Fix audio resampling functionality (#7858)

Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>

include/AudioEngine.h

@@ -105,40 +105,6 @@ public:
 		AudioEngine* m_audioEngine;
 	};
 
-	struct qualitySettings
-	{
-		enum class Interpolation
-		{
-			Linear,
-			SincFastest,
-			SincMedium,
-			SincBest
-		} ;
-
-		Interpolation interpolation;
-
-		qualitySettings(Interpolation i) :
-			interpolation(i)
-		{
-		}
-
-		int libsrcInterpolation() const
-		{
-			switch( interpolation )
-			{
-				case Interpolation::Linear:
-					return SRC_ZERO_ORDER_HOLD;
-				case Interpolation::SincFastest:
-					return SRC_SINC_FASTEST;
-				case Interpolation::SincMedium:
-					return SRC_SINC_MEDIUM_QUALITY;
-				case Interpolation::SincBest:
-					return SRC_SINC_BEST_QUALITY;
-			}
-			return SRC_LINEAR;
-		}
-	} ;
-
 	void initDevices();
 	void clear();
 	void clearNewPlayHandles();
@@ -160,10 +126,7 @@ public:
 
 	//! Set new audio device. Old device will be deleted,
 	//! unless it's stored using storeAudioDevice
-	void setAudioDevice( AudioDevice * _dev,
-				const struct qualitySettings & _qs,
-				bool _needs_fifo,
-				bool startNow );
+	void setAudioDevice(AudioDevice* _dev, bool _needs_fifo, bool startNow);
 	void storeAudioDevice();
 	void restoreAudioDevice();
 	inline AudioDevice * audioDev()
@@ -230,12 +193,6 @@ public:
 		return m_profiler.detailLoad(type);
 	}
 
-	const qualitySettings & currentQualitySettings() const
-	{
-		return m_qualitySettings;
-	}
-
-
 	sample_rate_t baseSampleRate() const { return m_baseSampleRate; }
 
 
@@ -300,8 +257,6 @@ public:
 		return hasFifoWriter() ? m_fifo->read() : renderNextBuffer();
 	}
 
-	void changeQuality(const struct qualitySettings & qs);
-
 	//! Block until a change in model can be done (i.e. wait for audio thread)
 	void requestChangeInModel();
 	void doneChangeInModel();
@@ -390,8 +345,6 @@ private:
 	LocklessList<PlayHandle *> m_newPlayHandles;
 	ConstPlayHandleList m_playHandlesToRemove;
 
-
-	struct qualitySettings m_qualitySettings;
 	float m_masterGain;
 
 	// audio device stuff

include/AudioResampler.h

@@ -1,7 +1,7 @@
 /*
- * AudioResampler.h - wrapper around libsamplerate
+ * AudioResampler.h
  *
- * Copyright (c) 2023 saker <sakertooth@gmail.com>
+ * Copyright (c) 2025 saker <sakertooth@gmail.com>
  *
  * This file is part of LMMS - https://lmms.io
  *
@@ -25,41 +25,104 @@
 #ifndef LMMS_AUDIO_RESAMPLER_H
 #define LMMS_AUDIO_RESAMPLER_H
 
-#include <samplerate.h>
-
+#include <memory>
+#include "AudioBufferView.h"
 #include "lmms_export.h"
 
 namespace lmms {
 
+/**
+ * @class AudioResampler
+ * @brief A utility class for resampling interleaved audio buffers using various resampling algorithms.
+ *
+ * This class provides support for zero-order hold, linear, and several levels of sinc-based resampling.
+ */
 class LMMS_EXPORT AudioResampler
 {
 public:
-	struct ProcessResult
+	/**
+	 * @enum Mode
+	 * @brief Defines the resampling method to use.
+	 */
+	enum class Mode
 	{
-		int error;
-		long inputFramesUsed;
-		long outputFramesGenerated;
+		ZOH,		 //!< Zero Order Hold (nearest-neighbor) interpolation.
+		Linear,		 //!< Linear interpolation.
+		SincFastest, //!< Fastest sinc-based resampling.
+		SincMedium,	 //!< Medium quality sinc-based resampling.
+		SincBest	 //!< Highest quality sinc-based resampling.
 	};
 
