Remove the FIFO thread

include/AudioAlsa.h

@@ -82,8 +82,8 @@ class AudioAlsa : public QThread, public AudioDevice
 	static DeviceInfoCollection getAvailableDevices();
 
 private:
-	void startProcessing() override;
-	void stopProcessing() override;
+	void startProcessingImpl() override;
+	void stopProcessingImpl() override;
 	void run() override;
 
 	int setHWParams( const ch_cnt_t _channels, snd_pcm_access_t _access );
@@ -98,9 +98,6 @@ class AudioAlsa : public QThread, public AudioDevice
 
 	snd_pcm_hw_params_t * m_hwParams;
 	snd_pcm_sw_params_t * m_swParams;
-
-	bool m_convertEndian;
-
 } ;
 
 } // namespace lmms

include/AudioBufferView.h

@@ -361,6 +361,12 @@ class InterleavedBufferView : public detail::BufferViewData<T, channelCount>
 		}
 	}
 
+	//! @return the sample at the given channel and frame position
+	constexpr auto sample(proc_ch_t channel, f_cnt_t frame) noexcept -> T&
+	{
+		return framePtr(frame)[channel];
+	}
+
 	/**
 	 * @return pointer to the frame at the given index.
 	 * The size of the frame is `channels()`.
@@ -501,6 +507,12 @@ class PlanarBufferView : public detail::BufferViewData<T* const, channelCount>
 		return !this->m_data || Base::channels() == 0 || this->m_frames == 0;
 	}
 
+	//! @return the sample at the given channel and frame position
+	constexpr auto sample(proc_ch_t channel, f_cnt_t frame) noexcept -> T&
+	{
+		return bufferPtr(channel)[frame];
+	}
+
 	//! @return the buffer of the given channel
 	constexpr auto buffer(proc_ch_t channel) const noexcept -> std::span<T>
 	{

include/AudioDevice.h

@@ -28,6 +28,7 @@
 #include <QMutex>
 #include <samplerate.h>
 
+#include "AudioBufferView.h"
 #include "LmmsTypes.h"
 
 class QThread;
@@ -75,66 +76,43 @@ class AudioDevice
 		return m_sampleRate;
 	}
 
-	void processNextBuffer();
+	void startProcessing();
 
-	virtual void startProcessing()
-	{
-		m_inProcess = true;
-	}
+	void stopProcessing();
 
-	virtual void stopProcessing();
+	bool isRunning() const { return m_running.test(std::memory_order_acquire); }
 
 protected:
-	// subclasses can re-implement this for being used in conjunction with
-	// processNextBuffer()
-	virtual void writeBuffer(const SampleFrame* /* _buf*/, const fpp_t /*_frames*/) {}
-
-	// called by according driver for fetching new sound-data
-	fpp_t getNextBuffer(SampleFrame* _ab);
-
 	// convert a given audio-buffer to a buffer in signed 16-bit samples
 	// returns num of bytes in outbuf
-	int convertToS16(const SampleFrame* _ab,
-						const fpp_t _frames,
-						int_sample_t * _output_buffer,
-						const bool _convert_endian = false );
+	int convertToS16(
+		const SampleFrame* _ab, const fpp_t _frames, int_sample_t* _output_buffer, const bool _convert_endian = false);
 
 	// clear given signed-int-16-buffer
-	void clearS16Buffer( int_sample_t * _outbuf,
-							const fpp_t _frames );
+	void clearS16Buffer(int_sample_t* _outbuf, const fpp_t _frames);
 
-	ch_cnt_t channels() const
-	{
-		return m_channels;
-	}
+	ch_cnt_t channels() const { return m_channels; }
 
-	inline void setSampleRate( const sample_rate_t _new_sr )
-	{
-		m_sampleRate = _new_sr;
-	}
+	AudioEngine* audioEngine() { return m_audioEngine; }
 
-	AudioEngine* audioEngine()
-	{
-		return m_audioEngine;
-	}
+	void setSampleRate(const sample_rate_t _new_sr) { m_sampleRate = _new_sr; }
 
 	static void stopProcessingThread( QThread * thread );
-
-
 protected:
 	bool m_supportsCapture;
 
-
 private:
+	virtual void startProcessingImpl() = 0;
+	virtual void stopProcessingImpl() = 0;
+
 	sample_rate_t m_sampleRate;
 	ch_cnt_t m_channels;
-	AudioEngine* m_audioEngine;
-	bool m_inProcess;
 
-	QMutex m_devMutex;
+	AudioEngine* m_audioEngine = nullptr;
 
-	SampleFrame* m_buffer;
+	QMutex m_devMutex;
 
+	std::atomic_flag m_running = ATOMIC_FLAG_INIT;
 };
 
