mirror of
https://github.com/dolphin-emu/dolphin.git
synced 2025-07-29 09:09:52 -06:00
This is a joined work of XK and me on improving the HLE plugin interface.
It allows run time selection of backends (AOSound, DSound and NullSound). It replaces the DSP_NULL plugin (works even better!) It also includes improved thread handling on asound, and using some common functions on both asound and windows. git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@2027 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
@ -1,4 +1,4 @@
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// Copyright (C) 2003-2008 Dolphin Project.
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// Copyright (C) 2003-2009 Dolphin Project.
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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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@ -15,78 +15,74 @@
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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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#include <ao/ao.h>
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#include <pthread.h>
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#include <string.h>
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#include "AOSoundStream.h"
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namespace AOSound
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#if defined(HAVE_AO) && HAVE_AO
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void AOSound::SoundLoop()
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{
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pthread_t thread;
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StreamCallback callback;
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int buf_size;
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ao_device *device;
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ao_sample_format format;
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int default_driver;
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int sampleRate;
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volatile int threadData;
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short realtimeBuffer[1024 * 1024];
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int AOSound_GetSampleRate()
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{
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return sampleRate;
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}
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void *soundThread(void *)
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{
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ao_initialize();
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default_driver = ao_default_driver_id();
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format.bits = 16;
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format.channels = 2;
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format.rate = sampleRate;
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format.byte_format = AO_FMT_LITTLE;
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ao_initialize();
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default_driver = ao_default_driver_id();
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format.bits = 16;
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format.channels = 2;
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format.rate = sampleRate;
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format.byte_format = AO_FMT_LITTLE;
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device = ao_open_live(default_driver, &format, NULL /* no options */);
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if (device == NULL)
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{
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fprintf(stderr, "DSP_HLE: Error opening AO device.\n");
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return false;
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}
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buf_size = format.bits/8 * format.channels * format.rate;
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device = ao_open_live(default_driver, &format, NULL /* no options */);
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if (device == NULL) {
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PanicAlert("DSP_HLE: Error opening AO device.\n");
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ao_shutdown();
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Stop();
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return;
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}
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while (!threadData)
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{
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uint_32 numBytesToRender = 256;
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(*callback)(realtimeBuffer, numBytesToRender >> 2, 16, sampleRate, 2);
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ao_play(device, (char*)realtimeBuffer, numBytesToRender);
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}
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buf_size = format.bits/8 * format.channels * format.rate;
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ao_close(device);
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device = 0;
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ao_shutdown();
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while (!threadData) {
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soundCriticalSection->Enter();
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uint_32 numBytesToRender = 256;
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(*callback)(realtimeBuffer, numBytesToRender >> 2, 16, sampleRate, 2);
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ao_play(device, (char*)realtimeBuffer, numBytesToRender);
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soundCriticalSection->Leave();
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soundSyncEvent->Wait();
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}
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return 0;
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}
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bool AOSound_StartSound(int _sampleRate, StreamCallback _callback)
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{
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callback = _callback;
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threadData = 0;
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sampleRate = _sampleRate;
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memset(realtimeBuffer, 0, sizeof(realtimeBuffer));
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pthread_create(&thread, NULL, soundThread, (void *)NULL);
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return true;
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}
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void AOSound_StopSound()
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{
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threadData = 1;
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void *retval;
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pthread_join(thread, &retval);
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}
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ao_close(device);
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device = NULL;
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ao_shutdown();
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}
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void *soundThread(void *args) {
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((AOSound *)args)->SoundLoop();
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return NULL;
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}
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bool AOSound::Start() {
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memset(realtimeBuffer, 0, sizeof(realtimeBuffer));
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soundCriticalSection = new Common::CriticalSection(1);
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thread = new Common::Thread(soundThread, (void *)this);
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soundSyncEvent = new Common::Event();
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soundSyncEvent->Init();
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return true;
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}
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void AOSound::Update() {
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soundSyncEvent->Set();
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}
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void AOSound::Stop()
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{
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soundCriticalSection->Enter();
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threadData = 1;
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soundSyncEvent->Set();
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soundCriticalSection->Leave();
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soundSyncEvent->Shutdown();
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delete soundCriticalSection;
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delete thread;
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delete soundSyncEvent;
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}
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#endif
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@ -1,4 +1,4 @@
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// Copyright (C) 2003-2008 Dolphin Project.
