modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java - src - Git at Google

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 package org.webrtc.voiceengine;

 import android.media.AudioFormat;
 import android.media.AudioRecord;
 import android.media.MediaRecorder.AudioSource;
 import android.os.Build;
 import android.os.Process;
 import androidx.annotation.Nullable;
 import java.lang.System;
 import java.nio.ByteBuffer;
 import java.util.Arrays;
 import java.util.concurrent.TimeUnit;
 import org.webrtc.Logging;
 import org.webrtc.ThreadUtils;

 public class WebRtcAudioRecord {
   private static final boolean DEBUG = false;

   private static final String TAG = "WebRtcAudioRecord";

   // Default audio data format is PCM 16 bit per sample.
   // Guaranteed to be supported by all devices.
   private static final int BITS_PER_SAMPLE = 16;

   // Requested size of each recorded buffer provided to the client.
   private static final int CALLBACK_BUFFER_SIZE_MS = 10;

   // Average number of callbacks per second.
   private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;

   // We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
   // buffer size). The extra space is allocated to guard against glitches under
   // high load.
   private static final int BUFFER_SIZE_FACTOR = 2;

   // The AudioRecordJavaThread is allowed to wait for successful call to join()
   // but the wait times out afther this amount of time.
   private static final long AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS = 2000;

   private static final int DEFAULT_AUDIO_SOURCE = getDefaultAudioSource();
   private static int audioSource = DEFAULT_AUDIO_SOURCE;

   private final long nativeAudioRecord;

   private @Nullable WebRtcAudioEffects effects;

   private ByteBuffer byteBuffer;

   private @Nullable AudioRecord audioRecord;
   private @Nullable AudioRecordThread audioThread;

   private static volatile boolean microphoneMute;
   private byte[] emptyBytes;

   // Audio recording error handler functions.
   public enum AudioRecordStartErrorCode {
     AUDIO_RECORD_START_EXCEPTION,
     AUDIO_RECORD_START_STATE_MISMATCH,
   }

   public static interface WebRtcAudioRecordErrorCallback {
     void onWebRtcAudioRecordInitError(String errorMessage);
     void onWebRtcAudioRecordStartError(AudioRecordStartErrorCode errorCode, String errorMessage);
     void onWebRtcAudioRecordError(String errorMessage);
   }

   private static @Nullable WebRtcAudioRecordErrorCallback errorCallback;

   public static void setErrorCallback(WebRtcAudioRecordErrorCallback errorCallback) {
     Logging.d(TAG, "Set error callback");
     WebRtcAudioRecord.errorCallback = errorCallback;
   }

   /**
    * Contains audio sample information. Object is passed using {@link
    * WebRtcAudioRecord.WebRtcAudioRecordSamplesReadyCallback}
    */
   public static class AudioSamples {
     /** See {@link AudioRecord#getAudioFormat()} */
     private final int audioFormat;
     /** See {@link AudioRecord#getChannelCount()} */
     private final int channelCount;
     /** See {@link AudioRecord#getSampleRate()} */
     private final int sampleRate;

     private final byte[] data;

     private AudioSamples(AudioRecord audioRecord, byte[] data) {
       this.audioFormat = audioRecord.getAudioFormat();
       this.channelCount = audioRecord.getChannelCount();
       this.sampleRate = audioRecord.getSampleRate();
       this.data = data;
     }

     public int getAudioFormat() {
       return audioFormat;
     }

     public int getChannelCount() {
       return channelCount;
     }

     public int getSampleRate() {
       return sampleRate;
     }

     public byte[] getData() {
       return data;
     }
   }

   /** Called when new audio samples are ready. This should only be set for debug purposes */
   public static interface WebRtcAudioRecordSamplesReadyCallback {
     void onWebRtcAudioRecordSamplesReady(AudioSamples samples);
   }

   private static @Nullable WebRtcAudioRecordSamplesReadyCallback audioSamplesReadyCallback;

   public static void setOnAudioSamplesReady(WebRtcAudioRecordSamplesReadyCallback callback) {
     audioSamplesReadyCallback = callback;
   }

