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set svn:eol-style=native for Plugins/**.cpp

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git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@1441 8ced0084-cf51-0410-be5f-012b33b47a6e
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bushing
2008-12-08 05:25:12 +00:00
parent 9146b9b261
commit 901fe7c00f
142 changed files with 43834 additions and 43834 deletions
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496
Source/Plugins/Plugin_DSP_HLE/Src/PCHW/DSoundStream.cpp
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@ -1,248 +1,248 @@
// 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/
#include "stdafx.h"
#include <mmsystem.h>
#include <dsound.h>
#include "DSoundStream.h"
namespace DSound
{
#define BUFSIZE 32768
#define MAXWAIT 70 //ms
CRITICAL_SECTION soundCriticalSection;
HANDLE soundSyncEvent;
HANDLE hThread;
StreamCallback callback;
IDirectSound8* ds;
IDirectSoundBuffer* dsBuffer;
int bufferSize; //i bytes
int totalRenderedBytes;
int sampleRate;
// playback position
int currentPos;
int lastPos;
short realtimeBuffer[1024 * 1024];
// We set this to shut down the sound thread.
// 0=keep playing, 1=stop playing NOW.
volatile int threadData;
inline int FIX128(int x)
{
return(x & (~127));
}
int DSound_GetSampleRate()
{
return(sampleRate);
}
bool 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 = sampleRate;
pcmwf.wf.nBlockAlign = 4;
pcmwf.wf.nAvgBytesPerSec = pcmwf.wf.nSamplesPerSec * pcmwf.wf.nBlockAlign;
pcmwf.wBitsPerSample = 16;
//buffer description
dsbdesc.dwSize = sizeof(DSBUFFERDESC);
dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_STICKYFOCUS; //VIKTIGT //DSBCAPS_CTRLPAN | DSBCAPS_CTRLVOLUME | DSBCAPS_CTRLFREQUENCY;
dsbdesc.dwBufferBytes = bufferSize = BUFSIZE;
dsbdesc.lpwfxFormat = (WAVEFORMATEX*)&pcmwf;
if (SUCCEEDED(ds->CreateSoundBuffer(&dsbdesc, &dsBuffer, NULL)))
{
dsBuffer->SetCurrentPosition(0);
return(true);
}
else
{
// Failed.
dsBuffer = NULL;
return(false);
}
}
bool WriteDataToBuffer(DWORD dwOffset, // Our own write cursor.
char* soundData, // Start of our data.
DWORD dwSoundBytes) // Size of block to copy.
{
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);
}
inline int ModBufferSize(int x)
{
return((x + bufferSize) % bufferSize);
}
// The audio thread.
DWORD WINAPI soundThread(void*)
{
currentPos = 0;
lastPos = 0;
// Prefill buffer?
//writeDataToBuffer(0,realtimeBuffer,bufferSize);
// dsBuffer->Lock(0, bufferSize, (void **)&p1, &num1, (void **)&p2, &num2, 0);
dsBuffer->Play(0, 0, DSBPLAY_LOOPING);
while (!threadData)
{
// No blocking inside the csection
EnterCriticalSection(&soundCriticalSection);
dsBuffer->GetCurrentPosition((DWORD*)&currentPos, 0);
int numBytesToRender = FIX128(ModBufferSize(currentPos - lastPos));
if (numBytesToRender >= 256)
{
if (numBytesToRender > sizeof(realtimeBuffer))
MessageBox(0,"soundThread: too big render call",0,0);
(*callback)(realtimeBuffer, numBytesToRender >> 2, 16, sampleRate, 2);
WriteDataToBuffer(lastPos, (char*)realtimeBuffer, numBytesToRender);
currentPos = ModBufferSize(lastPos + numBytesToRender);
totalRenderedBytes += numBytesToRender;
lastPos = currentPos;
}
LeaveCriticalSection(&soundCriticalSection);
WaitForSingleObject(soundSyncEvent, MAXWAIT);
}
dsBuffer->Stop();
return(0); //hurra!
}
bool DSound_StartSound(HWND window, int _sampleRate, StreamCallback _callback)
{
callback = _callback;
threadData = 0;
sampleRate = _sampleRate;
//no security attributes, automatic resetting, init state nonset, untitled
soundSyncEvent = CreateEvent(0, false, false, 0);
//vi initierar den...........