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// Copyright (C) 2003-2009 Dolphin Project.
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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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@ -18,18 +18,69 @@
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#ifndef __AOSOUNDSTREAM_H__
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#define __AOSOUNDSTREAM_H__
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namespace AOSound
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#include "SoundStream.h"
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#if defined(HAVE_AO) && HAVE_AO
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#include <ao/ao.h>
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#endif
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#include "Thread.h"
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class AOSound : public SoundStream
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{
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typedef void (*StreamCallback)(short* buffer, int numSamples, int bits, int rate, int channels);
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bool AOSound_StartSound(int sampleRate, StreamCallback _callback);
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void AOSound_UpdateSound();
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void AOSound_StopSound();
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#if defined(HAVE_AO) && HAVE_AO
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Common::Thread *thread;
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float AOSound_GetTimer();
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int AOSound_GetCurSample();
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int AOSound_GetSampleRate();
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}
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Common::CriticalSection *soundCriticalSection;
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Common::Event *soundSyncEvent;
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int buf_size;
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ao_device *device;
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ao_sample_format format;
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int default_driver;
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short realtimeBuffer[1024 * 1024];
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public:
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AOSound(int _sampleRate, StreamCallback _callback) :
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SoundStream(_sampleRate, _callback) {}
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virtual ~AOSound() {}
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virtual bool Start();
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virtual void SoundLoop();
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virtual void Stop();
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static bool isValid() {
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return true;
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}
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virtual bool usesMixer() {
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return true;
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}
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virtual void Update();
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virtual int GetSampleRate() {
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return sampleRate;
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}
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#else
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public:
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AOSound(int _sampleRate, StreamCallback _callback) :
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SoundStream(_sampleRate, _callback) {}
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#endif
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};
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#endif //__AOSOUNDSTREAM_H__
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@ -1,4 +1,4 @@
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// Copyright (C) 2003-2008 Dolphin Project.
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// Copyright (C) 2003-2009 Dolphin Project.
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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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@ -15,53 +15,12 @@
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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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#include "stdafx.h"
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#include <mmsystem.h>
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#include <dsound.h>
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#include "DSoundStream.h"
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#include "../main.h"
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namespace DSound
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{
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#include <dxerr.h>
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#define BUFSIZE 32768
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#define MAXWAIT 70 //ms
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CRITICAL_SECTION soundCriticalSection;
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HANDLE soundSyncEvent;
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HANDLE hThread;
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StreamCallback callback;
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IDirectSound8* ds;
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IDirectSoundBuffer* dsBuffer;
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int bufferSize; //i bytes
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int totalRenderedBytes;
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int sampleRate;
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// playback position
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int currentPos;
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int lastPos;
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short realtimeBuffer[1024 * 1024];
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// We set this to shut down the sound thread.
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// 0=keep playing, 1=stop playing NOW.
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volatile int threadData;
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inline int FIX128(int x)
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{
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return(x & (~127));
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}
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int DSound_GetSampleRate()
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{
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return(sampleRate);
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}
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bool CreateBuffer()
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bool DSound::CreateBuffer()
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{
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PCMWAVEFORMAT pcmwf;
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DSBUFFERDESC dsbdesc;
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@ -80,22 +39,23 @@ bool CreateBuffer()
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dsbdesc.dwSize = sizeof(DSBUFFERDESC);
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dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_STICKYFOCUS; //VIKTIGT //DSBCAPS_CTRLPAN | DSBCAPS_CTRLVOLUME | DSBCAPS_CTRLFREQUENCY;
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dsbdesc.dwBufferBytes = bufferSize = BUFSIZE;
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dsbdesc.lpwfxFormat = (WAVEFORMATEX*)&pcmwf;
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dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&pcmwf;
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if (SUCCEEDED(ds->CreateSoundBuffer(&dsbdesc, &dsBuffer, NULL)))
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HRESULT res = ds->CreateSoundBuffer(&dsbdesc, &dsBuffer, NULL);
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if (SUCCEEDED(res))
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{
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dsBuffer->SetCurrentPosition(0);
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return(true);
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return true;
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}
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else
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{
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else {
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// Failed.
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PanicAlert("Sound buffer creation failed: %s", DXGetErrorString(res));
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dsBuffer = NULL;
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return(false);
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return false;
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}
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}
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bool WriteDataToBuffer(DWORD dwOffset, // Our own write cursor.
