Refactored audio handling

Now audio clips can be passed around as const references
and don't carry state any more.

CMakeLists.txt

@@ -197,11 +197,10 @@ set(SOURCE_FILES
 	src/phoneExtraction.cpp src/phoneExtraction.h
 	src/platformTools.cpp src/platformTools.h
 	src/tools.cpp src/tools.h
-	src/audio/AudioStream.cpp src/audio/AudioStream.h
-	src/audio/AudioStreamSegment.cpp src/audio/AudioStreamSegment.h
+	src/audio/AudioClip.cpp src/audio/AudioClip.h
+	src/audio/AudioSegment.cpp src/audio/AudioSegment.h
 	src/audio/DCOffset.cpp src/audio/DCOffset.h
 	src/audio/SampleRateConverter.cpp src/audio/SampleRateConverter.h
-	src/audio/UnboundedStream.cpp src/audio/UnboundedStream.h
 	src/audio/voiceActivityDetection.cpp src/audio/voiceActivityDetection.h
 	src/audio/WaveFileReader.cpp src/audio/WaveFileReader.h
 	src/audio/waveFileWriting.cpp src/audio/waveFileWriting.h

src/audio/AudioClip.cpp

@@ -0,0 +1,65 @@
+#include "AudioClip.h"
+#include <format.h>
+
+using std::invalid_argument;
+
+TimeRange AudioClip::getTruncatedRange() const {
+	return TimeRange(0cs, centiseconds(100 * size() / getSampleRate()));
+}
+
+class SafeSampleReader {
+public:
+	SafeSampleReader(SampleReader unsafeRead, AudioClip::size_type size);
+	AudioClip::value_type operator()(AudioClip::size_type index);
+private:
+	SampleReader unsafeRead;
+	AudioClip::size_type size;
+	AudioClip::size_type lastIndex = -1;
+	AudioClip::value_type lastSample = 0;
+};
+
+SafeSampleReader::SafeSampleReader(SampleReader unsafeRead, AudioClip::size_type size) :
+	unsafeRead(unsafeRead),
+	size(size)
+{}
+
+inline AudioClip::value_type SafeSampleReader::operator()(AudioClip::size_type index) {
+	if (index < 0) {
+		throw invalid_argument(fmt::format("Cannot read from sample index {}. Index < 0.", index));
+	}
+	if (index >= size) {
+		throw invalid_argument(fmt::format("Cannot read from sample index {}. Clip size is {}.", index, size));
+	}
+	if (index == lastIndex) {
+		return lastSample;
+	}
+
+	lastIndex = index;
+	lastSample = unsafeRead(index);
+	return lastSample;
+}
+
+SampleReader AudioClip::createSampleReader() const {
+	return SafeSampleReader(createUnsafeSampleReader(), size());
+}
+
+AudioClip::iterator AudioClip::begin() const {
+	return SampleIterator(*this, 0);
+}
+
+AudioClip::iterator AudioClip::end() const {
+	return SampleIterator(*this, size());
+}
+
+std::unique_ptr<AudioClip> operator|(std::unique_ptr<AudioClip> clip, AudioEffect effect) {
+	return effect(std::move(clip));
+}
+
+SampleIterator::SampleIterator() :
+	sampleIndex(0)
+{}
+
+SampleIterator::SampleIterator(const AudioClip& audioClip, size_type sampleIndex) :
+	sampleReader([&audioClip] { return audioClip.createSampleReader(); }),
+	sampleIndex(sampleIndex)
+{}

src/audio/AudioClip.h

@@ -0,0 +1,141 @@
+#pragma once
+#include <memory>
+#include "TimeRange.h"
+#include <functional>
+#include "Lazy.h"
+
+class AudioClip;
+class SampleIterator;
+
+class AudioClip {
+public:
+	using value_type = float;
+	using size_type = int64_t;
+	using difference_type = int64_t;
+	using iterator = SampleIterator;
+	using SampleReader = std::function<value_type(size_type)>;
+
+	virtual ~AudioClip() {}
+	virtual std::unique_ptr<AudioClip> clone() const = 0;
+	virtual int getSampleRate() const = 0;
+	virtual size_type size() const = 0;
+	TimeRange getTruncatedRange() const;
+	SampleReader createSampleReader() const;
+	iterator begin() const;
+	iterator end() const;
+private:
+	virtual SampleReader createUnsafeSampleReader() const = 0;
+};
+
+using AudioEffect = std::function<std::unique_ptr<AudioClip>(std::unique_ptr<AudioClip>)>;
+
+std::unique_ptr<AudioClip> operator|(std::unique_ptr<AudioClip> clip, AudioEffect effect);
+
+using SampleReader = AudioClip::SampleReader;
+
+class SampleIterator {
+public:
+	using value_type = AudioClip::value_type;
+	using size_type = AudioClip::size_type;
+	using difference_type = AudioClip::difference_type;
+
+	SampleIterator();
+
+	size_type getSampleIndex() const;
+	void seek(size_type sampleIndex);
+	value_type operator*() const;
+	value_type operator[](difference_type n) const;
+
+private:
+	friend AudioClip;
+	SampleIterator(const AudioClip& audioClip, size_type sampleIndex);
+
+	Lazy<SampleReader> sampleReader;
+	size_type sampleIndex;
+};
+
+inline SampleIterator::size_type SampleIterator::getSampleIndex() const {
+	return sampleIndex;
+}
+
+inline void SampleIterator::seek(size_type sampleIndex) {
+	this->sampleIndex = sampleIndex;
+}
+
+inline SampleIterator::value_type SampleIterator::operator*() const {
+	return (*sampleReader)(sampleIndex);
+}
+
+inline SampleIterator::value_type SampleIterator::operator[](difference_type n) const {
+	return (*sampleReader)(sampleIndex + n);
+}
+
+inline bool operator==(const SampleIterator& lhs, const SampleIterator& rhs) {
+	return lhs.getSampleIndex() == rhs.getSampleIndex();
+}
+
+inline bool operator!=(const SampleIterator& lhs, const SampleIterator& rhs) {
+	return lhs.getSampleIndex() != rhs.getSampleIndex();
+}
+
+inline bool operator<(const SampleIterator& lhs, const SampleIterator& rhs) {
+	return lhs.getSampleIndex() < rhs.getSampleIndex();
+}
+
+inline bool operator>(const SampleIterator& lhs, const SampleIterator& rhs) {
+	return lhs.getSampleIndex() > rhs.getSampleIndex();
+}
+
+inline bool operator<=(const SampleIterator& lhs, const SampleIterator& rhs) {
+	return lhs.getSampleIndex() <= rhs.getSampleIndex();
+}
+
+inline bool operator>=(const SampleIterator& lhs, const SampleIterator& rhs) {
+	return lhs.getSampleIndex() >= rhs.getSampleIndex();
+}
+
+inline SampleIterator& operator+=(SampleIterator& it, SampleIterator::difference_type n) {
+	it.seek(it.getSampleIndex() + n);
+	return it;
+}
+
+inline SampleIterator& operator-=(SampleIterator& it, SampleIterator::difference_type n) {
+	it.seek(it.getSampleIndex() - n);
+	return it;
+}
+
+inline SampleIterator& operator++(SampleIterator& it) {
+	return operator+=(it, 1);
+}
+
+inline SampleIterator operator++(SampleIterator& it, int) {
+	SampleIterator tmp(it);
+	operator++(it);
+	return tmp;
+}
+
+inline SampleIterator& operator--(SampleIterator& it) {
+	return operator-=(it, 1);
+}
+
+inline SampleIterator operator--(SampleIterator& it, int) {
+	SampleIterator tmp(it);
+	operator--(it);
+	return tmp;
+}
+
+inline SampleIterator operator+(const SampleIterator& it, SampleIterator::difference_type n) {
+	SampleIterator result(it);
+	result += n;
+	return result;
+}
+
+inline SampleIterator operator-(const SampleIterator& it, SampleIterator::difference_type n) {
+	SampleIterator result(it);
+	result -= n;
+	return result;
+}
+
+inline SampleIterator::difference_type operator-(const SampleIterator& lhs, const SampleIterator& rhs) {
+	return lhs.getSampleIndex() - rhs.getSampleIndex();
+}

