mirror of
https://github.com/nzp-team/dquakeplus.git
synced 2024-11-26 13:51:07 +00:00
183 lines
4.6 KiB
C++
183 lines
4.6 KiB
C++
/*
|
|
Copyright (C) 1996-1997 Id Software, Inc.
|
|
Copyright (C) 2007 Peter Mackay and Chris Swindle.
|
|
|
|
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; either version 2
|
|
of the License, or (at your option) any later version.
|
|
|
|
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 for more details.
|
|
|
|
You should have received a copy of the GNU General Public License
|
|
along with this program; if not, write to the Free Software
|
|
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
|
|
|
|
*/
|
|
|
|
#include <pspaudiolib.h>
|
|
#include <pspdebug.h>
|
|
#include <pspkernel.h>
|
|
|
|
extern "C"
|
|
{
|
|
#include "../quakedef.h"
|
|
}
|
|
|
|
namespace quake
|
|
{
|
|
namespace sound
|
|
{
|
|
struct Sample
|
|
{
|
|
short left;
|
|
short right;
|
|
};
|
|
|
|
static const unsigned int channelCount = 2;
|
|
static const unsigned int inputBufferSize = 16384;
|
|
|
|
#if 1 //def NORMAL_MODE
|
|
static const unsigned int inputFrequency = 11025;
|
|
#else
|
|
static const unsigned int inputFrequency = 22050;
|
|
#endif
|
|
static const unsigned int outputFrequency = 44100;
|
|
static const unsigned int inputSamplesPerOutputSample = outputFrequency / inputFrequency;
|
|
static Sample inputBuffer[inputBufferSize];
|
|
static volatile unsigned int samplesRead;
|
|
|
|
static inline void copySamples(const Sample* first, const Sample* last, Sample* destination)
|
|
{
|
|
switch (inputSamplesPerOutputSample)
|
|
{
|
|
case 1:
|
|
memcpy(destination, first, (last - first) * sizeof(Sample));
|
|
break;
|
|
|
|
case 2:
|
|
for (const Sample* source = first; source != last; ++source)
|
|
{
|
|
const Sample sample = *source;
|
|
*destination++ = sample;
|
|
*destination++ = sample;
|
|
}
|
|
break;
|
|
|
|
case 4:
|
|
for (const Sample* source = first; source != last; ++source)
|
|
{
|
|
const Sample sample = *source;
|
|
*destination++ = sample;
|
|
*destination++ = sample;
|
|
*destination++ = sample;
|
|
*destination++ = sample;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void fillOutputBuffer(void* buffer, unsigned int samplesToWrite, void* userData)
|
|
{
|
|
// Where are we writing to?
|
|
Sample* const destination = static_cast<Sample*> (buffer);
|
|
|
|
// Where are we reading from?
|
|
const Sample* const firstSampleToRead = &inputBuffer[samplesRead];
|
|
|
|
// How many samples to read?
|
|
const unsigned int samplesToRead = samplesToWrite / inputSamplesPerOutputSample;
|
|
|
|
// Going to wrap past the end of the input buffer?
|
|
const unsigned int samplesBeforeEndOfInput = inputBufferSize - samplesRead;
|
|
if (samplesToRead > samplesBeforeEndOfInput)
|
|
{
|
|
// Yes, so write the first chunk from the end of the input buffer.
|
|
copySamples(
|
|
firstSampleToRead,
|
|
firstSampleToRead + samplesBeforeEndOfInput,
|
|
&destination[0]);
|
|
|
|
// Write the second chunk from the start of the input buffer.
|
|
const unsigned int samplesToReadFromBeginning = samplesToRead - samplesBeforeEndOfInput;
|
|
copySamples(
|
|
&inputBuffer[0],
|
|
&inputBuffer[samplesToReadFromBeginning],
|
|
&destination[samplesBeforeEndOfInput * inputSamplesPerOutputSample]);
|
|
}
|
|
else
|
|
{
|
|
// No wrapping, just copy.
|
|
copySamples(
|
|
firstSampleToRead,
|
|
firstSampleToRead + samplesToRead,
|
|
&destination[0]);
|
|
}
|
|
|
|
// Update the read offset.
|
|
samplesRead = (samplesRead + samplesToRead) % inputBufferSize;
|
|
}
|
|
}
|
|
}
|
|
|
|
using namespace quake;
|
|
using namespace quake::sound;
|
|
|
|
qboolean SNDDMA_Init(void)
|
|
{
|
|
// Set up Quake's audio.
|
|
shm = &sn;
|
|
shm->channels = channelCount;
|
|
shm->samplebits = 16;
|
|
shm->speed = inputFrequency;
|
|
shm->soundalive = qtrue;
|
|
shm->splitbuffer = qfalse;
|
|
shm->samples = inputBufferSize * channelCount;
|
|
shm->samplepos = 0;
|
|
shm->submission_chunk = 1;
|
|
shm->buffer = (unsigned char *) inputBuffer;
|
|
|
|
// Initialise the audio system. This initialises it for the CD audio module
|
|
// too.
|
|
pspAudioInit();
|
|
|
|
// Set the channel callback.
|
|
// Sound effects use channel 0, CD audio uses channel 1.
|
|
pspAudioSetChannelCallback(0, fillOutputBuffer, 0);
|
|
|
|
return qtrue;
|
|
}
|
|
|
|
void SNDDMA_Shutdown(void)
|
|
{
|
|
// Clear the mixing buffer so we don't get any noise during cleanup.
|
|
memset(inputBuffer, 0, sizeof(inputBuffer));
|
|
|
|
// Clear the channel callback.
|
|
pspAudioSetChannelCallback(0, 0, 0);
|
|
|
|
// Stop the audio system?
|
|
pspAudioEndPre();
|
|
|
|
// Insert a false delay so the thread can be cleaned up.
|
|
sceKernelDelayThread(50 * 1000);
|
|
|
|
// Shut down the audio system.
|
|
pspAudioEnd();
|
|
}
|
|
|
|
int SNDDMA_GetDMAPos(void)
|
|
{
|
|
return samplesRead * channelCount;
|
|
}
|
|
|
|
void SNDDMA_Submit(void)
|
|
{
|
|
}
|