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:
nakeee
2009-01-29 00:57:55 +00:00
parent 121be22532
commit 7219bcd4d5
46 changed files with 705 additions and 4584 deletions

View File

@ -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();
}