |
| * Returns the minimum buffer size required for the successful creation of an AudioTrack |
| * object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't |
| * guarantee a smooth playback under load, and higher values should be chosen according to |
| * the expected frequency at which the buffer will be refilled with additional data to play. |
| * For example, if you intend to dynamically set the source sample rate of an AudioTrack |
| * to a higher value than the initial source sample rate, be sure to configure the buffer size |
| * based on the highest planned sample rate. |
| * @param sampleRateInHz the source sample rate expressed in Hz. |
| * @param channelConfig describes the configuration of the audio channels. |
| * See {@link AudioFormat#CHANNEL_OUT_MONO} and |
| * {@link AudioFormat#CHANNEL_OUT_STEREO} |
| * @param audioFormat the format in which the audio data is represented. |
| * See {@link AudioFormat#ENCODING_PCM_16BIT} and |
| * {@link AudioFormat#ENCODING_PCM_8BIT}, |
| * and {@link AudioFormat#ENCODING_PCM_FLOAT}. |
| * @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed, |
| * or {@link #ERROR} if unable to query for output properties, |
| * or the minimum buffer size expressed in bytes. |
| */ |
| static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) { |
| int channelCount = 0; |
| switch(channelConfig) { |
| case AudioFormat.CHANNEL_OUT_MONO: |
| case AudioFormat.CHANNEL_CONFIGURATION_MONO: |
| channelCount = 1; |
| break; |
| case AudioFormat.CHANNEL_OUT_STEREO: |
| case AudioFormat.CHANNEL_CONFIGURATION_STEREO: |
| channelCount = 2; |
| break; |
| default: |
| if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { |
| // input channel configuration features unsupported channels |
| loge("getMinBufferSize(): Invalid channel configuration."); |
| return ERROR_BAD_VALUE; |
| } else { |
| channelCount = Integer.bitCount(channelConfig); |
| } |
| } |
| |
| if (!AudioFormat.isValidEncoding(audioFormat)) { |
| loge("getMinBufferSize(): Invalid audio format."); |
| return ERROR_BAD_VALUE; |
| } |
| |
| // sample rate, note these values are subject to change |
| if ( (sampleRateInHz < SAMPLE_RATE_HZ_MIN) || (sampleRateInHz > SAMPLE_RATE_HZ_MAX) ) { |
| loge("getMinBufferSize(): " + sampleRateInHz + " Hz is not a supported sample rate."); |
| return ERROR_BAD_VALUE; |
| } |
| |
| int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat); |
| if (size <= 0) { |
| loge("getMinBufferSize(): error querying hardware"); |
| return ERROR; |
| } |
| else { |
| return size; |
| } |
| } |
| |
| /** |
| * Returns the audio session ID. |
| * |
| * @return the ID of the audio session this AudioTrack belongs to. |
| */ |
| public int getAudioSessionId() { |
| return mSessionId; |
| } |
| |
| /** |
| * Poll for a timestamp on demand. |
| * <p> |
| * If you need to track timestamps during initial warmup or after a routing or mode change, |
| * you should request a new timestamp once per second until the reported timestamps |
| * show that the audio clock is stable. |
| * Thereafter, query for a new timestamp approximately once every 10 seconds to once per minute. |
| * Calling this method more often is inefficient. |
| * It is also counter-productive to call this method more often than recommended, |
| * because the short-term differences between successive timestamp reports are not meaningful. |
| * If you need a high-resolution mapping between frame position and presentation time, |
| * consider implementing that at application level, based on low-resolution timestamps. |
| * <p> |
| * The audio data at the returned position may either already have been |
| * presented, or may have not yet been presented but is committed to be presented. |
| * It is not possible to request the time corresponding to a particular position, |
| * or to request the (fractional) position corresponding to a particular time. |
| * If you need such features, consider implementing them at application level. |
| * |
| * @param timestamp a reference to a non-null AudioTimestamp instance allocated |
| * and owned by caller. |
| * @return true if a timestamp is available, or false if no timestamp is available. |
| * If a timestamp if available, |
| * the AudioTimestamp instance is filled in with a position in frame units, together |
| * with the estimated time when that frame was presented or is committed to |
| * be presented. |
| * In the case that no timestamp is available, any supplied instance is left unaltered. |
