daedalus/Source/HLEAudio/AudioBuffer.cpp
2024-08-05 21:15:57 +10:00

210 lines
6.3 KiB
C++

/*
Copyright (C) 2007 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 "Base/Types.h"
#include "Interface/ConfigOptions.h"
#include "Debug/DBGConsole.h"
#include "HLEAudio/AudioBuffer.h"
#include "System/Thread.h"
#include <cstring>
#include <fstream>
#ifdef DAEDALUS_PSP
#include "SysPSP/Utility/CacheUtil.h"
#endif
CAudioBuffer::CAudioBuffer(u32 buffer_size)
: mBufferBegin(new Sample[buffer_size]),
mBufferEnd(mBufferBegin + buffer_size), mReadPtr(mBufferBegin),
mWritePtr(mBufferBegin) {}
CAudioBuffer::~CAudioBuffer() { delete[] mBufferBegin; }
u32 CAudioBuffer::GetNumBufferedSamples() const {
// Todo: Check Cache Routines
// #ifdef DAEDALUS_PSP
// dcache_wbinv_all();
// #endif
// Safe? What if we read mWrite, and then mRead moves to start of buffer?
s32 diff = mWritePtr - mReadPtr;
if (diff < 0) {
diff += (mBufferEnd - mBufferBegin); // Add on buffer length
}
return diff;
}
void CAudioBuffer::AddSamples(const Sample *samples, u32 num_samples,
u32 frequency, u32 output_freq) {
#ifdef DAEDALUS_ENABLE_ASSERTS
DAEDALUS_ASSERT(frequency <= output_freq, "Input frequency is too high");
#endif
#ifdef DAEDALUS_DEBUG_AUDIO
std::ofstream fh;
if (!fh.is_open())
{
fh.open("audio_in.raw", std::ios::binary);
fh.write(reinterpret_cast<const char*>(samples), sizeof(Sample) * num_samples);
fh.flush();
}
#endif
// clear the Cache
#ifdef DAEDALUS_PSP
// sceKernelDcacheWritebackInvalidateAll();
#endif
const Sample *read_ptr(
mReadPtr); // No need to invalidate, as this is uncached/volatile
Sample *write_ptr(mWritePtr);
//
// 'r' is the number of input samples we progress through for each output
//sample. 's' keeps track of how far between the current two input samples we
//are. We increment it by 'r' for each output sample we generate. When it
//reaches 1.0, we know we've hit the next sample, so we increment in_idx and
//reduce s by 1.0 (to keep it in the range 0.0 .. 1.0) Principle is the same
//but rewritten to integer mode (faster & less ASM) //Corn
const s32 r = (frequency << 12) / output_freq;
s32 s = 0;
u32 in_idx = 0;
u32 output_samples = ((num_samples * output_freq) / frequency) - 1;
for (u32 i = output_samples; i != 0; i--) {
#ifdef DAEDALUS_ENABLE_ASSERTS
DAEDALUS_ASSERT(in_idx + 1 < num_samples,
"Input index out of range - %d / %d", in_idx + 1,
num_samples);
#endif
//#if 0 // 1->Sine tone, 0->Normal
// static float c= 0.0f;
// c += 100.0f / 44100.0f;
// if( c >= 1.0f )
// c-=1.f;
// s16 v( s16( SHRT_MAX * sinf( c * 3.141f*2 ) ) );
// Sample out;
// s16 v = WriteCounter++;
// if( WriteCounter >= MAX_COUNTER )
// {
// printf( "Loop write\n" );
// WriteCounter = 0;
// }
// out.L = out.R = v;
//
// #else
// Resample in integer mode (faster & less ASM code) //Corn
Sample out;
out.L = samples[in_idx].L +
(((samples[in_idx + 1].L - samples[in_idx].L) * s) >> 12);
out.R = samples[in_idx].R +
(((samples[in_idx + 1].R - samples[in_idx].R) * s) >> 12);
s += r;
in_idx += s >> 12;
s &= 4095;
// #endif
write_ptr++;
if (write_ptr >= mBufferEnd)
write_ptr = mBufferBegin;
while (write_ptr == read_ptr) {
// The buffer is full - spin until the read pointer advances.
// Note - spends a lot of time here if program is running
// fast. This loop locks the speed to the playback rate
// as the program winds up waiting for the buffer to empty.
// ToDo: Adjust Audio Frequency/ Look at Turok in this regard.
// We might want to put a Sleep in when executing on the SC?
// Give time to other threads when using SYNC mode.
// ThreadYield();
read_ptr = mReadPtr;
}
*write_ptr = out;
}
// Todo: Check Cache Routines
// Ensure samples array is written back before mWritePtr
// dcache_wbinv_range_unaligned( mBufferBegin, mBufferEnd );
mWritePtr = write_ptr; // Needs cache wbinv
}
u32 CAudioBuffer::Drain(Sample *samples, u32 num_samples) {
// Todo: Check Cache Routines
// Ideally we could just invalidate this range?
// clear the Cache
#ifdef DAEDALUS_PSP
// sceKernelDcacheWritebackInvalidateAll();
#endif
const Sample *read_ptr(mReadPtr); // No need to invalidate, as this is uncached/volatile
const Sample *write_ptr(mWritePtr); //
Sample *out_ptr(samples);
u32 samples_required(num_samples);
while (samples_required > 0) {
// Check if empty
if (read_ptr == write_ptr)
break;
*out_ptr++ = *read_ptr++;
if (read_ptr >= mBufferEnd)
read_ptr = mBufferBegin;
samples_required--;
}
#ifdef DAEDALUS_DEBUG_AUDIO
std::ofstream fh;
if (!fh.is_open())
{
fh.open("audio_out.raw", std::ios::binary);
fh.write(reinterpret_cast<const char*>(samples), sizeof(Sample) * num_samples - samples_required);
fh.flush();
}
#endif
mReadPtr = read_ptr; // No need to invalidate, as this is uncached
// clear the Cache
#ifdef DAEDALUS_PSP
// sceKernelDcacheWritebackInvalidateAll();
#endif
//
// If there weren't enough samples, zero out the buffer
// FIXME(strmnnrmn): Unnecessary on OSX...
//
if (samples_required > 0) {
// DBGConsole_Msg( 0, "Buffer underflow (%d samples)\n", samples_required );
// printf( "Buffer underflow (%d samples)\n", samples_required );
memset(out_ptr, 0, samples_required * sizeof(Sample));
}
// Return the number of samples written
return num_samples - samples_required;
}