src/audio/AudioSegment.cpp

@@ -0,0 +1,30 @@
+#include "AudioSegment.h"
+
+using std::unique_ptr;
+using std::make_unique;
+
+AudioSegment::AudioSegment(std::unique_ptr<AudioClip> inputClip, const TimeRange& range) :
+	inputClip(std::move(inputClip)),
+	sampleOffset(static_cast<int64_t>(range.getStart().count()) * this->inputClip->getSampleRate() / 100),
+	sampleCount(static_cast<int64_t>(range.getLength().count()) * this->inputClip->getSampleRate() / 100)
+{
+	if (sampleOffset < 0 || sampleOffset + sampleCount > this->inputClip->size()) {
+		throw std::invalid_argument("Segment extends beyond input clip.");
+	}
+}
+
+unique_ptr<AudioClip> AudioSegment::clone() const {
+	return make_unique<AudioSegment>(*this);
+}
+
+SampleReader AudioSegment::createUnsafeSampleReader() const {
+	return [read = inputClip->createSampleReader(), sampleOffset = sampleOffset](size_type index) {
+		return read(index + sampleOffset);
+	};
+}
+
+AudioEffect segment(const TimeRange& range) {
+	return [range](unique_ptr<AudioClip> inputClip) {
+		return make_unique<AudioSegment>(std::move(inputClip), range);
+	};
+}

src/audio/AudioSegment.h

@@ -0,0 +1,26 @@
+#pragma once
+#include "AudioClip.h"
+
+class AudioSegment : public AudioClip {
+public:
+	AudioSegment(std::unique_ptr<AudioClip> inputClip, const TimeRange& range);
+	std::unique_ptr<AudioClip> clone() const override;
+	int getSampleRate() const override;
+	size_type size() const override;
+
+private:
+	SampleReader createUnsafeSampleReader() const override;
+
+	std::shared_ptr<AudioClip> inputClip;
+	size_type sampleOffset, sampleCount;
+};
+
+inline int AudioSegment::getSampleRate() const {
+	return inputClip->getSampleRate();
+}
+
+inline AudioClip::size_type AudioSegment::size() const {
+	return sampleCount;
+}
+
+AudioEffect segment(const TimeRange& range);

src/audio/AudioStream.cpp

@@ -1,9 +0,0 @@
-#include "AudioStream.h"
-
-TimeRange AudioStream::getTruncatedRange() const {
-	return TimeRange(0cs, centiseconds(100 * getSampleCount() / getSampleRate()));
-}
-
-bool AudioStream::endOfStream() const {
-	return getSampleIndex() >= getSampleCount();
-}

src/audio/AudioStream.h

@@ -1,18 +0,0 @@
-#pragma once
-
-#include <memory>
-#include "TimeRange.h"
-
-// A mono stream of floating-point samples.
-class AudioStream {
-public:
-	virtual ~AudioStream() {}
-	virtual std::unique_ptr<AudioStream> clone(bool reset) const = 0;
-	virtual int getSampleRate() const = 0;
-	virtual int64_t getSampleCount() const = 0;
-	TimeRange getTruncatedRange() const;
-	virtual int64_t getSampleIndex() const = 0;
-	virtual void seek(int64_t sampleIndex) = 0;
-	bool endOfStream() const;
-	virtual float readSample() = 0;
-};

src/audio/AudioStreamSegment.cpp

@@ -1,50 +0,0 @@
-#include "AudioStreamSegment.h"
-#include <stdexcept>
-
-AudioStreamSegment::AudioStreamSegment(std::unique_ptr<AudioStream> audioStream, const TimeRange& range) :
-	audioStream(std::move(audioStream)),
-	sampleOffset(static_cast<int64_t>(range.getStart().count()) * this->audioStream->getSampleRate() / 100),
-	sampleCount(static_cast<int64_t>(range.getLength().count()) * this->audioStream->getSampleRate() / 100)
-{
-	seek(0);
-
-	if (sampleOffset < 0 || sampleOffset + sampleCount > this->audioStream->getSampleCount()) {
-		throw std::invalid_argument("Segment extends beyond input stream.");
-	}
-}
-
-AudioStreamSegment::AudioStreamSegment(const AudioStreamSegment& rhs, bool reset) :
-	audioStream(rhs.audioStream->clone(false)),
-	sampleOffset(rhs.sampleOffset),
-	sampleCount(rhs.sampleCount)
-{
-	if (reset) seek(0);
-}
-
-std::unique_ptr<AudioStream> AudioStreamSegment::clone(bool reset) const {
-	return std::make_unique<AudioStreamSegment>(*this, reset);
-}
-
-int AudioStreamSegment::getSampleRate() const {
-	return audioStream->getSampleRate();
-}
-
-int64_t AudioStreamSegment::getSampleCount() const {
-	return sampleCount;
-}
-
-int64_t AudioStreamSegment::getSampleIndex() const {
-	return audioStream->getSampleIndex() - sampleOffset;
-}
-
-void AudioStreamSegment::seek(int64_t sampleIndex) {
-	audioStream->seek(sampleIndex + sampleOffset);
-}
-
-float AudioStreamSegment::readSample() {
-	return audioStream->readSample();
-}
-
-std::unique_ptr<AudioStream> createSegment(std::unique_ptr<AudioStream> audioStream, const TimeRange& range) {
-	return std::make_unique<AudioStreamSegment>(std::move(audioStream), range);
-}

src/audio/AudioStreamSegment.h

@@ -1,21 +0,0 @@
-#pragma once
-#include <audio/AudioStream.h>
-#include <TimeRange.h>
-
-class AudioStreamSegment : public AudioStream {
-public:
-	AudioStreamSegment(std::unique_ptr<AudioStream> audioStream, const TimeRange& range);
-	AudioStreamSegment(const AudioStreamSegment& rhs, bool reset);
-	std::unique_ptr<AudioStream> clone(bool reset) const override;
-	int getSampleRate() const override;
-	int64_t getSampleCount() const override;
-	int64_t getSampleIndex() const override;
-	void seek(int64_t sampleIndex) override;
-	float readSample() override;
-
-private:
-	std::unique_ptr<AudioStream> audioStream;
-	const int64_t sampleOffset, sampleCount;
-};
-
-std::unique_ptr<AudioStream> createSegment(std::unique_ptr<AudioStream> audioStream, const TimeRange& range);

src/audio/DCOffset.cpp

@@ -1,73 +1,46 @@
 #include "DCOffset.h"
-#include <gsl_util.h>
-#include <cmath>
 
-DCOffset::DCOffset(std::unique_ptr<AudioStream> inputStream, float offset) :
-	inputStream(std::move(inputStream)),
+using std::unique_ptr;
+using std::make_unique;
+
+DCOffset::DCOffset(unique_ptr<AudioClip> inputClip, float offset) :
+	inputClip(std::move(inputClip)),
 	offset(offset),
 	factor(1 / (1 + std::abs(offset)))
 {}
 
-DCOffset::DCOffset(const DCOffset& rhs, bool reset) :
-	inputStream(rhs.inputStream->clone(reset)),
-	offset(rhs.offset),
-	factor(rhs.factor)
-{}
-
-std::unique_ptr<AudioStream> DCOffset::clone(bool reset) const {
-	return std::make_unique<DCOffset>(*this, reset);
+unique_ptr<AudioClip> DCOffset::clone() const {
+	return make_unique<DCOffset>(*this);
 }
 
-int DCOffset::getSampleRate() const {
-	return inputStream->getSampleRate();
+SampleReader DCOffset::createUnsafeSampleReader() const {
+	return [read = inputClip->createSampleReader(), factor = factor, offset = offset](size_type index) {
+		float sample = read(index);
+		return sample * factor + offset;
+	};
 }
 
