Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 743 insertions, 0 deletions

diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
new file mode 100644
index 0000000000..6647e5fcbb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
@@ -0,0 +1,743 @@
+/*
+ *  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.audio;
+
+import android.annotation.TargetApi;
+import android.content.Context;
+import android.media.AudioDeviceInfo;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioRecordingConfiguration;
+import android.media.AudioTimestamp;
+import android.media.MediaRecorder.AudioSource;
+import android.os.Build;
+import android.os.Process;
+import androidx.annotation.Nullable;
+import androidx.annotation.RequiresApi;
+import java.lang.System;
+import java.nio.ByteBuffer;
+import java.util.Arrays;
+import java.util.Iterator;
+import java.util.List;
+import java.util.concurrent.Callable;
+import java.util.concurrent.Executors;
+import java.util.concurrent.ScheduledExecutorService;
+import java.util.concurrent.ScheduledFuture;
+import java.util.concurrent.ThreadFactory;
+import java.util.concurrent.TimeUnit;
+import java.util.concurrent.atomic.AtomicInteger;
+import java.util.concurrent.atomic.AtomicReference;
+import org.webrtc.CalledByNative;
+import org.webrtc.Logging;
+import org.webrtc.ThreadUtils;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStartErrorCode;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStateCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback;
+
+class WebRtcAudioRecord {
+  private static final String TAG = "WebRtcAudioRecordExternal";
+
+  // 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;
+
+  public static final int DEFAULT_AUDIO_SOURCE = AudioSource.VOICE_COMMUNICATION;
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  public static final int DEFAULT_AUDIO_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
+
+  // Indicates AudioRecord has started recording audio.
+  private static final int AUDIO_RECORD_START = 0;
+
+  // Indicates AudioRecord has stopped recording audio.
+  private static final int AUDIO_RECORD_STOP = 1;
+
+  // Time to wait before checking recording status after start has been called. Tests have
+  // shown that the result can sometimes be invalid (our own status might be missing) if we check
+  // directly after start.
+  private static final int CHECK_REC_STATUS_DELAY_MS = 100;
+
+  private final Context context;
+  private final AudioManager audioManager;
+  private final int audioSource;
+  private final int audioFormat;
+
+  private long nativeAudioRecord;
+
+  private final WebRtcAudioEffects effects = new WebRtcAudioEffects();
+
+  private @Nullable ByteBuffer byteBuffer;
+
+  private @Nullable AudioRecord audioRecord;
+  private @Nullable AudioRecordThread audioThread;
+  private @Nullable AudioDeviceInfo preferredDevice;
+
+  private final ScheduledExecutorService executor;
+  private @Nullable ScheduledFuture<String> future;
+
+  private volatile boolean microphoneMute;
+  private final AtomicReference<Boolean> audioSourceMatchesRecordingSessionRef =
+      new AtomicReference<>();
+  private byte[] emptyBytes;
+
+  private final @Nullable AudioRecordErrorCallback errorCallback;
+  private final @Nullable AudioRecordStateCallback stateCallback;
+  private final @Nullable SamplesReadyCallback audioSamplesReadyCallback;
+  private final boolean isAcousticEchoCancelerSupported;
+  private final boolean isNoiseSuppressorSupported;
+
+  /**
+   * 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);
+    }
+
+    @Override
+    public void run() {
+      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
+      Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
+      assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);
+
+      // Audio recording has started and the client is informed about it.
+      doAudioRecordStateCallback(AUDIO_RECORD_START);
+
+      long lastTime = System.nanoTime();
+      AudioTimestamp audioTimestamp = null;
+      if (Build.VERSION.SDK_INT >= 24) {
+        audioTimestamp = new AudioTimestamp();
+      }
+      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) {
+            long captureTimeNs = 0;
+            if (Build.VERSION.SDK_INT >= 24) {
+              if (audioRecord.getTimestamp(audioTimestamp, AudioTimestamp.TIMEBASE_MONOTONIC)
+                  == AudioRecord.SUCCESS) {
+                captureTimeNs = audioTimestamp.nanoTime;
+              }
+            }
+            nativeDataIsRecorded(nativeAudioRecord, bytesRead, captureTimeNs);
+          }
+          if (audioSamplesReadyCallback != null) {
+            // Copy the entire byte buffer array. The start of the byteBuffer is not necessarily
+            // at index 0.