-	AudioResampler(int interpolationMode, int channels);
-	AudioResampler(const AudioResampler&) = delete;
-	AudioResampler(AudioResampler&&) = delete;
-	~AudioResampler();
+	/**
+	 * @struct Result
+	 * @brief Result of a resampling operation.
+	 */
+	struct Result
+	{
+		f_cnt_t inputFramesUsed;	   //!< The number of input frames used during processing.
+		f_cnt_t outputFramesGenerated; //!< The number of output frames generated during processing.
+	};
 
-	AudioResampler& operator=(const AudioResampler&) = delete;
-	AudioResampler& operator=(AudioResampler&&) = delete;
+	/**
+	 * @brief Constructs an `AudioResampler` instance.
+	 * @param mode The resampling mode to use.
+	 * @param channels Number of audio channels. Defaults to `2` (stereo).
+	 */
+	AudioResampler(Mode mode, ch_cnt_t channels = 2);
 
-	auto resample(const float* in, long inputFrames, float* out, long outputFrames, double ratio) -> ProcessResult;
-	auto interpolationMode() const -> int { return m_interpolationMode; }
-	auto channels() const -> int { return m_channels; }
-	void setRatio(double ratio);
+	/**
+	 * @brief Process a block of interleaved audio input from `input` and resample it into `output`.
+	 *
+	 * @param input The interleaved audio input.
+	 * @param output The interleaved audio output.
+	 *
+	 * @throws `std::invalid_argument` if a channel mismatch has been detected.
+	 * @throws `std::runtime_error` if the resampling process has failed.
+	 *
+	 * @remark This utility class does not cache the input and output buffers, making it stateless. In other words,
+	 * `input` is directly resampled into the `output`.
+	 *
+	 * @returns the result of the resampling process. See @ref Result for more details.
+	 */
+	[[nodiscard]] auto process(InterleavedBufferView<const float> input, InterleavedBufferView<float> output) -> Result;
+
+	/**
+	 * @brief Resets the internal resampler state.
+	 * Useful when working with unreleated pieces of audio.
+	 */
+	void reset();
+
+	/**
+	 * @brief Sets the resampling ratio to `ratio`.
+	 * @param ratio Output sample rate divided by input sample rate.
+	 */
+	void setRatio(double ratio) { m_ratio = ratio; }
+
+	/**
+	 * @brief Sets the resampling ratio to `output / input`.
+	 * @param input Input sample rate.
+	 * @param output Output sample rate.
+	 */
+	void setRatio(sample_rate_t input, sample_rate_t output) { m_ratio = static_cast<double>(output) / input; }
+
+	//! @returns the resampling ratio.
+	auto ratio() const -> double { return m_ratio; }
+
+	//! @returns the number of channels expected by the resampler.
+	auto channels() const -> ch_cnt_t { return m_channels; }
+
+	//! @returns the interpolation mode used by this resampler.
+	auto mode() const -> Mode { return m_mode; }
 
 private:
-	int m_interpolationMode = -1;
-	int m_channels = 0;
+	struct LMMS_EXPORT StateDeleter { void operator()(void* state); };
+	std::unique_ptr<void, StateDeleter> m_state;
+	Mode m_mode;
+	ch_cnt_t m_channels = 0;
+	double m_ratio = 1.0;
 	int m_error = 0;
-	SRC_STATE* m_state = nullptr;
 };
+
 } // namespace lmms
 
 #endif // LMMS_AUDIO_RESAMPLER_H

include/Effect.h

@@ -55,7 +55,6 @@ public:
 	Effect( const Plugin::Descriptor * _desc,
 			Model * _parent,
 			const Descriptor::SubPluginFeatures::Key * _key );
-	~Effect() override;
 
 	void saveSettings( QDomDocument & _doc, QDomElement & _parent ) override;
 	void loadSettings( const QDomElement & _this ) override;
@@ -176,29 +175,6 @@ protected:
 
 	gui::PluginView* instantiateView( QWidget * ) override;
 