-int64_t DCOffset::getSampleCount() const {
-	return inputStream->getSampleCount();
-}
-
-int64_t DCOffset::getSampleIndex() const {
-	return inputStream->getSampleIndex();
-}
-
-void DCOffset::seek(int64_t sampleIndex) {
-	inputStream->seek(sampleIndex);
-}
-
-float DCOffset::readSample() {
-	float sample = inputStream->readSample();
-	return sample * factor + offset;
-}
-
-std::unique_ptr<AudioStream> addDCOffset(std::unique_ptr<AudioStream> audioStream, float offset, float epsilon) {
-	if (std::abs(offset) < epsilon) return audioStream;
-	return std::make_unique<DCOffset>(std::move(audioStream), offset);
-}
-
-float getDCOffset(AudioStream& audioStream) {
+float getDCOffset(const AudioClip& audioClip) {
 	int flatMeanSampleCount, fadingMeanSampleCount;
-	int sampleRate = audioStream.getSampleRate();
-	if (audioStream.getSampleCount() > 4 * sampleRate) {
+	int sampleRate = audioClip.getSampleRate();
+	if (audioClip.size() > 4 * sampleRate) {
 		// Long audio file. Average over the first 3 seconds, then fade out over the 4th.
 		flatMeanSampleCount = 3 * sampleRate;
 		fadingMeanSampleCount = 1 * sampleRate;
 	} else {
 		// Short audio file. Average over the entire length.
-		flatMeanSampleCount = static_cast<int>(audioStream.getSampleCount());
+		flatMeanSampleCount = static_cast<int>(audioClip.size());
 		fadingMeanSampleCount = 0;
 	}
 
-	int64_t originalSampleIndex = audioStream.getSampleIndex();
-	audioStream.seek(0);
-	auto restorePosition = gsl::finally([&]() { audioStream.seek(originalSampleIndex); });
-
+	auto read = audioClip.createSampleReader();
 	double sum = 0;
-	for (int i = 0; i < flatMeanSampleCount; i++) {
-		sum += audioStream.readSample();
+	for (int i = 0; i < flatMeanSampleCount; ++i) {
+		sum += read(i);
 	}
-	for (int i = 0; i < fadingMeanSampleCount; i++) {
+	for (int i = 0; i < fadingMeanSampleCount; ++i) {
 		double weight = static_cast<double>(fadingMeanSampleCount - i) / fadingMeanSampleCount;
-		sum += audioStream.readSample() * weight;
+		sum += read(flatMeanSampleCount + i) * weight;
 	}
 
 	double totalWeight = flatMeanSampleCount + (fadingMeanSampleCount + 1) / 2.0;
@@ -75,7 +48,16 @@ float getDCOffset(AudioStream& audioStream) {
 	return static_cast<float>(offset);
 }
 
-std::unique_ptr<AudioStream> removeDCOffset(std::unique_ptr<AudioStream> inputStream) {
-	float offset = getDCOffset(*inputStream.get());
-	return addDCOffset(std::move(inputStream), -offset);
+AudioEffect addDCOffset(float offset, float epsilon) {
+	return [offset, epsilon](unique_ptr<AudioClip> inputClip) -> unique_ptr<AudioClip> {
+		if (std::abs(offset) < epsilon) return std::move(inputClip);
+		return make_unique<DCOffset>(std::move(inputClip), offset);
+	};
+}
+
+AudioEffect removeDCOffset(float epsilon) {
+	return [epsilon](unique_ptr<AudioClip> inputClip) {
+		float offset = getDCOffset(*inputClip);
+		return std::move(inputClip) | addDCOffset(-offset, epsilon);
+	};
 }

src/audio/DCOffset.h

@@ -1,26 +1,32 @@
 #pragma once
 
-#include "AudioStream.h"
+#include "AudioClip.h"
 
-// Applies a constant DC offset to an audio stream and reduces its amplitude
+// Applies a constant DC offset to an audio clip and reduces its amplitude
 // to prevent clipping
-class DCOffset : public AudioStream {
+class DCOffset : public AudioClip {
 public:
-	DCOffset(std::unique_ptr<AudioStream> inputStream, float offset);
-	DCOffset(const DCOffset& rhs, bool reset);
-	std::unique_ptr<AudioStream> clone(bool reset) const override;
+	DCOffset(std::unique_ptr<AudioClip> inputClip, float offset);
+	std::unique_ptr<AudioClip> clone() const override;
 	int getSampleRate() const override;
-	int64_t getSampleCount() const override;
-	int64_t getSampleIndex() const override;
-	void seek(int64_t sampleIndex) override;
-	float readSample() override;
-
+	size_type size() const override;
 private:
-	std::unique_ptr<AudioStream> inputStream;
+	SampleReader createUnsafeSampleReader() const override;
+
+	std::shared_ptr<AudioClip> inputClip;
 	float offset;
 	float factor;
 };
 
-std::unique_ptr<AudioStream> addDCOffset(std::unique_ptr<AudioStream> audioStream, float offset, float epsilon = 1.0f / 15000);
+inline int DCOffset::getSampleRate() const {
+	return inputClip->getSampleRate();
+}
 
-std::unique_ptr<AudioStream> removeDCOffset(std::unique_ptr<AudioStream> audioStream);
+inline AudioClip::size_type DCOffset::size() const {
+	return inputClip->size();
+}
+
+float getDCOffset(const AudioClip& audioClip);
+
+AudioEffect addDCOffset(float offset, float epsilon = 1.0f / 15000);
+AudioEffect removeDCOffset(float epsilon = 1.0f / 15000);

src/audio/SampleRateConverter.cpp

@@ -1,105 +1,62 @@
 #include <cmath>
 #include "SampleRateConverter.h"
 #include <stdexcept>
-#include <algorithm>
 #include <format.h>
 
 using std::invalid_argument;
+using std::unique_ptr;
+using std::make_unique;
 
-SampleRateConverter::SampleRateConverter(std::unique_ptr<AudioStream> inputStream, int outputSampleRate) :
-	inputStream(std::move(inputStream)),
-	downscalingFactor(static_cast<double>(this->inputStream->getSampleRate()) / outputSampleRate),
+SampleRateConverter::SampleRateConverter(unique_ptr<AudioClip> inputClip, int outputSampleRate) :
+	inputClip(std::move(inputClip)),
+	downscalingFactor(static_cast<double>(this->inputClip->getSampleRate()) / outputSampleRate),
 	outputSampleRate(outputSampleRate),
-	outputSampleCount(std::lround(this->inputStream->getSampleCount() / downscalingFactor)),
-	lastInputSample(0),
-	lastInputSampleIndex(-1),
-	nextOutputSampleIndex(0)
+	outputSampleCount(std::lround(this->inputClip->size() / downscalingFactor))
 {
 	if (outputSampleRate <= 0) {
 		throw invalid_argument("Sample rate must be positive.");
 	}
-	if (this->inputStream->getSampleRate() < outputSampleRate) {
-		throw invalid_argument(fmt::format("Upsampling not supported. Audio sample rate must not be below {}Hz.", outputSampleRate));
+	if (this->inputClip->getSampleRate() < outputSampleRate) {
+		throw invalid_argument(fmt::format("Upsampling not supported. Input sample rate must not be below {}Hz.", outputSampleRate));
 	}
 }
 
-SampleRateConverter::SampleRateConverter(const SampleRateConverter& rhs, bool reset) :
-	SampleRateConverter(rhs.inputStream->clone(reset), rhs.outputSampleRate)
-{
-	nextOutputSampleIndex = reset ? 0 : rhs.nextOutputSampleIndex;
+unique_ptr<AudioClip> SampleRateConverter::clone() const {
+	return make_unique<SampleRateConverter>(*this);
 }
 