InitializeCriticalSection(&soundCriticalSection);
//vi vill ha access till DSOUND s<>...
if (FAILED(DirectSoundCreate8(0, &ds, 0)))
return false;
ds->SetCooperativeLevel(window, DSSCL_NORMAL);
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);
totalRenderedBytes = -bufferSize;
DWORD h;
hThread = CreateThread(0, 0, soundThread, 0, 0, &h);
SetThreadPriority(hThread, THREAD_PRIORITY_ABOVE_NORMAL);
return true;
}
void DSound_UpdateSound()
{
SetEvent(soundSyncEvent);
}
void DSound_StopSound()
{
EnterCriticalSection(&soundCriticalSection);
threadData = 1;
// kick the thread if it's waiting
SetEvent(soundSyncEvent);
LeaveCriticalSection(&soundCriticalSection);
WaitForSingleObject(hThread, INFINITE);
CloseHandle(hThread);
dsBuffer->Release();
ds->Release();
CloseHandle(soundSyncEvent);
soundSyncEvent = INVALID_HANDLE_VALUE;
hThread = INVALID_HANDLE_VALUE;
}
int DSound_GetCurSample()
{
EnterCriticalSection(&soundCriticalSection);
int playCursor;
dsBuffer->GetCurrentPosition((DWORD*)&playCursor, 0);
playCursor = ModBufferSize(playCursor - lastPos) + totalRenderedBytes;
LeaveCriticalSection(&soundCriticalSection);
return(playCursor);
}
float DSound_GetTimer()
{
return((float)DSound_GetCurSample() * (1.0f / (4.0f * sampleRate)));
}
} // namespace
// 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/
#include "stdafx.h"
#include <mmsystem.h>
#include <dsound.h>
#include "DSoundStream.h"
namespace DSound
{
#define BUFSIZE 32768
#define MAXWAIT 70 //ms
CRITICAL_SECTION soundCriticalSection;
HANDLE soundSyncEvent;
HANDLE hThread;
StreamCallback callback;
IDirectSound8* ds;
IDirectSoundBuffer* dsBuffer;
int bufferSize; //i bytes
int totalRenderedBytes;
int sampleRate;
// playback position
int currentPos;
int lastPos;
short realtimeBuffer[1024 * 1024];
// We set this to shut down the sound thread.
// 0=keep playing, 1=stop playing NOW.
volatile int threadData;
inline int FIX128(int x)
{
return(x & (~127));
}
int DSound_GetSampleRate()
{
return(sampleRate);
}
bool 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 = sampleRate;
pcmwf.wf.nBlockAlign = 4;
pcmwf.wf.nAvgBytesPerSec = pcmwf.wf.nSamplesPerSec * pcmwf.wf.nBlockAlign;
pcmwf.wBitsPerSample = 16;
//buffer description
dsbdesc.dwSize = sizeof(DSBUFFERDESC);
dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_STICKYFOCUS; //VIKTIGT //DSBCAPS_CTRLPAN | DSBCAPS_CTRLVOLUME | DSBCAPS_CTRLFREQUENCY;
dsbdesc.dwBufferBytes = bufferSize = BUFSIZE;
dsbdesc.lpwfxFormat = (WAVEFORMATEX*)&pcmwf;
if (SUCCEEDED(ds->CreateSoundBuffer(&dsbdesc, &dsBuffer, NULL)))
{
dsBuffer->SetCurrentPosition(0);
return(true);
}
else
{
// Failed.
dsBuffer = NULL;
return(false);
}
}
bool WriteDataToBuffer(DWORD dwOffset, // Our own write cursor.
char* soundData, // Start of our data.
DWORD dwSoundBytes) // Size of block to copy.
{
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);
}
inline int ModBufferSize(int x)
{
return((x + bufferSize) % bufferSize);
}
// The audio thread.
DWORD WINAPI soundThread(void*)
{
currentPos = 0;
lastPos = 0;
// Prefill buffer?
//writeDataToBuffer(0,realtimeBuffer,bufferSize);
// dsBuffer->Lock(0, bufferSize, (void **)&p1, &num1, (void **)&p2, &num2, 0);
dsBuffer->Play(0, 0, DSBPLAY_LOOPING);
while (!threadData)
{
// No blocking inside the csection
EnterCriticalSection(&soundCriticalSection);
dsBuffer->GetCurrentPosition((DWORD*)&currentPos, 0);
int numBytesToRender = FIX128(ModBufferSize(currentPos - lastPos));
if (numBytesToRender >= 256)
{
if (numBytesToRender > sizeof(realtimeBuffer))
MessageBox(0,"soundThread: too big render call",0,0);
(*callback)(realtimeBuffer, numBytesToRender >> 2, 16, sampleRate, 2);
WriteDataToBuffer(lastPos, (char*)realtimeBuffer, numBytesToRender);
currentPos = ModBufferSize(lastPos + numBytesToRender);
totalRenderedBytes += numBytesToRender;
lastPos = currentPos;
}
LeaveCriticalSection(&soundCriticalSection);
WaitForSingleObject(soundSyncEvent, MAXWAIT);
}
dsBuffer->Stop();
return(0); //hurra!
