/* * 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.util.Collection; import java.util.Iterator; import android.annotation.IntDef; import android.annotation.NonNull; import android.annotation.SystemApi; import android.app.ActivityThread; import android.os.Binder; import android.os.Handler; import android.os.IBinder; import android.os.Looper; import android.os.Message; import android.os.RemoteException; import android.os.ServiceManager; import android.util.ArrayMap; import android.util.Log; import com.android.internal.annotations.GuardedBy; /** * The AudioRecord class manages the audio resources for Java applications * to record audio from the audio input hardware of the platform. This is * achieved by "pulling" (reading) the data from the AudioRecord object. The * application is responsible for polling the AudioRecord object in time using one of * the following three methods: {@link #read(byte[],int, int)}, {@link #read(short[], int, int)} * or {@link #read(ByteBuffer, int)}. The choice of which method to use will be based * on the audio data storage format that is the most convenient for the user of AudioRecord. *
Upon creation, an AudioRecord object initializes its associated audio buffer that it will
* fill with the new audio data. The size of this buffer, specified during the construction,
* determines how long an AudioRecord can record before "over-running" data that has not
* been read yet. Data should be read from the audio hardware in chunks of sizes inferior to
* the total recording buffer size.
*/
public class AudioRecord implements AudioRouting
{
//---------------------------------------------------------
// Constants
//--------------------
/**
* indicates AudioRecord state is not successfully initialized.
*/
public static final int STATE_UNINITIALIZED = 0;
/**
* indicates AudioRecord state is ready to be used
*/
public static final int STATE_INITIALIZED = 1;
/**
* indicates AudioRecord recording state is not recording
*/
public static final int RECORDSTATE_STOPPED = 1; // matches SL_RECORDSTATE_STOPPED
/**
* indicates AudioRecord recording state is recording
*/
public static final int RECORDSTATE_RECORDING = 3;// matches SL_RECORDSTATE_RECORDING
/**
* 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;
/**
* An error code indicating that the object reporting it is no longer valid and needs to
* be recreated.
*/
public static final int ERROR_DEAD_OBJECT = AudioSystem.DEAD_OBJECT;
// Error codes:
// to keep in sync with frameworks/base/core/jni/android_media_AudioRecord.cpp
private static final int AUDIORECORD_ERROR_SETUP_ZEROFRAMECOUNT = -16;
private static final int AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK = -17;
private static final int AUDIORECORD_ERROR_SETUP_INVALIDFORMAT = -18;
private static final int AUDIORECORD_ERROR_SETUP_INVALIDSOURCE = -19;
private static final int AUDIORECORD_ERROR_SETUP_NATIVEINITFAILED = -20;
// Events:
// to keep in sync with frameworks/av/include/media/AudioRecord.h
/**
* Event id denotes when record head has reached a previously set marker.
*/
private static final int NATIVE_EVENT_MARKER = 2;
/**
* Event id denotes when previously set update period has elapsed during recording.
*/
private static final int NATIVE_EVENT_NEW_POS = 3;
private final static String TAG = "android.media.AudioRecord";
/** @hide */
public final static String SUBMIX_FIXED_VOLUME = "fixedVolume";
/** @hide */
@IntDef({
READ_BLOCKING,
READ_NON_BLOCKING
})
@Retention(RetentionPolicy.SOURCE)
public @interface ReadMode {}
/**
* The read mode indicating the read operation will block until all data
* requested has been read.
*/
public final static int READ_BLOCKING = 0;
/**
* The read mode indicating the read operation will return immediately after
* reading as much audio data as possible without blocking.
*/
public final static int READ_NON_BLOCKING = 1;
//---------------------------------------------------------
// Used exclusively by native code
//--------------------
/**
* Accessed by native methods: provides access to C++ AudioRecord object
*/
@SuppressWarnings("unused")
private long mNativeRecorderInJavaObj;
/**
* Accessed by native methods: provides access to the callback data.
*/
@SuppressWarnings("unused")
private long mNativeCallbackCookie;
/**
* Accessed by native methods: provides access to the JNIDeviceCallback instance.
*/
@SuppressWarnings("unused")
private long mNativeDeviceCallback;
//---------------------------------------------------------
// Member variables
//--------------------
/**
* The audio data sampling rate in Hz.
* Never {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED}.
*/
private int mSampleRate; // initialized by all constructors via audioParamCheck()
/**
* The number of input audio channels (1 is mono, 2 is stereo)
*/
private int mChannelCount;
/**
* The audio channel position mask
*/
private int mChannelMask;
/**
* The audio channel index mask
*/
private int mChannelIndexMask;
/**
* The encoding of the audio samples.
* @see AudioFormat#ENCODING_PCM_8BIT
* @see AudioFormat#ENCODING_PCM_16BIT
* @see AudioFormat#ENCODING_PCM_FLOAT
*/
private int mAudioFormat;
/**
* Where the audio data is recorded from.
*/
private int mRecordSource;
/**
* Indicates the state of the AudioRecord instance.
*/
private int mState = STATE_UNINITIALIZED;
/**
* Indicates the recording state of the AudioRecord instance.
*/
private int mRecordingState = RECORDSTATE_STOPPED;
/**
* Lock to make sure mRecordingState updates are reflecting the actual state of the object.
*/
private final Object mRecordingStateLock = new Object();
/**
* The listener the AudioRecord notifies when the record position reaches a marker
* or for periodic updates during the progression of the record head.
* @see #setRecordPositionUpdateListener(OnRecordPositionUpdateListener)
* @see #setRecordPositionUpdateListener(OnRecordPositionUpdateListener, Handler)
*/
private OnRecordPositionUpdateListener mPositionListener = null;
/**
* Lock to protect position listener updates against event notifications
*/
private final Object mPositionListenerLock = new Object();
/**
* Handler for marker events coming from the native code
*/
private NativeEventHandler mEventHandler = null;
/**
* Looper associated with the thread that creates the AudioRecord instance
*/
private Looper mInitializationLooper = null;
/**
* Size of the native audio buffer.
*/
private int mNativeBufferSizeInBytes = 0;
/**
* Audio session ID
*/
private int mSessionId = AudioManager.AUDIO_SESSION_ID_GENERATE;
/**
* AudioAttributes
*/
private AudioAttributes mAudioAttributes;
private boolean mIsSubmixFullVolume = false;
//---------------------------------------------------------
// Constructor, Finalize
//--------------------
/**
* Class constructor.