-std::unique_ptr<AudioStream> SampleRateConverter::clone(bool reset) const {
-	return std::make_unique<SampleRateConverter>(*this, reset);
-}
-
-int SampleRateConverter::getSampleRate() const {
-	return outputSampleRate;
-}
-
-int64_t SampleRateConverter::getSampleCount() const {
-	return outputSampleCount;
-}
-
-int64_t SampleRateConverter::getSampleIndex() const {
-	return nextOutputSampleIndex;
-}
-
-void SampleRateConverter::seek(int64_t sampleIndex) {
-	if (sampleIndex < 0 || sampleIndex >= outputSampleCount) throw std::invalid_argument("sampleIndex out of range.");
-
-	nextOutputSampleIndex = sampleIndex;
-}
-
-float SampleRateConverter::readSample() {
-	if (nextOutputSampleIndex >= outputSampleCount) throw std::out_of_range("End of stream.");
-
-	double inputStart = nextOutputSampleIndex * downscalingFactor;
-	double inputEnd = (nextOutputSampleIndex + 1) * downscalingFactor;
-
-	nextOutputSampleIndex++;
-	return mean(inputStart, inputEnd);
-}
-
-float SampleRateConverter::mean(double inputStart, double inputEnd) {
+float mean(double inputStart, double inputEnd, const SampleReader& read) {
 	// Calculate weighted sum...
 	double sum = 0;
 
 	// ... first sample (weight <= 1)
 	int64_t startIndex = static_cast<int64_t>(inputStart);
-	sum += getInputSample(startIndex) * ((startIndex + 1) - inputStart);
+	sum += read(startIndex) * ((startIndex + 1) - inputStart);
 
 	// ... middle samples (weight 1 each)
 	int64_t endIndex = static_cast<int64_t>(inputEnd);
 	for (int64_t index = startIndex + 1; index < endIndex; ++index) {
-		sum += getInputSample(index);
+		sum += read(index);
 	}
 
 	// ... last sample (weight < 1)
-	sum += getInputSample(endIndex) * (inputEnd - endIndex);
+	if (endIndex < inputEnd) {
+		sum += read(endIndex) * (inputEnd - endIndex);
+	}
 
 	return static_cast<float>(sum / (inputEnd - inputStart));
 }
 
-float SampleRateConverter::getInputSample(int64_t sampleIndex) {
-	sampleIndex = std::min(sampleIndex, inputStream->getSampleCount() - 1);
-	if (sampleIndex < 0) return 0.0f;
-
-	if (sampleIndex == lastInputSampleIndex) {
-		return lastInputSample;
-	}
-
-	if (sampleIndex != inputStream->getSampleIndex()) {
-		inputStream->seek(sampleIndex);
-	}
-	lastInputSample = inputStream->readSample();
-	lastInputSampleIndex = sampleIndex;
-	return lastInputSample;
+SampleReader SampleRateConverter::createUnsafeSampleReader() const {
+	return[read = inputClip->createSampleReader(), downscalingFactor = downscalingFactor, size = size()](size_type index) {
+		double inputStart = index * downscalingFactor;
+		double inputEnd = std::min((index + 1) * downscalingFactor, static_cast<double>(size));
+		return mean(inputStart, inputEnd, read);
+	};
 }
 
-std::unique_ptr<AudioStream> convertSampleRate(std::unique_ptr<AudioStream> audioStream, int sampleRate) {
-	if (sampleRate == audioStream->getSampleRate()) {
-		return audioStream;
-	}
-	return std::make_unique<SampleRateConverter>(std::move(audioStream), sampleRate);
+AudioEffect resample(int sampleRate) {
+	return [sampleRate](unique_ptr<AudioClip> inputClip) {
+		return make_unique<SampleRateConverter>(std::move(inputClip), sampleRate);
+	};
 }

src/audio/SampleRateConverter.h

@@ -1,32 +1,29 @@
 #pragma once
 
 #include <memory>
-#include "AudioStream.h"
+#include "AudioClip.h"
 
-class SampleRateConverter : public AudioStream {
+class SampleRateConverter : public AudioClip {
 public:
-	SampleRateConverter(std::unique_ptr<AudioStream> inputStream, int outputSampleRate);
-	SampleRateConverter(const SampleRateConverter& rhs, bool reset);
-	std::unique_ptr<AudioStream> clone(bool reset) const override;
+	SampleRateConverter(std::unique_ptr<AudioClip> inputClip, int outputSampleRate);
+	std::unique_ptr<AudioClip> clone() const override;
 	int getSampleRate() const override;
-	int64_t getSampleCount() const override;
-	int64_t getSampleIndex() const override;
-	void seek(int64_t sampleIndex) override;
-	float readSample() override;
+	size_type size() const override;
 private:
-	std::unique_ptr<AudioStream> inputStream;
-	double downscalingFactor;					// input sample rate / output sample rate
+	SampleReader createUnsafeSampleReader() const override;
 
+	std::shared_ptr<AudioClip> inputClip;
+	double downscalingFactor; // input sample rate / output sample rate
 	int outputSampleRate;
 	int64_t outputSampleCount;
-
-	float lastInputSample;
-	int64_t lastInputSampleIndex;
-
-	int64_t nextOutputSampleIndex;
-
-	float mean(double start, double end);
-	float getInputSample(int64_t sampleIndex);
 };
 
-std::unique_ptr<AudioStream> convertSampleRate(std::unique_ptr<AudioStream> audioStream, int sampleRate);
+AudioEffect resample(int sampleRate);
+
+inline int SampleRateConverter::getSampleRate() const {
+	return outputSampleRate;
+}
+
+inline AudioClip::size_type SampleRateConverter::size() const {
+	return outputSampleCount;
+}

src/audio/UnboundedStream.cpp

@@ -1,59 +0,0 @@
-#include "UnboundedStream.h"
-
-using boost::optional;
-
-UnboundedStream::UnboundedStream(std::unique_ptr<AudioStream> inputStream) :
-	innerStream(std::move(inputStream)),
-	sampleIndex(innerStream->getSampleIndex()),
-	firstSample(inputStream->getSampleCount() ? optional<float>() : 0.0f),
-	lastSample(inputStream->getSampleCount() ? optional<float>() : 0.0f)
-{}
-
-UnboundedStream::UnboundedStream(const UnboundedStream& rhs, bool reset) :
-	innerStream(rhs.innerStream->clone(reset)),
-	sampleIndex(rhs.sampleIndex),
-	firstSample(rhs.firstSample),
-	lastSample(rhs.lastSample)
-{}
-
-std::unique_ptr<AudioStream> UnboundedStream::clone(bool reset) const {
-	return std::make_unique<UnboundedStream>(*this, reset);
-}
-
-int UnboundedStream::getSampleRate() const {
-	return innerStream->getSampleRate();
-}
-
-int64_t UnboundedStream::getSampleCount() const {
-	return innerStream->getSampleCount();
-}
-
-int64_t UnboundedStream::getSampleIndex() const {
-	return sampleIndex;
-}
-
-void UnboundedStream::seek(int64_t sampleIndex) {
-	this->sampleIndex = sampleIndex;
-}
-
-float UnboundedStream::readSample() {
-	if (sampleIndex < 0) {
-		if (!firstSample) {
-			innerStream->seek(0);
-			firstSample = innerStream->readSample();
-		}
-		return firstSample.get();
-	}
-	if (sampleIndex >= innerStream->getSampleCount()) {
-		if (!lastSample) {
-			innerStream->seek(innerStream->getSampleCount() - 1);
-			lastSample = innerStream->readSample();
-		}
-		return lastSample.get();
-	}
-
-	if (sampleIndex != innerStream->getSampleIndex()) {
-		innerStream->seek(sampleIndex);
-	}
-	return innerStream->readSample();
-}

src/audio/UnboundedStream.h

@@ -1,22 +0,0 @@
-#pragma once
-
-#include "AudioStream.h"
-#include <boost/optional/optional.hpp>
-
-// Stream wrapper that allows reading before the start and past the end of the input stream.
-class UnboundedStream : public AudioStream {
-public:
-	UnboundedStream(std::unique_ptr<AudioStream> inputStream);
-	UnboundedStream(const UnboundedStream& rhs, bool reset);
-	std::unique_ptr<AudioStream> clone(bool reset) const override;
-	int getSampleRate() const override;
-	int64_t getSampleCount() const override;
-	int64_t getSampleIndex() const override;
-	void seek(int64_t sampleIndex) override;
-	float readSample() override;
-
-private:
-	std::unique_ptr<AudioStream> innerStream;
-	int64_t sampleIndex;
-	boost::optional<float> firstSample, lastSample;
-};

src/audio/WaveFileReader.cpp

@@ -7,6 +7,10 @@ using std::runtime_error;
 using fmt::format;
 using std::string;
 using namespace little_endian;
+using std::unique_ptr;
+using std::make_unique;
+using std::make_shared;
+using boost::filesystem::path;
 