   /**
    * Audio thread which keeps calling ByteBuffer.read() waiting for audio
    * to be recorded. Feeds recorded data to the native counterpart as a
    * periodic sequence of callbacks using DataIsRecorded().
    * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
    */
   private class AudioRecordThread extends Thread {
     private volatile boolean keepAlive = true;

     public AudioRecordThread(String name) {
       super(name);
     }

     // TODO(titovartem) make correct fix during webrtc:9175
     @SuppressWarnings("ByteBufferBackingArray")
     @Override
     public void run() {
       Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
       Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
       assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);

       long lastTime = System.nanoTime();
       while (keepAlive) {
         int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
         if (bytesRead == byteBuffer.capacity()) {
           if (microphoneMute) {
             byteBuffer.clear();
             byteBuffer.put(emptyBytes);
           }
           // It's possible we've been shut down during the read, and stopRecording() tried and
           // failed to join this thread. To be a bit safer, try to avoid calling any native methods
           // in case they've been unregistered after stopRecording() returned.
           if (keepAlive) {
             nativeDataIsRecorded(bytesRead, nativeAudioRecord);
           }
           if (audioSamplesReadyCallback != null) {
             // Copy the entire byte buffer array.  Assume that the start of the byteBuffer is
             // at index 0.
             byte[] data = Arrays.copyOf(byteBuffer.array(), byteBuffer.capacity());
             audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
                 new AudioSamples(audioRecord, data));
           }
         } else {
           String errorMessage = "AudioRecord.read failed: " + bytesRead;
           Logging.e(TAG, errorMessage);
           if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
             keepAlive = false;
             reportWebRtcAudioRecordError(errorMessage);
           }
         }
         if (DEBUG) {
           long nowTime = System.nanoTime();
           long durationInMs = TimeUnit.NANOSECONDS.toMillis((nowTime - lastTime));
           lastTime = nowTime;
           Logging.d(TAG, "bytesRead[" + durationInMs + "] " + bytesRead);
         }
       }

       try {
         if (audioRecord != null) {
           audioRecord.stop();
         }
       } catch (IllegalStateException e) {
         Logging.e(TAG, "AudioRecord.stop failed: " + e.getMessage());
       }
     }

     // Stops the inner thread loop and also calls AudioRecord.stop().
     // Does not block the calling thread.
     public void stopThread() {
       Logging.d(TAG, "stopThread");
       keepAlive = false;
     }
   }

   WebRtcAudioRecord(long nativeAudioRecord) {
     Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
     this.nativeAudioRecord = nativeAudioRecord;
     if (DEBUG) {
       WebRtcAudioUtils.logDeviceInfo(TAG);
     }
     effects = WebRtcAudioEffects.create();
   }

   private boolean enableBuiltInAEC(boolean enable) {
     Logging.d(TAG, "enableBuiltInAEC(" + enable + ')');
     if (effects == null) {
       Logging.e(TAG, "Built-in AEC is not supported on this platform");
       return false;
     }
     return effects.setAEC(enable);
   }

   private boolean enableBuiltInNS(boolean enable) {
     Logging.d(TAG, "enableBuiltInNS(" + enable + ')');
     if (effects == null) {
       Logging.e(TAG, "Built-in NS is not supported on this platform");
       return false;
     }
     return effects.setNS(enable);
   }

   private int initRecording(int sampleRate, int channels) {
     Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
     if (audioRecord != null) {
       reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
       return -1;
     }
     final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
     final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
     byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
     Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
     emptyBytes = new byte[byteBuffer.capacity()];
     // Rather than passing the ByteBuffer with every callback (requiring
     // the potentially expensive GetDirectBufferAddress) we simply have the
     // the native class cache the address to the memory once.
     nativeCacheDirectBufferAddress(byteBuffer, nativeAudioRecord);

     // Get 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.
     final int channelConfig = channelCountToConfiguration(channels);
     int minBufferSize =
         AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
     if (minBufferSize == AudioRecord.ERROR || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
       reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
       return -1;
     }
     Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);

