dolphin/Source/Plugins/Plugin_DSP_HLE/Src/PCHW/Mixer.cpp

200 lines
6.1 KiB
C++

// 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 "ConsoleWindow.h"
#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;
bool bThrottling = false;
void UpdateThrottle(bool update) {
bThrottling = update;
}
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) {
// 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;
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);
}
//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();
}
}