docs: a microphone track hands no audio to sinks

docs/tools/audio/voice-activity-detector.md

@@ -147,9 +147,9 @@ peer.onTrack(track -> {
 });
 ```
 
-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:
+An `AudioTrack` of the native API takes the adapter as an `AudioTrackSink`, as
+long as the track's source hands its audio to sinks. A remote track and a track
+over a `CustomAudioSource` do:
 
 ```java
 VoiceActivityExample sink = new VoiceActivityExample(speech -> {
@@ -165,6 +165,11 @@ finally {
 }
 ```
 
+A microphone track does not: WebRTC passes captured audio from the audio device
+module straight to the senders, and the source of the track hands nothing to
+sinks. An [`AudioRecorder`](/tools/audio/audio-recorder) captures a microphone
+on its own, apart from any peer connection.
+
 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

webrtc-examples/src/main/java/dev/onvoid/webrtc/examples/VoiceActivityExample.java

@@ -31,8 +31,9 @@
 /**
  * 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().
+ * before close(). An AudioTrack of the native API takes it with addSink(),
+ * then removeSink() before close(), if its source hands audio to sinks: that
+ * of a remote track or a CustomAudioSource does, a microphone's does not.
  * 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 thread that delivers the audio and should return promptly. It receives raw