| * A timestamp may be temporarily unavailable while the audio clock is stabilizing, |
| * or during and immediately after a route change. |
| */ |
| // Add this text when the "on new timestamp" API is added: |
| // Use if you need to get the most recent timestamp outside of the event callback handler. |
| public boolean getTimestamp(AudioTimestamp timestamp) |
| { |
| if (timestamp == null) { |
| throw new IllegalArgumentException(); |
| } |
| // It's unfortunate, but we have to either create garbage every time or use synchronized |
| long[] longArray = new long[2]; |
| int ret = native_get_timestamp(longArray); |
| if (ret != SUCCESS) { |
| return false; |
| } |
| timestamp.framePosition = longArray[0]; |
| timestamp.nanoTime = longArray[1]; |
| return true; |
| } |
| |
| |
| //-------------------------------------------------------------------------- |
| // Initialization / configuration |
| //-------------------- |
| /** |
| * Sets the listener the AudioTrack notifies when a previously set marker is reached or |
| * for each periodic playback head position update. |
| * Notifications will be received in the same thread as the one in which the AudioTrack |
| * instance was created. |
| * @param listener |
| */ |
| public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener) { |
| setPlaybackPositionUpdateListener(listener, null); |
| } |
| |
| /** |
| * Sets the listener the AudioTrack notifies when a previously set marker is reached or |
| * for each periodic playback head position update. |
| * Use this method to receive AudioTrack events in the Handler associated with another |
| * thread than the one in which you created the AudioTrack instance. |
| * @param listener |
| * @param handler the Handler that will receive the event notification messages. |
| */ |
| public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener, |
| Handler handler) { |
| if (listener != null) { |
| mEventHandlerDelegate = new NativeEventHandlerDelegate(this, listener, handler); |
| } else { |
| mEventHandlerDelegate = null; |
| } |
| } |
| |
| |
| private static float clampGainOrLevel(float gainOrLevel) { |
| if (Float.isNaN(gainOrLevel)) { |
| throw new IllegalArgumentException(); |
| } |
| if (gainOrLevel < GAIN_MIN) { |
| gainOrLevel = GAIN_MIN; |
| } else if (gainOrLevel > GAIN_MAX) { |
| gainOrLevel = GAIN_MAX; |
| } |
| return gainOrLevel; |
| } |
| |
| |
| /** |
| * Sets the specified left and right output gain values on the AudioTrack. |
| * <p>Gain values are clamped to the closed interval [0.0, max] where |
| * max is the value of {@link #getMaxVolume}. |
| * A value of 0.0 results in zero gain (silence), and |
| * a value of 1.0 means unity gain (signal unchanged). |
| * The default value is 1.0 meaning unity gain. |
| * <p>The word "volume" in the API name is historical; this is actually a linear gain. |
| * @param leftGain output gain for the left channel. |
| * @param rightGain output gain for the right channel |
| * @return error code or success, see {@link #SUCCESS}, |
| * {@link #ERROR_INVALID_OPERATION} |
| * @deprecated Applications should use {@link #setVolume} instead, as it |
| * more gracefully scales down to mono, and up to multi-channel content beyond stereo. |
| */ |
| public int setStereoVolume(float leftGain, float rightGain) { |
| if (isRestricted()) { |
| return SUCCESS; |
| } |
| if (mState == STATE_UNINITIALIZED) { |
| return ERROR_INVALID_OPERATION; |
| } |
| |
| leftGain = clampGainOrLevel(leftGain); |
| rightGain = clampGainOrLevel(rightGain); |
| |
| native_setVolume(leftGain, rightGain); |
| |
| return SUCCESS; |
| } |
| |
| |
| /** |
| * Sets the specified output gain value on all channels of this track. |
| * <p>Gain values are clamped to the closed interval [0.0, max] where |
| * max is the value of {@link #getMaxVolume}. |
| * A value of 0.0 results in zero gain (silence), and |
| * a value of 1.0 means unity gain (signal unchanged). |
| * The default value is 1.0 meaning unity gain. |
| * <p>This API is preferred over {@link #setStereoVolume}, as it |
| * more gracefully scales down to mono, and up to multi-channel content beyond stereo. |
| * <p>The word "volume" in the API name is historical; this is actually a linear gain. |
| * @param gain output gain for all channels. |
| * @return error code or success, see {@link #SUCCESS}, |
| * {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int setVolume(float gain) { |
| return setStereoVolume(gain, gain); |
| } |
| |
| |
| /** |