+            byte[] data = Arrays.copyOfRange(byteBuffer.array(), byteBuffer.arrayOffset(),
+                byteBuffer.capacity() + byteBuffer.arrayOffset());
+            audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
+                new JavaAudioDeviceModule.AudioSamples(audioRecord.getAudioFormat(),
+                    audioRecord.getChannelCount(), audioRecord.getSampleRate(), data));
+          }
+        } else {
+          String errorMessage = "AudioRecord.read failed: " + bytesRead;
+          Logging.e(TAG, errorMessage);
+          if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
+            keepAlive = false;
+            reportWebRtcAudioRecordError(errorMessage);
+          }
+        }
+      }
+
+      try {
+        if (audioRecord != null) {
+          audioRecord.stop();
+          doAudioRecordStateCallback(AUDIO_RECORD_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;
+    }
+  }
+
+  @CalledByNative
+  WebRtcAudioRecord(Context context, AudioManager audioManager) {
+    this(context, newDefaultScheduler() /* scheduler */, audioManager, DEFAULT_AUDIO_SOURCE,
+        DEFAULT_AUDIO_FORMAT, null /* errorCallback */, null /* stateCallback */,
+        null /* audioSamplesReadyCallback */, WebRtcAudioEffects.isAcousticEchoCancelerSupported(),
+        WebRtcAudioEffects.isNoiseSuppressorSupported());
+  }
+
+  public WebRtcAudioRecord(Context context, ScheduledExecutorService scheduler,
+      AudioManager audioManager, int audioSource, int audioFormat,
+      @Nullable AudioRecordErrorCallback errorCallback,
+      @Nullable AudioRecordStateCallback stateCallback,
+      @Nullable SamplesReadyCallback audioSamplesReadyCallback,
+      boolean isAcousticEchoCancelerSupported, boolean isNoiseSuppressorSupported) {
+    if (isAcousticEchoCancelerSupported && !WebRtcAudioEffects.isAcousticEchoCancelerSupported()) {
+      throw new IllegalArgumentException("HW AEC not supported");
+    }
+    if (isNoiseSuppressorSupported && !WebRtcAudioEffects.isNoiseSuppressorSupported()) {
+      throw new IllegalArgumentException("HW NS not supported");
+    }
+    this.context = context;
+    this.executor = scheduler;
+    this.audioManager = audioManager;
+    this.audioSource = audioSource;
+    this.audioFormat = audioFormat;
+    this.errorCallback = errorCallback;
+    this.stateCallback = stateCallback;
+    this.audioSamplesReadyCallback = audioSamplesReadyCallback;
+    this.isAcousticEchoCancelerSupported = isAcousticEchoCancelerSupported;
+    this.isNoiseSuppressorSupported = isNoiseSuppressorSupported;
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+  }
+
+  @CalledByNative
+  public void setNativeAudioRecord(long nativeAudioRecord) {
+    this.nativeAudioRecord = nativeAudioRecord;
+  }
+
+  @CalledByNative
+  boolean isAcousticEchoCancelerSupported() {
+    return isAcousticEchoCancelerSupported;
+  }
+
+  @CalledByNative
+  boolean isNoiseSuppressorSupported() {
+    return isNoiseSuppressorSupported;
+  }
+
+  // Returns true if a valid call to verifyAudioConfig() has been done. Should always be
+  // checked before using the returned value of isAudioSourceMatchingRecordingSession().
+  @CalledByNative
+  boolean isAudioConfigVerified() {
+    return audioSourceMatchesRecordingSessionRef.get() != null;
+  }
+
+  // Returns true if verifyAudioConfig() succeeds. This value is set after a specific delay when
+  // startRecording() has been called. Hence, should preferably be called in combination with
+  // stopRecording() to ensure that it has been set properly. `isAudioConfigVerified` is
+  // enabled in WebRtcAudioRecord to ensure that the returned value is valid.