     // Use a larger buffer size than the minimum required when creating the
     // AudioRecord instance to ensure smooth recording under load. It has been
     // verified that it does not increase the actual recording latency.
     int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
     Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
     try {
       audioRecord = new AudioRecord(audioSource, sampleRate, channelConfig,
           AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes);
     } catch (IllegalArgumentException e) {
       reportWebRtcAudioRecordInitError("AudioRecord ctor error: " + e.getMessage());
       releaseAudioResources();
       return -1;
     }
     if (audioRecord == null || audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
       reportWebRtcAudioRecordInitError("Failed to create a new AudioRecord instance");
       releaseAudioResources();
       return -1;
     }
     if (effects != null) {
       effects.enable(audioRecord.getAudioSessionId());
     }
     logMainParameters();
     logMainParametersExtended();
     return framesPerBuffer;
   }

   private boolean startRecording() {
     Logging.d(TAG, "startRecording");
     assertTrue(audioRecord != null);
     assertTrue(audioThread == null);
     try {
       audioRecord.startRecording();
     } catch (IllegalStateException e) {
       reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_EXCEPTION,
           "AudioRecord.startRecording failed: " + e.getMessage());
       return false;
     }
     if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
       reportWebRtcAudioRecordStartError(
           AudioRecordStartErrorCode.AUDIO_RECORD_START_STATE_MISMATCH,
           "AudioRecord.startRecording failed - incorrect state :"
           + audioRecord.getRecordingState());
       return false;
     }
     audioThread = new AudioRecordThread("AudioRecordJavaThread");
     audioThread.start();
     return true;
   }

   private boolean stopRecording() {
     Logging.d(TAG, "stopRecording");
     assertTrue(audioThread != null);
     audioThread.stopThread();
     if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
       Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
       WebRtcAudioUtils.logAudioState(TAG);
     }
     audioThread = null;
     if (effects != null) {
       effects.release();
     }
     releaseAudioResources();
     return true;
   }

   private void logMainParameters() {
     Logging.d(TAG, "AudioRecord: "
             + "session ID: " + audioRecord.getAudioSessionId() + ", "
             + "channels: " + audioRecord.getChannelCount() + ", "
             + "sample rate: " + audioRecord.getSampleRate());
   }

   private void logMainParametersExtended() {
     if (Build.VERSION.SDK_INT >= 23) {
       Logging.d(TAG, "AudioRecord: "
               // The frame count of the native AudioRecord buffer.
               + "buffer size in frames: " + audioRecord.getBufferSizeInFrames());
     }
   }

   // Helper method which throws an exception  when an assertion has failed.
   private static void assertTrue(boolean condition) {
     if (!condition) {
       throw new AssertionError("Expected condition to be true");
     }
   }

   private int channelCountToConfiguration(int channels) {
     return (channels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
   }

   private native void nativeCacheDirectBufferAddress(ByteBuffer byteBuffer, long nativeAudioRecord);

   private native void nativeDataIsRecorded(int bytes, long nativeAudioRecord);

   @SuppressWarnings("NoSynchronizedMethodCheck")
   public static synchronized void setAudioSource(int source) {
     Logging.w(TAG, "Audio source is changed from: " + audioSource
             + " to " + source);
     audioSource = source;
   }

   private static int getDefaultAudioSource() {
     return AudioSource.VOICE_COMMUNICATION;
   }

   // Sets all recorded samples to zero if `mute` is true, i.e., ensures that
   // the microphone is muted.
   public static void setMicrophoneMute(boolean mute) {
     Logging.w(TAG, "setMicrophoneMute(" + mute + ")");
     microphoneMute = mute;
   }

   // Releases the native AudioRecord resources.
   private void releaseAudioResources() {
     Logging.d(TAG, "releaseAudioResources");
     if (audioRecord != null) {
       audioRecord.release();
       audioRecord = null;
     }
   }

   private void reportWebRtcAudioRecordInitError(String errorMessage) {
     Logging.e(TAG, "Init recording error: " + errorMessage);
     WebRtcAudioUtils.logAudioState(TAG);
     if (errorCallback != null) {
       errorCallback.onWebRtcAudioRecordInitError(errorMessage);
     }
   }

   private void reportWebRtcAudioRecordStartError(
       AudioRecordStartErrorCode errorCode, String errorMessage) {
     Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
     WebRtcAudioUtils.logAudioState(TAG);
     if (errorCallback != null) {
       errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
     }
   }

   private void reportWebRtcAudioRecordError(String errorMessage) {
     Logging.e(TAG, "Run-time recording error: " + errorMessage);
     WebRtcAudioUtils.logAudioState(TAG);
     if (errorCallback != null) {
       errorCallback.onWebRtcAudioRecordError(errorMessage);
     }
   }
 }
	/*
	* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	package org.webrtc.voiceengine;

	import android.media.AudioFormat;
	import android.media.AudioRecord;
	import android.media.MediaRecorder.AudioSource;
	import android.os.Build;
	import android.os.Process;
	import androidx.annotation.Nullable;
	import java.lang.System;
	import java.nio.ByteBuffer;
	import java.util.Arrays;
	import java.util.concurrent.TimeUnit;
	import org.webrtc.Logging;
	import org.webrtc.ThreadUtils;

	public class WebRtcAudioRecord {
	private static final boolean DEBUG = false;

	private static final String TAG = "WebRtcAudioRecord";