* Though some invalid parameters will result in an {@link IllegalArgumentException} exception,
* other errors do not. Thus you should call {@link #getState()} immediately after construction
* to confirm that the object is usable.
* @param audioSource the recording source.
* See {@link MediaRecorder.AudioSource} for the recording source definitions.
* @param sampleRateInHz the sample rate expressed in Hertz. 44100Hz is currently the only
* rate that is guaranteed to work on all devices, but other rates such as 22050,
* 16000, and 11025 may work on some devices.
* {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED} means to use a route-dependent value
* which is usually the sample rate of the source.
* {@link #getSampleRate()} can be used to retrieve the actual sample rate chosen.
* @param channelConfig describes the configuration of the audio channels.
* See {@link AudioFormat#CHANNEL_IN_MONO} and
* {@link AudioFormat#CHANNEL_IN_STEREO}. {@link AudioFormat#CHANNEL_IN_MONO} is guaranteed
* to work on all devices.
* @param audioFormat the format in which the audio data is to be returned.
* See {@link AudioFormat#ENCODING_PCM_8BIT}, {@link AudioFormat#ENCODING_PCM_16BIT},
* and {@link AudioFormat#ENCODING_PCM_FLOAT}.
* @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is written
* to during the recording. New audio data can be read from this buffer in smaller chunks
* than this size. See {@link #getMinBufferSize(int, int, int)} to determine the minimum
* required buffer size for the successful creation of an AudioRecord instance. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @throws java.lang.IllegalArgumentException
*/
public AudioRecord(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes)
throws IllegalArgumentException {
this((new AudioAttributes.Builder())
.setInternalCapturePreset(audioSource)
.build(),
(new AudioFormat.Builder())
.setChannelMask(getChannelMaskFromLegacyConfig(channelConfig,
true/*allow legacy configurations*/))
.setEncoding(audioFormat)
.setSampleRate(sampleRateInHz)
.build(),
bufferSizeInBytes,
AudioManager.AUDIO_SESSION_ID_GENERATE);
}
/**
* @hide
* Class constructor with {@link AudioAttributes} and {@link AudioFormat}.
* @param attributes a non-null {@link AudioAttributes} instance. Use
* {@link AudioAttributes.Builder#setAudioSource(int)} for configuring the audio
* source for this instance.
* @param format a non-null {@link AudioFormat} instance describing the format of the data
* that will be recorded through this AudioRecord. 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 written
* to during the recording. New audio data can be read from this buffer in smaller chunks
* than this size. See {@link #getMinBufferSize(int, int, int)} to determine the minimum
* required buffer size for the successful creation of an AudioRecord instance. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @param sessionId ID of audio session the AudioRecord 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
*/
@SystemApi
public AudioRecord(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int sessionId) throws IllegalArgumentException {
mRecordingState = RECORDSTATE_STOPPED;
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 AudioRecord instanciation
if ((mInitializationLooper = Looper.myLooper()) == null) {
mInitializationLooper = Looper.getMainLooper();
}
// is this AudioRecord using REMOTE_SUBMIX at full volume?
if (attributes.getCapturePreset() == MediaRecorder.AudioSource.REMOTE_SUBMIX) {
final AudioAttributes.Builder filteredAttr = new AudioAttributes.Builder();
final Iterator Here is an example where
* If the audio source is not set with {@link #setAudioSource(int)},
* {@link MediaRecorder.AudioSource#DEFAULT} is used.
* See {@link AudioFormat#CHANNEL_IN_MONO}
* and {@link AudioFormat#CHANNEL_IN_STEREO}.
* This method may return {@link AudioFormat#CHANNEL_INVALID} if
* a channel index mask is used.
* Consider {@link #getFormat()} instead, to obtain an {@link AudioFormat},
* which contains both the channel position mask and the channel index mask.
*/
public int getChannelConfiguration() {
return mChannelMask;
}
/**
* Returns the configured
* The AudioTimestamp reflects the frame delivery information at
* the earliest point available in the capture pipeline.
*
* Calling {@link #startRecording()} following a {@link #stop()} will reset
* the frame count to 0.
*
* @param outTimestamp a caller provided non-null AudioTimestamp instance,
* which is updated with the AudioRecord frame delivery information upon success.
* @param timebase one of
* {@link AudioTimestamp#TIMEBASE_BOOTTIME AudioTimestamp.TIMEBASE_BOOTTIME} or
* {@link AudioTimestamp#TIMEBASE_MONOTONIC AudioTimestamp.TIMEBASE_MONOTONIC},
* used to select the clock for the AudioTimestamp time.
* @return {@link #SUCCESS} if a timestamp is available,
* or {@link #ERROR_INVALID_OPERATION} if a timestamp not available.
*/
public int getTimestamp(@NonNull AudioTimestamp outTimestamp,
@AudioTimestamp.Timebase int timebase)
{
if (outTimestamp == null ||
(timebase != AudioTimestamp.TIMEBASE_BOOTTIME
&& timebase != AudioTimestamp.TIMEBASE_MONOTONIC)) {
throw new IllegalArgumentException();
}
return native_get_timestamp(outTimestamp, timebase);
}
/**
* Returns the minimum buffer size required for the successful creation of an AudioRecord
* object, in byte units.
* Note that this size doesn't guarantee a smooth recording under load, and higher values
* should be chosen according to the expected frequency at which the AudioRecord instance
* will be polled for new data.
* See {@link #AudioRecord(int, int, int, int, int)} for more information on valid
* configuration values.
* @param sampleRateInHz the sample rate expressed in Hertz.
* {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED} is not permitted.
* @param channelConfig describes the configuration of the audio channels.
* See {@link AudioFormat#CHANNEL_IN_MONO} and
* {@link AudioFormat#CHANNEL_IN_STEREO}
* @param audioFormat the format in which the audio data is represented.
* See {@link AudioFormat#ENCODING_PCM_16BIT}.
* @return {@link #ERROR_BAD_VALUE} if the recording parameters are not supported by the
* hardware, or an invalid parameter was passed,
* or {@link #ERROR} if the implementation was unable to query the hardware for its
* input properties,
* or the minimum buffer size expressed in bytes.