+  @CalledByNative
+  boolean isAudioSourceMatchingRecordingSession() {
+    Boolean audioSourceMatchesRecordingSession = audioSourceMatchesRecordingSessionRef.get();
+    if (audioSourceMatchesRecordingSession == null) {
+      Logging.w(TAG, "Audio configuration has not yet been verified");
+      return false;
+    }
+    return audioSourceMatchesRecordingSession;
+  }
+
+  @CalledByNative
+  private boolean enableBuiltInAEC(boolean enable) {
+    Logging.d(TAG, "enableBuiltInAEC(" + enable + ")");
+    return effects.setAEC(enable);
+  }
+
+  @CalledByNative
+  private boolean enableBuiltInNS(boolean enable) {
+    Logging.d(TAG, "enableBuiltInNS(" + enable + ")");
+    return effects.setNS(enable);
+  }
+
+  @CalledByNative
+  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 * getBytesPerSample(audioFormat);
+    final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
+    byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
+    if (!(byteBuffer.hasArray())) {
+      reportWebRtcAudioRecordInitError("ByteBuffer does not have backing array.");
+      return -1;
+    }
+    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(nativeAudioRecord, byteBuffer);
+
+    // 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);
+    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 {
+      if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+        // Use the AudioRecord.Builder class on Android M (23) and above.
+        // Throws IllegalArgumentException.
+        audioRecord = createAudioRecordOnMOrHigher(
+            audioSource, sampleRate, channelConfig, audioFormat, bufferSizeInBytes);
+        audioSourceMatchesRecordingSessionRef.set(null);
+        if (preferredDevice != null) {
+          setPreferredDevice(preferredDevice);
+        }
+      } else {
+        // Use the old AudioRecord constructor for API levels below 23.
+        // Throws UnsupportedOperationException.
+        audioRecord = createAudioRecordOnLowerThanM(
+            audioSource, sampleRate, channelConfig, audioFormat, bufferSizeInBytes);
+        audioSourceMatchesRecordingSessionRef.set(null);
+      }
+    } catch (IllegalArgumentException | UnsupportedOperationException e) {
+      // Report of exception message is sufficient. Example: "Cannot create AudioRecord".
+      reportWebRtcAudioRecordInitError(e.getMessage());
+      releaseAudioResources();
+      return -1;
+    }
+    if (audioRecord == null || audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
+      reportWebRtcAudioRecordInitError("Creation or initialization of audio recorder failed.");
+      releaseAudioResources();
+      return -1;
+    }
+    effects.enable(audioRecord.getAudioSessionId());
+    logMainParameters();
+    logMainParametersExtended();
+    // Check number of active recording sessions. Should be zero but we have seen conflict cases
+    // and adding a log for it can help us figure out details about conflicting sessions.
+    final int numActiveRecordingSessions =
+        logRecordingConfigurations(audioRecord, false /* verifyAudioConfig */);
+    if (numActiveRecordingSessions != 0) {
+      // Log the conflict as a warning since initialization did in fact succeed. Most likely, the
+      // upcoming call to startRecording() will fail under these conditions.
+      Logging.w(
+          TAG, "Potential microphone conflict. Active sessions: " + numActiveRecordingSessions);
+    }
+    return framesPerBuffer;
+  }
+
+  /**
+   * Prefer a specific {@link AudioDeviceInfo} device for recording. Calling after recording starts
+   * is valid but may cause a temporary interruption if the audio routing changes.
+   */
+  @RequiresApi(Build.VERSION_CODES.M)
+  @TargetApi(Build.VERSION_CODES.M)
+  void setPreferredDevice(@Nullable AudioDeviceInfo preferredDevice) {
+    Logging.d(
+        TAG, "setPreferredDevice " + (preferredDevice != null ? preferredDevice.getId() : null));
+    this.preferredDevice = preferredDevice;
+    if (audioRecord != null) {
+      if (!audioRecord.setPreferredDevice(preferredDevice)) {
+        Logging.e(TAG, "setPreferredDevice failed");
+      }
+    }
+  }
+
+  @CalledByNative
+  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();
+    scheduleLogRecordingConfigurationsTask(audioRecord);
+    return true;
+  }
+
+  @CalledByNative
+  private boolean stopRecording() {
+    Logging.d(TAG, "stopRecording");
+    assertTrue(audioThread != null);
+    if (future != null) {
+      if (!future.isDone()) {
+        // Might be needed if the client calls startRecording(), stopRecording() back-to-back.