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bool DSound::WriteDataToBuffer(DWORD dwOffset, // Our own write cursor.
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char* soundData, // Start of our data.
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DWORD dwSoundBytes) // Size of block to copy.
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{
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@ -122,20 +82,22 @@ bool WriteDataToBuffer(DWORD dwOffset, // Our own write cursor.
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// Release the data back to DirectSound.
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dsBuffer->Unlock(ptr1, numBytes1, ptr2, numBytes2);
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return(true);
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return true;
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}
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return(false);
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}
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inline int ModBufferSize(int x)
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{
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return((x + bufferSize) % bufferSize);
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return false;
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}
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// The audio thread.
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DWORD WINAPI soundThread(void*)
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DWORD WINAPI soundThread(void* args)
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{
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((DSound *)args)->SoundLoop();
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return 0; //huzzah! :D
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}
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void DSound::SoundLoop() {
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currentPos = 0;
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lastPos = 0;
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@ -144,18 +106,19 @@ DWORD WINAPI soundThread(void*)
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// dsBuffer->Lock(0, bufferSize, (void **)&p1, &num1, (void **)&p2, &num2, 0);
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dsBuffer->Play(0, 0, DSBPLAY_LOOPING);
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while (!threadData)
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{
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while (!threadData) {
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// No blocking inside the csection
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EnterCriticalSection(&soundCriticalSection);
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soundCriticalSection->Enter();
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dsBuffer->GetCurrentPosition((DWORD*)¤tPos, 0);
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int numBytesToRender = FIX128(ModBufferSize(currentPos - lastPos));
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int numBytesToRender = FIX128(
|
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ModBufferSize(currentPos - lastPos));
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if (numBytesToRender >= 256)
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{
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if (numBytesToRender > sizeof(realtimeBuffer))
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MessageBox(0,"soundThread: too big render call",0,0);
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(*callback)(realtimeBuffer, numBytesToRender >> 2, 16, sampleRate, 2);
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PanicAlert("soundThread: too big render call");
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(*callback)(realtimeBuffer, numBytesToRender >> 2, 16,
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sampleRate, 2);
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WriteDataToBuffer(lastPos, (char*)realtimeBuffer, numBytesToRender);
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currentPos = ModBufferSize(lastPos + numBytesToRender);
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@ -164,36 +127,32 @@ DWORD WINAPI soundThread(void*)
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lastPos = currentPos;
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}
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LeaveCriticalSection(&soundCriticalSection);
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WaitForSingleObject(soundSyncEvent, MAXWAIT);
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soundCriticalSection->Leave();
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soundSyncEvent->Wait();
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}
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|
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dsBuffer->Stop();
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return(0); //hurra!
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}
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|
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bool DSound_StartSound(HWND window, int _sampleRate, StreamCallback _callback)
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bool DSound::Start()
|
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{
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callback = _callback;
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threadData = 0;
|
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sampleRate = _sampleRate;
|
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|
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//no security attributes, automatic resetting, init state nonset, untitled
|
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soundSyncEvent = CreateEvent(0, false, false, 0);
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soundSyncEvent = new Common::Event();
|
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soundSyncEvent->Init();
|
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|
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//vi initierar den...........
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InitializeCriticalSection(&soundCriticalSection);
|
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soundCriticalSection = new Common::CriticalSection();
|
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|
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//vi vill ha access till DSOUND s<>...
