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Attempt to move mixer to audio common, it's a bit more complicated than I expected

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so please check I didn't break anything in hle



git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@2756 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
nakeee
2009-03-26 09:29:14 +00:00
parent d7038fea17
commit fff663e8c7
35 changed files with 386 additions and 619 deletions
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25
Source/Plugins/Plugin_DSP_HLE/Src/HLEMixer.cpp Normal file
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@ -0,0 +1,25 @@
#include "Config.h" // Local
#include "Globals.h"
#include "DSPHandler.h"
#include "HLEMixer.h"
void HLEMixer::MixUCode(short *samples, int numSamples) {
//if this was called directly from the HLE, and not by timeout
if (g_Config.m_EnableHLEAudio && IsHLEReady()) {
IUCode* pUCode = CDSPHandler::GetInstance().GetUCode();
if (pUCode != NULL)
pUCode->MixAdd(samples, numSamples);
}
}
void HLEMixer::Premix(short *samples, int numSamples) {
// first get the DTK Music
if (g_Config.m_EnableDTKMusic) {
g_dspInitialize.pGetAudioStreaming(samples, numSamples);
}
MixUCode(samples, numSamples);
}

16
Source/Plugins/Plugin_DSP_HLE/Src/HLEMixer.h Normal file
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@ -0,0 +1,16 @@
#ifndef HLEMIXER_H
#define HLEMIXER_H
#include "AudioCommon.h"
#include "Mixer.h"
class HLEMixer : public CMixer
{
public:
void MixUCode(short *samples, int numSamples);
virtual void Premix(short *samples, int numSamples);
};
#endif // HLEMIXER_H

200
Source/Plugins/Plugin_DSP_HLE/Src/PCHW/Mixer.cpp
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@ -1,200 +0,0 @@
// Copyright (C) 2003-2008 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/
// This queue solution is temporary. I'll implement something more efficient later.
#include <queue> // System
#include "Thread.h" // Common
#include "../Config.h" // Local
#include "../Globals.h"
#include "../DSPHandler.h"
#include "../Debugger/File.h"
#include "../main.h"
#include "Mixer.h"
#include "FixedSizeQueue.h"
namespace {
Common::CriticalSection push_sync;
// On real hardware, this fifo is much, much smaller. But timing is also tighter than under Windows, so...
const int queue_minlength = 1024 * 4;
const int queue_maxlength = 1024 * 28;
FixedSizeQueue<s16, queue_maxlength> sample_queue;
} // namespace
volatile bool mixer_HLEready = false;
volatile int queue_size = 0;
void Mixer(short *buffer, int numSamples, int bits, int rate, int channels)
{
// silence
memset(buffer, 0, numSamples * 2 * sizeof(short));
if (g_dspInitialize.pEmulatorState) {
if (*g_dspInitialize.pEmulatorState != 0)
return;
}
// first get the DTK Music
if (g_Config.m_EnableDTKMusic)
{
g_dspInitialize.pGetAudioStreaming(buffer, numSamples);
}
Mixer_MixUCode(buffer, numSamples, bits, rate, channels);
push_sync.Enter();
int count = 0;
while (queue_size > queue_minlength && count < numSamples * 2) {
int x = buffer[count];
x += sample_queue.front();
if (x > 32767) x = 32767;
if (x < -32767) x = -32767;
buffer[count++] = x;
sample_queue.pop();
x = buffer[count];
x += sample_queue.front();
if (x > 32767) x = 32767;
if (x < -32767) x = -32767;
buffer[count++] = x;
sample_queue.pop();
queue_size-=2;
}
push_sync.Leave();
}
void Mixer_MixUCode(short *buffer, int numSamples, int bits, int rate,
int channels) {
//if this was called directly from the HLE, and not by timeout
if (g_Config.m_EnableHLEAudio && mixer_HLEready)
{
IUCode* pUCode = CDSPHandler::GetInstance().GetUCode();
if (pUCode != NULL)
pUCode->MixAdd(buffer, numSamples);
}
}
void Mixer_PushSamples(short *buffer, int num_stereo_samples, int sample_rate)
{
if (!soundStream)
return;
if (queue_size == 0)
{
queue_size = queue_minlength;
for (int i = 0; i < queue_minlength; i++)
sample_queue.push((s16)0);
}
static int PV1l=0,PV2l=0,PV3l=0,PV4l=0;
static int PV1r=0,PV2r=0,PV3r=0,PV4r=0;
static int acc=0;
#ifdef _WIN32
if (GetAsyncKeyState(VK_TAB))
return;
#endif
// Write Other Audio
if (g_Config.m_EnableThrottle)
{
/* 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) {
// Urgh.
if (g_dspInitialize.pEmulatorState) {
if (*g_dspInitialize.pEmulatorState != 0)
return;
}
soundStream->Update();
Common::SleepCurrentThread(0);
}
//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;
}
// 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;
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;
}
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();
}
}