* @see #AudioRecord(int, int, int, int, int)
*/
static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
int channelCount = 0;
switch (channelConfig) {
case AudioFormat.CHANNEL_IN_DEFAULT: // AudioFormat.CHANNEL_CONFIGURATION_DEFAULT
case AudioFormat.CHANNEL_IN_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
channelCount = 1;
break;
case AudioFormat.CHANNEL_IN_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
case (AudioFormat.CHANNEL_IN_FRONT | AudioFormat.CHANNEL_IN_BACK):
channelCount = 2;
break;
case AudioFormat.CHANNEL_INVALID:
default:
loge("getMinBufferSize(): Invalid channel configuration.");
return ERROR_BAD_VALUE;
}
int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
if (size == 0) {
return ERROR_BAD_VALUE;
}
else if (size == -1) {
return ERROR;
}
else {
return size;
}
}
/**
* Returns the audio session ID.
*
* @return the ID of the audio session this AudioRecord belongs to.
*/
public int getAudioSessionId() {
return mSessionId;
}
//---------------------------------------------------------
// Transport control methods
//--------------------
/**
* Starts recording from the AudioRecord instance.
* @throws IllegalStateException
*/
public void startRecording()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("startRecording() called on an "
+ "uninitialized AudioRecord.");
}
// start recording
synchronized(mRecordingStateLock) {
if (native_start(MediaSyncEvent.SYNC_EVENT_NONE, 0) == SUCCESS) {
handleFullVolumeRec(true);
mRecordingState = RECORDSTATE_RECORDING;
}
}
}
/**
* Starts recording from the AudioRecord instance when the specified synchronization event
* occurs on the specified audio session.
* @throws IllegalStateException
* @param syncEvent event that triggers the capture.
* @see MediaSyncEvent
*/
public void startRecording(MediaSyncEvent syncEvent)
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("startRecording() called on an "
+ "uninitialized AudioRecord.");
}
// start recording
synchronized(mRecordingStateLock) {
if (native_start(syncEvent.getType(), syncEvent.getAudioSessionId()) == SUCCESS) {
handleFullVolumeRec(true);
mRecordingState = RECORDSTATE_RECORDING;
}
}
}
/**
* Stops recording.
* @throws IllegalStateException
*/
public void stop()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("stop() called on an uninitialized AudioRecord.");
}
// stop recording
synchronized(mRecordingStateLock) {
handleFullVolumeRec(false);
native_stop();
mRecordingState = RECORDSTATE_STOPPED;
}
}
private final IBinder mICallBack = new Binder();
private void handleFullVolumeRec(boolean starting) {
if (!mIsSubmixFullVolume) {
return;
}
final IBinder b = ServiceManager.getService(android.content.Context.AUDIO_SERVICE);
final IAudioService ias = IAudioService.Stub.asInterface(b);
try {
ias.forceRemoteSubmixFullVolume(starting, mICallBack);
} catch (RemoteException e) {
Log.e(TAG, "Error talking to AudioService when handling full submix volume", e);
}
}
//---------------------------------------------------------
// Audio data supply
//--------------------
/**
* Reads audio data from the audio hardware for recording into a byte array.
* The format specified in the AudioRecord constructor should be
* {@link AudioFormat#ENCODING_PCM_8BIT} to correspond to the data in the array.
* @param audioData the array to which the recorded audio data is written.
* @param offsetInBytes index in audioData from which the data is written expressed in bytes.
* @param sizeInBytes the number of requested bytes.
* @return zero or the positive number of bytes that were read, or one of the following
* error codes. The number of bytes will not exceed sizeInBytes.
* AudioRecord
instance. By setting the
* recording source and audio format parameters, you indicate which of
* those vary from the default behavior on the device.
* Builder
is used to specify all {@link AudioFormat}
* parameters, to be used by a new AudioRecord
instance:
*
*
* AudioRecord recorder = new AudioRecord.Builder()
* .setAudioSource(MediaRecorder.AudioSource.VOICE_COMMUNICATION)
* .setAudioFormat(new AudioFormat.Builder()
* .setEncoding(AudioFormat.ENCODING_PCM_16BIT)
* .setSampleRate(32000)
* .setChannelMask(AudioFormat.CHANNEL_IN_MONO)
* .build())
* .setBufferSize(2*minBuffSize)
* .build();
*
*
If the audio format is not specified or is incomplete, its channel configuration will be
* {@link AudioFormat#CHANNEL_IN_MONO}, and the encoding will be
* {@link AudioFormat#ENCODING_PCM_16BIT}.
* The sample rate will depend on the device actually selected for capture and can be queried
* with {@link #getSampleRate()} method.
*
If the buffer size is not specified with {@link #setBufferSizeInBytes(int)},
* the minimum buffer size for the source is used.
*/
public static class Builder {
private AudioAttributes mAttributes;
private AudioFormat mFormat;
private int mBufferSizeInBytes;
private int mSessionId = AudioManager.AUDIO_SESSION_ID_GENERATE;
/**
* Constructs a new Builder with the default values as described above.
*/
public Builder() {
}
/**
* @param source the audio source.
* See {@link MediaRecorder.AudioSource} for the supported audio source definitions.
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public Builder setAudioSource(int source) throws IllegalArgumentException {
if ( (source < MediaRecorder.AudioSource.DEFAULT) ||
(source > MediaRecorder.getAudioSourceMax()) ) {
throw new IllegalArgumentException("Invalid audio source " + source);
}
mAttributes = new AudioAttributes.Builder()
.setInternalCapturePreset(source)
.build();
return this;
}
/**
* @hide
* To be only used by system components. Allows specifying non-public capture presets
* @param attributes a non-null {@link AudioAttributes} instance that contains the capture
* preset to be used.
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
@SystemApi
public Builder setAudioAttributes(@NonNull AudioAttributes attributes)
throws IllegalArgumentException {
if (attributes == null) {
throw new IllegalArgumentException("Illegal null AudioAttributes argument");
}
if (attributes.getCapturePreset() == MediaRecorder.AudioSource.AUDIO_SOURCE_INVALID) {
throw new IllegalArgumentException(
"No valid capture preset in AudioAttributes argument");
}
// keep reference, we only copy the data when building
mAttributes = attributes;
return this;
}
/**
* Sets the format of the audio data to be captured.