|
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if (FAILED(DirectSoundCreate8(0, &ds, 0)))
|
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return false;
|
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return false;
|
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|
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ds->SetCooperativeLevel(window, DSSCL_NORMAL);
|
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if(hWnd)
|
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ds->SetCooperativeLevel((HWND)hWnd, DSSCL_NORMAL);
|
||||
|
||||
if (!CreateBuffer())
|
||||
{
|
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return false;
|
||||
}
|
||||
|
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DWORD num1;
|
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short* p1;
|
||||
@ -201,48 +160,43 @@ bool DSound_StartSound(HWND window, int _sampleRate, StreamCallback _callback)
|
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memset(p1, 0, num1);
|
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dsBuffer->Unlock(p1, num1, 0, 0);
|
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totalRenderedBytes = -bufferSize;
|
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DWORD h;
|
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hThread = CreateThread(0, 0, soundThread, 0, 0, &h);
|
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SetThreadPriority(hThread, THREAD_PRIORITY_ABOVE_NORMAL);
|
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|
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thread = new Common::Thread(soundThread, (void *)this);
|
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return true;
|
||||
}
|
||||
|
||||
void DSound_UpdateSound()
|
||||
void DSound::Update()
|
||||
{
|
||||
SetEvent(soundSyncEvent);
|
||||
soundSyncEvent->Set();
|
||||
}
|
||||
|
||||
void DSound_StopSound()
|
||||
void DSound::Stop()
|
||||
{
|
||||
EnterCriticalSection(&soundCriticalSection);
|
||||
soundCriticalSection->Enter();
|
||||
threadData = 1;
|
||||
// kick the thread if it's waiting
|
||||
SetEvent(soundSyncEvent);
|
||||
LeaveCriticalSection(&soundCriticalSection);
|
||||
WaitForSingleObject(hThread, INFINITE);
|
||||
CloseHandle(hThread);
|
||||
soundSyncEvent->Set();
|
||||
soundCriticalSection->Leave();
|
||||
delete soundCriticalSection;
|
||||
delete thread;
|
||||
|
||||
dsBuffer->Release();
|
||||
ds->Release();
|
||||
|
||||
CloseHandle(soundSyncEvent);
|
||||
soundSyncEvent = INVALID_HANDLE_VALUE;
|
||||
hThread = INVALID_HANDLE_VALUE;
|
||||
soundSyncEvent->Shutdown();
|
||||
delete soundSyncEvent;
|
||||
soundSyncEvent = NULL;
|
||||
thread = NULL;
|
||||
}
|
||||
|
||||
int DSound_GetCurSample()
|
||||
/* Unused, is it needed?
|
||||
int DSound::GetCurSample()
|
||||
{
|
||||
EnterCriticalSection(&soundCriticalSection);
|
||||
soundCriticalSection->Enter();
|
||||
int playCursor;
|
||||
dsBuffer->GetCurrentPosition((DWORD*)&playCursor, 0);
|
||||
playCursor = ModBufferSize(playCursor - lastPos) + totalRenderedBytes;
|
||||
LeaveCriticalSection(&soundCriticalSection);
|
||||
return(playCursor);
|
||||
soundCriticalSection->Leave();
|
||||
return playCursor;
|
||||
}
|
||||
|
||||
float DSound_GetTimer()
|
||||
{
|
||||
return((float)DSound_GetCurSample() * (1.0f / (4.0f * sampleRate)));
|
||||
}
|
||||
|
||||
} // namespace
|
||||
*/
|
||||
|
@ -1,4 +1,4 @@
|
||||
// Copyright (C) 2003-2008 Dolphin Project.
|
||||
// Copyright (C) 2003-2009 Dolphin Project.
|
||||
|
||||
// This program is free software: you can redistribute it and/or modify
|
||||
// it under the terms of the GNU General Public License as published by
|
||||
@ -15,21 +15,90 @@
|
||||
// Official SVN repository and contact information can be found at
|
||||
// http://code.google.com/p/dolphin-emu/
|
||||
|
||||
#ifndef __SOUNDSTREAM_H__
|
||||
#define __SOUNDSTREAM_H__
|
||||
#ifndef __DSOUNDSTREAM_H__
|
||||
#define __DSOUNDSTREAM_H__
|
||||
#include "SoundStream.h"
|
||||
|
||||
namespace DSound
|
||||
#include "Thread.h"
|
||||
|
||||
#ifdef _WIN32