 #define INT24_MIN (-8388608)
 #define INT24_MAX 8388607
@@ -25,12 +29,34 @@ enum class Codec {
 	Float = 0x03
 };
 
-WaveFileReader::WaveFileReader(boost::filesystem::path filePath) :
+std::ifstream openFile(path filePath) {
+	try {
+		std::ifstream file;
+		file.exceptions(std::ifstream::failbit | std::ifstream::badbit);
+		file.open(filePath.c_str(), std::ios::binary);
+
+		// Error messages on stream exceptions are mostly useless.
+		// Read some dummy data so that we can throw a decent exception in case the file is missing, locked, etc.
+		file.seekg(0, std::ios_base::end);
+		if (file.tellg()) {
+			file.seekg(0);
+			file.get();
+			file.seekg(0);
+		}
+
+		return std::move(file);
+	} catch (const std::ifstream::failure&) {
+		char message[256];
+		strerror_s(message, sizeof message, errno);
+		throw runtime_error(message);
+	}
+}
+
+WaveFileReader::WaveFileReader(path filePath) :
 	filePath(filePath),
-	file(),
-	frameIndex(0)
+	formatInfo{}
 {
-	openFile();
+	auto file = openFile(filePath);
 
 	file.seekg(0, std::ios_base::end);
 	std::streamoff fileSize = file.tellg();
@@ -57,16 +83,15 @@ WaveFileReader::WaveFileReader(boost::filesystem::path filePath) :
 
 	// Read chunks until we reach the data chunk
 	bool reachedDataChunk = false;
-	bytesPerSample = 0;
 	while (!reachedDataChunk && remaining(8)) {
 		uint32_t chunkId = read<uint32_t>(file);
 		int chunkSize = read<uint32_t>(file);
 		switch (chunkId) {
 		case fourcc('f', 'm', 't', ' '): {
 			// Read relevant data
-			Codec codec = (Codec)read<uint16_t>(file);
-			channelCount = read<uint16_t>(file);
-			frameRate = read<uint32_t>(file);
+			Codec codec = static_cast<Codec>(read<uint16_t>(file));
+			formatInfo.channelCount = read<uint16_t>(file);
+			formatInfo.frameRate = read<uint32_t>(file);
 			read<uint32_t>(file); // Bytes per second
 			int frameSize = read<uint16_t>(file);
 			int bitsPerSample = read<uint16_t>(file);
@@ -75,31 +100,32 @@ WaveFileReader::WaveFileReader(boost::filesystem::path filePath) :
 			file.seekg(roundToEven(chunkSize) - 16, file.cur);
 
 			// Determine sample format
+			int bytesPerSample;
 			switch (codec) {
 			case Codec::PCM:
 				// Determine sample size.
 				// According to the WAVE standard, sample sizes that are not multiples of 8 bits
 				// (e.g. 12 bits) can be treated like the next-larger byte size.
 				if (bitsPerSample == 8) {
-					sampleFormat = SampleFormat::UInt8;
+					formatInfo.sampleFormat = SampleFormat::UInt8;
 					bytesPerSample = 1;
 				} else if (bitsPerSample <= 16) {
-					sampleFormat = SampleFormat::Int16;
+					formatInfo.sampleFormat = SampleFormat::Int16;
 					bytesPerSample = 2;
 				} else if (bitsPerSample <= 24) {
-					sampleFormat = SampleFormat::Int24;
+					formatInfo.sampleFormat = SampleFormat::Int24;
 					bytesPerSample = 3;
 				} else {
 					throw runtime_error(
 						format("Unsupported sample format: {}-bit integer samples.", bitsPerSample));
 				}
-				if (bytesPerSample != frameSize / channelCount) {
+				if (bytesPerSample != frameSize / formatInfo.channelCount) {
 					throw runtime_error("Unsupported sample organization.");
 				}
 				break;
 			case Codec::Float:
 				if (bitsPerSample == 32) {
-					sampleFormat = SampleFormat::Float32;
+					formatInfo.sampleFormat = SampleFormat::Float32;
 					bytesPerSample = 4;
 				} else {
 					throw runtime_error(format("Unsupported sample format: {}-bit floating-point samples.", bitsPerSample));
@@ -108,13 +134,13 @@ WaveFileReader::WaveFileReader(boost::filesystem::path filePath) :
 			default:
 				throw runtime_error("Unsupported sample format. Only uncompressed formats are supported.");
 			}
+			formatInfo.bytesPerFrame = bytesPerSample * formatInfo.channelCount;
 			break;
 		}
 		case fourcc('d', 'a', 't', 'a'): {
 			reachedDataChunk = true;
-			dataOffset = file.tellg();
-			int sampleCount = chunkSize / bytesPerSample;
-			frameCount = sampleCount / channelCount;
+			formatInfo.dataOffset = file.tellg();
+			formatInfo.frameCount = chunkSize / formatInfo.bytesPerFrame;
 			break;
 		}
 		default: {
@@ -124,75 +150,13 @@ WaveFileReader::WaveFileReader(boost::filesystem::path filePath) :
 		}
 		}
 	}
-
-	if (!reachedDataChunk) {
-		dataOffset = file.tellg();
-		frameCount = 0;
-	}
 }
 
-WaveFileReader::WaveFileReader(const WaveFileReader& rhs, bool reset) :
-	filePath(rhs.filePath),
-	file(),
-	bytesPerSample(rhs.bytesPerSample),
-	sampleFormat(rhs.sampleFormat),
-	frameRate(rhs.frameRate),
-	frameCount(rhs.frameCount),
-	channelCount(rhs.channelCount),
-	dataOffset(rhs.dataOffset),
-	frameIndex(-1)
-{
-	openFile();
-	seek(reset ? 0 : rhs.frameIndex);
+unique_ptr<AudioClip> WaveFileReader::clone() const {
+	return make_unique<WaveFileReader>(*this);
 }
 
-std::unique_ptr<AudioStream> WaveFileReader::clone(bool reset) const {
-	return std::make_unique<WaveFileReader>(*this, reset);
-}
-
-void WaveFileReader::openFile() {
-	try {
-		file.exceptions(std::ifstream::failbit | std::ifstream::badbit);
-		file.open(filePath, std::ios::binary);
-
-		// Error messages on stream exceptions are mostly useless.
-		// Read some dummy data so that we can throw a decent exception in case the file is missing, locked, etc.
-		file.seekg(0, std::ios_base::end);
-		if (file.tellg()) {
-			file.seekg(0);
-			file.get();
-			file.seekg(0);
-		}
-	} catch (const std::ifstream::failure&) {
-		char message[256];
-		strerror_s(message, sizeof message, errno);
-		throw runtime_error(message);
-	}
-}
-
-int WaveFileReader::getSampleRate() const {
-	return frameRate;
-}
-
-int64_t WaveFileReader::getSampleCount() const {
-	return frameCount;
-}
-
-int64_t WaveFileReader::getSampleIndex() const {
-	return frameIndex;
-}
-
-void WaveFileReader::seek(int64_t frameIndex) {
-	if (frameIndex < 0 || frameIndex > frameCount) throw std::invalid_argument("frameIndex out of range.");
-
-	file.seekg(dataOffset + static_cast<std::streamoff>(frameIndex * channelCount * bytesPerSample));
-	this->frameIndex = frameIndex;
-}
-
-float WaveFileReader::readSample() {
-	if (frameIndex >= frameCount) throw std::out_of_range("End of stream.");
-	++frameIndex;
-
+inline AudioClip::value_type readSample(std::ifstream& file, SampleFormat sampleFormat, int channelCount) {
 	float sum = 0;
 	for (int channelIndex = 0; channelIndex < channelCount; channelIndex++) {
 		switch (sampleFormat) {
@@ -221,3 +185,13 @@ float WaveFileReader::readSample() {
 
 	return sum / channelCount;
 }
+
+SampleReader WaveFileReader::createUnsafeSampleReader() const {
+	return [formatInfo = formatInfo, file = std::make_shared<std::ifstream>(openFile(filePath)), filePos = std::streampos(0)](size_type index) mutable {
+		std::streampos newFilePos = formatInfo.dataOffset + static_cast<std::streamoff>(index * formatInfo.bytesPerFrame);
+		file->seekg(newFilePos);
+		value_type result = readSample(*file, formatInfo.sampleFormat, formatInfo.channelCount);
+		filePos = newFilePos + static_cast<std::streamoff>(formatInfo.bytesPerFrame);
+		return result;
+	};
+}

src/audio/WaveFileReader.h

@@ -1,8 +1,7 @@
 #pragma once
 
 #include <boost/filesystem/path.hpp>
-#include <boost/filesystem/fstream.hpp>
-#include "AudioStream.h"
+#include "AudioClip.h"
 
 enum class SampleFormat {
 	UInt8,
@@ -11,28 +10,33 @@ enum class SampleFormat {
 	Float32
 };
 