* @param format a non-null {@link AudioFormat} instance
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public Builder setAudioFormat(@NonNull AudioFormat format) throws IllegalArgumentException {
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat argument");
}
// keep reference, we only copy the data when building
mFormat = format;
return this;
}
/**
* Sets the total size (in bytes) of the buffer where audio data is written
* during the recording. New audio data can be read from this buffer in smaller chunks
* than this size. See {@link #getMinBufferSize(int, int, int)} to determine the minimum
* required buffer size for the successful creation of an AudioRecord instance.
* Since bufferSizeInBytes may be internally increased to accommodate the source
* requirements, use {@link #getBufferSizeInFrames()} to determine the actual buffer size
* in frames.
* @param bufferSizeInBytes a value strictly greater than 0
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public Builder setBufferSizeInBytes(int bufferSizeInBytes) throws IllegalArgumentException {
if (bufferSizeInBytes <= 0) {
throw new IllegalArgumentException("Invalid buffer size " + bufferSizeInBytes);
}
mBufferSizeInBytes = bufferSizeInBytes;
return this;
}
/**
* @hide
* To be only used by system components.
* @param sessionId ID of audio session the AudioRecord must be attached to, or
* {@link AudioManager#AUDIO_SESSION_ID_GENERATE} if the session isn't known at
* construction time.
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
@SystemApi
public Builder setSessionId(int sessionId) throws IllegalArgumentException {
if (sessionId < 0) {
throw new IllegalArgumentException("Invalid session ID " + sessionId);
}
mSessionId = sessionId;
return this;
}
/**
* @return a new {@link AudioRecord} instance successfully initialized with all
* the parameters set on this Builder
.
* @throws UnsupportedOperationException if the parameters set on the Builder
* were incompatible, or if they are not supported by the device,
* or if the device was not available.
*/
public AudioRecord build() throws UnsupportedOperationException {
if (mFormat == null) {
mFormat = new AudioFormat.Builder()
.setEncoding(AudioFormat.ENCODING_PCM_16BIT)
.setChannelMask(AudioFormat.CHANNEL_IN_MONO)
.build();
} else {
if (mFormat.getEncoding() == AudioFormat.ENCODING_INVALID) {
mFormat = new AudioFormat.Builder(mFormat)
.setEncoding(AudioFormat.ENCODING_PCM_16BIT)
.build();
}
if (mFormat.getChannelMask() == AudioFormat.CHANNEL_INVALID
&& mFormat.getChannelIndexMask() == AudioFormat.CHANNEL_INVALID) {
mFormat = new AudioFormat.Builder(mFormat)
.setChannelMask(AudioFormat.CHANNEL_IN_MONO)
.build();
}
}
if (mAttributes == null) {
mAttributes = new AudioAttributes.Builder()
.setInternalCapturePreset(MediaRecorder.AudioSource.DEFAULT)
.build();
}
try {
// If the buffer size is not specified,
// use a single frame for the buffer size and let the
// native code figure out the minimum buffer size.
if (mBufferSizeInBytes == 0) {
mBufferSizeInBytes = mFormat.getChannelCount()
* mFormat.getBytesPerSample(mFormat.getEncoding());
}
final AudioRecord record = new AudioRecord(
mAttributes, mFormat, mBufferSizeInBytes, mSessionId);
if (record.getState() == STATE_UNINITIALIZED) {
// release is not necessary
throw new UnsupportedOperationException("Cannot create AudioRecord");
}
return record;
} catch (IllegalArgumentException e) {
throw new UnsupportedOperationException(e.getMessage());
}
}
}
// Convenience method for the constructor's parameter checks.
// This, getChannelMaskFromLegacyConfig and audioBuffSizeCheck are where constructor
// IllegalArgumentException-s are thrown
private static int getChannelMaskFromLegacyConfig(int inChannelConfig,
boolean allowLegacyConfig) {
int mask;
switch (inChannelConfig) {
case AudioFormat.CHANNEL_IN_DEFAULT: // AudioFormat.CHANNEL_CONFIGURATION_DEFAULT
case AudioFormat.CHANNEL_IN_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
mask = AudioFormat.CHANNEL_IN_MONO;
break;
case AudioFormat.CHANNEL_IN_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
mask = AudioFormat.CHANNEL_IN_STEREO;
break;
case (AudioFormat.CHANNEL_IN_FRONT | AudioFormat.CHANNEL_IN_BACK):
mask = inChannelConfig;
break;
default:
throw new IllegalArgumentException("Unsupported channel configuration.");
}
if (!allowLegacyConfig && ((inChannelConfig == AudioFormat.CHANNEL_CONFIGURATION_MONO)
|| (inChannelConfig == AudioFormat.CHANNEL_CONFIGURATION_STEREO))) {
// only happens with the constructor that uses AudioAttributes and AudioFormat
throw new IllegalArgumentException("Unsupported deprecated configuration.");
}
return mask;
}
// postconditions:
// mRecordSource is valid
// mAudioFormat is valid
// mSampleRate is valid
private void audioParamCheck(int audioSource, int sampleRateInHz, int audioFormat)
throws IllegalArgumentException {
//--------------
// audio source
if ( (audioSource < MediaRecorder.AudioSource.DEFAULT) ||
((audioSource > MediaRecorder.getAudioSourceMax()) &&
(audioSource != MediaRecorder.AudioSource.RADIO_TUNER) &&
(audioSource != MediaRecorder.AudioSource.HOTWORD)) ) {
throw new IllegalArgumentException("Invalid audio source.");
}
mRecordSource = audioSource;
//--------------
// sample rate
if ((sampleRateInHz < AudioFormat.SAMPLE_RATE_HZ_MIN ||
sampleRateInHz > AudioFormat.SAMPLE_RATE_HZ_MAX) &&
sampleRateInHz != AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
throw new IllegalArgumentException(sampleRateInHz
+ "Hz is not a supported sample rate.");
}
mSampleRate = sampleRateInHz;
//--------------
// audio format
switch (audioFormat) {
case AudioFormat.ENCODING_DEFAULT:
mAudioFormat = AudioFormat.ENCODING_PCM_16BIT;
break;
case AudioFormat.ENCODING_PCM_FLOAT:
case AudioFormat.ENCODING_PCM_16BIT:
case AudioFormat.ENCODING_PCM_8BIT:
mAudioFormat = audioFormat;
break;
default:
throw new IllegalArgumentException("Unsupported sample encoding."
+ " Should be ENCODING_PCM_8BIT, ENCODING_PCM_16BIT, or ENCODING_PCM_FLOAT.");
}
}
// Convenience method for the contructor's audio buffer size check.