|
||||
#include "stdafx.h"
|
||||
|
||||
#include <mmsystem.h>
|
||||
#include <dsound.h>
|
||||
|
||||
#define BUFSIZE 32768
|
||||
#define MAXWAIT 70 //ms
|
||||
|
||||
#endif
|
||||
|
||||
class DSound : public SoundStream
|
||||
{
|
||||
typedef void (*StreamCallback)(short* buffer, int numSamples, int bits, int rate, int channels);
|
||||
|
||||
#ifdef _WIN32
|
||||
|
||||
bool DSound_StartSound(HWND window, int sampleRate, StreamCallback _callback);
|
||||
void DSound_UpdateSound();
|
||||
void DSound_StopSound();
|
||||
Common::Thread *thread;
|
||||
|
||||
Common::CriticalSection *soundCriticalSection;
|
||||
Common::Event *soundSyncEvent;
|
||||
void *hWnd;
|
||||
|
||||
float DSound_GetTimer();
|
||||
int DSound_GetCurSample();
|
||||
int DSound_GetSampleRate();
|
||||
}
|
||||
IDirectSound8* ds;
|
||||
IDirectSoundBuffer* dsBuffer;
|
||||
|
||||
int bufferSize; //i bytes
|
||||
int totalRenderedBytes;
|
||||
|
||||
// playback position
|
||||
int currentPos;
|
||||
int lastPos;
|
||||
short realtimeBuffer[1024 * 1024];
|
||||
|
||||
inline int FIX128(int x) {
|
||||
return x & (~127);
|
||||
}
|
||||
|
||||
inline int ModBufferSize(int x) {
|
||||
return (x + bufferSize) % bufferSize;
|
||||
}
|
||||
|
||||
bool CreateBuffer();
|
||||
|
||||
bool WriteDataToBuffer(DWORD dwOffset, char* soundData,
|
||||
DWORD dwSoundBytes);
|
||||
|
||||
public:
|
||||
|
||||
DSound(int _sampleRate, StreamCallback _callback) :
|
||||
SoundStream(_sampleRate, _callback) {}
|
||||
|
||||
DSound(int _sampleRate, StreamCallback _callback, void *_hWnd) :
|
||||
SoundStream(_sampleRate, _callback), hWnd(_hWnd) {}
|
||||
|
||||
virtual ~DSound() {}
|
||||
|
||||
virtual bool Start();
|
||||
|
||||
virtual void SoundLoop();
|
||||
|
||||
virtual void Stop();
|
||||
|
||||
static bool isValid() { return true; }
|
||||
|
||||
virtual bool usesMixer() { return true; }
|
||||
|
||||
virtual void Update();
|
||||
|
||||
#else
|
||||
public:
|
||||
|
||||
DSound(int _sampleRate, StreamCallback _callback, void *hWnd = NULL) :
|
||||
SoundStream(_sampleRate, _callback) {}
|
||||
|
||||
|
||||
#endif //__SOUNDSTREAM_H__
|
||||
|
||||
#endif
|
||||
|
||||
};
|
||||
|
||||
#endif //__DSOUNDSTREAM_H__
|
||||
|
@ -16,9 +16,6 @@
|
||||
// http://code.google.com/p/dolphin-emu/
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////////////////
|
||||
// Includes
|
||||
// -------------
|
||||
// This queue solution is temporary. I'll implement something more efficient later.
|
||||
#include <queue> // System
|
||||
|
||||
@ -29,14 +26,11 @@
|
||||
#include "../Globals.h"
|
||||
#include "../DSPHandler.h"
|
||||
#include "../Debugger/File.h"
|
||||
#include "../main.h"
|
||||
|
||||
#include "Mixer.h"
|
||||
#include "FixedSizeQueue.h"
|
||||
|
||||
#ifdef _WIN32
|
||||
#include "../PCHW/DSoundStream.h"
|
||||
#endif
|
||||
///////////////////////
|
||||
|
||||
|
||||
namespace {
|
||||
@ -52,6 +46,11 @@ FixedSizeQueue<s16, queue_maxlength> sample_queue;
|
||||
|
||||
volatile bool mixer_HLEready = false;
|
||||
volatile int queue_size = 0;
|
||||
bool bThrottling = false;
|
||||
|
||||
void UpdateThrottle(bool update) {
|
||||
bThrottling = update;
|
||||
}
|
||||
|
||||
void Mixer(short *buffer, int numSamples, int bits, int rate, int channels)
|
||||
{
|
||||
@ -111,87 +110,85 @@ void Mixer_PushSamples(short *buffer, int num_stereo_samples, int sample_rate) {
|
||||
|
||||
static int PV1l=0,PV2l=0,PV3l=0,PV4l=0;
|
||||
static int PV1r=0,PV2r=0,PV3r=0,PV4r=0;
|
||||
static int acc=0;
|
||||
static int acc=0;
|
||||
|
||||
bThrottling = g_Config.m_EnableThrottle;
|
||||
|
||||
if(bThrottling) {
|
||||
|
||||
/* This is only needed for non-AX sound, currently directly
|
||||
streamed and DTK sound. For AX we call SoundStream::Update in
|
||||
AXTask() for example. */
|
||||
while (queue_size > queue_maxlength / 2) {
|
||||
soundStream->Update();
|
||||
Common::SleepCurrentThread(0);
|
||||
}
|
||||
|
||||
#ifdef _WIN32
|
||||
if (! (GetAsyncKeyState(VK_TAB)) && g_Config.m_EnableThrottle) {
|
||||
//convert into config option?