-	// some effects might not be capable of higher sample-rates so they can
-	// sample it down before processing and back after processing
-	inline void sampleDown( const SampleFrame* _src_buf,
-							SampleFrame* _dst_buf,
-							sample_rate_t _dst_sr )
-	{
-		resample( 0, _src_buf,
-				Engine::audioEngine()->outputSampleRate(),
-					_dst_buf, _dst_sr,
-					Engine::audioEngine()->framesPerPeriod() );
-	}
-
-	inline void sampleBack( const SampleFrame* _src_buf,
-							SampleFrame* _dst_buf,
-							sample_rate_t _src_sr )
-	{
-		resample( 1, _src_buf, _src_sr, _dst_buf,
-				Engine::audioEngine()->outputSampleRate(),
-			Engine::audioEngine()->framesPerPeriod() * _src_sr /
-				Engine::audioEngine()->outputSampleRate() );
-	}
-	void reinitSRC();
-
 	virtual void onEnabledChanged() {}
 
 
@@ -212,10 +188,6 @@ private:
 
 
 	EffectChain * m_parent;
-	void resample( int _i, const SampleFrame* _src_buf,
-					sample_rate_t _src_sr,
-					SampleFrame* _dst_buf, sample_rate_t _dst_sr,
-					const f_cnt_t _frames );
 
 	bool m_okay;
 	bool m_noRun;
@@ -230,10 +202,6 @@ private:
 
 	bool m_autoQuitEnabled = false;
 
-	SRC_DATA m_srcData[2];
-	SRC_STATE * m_srcState[2];
-
-
 	friend class gui::EffectView;
 	friend class EffectChain;
 

include/ProjectRenderer.h

@@ -59,11 +59,7 @@ public:
 		AudioFileDeviceInstantiaton m_getDevInst;
 	} ;
 
-
-	ProjectRenderer( const AudioEngine::qualitySettings & _qs,
-				const OutputSettings & _os,
-				ExportFileFormat _file_format,
-				const QString & _out_file );
+	ProjectRenderer(const OutputSettings& _os, ExportFileFormat _file_format, const QString& _out_file);
 	~ProjectRenderer() override = default;
 
 	bool isReady() const
@@ -93,7 +89,6 @@ private:
 	void run() override;
 
 	AudioFileDevice * m_fileDev;
-	AudioEngine::qualitySettings m_qualitySettings;
 
 	volatile int m_progress;
 	volatile bool m_abort;

include/RenderManager.h

@@ -40,11 +40,7 @@ class RenderManager : public QObject
 {
 	Q_OBJECT
 public:
-	RenderManager(
-		const AudioEngine::qualitySettings & qualitySettings,
-		const OutputSettings & outputSettings,
-		ProjectRenderer::ExportFileFormat fmt,
-		QString outputPath);
+	RenderManager(const OutputSettings& outputSettings, ProjectRenderer::ExportFileFormat fmt, QString outputPath);
 
 	~RenderManager() override;
 
@@ -70,8 +66,6 @@ private:
 
 	void render( QString outputPath );
 
-	const AudioEngine::qualitySettings m_qualitySettings;
-	const AudioEngine::qualitySettings m_oldQualitySettings;
 	const OutputSettings m_outputSettings;
 	ProjectRenderer::ExportFileFormat m_format;
 	QString m_outputPath;