	// Default audio data format is PCM 16 bit per sample.
	// Guaranteed to be supported by all devices.
	private static final int BITS_PER_SAMPLE = 16;

	// Requested size of each recorded buffer provided to the client.
	private static final int CALLBACK_BUFFER_SIZE_MS = 10;

	// Average number of callbacks per second.
	private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;

	// We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
	// buffer size). The extra space is allocated to guard against glitches under
	// high load.
	private static final int BUFFER_SIZE_FACTOR = 2;

	// The AudioRecordJavaThread is allowed to wait for successful call to join()
	// but the wait times out afther this amount of time.
	private static final long AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS = 2000;

	private static final int DEFAULT_AUDIO_SOURCE = getDefaultAudioSource();
	private static int audioSource = DEFAULT_AUDIO_SOURCE;

	private final long nativeAudioRecord;

	private @Nullable WebRtcAudioEffects effects;

	private ByteBuffer byteBuffer;

	private @Nullable AudioRecord audioRecord;
	private @Nullable AudioRecordThread audioThread;

	private static volatile boolean microphoneMute;
	private byte[] emptyBytes;

	// Audio recording error handler functions.
	public enum AudioRecordStartErrorCode {
	AUDIO_RECORD_START_EXCEPTION,
	AUDIO_RECORD_START_STATE_MISMATCH,
	}

	public static interface WebRtcAudioRecordErrorCallback {
	void onWebRtcAudioRecordInitError(String errorMessage);
	void onWebRtcAudioRecordStartError(AudioRecordStartErrorCode errorCode, String errorMessage);
	void onWebRtcAudioRecordError(String errorMessage);
	}

	private static @Nullable WebRtcAudioRecordErrorCallback errorCallback;

	public static void setErrorCallback(WebRtcAudioRecordErrorCallback errorCallback) {
	Logging.d(TAG, "Set error callback");
	WebRtcAudioRecord.errorCallback = errorCallback;
	}

	/**
	* Contains audio sample information. Object is passed using {@link
	* WebRtcAudioRecord.WebRtcAudioRecordSamplesReadyCallback}
	*/
	public static class AudioSamples {
	/** See {@link AudioRecord#getAudioFormat()} */
	private final int audioFormat;
	/** See {@link AudioRecord#getChannelCount()} */
	private final int channelCount;
	/** See {@link AudioRecord#getSampleRate()} */
	private final int sampleRate;

	private final byte[] data;

	private AudioSamples(AudioRecord audioRecord, byte[] data) {
	this.audioFormat = audioRecord.getAudioFormat();
	this.channelCount = audioRecord.getChannelCount();
	this.sampleRate = audioRecord.getSampleRate();
	this.data = data;
	}

	public int getAudioFormat() {
	return audioFormat;
	}

	public int getChannelCount() {
	return channelCount;
	}

	public int getSampleRate() {
	return sampleRate;
	}

	public byte[] getData() {
	return data;
	}
	}

	/** Called when new audio samples are ready. This should only be set for debug purposes */
	public static interface WebRtcAudioRecordSamplesReadyCallback {
	void onWebRtcAudioRecordSamplesReady(AudioSamples samples);
	}

	private static @Nullable WebRtcAudioRecordSamplesReadyCallback audioSamplesReadyCallback;

	public static void setOnAudioSamplesReady(WebRtcAudioRecordSamplesReadyCallback callback) {
	audioSamplesReadyCallback = callback;
	}