| * Sets the playback sample rate for this track. This sets the sampling rate at which |
| * the audio data will be consumed and played back |
| * (as set by the sampleRateInHz parameter in the |
| * {@link #AudioTrack(int, int, int, int, int, int)} constructor), |
| * not the original sampling rate of the |
| * content. For example, setting it to half the sample rate of the content will cause the |
| * playback to last twice as long, but will also result in a pitch shift down by one octave. |
| * The valid sample rate range is from 1 Hz to twice the value returned by |
| * {@link #getNativeOutputSampleRate(int)}. |
| * @param sampleRateInHz the sample rate expressed in Hz |
| * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, |
| * {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int setPlaybackRate(int sampleRateInHz) { |
| if (mState != STATE_INITIALIZED) { |
| return ERROR_INVALID_OPERATION; |
| } |
| if (sampleRateInHz <= 0) { |
| return ERROR_BAD_VALUE; |
| } |
| return native_set_playback_rate(sampleRateInHz); |
| } |
| |
| |
| /** |
| * Sets the position of the notification marker. At most one marker can be active. |
| * @param markerInFrames marker position in wrapping frame units similar to |
| * {@link #getPlaybackHeadPosition}, or zero to disable the marker. |
| * To set a marker at a position which would appear as zero due to wraparound, |
| * a workaround is to use a non-zero position near zero, such as -1 or 1. |
| * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, |
| * {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int setNotificationMarkerPosition(int markerInFrames) { |
| if (mState == STATE_UNINITIALIZED) { |
| return ERROR_INVALID_OPERATION; |
| } |
| return native_set_marker_pos(markerInFrames); |
| } |
| |
| |
| /** |
| * Sets the period for the periodic notification event. |
| * @param periodInFrames update period expressed in frames |
| * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int setPositionNotificationPeriod(int periodInFrames) { |
| if (mState == STATE_UNINITIALIZED) { |
| return ERROR_INVALID_OPERATION; |
| } |
| return native_set_pos_update_period(periodInFrames); |
| } |
| |
| |
| /** |
| * Sets the playback head position. |
| * The track must be stopped or paused for the position to be changed, |
| * and must use the {@link #MODE_STATIC} mode. |
| * @param positionInFrames playback head position expressed in frames |
| * Zero corresponds to start of buffer. |
| * The position must not be greater than the buffer size in frames, or negative. |
| * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, |
| * {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int setPlaybackHeadPosition(int positionInFrames) { |
| if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED || |
| getPlayState() == PLAYSTATE_PLAYING) { |
| return ERROR_INVALID_OPERATION; |
| } |
| if (!(0 <= positionInFrames && positionInFrames <= mNativeBufferSizeInFrames)) { |
| return ERROR_BAD_VALUE; |
| } |
| return native_set_position(positionInFrames); |
| } |
| |
| /** |
| * Sets the loop points and the loop count. The loop can be infinite. |
| * Similarly to setPlaybackHeadPosition, |
| * the track must be stopped or paused for the loop points to be changed, |
| * and must use the {@link #MODE_STATIC} mode. |
| * @param startInFrames loop start marker expressed in frames |
| * Zero corresponds to start of buffer. |
| * The start marker must not be greater than or equal to the buffer size in frames, or negative. |
| * @param endInFrames loop end marker expressed in frames |
| * The total buffer size in frames corresponds to end of buffer. |
| * The end marker must not be greater than the buffer size in frames. |
| * For looping, the end marker must not be less than or equal to the start marker, |
| * but to disable looping |
| * it is permitted for start marker, end marker, and loop count to all be 0. |
| * @param loopCount the number of times the loop is looped. |
| * A value of -1 means infinite looping, and 0 disables looping. |
| * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, |
| * {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int setLoopPoints(int startInFrames, int endInFrames, int loopCount) { |
| if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED || |
| getPlayState() == PLAYSTATE_PLAYING) { |
| return ERROR_INVALID_OPERATION; |
| } |
| if (loopCount == 0) { |
| ; // explicitly allowed as an exception to the loop region range check |
