From bcb6cce7dcc8b3dd9f79ed275715fa99d5a8002c Mon Sep 17 00:00:00 2001
From: Alex Andres <andres.alex@protonmail.com>
Date: Thu, 29 Oct 2026 07:51:57 +0100
Subject: [PATCH] feat: feed the voice activity example from client API audio

VoiceActivityExample also takes the AudioChunks of RemoteAudio.onAudio. Local tracks still attach it as an AudioTrackSink.
---
 docs/tools/audio/voice-activity-detector.md   | 39 ++++++++++++++++---
 .../webrtc/examples/VoiceActivityExample.java | 21 +++++++---
 2 files changed, 49 insertions(+), 11 deletions(-)

diff --git a/docs/tools/audio/voice-activity-detector.md b/docs/tools/audio/voice-activity-detector.md
index 118b2a84f..b658c84b4 100644
--- a/docs/tools/audio/voice-activity-detector.md
+++ b/docs/tools/audio/voice-activity-detector.md
@@ -120,14 +120,38 @@ state, or transmitted audio.
 
 ## Audio track integration
 
-The examples module includes `VoiceActivityExample`, an `AudioTrackSink` adapter
-using `StandaloneVoiceActivityDetector`. It buffers callback data into 10 ms
-frames, converts mono or stereo input to 16 kHz mono with `AudioConverter`, and
-reports each boolean decision. It also supports 44.1 kHz input.
+The examples module includes `VoiceActivityExample`, an adapter using
+`StandaloneVoiceActivityDetector`. It buffers incoming audio into 10 ms frames,
+converts mono or stereo input to 16 kHz mono with `AudioConverter`, and reports
+each boolean decision. It also supports 44.1 kHz input.
+
+With the [Client API](/guide/client-api), it takes the `AudioChunk`s of a remote
+track's `onAudio` subscription:
 
 ```java
+import dev.onvoid.webrtc.client.Subscription;
 import dev.onvoid.webrtc.examples.VoiceActivityExample;
 
+peer.onTrack(track -> {
+    if (track.isAudio()) {
+        VoiceActivityExample vad = new VoiceActivityExample(speech -> {
+            // Enqueue the decision for your application/UI; return promptly.
+        });
+        Subscription audio = track.asAudio().onAudio(vad);
+
+        track.onEnded(() -> {
+            audio.close();
+            vad.close();
+        });
+    }
+});
+```
+
+The client API has no audio subscription on a local track. To detect speech at
+the microphone, attach the adapter as an `AudioTrackSink` to an `AudioTrack` of
+the native API:
+
+```java
 VoiceActivityExample sink = new VoiceActivityExample(speech -> {
     // Enqueue the decision for your application/UI; return promptly.
 });
@@ -141,8 +165,11 @@ finally {
 }
 ```
 
-The adapter invokes the consumer on the audio callback thread. Keep work there
-short. Apply application-level timing to produce stable speech-start/speech-end
+The adapter invokes the consumer on the thread that delivers the audio: a thread
+of the `onAudio` subscription, or the audio callback thread of a sink. Keep work
+there short. A subscription that falls behind loses the oldest audio beyond its
+buffer, so a detector that must stay live passes a short one, as in
+`onAudio(Duration.ofMillis(200), vad)`. Apply application-level timing to produce stable speech-start/speech-end
 events, rather than toggling a UI or recording for every frame. Audio format
 changes clear the adapter's partial frame and reset its detector.
 
diff --git a/webrtc-examples/src/main/java/dev/onvoid/webrtc/examples/VoiceActivityExample.java b/webrtc-examples/src/main/java/dev/onvoid/webrtc/examples/VoiceActivityExample.java
index 73c2ead6d..90c03fbe6 100644
--- a/webrtc-examples/src/main/java/dev/onvoid/webrtc/examples/VoiceActivityExample.java
+++ b/webrtc-examples/src/main/java/dev/onvoid/webrtc/examples/VoiceActivityExample.java
@@ -21,18 +21,24 @@
 import java.util.function.Consumer;
 import java.util.Objects;
 
+import dev.onvoid.webrtc.client.AudioChunk;
+import dev.onvoid.webrtc.client.RemoteAudio;
+import dev.onvoid.webrtc.client.Subscription;
 import dev.onvoid.webrtc.media.audio.AudioConverter;
 import dev.onvoid.webrtc.media.audio.AudioTrackSink;
 import dev.onvoid.webrtc.media.audio.StandaloneVoiceActivityDetector;
 
 /**
- * Attach to an AudioTrack with addSink(), then removeSink() before close().
- * Buffers arbitrary callback lengths into 10 ms frames and converts interleaved
+ * With the client API, subscribe it to a remote track with
+ * {@link RemoteAudio#onAudio(Consumer)}, then close the {@link Subscription}
+ * before close(). An AudioTrack of the native API, such as a microphone's,
+ * takes it with addSink(), then removeSink() before close().
+ * Buffers arbitrary chunk lengths into 10 ms frames and converts interleaved
  * signed 16-bit PCM to 16 kHz mono, including input at 44.1 kHz. The consumer runs
- * on the audio callback thread and should return promptly. It receives raw frame
- * decisions; a UI should apply its own speaking-state timing off this thread.
+ * on the thread that delivers the audio and should return promptly. It receives raw
+ * frame decisions; a UI should apply its own speaking-state timing off this thread.
  */
-public final class VoiceActivityExample implements AudioTrackSink, AutoCloseable {
+public final class VoiceActivityExample implements Consumer<AudioChunk>, AudioTrackSink, AutoCloseable {
 
     private final Consumer<Boolean> onDecision;
     private final StandaloneVoiceActivityDetector detector;
@@ -51,6 +57,11 @@ public VoiceActivityExample(Consumer<Boolean> onDecision) {
         detector = new StandaloneVoiceActivityDetector(StandaloneVoiceActivityDetector.Mode.AGGRESSIVE);
     }
 
+    @Override
+    public void accept(AudioChunk chunk) {
+        onData(chunk.samples(), 16, chunk.sampleRate(), chunk.channels(), chunk.frames());
+    }
+
     @Override
     public synchronized void onData(byte[] data, int bitsPerSample, int sampleRate, int channels, int frames) {
         if (closed) {