+        future.cancel(true /* mayInterruptIfRunning */);
+      }
+      future = null;
+    }
+    audioThread.stopThread();
+    if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
+      Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
+      WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    }
+    audioThread = null;
+    effects.release();
+    releaseAudioResources();
+    return true;
+  }
+
+  @TargetApi(Build.VERSION_CODES.M)
+  private static AudioRecord createAudioRecordOnMOrHigher(
+      int audioSource, int sampleRate, int channelConfig, int audioFormat, int bufferSizeInBytes) {
+    Logging.d(TAG, "createAudioRecordOnMOrHigher");
+    return new AudioRecord.Builder()
+        .setAudioSource(audioSource)
+        .setAudioFormat(new AudioFormat.Builder()
+                            .setEncoding(audioFormat)
+                            .setSampleRate(sampleRate)
+                            .setChannelMask(channelConfig)
+                            .build())
+        .setBufferSizeInBytes(bufferSizeInBytes)
+        .build();
+  }
+
+  private static AudioRecord createAudioRecordOnLowerThanM(
+      int audioSource, int sampleRate, int channelConfig, int audioFormat, int bufferSizeInBytes) {
+    Logging.d(TAG, "createAudioRecordOnLowerThanM");
+    return new AudioRecord(audioSource, sampleRate, channelConfig, audioFormat, bufferSizeInBytes);
+  }
+
+  private void logMainParameters() {
+    Logging.d(TAG,
+        "AudioRecord: "
+            + "session ID: " + audioRecord.getAudioSessionId() + ", "
+            + "channels: " + audioRecord.getChannelCount() + ", "
+            + "sample rate: " + audioRecord.getSampleRate());
+  }
+
+  @TargetApi(Build.VERSION_CODES.M)
+  private void logMainParametersExtended() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+      Logging.d(TAG,
+          "AudioRecord: "
+              // The frame count of the native AudioRecord buffer.
+              + "buffer size in frames: " + audioRecord.getBufferSizeInFrames());
+    }
+  }
+
+  @TargetApi(Build.VERSION_CODES.N)
+  // Checks the number of active recording sessions and logs the states of all active sessions.
+  // Returns number of active sessions. Note that this could occur on arbituary thread.
+  private int logRecordingConfigurations(AudioRecord audioRecord, boolean verifyAudioConfig) {
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.N) {
+      Logging.w(TAG, "AudioManager#getActiveRecordingConfigurations() requires N or higher");
+      return 0;
+    }
+    if (audioRecord == null) {
+      return 0;
+    }
+
+    // Get a list of the currently active audio recording configurations of the device (can be more
+    // than one). An empty list indicates there is no recording active when queried.
+    List<AudioRecordingConfiguration> configs = audioManager.getActiveRecordingConfigurations();
+    final int numActiveRecordingSessions = configs.size();
+    Logging.d(TAG, "Number of active recording sessions: " + numActiveRecordingSessions);
+    if (numActiveRecordingSessions > 0) {
+      logActiveRecordingConfigs(audioRecord.getAudioSessionId(), configs);
+      if (verifyAudioConfig) {
+        // Run an extra check to verify that the existing audio source doing the recording (tied
+        // to the AudioRecord instance) is matching what the audio recording configuration lists
+        // as its client parameters. If these do not match, recording might work but under invalid
+        // conditions.
+        audioSourceMatchesRecordingSessionRef.set(
+            verifyAudioConfig(audioRecord.getAudioSource(), audioRecord.getAudioSessionId(),
+                audioRecord.getFormat(), audioRecord.getRoutedDevice(), configs));
+      }
+    }
+    return numActiveRecordingSessions;
+  }
+
+  // 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(
+      long nativeAudioRecordJni, ByteBuffer byteBuffer);
+  private native void nativeDataIsRecorded(
+      long nativeAudioRecordJni, int bytes, long captureTimestampNs);
+
+  // Sets all recorded samples to zero if `mute` is true, i.e., ensures that
+  // the microphone is muted.
+  public 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;
+    }
+    audioSourceMatchesRecordingSessionRef.set(null);
+  }
+
+  private void reportWebRtcAudioRecordInitError(String errorMessage) {
+    Logging.e(TAG, "Init recording error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    logRecordingConfigurations(audioRecord, false /* verifyAudioConfig */);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordInitError(errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioRecordStartError(
+      AudioRecordStartErrorCode errorCode, String errorMessage) {
+    Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    logRecordingConfigurations(audioRecord, false /* verifyAudioConfig */);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioRecordError(String errorMessage) {
+    Logging.e(TAG, "Run-time recording error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordError(errorMessage);
+    }
+  }
+
+  private void doAudioRecordStateCallback(int audioState) {
+    Logging.d(TAG, "doAudioRecordStateCallback: " + audioStateToString(audioState));
+    if (stateCallback != null) {
+      if (audioState == WebRtcAudioRecord.AUDIO_RECORD_START) {
+        stateCallback.onWebRtcAudioRecordStart();
+      } else if (audioState == WebRtcAudioRecord.AUDIO_RECORD_STOP) {
+        stateCallback.onWebRtcAudioRecordStop();
+      } else {
+        Logging.e(TAG, "Invalid audio state");
+      }
+    }
+  }
+
+  // Reference from Android code, AudioFormat.getBytesPerSample. BitPerSample / 8
+  // Default audio data format is PCM 16 bits per sample.