| } else if (!(0 <= startInFrames && startInFrames < mNativeBufferSizeInFrames && |
| startInFrames < endInFrames && endInFrames <= mNativeBufferSizeInFrames)) { |
| return ERROR_BAD_VALUE; |
| } |
| return native_set_loop(startInFrames, endInFrames, loopCount); |
| } |
| |
| /** |
| * Sets the initialization state of the instance. This method was originally intended to be used |
| * in an AudioTrack subclass constructor to set a subclass-specific post-initialization state. |
| * However, subclasses of AudioTrack are no longer recommended, so this method is obsolete. |
| * @param state the state of the AudioTrack instance |
| * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. |
| */ |
| @Deprecated |
| protected void setState(int state) { |
| mState = state; |
| } |
| |
| |
| //--------------------------------------------------------- |
| // Transport control methods |
| //-------------------- |
| /** |
| * Starts playing an AudioTrack. |
| * If track's creation mode is {@link #MODE_STATIC}, you must have called write() prior. |
| * |
| * @throws IllegalStateException |
| */ |
| public void play() |
| throws IllegalStateException { |
| if (mState != STATE_INITIALIZED) { |
| throw new IllegalStateException("play() called on uninitialized AudioTrack."); |
| } |
| if (isRestricted()) { |
| setVolume(0); |
| } |
| synchronized(mPlayStateLock) { |
| native_start(); |
| mPlayState = PLAYSTATE_PLAYING; |
| } |
| } |
| |
| private boolean isRestricted() { |
| try { |
| final int usage = AudioAttributes.usageForLegacyStreamType(mStreamType); |
| final int mode = mAppOps.checkAudioOperation(AppOpsManager.OP_PLAY_AUDIO, usage, |
| Process.myUid(), ActivityThread.currentPackageName()); |
| return mode != AppOpsManager.MODE_ALLOWED; |
| } catch (RemoteException e) { |
| return false; |
| } |
| } |
| |
| /** |
| * Stops playing the audio data. |
| * When used on an instance created in {@link #MODE_STREAM} mode, audio will stop playing |
| * after the last buffer that was written has been played. For an immediate stop, use |
| * {@link #pause()}, followed by {@link #flush()} to discard audio data that hasn't been played |
| * back yet. |
| * @throws IllegalStateException |
| */ |
| public void stop() |
| throws IllegalStateException { |
| if (mState != STATE_INITIALIZED) { |
| throw new IllegalStateException("stop() called on uninitialized AudioTrack."); |
| } |
| |
| // stop playing |
| synchronized(mPlayStateLock) { |
| native_stop(); |
| mPlayState = PLAYSTATE_STOPPED; |
| } |
| } |
| |
| /** |
| * Pauses the playback of the audio data. Data that has not been played |
| * back will not be discarded. Subsequent calls to {@link #play} will play |
| * this data back. See {@link #flush()} to discard this data. |
| * |
| * @throws IllegalStateException |
| */ |
| public void pause() |
| throws IllegalStateException { |
| if (mState != STATE_INITIALIZED) { |
| throw new IllegalStateException("pause() called on uninitialized AudioTrack."); |
| } |
| //logd("pause()"); |
| |
| // pause playback |
| synchronized(mPlayStateLock) { |
| native_pause(); |
| mPlayState = PLAYSTATE_PAUSED; |
| } |
| } |
| |
| |
| //--------------------------------------------------------- |
| // Audio data supply |
| //-------------------- |
| |
| /** |
| * Flushes the audio data currently queued for playback. Any data that has |
| * not been played back will be discarded. No-op if not stopped or paused, |
| * or if the track's creation mode is not {@link #MODE_STREAM}. |
| */ |
| public void flush() { |
| if (mState == STATE_INITIALIZED) { |
| // flush the data in native layer |
| native_flush(); |
| } |
| |
| } |
| |
| /** |
| * Writes the audio data to the audio sink for playback (streaming mode), |
| * or copies audio data for later playback (static buffer mode). |
| * In streaming mode, will block until all data has been written to the audio sink. |
| * In static buffer mode, copies the data to the buffer starting at offset 0. |
| * Note that the actual playback of this data might occur after this function |
| * returns. This function is thread safe with respect to {@link #stop} calls, |
| * in which case all of the specified data might not be written to the audio sink. |
| * |
| * @param audioData the array that holds the data to play. |
| * @param offsetInBytes the offset expressed in bytes in audioData where the data to play |
| * starts. |
| * @param sizeInBytes the number of bytes to read in audioData after the offset. |
| * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION} |
| * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if |
| * the parameters don't resolve to valid data and indexes, or |
| * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and |
| * needs to be recreated. |
| */ |
| |
| public int write(byte[] audioData, int offsetInBytes, int sizeInBytes) { |
| |
| if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { |
| return ERROR_INVALID_OPERATION; |
| } |
| |
| if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0) |
| || (offsetInBytes + sizeInBytes < 0) // detect integer overflow |
| || (offsetInBytes + sizeInBytes > audioData.length)) { |
| return ERROR_BAD_VALUE; |
| } |
| |
| int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat, |
| true /*isBlocking*/); |
| |
| if ((mDataLoadMode == MODE_STATIC) |
| && (mState == STATE_NO_STATIC_DATA) |
| && (ret > 0)) { |
| // benign race with respect to other APIs that read mState |
| mState = STATE_INITIALIZED; |
| } |
| |
| return ret; |
| } |
| |
| |
| /** |
| * Writes the audio data to the audio sink for playback (streaming mode), |
| * or copies audio data for later playback (static buffer mode). |
| * In streaming mode, will block until all data has been written to the audio sink. |
| * In static buffer mode, copies the data to the buffer starting at offset 0. |
| * Note that the actual playback of this data might occur after this function |
| * returns. This function is thread safe with respect to {@link #stop} calls, |
| * in which case all of the specified data might not be written to the audio sink. |
| * |
| * @param audioData the array that holds the data to play. |
| * @param offsetInShorts the offset expressed in shorts in audioData where the data to play |
| * starts. |
| * @param sizeInShorts the number of shorts to read in audioData after the offset. |
| * @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION} |
| * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if |
| * the parameters don't resolve to valid data and indexes. |
| */ |
| |
| public int write(short[] audioData, int offsetInShorts, int sizeInShorts) { |
| |
| if (mState == STATE_UNINITIALIZED || mAudioFormat != AudioFormat.ENCODING_PCM_16BIT) { |
| return ERROR_INVALID_OPERATION; |
| } |
| |
| if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0) |
| || (offsetInShorts + sizeInShorts < 0) // detect integer overflow |
| || (offsetInShorts + sizeInShorts > audioData.length)) { |
| return ERROR_BAD_VALUE; |
| } |
| |
| int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat); |
| |
| if ((mDataLoadMode == MODE_STATIC) |
| && (mState == STATE_NO_STATIC_DATA) |
| && (ret > 0)) { |
| // benign race with respect to other APIs that read mState |
| mState = STATE_INITIALIZED; |
| } |
| |
| return ret; |
| } |
| |
| |
| /** |
| * Writes the audio data to the audio sink for playback (streaming mode), |
| * or copies audio data for later playback (static buffer mode). |
| * In static buffer mode, copies the data to the buffer starting at offset 0, |
| * and the write mode is ignored. |
| * In streaming mode, the blocking behavior will depend on the write mode. |
| * <p> |
| * Note that the actual playback of this data might occur after this function |
| * returns. This function is thread safe with respect to {@link #stop} calls, |
| * in which case all of the specified data might not be written to the audio sink. |
| * <p> |
| * @param audioData the array that holds the data to play. |
| * The implementation does not clip for sample values within the nominal range |
| * [-1.0f, 1.0f], provided that all gains in the audio pipeline are |
| * less than or equal to unity (1.0f), and in the absence of post-processing effects |
| * that could add energy, such as reverb. For the convenience of applications |
| * that compute samples using filters with non-unity gain, |
| * sample values +3 dB beyond the nominal range are permitted. |
| * However such values may eventually be limited or clipped, depending on various gains |
| * and later processing in the audio path. Therefore applications are encouraged |
| * to provide samples values within the nominal range. |
| * @param offsetInFloats the offset, expressed as a number of floats, |
| * in audioData where the data to play starts. |
| * @param sizeInFloats the number of floats to read in audioData after the offset. |
| * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no |
| * effect in static mode. |
| * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written |