|
||||
const int mode = 2;
|
||||
|
||||
/* This is only needed for non-AX sound, currently directly streamed and
|
||||
DTK sound. For AX we call DSound_UpdateSound in AXTask() for example. */
|
||||
while (queue_size > queue_maxlength / 2) {
|
||||
DSound::DSound_UpdateSound();
|
||||
Sleep(0);
|
||||
}
|
||||
} else {
|
||||
return;
|
||||
}
|
||||
#else
|
||||
while (queue_size > queue_maxlength) {
|
||||
usleep(1000);
|
||||
}
|
||||
#endif
|
||||
//convert into config option?
|
||||
const int mode = 2;
|
||||
push_sync.Enter();
|
||||
while (num_stereo_samples)
|
||||
{
|
||||
acc += sample_rate;
|
||||
while (num_stereo_samples && (acc >= 48000))
|
||||
{
|
||||
PV4l=PV3l;
|
||||
PV3l=PV2l;
|
||||
PV2l=PV1l;
|
||||
PV1l=*(buffer++); //32bit processing
|
||||
PV4r=PV3r;
|
||||
PV3r=PV2r;
|
||||
PV2r=PV1r;
|
||||
PV1r=*(buffer++); //32bit processing
|
||||
num_stereo_samples--;
|
||||
acc-=48000;
|
||||
}
|
||||
|
||||
push_sync.Enter();
|
||||
while (num_stereo_samples)
|
||||
{
|
||||
acc += sample_rate;
|
||||
while (num_stereo_samples && (acc >= 48000))
|
||||
{
|
||||
PV4l=PV3l;
|
||||
PV3l=PV2l;
|
||||
PV2l=PV1l;
|
||||
PV1l=*(buffer++); //32bit processing
|
||||
PV4r=PV3r;
|
||||
PV3r=PV2r;
|
||||
PV2r=PV1r;
|
||||
PV1r=*(buffer++); //32bit processing
|
||||
num_stereo_samples--;
|
||||
acc-=48000;
|
||||
}
|
||||
// defaults to nearest
|
||||
s32 DataL = PV1l;
|
||||
s32 DataR = PV1r;
|
||||
|
||||
// defaults to nearest
|
||||
s32 DataL = PV1l;
|
||||
s32 DataR = PV1r;
|
||||
|
||||
if (mode == 1) //linear
|
||||
{
|
||||
DataL = PV1l + ((PV2l - PV1l)*acc)/48000;
|
||||
DataR = PV1r + ((PV2r - PV1r)*acc)/48000;
|
||||
}
|
||||
else if (mode == 2) //cubic
|
||||
{
|
||||
s32 a0l = PV1l - PV2l - PV4l + PV3l;
|
||||
s32 a0r = PV1r - PV2r - PV4r + PV3r;
|
||||
s32 a1l = PV4l - PV3l - a0l;
|
||||
s32 a1r = PV4r - PV3r - a0r;
|
||||
s32 a2l = PV1l - PV4l;
|
||||
s32 a2r = PV1r - PV4r;
|
||||
s32 a3l = PV2l;
|
||||
s32 a3r = PV2r;
|
||||
if (mode == 1) //linear
|
||||
{
|
||||
DataL = PV1l + ((PV2l - PV1l)*acc)/48000;
|
||||
DataR = PV1r + ((PV2r - PV1r)*acc)/48000;
|
||||
}
|
||||
else if (mode == 2) //cubic
|
||||
{
|
||||
s32 a0l = PV1l - PV2l - PV4l + PV3l;
|
||||
s32 a0r = PV1r - PV2r - PV4r + PV3r;
|
||||
s32 a1l = PV4l - PV3l - a0l;
|
||||
s32 a1r = PV4r - PV3r - a0r;
|
||||
s32 a2l = PV1l - PV4l;
|
||||
s32 a2r = PV1r - PV4r;
|
||||
s32 a3l = PV2l;
|
||||
s32 a3r = PV2r;
|
||||
|
||||
s32 t0l = ((a0l )*acc)/48000;
|
||||
s32 t0r = ((a0r )*acc)/48000;
|
||||
s32 t1l = ((t0l+a1l)*acc)/48000;
|
||||
s32 t1r = ((t0r+a1r)*acc)/48000;
|
||||
s32 t2l = ((t1l+a2l)*acc)/48000;
|
||||
s32 t2r = ((t1r+a2r)*acc)/48000;
|
||||
s32 t3l = ((t2l+a3l));
|
||||
s32 t3r = ((t2r+a3r));
|
||||
s32 t0l = ((a0l )*acc)/48000;
|
||||
s32 t0r = ((a0r )*acc)/48000;
|
||||
s32 t1l = ((t0l+a1l)*acc)/48000;
