diff --git a/rhubarb/resharper.DotSettings b/rhubarb/resharper.DotSettings
index 103065f..b695093 100644
--- a/rhubarb/resharper.DotSettings
+++ b/rhubarb/resharper.DotSettings
@@ -212,6 +212,7 @@
 	<s:Boolean x:Key="/Default/UserDictionary/Words/=qwhy/@EntryIndexedValue">True</s:Boolean>
 	<s:Boolean x:Key="/Default/UserDictionary/Words/=rbegin/@EntryIndexedValue">True</s:Boolean>
 	<s:Boolean x:Key="/Default/UserDictionary/Words/=resample/@EntryIndexedValue">True</s:Boolean>
+	<s:Boolean x:Key="/Default/UserDictionary/Words/=resamples/@EntryIndexedValue">True</s:Boolean>
 	<s:Boolean x:Key="/Default/UserDictionary/Words/=retime/@EntryIndexedValue">True</s:Boolean>
 	<s:Boolean x:Key="/Default/UserDictionary/Words/=retimed/@EntryIndexedValue">True</s:Boolean>
 	<s:Boolean x:Key="/Default/UserDictionary/Words/=synth/@EntryIndexedValue">True</s:Boolean>
diff --git a/rhubarb/src/audio/voiceActivityDetection.cpp b/rhubarb/src/audio/voiceActivityDetection.cpp
index 8de372c..2e5ae17 100644
--- a/rhubarb/src/audio/voiceActivityDetection.cpp
+++ b/rhubarb/src/audio/voiceActivityDetection.cpp
@@ -8,7 +8,7 @@
 #include "processing.h"
 #include <gsl_util.h>
 #include "tools/parallel.h"
-#include "AudioSegment.h"
+#include <webrtc/common_audio/vad/vad_core.h>
 
 using std::vector;
 using boost::adaptors::transformed;
@@ -16,124 +16,65 @@ using fmt::format;
 using std::runtime_error;
 using std::unique_ptr;
 
-JoiningBoundedTimeline<void> webRtcDetectVoiceActivity(
-	const AudioClip& audioClip,
+JoiningBoundedTimeline<void> detectVoiceActivity(
+	const AudioClip& inputAudioClip,
 	ProgressSink& progressSink
 ) {
+	// Prepare audio for VAD
+	constexpr int webRtcSamplingRate = 8000;
+	const unique_ptr<AudioClip> audioClip = inputAudioClip.clone()
+		| resample(webRtcSamplingRate)
+		| removeDcOffset();
+
 	VadInst* vadHandle = WebRtcVad_Create();
 	if (!vadHandle) throw runtime_error("Error creating WebRTC VAD handle.");
 
 	auto freeHandle = gsl::finally([&]() { WebRtcVad_Free(vadHandle); });
 
 	int error = WebRtcVad_Init(vadHandle);
-	if (error) throw runtime_error("Error initializing WebRTC VAD handle.");
+	if (error) throw runtime_error("Error initializing WebRTC VAD.");
 
 	const int aggressiveness = 2; // 0..3. The higher, the more is cut off.
 	error = WebRtcVad_set_mode(vadHandle, aggressiveness);
 	if (error) throw runtime_error("Error setting WebRTC VAD aggressiveness.");
 
-	ProgressMerger progressMerger(progressSink);
-	ProgressSink& pass1ProgressSink = progressMerger.addSource("VAD pass 1", 1.0);
-	ProgressSink& pass2ProgressSink = progressMerger.addSource("VAD pass 2", 0.3);
-
 	// Detect activity
-	JoiningBoundedTimeline<void> activity(audioClip.getTruncatedRange());
+	JoiningBoundedTimeline<void> activity(audioClip->getTruncatedRange());
 	centiseconds time = 0_cs;
-	const size_t bufferCapacity = audioClip.getSampleRate() / 100;
+	const size_t frameSize = webRtcSamplingRate / 100;
 	const auto processBuffer = [&](const vector<int16_t>& buffer) {
 		// WebRTC is picky regarding buffer size
-		if (buffer.size() < bufferCapacity) return;
+		if (buffer.size() < frameSize) return;
 
 		const int result = WebRtcVad_Process(
 			vadHandle,
-			audioClip.getSampleRate(),
+			webRtcSamplingRate,
 			buffer.data(),
 			buffer.size()
-		) == 1;
+		);
 		if (result == -1) throw runtime_error("Error processing audio buffer using WebRTC VAD.");
 