32
Source/Plugins/Plugin_DSP_HLE/Src/PCHW/Mixer.h
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@ -1,32 +0,0 @@
// Copyright (C) 2003-2008 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 _MIXER_H
#define _MIXER_H
extern volatile bool mixer_HLEready;
// Called from audio threads
void Mixer(short* buffer, int numSamples, int bits, int rate, int channels);
void Mixer_MixUCode(short *buffer, int numSamples, int bits, int rate, int channels);
// Called from main thread
void Mixer_PushSamples(short *buffer, int num_stereo_samples, int sample_rate);
#endif

3
Source/Plugins/Plugin_DSP_HLE/Src/SConscript
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@ -8,11 +8,10 @@ name = "Plugin_DSP_HLE"
files = [
'DSPHandler.cpp',
'MailHandler.cpp',
'HLEMixer.cpp',
'main.cpp',
'Config.cpp',
'Globals.cpp',
# 'PCHW/AOSoundStream.cpp',
'PCHW/Mixer.cpp',
'Debugger/File.cpp',
'UCodes/UCode_AX.cpp',
'UCodes/UCode_AXWii.cpp',

4
Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AX.cpp
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@ -25,7 +25,7 @@ extern CDebugger* m_frame;
#include <sstream>
#include "../Globals.h"
#include "../PCHW/Mixer.h"
#include "Mixer.h"
#include "../MailHandler.h"
#include "UCodes.h"
@ -513,7 +513,7 @@ bool CUCode_AX::AXTask(u32& _uMail)
m_addressPBs = Memory_Read_U32(uAddress);
uAddress += 4;
mixer_HLEready = true;
soundStream->GetMixer()->SetHLEReady(true);
SaveLog("%08x : AXLIST PB address: %08x", uAddress, m_addressPBs);
SaveLog("Update the SoundThread to be in sync");

4
Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AXWii.cpp
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@ -23,8 +23,8 @@
extern CDebugger * m_frame;
#endif
#include "../PCHW/Mixer.h"
#include "../MailHandler.h"
#include "Mixer.h"
#include "UCodes.h"
#include "UCode_AXStructs.h"
@ -324,7 +324,7 @@ bool CUCode_AXWii::AXTask(u32& _uMail)
case 0x0004: // PBs are here now
m_addressPBs = Memory_Read_U32(uAddress);
lCUCode_AX->m_addressPBs = m_addressPBs; // for the sake of logging
mixer_HLEready = true;
soundStream->GetMixer()->SetHLEReady(true);
SaveLog("%08x : AXLIST PB address: %08x", uAddress, m_addressPBs);
soundStream->Update();
uAddress += 4;

5
Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_AX_Voice.h
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@ -21,7 +21,7 @@
#include "UCode_AX_ADPCM.h"
#include "UCode_AX.h"
#include "../main.h"
#include "Mixer.h"
// ----------------------------------------------------
// Externals
@ -107,7 +107,7 @@ inline void WriteBackPBsWii(u32 pbs_address, ParamBlockType& _pPBs, int _num)
template<class ParamBlockType>
inline void MixAddVoice(ParamBlockType &pb, int *templbuffer, int *temprbuffer, int _iSize, bool Wii)
{
ratioFactor = 32000.0f / (float)soundStream->GetSampleRate();
ratioFactor = 32000.0f / (float)soundStream->GetMixer()->GetSampleRate();
DoVoiceHacks(pb, Wii);
@ -115,7 +115,6 @@ inline void MixAddVoice(ParamBlockType &pb, int *templbuffer, int *temprbuffer,
if (pb.running)
{
// =======================================================================================
// Read initial parameters
// ------------
//constants