// preconditions:
// mChannelCount is valid
// mAudioFormat is AudioFormat.ENCODING_PCM_8BIT, AudioFormat.ENCODING_PCM_16BIT,
// or AudioFormat.ENCODING_PCM_FLOAT
// postcondition:
// mNativeBufferSizeInBytes is valid (multiple of frame size, positive)
private void audioBuffSizeCheck(int audioBufferSize) throws IllegalArgumentException {
// NB: this section is only valid with PCM data.
// To update when supporting compressed formats
int frameSizeInBytes = mChannelCount
* (AudioFormat.getBytesPerSample(mAudioFormat));
if ((audioBufferSize % frameSizeInBytes != 0) || (audioBufferSize < 1)) {
throw new IllegalArgumentException("Invalid audio buffer size.");
}
mNativeBufferSizeInBytes = audioBufferSize;
}
/**
* Releases the native AudioRecord resources.
* The object can no longer be used and the reference should be set to null
* after a call to release()
*/
public void release() {
try {
stop();
} catch(IllegalStateException ise) {
// don't raise an exception, we're releasing the resources.
}
native_release();
mState = STATE_UNINITIALIZED;
}
@Override
protected void finalize() {
// will cause stop() to be called, and if appropriate, will handle fixed volume recording
release();
}
//--------------------------------------------------------------------------
// Getters
//--------------------
/**
* Returns the configured audio sink sample rate in Hz.
* The sink sample rate never changes after construction.
* If the constructor had a specific sample rate, then the sink sample rate is that value.
* If the constructor had {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED},
* then the sink sample rate is a route-dependent default value based on the source [sic].
*/
public int getSampleRate() {
return mSampleRate;
}
/**
* Returns the audio recording source.
* @see MediaRecorder.AudioSource
*/
public int getAudioSource() {
return mRecordSource;
}
/**
* Returns the configured audio data encoding. See {@link AudioFormat#ENCODING_PCM_8BIT},
* {@link AudioFormat#ENCODING_PCM_16BIT}, and {@link AudioFormat#ENCODING_PCM_FLOAT}.
*/
public int getAudioFormat() {
return mAudioFormat;
}
/**
* Returns the configured channel position mask.
* AudioRecord
format.
* @return an {@link AudioFormat} containing the
* AudioRecord
parameters at the time of configuration.
*/
public @NonNull AudioFormat getFormat() {
AudioFormat.Builder builder = new AudioFormat.Builder()
.setSampleRate(mSampleRate)
.setEncoding(mAudioFormat);
if (mChannelMask != AudioFormat.CHANNEL_INVALID) {
builder.setChannelMask(mChannelMask);
}
if (mChannelIndexMask != AudioFormat.CHANNEL_INVALID /* 0 */) {
builder.setChannelIndexMask(mChannelIndexMask);
}
return builder.build();
}
/**
* Returns the configured number of channels.
*/
public int getChannelCount() {
return mChannelCount;
}
/**
* Returns the state of the AudioRecord instance. This is useful after the
* AudioRecord instance has been created to check if it was initialized
* properly. This ensures that the appropriate hardware resources have been
* acquired.
* @see AudioRecord#STATE_INITIALIZED
* @see AudioRecord#STATE_UNINITIALIZED
*/
public int getState() {
return mState;
}
/**
* Returns the recording state of the AudioRecord instance.
* @see AudioRecord#RECORDSTATE_STOPPED
* @see AudioRecord#RECORDSTATE_RECORDING
*/
public int getRecordingState() {
synchronized (mRecordingStateLock) {
return mRecordingState;
}
}
/**
* Returns the frame count of the native AudioRecord
buffer.
* This is greater than or equal to the bufferSizeInBytes converted to frame units
* specified in the AudioRecord
constructor or Builder.
* The native frame count may be enlarged to accommodate the requirements of the
* source on creation or if the AudioRecord
* is subsequently rerouted.
* @return current size in frames of the AudioRecord
buffer.
* @throws IllegalStateException
*/
public int getBufferSizeInFrames() {
return native_get_buffer_size_in_frames();
}
/**
* Returns the notification marker position expressed in frames.
*/
public int getNotificationMarkerPosition() {
return native_get_marker_pos();
}
/**
* Returns the notification update period expressed in frames.
*/
public int getPositionNotificationPeriod() {
return native_get_pos_update_period();
}
/**
* Poll for an {@link AudioTimestamp} on demand.
*
*
*/
public int read(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes) {
return read(audioData, offsetInBytes, sizeInBytes, READ_BLOCKING);
}
/**
* Reads audio data from the audio hardware for recording into a byte array.
* The format specified in the AudioRecord constructor should be
* {@link AudioFormat#ENCODING_PCM_8BIT} to correspond to the data in the array.
* The format can be {@link AudioFormat#ENCODING_PCM_16BIT}, but this is deprecated.
* @param audioData the array to which the recorded audio data is written.
* @param offsetInBytes index in audioData to which the data is written expressed in bytes.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @param sizeInBytes the number of requested bytes.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @param readMode one of {@link #READ_BLOCKING}, {@link #READ_NON_BLOCKING}.
*
With {@link #READ_BLOCKING}, the read will block until all the requested data
* is read.
*
With {@link #READ_NON_BLOCKING}, the read will return immediately after
* reading as much audio data as possible without blocking.
* @return zero or the positive number of bytes that were read, or one of the following
* error codes. The number of bytes will be a multiple of the frame size in bytes
* not to exceed sizeInBytes.
*
*
*/
public int read(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes,
@ReadMode int readMode) {
if (mState != STATE_INITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) {
return ERROR_INVALID_OPERATION;
}
if ((readMode != READ_BLOCKING) && (readMode != READ_NON_BLOCKING)) {
Log.e(TAG, "AudioRecord.read() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
|| (offsetInBytes + sizeInBytes < 0) // detect integer overflow
|| (offsetInBytes + sizeInBytes > audioData.length)) {
return ERROR_BAD_VALUE;
}
return native_read_in_byte_array(audioData, offsetInBytes, sizeInBytes,
readMode == READ_BLOCKING);
}
/**
* Reads audio data from the audio hardware for recording into a short array.
* The format specified in the AudioRecord constructor should be
* {@link AudioFormat#ENCODING_PCM_16BIT} to correspond to the data in the array.
* @param audioData the array to which the recorded audio data is written.