-class WaveFileReader : public AudioStream {
+class WaveFileReader : public AudioClip {
 public:
 	WaveFileReader(boost::filesystem::path filePath);
-	WaveFileReader(const WaveFileReader& rhs, bool reset);
-	std::unique_ptr<AudioStream> clone(bool reset) const override;
-	int getSampleRate() const override ;
-	int64_t getSampleCount() const override;
-	int64_t getSampleIndex() const override;
-	void seek(int64_t sampleIndex) override;
-	float readSample() override;
+	std::unique_ptr<AudioClip> clone() const override;
+	int getSampleRate() const override;
+	size_type size() const override;
 
 private:
-	void openFile();
+	SampleReader createUnsafeSampleReader() const override;
+
+	struct WaveFormatInfo {
+		int bytesPerFrame;
+		SampleFormat sampleFormat;
+		int frameRate;
+		int64_t frameCount;
+		int channelCount;
+		std::streampos dataOffset;
+	};
 
-private:
 	boost::filesystem::path filePath;
-	boost::filesystem::ifstream file;
-	int bytesPerSample;
-	SampleFormat sampleFormat;
-	int frameRate;
-	int64_t frameCount;
-	int channelCount;
-	std::streampos dataOffset;
-	int64_t frameIndex;
+	WaveFormatInfo formatInfo;
 };
+
+inline int WaveFileReader::getSampleRate() const {
+	return formatInfo.frameRate;
+}
+
+inline AudioClip::size_type WaveFileReader::size() const {
+	return formatInfo.frameCount;
+}

src/audio/processing.cpp

@@ -10,31 +10,31 @@ inline int16_t floatSampleToInt16(float sample) {
 	return static_cast<int16_t>(((sample + 1) / 2) * (INT16_MAX - INT16_MIN) + INT16_MIN);
 }
 
-void process16bitAudioStream(AudioStream& audioStream, function<void(const vector<int16_t>&)> processBuffer, size_t bufferCapacity, ProgressSink& progressSink) {
+void process16bitAudioClip(const AudioClip& audioClip, function<void(const vector<int16_t>&)> processBuffer, size_t bufferCapacity, ProgressSink& progressSink) {
 	// Process entire sound stream
 	vector<int16_t> buffer;
 	buffer.reserve(bufferCapacity);
 	int sampleCount = 0;
+	auto it = audioClip.begin();
+	auto end = audioClip.end();
 	do {
 		// Read to buffer
 		buffer.clear();
-		while (buffer.size() < bufferCapacity && !audioStream.endOfStream()) {
-			// Read sample
-			float floatSample = audioStream.readSample();
-			int16_t sample = floatSampleToInt16(floatSample);
-			buffer.push_back(sample);
+		for (; buffer.size() < bufferCapacity && it != end; ++it) {
+			// Read sample to buffer
+			buffer.push_back(floatSampleToInt16(*it));
 		}
 
 		// Process buffer
 		processBuffer(buffer);
 
 		sampleCount += buffer.size();
-		progressSink.reportProgress(static_cast<double>(sampleCount) / audioStream.getSampleCount());
+		progressSink.reportProgress(static_cast<double>(sampleCount) / audioClip.size());
 	} while (buffer.size());
 }
 
-void process16bitAudioStream(AudioStream& audioStream, function<void(const vector<int16_t>&)> processBuffer, ProgressSink& progressSink) {
+void process16bitAudioClip(const AudioClip& audioClip, function<void(const vector<int16_t>&)> processBuffer, ProgressSink& progressSink) {
 	const size_t capacity = 1600; // 0.1 second capacity
-	process16bitAudioStream(audioStream, processBuffer, capacity, progressSink);
+	process16bitAudioClip(audioClip, processBuffer, capacity, progressSink);
 }
 

src/audio/processing.h

@@ -2,8 +2,8 @@
 
 #include <vector>
 #include <functional>
-#include "audio/AudioStream.h"
+#include "audio/AudioClip.h"
 #include "ProgressBar.h"
 
-void process16bitAudioStream(AudioStream& audioStream, std::function<void(const std::vector<int16_t>&)> processBuffer, size_t bufferCapacity, ProgressSink& progressSink);
-void process16bitAudioStream(AudioStream& audioStream, std::function<void(const std::vector<int16_t>&)> processBuffer, ProgressSink& progressSink);
+void process16bitAudioClip(const AudioClip& audioClip, std::function<void(const std::vector<int16_t>&)> processBuffer, size_t bufferCapacity, ProgressSink& progressSink);
+void process16bitAudioClip(const AudioClip& audioClip, std::function<void(const std::vector<int16_t>&)> processBuffer, ProgressSink& progressSink);

src/audio/voiceActivityDetection.cpp

@@ -8,7 +8,7 @@
 #include "processing.h"
 #include <gsl_util.h>
 #include <parallel.h>
-#include "AudioStreamSegment.h"
+#include "AudioSegment.h"
 
 using std::vector;
 using boost::adaptors::transformed;
@@ -16,7 +16,7 @@ using fmt::format;
 using std::runtime_error;
 using std::unique_ptr;
 
-BoundedTimeline<void> webRtcDetectVoiceActivity(AudioStream& audioStream, ProgressSink& progressSink) {
+BoundedTimeline<void> webRtcDetectVoiceActivity(const AudioClip& audioClip, ProgressSink& progressSink) {
 	VadInst* vadHandle = WebRtcVad_Create();
 	if (!vadHandle) throw runtime_error("Error creating WebRTC VAD handle.");
 
@@ -30,14 +30,14 @@ BoundedTimeline<void> webRtcDetectVoiceActivity(AudioStream& audioStream, Progre
 	if (error) throw runtime_error("Error setting WebRTC VAD aggressiveness.");
 
 	// Detect activity
-	BoundedTimeline<void> activity(audioStream.getTruncatedRange());
+	BoundedTimeline<void> activity(audioClip.getTruncatedRange());
 	centiseconds time = 0cs;
-	const size_t bufferCapacity = audioStream.getSampleRate() / 100;
+	const size_t bufferCapacity = audioClip.getSampleRate() / 100;
 	auto processBuffer = [&](const vector<int16_t>& buffer) {
 		// WebRTC is picky regarding buffer size
 		if (buffer.size() < bufferCapacity) return;
 
-		int result = WebRtcVad_Process(vadHandle, audioStream.getSampleRate(), buffer.data(), buffer.size()) == 1;
+		int result = WebRtcVad_Process(vadHandle, audioClip.getSampleRate(), buffer.data(), buffer.size()) == 1;
 		if (result == -1) throw runtime_error("Error processing audio buffer using WebRTC VAD.");
 
 		bool isActive = result != 0;
@@ -46,7 +46,7 @@ BoundedTimeline<void> webRtcDetectVoiceActivity(AudioStream& audioStream, Progre
 		}
 		time += 1cs;
 	};
-	process16bitAudioStream(*audioStream.clone(true), processBuffer, bufferCapacity, progressSink);
+	process16bitAudioClip(audioClip, processBuffer, bufferCapacity, progressSink);
 