5
Source/Plugins/Plugin_DSP_HLE/Src/UCodes/UCode_Zelda.cpp
View File

@ -24,7 +24,7 @@
#include "../MailHandler.h"
#include "../main.h"
#include "../PCHW/Mixer.h"
#include "Mixer.h"
CUCode_Zelda::CUCode_Zelda(CMailHandler& _rMailHandler)
@ -157,8 +157,7 @@ void CUCode_Zelda::ExecuteList()
tmp[2] = Read32();
// We're ready to mix
mixer_HLEready = true;
soundStream->GetMixer()->SetHLEReady(true);
DEBUG_LOG(DSPHLE, "Update the SoundThread to be in sync");
soundStream->Update(); //do it in this thread to avoid sync problems

57
Source/Plugins/Plugin_DSP_HLE/Src/main.cpp
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@ -28,16 +28,17 @@ CDebugger* m_frame = NULL;
#include "ChunkFile.h"
#include "WaveFile.h"
#include "PCHW/Mixer.h"
#include "HLEMixer.h"
#include "DSPHandler.h"
#include "Config.h"
#include "Setup.h"
#include "StringUtil.h"
#include "AudioCommon.h"
#include "AOSoundStream.h"
#include "DSoundStream.h"
#include "NullSoundStream.h"
// Declarations and definitions
PLUGIN_GLOBALS* globals = NULL;
DSPInitialize g_dspInitialize;
@ -192,6 +193,7 @@ void DllConfig(HWND _hParent)
#endif
}
void Initialize(void *init)
{
g_dspInitialize = *(DSPInitialize*)init;
@ -201,45 +203,9 @@ void Initialize(void *init)
CDSPHandler::CreateInstance();
if (g_Config.sBackend == "DSound")
{
if (DSound::isValid())
soundStream = new DSound(48000, Mixer, g_dspInitialize.hWnd);
}
else if (g_Config.sBackend == "AOSound")
{
if (AOSound::isValid())
soundStream = new AOSound(48000, Mixer);
}
else if (g_Config.sBackend == "NullSound")
{
soundStream = new NullSound(48000, Mixer_MixUCode);
}
else
{
PanicAlert("Cannot recognize backend %s", g_Config.sBackend.c_str());
return;
}
if (soundStream)
{
if (!soundStream->Start())
{
PanicAlert("Could not initialize backend %s, falling back to NULL",
g_Config.sBackend.c_str());
delete soundStream;
soundStream = new NullSound(48000, Mixer);
soundStream->Start();
}
}
else
{
PanicAlert("Sound backend %s is not valid, falling back to NULL",
g_Config.sBackend.c_str());
delete soundStream;
soundStream = new NullSound(48000, Mixer);
soundStream->Start();
}
soundStream = AudioCommon::InitSoundStream(g_Config.sBackend,
new HLEMixer());
soundStream->GetMixer()->SetThrottle(g_Config.m_EnableThrottle);
// Start the sound recording
if (log_ai)
@ -251,15 +217,20 @@ void Initialize(void *init)
void DSP_StopSoundStream()
{
// fprintf(stderr, "in dsp stop\n");
if (!soundStream)
PanicAlert("Can't stop non running SoundStream!");
soundStream->Stop();
delete soundStream;
soundStream = NULL;
// fprintf(stderr, "in dsp stop end\n");
}
void Shutdown()
{
// FIXME: called before stop is finished????
// fprintf(stderr, "in dsp shutdown\n");
// Check that soundstream already is stopped.
if (soundStream)
PanicAlert("SoundStream alive in DSP::Shutdown!");
@ -384,7 +355,7 @@ void DSP_SendAIBuffer(unsigned int address, int sample_rate)
return;
}
if (soundStream->usesMixer())
if (soundStream->GetMixer())
{
short samples[16] = {0}; // interleaved stereo
if (address)
@ -398,7 +369,7 @@ void DSP_SendAIBuffer(unsigned int address, int sample_rate)
if (log_ai)
g_wave_writer.AddStereoSamples(samples, 8);
}
Mixer_PushSamples(samples, 32 / 4, sample_rate);
soundStream->GetMixer()->PushSamples(samples, 32 / 4);
}
// SoundStream is updated only when necessary (there is no 70 ms limit