include/Sample.h

@@ -1,7 +1,7 @@
 /*
- * Sample.h - State for container-class SampleBuffer
+ * Sample.h
  *
- * Copyright (c) 2023 saker <sakertooth@gmail.com>
+ * Copyright (c) 2025 saker <sakertooth@gmail.com>
  *
  * This file is part of LMMS - https://lmms.io
  *
@@ -36,12 +36,6 @@ namespace lmms {
 class LMMS_EXPORT Sample
 {
 public:
-	// values for buffer margins, used for various libsamplerate interpolation modes
-	// the array positions correspond to the converter_type parameter values in libsamplerate
-	// if there appears problems with playback on some interpolation mode, then the value for that mode
-	// may need to be higher - conversely, to optimize, some may work with lower values
-	static constexpr auto s_interpolationMargins = std::array<int, 5>{64, 64, 64, 4, 4};
-
 	enum class Loop
 	{
 		Off,
@@ -52,42 +46,41 @@ public:
 	class LMMS_EXPORT PlaybackState
 	{
 	public:
-		PlaybackState(bool varyingPitch = false, int interpolationMode = SRC_LINEAR)
-			: m_resampler(interpolationMode, DEFAULT_CHANNELS)
-			, m_varyingPitch(varyingPitch)
+		PlaybackState(AudioResampler::Mode interpolationMode = AudioResampler::Mode::Linear, int frameIndex = 0)
+			: m_resampler(interpolationMode)
+			, m_frameIndex(frameIndex)
 		{
 		}
 
-		auto resampler() -> AudioResampler& { return m_resampler; }
 		auto frameIndex() const -> int { return m_frameIndex; }
-		auto varyingPitch() const -> bool { return m_varyingPitch; }
 		auto backwards() const -> bool { return m_backwards; }
 
-		void setFrameIndex(int frameIndex) { m_frameIndex = frameIndex; }
-		void setVaryingPitch(bool varyingPitch) { m_varyingPitch = varyingPitch; }
+		void setFrameIndex(int index) { m_frameIndex = index; }
 		void setBackwards(bool backwards) { m_backwards = backwards; }
 
 	private:
 		AudioResampler m_resampler;
+		std::array<SampleFrame, DEFAULT_BUFFER_SIZE> m_buffer;
+		std::span<SampleFrame> m_bufferView;
 		int m_frameIndex = 0;
-		bool m_varyingPitch = false;
 		bool m_backwards = false;
 		friend class Sample;
 	};
 
 	Sample() = default;
+
 	Sample(const QByteArray& base64, int sampleRate = Engine::audioEngine()->outputSampleRate());
 	Sample(const SampleFrame* data, size_t numFrames, int sampleRate = Engine::audioEngine()->outputSampleRate());
 	Sample(const Sample& other);
-	Sample(Sample&& other);
+	Sample(Sample&& other) noexcept;
 	explicit Sample(const QString& audioFile);
 	explicit Sample(std::shared_ptr<const SampleBuffer> buffer);
 
 	auto operator=(const Sample&) -> Sample&;
-	auto operator=(Sample&&) -> Sample&;
+	auto operator=(Sample&&) noexcept -> Sample&;
 
-	auto play(SampleFrame* dst, PlaybackState* state, size_t numFrames, float desiredFrequency = DefaultBaseFreq,
-		Loop loopMode = Loop::Off) const -> bool;
+	auto play(SampleFrame* dst, PlaybackState* state, size_t numFrames, Loop loopMode = Loop::Off,
+		double ratio = 1.0) const -> bool;
 
 	auto sampleDuration() const -> std::chrono::milliseconds;
 	auto sampleFile() const -> const QString& { return m_buffer->audioFile(); }
@@ -116,10 +109,7 @@ public:
 	void setReversed(bool reversed) { m_reversed.store(reversed, std::memory_order_relaxed); }
 
 private:
-	void playRaw(SampleFrame* dst, size_t numFrames, const PlaybackState* state, Loop loopMode) const;
-	void advance(PlaybackState* state, size_t advanceAmount, Loop loopMode) const;
-
-private:
+	f_cnt_t render(SampleFrame* dst, f_cnt_t size, PlaybackState* state, Loop loop) const;
 	std::shared_ptr<const SampleBuffer> m_buffer = SampleBuffer::emptyBuffer();
 	std::atomic<int> m_startFrame = 0;
 	std::atomic<int> m_endFrame = 0;