[AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule

Added audio format field and set method to Builder. - WebRTCAudioRecord. Added audio format field, added to constructor. Default audio format value AudioFormat.ENCODING_PCM_16BIT. initRecord(), added how to calculate bytesPerFrame, depends on audioFormat.

First commit and contribution, updated AUTHORS file

Bug: None
Change-Id: I16f660d42350ec9ce2e329b239bd7f6324e76dfe
Reviewed-on: https://webrtc-review.googlesource.com/c/122302
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26775}

AUTHORS

@@ -108,3 +108,4 @@ Vewd Software AS <*@vewd.com>
 Highfive, Inc. <*@highfive.com>
 CoSMo Software Consulting, Pte Ltd <*@cosmosoftware.io>
 Tuple, LLC <*@tuple.app>
+Videona Socialmedia <*@videona.com>

sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java

@@ -31,6 +31,7 @@ public class JavaAudioDeviceModule implements AudioDeviceModule {
     private final AudioManager audioManager;
     private int sampleRate;
     private int audioSource = WebRtcAudioRecord.DEFAULT_AUDIO_SOURCE;
+    private int audioFormat = WebRtcAudioRecord.DEFAULT_AUDIO_FORMAT;
     private AudioTrackErrorCallback audioTrackErrorCallback;
     private AudioRecordErrorCallback audioRecordErrorCallback;
     private SamplesReadyCallback samplesReadyCallback;
@@ -65,6 +66,17 @@ public class JavaAudioDeviceModule implements AudioDeviceModule {
       return this;
     }
 
+    /**
+     * Call this to change the audio format. The argument should be one of the values from
+     * android.media.AudioFormat ENCODING_PCM_8BIT, ENCODING_PCM_16BIT or ENCODING_PCM_FLOAT.
+     * Default audio data format is PCM 16 bit per sample.
+     * Guaranteed to be supported by all devices.
+     */
+    public Builder setAudioFormat(int audioFormat) {
+      this.audioFormat = audioFormat;
+      return this;
+    }
+
     /**
      * Set a callback to retrieve errors from the AudioTrack.
      */
@@ -154,9 +166,9 @@ public class JavaAudioDeviceModule implements AudioDeviceModule {
         }
         Logging.d(TAG, "HW AEC will not be used.");
       }
-      final WebRtcAudioRecord audioInput =
+      final WebRtcAudioRecord audioInput = new WebRtcAudioRecord(context, audioManager, audioSource,
-          new WebRtcAudioRecord(context, audioManager, audioSource, audioRecordErrorCallback,
+          audioFormat, audioRecordErrorCallback, samplesReadyCallback,
-              samplesReadyCallback, useHardwareAcousticEchoCanceler, useHardwareNoiseSuppressor);
+          useHardwareAcousticEchoCanceler, useHardwareNoiseSuppressor);
       final WebRtcAudioTrack audioOutput =
           new WebRtcAudioTrack(context, audioManager, audioTrackErrorCallback);
       return new JavaAudioDeviceModule(context, audioManager, audioInput, audioOutput, sampleRate,

sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java

@@ -32,10 +32,6 @@ import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback;
 class WebRtcAudioRecord {
   private static final String TAG = "WebRtcAudioRecordExternal";
 
-  // 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;
 
@@ -53,9 +49,14 @@ class WebRtcAudioRecord {
 
   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;
+
   private final Context context;
   private final AudioManager audioManager;
   private final int audioSource;
+  private final int audioFormat;
 
   private long nativeAudioRecord;
 
@@ -145,13 +146,14 @@ class WebRtcAudioRecord {
 
   @CalledByNative
   WebRtcAudioRecord(Context context, AudioManager audioManager) {
-    this(context, audioManager, DEFAULT_AUDIO_SOURCE, null /* errorCallback */,
+    this(context, audioManager, DEFAULT_AUDIO_SOURCE, DEFAULT_AUDIO_FORMAT,
-        null /* audioSamplesReadyCallback */, WebRtcAudioEffects.isAcousticEchoCancelerSupported(),
+        null /* errorCallback */, null /* audioSamplesReadyCallback */,
+        WebRtcAudioEffects.isAcousticEchoCancelerSupported(),
         WebRtcAudioEffects.isNoiseSuppressorSupported());
   }
 
   public WebRtcAudioRecord(Context context, AudioManager audioManager, int audioSource,
-      @Nullable AudioRecordErrorCallback errorCallback,
+      int audioFormat, @Nullable AudioRecordErrorCallback errorCallback,
       @Nullable SamplesReadyCallback audioSamplesReadyCallback,
       boolean isAcousticEchoCancelerSupported, boolean isNoiseSuppressorSupported) {
     if (isAcousticEchoCancelerSupported && !WebRtcAudioEffects.isAcousticEchoCancelerSupported()) {
@@ -163,6 +165,7 @@ class WebRtcAudioRecord {
     this.context = context;
     this.audioManager = audioManager;
     this.audioSource = audioSource;
+    this.audioFormat = audioFormat;
     this.errorCallback = errorCallback;
     this.audioSamplesReadyCallback = audioSamplesReadyCallback;
     this.isAcousticEchoCancelerSupported = isAcousticEchoCancelerSupported;
@@ -203,7 +206,7 @@ class WebRtcAudioRecord {
       reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
       return -1;
     }
-    final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
+    final int bytesPerFrame = channels * getBytesPerSample(audioFormat);
     final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
     byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
     if (!(byteBuffer.hasArray())) {
@@ -221,8 +224,7 @@ class WebRtcAudioRecord {
     // 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 =
+    int minBufferSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat);
-        AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
     if (minBufferSize == AudioRecord.ERROR || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
       reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
       return -1;
@@ -235,8 +237,8 @@ class WebRtcAudioRecord {
     int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
     Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
     try {
-      audioRecord = new AudioRecord(audioSource, sampleRate, channelConfig,
+      audioRecord =
-          AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes);
+          new AudioRecord(audioSource, sampleRate, channelConfig, audioFormat, bufferSizeInBytes);
     } catch (IllegalArgumentException e) {
       reportWebRtcAudioRecordInitError("AudioRecord ctor error: " + e.getMessage());
       releaseAudioResources();
@@ -363,4 +365,23 @@ class WebRtcAudioRecord {
       errorCallback.onWebRtcAudioRecordError(errorMessage);
     }
   }
+
+  // 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);
+    }
+  }
 }