AudioProcessing to use only 16 Bit audio samples

Author: devopvoid

webrtc-jni/src/main/cpp/include/JNI_AudioProcessing.h

@@ -20,12 +20,11 @@
 import dev.onvoid.webrtc.internal.NativeLoader;
 
 /**
- * Audio Processing provides a collection of voice processing components
- * designed for real-time communications software.
- * <br/>
- * Audio Processing
- * accepts only linear PCM audio data in chunks of 10 ms. The int16 interfaces
- * use interleaved data, while the float interfaces use deinterleaved data.
+ * AudioProcessing provides a collection of voice processing components designed
+ * for real-time communications software.
+ * <p>
+ * AudioProcessing accepts only linear PCM audio data in chunks of 10 ms. The
+ * int16 interfaces use interleaved data.
  *
  * @author Alex Andres
  */
@@ -56,21 +55,13 @@ public AudioProcessingStats getStatistics() {
 		return stats;
 	}
 
-	public native int ProcessStream(short[] src,
+	public native int processStream(byte[] src,
 			AudioProcessingStreamConfig inputConfig,
-			AudioProcessingStreamConfig outputConfig, short[] dest);
+			AudioProcessingStreamConfig outputConfig, byte[] dest);
 
-	public native int ProcessStream(float[] src,
+	public native int processReverseStream(byte[] src,
 			AudioProcessingStreamConfig inputConfig,
-			AudioProcessingStreamConfig outputConfig, float[] dest);
-
-	public native int ProcessReverseStream(short[] src,
-			AudioProcessingStreamConfig inputConfig,
-			AudioProcessingStreamConfig outputConfig, short[] dest);
-
-	public native int ProcessReverseStream(float[] src,
-			AudioProcessingStreamConfig inputConfig,
-			AudioProcessingStreamConfig outputConfig, float[] dest);
+			AudioProcessingStreamConfig outputConfig, byte[] dest);
 
 	@Override
 	public native void dispose();

webrtc/src/main/java/dev/onvoid/webrtc/media/audio/AudioProcessing.java

@@ -16,6 +16,7 @@
 
 package dev.onvoid.webrtc.media.audio;
 
+import static org.junit.jupiter.api.Assertions.assertEquals;
 import static org.junit.jupiter.api.Assertions.assertNotNull;
 
 import dev.onvoid.webrtc.TestBase;
@@ -24,7 +25,9 @@
 import org.junit.jupiter.api.BeforeEach;
 import org.junit.jupiter.api.Test;
 
-class AudioProcessingTests extends TestBase {
+class AudioProcessingTest extends TestBase {
+
+	private final AudioProcessingStreamConfig streamConfig = new AudioProcessingStreamConfig(48000, 1);
 
 	private AudioProcessing audioProcessing;
 
@@ -59,4 +62,25 @@ void getStats() {
 		assertNotNull(audioProcessing.getStatistics());
 	}
 
+	@Test
+	void processByteStream() {
+		int sampleRate = streamConfig.sampleRate;
+		int bytesPerFrame = 2;
+		int frameSize = sampleRate / 100 * bytesPerFrame; // 10 ms frame
+
+		byte[] data = new byte[frameSize];
+
+		assertEquals(0, audioProcessing.processStream(data, streamConfig, streamConfig, data));
+	}
+
+	@Test
+	void processReverseStream() {
+		int sampleRate = streamConfig.sampleRate;
+		int bytesPerFrame = 2;
+		int frameSize = sampleRate / 100 * bytesPerFrame; // 10 ms frame
+
+		byte[] data = new byte[frameSize];
+
+		assertEquals(0, audioProcessing.processReverseStream(data, streamConfig, streamConfig, data));
+	}
 }