daedalus/Source/SysCTR/HLEAudio/AudioOutput.cpp
2022-04-14 15:01:56 +10:00

183 lines
4 KiB
C++

/*
Copyright (C) 2003 Azimer
Copyright (C) 2001,2006 StrmnNrmn
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 "stdafx.h"
#include "AudioOutput.h"
#include <stdio.h>
#include <new>
#include <3ds.h>
#include "Config/ConfigOptions.h"
#include "Debug/DBGConsole.h"
#include "HLEAudio/AudioBuffer.h"
#include "Utility/FramerateLimiter.h"
#include "Utility/Thread.h"
extern u32 gSoundSync;
static const u32 DESIRED_OUTPUT_FREQUENCY = 44100;
// Large BUFFER_SIZE creates huge delay on sound //Corn
static const u32 BUFFER_SIZE = 1024 * 2;
static const u32 CTR_NUM_SAMPLES = 512;
static ndspWaveBuf waveBuf[2];
static unsigned int waveBuf_id;
bool audioOpen = false;
static AudioOutput * ac;
CAudioBuffer *mAudioBuffer;
static void audioCallback(void *arg)
{
(void)arg;
if(waveBuf[waveBuf_id].status == NDSP_WBUF_DONE)
{
mAudioBuffer->Drain( reinterpret_cast< Sample * >( waveBuf[waveBuf_id].data_pcm16 ), CTR_NUM_SAMPLES );
DSP_FlushDataCache(waveBuf[waveBuf_id].data_pcm16, CTR_NUM_SAMPLES << 2);
ndspChnWaveBufAdd( 0, &waveBuf[waveBuf_id] );
waveBuf_id = !waveBuf_id;
}
}
static void AudioInit()
{
if (ndspInit() != 0)
return;
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
ndspChnSetFormat(0, NDSP_FORMAT_STEREO_PCM16);
ndspChnSetRate(0, 44100.0f);
waveBuf[0].data_vaddr = linearAlloc(CTR_NUM_SAMPLES * 4);
waveBuf[0].nsamples = CTR_NUM_SAMPLES;
waveBuf[0].status = 0;
waveBuf[1].data_vaddr = linearAlloc(CTR_NUM_SAMPLES * 4);
waveBuf[1].nsamples = CTR_NUM_SAMPLES;
waveBuf[1].status = 0;
memset(waveBuf[0].data_pcm16, 0, CTR_NUM_SAMPLES * 4);
memset(waveBuf[1].data_pcm16, 0, CTR_NUM_SAMPLES * 4);
waveBuf_id = 0;
ndspSetCallback(&audioCallback, nullptr);
ndspChnWaveBufAdd(0, &waveBuf[0]);
ndspChnWaveBufAdd(0, &waveBuf[1]);
// Everything OK
audioOpen = true;
}
static void AudioExit()
{
// Stop stream
ndspChnWaveBufClear(0);
ndspExit();
linearFree((void*)waveBuf[0].data_vaddr);
linearFree((void*)waveBuf[1].data_vaddr);
audioOpen = false;
}
AudioOutput::AudioOutput()
: mAudioPlaying( false )
, mFrequency( 44100 )
{
// Allocate audio buffer with malloc_64 to avoid cached/uncached aliasing
void * mem = malloc( sizeof( CAudioBuffer ) );
mAudioBuffer = new( mem ) CAudioBuffer( BUFFER_SIZE );
}
AudioOutput::~AudioOutput( )
{
StopAudio();
mAudioBuffer->~CAudioBuffer();
free( mAudioBuffer );
}
void AudioOutput::SetFrequency( u32 frequency )
{
mFrequency = frequency;
}
void AudioOutput::AddBuffer( u8 *start, u32 length )
{
if (length == 0)
return;
if (!mAudioPlaying)
StartAudio();
u32 num_samples = length / sizeof( Sample );
u32 output_freq = DESIRED_OUTPUT_FREQUENCY;
if (gAudioRateMatch)
{
if (gSoundSync > 88200) output_freq = 88200; //limit upper rate
else if (gSoundSync < DESIRED_OUTPUT_FREQUENCY) output_freq = DESIRED_OUTPUT_FREQUENCY; //limit lower rate
}
switch( gAudioPluginEnabled )
{
case APM_DISABLED:
break;
case APM_ENABLED_ASYNC:
mAudioBuffer->AddSamples( reinterpret_cast< const Sample * >( start ), num_samples, mFrequency, output_freq );
break;
case APM_ENABLED_SYNC:
mAudioBuffer->AddSamples( reinterpret_cast< const Sample * >( start ), num_samples, mFrequency, output_freq );
break;
}
}
void AudioOutput::StartAudio()
{
if (mAudioPlaying)
return;
mAudioPlaying = true;
ac = this;
AudioInit();
}
void AudioOutput::StopAudio()
{
if (!mAudioPlaying)
return;
mAudioPlaying = false;
AudioExit();
}