| * to the audio sink. |
| * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after |
| * queuing as much audio data for playback as possible without blocking. |
| * @return the number of floats that were written, or {@link #ERROR_INVALID_OPERATION} |
| * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if |
| * the parameters don't resolve to valid data and indexes. |
| */ |
| public int write(float[] audioData, int offsetInFloats, int sizeInFloats, |
| @WriteMode int writeMode) { |
| |
| if (mState == STATE_UNINITIALIZED) { |
| Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); |
| return ERROR_INVALID_OPERATION; |
| } |
| |
| if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) { |
| Log.e(TAG, "AudioTrack.write(float[] ...) requires format ENCODING_PCM_FLOAT"); |
| return ERROR_INVALID_OPERATION; |
| } |
| |
| if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { |
| Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); |
| return ERROR_BAD_VALUE; |
| } |
| |
| if ( (audioData == null) || (offsetInFloats < 0 ) || (sizeInFloats < 0) |
| || (offsetInFloats + sizeInFloats < 0) // detect integer overflow |
| || (offsetInFloats + sizeInFloats > audioData.length)) { |
| Log.e(TAG, "AudioTrack.write() called with invalid array, offset, or size"); |
| return ERROR_BAD_VALUE; |
| } |
| |
| int ret = native_write_float(audioData, offsetInFloats, sizeInFloats, mAudioFormat, |
| writeMode == WRITE_BLOCKING); |
| |
| if ((mDataLoadMode == MODE_STATIC) |
| && (mState == STATE_NO_STATIC_DATA) |
| && (ret > 0)) { |
| // benign race with respect to other APIs that read mState |
| mState = STATE_INITIALIZED; |
| } |
| |
| return ret; |
| } |
| |
| |
| /** |
| * Writes the audio data to the audio sink for playback (streaming mode), |
| * or copies audio data for later playback (static buffer mode). |
| * In static buffer mode, copies the data to the buffer starting at its 0 offset, and the write |
| * mode is ignored. |
| * In streaming mode, the blocking behavior will depend on the write mode. |
| * @param audioData the buffer that holds the data to play, starting at the position reported |
| * by <code>audioData.position()</code>. |
| * <BR>Note that upon return, the buffer position (<code>audioData.position()</code>) will |
| * have been advanced to reflect the amount of data that was successfully written to |
| * the AudioTrack. |
| * @param sizeInBytes number of bytes to write. |
| * <BR>Note this may differ from <code>audioData.remaining()</code>, but cannot exceed it. |
| * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no |
| * effect in static mode. |
| * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written |
| * to the audio sink. |
| * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after |
| * queuing as much audio data for playback as possible without blocking. |
| * @return 0 or a positive number of bytes that were written, or |
| * {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int write(ByteBuffer audioData, int sizeInBytes, |
| @WriteMode int writeMode) { |
| |
| if (mState == STATE_UNINITIALIZED) { |
| Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); |
| return ERROR_INVALID_OPERATION; |
| } |
| |
| if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { |
| Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); |
| return ERROR_BAD_VALUE; |
| } |
| |
| if ( (audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) { |
| Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value"); |
| return ERROR_BAD_VALUE; |
| } |
| |
| int ret = 0; |
| if (audioData.isDirect()) { |
| ret = native_write_native_bytes(audioData, |
| audioData.position(), sizeInBytes, mAudioFormat, |
| writeMode == WRITE_BLOCKING); |
| } else { |
| ret = native_write_byte(NioUtils.unsafeArray(audioData), |
| NioUtils.unsafeArrayOffset(audioData) + audioData.position(), |
| sizeInBytes, mAudioFormat, |
| writeMode == WRITE_BLOCKING); |
| } |
| |
| if ((mDataLoadMode == MODE_STATIC) |
| && (mState == STATE_NO_STATIC_DATA) |
| && (ret > 0)) { |
| // benign race with respect to other APIs that read mState |
| mState = STATE_INITIALIZED; |
| } |
| |
| if (ret > 0) { |
| audioData.position(audioData.position() + ret); |
| } |
| |
| return ret; |
| } |
| |
| /** |
| * Notifies the native resource to reuse the audio data already loaded in the native |
| * layer, that is to rewind to start of buffer. |
| * The track's creation mode must be {@link #MODE_STATIC}. |
| * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, |