+  // Guaranteed to be supported by all devices
+  private static int getBytesPerSample(int audioFormat) {
+    switch (audioFormat) {
+      case AudioFormat.ENCODING_PCM_8BIT:
+        return 1;
+      case AudioFormat.ENCODING_PCM_16BIT:
+      case AudioFormat.ENCODING_IEC61937:
+      case AudioFormat.ENCODING_DEFAULT:
+        return 2;
+      case AudioFormat.ENCODING_PCM_FLOAT:
+        return 4;
+      case AudioFormat.ENCODING_INVALID:
+      default:
+        throw new IllegalArgumentException("Bad audio format " + audioFormat);
+    }
+  }
+
+  // Use an ExecutorService to schedule a task after a given delay where the task consists of
+  // checking (by logging) the current status of active recording sessions.
+  private void scheduleLogRecordingConfigurationsTask(AudioRecord audioRecord) {
+    Logging.d(TAG, "scheduleLogRecordingConfigurationsTask");
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.N) {
+      return;
+    }
+
+    Callable<String> callable = () -> {
+      if (this.audioRecord == audioRecord) {
+        logRecordingConfigurations(audioRecord, true /* verifyAudioConfig */);
+      } else {
+        Logging.d(TAG, "audio record has changed");
+      }
+      return "Scheduled task is done";
+    };
+
+    if (future != null && !future.isDone()) {
+      future.cancel(true /* mayInterruptIfRunning */);
+    }
+    // Schedule call to logRecordingConfigurations() from executor thread after fixed delay.
+    future = executor.schedule(callable, CHECK_REC_STATUS_DELAY_MS, TimeUnit.MILLISECONDS);
+  };
+
+  @TargetApi(Build.VERSION_CODES.N)
+  private static boolean logActiveRecordingConfigs(
+      int session, List<AudioRecordingConfiguration> configs) {
+    assertTrue(!configs.isEmpty());
+    final Iterator<AudioRecordingConfiguration> it = configs.iterator();
+    Logging.d(TAG, "AudioRecordingConfigurations: ");
+    while (it.hasNext()) {
+      final AudioRecordingConfiguration config = it.next();
+      StringBuilder conf = new StringBuilder();
+      // The audio source selected by the client.
+      final int audioSource = config.getClientAudioSource();
+      conf.append("  client audio source=")
+          .append(WebRtcAudioUtils.audioSourceToString(audioSource))
+          .append(", client session id=")
+          .append(config.getClientAudioSessionId())
+          // Compare with our own id (based on AudioRecord#getAudioSessionId()).
+          .append(" (")
+          .append(session)
+          .append(")")
+          .append("\n");
+      // Audio format at which audio is recorded on this Android device. Note that it may differ
+      // from the client application recording format (see getClientFormat()).
+      AudioFormat format = config.getFormat();
+      conf.append("  Device AudioFormat: ")
+          .append("channel count=")
+          .append(format.getChannelCount())
+          .append(", channel index mask=")
+          .append(format.getChannelIndexMask())
+          // Only AudioFormat#CHANNEL_IN_MONO is guaranteed to work on all devices.
+          .append(", channel mask=")
+          .append(WebRtcAudioUtils.channelMaskToString(format.getChannelMask()))
+          .append(", encoding=")
+          .append(WebRtcAudioUtils.audioEncodingToString(format.getEncoding()))
+          .append(", sample rate=")
+          .append(format.getSampleRate())
+          .append("\n");
+      // Audio format at which the client application is recording audio.
+      format = config.getClientFormat();
+      conf.append("  Client AudioFormat: ")
+          .append("channel count=")
+          .append(format.getChannelCount())
+          .append(", channel index mask=")
+          .append(format.getChannelIndexMask())
+          // Only AudioFormat#CHANNEL_IN_MONO is guaranteed to work on all devices.