 } // namespace lmms

include/AudioDummy.h

@@ -78,31 +78,23 @@ class AudioDummy : public QThread, public AudioDevice
 
 
 private:
-	void startProcessing() override
+	void startProcessingImpl() override
 	{
 		start();
 	}
 
-	void stopProcessing() override
+	void stopProcessingImpl() override
 	{
 		stopProcessingThread( this );
 	}
 
 	void run() override
 	{
 		MicroTimer timer;
-		while( true )
+		while (AudioDevice::isRunning())
 		{
 			timer.reset();
-			const SampleFrame* b = audioEngine()->nextBuffer();
-			if( !b )
-			{
-				break;
-			}
-			if( audioEngine()->hasFifoWriter() )
-			{
-				delete[] b;
-			}
+			audioEngine()->renderNextPeriod();
 
 			const int microseconds = static_cast<int>( audioEngine()->framesPerPeriod() * 1000000.0f / audioEngine()->outputSampleRate() - timer.elapsed() );
 			if( microseconds > 0 )

include/AudioEngine.h

@@ -37,7 +37,6 @@
 #include "LmmsTypes.h"
 #include "SampleFrame.h"
 #include "LocklessList.h"
-#include "FifoBuffer.h"
 #include "AudioEngineProfiler.h"
 #include "PlayHandle.h"
 
@@ -59,6 +58,8 @@ constexpr int BYTES_PER_FRAME = sizeof(SampleFrame);
 
 constexpr float OUTPUT_SAMPLE_MULTIPLIER = 32767.0f;
 
+constexpr sample_rate_t DEFAULT_SAMPLE_RATE = 44100;
+
 constexpr auto SUPPORTED_SAMPLERATES = std::array{44100, 48000, 88200, 96000, 192000}; 
 
 class LMMS_EXPORT AudioEngine : public QObject
@@ -126,7 +127,7 @@ class LMMS_EXPORT AudioEngine : public QObject
 
 	//! Set new audio device. Old device will be deleted,
 	//! unless it's stored using storeAudioDevice
-	void setAudioDevice(AudioDevice* _dev, bool _needs_fifo, bool startNow);
+	void setAudioDevice(AudioDevice* _dev, bool startNow);
 	void storeAudioDevice();
 	void restoreAudioDevice();
 	inline AudioDevice * audioDev()
@@ -170,13 +171,19 @@ class LMMS_EXPORT AudioEngine : public QObject
 
 	void removePlayHandlesOfTypes(Track * track, PlayHandle::Types types);
 
-
-	// methods providing information for other classes
-	inline fpp_t framesPerPeriod() const
+	//! @return the number of frames rendered per period
+	//! @note this represents the size of the buffer for each call to @ref renderNextBuffer
+	fpp_t framesPerPeriod() const
 	{
 		return m_framesPerPeriod;
 	}
 
+	//! @returns the number of audio frames per audio buffer
+	//! @note this represents the size of the underlying device buffer
+	fpp_t framesPerAudioBuffer() const
+	{
+		return m_framesPerAudioBuffer;
+	}
 
 	AudioEngineProfiler& profiler()
 	{
@@ -235,11 +242,6 @@ class LMMS_EXPORT AudioEngine : public QObject
 
 	bool criticalXRuns() const;
 
-	inline bool hasFifoWriter() const
-	{
-		return m_fifoWriter != nullptr;
-	}
-
 	void pushInputFrames( SampleFrame* _ab, const f_cnt_t _frames );
 
 	inline const SampleFrame* inputBuffer()
@@ -252,10 +254,16 @@ class LMMS_EXPORT AudioEngine : public QObject
 		return m_inputBufferFrames[ m_inputBufferRead ];
 	}
 
-	inline const SampleFrame* nextBuffer()
-	{
-		return hasFifoWriter() ? m_fifo->read() : renderNextBuffer();
-	}
+	//! Renders the next period of the audio buffer
+	//! The audio buffer can contain many periods, depending on the chosen buffer size
+	//! @returns The next audio period
+	const SampleFrame* renderNextPeriod();
+
+	//! Renders the stereo, interleaved audio buffer into @a dst
+	//! @note @a dst is expected to have a frame count of `framesPerAudioBuffer()`
+	//! @note if @a dst is mono, the buffer is converted to mono
+	//! @note if @a dst is non stereo, the other channels are zero-filled
+	void renderNextBuffer(AudioBufferView<float> auto dst);
 