* @param offsetInShorts index in audioData to which the data is written expressed in shorts.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @param sizeInShorts the number of requested shorts.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @return zero or the positive number of shorts that were read, or one of the following
* error codes. The number of shorts will be a multiple of the channel count not to exceed
* sizeInShorts.
*
*
*/
public int read(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts) {
return read(audioData, offsetInShorts, sizeInShorts, READ_BLOCKING);
}
/**
* Reads audio data from the audio hardware for recording into a short array.
* The format specified in the AudioRecord constructor should be
* {@link AudioFormat#ENCODING_PCM_16BIT} to correspond to the data in the array.
* @param audioData the array to which the recorded audio data is written.
* @param offsetInShorts index in audioData from which the data is written expressed in shorts.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @param sizeInShorts the number of requested shorts.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @param readMode one of {@link #READ_BLOCKING}, {@link #READ_NON_BLOCKING}.
*
With {@link #READ_BLOCKING}, the read will block until all the requested data
* is read.
*
With {@link #READ_NON_BLOCKING}, the read will return immediately after
* reading as much audio data as possible without blocking.
* @return zero or the positive number of shorts that were read, or one of the following
* error codes. The number of shorts will be a multiple of the channel count not to exceed
* sizeInShorts.
*
*
*/
public int read(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts,
@ReadMode int readMode) {
if (mState != STATE_INITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) {
return ERROR_INVALID_OPERATION;
}
if ((readMode != READ_BLOCKING) && (readMode != READ_NON_BLOCKING)) {
Log.e(TAG, "AudioRecord.read() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0)
|| (offsetInShorts + sizeInShorts < 0) // detect integer overflow
|| (offsetInShorts + sizeInShorts > audioData.length)) {
return ERROR_BAD_VALUE;
}
return native_read_in_short_array(audioData, offsetInShorts, sizeInShorts,
readMode == READ_BLOCKING);
}
/**
* Reads audio data from the audio hardware for recording into a float array.
* The format specified in the AudioRecord constructor should be
* {@link AudioFormat#ENCODING_PCM_FLOAT} to correspond to the data in the array.
* @param audioData the array to which the recorded audio data is written.
* @param offsetInFloats index in audioData from which the data is written.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @param sizeInFloats the number of requested floats.
* Must not be negative, or cause the data access to go out of bounds of the array.
* @param readMode one of {@link #READ_BLOCKING}, {@link #READ_NON_BLOCKING}.
*
With {@link #READ_BLOCKING}, the read will block until all the requested data
* is read.
*
With {@link #READ_NON_BLOCKING}, the read will return immediately after
* reading as much audio data as possible without blocking.
* @return zero or the positive number of floats that were read, or one of the following
* error codes. The number of floats will be a multiple of the channel count not to exceed
* sizeInFloats.
*
*
*/
public int read(@NonNull float[] audioData, int offsetInFloats, int sizeInFloats,
@ReadMode int readMode) {
if (mState == STATE_UNINITIALIZED) {
Log.e(TAG, "AudioRecord.read() called in invalid state STATE_UNINITIALIZED");
return ERROR_INVALID_OPERATION;
}
if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) {
Log.e(TAG, "AudioRecord.read(float[] ...) requires format ENCODING_PCM_FLOAT");
return ERROR_INVALID_OPERATION;
}
if ((readMode != READ_BLOCKING) && (readMode != READ_NON_BLOCKING)) {
Log.e(TAG, "AudioRecord.read() 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)) {
return ERROR_BAD_VALUE;
}
return native_read_in_float_array(audioData, offsetInFloats, sizeInFloats,
readMode == READ_BLOCKING);
}
/**
* Reads audio data from the audio hardware for recording into a direct buffer. If this buffer
* is not a direct buffer, this method will always return 0.
* Note that the value returned by {@link java.nio.Buffer#position()} on this buffer is
* unchanged after a call to this method.
* The representation of the data in the buffer will depend on the format specified in
* the AudioRecord constructor, and will be native endian.
* @param audioBuffer the direct buffer to which the recorded audio data is written.
* Data is written to audioBuffer.position().
* @param sizeInBytes the number of requested bytes. It is recommended but not enforced
* that the number of bytes requested be a multiple of the frame size (sample size in
* bytes multiplied by the channel count).
* @return zero or the positive number of bytes that were read, or one of the following
* error codes. The number of bytes will not exceed sizeInBytes and will be truncated to be
* a multiple of the frame size.
*
*
*/
public int read(@NonNull ByteBuffer audioBuffer, int sizeInBytes) {
return read(audioBuffer, sizeInBytes, READ_BLOCKING);
}
/**
* Reads audio data from the audio hardware for recording into a direct buffer. If this buffer
* is not a direct buffer, this method will always return 0.
* Note that the value returned by {@link java.nio.Buffer#position()} on this buffer is
* unchanged after a call to this method.
* The representation of the data in the buffer will depend on the format specified in
* the AudioRecord constructor, and will be native endian.
* @param audioBuffer the direct buffer to which the recorded audio data is written.
* Data is written to audioBuffer.position().
* @param sizeInBytes the number of requested bytes. It is recommended but not enforced
* that the number of bytes requested be a multiple of the frame size (sample size in
* bytes multiplied by the channel count).
* @param readMode one of {@link #READ_BLOCKING}, {@link #READ_NON_BLOCKING}.
*
With {@link #READ_BLOCKING}, the read will block until all the requested data
* is read.
*
With {@link #READ_NON_BLOCKING}, the read will return immediately after
* reading as much audio data as possible without blocking.
* @return zero or the positive number of bytes that were read, or one of the following
* error codes. The number of bytes will not exceed sizeInBytes and will be truncated to be
* a multiple of the frame size.
*
*
*/
public int read(@NonNull ByteBuffer audioBuffer, int sizeInBytes, @ReadMode int readMode) {
if (mState != STATE_INITIALIZED) {
return ERROR_INVALID_OPERATION;
}
if ((readMode != READ_BLOCKING) && (readMode != READ_NON_BLOCKING)) {
Log.e(TAG, "AudioRecord.read() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioBuffer == null) || (sizeInBytes < 0) ) {
return ERROR_BAD_VALUE;
}
return native_read_in_direct_buffer(audioBuffer, sizeInBytes, readMode == READ_BLOCKING);
}
//--------------------------------------------------------------------------
// Initialization / configuration
//--------------------
/**
* Sets the listener the AudioRecord notifies when a previously set marker is reached or
* for each periodic record head position update.