| * {@link #ERROR_INVALID_OPERATION} |
| */ |
| public int reloadStaticData() { |
| if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED) { |
| return ERROR_INVALID_OPERATION; |
| } |
| return native_reload_static(); |
| } |
| |
| //-------------------------------------------------------------------------- |
| // Audio effects management |
| //-------------------- |
| |
| /** |
| * Attaches an auxiliary effect to the audio track. A typical auxiliary |
| * effect is a reverberation effect which can be applied on any sound source |
| * that directs a certain amount of its energy to this effect. This amount |
| * is defined by setAuxEffectSendLevel(). |
| * {@see #setAuxEffectSendLevel(float)}. |
| * <p>After creating an auxiliary effect (e.g. |
| * {@link android.media.audiofx.EnvironmentalReverb}), retrieve its ID with |
| * {@link android.media.audiofx.AudioEffect#getId()} and use it when calling |
| * this method to attach the audio track to the effect. |
| * <p>To detach the effect from the audio track, call this method with a |
| * null effect id. |
| * |
| * @param effectId system wide unique id of the effect to attach |
| * @return error code or success, see {@link #SUCCESS}, |
| * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR_BAD_VALUE} |
| */ |
| public int attachAuxEffect(int effectId) { |
| if (mState == STATE_UNINITIALIZED) { |
| return ERROR_INVALID_OPERATION; |
| } |
| return native_attachAuxEffect(effectId); |
| } |
| |
| /** |
| * Sets the send level of the audio track to the attached auxiliary effect |
| * {@link #attachAuxEffect(int)}. Effect levels |
| * are clamped to the closed interval [0.0, max] where |
| * max is the value of {@link #getMaxVolume}. |
| * A value of 0.0 results in no effect, and a value of 1.0 is full send. |
| * <p>By default the send level is 0.0f, so even if an effect is attached to the player |
| * this method must be called for the effect to be applied. |
| * <p>Note that the passed level value is a linear scalar. UI controls should be scaled |
| * logarithmically: the gain applied by audio framework ranges from -72dB to at least 0dB, |
| * so an appropriate conversion from linear UI input x to level is: |
| * x == 0 -> level = 0 |
| * 0 < x <= R -> level = 10^(72*(x-R)/20/R) |
| * |
| * @param level linear send level |
| * @return error code or success, see {@link #SUCCESS}, |
| * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR} |
| */ |
| public int setAuxEffectSendLevel(float level) { |
| if (isRestricted()) { |
| return SUCCESS; |
| } |
| if (mState == STATE_UNINITIALIZED) { |
| return ERROR_INVALID_OPERATION; |
| } |
| level = clampGainOrLevel(level); |
| int err = native_setAuxEffectSendLevel(level); |
| return err == 0 ? SUCCESS : ERROR; |
| } |
| |
| //--------------------------------------------------------- |
| // Interface definitions |
| //-------------------- |
| /** |
| * Interface definition for a callback to be invoked when the playback head position of |
| * an AudioTrack has reached a notification marker or has increased by a certain period. |
| */ |
| public interface OnPlaybackPositionUpdateListener { |
| /** |
| * Called on the listener to notify it that the previously set marker has been reached |
| * by the playback head. |
| */ |
| void onMarkerReached(AudioTrack track); |
| |
| /** |
| * Called on the listener to periodically notify it that the playback head has reached |
| * a multiple of the notification period. |
| */ |
| void onPeriodicNotification(AudioTrack track); |
| } |
| |
| //--------------------------------------------------------- |
| // Inner classes |
| //-------------------- |
| /** |
| * Helper class to handle the forwarding of native events to the appropriate listener |
| * (potentially) handled in a different thread |
| */ |
| private class NativeEventHandlerDelegate { |
| private final Handler mHandler; |
| |
| NativeEventHandlerDelegate(final AudioTrack track, |
| final OnPlaybackPositionUpdateListener listener, |
| Handler handler) { |
| // find the looper for our new event handler |
| Looper looper; |
| if (handler != null) { |
| looper = handler.getLooper(); |
| } else { |
| // no given handler, use the looper the AudioTrack was created in |
| looper = mInitializationLooper; |
| } |
| |
| // construct the event handler with this looper |
| if (looper != null) { |
| // implement the event handler delegate |
| mHandler = new Handler(looper) { |
| @Override |
| public void handleMessage(Message msg) { |