	/**
	* Audio thread which keeps calling ByteBuffer.read() waiting for audio
	* to be recorded. Feeds recorded data to the native counterpart as a
	* periodic sequence of callbacks using DataIsRecorded().
	* This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
	*/
	private class AudioRecordThread extends Thread {
	private volatile boolean keepAlive = true;

	public AudioRecordThread(String name) {
	super(name);
	}

	// TODO(titovartem) make correct fix during webrtc:9175
	@SuppressWarnings("ByteBufferBackingArray")
	@Override
	public void run() {
	Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
	Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
	assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);

	long lastTime = System.nanoTime();
	while (keepAlive) {
	int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
	if (bytesRead == byteBuffer.capacity()) {
	if (microphoneMute) {
	byteBuffer.clear();
	byteBuffer.put(emptyBytes);
	}
	// It's possible we've been shut down during the read, and stopRecording() tried and
	// failed to join this thread. To be a bit safer, try to avoid calling any native methods
	// in case they've been unregistered after stopRecording() returned.
	if (keepAlive) {
	nativeDataIsRecorded(bytesRead, nativeAudioRecord);
	}
	if (audioSamplesReadyCallback != null) {
	// Copy the entire byte buffer array. Assume that the start of the byteBuffer is
	// at index 0.
	byte[] data = Arrays.copyOf(byteBuffer.array(), byteBuffer.capacity());
	audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
	new AudioSamples(audioRecord, data));
	}
	} else {
	String errorMessage = "AudioRecord.read failed: " + bytesRead;
	Logging.e(TAG, errorMessage);
	if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
	keepAlive = false;
	reportWebRtcAudioRecordError(errorMessage);
	}
	}
	if (DEBUG) {
	long nowTime = System.nanoTime();
	long durationInMs = TimeUnit.NANOSECONDS.toMillis((nowTime - lastTime));
	lastTime = nowTime;
	Logging.d(TAG, "bytesRead[" + durationInMs + "] " + bytesRead);
	}
	}

	try {
	if (audioRecord != null) {
	audioRecord.stop();
	}
	} catch (IllegalStateException e) {
	Logging.e(TAG, "AudioRecord.stop failed: " + e.getMessage());
	}
	}

	// Stops the inner thread loop and also calls AudioRecord.stop().
	// Does not block the calling thread.
	public void stopThread() {
	Logging.d(TAG, "stopThread");
	keepAlive = false;
	}
	}

	WebRtcAudioRecord(long nativeAudioRecord) {
	Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
	this.nativeAudioRecord = nativeAudioRecord;
	if (DEBUG) {
	WebRtcAudioUtils.logDeviceInfo(TAG);
	}
	effects = WebRtcAudioEffects.create();
	}

	private boolean enableBuiltInAEC(boolean enable) {
	Logging.d(TAG, "enableBuiltInAEC(" + enable + ')');
	if (effects == null) {
	Logging.e(TAG, "Built-in AEC is not supported on this platform");
	return false;
	}
	return effects.setAEC(enable);
	}

	private boolean enableBuiltInNS(boolean enable) {
	Logging.d(TAG, "enableBuiltInNS(" + enable + ')');
	if (effects == null) {
	Logging.e(TAG, "Built-in NS is not supported on this platform");
	return false;
	}
	return effects.setNS(enable);
	}

	private int initRecording(int sampleRate, int channels) {
	Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
	if (audioRecord != null) {
	reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
	return -1;
	}
	final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
	final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
	byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
	Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
	emptyBytes = new byte[byteBuffer.capacity()];
	// Rather than passing the ByteBuffer with every callback (requiring
	// the potentially expensive GetDirectBufferAddress) we simply have the
	// the native class cache the address to the memory once.
	nativeCacheDirectBufferAddress(byteBuffer, nativeAudioRecord);

	// Get 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.
	final int channelConfig = channelCountToConfiguration(channels);
	int minBufferSize =
	AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
	if (minBufferSize == AudioRecord.ERROR \|\| minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
	reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
	return -1;
	}
	Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);