|
||||
s32 t1r = ((t0r+a1r)*acc)/48000;
|
||||
s32 t2l = ((t1l+a2l)*acc)/48000;
|
||||
s32 t2r = ((t1r+a2r)*acc)/48000;
|
||||
s32 t3l = ((t2l+a3l));
|
||||
s32 t3r = ((t2r+a3r));
|
||||
|
||||
DataL = t3l;
|
||||
DataR = t3r;
|
||||
}
|
||||
DataL = t3l;
|
||||
DataR = t3r;
|
||||
}
|
||||
|
||||
int l = DataL, r = DataR;
|
||||
if (l < -32767) l = -32767;
|
||||
if (r < -32767) r = -32767;
|
||||
if (l > 32767) l = 32767;
|
||||
if (r > 32767) r = 32767;
|
||||
sample_queue.push(l);
|
||||
sample_queue.push(r);
|
||||
queue_size += 2;
|
||||
}
|
||||
push_sync.Leave();
|
||||
}
|
||||
|
||||
int l = DataL, r = DataR;
|
||||
if (l < -32767) l = -32767;
|
||||
if (r < -32767) r = -32767;
|
||||
if (l > 32767) l = 32767;
|
||||
if (r > 32767) r = 32767;
|
||||
sample_queue.push(l);
|
||||
sample_queue.push(r);
|
||||
queue_size += 2;
|
||||
}
|
||||
push_sync.Leave();
|
||||
}
|
||||
|
||||
|
58
Source/Plugins/Plugin_DSP_HLE/Src/PCHW/SoundStream.h
Normal file
58
Source/Plugins/Plugin_DSP_HLE/Src/PCHW/SoundStream.h
Normal file
@ -0,0 +1,58 @@
|
||||
// Copyright (C) 2003-2009 Dolphin Project.
|
||||
|
||||
// This program is free software: you can redistribute it and/or modify
|
||||
// it under the terms of the GNU General Public License as published by
|
||||
// the Free Software Foundation, version 2.0.
|
||||
|
||||
// This program is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
||||
// GNU General Public License 2.0 for more details.
|
||||
|
||||
// A copy of the GPL 2.0 should have been included with the program.
|
||||
// If not, see http://www.gnu.org/licenses/
|
||||
|
||||
// Official SVN repository and contact information can be found at
|
||||
// http://code.google.com/p/dolphin-emu/
|
||||
|
||||
#ifndef __SOUNDSTREAM_H__
|
||||
#define __SOUNDSTREAM_H__
|
||||
|
||||
#include "Common.h"
|
||||
|
||||
typedef void (*StreamCallback)(short* buffer, int numSamples, int bits, int rate, int channels);
|
||||
|
||||
class SoundStream {
|
||||
|
||||
protected:
|
||||
int sampleRate;
|
||||
|
||||
StreamCallback callback;
|
||||
|
||||
// We set this to shut down the sound thread.
|
||||
// 0=keep playing, 1=stop playing NOW.
|
||||
volatile int threadData;
|
||||
|
||||
public:
|
||||
SoundStream(int _sampleRate, StreamCallback _callback) :
|
||||
sampleRate(_sampleRate), callback(_callback), threadData(0) {}
|
||||
|
||||
virtual ~SoundStream() {}
|
||||
|
||||
static bool isValid() { return false; }
|
||||
|
||||
virtual bool usesMixer() { return false; }
|
||||
|
||||
virtual bool Start() { return false; }
|
||||
|
||||
virtual void SoundLoop() { }
|
||||
|
||||
virtual void Stop() {}
|
||||
|
||||
virtual void Update() {}
|
||||
|
||||
virtual int GetSampleRate() { return sampleRate; }
|
||||
|
||||
};
|
||||
|
||||
#endif
|
Reference in New Issue
Block a user