+          .append(", channel mask=")
+          .append(WebRtcAudioUtils.channelMaskToString(format.getChannelMask()))
+          .append(", encoding=")
+          .append(WebRtcAudioUtils.audioEncodingToString(format.getEncoding()))
+          .append(", sample rate=")
+          .append(format.getSampleRate())
+          .append("\n");
+      // Audio input device used for this recording session.
+      final AudioDeviceInfo device = config.getAudioDevice();
+      if (device != null) {
+        assertTrue(device.isSource());
+        conf.append("  AudioDevice: ")
+            .append("type=")
+            .append(WebRtcAudioUtils.deviceTypeToString(device.getType()))
+            .append(", id=")
+            .append(device.getId());
+      }
+      Logging.d(TAG, conf.toString());
+    }
+    return true;
+  }
+
+  // Verify that the client audio configuration (device and format) matches the requested
+  // configuration (same as AudioRecord's).
+  @TargetApi(Build.VERSION_CODES.N)
+  private static boolean verifyAudioConfig(int source, int session, AudioFormat format,
+      AudioDeviceInfo device, List<AudioRecordingConfiguration> configs) {
+    assertTrue(!configs.isEmpty());
+    final Iterator<AudioRecordingConfiguration> it = configs.iterator();
+    while (it.hasNext()) {
+      final AudioRecordingConfiguration config = it.next();
+      final AudioDeviceInfo configDevice = config.getAudioDevice();
+      if (configDevice == null) {
+        continue;
+      }
+      if ((config.getClientAudioSource() == source)
+          && (config.getClientAudioSessionId() == session)
+          // Check the client format (should match the format of the AudioRecord instance).
+          && (config.getClientFormat().getEncoding() == format.getEncoding())
+          && (config.getClientFormat().getSampleRate() == format.getSampleRate())
+          && (config.getClientFormat().getChannelMask() == format.getChannelMask())
+          && (config.getClientFormat().getChannelIndexMask() == format.getChannelIndexMask())
+          // Ensure that the device format is properly configured.
+          && (config.getFormat().getEncoding() != AudioFormat.ENCODING_INVALID)
+          && (config.getFormat().getSampleRate() > 0)
+          //  For the channel mask, either the position or index-based value must be valid.
+          && ((config.getFormat().getChannelMask() != AudioFormat.CHANNEL_INVALID)
+              || (config.getFormat().getChannelIndexMask() != AudioFormat.CHANNEL_INVALID))
+          && checkDeviceMatch(configDevice, device)) {
+        Logging.d(TAG, "verifyAudioConfig: PASS");
+        return true;
+      }
+    }
+    Logging.e(TAG, "verifyAudioConfig: FAILED");
+    return false;
+  }
+
+  @TargetApi(Build.VERSION_CODES.N)
+  // Returns true if device A parameters matches those of device B.
+  // TODO(henrika): can be improved by adding AudioDeviceInfo#getAddress() but it requires API 29.
+  private static boolean checkDeviceMatch(AudioDeviceInfo devA, AudioDeviceInfo devB) {
+    return ((devA.getId() == devB.getId() && (devA.getType() == devB.getType())));
+  }
+
+  private static String audioStateToString(int state) {
+    switch (state) {
+      case WebRtcAudioRecord.AUDIO_RECORD_START:
+        return "START";
+      case WebRtcAudioRecord.AUDIO_RECORD_STOP:
+        return "STOP";
+      default:
+        return "INVALID";
+    }
+  }
+
+  private static final AtomicInteger nextSchedulerId = new AtomicInteger(0);
+
+  static ScheduledExecutorService newDefaultScheduler() {
+    AtomicInteger nextThreadId = new AtomicInteger(0);
+    return Executors.newScheduledThreadPool(0, new ThreadFactory() {
+      /**
+       * Constructs a new {@code Thread}
+       */
+      @Override
+      public Thread newThread(Runnable r) {
+        Thread thread = Executors.defaultThreadFactory().newThread(r);
+        thread.setName(String.format("WebRtcAudioRecordScheduler-%s-%s",
+            nextSchedulerId.getAndIncrement(), nextThreadId.getAndIncrement()));
+        return thread;
+      }
+    });
+  }
+}