diff --git a/src/audio/voiceActivityDetection.cpp b/src/audio/voiceActivityDetection.cpp
index 59324ef..b121071 100644
--- a/src/audio/voiceActivityDetection.cpp
+++ b/src/audio/voiceActivityDetection.cpp
@@ -7,16 +7,15 @@
 #include <webrtc/common_audio/vad/include/webrtc_vad.h>
 #include "processing.h"
 #include <gsl_util.h>
+#include <ThreadPool.h>
+#include "AudioStreamSegment.h"
 
 using std::vector;
 using boost::adaptors::transformed;
 using fmt::format;
 using std::runtime_error;
 
-BoundedTimeline<void> detectVoiceActivity(std::unique_ptr<AudioStream> audioStream, ProgressSink& progressSink) {
-	// Prepare audio for VAD
-	audioStream = removeDCOffset(convertSampleRate(std::move(audioStream), 16000));
-
+BoundedTimeline<void> webRtcDetectVoiceActivity(AudioStream& audioStream, ProgressSink& progressSink) {
 	VadInst* vadHandle = WebRtcVad_Create();
 	if (!vadHandle) throw runtime_error("Error creating WebRTC VAD handle.");
 
@@ -30,17 +29,48 @@ BoundedTimeline<void> detectVoiceActivity(std::unique_ptr<AudioStream> audioStre
 	if (error) throw runtime_error("Error setting WebRTC VAD aggressiveness.");
 
 	// Detect activity
-	BoundedTimeline<void> activity(audioStream->getTruncatedRange());
+	BoundedTimeline<void> activity(audioStream.getTruncatedRange());
 	centiseconds time = centiseconds::zero();
 	auto processBuffer = [&](const vector<int16_t>& buffer) {
-		bool isActive = WebRtcVad_Process(vadHandle, audioStream->getSampleRate(), buffer.data(), buffer.size()) == 1;
+		bool isActive = WebRtcVad_Process(vadHandle, audioStream.getSampleRate(), buffer.data(), buffer.size()) == 1;
 		if (isActive) {
 			activity.set(time, time + centiseconds(1));
 		}
 		time += centiseconds(1);
 	};
-	const size_t bufferCapacity = audioStream->getSampleRate() / 100;
-	process16bitAudioStream(*audioStream.get(), processBuffer, bufferCapacity, progressSink);
+	const size_t bufferCapacity = audioStream.getSampleRate() / 100;
+	process16bitAudioStream(audioStream, processBuffer, bufferCapacity, progressSink);
+
+	return activity;
+}
+
+BoundedTimeline<void> detectVoiceActivity(std::unique_ptr<AudioStream> audioStream, ProgressSink& progressSink) {
+	// Prepare audio for VAD
+	audioStream = removeDCOffset(convertSampleRate(std::move(audioStream), 16000));
+
+	BoundedTimeline<void> activity(audioStream->getTruncatedRange());
+	std::mutex activityMutex;
+
+	// Split audio into segments and perform parallel VAD
+	ThreadPool threadPool;
+	int segmentCount = threadPool.getThreadCount();
+	centiseconds audioLength = audioStream->getTruncatedRange().getLength();
+	ProgressMerger progressMerger(progressSink);
+	for (int i = 0; i < segmentCount; ++i) {
+		TimeRange segmentRange = TimeRange(i * audioLength / segmentCount, (i + 1) * audioLength / segmentCount);
+		ProgressSink& segmentProgressSink = progressMerger.addSink(1.0);
+		threadPool.addJob([segmentRange, &audioStream, &segmentProgressSink, &activityMutex, &activity] {
+			std::unique_ptr<AudioStream> audioSegment = createSegment(audioStream->clone(false), segmentRange);
+			BoundedTimeline<void> activitySegment = webRtcDetectVoiceActivity(*audioSegment, segmentProgressSink);
+
+			std::lock_guard<std::mutex> lock(activityMutex);
+			for (auto activityRange : activitySegment) {
+				activityRange.getTimeRange().shift(segmentRange.getStart());
+				activity.set(activityRange);
+			}
+		});
+	}
+	threadPool.waitAll();
 
 	// Fill small gaps in activity
 	const centiseconds maxGap(5);