forked from dolphin-emu/dolphin
		
	the intent is to replace the haphazard scheduling and finger-crossing associated with saving/loading with the correct and minimal necessary wait for each thread to reach a known safe location before commencing the savestate operation, and for any already-paused components to not need to be resumed to do so.
		
			
				
	
	
		
			121 lines
		
	
	
		
			3.1 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			121 lines
		
	
	
		
			3.1 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
| // Copyright (C) 2003 Dolphin Project.
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| 
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| // This program is free software: you can redistribute it and/or modify
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| // it under the terms of the GNU General Public License as published by
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| // the Free Software Foundation, version 2.0.
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| 
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| // This program is distributed in the hope that it will be useful,
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| // but WITHOUT ANY WARRANTY; without even the implied warranty of
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| // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
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| // GNU General Public License 2.0 for more details.
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| 
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| // A copy of the GPL 2.0 should have been included with the program.
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| // If not, see http://www.gnu.org/licenses/
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| 
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| // Official SVN repository and contact information can be found at
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| // http://code.google.com/p/dolphin-emu/
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| 
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| #ifndef _MIXER_H_
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| #define _MIXER_H_
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| 
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| #include "WaveFile.h"
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| #include "StdMutex.h"
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| 
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| // 16 bit Stereo
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| #define MAX_SAMPLES			(1024 * 8)
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| #define INDEX_MASK			(MAX_SAMPLES * 2 - 1)
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| #define RESERVED_SAMPLES	(256)
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| 
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| class CMixer {
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| 	
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| public:
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| 	CMixer(unsigned int AISampleRate = 48000, unsigned int DACSampleRate = 48000, unsigned int BackendSampleRate = 32000)
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| 		: m_aiSampleRate(AISampleRate)
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| 		, m_dacSampleRate(DACSampleRate)
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| 		, m_bits(16)
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| 		, m_channels(2)
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| 		, m_HLEready(false)
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| 		, m_logAudio(0)
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| 		, m_numSamples(0)
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| 		, m_indexW(0)
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| 		, m_indexR(0)
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| 		, m_AIplaying(true)
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| 	{
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| 		// AyuanX: The internal (Core & DSP) sample rate is fixed at 32KHz
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| 		// So when AI/DAC sample rate differs than 32KHz, we have to do re-sampling
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| 		m_sampleRate = BackendSampleRate;
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| 
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| 		memset(m_buffer, 0, sizeof(m_buffer));
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| 
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| 		INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized (AISampleRate:%i, DACSampleRate:%i)", AISampleRate, DACSampleRate);
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| 	}
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| 
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| 	virtual ~CMixer() {}
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| 
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| 	// Called from audio threads
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| 	virtual unsigned int Mix(short* samples, unsigned int numSamples);
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| 	virtual void Premix(short * /*samples*/, unsigned int /*numSamples*/) {}
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| 	unsigned int GetNumSamples();
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| 
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| 	// Called from main thread
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| 	virtual void PushSamples(const short* samples, unsigned int num_samples);
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| 	unsigned int GetSampleRate() {return m_sampleRate;}
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| 
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| 	void SetThrottle(bool use) { m_throttle = use;}
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| 
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| 	// TODO: do we need this
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| 	bool IsHLEReady() { return m_HLEready;}
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| 	void SetHLEReady(bool ready) { m_HLEready = ready;}
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| 	// ---------------------
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| 
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| 
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| 	virtual void StartLogAudio(const char *filename) {
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| 		if (! m_logAudio) {
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| 			m_logAudio = true;
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| 			g_wave_writer.Start(filename, GetSampleRate());
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| 			g_wave_writer.SetSkipSilence(false);
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| 			NOTICE_LOG(DSPHLE, "Starting Audio logging");
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| 		} else {
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| 			WARN_LOG(DSPHLE, "Audio logging already started");
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| 		}
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| 	}
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| 
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| 	virtual void StopLogAudio() {
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| 		if (m_logAudio) {
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| 			m_logAudio = false;
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| 			g_wave_writer.Stop();
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| 			NOTICE_LOG(DSPHLE, "Stopping Audio logging");
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| 		} else {
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| 			WARN_LOG(DSPHLE, "Audio logging already stopped");
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| 		}
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| 	}
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| 
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| 	std::mutex& MixerCritical() { return m_csMixing; }
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| 
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| protected:
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| 	unsigned int m_sampleRate;
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| 	unsigned int m_aiSampleRate;
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| 	unsigned int m_dacSampleRate;
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| 	int m_bits;
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| 	int m_channels;
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| 
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| 	WaveFileWriter g_wave_writer;
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| 	
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| 	bool m_HLEready;
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| 	bool m_logAudio;
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| 
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| 	bool m_throttle;
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| 
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| 	short m_buffer[MAX_SAMPLES * 2];
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| 	volatile u32 m_numSamples;
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| 	u32 m_indexW;
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| 	u32 m_indexR;
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| 
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| 	bool m_AIplaying;
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| 	std::mutex m_csMixing;
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| private:
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| 
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| };
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| 
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| #endif // _MIXER_H_
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