| if (track == null) { |
| return; |
| } |
| switch(msg.what) { |
| case NATIVE_EVENT_MARKER: |
| if (listener != null) { |
| listener.onMarkerReached(track); |
| } |
| break; |
| case NATIVE_EVENT_NEW_POS: |
| if (listener != null) { |
| listener.onPeriodicNotification(track); |
| } |
| break; |
| default: |
| loge("Unknown native event type: " + msg.what); |
| break; |
| } |
| } |
| }; |
| } else { |
| mHandler = null; |
| } |
| } |
| |
| Handler getHandler() { |
| return mHandler; |
| } |
| } |
| |
| |
| //--------------------------------------------------------- |
| // Java methods called from the native side |
| //-------------------- |
| @SuppressWarnings("unused") |
| private static void postEventFromNative(Object audiotrack_ref, |
| int what, int arg1, int arg2, Object obj) { |
| //logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2); |
| AudioTrack track = (AudioTrack)((WeakReference)audiotrack_ref).get(); |
| if (track == null) { |
| return; |
| } |
| |
| NativeEventHandlerDelegate delegate = track.mEventHandlerDelegate; |
| if (delegate != null) { |
| Handler handler = delegate.getHandler(); |
| if (handler != null) { |
| Message m = handler.obtainMessage(what, arg1, arg2, obj); |
| handler.sendMessage(m); |
| } |
| } |
| |
| } |
| |
| |
| //--------------------------------------------------------- |
| // Native methods called from the Java side |
| //-------------------- |
| |
| // post-condition: mStreamType is overwritten with a value |
| // that reflects the audio attributes (e.g. an AudioAttributes object with a usage of |
| // AudioAttributes.USAGE_MEDIA will map to AudioManager.STREAM_MUSIC |
| private native final int native_setup(Object /*WeakReference<AudioTrack>*/ audiotrack_this, |
| Object /*AudioAttributes*/ attributes, |
| int sampleRate, int channelMask, int audioFormat, |
| int buffSizeInBytes, int mode, int[] sessionId); |
| |
| private native final void native_finalize(); |
| |
| private native final void native_release(); |
| |
| private native final void native_start(); |
| |
| private native final void native_stop(); |
| |
| private native final void native_pause(); |
| |
| private native final void native_flush(); |
| |
| private native final int native_write_byte(byte[] audioData, |
| int offsetInBytes, int sizeInBytes, int format, |
| boolean isBlocking); |
| |
| private native final int native_write_short(short[] audioData, |
| int offsetInShorts, int sizeInShorts, int format); |
| |
| private native final int native_write_float(float[] audioData, |
| int offsetInFloats, int sizeInFloats, int format, |
| boolean isBlocking); |
| |
| private native final int native_write_native_bytes(Object audioData, |
| int positionInBytes, int sizeInBytes, int format, boolean blocking); |
| |
| private native final int native_reload_static(); |
| |
| private native final int native_get_native_frame_count(); |
| |
| private native final void native_setVolume(float leftVolume, float rightVolume); |
| |
| private native final int native_set_playback_rate(int sampleRateInHz); |
| private native final int native_get_playback_rate(); |
| |
| private native final int native_set_marker_pos(int marker); |
| private native final int native_get_marker_pos(); |
| |
| private native final int native_set_pos_update_period(int updatePeriod); |
| private native final int native_get_pos_update_period(); |
| |
| private native final int native_set_position(int position); |
| private native final int native_get_position(); |
| |
| private native final int native_get_latency(); |
| |
| // longArray must be a non-null array of length >= 2 |
| // [0] is assigned the frame position |
| // [1] is assigned the time in CLOCK_MONOTONIC nanoseconds |
| private native final int native_get_timestamp(long[] longArray); |
| |
| private native final int native_set_loop(int start, int end, int loopCount); |
| |
| static private native final int native_get_output_sample_rate(int streamType); |
| static private native final int native_get_min_buff_size( |
| int sampleRateInHz, int channelConfig, int audioFormat); |
| |
| private native final int native_attachAuxEffect(int effectId); |
| private native final int native_setAuxEffectSendLevel(float level); |
| |
| //--------------------------------------------------------- |
| // Utility methods |
| //------------------ |
| |
| private static void logd(String msg) { |
| Log.d(TAG, msg); |
| } |
| |
| private static void loge(String msg) { |
| Log.e(TAG, msg); |
| } |
| |
| } |