 	//! Block until a change in model can be done (i.e. wait for audio thread)
 	void requestChangeInModel();
@@ -277,32 +285,11 @@ class LMMS_EXPORT AudioEngine : public QObject
 
 
 private:
-	using Fifo = FifoBuffer<SampleFrame*>;
-
-	class fifoWriter : public QThread
-	{
-	public:
-		fifoWriter( AudioEngine * audioEngine, Fifo * fifo );
-
-		void finish();
-
-
-	private:
-		AudioEngine * m_audioEngine;
-		Fifo * m_fifo;
-		volatile bool m_writing;
-
-		void run() override;
-
-		void write(SampleFrame* buffer);
-	} ;
-
-
 	AudioEngine( bool renderOnly );
 	~AudioEngine() override;
 
-	void startProcessing(bool needsFifo = true);
-	void stopProcessing();
+	void startProcessing() { m_audioDev->startProcessing(); }
+	void stopProcessing() { m_audioDev->stopProcessing(); }
 
 
 	AudioDevice * tryAudioDevices();
@@ -313,7 +300,6 @@ class LMMS_EXPORT AudioEngine : public QObject
 	void renderStageEffects();
 	void renderStageMix();
 
-	const SampleFrame* renderNextBuffer();
 
 	void swapBuffers();
 
@@ -323,12 +309,13 @@ class LMMS_EXPORT AudioEngine : public QObject
 
 	std::vector<AudioBusHandle*> m_audioBusHandles;
 
+	fpp_t m_framesPerAudioBuffer;
 	fpp_t m_framesPerPeriod;
+	sample_rate_t m_baseSampleRate;
 
 	SampleFrame* m_inputBuffer[2];
 	f_cnt_t m_inputBufferFrames[2];
 	f_cnt_t m_inputBufferSize[2];
-	sample_rate_t m_baseSampleRate;
 	int m_inputBufferRead;
 	int m_inputBufferWrite;
 
@@ -358,10 +345,6 @@ class LMMS_EXPORT AudioEngine : public QObject
 	MidiClient * m_midiClient;
 	QString m_midiClientName;
 
-	// FIFO stuff
-	Fifo * m_fifo;
-	fifoWriter * m_fifoWriter;
-
 	AudioEngineProfiler m_profiler;
 
 	bool m_clearSignal;

include/AudioFileDevice.h

@@ -49,6 +49,8 @@ class AudioFileDevice : public AudioDevice
 
 	OutputSettings const & getOutputSettings() const { return m_outputSettings; }
 
+	//! Write `size` sample frames from `buf` into the output file.
+	virtual void writeBuffer(const SampleFrame* buf, std::size_t size) = 0;
 
 protected:
 	int writeData( const void* data, int len );
@@ -64,6 +66,9 @@ class AudioFileDevice : public AudioDevice
 	}
 
 private:
+	void startProcessingImpl() override {}
+	void stopProcessingImpl() override {}
+
 	QFile m_outputFile;
 	OutputSettings m_outputSettings;
 } ;

include/AudioFileMP3.h

@@ -57,10 +57,8 @@ class AudioFileMP3 : public AudioFileDevice
 					 outputFilename, audioEngine );
 	}
 
-protected:
-	void writeBuffer(const SampleFrame* /* _buf*/, const fpp_t /*_frames*/) override;
-
 private:
+	void writeBuffer(const SampleFrame* /* _buf*/, const fpp_t /*_frames*/) override;
 	void flushRemainingBuffers();
 	bool initEncoder();
 	void tearDownEncoder();

include/AudioJack.h

@@ -103,13 +103,13 @@ private slots:
 	bool initJackClient();
 	void resizeInputBuffer(jack_nframes_t nframes);
 
+	void startProcessingImpl() override;
+	void stopProcessingImpl() override;
+
 	void attemptToConnect(size_t index, const char *lmms_port_type, const char *source_port, const char *destination_port);
 	void attemptToReconnectOutput(size_t outputIndex, const QString& targetPort);
 	void attemptToReconnectInput(size_t inputIndex, const QString& sourcePort);
 
-	void startProcessing() override;
-	void stopProcessing() override;
-
 	void registerPort(AudioBusHandle* port) override;
 	void unregisterPort(AudioBusHandle* port) override;
 	void renamePort(AudioBusHandle* port) override;
@@ -122,14 +122,12 @@ private slots:
 	jack_client_t* m_client;
 
 	bool m_active;
-	std::atomic<bool> m_stopped;
 
 	std::atomic<MidiJack*> m_midiClient;
 	std::vector<jack_port_t*> m_outputPorts;
 	std::vector<jack_port_t*> m_inputPorts;
 	jack_default_audio_sample_t** m_tempOutBufs;
 	std::vector<SampleFrame> m_inputFrameBuffer;
-	SampleFrame* m_outBuf;
 
 	f_cnt_t m_framesDoneInCurBuf;
 	f_cnt_t m_framesToDoInCurBuf;