 	// 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.
@@ -54,31 +54,31 @@ BoundedTimeline<void> webRtcDetectVoiceActivity(AudioStream& audioStream, Progre
 	if (!activity.empty()) {
 		TimeRange firstActivity = activity.begin()->getTimeRange();
 		activity.clear(firstActivity);
-		unique_ptr<AudioStream> streamStart = createSegment(audioStream.clone(true), TimeRange(0cs, firstActivity.getEnd()));
+		unique_ptr<AudioClip> streamStart = audioClip.clone() | segment(TimeRange(0cs, firstActivity.getEnd()));
 		time = 0cs;
-		process16bitAudioStream(*streamStart, processBuffer, bufferCapacity, progressSink);
+		process16bitAudioClip(*streamStart, processBuffer, bufferCapacity, progressSink);
 	}
 
 	return activity;
 }
 
-BoundedTimeline<void> detectVoiceActivity(std::unique_ptr<AudioStream> audioStream, ProgressSink& progressSink) {
+BoundedTimeline<void> detectVoiceActivity(const AudioClip& inputAudioClip, ProgressSink& progressSink) {
 	// Prepare audio for VAD
-	audioStream = removeDCOffset(convertSampleRate(std::move(audioStream), 16000));
+	const unique_ptr<AudioClip> audioClip = inputAudioClip.clone() | resample(16000) | removeDCOffset();
 
-	BoundedTimeline<void> activity(audioStream->getTruncatedRange());
+	BoundedTimeline<void> activity(audioClip->getTruncatedRange());
 	std::mutex activityMutex;
 
 	// Split audio into segments and perform parallel VAD
 	int segmentCount = getProcessorCoreCount();
-	centiseconds audioLength = audioStream->getTruncatedRange().getLength();
+	centiseconds audioLength = audioClip->getTruncatedRange().getLength();
 	vector<TimeRange> audioSegments;
 	for (int i = 0; i < segmentCount; ++i) {
 		TimeRange segmentRange = TimeRange(i * audioLength / segmentCount, (i + 1) * audioLength / segmentCount);
 		audioSegments.push_back(segmentRange);
 	}
 	runParallel([&](const TimeRange& segmentRange, ProgressSink& segmentProgressSink) {
-		unique_ptr<AudioStream> audioSegment = createSegment(audioStream->clone(false), segmentRange);
+		unique_ptr<AudioClip> audioSegment = audioClip->clone() | segment(segmentRange);
 		BoundedTimeline<void> activitySegment = webRtcDetectVoiceActivity(*audioSegment, segmentProgressSink);
 
 		std::lock_guard<std::mutex> lock(activityMutex);

src/audio/voiceActivityDetection.h

@@ -1,7 +1,6 @@
 #pragma once
-#include <memory>
-#include "AudioStream.h"
+#include "AudioClip.h"
 #include <BoundedTimeline.h>
 #include <ProgressBar.h>
 
-BoundedTimeline<void> detectVoiceActivity(std::unique_ptr<AudioStream> audioStream, ProgressSink& progressSink);
+BoundedTimeline<void> detectVoiceActivity(const AudioClip& audioClip, ProgressSink& progressSink);

src/audio/waveFileWriting.cpp

@@ -4,7 +4,7 @@
 
 using namespace little_endian;
 
-void createWaveFile(std::unique_ptr<AudioStream> inputStream, std::string fileName) {
+void createWaveFile(const AudioClip& audioClip, std::string fileName) {
 	// Open file
 	std::ofstream file;
 	file.exceptions(std::ofstream::failbit | std::ofstream::badbit);
@@ -15,7 +15,7 @@ void createWaveFile(std::unique_ptr<AudioStream> inputStream, std::string fileNa
 	uint32_t formatChunkSize = 16;
 	uint16_t channelCount = 1;
 	uint16_t frameSize = static_cast<uint16_t>(channelCount * sizeof(float));
-	uint32_t dataChunkSize = static_cast<uint32_t>(inputStream->getSampleCount() * frameSize);
+	uint32_t dataChunkSize = static_cast<uint32_t>(audioClip.size() * frameSize);
 	uint32_t riffChunkSize = 4 + (8 + formatChunkSize) + (8 + dataChunkSize);
 	write<uint32_t>(riffChunkSize, file);
 	write<uint32_t>(fourcc('W', 'A', 'V', 'E'), file);
@@ -26,7 +26,7 @@ void createWaveFile(std::unique_ptr<AudioStream> inputStream, std::string fileNa
 	uint16_t codec = 0x03; // 32-bit float
 	write<uint16_t>(codec, file);
 	write<uint16_t>(channelCount, file);
-	uint32_t frameRate = static_cast<uint16_t>(inputStream->getSampleRate());
+	uint32_t frameRate = static_cast<uint16_t>(audioClip.getSampleRate());
 	write<uint32_t>(frameRate, file);
 	uint32_t bytesPerSecond = frameRate * frameSize;
 	write<uint32_t>(bytesPerSecond, file);
@@ -37,8 +37,7 @@ void createWaveFile(std::unique_ptr<AudioStream> inputStream, std::string fileNa
 	// Write data chunk
 	write<uint32_t>(fourcc('d', 'a', 't', 'a'), file);
 	write<uint32_t>(dataChunkSize, file);
-	while (!inputStream->endOfStream()) {
-		float sample = inputStream->readSample();
+	for (float sample : audioClip) {
 		write<float>(sample, file);
 	}
 }

src/audio/waveFileWriting.h

@@ -1,7 +1,5 @@
 #pragma once
 
-#include <memory>
-#include <string>
-#include "AudioStream.h"
+#include "AudioClip.h"
 
-void createWaveFile(std::unique_ptr<AudioStream> inputStream, std::string fileName);
+void createWaveFile(const AudioClip& audioClip, std::string fileName);

src/main.cpp

@@ -15,6 +15,7 @@
 #include <boost/filesystem/operations.hpp>
 #include "stringTools.h"
 #include <boost/range/adaptor/transformed.hpp>
+#include <boost/filesystem/fstream.hpp>
 
 using std::exception;
 using std::string;
@@ -43,7 +44,7 @@ string getMessage(const exception& e) {
 	return result;
 }
 
-unique_ptr<AudioStream> createAudioStream(path filePath) {
+unique_ptr<AudioClip> createAudioClip(path filePath) {
 	try {
 		return std::make_unique<WaveFileReader>(filePath);
 	} catch (...) {
@@ -144,7 +145,7 @@ int main(int argc, char *argv[]) {
 		{
 			ProgressBar progressBar;
 			phones = detectPhones(
-				createAudioStream(inputFileName.getValue()),
+				*createAudioClip(inputFileName.getValue()),
 				dialogFile.isSet() ? readTextFile(path(dialogFile.getValue())) : boost::optional<u32string>(),
 				progressBar);
 		}

src/phoneExtraction.cpp

@@ -11,7 +11,7 @@
 #include <audio/DCOffset.h>
 #include <Timeline.h>
 #include <audio/voiceActivityDetection.h>
-#include <audio/AudioStreamSegment.h>
+#include "audio/AudioSegment.h"
 #include "languageModels.h"
 #include "tokenization.h"
 #include "g2p.h"
@@ -95,9 +95,9 @@ void sphinxLogCallback(void* user_data, err_lvl_t errorLevel, const char* format
 	logging::log(logLevel, message);
 }
 
-BoundedTimeline<string> recognizeWords(unique_ptr<AudioStream> audioStream, ps_decoder_t& decoder, bool& decoderIsStillUsable, ProgressSink& progressSink) {
+BoundedTimeline<string> recognizeWords(const AudioClip& inputAudioClip, ps_decoder_t& decoder, bool& decoderIsStillUsable, ProgressSink& progressSink) {
 	// Convert audio stream to the exact format PocketSphinx requires
-	audioStream = convertSampleRate(std::move(audioStream), sphinxSampleRate);
+	const unique_ptr<AudioClip> audioClip = inputAudioClip.clone() | resample(sphinxSampleRate);
 