	// Use a larger buffer size than the minimum required when creating the
	// AudioRecord instance to ensure smooth recording under load. It has been
	// verified that it does not increase the actual recording latency.
	int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
	Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
	try {
	audioRecord = new AudioRecord(audioSource, sampleRate, channelConfig,
	AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes);
	} catch (IllegalArgumentException e) {
	reportWebRtcAudioRecordInitError("AudioRecord ctor error: " + e.getMessage());
	releaseAudioResources();
	return -1;
	}
	if (audioRecord == null \|\| audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
	reportWebRtcAudioRecordInitError("Failed to create a new AudioRecord instance");
	releaseAudioResources();
	return -1;
	}
	if (effects != null) {
	effects.enable(audioRecord.getAudioSessionId());
	}
	logMainParameters();
	logMainParametersExtended();
	return framesPerBuffer;
	}

	private boolean startRecording() {
	Logging.d(TAG, "startRecording");
	assertTrue(audioRecord != null);
	assertTrue(audioThread == null);
	try {
	audioRecord.startRecording();
	} catch (IllegalStateException e) {
	reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_EXCEPTION,
	"AudioRecord.startRecording failed: " + e.getMessage());
	return false;
	}
	if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
	reportWebRtcAudioRecordStartError(
	AudioRecordStartErrorCode.AUDIO_RECORD_START_STATE_MISMATCH,
	"AudioRecord.startRecording failed - incorrect state :"
	+ audioRecord.getRecordingState());
	return false;
	}
	audioThread = new AudioRecordThread("AudioRecordJavaThread");
	audioThread.start();
	return true;
	}

	private boolean stopRecording() {
	Logging.d(TAG, "stopRecording");
	assertTrue(audioThread != null);
	audioThread.stopThread();
	if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
	Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
	WebRtcAudioUtils.logAudioState(TAG);
	}
	audioThread = null;
	if (effects != null) {
	effects.release();
	}
	releaseAudioResources();
	return true;
	}

	private void logMainParameters() {
	Logging.d(TAG, "AudioRecord: "
	+ "session ID: " + audioRecord.getAudioSessionId() + ", "
	+ "channels: " + audioRecord.getChannelCount() + ", "
	+ "sample rate: " + audioRecord.getSampleRate());
	}

	private void logMainParametersExtended() {
	if (Build.VERSION.SDK_INT >= 23) {
	Logging.d(TAG, "AudioRecord: "
	// The frame count of the native AudioRecord buffer.
	+ "buffer size in frames: " + audioRecord.getBufferSizeInFrames());
	}
	}

	// Helper method which throws an exception when an assertion has failed.
	private static void assertTrue(boolean condition) {
	if (!condition) {
	throw new AssertionError("Expected condition to be true");
	}
	}

	private int channelCountToConfiguration(int channels) {
	return (channels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
	}

	private native void nativeCacheDirectBufferAddress(ByteBuffer byteBuffer, long nativeAudioRecord);

	private native void nativeDataIsRecorded(int bytes, long nativeAudioRecord);

	@SuppressWarnings("NoSynchronizedMethodCheck")
	public static synchronized void setAudioSource(int source) {
	Logging.w(TAG, "Audio source is changed from: " + audioSource
	+ " to " + source);
	audioSource = source;
	}

	private static int getDefaultAudioSource() {
	return AudioSource.VOICE_COMMUNICATION;
	}

	// Sets all recorded samples to zero if `mute` is true, i.e., ensures that
	// the microphone is muted.
	public static void setMicrophoneMute(boolean mute) {
	Logging.w(TAG, "setMicrophoneMute(" + mute + ")");
	microphoneMute = mute;
	}

	// Releases the native AudioRecord resources.
	private void releaseAudioResources() {
	Logging.d(TAG, "releaseAudioResources");
	if (audioRecord != null) {
	audioRecord.release();
	audioRecord = null;
	}
	}

	private void reportWebRtcAudioRecordInitError(String errorMessage) {
	Logging.e(TAG, "Init recording error: " + errorMessage);
	WebRtcAudioUtils.logAudioState(TAG);
	if (errorCallback != null) {
	errorCallback.onWebRtcAudioRecordInitError(errorMessage);
	}
	}

	private void reportWebRtcAudioRecordStartError(
	AudioRecordStartErrorCode errorCode, String errorMessage) {
	Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
	WebRtcAudioUtils.logAudioState(TAG);
	if (errorCallback != null) {
	errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
	}
	}

	private void reportWebRtcAudioRecordError(String errorMessage) {
	Logging.e(TAG, "Run-time recording error: " + errorMessage);
	WebRtcAudioUtils.logAudioState(TAG);
	if (errorCallback != null) {
	errorCallback.onWebRtcAudioRecordError(errorMessage);
	}
	}
	}