* @param listener
*/
public void setRecordPositionUpdateListener(OnRecordPositionUpdateListener listener) {
setRecordPositionUpdateListener(listener, null);
}
/**
* Sets the listener the AudioRecord notifies when a previously set marker is reached or
* for each periodic record head position update.
* Use this method to receive AudioRecord events in the Handler associated with another
* thread than the one in which you created the AudioRecord instance.
* @param listener
* @param handler the Handler that will receive the event notification messages.
*/
public void setRecordPositionUpdateListener(OnRecordPositionUpdateListener listener,
Handler handler) {
synchronized (mPositionListenerLock) {
mPositionListener = listener;
if (listener != null) {
if (handler != null) {
mEventHandler = new NativeEventHandler(this, handler.getLooper());
} else {
// no given handler, use the looper the AudioRecord was created in
mEventHandler = new NativeEventHandler(this, mInitializationLooper);
}
} else {
mEventHandler = null;
}
}
}
/**
* Sets the marker position at which the listener is called, if set with
* {@link #setRecordPositionUpdateListener(OnRecordPositionUpdateListener)} or
* {@link #setRecordPositionUpdateListener(OnRecordPositionUpdateListener, Handler)}.
* @param markerInFrames marker position expressed in frames
* @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);
}
/**
* Returns an {@link AudioDeviceInfo} identifying the current routing of this AudioRecord.
* Note: The query is only valid if the AudioRecord is currently recording. If it is not,
* getRoutedDevice()
will return null.
*/
@Override
public AudioDeviceInfo getRoutedDevice() {
int deviceId = native_getRoutedDeviceId();
if (deviceId == 0) {
return null;
}
AudioDeviceInfo[] devices =
AudioManager.getDevicesStatic(AudioManager.GET_DEVICES_INPUTS);
for (int i = 0; i < devices.length; i++) {
if (devices[i].getId() == deviceId) {
return devices[i];
}
}
return null;
}
/*
* Call BEFORE adding a routing callback handler.
*/
private void testEnableNativeRoutingCallbacksLocked() {
if (mRoutingChangeListeners.size() == 0) {
native_enableDeviceCallback();
}
}
/*
* Call AFTER removing a routing callback handler.
*/
private void testDisableNativeRoutingCallbacksLocked() {
if (mRoutingChangeListeners.size() == 0) {
native_disableDeviceCallback();
}
}
//--------------------------------------------------------------------------
// (Re)Routing Info
//--------------------
/**
* The list of AudioRouting.OnRoutingChangedListener interfaces added (with
* {@link AudioRecord#addOnRoutingChangedListener} by an app to receive
* (re)routing notifications.
*/
@GuardedBy("mRoutingChangeListeners")
private ArrayMapnull
, the {@link Handler} associated with the main
* {@link Looper} will be used.
*/
@Override
public void addOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener,
android.os.Handler handler) {
synchronized (mRoutingChangeListeners) {
if (listener != null && !mRoutingChangeListeners.containsKey(listener)) {
testEnableNativeRoutingCallbacksLocked();
mRoutingChangeListeners.put(
listener, new NativeRoutingEventHandlerDelegate(this, listener,
handler != null ? handler : new Handler(mInitializationLooper)));
}
}
}
/**
* Removes an {@link AudioRouting.OnRoutingChangedListener} which has been previously added
* to receive rerouting notifications.
* @param listener The previously added {@link AudioRouting.OnRoutingChangedListener} interface
* to remove.
*/
@Override
public void removeOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener) {
synchronized (mRoutingChangeListeners) {
if (mRoutingChangeListeners.containsKey(listener)) {
mRoutingChangeListeners.remove(listener);
testDisableNativeRoutingCallbacksLocked();
}
}
}
//--------------------------------------------------------------------------
// (Re)Routing Info
//--------------------
/**
* Defines the interface by which applications can receive notifications of
* routing changes for the associated {@link AudioRecord}.
*
* @deprecated users should switch to the general purpose
* {@link AudioRouting.OnRoutingChangedListener} class instead.
*/
@Deprecated
public interface OnRoutingChangedListener extends AudioRouting.OnRoutingChangedListener {
/**
* Called when the routing of an AudioRecord changes from either and
* explicit or policy rerouting. Use {@link #getRoutedDevice()} to
* retrieve the newly routed-from device.
*/
public void onRoutingChanged(AudioRecord audioRecord);
@Override
default public void onRoutingChanged(AudioRouting router) {
if (router instanceof AudioRecord) {
onRoutingChanged((AudioRecord) router);
}
}
}
/**
* Adds an {@link OnRoutingChangedListener} to receive notifications of routing changes
* on this AudioRecord.
* @param listener The {@link OnRoutingChangedListener} interface to receive notifications
* of rerouting events.
* @param handler Specifies the {@link Handler} object for the thread on which to execute
* the callback. If null
, the {@link Handler} associated with the main
* {@link Looper} will be used.
* @deprecated users should switch to the general purpose
* {@link AudioRouting.OnRoutingChangedListener} class instead.
*/
@Deprecated
public void addOnRoutingChangedListener(OnRoutingChangedListener listener,
android.os.Handler handler) {
addOnRoutingChangedListener((AudioRouting.OnRoutingChangedListener) listener, handler);
}
/**
* Removes an {@link OnRoutingChangedListener} which has been previously added
* to receive rerouting notifications.
* @param listener The previously added {@link OnRoutingChangedListener} interface to remove.
* @deprecated users should switch to the general purpose
* {@link AudioRouting.OnRoutingChangedListener} class instead.
*/
@Deprecated
public void removeOnRoutingChangedListener(OnRoutingChangedListener listener) {
removeOnRoutingChangedListener((AudioRouting.OnRoutingChangedListener) listener);
}
/**
* Helper class to handle the forwarding of native events to the appropriate listener
* (potentially) handled in a different thread
*/
private class NativeRoutingEventHandlerDelegate {
private final Handler mHandler;
NativeRoutingEventHandlerDelegate(final AudioRecord record,
final AudioRouting.OnRoutingChangedListener 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 AudioRecord 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 (record == null) {
return;
}
switch(msg.what) {
case AudioSystem.NATIVE_EVENT_ROUTING_CHANGE:
if (listener != null) {
listener.onRoutingChanged(record);
}
break;
default:
loge("Unknown native event type: " + msg.what);
break;
}
}
};
} else {
mHandler = null;
}
}
Handler getHandler() {
return mHandler;
}
}
/**
* Sends device list change notification to all listeners.