-		const bool isActive = result != 0;
+		// Ignore the result of WebRtcVad_Process, instead directly interpret the internal VAD flag.
+		// The result of WebRtcVad_Process stays 1 for a number of frames after the last detected
+		// activity.
+		const bool isActive = reinterpret_cast<VadInstT*>(vadHandle)->vad == 1;
+
 		if (isActive) {
 			activity.set(time, time + 1_cs);
 		}
+
 		time += 1_cs;
 	};
-	process16bitAudioClip(audioClip, processBuffer, bufferCapacity, pass1ProgressSink);
-
-	// WebRTC adapts to the audio. This means results may not be correct at the very beginning.
-	// It sometimes returns false activity at the very beginning, mistaking the background noise for
-	// speech.
-	// So we delete the first recognized utterance and re-process the corresponding audio segment.
-	if (!activity.empty()) {
-		TimeRange firstActivity = activity.begin()->getTimeRange();
-		activity.clear(firstActivity);
-		const unique_ptr<AudioClip> streamStart = audioClip.clone()
-			| segment(TimeRange(0_cs, firstActivity.getEnd()));
-		time = 0_cs;
-		process16bitAudioClip(*streamStart, processBuffer, bufferCapacity, pass2ProgressSink);
-	}
-
-	return activity;
-}
-
-JoiningBoundedTimeline<void> detectVoiceActivity(
-	const AudioClip& inputAudioClip,
-	int maxThreadCount,
-	ProgressSink& progressSink
-) {
-	// Prepare audio for VAD
-	const unique_ptr<AudioClip> audioClip = inputAudioClip.clone()
-		| resample(8000) // Convert to the internal sampling rate of WebRTC
-		| removeDcOffset();
-
-	JoiningBoundedTimeline<void> activity(audioClip->getTruncatedRange());
-	std::mutex activityMutex;
-
-	// Split audio into segments and perform parallel VAD
-	const int segmentCount = maxThreadCount;
-	const centiseconds audioDuration = audioClip->getTruncatedRange().getDuration();
-	vector<TimeRange> audioSegments;
-	for (int i = 0; i < segmentCount; ++i) {
-		TimeRange segmentRange = TimeRange(
-			i * audioDuration / segmentCount,
-			(i + 1) * audioDuration / segmentCount
-		);
-		audioSegments.push_back(segmentRange);
-	}
-	runParallel(
-		"VAD",
-		[&](const TimeRange& segmentRange, ProgressSink& segmentProgressSink) {
-			const unique_ptr<AudioClip> audioSegment = audioClip->clone() | segment(segmentRange);
-			JoiningBoundedTimeline<void> activitySegment =
-				webRtcDetectVoiceActivity(*audioSegment, segmentProgressSink);
-
-			std::lock_guard<std::mutex> lock(activityMutex);
-			for (auto activityRange : activitySegment) {
-				activityRange.getTimeRange().shift(segmentRange.getStart());
-				activity.set(activityRange);
-			}
-		},
-		audioSegments,
-		segmentCount,
-		progressSink
-	);
+	process16bitAudioClip(*audioClip, processBuffer, frameSize, progressSink);
 
 	// Fill small gaps in activity
-	const centiseconds maxGap(5);
+	const centiseconds maxGap(10);
 	for (const auto& pair : getPairs(activity)) {
 		if (pair.second.getStart() - pair.first.getEnd() <= maxGap) {
 			activity.set(pair.first.getEnd(), pair.second.getStart());
 		}
 	}
 
-	// Shorten activities. WebRTC adds a bit of buffer at the end.
-	const centiseconds tail(5);
-	for (const auto& utterance : JoiningBoundedTimeline<void>(activity)) {
-		if (utterance.getDuration() > tail && utterance.getEnd() < audioDuration) {
-			activity.clear(utterance.getEnd() - tail, utterance.getEnd());
-		}
-	}
-
 	logging::debugFormat(
 		"Found {} sections of voice activity: {}",
 		activity.size(),
diff --git a/rhubarb/src/audio/voiceActivityDetection.h b/rhubarb/src/audio/voiceActivityDetection.h
index 02658e0..4e85f9f 100644
--- a/rhubarb/src/audio/voiceActivityDetection.h
+++ b/rhubarb/src/audio/voiceActivityDetection.h
@@ -5,6 +5,5 @@
 
 JoiningBoundedTimeline<void> detectVoiceActivity(
 	const AudioClip& audioClip,
-	int maxThreadCount,
 	ProgressSink& progressSink
 );
diff --git a/rhubarb/src/recognition/pocketSphinxTools.cpp b/rhubarb/src/recognition/pocketSphinxTools.cpp
index 4e9c117..cdb91b5 100644
--- a/rhubarb/src/recognition/pocketSphinxTools.cpp
+++ b/rhubarb/src/recognition/pocketSphinxTools.cpp
@@ -102,7 +102,7 @@ BoundedTimeline<Phone> recognizePhones(
 	// Split audio into utterances
 	JoiningBoundedTimeline<void> utterances;
 	try {
-		utterances = detectVoiceActivity(*audioClip, maxThreadCount, voiceActivationProgressSink);
+		utterances = detectVoiceActivity(*audioClip, voiceActivationProgressSink);
 	} catch (...) {
 		std::throw_with_nested(runtime_error("Error detecting segments of speech."));
 	}