dolphin/Source/Core/AudioCommon/Src/DSoundStream.cpp
hrydgard 419d6a244b DSP plugin merge - the two DSP plugins are now gone and all the code has been merged into Dolphin.
This WILL temporarily break the Linux and MacOSX builds but should be easy to fix.

Things left to do:
  * The UI on the new Audio tab for the LLE/HLE choice is ugly
  * At times the code still look "plugin-y" and needs cleanup
  * The two plugins should be merged further. DSPHLE should use the emulated memory etc of DSPLLE as much as possible, so that simply saving the DSPLLE state is enough. This would also bring the possibility of savestate compatibility between the two plugins.


git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@6947 8ced0084-cf51-0410-be5f-012b33b47a6e
2011-01-28 18:39:30 +00:00

188 lines
4.5 KiB
C++

// Copyright (C) 2003 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/
#include <windows.h>
#include <cmath>
#include <dxerr.h>
#include "AudioCommon.h"
#include "DSoundStream.h"
bool DSound::CreateBuffer()
{
PCMWAVEFORMAT pcmwf;
DSBUFFERDESC dsbdesc;
memset(&pcmwf, 0, sizeof(PCMWAVEFORMAT));
memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
pcmwf.wf.wFormatTag = WAVE_FORMAT_PCM;
pcmwf.wf.nChannels = 2;
pcmwf.wf.nSamplesPerSec = m_mixer->GetSampleRate();
pcmwf.wf.nBlockAlign = 4;
pcmwf.wf.nAvgBytesPerSec = pcmwf.wf.nSamplesPerSec * pcmwf.wf.nBlockAlign;
pcmwf.wBitsPerSample = 16;
// Fill out DSound buffer description.
dsbdesc.dwSize = sizeof(DSBUFFERDESC);
dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_STICKYFOCUS | DSBCAPS_CTRLVOLUME;
dsbdesc.dwBufferBytes = bufferSize = BUFSIZE;
dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&pcmwf;
dsbdesc.guid3DAlgorithm = DS3DALG_DEFAULT;
HRESULT res = ds->CreateSoundBuffer(&dsbdesc, &dsBuffer, NULL);
if (SUCCEEDED(res))
{
dsBuffer->SetCurrentPosition(0);
dsBuffer->SetVolume(m_volume);
return true;
}
else
{
// Failed.
PanicAlertT("Sound buffer creation failed: %s", DXGetErrorString(res));
dsBuffer = NULL;
return false;
}
}
bool DSound::WriteDataToBuffer(DWORD dwOffset, // Our own write cursor.
char* soundData, // Start of our data.
DWORD dwSoundBytes) // Size of block to copy.
{
// I want to record the regular audio to, how do I do that?
void *ptr1, *ptr2;
DWORD numBytes1, numBytes2;
// Obtain memory address of write block. This will be in two parts if the block wraps around.
HRESULT hr = dsBuffer->Lock(dwOffset, dwSoundBytes, &ptr1, &numBytes1, &ptr2, &numBytes2, 0);
// If the buffer was lost, restore and retry lock.
if (DSERR_BUFFERLOST == hr)
{
dsBuffer->Restore();
hr = dsBuffer->Lock(dwOffset, dwSoundBytes, &ptr1, &numBytes1, &ptr2, &numBytes2, 0);
}
if (SUCCEEDED(hr))
{
memcpy(ptr1, soundData, numBytes1);
if (ptr2 != 0)
memcpy(ptr2, soundData + numBytes1, numBytes2);
// Release the data back to DirectSound.
dsBuffer->Unlock(ptr1, numBytes1, ptr2, numBytes2);
return true;
}
return false;
}
// The audio thread.
void soundThread(DSound* dsound)
{
dsound->SoundLoop();
}
void DSound::SoundLoop()
{
currentPos = 0;
lastPos = 0;
dsBuffer->Play(0, 0, DSBPLAY_LOOPING);
while (!threadData)
{
// No blocking inside the csection
dsBuffer->GetCurrentPosition((DWORD*)&currentPos, 0);
int numBytesToRender = FIX128(ModBufferSize(currentPos - lastPos));
if (numBytesToRender >= 256)
{
if (numBytesToRender > sizeof(realtimeBuffer))
PanicAlert("soundThread: too big render call");
m_mixer->Mix(realtimeBuffer, numBytesToRender / 4);
WriteDataToBuffer(lastPos, (char*)realtimeBuffer, numBytesToRender);
lastPos = ModBufferSize(lastPos + numBytesToRender);
}
soundSyncEvent.Wait();
}
}
bool DSound::Start()
{
soundSyncEvent.Init();
if (FAILED(DirectSoundCreate8(0, &ds, 0)))
return false;
if (hWnd)
{
HRESULT hr = ds->SetCooperativeLevel((HWND)hWnd, DSSCL_PRIORITY);
}
if (!CreateBuffer())
return false;
DWORD num1;
short* p1;
dsBuffer->Lock(0, bufferSize, (void* *)&p1, &num1, 0, 0, 0);
memset(p1, 0, num1);
dsBuffer->Unlock(p1, num1, 0, 0);
thread = std::thread(soundThread, this);
return true;
}
void DSound::SetVolume(int volume)
{
// This is in "dBA attenuation" from 0 to -10000, logarithmic
m_volume = (int)floor(log10((float)volume) * 5000.0f) - 10000;
if (dsBuffer != NULL)
dsBuffer->SetVolume(m_volume);
}
void DSound::Update()
{
soundSyncEvent.Set();
}
void DSound::Clear(bool mute)
{
m_muted = mute;
if (dsBuffer != NULL)
{
if (m_muted)
{
dsBuffer->Stop();
}
else
{
dsBuffer->Play(0, 0, DSBPLAY_LOOPING);
}
}
}
void DSound::Stop()
{
threadData = 1;
// kick the thread if it's waiting
soundSyncEvent.Set();
thread.join();
dsBuffer->Stop();
dsBuffer->Release();
ds->Release();
soundSyncEvent.Shutdown();
}