}
bool DSound_StartSound(HWND window, int _sampleRate, StreamCallback _callback)
{
callback = _callback;
threadData = 0;
sampleRate = _sampleRate;
//no security attributes, automatic resetting, init state nonset, untitled
soundSyncEvent = CreateEvent(0, false, false, 0);
//vi initierar den...........
InitializeCriticalSection(&soundCriticalSection);
//vi vill ha access till DSOUND s<>...
if (FAILED(DirectSoundCreate8(0, &ds, 0)))
return false;
ds->SetCooperativeLevel(window, DSSCL_NORMAL);
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);
totalRenderedBytes = -bufferSize;
DWORD h;
hThread = CreateThread(0, 0, soundThread, 0, 0, &h);
SetThreadPriority(hThread, THREAD_PRIORITY_ABOVE_NORMAL);
return true;
}
void DSound_UpdateSound()
{
SetEvent(soundSyncEvent);
}
void DSound_StopSound()
{
EnterCriticalSection(&soundCriticalSection);
threadData = 1;
// kick the thread if it's waiting
SetEvent(soundSyncEvent);
LeaveCriticalSection(&soundCriticalSection);
WaitForSingleObject(hThread, INFINITE);
CloseHandle(hThread);
dsBuffer->Release();
ds->Release();
CloseHandle(soundSyncEvent);
soundSyncEvent = INVALID_HANDLE_VALUE;
hThread = INVALID_HANDLE_VALUE;
}
int DSound_GetCurSample()
{
EnterCriticalSection(&soundCriticalSection);
int playCursor;
dsBuffer->GetCurrentPosition((DWORD*)&playCursor, 0);
playCursor = ModBufferSize(playCursor - lastPos) + totalRenderedBytes;
LeaveCriticalSection(&soundCriticalSection);
return(playCursor);
}
float DSound_GetTimer()
{
return((float)DSound_GetCurSample() * (1.0f / (4.0f * sampleRate)));
}
} // namespace

366
Source/Plugins/Plugin_DSP_HLE/Src/PCHW/Mixer.cpp
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@ -1,183 +1,183 @@
// 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>
#include "../Config.h"
#include "../Globals.h"
#include "../DSPHandler.h"
#include "../Logging/Console.h"
#include "Thread.h"
#include "Mixer.h"
#include "FixedSizeQueue.h"
#ifdef _WIN32
#include "../PCHW/DSoundStream.h"
#endif
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));
// first get the DTK Music
if (g_Config.m_EnableDTKMusic)
{
g_dspInitialize.pGetAudioStreaming(buffer, numSamples);
}
//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);
}
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_PushSamples(short *buffer, int num_stereo_samples, int sample_rate) {
// static FILE *f;
// if (!f)
// f = fopen("d:\\hello.raw", "wb");
// fwrite(buffer, num_stereo_samples * 4, 1, f);
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)) && g_Config.m_EnableThrottle) {
/* 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) {
sleep(0);
}
#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;
}
// 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();
}
// 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>
#include "../Config.h"
#include "../Globals.h"
#include "../DSPHandler.h"
#include "../Logging/Console.h"
#include "Thread.h"
#include "Mixer.h"
#include "FixedSizeQueue.h"
#ifdef _WIN32
#include "../PCHW/DSoundStream.h"
#endif
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));
// first get the DTK Music
if (g_Config.m_EnableDTKMusic)
{
g_dspInitialize.pGetAudioStreaming(buffer, numSamples);
}
//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);
}
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_PushSamples(short *buffer, int num_stereo_samples, int sample_rate) {
// static FILE *f;
// if (!f)
// f = fopen("d:\\hello.raw", "wb");
// fwrite(buffer, num_stereo_samples * 4, 1, f);
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)) && g_Config.m_EnableThrottle) {
/* 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) {
sleep(0);
}
#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;
}
// 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();
}