*/
private void broadcastRoutingChange() {
AudioManager.resetAudioPortGeneration();
synchronized (mRoutingChangeListeners) {
for (NativeRoutingEventHandlerDelegate delegate : mRoutingChangeListeners.values()) {
Handler handler = delegate.getHandler();
if (handler != null) {
handler.sendEmptyMessage(AudioSystem.NATIVE_EVENT_ROUTING_CHANGE);
}
}
}
}
/**
* Sets the period at which the listener is called, if set with
* {@link #setRecordPositionUpdateListener(OnRecordPositionUpdateListener)} or
* {@link #setRecordPositionUpdateListener(OnRecordPositionUpdateListener, Handler)}.
* It is possible for notifications to be lost if the period is too small.
* @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);
}
//--------------------------------------------------------------------------
// Explicit Routing
//--------------------
private AudioDeviceInfo mPreferredDevice = null;
/**
* Specifies an audio device (via an {@link AudioDeviceInfo} object) to route
* the input to this AudioRecord.
* @param deviceInfo The {@link AudioDeviceInfo} specifying the audio source.
* If deviceInfo is null, default routing is restored.
* @return true if successful, false if the specified {@link AudioDeviceInfo} is non-null and
* does not correspond to a valid audio input device.
*/
@Override
public boolean setPreferredDevice(AudioDeviceInfo deviceInfo) {
// Do some validation....
if (deviceInfo != null && !deviceInfo.isSource()) {
return false;
}
int preferredDeviceId = deviceInfo != null ? deviceInfo.getId() : 0;
boolean status = native_setInputDevice(preferredDeviceId);
if (status == true) {
synchronized (this) {
mPreferredDevice = deviceInfo;
}
}
return status;
}
/**
* Returns the selected input specified by {@link #setPreferredDevice}. Note that this
* is not guarenteed to correspond to the actual device being used for recording.
*/
@Override
public AudioDeviceInfo getPreferredDevice() {
synchronized (this) {
return mPreferredDevice;
}
}
//---------------------------------------------------------
// Interface definitions
//--------------------
/**
* Interface definition for a callback to be invoked when an AudioRecord has
* reached a notification marker set by {@link AudioRecord#setNotificationMarkerPosition(int)}
* or for periodic updates on the progress of the record head, as set by
* {@link AudioRecord#setPositionNotificationPeriod(int)}.
*/
public interface OnRecordPositionUpdateListener {
/**
* Called on the listener to notify it that the previously set marker has been reached
* by the recording head.
*/
void onMarkerReached(AudioRecord recorder);
/**
* Called on the listener to periodically notify it that the record head has reached
* a multiple of the notification period.
*/
void onPeriodicNotification(AudioRecord recorder);
}
//---------------------------------------------------------
// Inner classes
//--------------------
/**
* Helper class to handle the forwarding of native events to the appropriate listener
* (potentially) handled in a different thread
*/
private class NativeEventHandler extends Handler {
private final AudioRecord mAudioRecord;
NativeEventHandler(AudioRecord recorder, Looper looper) {
super(looper);
mAudioRecord = recorder;
}
@Override
public void handleMessage(Message msg) {
OnRecordPositionUpdateListener listener = null;
synchronized (mPositionListenerLock) {
listener = mAudioRecord.mPositionListener;
}
switch (msg.what) {
case NATIVE_EVENT_MARKER:
if (listener != null) {
listener.onMarkerReached(mAudioRecord);
}
break;
case NATIVE_EVENT_NEW_POS:
if (listener != null) {
listener.onPeriodicNotification(mAudioRecord);
}
break;
default:
loge("Unknown native event type: " + msg.what);
break;
}
}
}
//---------------------------------------------------------
// Java methods called from the native side
//--------------------
@SuppressWarnings("unused")
private static void postEventFromNative(Object audiorecord_ref,
int what, int arg1, int arg2, Object obj) {
//logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2);
AudioRecord recorder = (AudioRecord)((WeakReference)audiorecord_ref).get();
if (recorder == null) {
return;
}
if (what == AudioSystem.NATIVE_EVENT_ROUTING_CHANGE) {
recorder.broadcastRoutingChange();
return;
}
if (recorder.mEventHandler != null) {
Message m =
recorder.mEventHandler.obtainMessage(what, arg1, arg2, obj);
recorder.mEventHandler.sendMessage(m);
}
}
//---------------------------------------------------------
// Native methods called from the Java side
//--------------------
private native final int native_setup(Object audiorecord_this,
Object /*AudioAttributes*/ attributes,
int[] sampleRate, int channelMask, int channelIndexMask, int audioFormat,
int buffSizeInBytes, int[] sessionId, String opPackageName,
long nativeRecordInJavaObj);
// TODO remove: implementation calls directly into implementation of native_release()
private native final void native_finalize();
/**
* @hide
*/
public native final void native_release();
private native final int native_start(int syncEvent, int sessionId);
private native final void native_stop();
private native final int native_read_in_byte_array(byte[] audioData,
int offsetInBytes, int sizeInBytes, boolean isBlocking);
private native final int native_read_in_short_array(short[] audioData,
int offsetInShorts, int sizeInShorts, boolean isBlocking);
private native final int native_read_in_float_array(float[] audioData,
int offsetInFloats, int sizeInFloats, boolean isBlocking);
private native final int native_read_in_direct_buffer(Object jBuffer,
int sizeInBytes, boolean isBlocking);
private native final int native_get_buffer_size_in_frames();
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();
static private native final int native_get_min_buff_size(
int sampleRateInHz, int channelCount, int audioFormat);
private native final boolean native_setInputDevice(int deviceId);
private native final int native_getRoutedDeviceId();
private native final void native_enableDeviceCallback();
private native final void native_disableDeviceCallback();
private native final int native_get_timestamp(@NonNull AudioTimestamp outTimestamp,
@AudioTimestamp.Timebase int timebase);
//---------------------------------------------------------
// Utility methods
//------------------
private static void logd(String msg) {
Log.d(TAG, msg);
}
private static void loge(String msg) {
Log.e(TAG, msg);
}
}