 	// Restart timing at 0
 	ps_start_stream(&decoder);
@@ -111,7 +111,7 @@ BoundedTimeline<string> recognizeWords(unique_ptr<AudioStream> audioStream, ps_d
 		int searchedFrameCount = ps_process_raw(&decoder, buffer.data(), buffer.size(), false, false);
 		if (searchedFrameCount < 0) throw runtime_error("Error analyzing raw audio data for word recognition.");
 	};
-	process16bitAudioStream(*audioStream.get(), processBuffer, progressSink);
+	process16bitAudioClip(*audioClip, processBuffer, progressSink);
 
 	// End recognition
 	error = ps_end_utt(&decoder);
@@ -121,7 +121,7 @@ BoundedTimeline<string> recognizeWords(unique_ptr<AudioStream> audioStream, ps_d
 	// As a result, the following utterance will be garbage.
 	// As a workaround, we throw away the decoder in this case.
 	// See https://sourceforge.net/p/cmusphinx/discussion/help/thread/f1dd91c5/#7529
-	BoundedTimeline<string> result(audioStream->getTruncatedRange());
+	BoundedTimeline<string> result(audioClip->getTruncatedRange());
 	bool noWordsRecognized = reinterpret_cast<ngram_search_t*>(decoder.search)->bpidx == 0;
 	if (noWordsRecognized) {
 		decoderIsStillUsable = false;
@@ -147,7 +147,7 @@ s3wid_t getWordId(const string& word, dict_t& dictionary) {
 
 optional<Timeline<Phone>> getPhoneAlignment(
 	const vector<s3wid_t>& wordIds,
-	unique_ptr<AudioStream> audioStream,
+	const AudioClip& inputAudioClip,
 	ps_decoder_t& decoder,
 	ProgressSink& progressSink)
 {
@@ -164,7 +164,7 @@ optional<Timeline<Phone>> getPhoneAlignment(
 	if (error) throw runtime_error("Error populating alignment struct.");
 
 	// Convert audio stream to the exact format PocketSphinx requires
-	audioStream = convertSampleRate(std::move(audioStream), sphinxSampleRate);
+	const unique_ptr<AudioClip> audioClip = inputAudioClip.clone() | resample(sphinxSampleRate);
 
 	// Create search structure
 	acmod_t* acousticModel = decoder.acmod;
@@ -195,7 +195,7 @@ optional<Timeline<Phone>> getPhoneAlignment(
 				}
 			}
 		};
-		process16bitAudioStream(*audioStream.get(), processBuffer, progressSink);
+		process16bitAudioClip(*audioClip, processBuffer, progressSink);
 
 		// End search
 		error = ps_search_finish(search.get());
@@ -288,7 +288,7 @@ lambda_unique_ptr<ps_decoder_t> createDecoder(optional<u32string> dialog) {
 }
 
 Timeline<Phone> utteranceToPhones(
-	AudioStream& audioStream,
+	const AudioClip& audioClip,
 	TimeRange utterance,
 	ps_decoder_t& decoder,
 	bool& decoderIsStillUsable,
@@ -298,10 +298,10 @@ Timeline<Phone> utteranceToPhones(
 	ProgressSink& wordRecognitionProgressSink = utteranceProgressMerger.addSink(1.0);
 	ProgressSink& alignmentProgressSink = utteranceProgressMerger.addSink(0.5);
 
-	auto streamSegment = createSegment(audioStream.clone(true), utterance);
+	const unique_ptr<AudioClip> clipSegment = audioClip.clone() | segment(utterance);
 
 	// Get words
-	BoundedTimeline<string> words = recognizeWords(streamSegment->clone(true), decoder, decoderIsStillUsable, wordRecognitionProgressSink);
+	BoundedTimeline<string> words = recognizeWords(*clipSegment, decoder, decoderIsStillUsable, wordRecognitionProgressSink);
 	for (Timed<string> timedWord : words) {
 		timedWord.getTimeRange().shift(utterance.getStart());
 		logging::logTimedEvent("word", timedWord);
@@ -315,8 +315,8 @@ Timeline<Phone> utteranceToPhones(
 	if (wordIds.empty()) return Timeline<Phone>();
 
 	// Align the words' phones with speech
-	Timeline<Phone> segmentPhones = getPhoneAlignment(wordIds, std::move(streamSegment), decoder, alignmentProgressSink)
-		.value_or(ContinuousTimeline<Phone>(streamSegment->getTruncatedRange(), Phone::Unknown));
+	Timeline<Phone> segmentPhones = getPhoneAlignment(wordIds, *clipSegment, decoder, alignmentProgressSink)
+		.value_or(ContinuousTimeline<Phone>(clipSegment->getTruncatedRange(), Phone::Unknown));
 	segmentPhones.shift(utterance.getStart());
 	for (const auto& timedPhone : segmentPhones) {
 		logging::logTimedEvent("phone", timedPhone);
@@ -326,7 +326,7 @@ Timeline<Phone> utteranceToPhones(
 }
 
 BoundedTimeline<Phone> detectPhones(
-	unique_ptr<AudioStream> audioStream,
+	const AudioClip& inputAudioClip,
 	optional<u32string> dialog,
 	ProgressSink& progressSink)
 {
@@ -335,12 +335,12 @@ BoundedTimeline<Phone> detectPhones(
 	ProgressSink& dialogProgressSink = totalProgressMerger.addSink(15);
 
 	// Make sure audio stream has no DC offset
-	audioStream = removeDCOffset(std::move(audioStream));
+	const unique_ptr<AudioClip> audioClip = inputAudioClip.clone() | removeDCOffset();
 
 	// Split audio into utterances
 	BoundedTimeline<void> utterances;
 	try {
-		utterances = detectVoiceActivity(audioStream->clone(true), voiceActivationProgressSink);
+		utterances = detectVoiceActivity(*audioClip, voiceActivationProgressSink);
 	}
 	catch (...) {
 		std::throw_with_nested(runtime_error("Error detecting segments of speech."));
@@ -369,17 +369,16 @@ BoundedTimeline<Phone> detectPhones(
 		decoderPool.push(std::move(decoder));
 	};
 
-	BoundedTimeline<Phone> result(audioStream->getTruncatedRange());
+	BoundedTimeline<Phone> result(audioClip->getTruncatedRange());
 	std::mutex resultMutex;
 	auto processUtterance = [&](Timed<void> timedUtterance, ProgressSink& utteranceProgressSink) {
 		logging::logTimedEvent("utterance", timedUtterance.getTimeRange(), string(""));
 
 		// Detect phones for utterance
 		auto decoder = getDecoder();
-		auto audioStreamCopy = audioStream->clone(true);
 		bool decoderIsStillUsable = true;
 		Timeline<Phone> phones =
-			utteranceToPhones(*audioStreamCopy, timedUtterance.getTimeRange(), *decoder, decoderIsStillUsable, utteranceProgressSink);
+			utteranceToPhones(*audioClip, timedUtterance.getTimeRange(), *decoder, decoderIsStillUsable, utteranceProgressSink);
 		if (decoderIsStillUsable) {
 			returnDecoder(std::move(decoder));
 		}
@@ -404,7 +403,7 @@ BoundedTimeline<Phone> detectPhones(
 			// Don't use more threads than there are utterances to be processed
 			static_cast<int>(utterances.size()),
 			// Don't waste time creating additional threads (and decoders!) if the recording is short
-			static_cast<int>(duration_cast<std::chrono::seconds>(audioStream->getTruncatedRange().getLength()).count() / 10)
+			static_cast<int>(duration_cast<std::chrono::seconds>(audioClip->getTruncatedRange().getLength()).count() / 10)
 		});
 		logging::debug("Speech recognition -- start");
 		runParallel(processUtterance, utterances, threadCount, dialogProgressSink, getUtteranceProgressWeight);

src/phoneExtraction.h

@@ -1,12 +1,11 @@
 #pragma once
 
-#include <memory>
-#include "audio/AudioStream.h"
+#include "audio/AudioClip.h"
 #include "Phone.h"
 #include "progressBar.h"
 #include "BoundedTimeline.h"
 
 BoundedTimeline<Phone> detectPhones(
-	std::unique_ptr<AudioStream> audioStream,
+	const AudioClip& audioClip,
 	boost::optional<std::u32string> dialog,
 	ProgressSink& progressSink);