mirror of
https://github.com/DaedalusX64/daedalus.git
synced 2025-04-02 10:21:48 -04:00
204 lines
6.2 KiB
C++
204 lines
6.2 KiB
C++
/*
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Copyright (C) 2007 StrmnNrmn
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#include "Base/Types.h"
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#include "BuildOptions.h"
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#include "Config/ConfigOptions.h"
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#include "Debug/DBGConsole.h"
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#include "HLEAudio/AudioBuffer.h"
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#include "System/Thread.h"
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#ifdef DAEDALUS_PSP
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#include "SysPSP/Utility/CacheUtil.h"
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#include <pspkernel.h>
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#include <pspsdk.h>
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#endif
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CAudioBuffer::CAudioBuffer(u32 buffer_size)
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: mBufferBegin(new Sample[buffer_size]),
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mBufferEnd(mBufferBegin + buffer_size), mReadPtr(mBufferBegin),
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mWritePtr(mBufferBegin) {}
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CAudioBuffer::~CAudioBuffer() { delete[] mBufferBegin; }
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u32 CAudioBuffer::GetNumBufferedSamples() const {
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// Todo: Check Cache Routines
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// #ifdef DAEDALUS_PSP
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// dcache_wbinv_all();
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// #endif
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// Safe? What if we read mWrite, and then mRead moves to start of buffer?
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s32 diff = mWritePtr - mReadPtr;
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if (diff < 0) {
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diff += (mBufferEnd - mBufferBegin); // Add on buffer length
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}
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return diff;
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}
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void CAudioBuffer::AddSamples(const Sample *samples, u32 num_samples,
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u32 frequency, u32 output_freq) {
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#ifdef DAEDALUS_ENABLE_ASSERTS
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DAEDALUS_ASSERT(frequency <= output_freq, "Input frequency is too high");
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#endif
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// static FILE * fh = nullptr;
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// if( !fh )
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//{
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// fh = fopen( "audio_in.raw", "wb" );
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// }
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// fwrite( samples, sizeof( Sample ), num_samples, fh );
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// fflush( fh );
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// clear the Cache
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#ifdef DAEDALUS_PSP
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// sceKernelDcacheWritebackInvalidateAll();
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#endif
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const Sample *read_ptr(
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mReadPtr); // No need to invalidate, as this is uncached/volatile
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Sample *write_ptr(mWritePtr);
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//
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// 'r' is the number of input samples we progress through for each output
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//sample. 's' keeps track of how far between the current two input samples we
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//are. We increment it by 'r' for each output sample we generate. When it
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//reaches 1.0, we know we've hit the next sample, so we increment in_idx and
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//reduce s by 1.0 (to keep it in the range 0.0 .. 1.0) Principle is the same
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//but rewritten to integer mode (faster & less ASM) //Corn
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const s32 r = (frequency << 12) / output_freq;
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s32 s = 0;
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u32 in_idx = 0;
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u32 output_samples = ((num_samples * output_freq) / frequency) - 1;
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for (u32 i = output_samples; i != 0; i--) {
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#ifdef DAEDALUS_ENABLE_ASSERTS
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DAEDALUS_ASSERT(in_idx + 1 < num_samples,
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"Input index out of range - %d / %d", in_idx + 1,
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num_samples);
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#endif
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//#if 0 // 1->Sine tone, 0->Normal
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// static float c= 0.0f;
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// c += 100.0f / 44100.0f;
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// if( c >= 1.0f )
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// c-=1.f;
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// s16 v( s16( SHRT_MAX * sinf( c * 3.141f*2 ) ) );
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// Sample out;
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// s16 v = WriteCounter++;
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// if( WriteCounter >= MAX_COUNTER )
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// {
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// printf( "Loop write\n" );
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// WriteCounter = 0;
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// }
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// out.L = out.R = v;
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//
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// #else
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// Resample in integer mode (faster & less ASM code) //Corn
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Sample out;
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out.L = samples[in_idx].L +
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(((samples[in_idx + 1].L - samples[in_idx].L) * s) >> 12);
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out.R = samples[in_idx].R +
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(((samples[in_idx + 1].R - samples[in_idx].R) * s) >> 12);
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s += r;
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in_idx += s >> 12;
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s &= 4095;
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// #endif
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write_ptr++;
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if (write_ptr >= mBufferEnd)
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write_ptr = mBufferBegin;
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while (write_ptr == read_ptr) {
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// The buffer is full - spin until the read pointer advances.
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// Note - spends a lot of time here if program is running
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// fast. This loop locks the speed to the playback rate
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// as the program winds up waiting for the buffer to empty.
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// ToDo: Adjust Audio Frequency/ Look at Turok in this regard.
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// We might want to put a Sleep in when executing on the SC?
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// Give time to other threads when using SYNC mode.
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// ThreadYield();
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read_ptr = mReadPtr;
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}
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*write_ptr = out;
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}
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// Todo: Check Cache Routines
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// Ensure samples array is written back before mWritePtr
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// dcache_wbinv_range_unaligned( mBufferBegin, mBufferEnd );
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mWritePtr = write_ptr; // Needs cache wbinv
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}
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u32 CAudioBuffer::Drain(Sample *samples, u32 num_samples) {
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// Todo: Check Cache Routines
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// Ideally we could just invalidate this range?
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// clear the Cache
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#ifdef DAEDALUS_PSP
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// sceKernelDcacheWritebackInvalidateAll();
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#endif
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const Sample *read_ptr(mReadPtr); // No need to invalidate, as this is uncached/volatile
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const Sample *write_ptr(mWritePtr); //
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Sample *out_ptr(samples);
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u32 samples_required(num_samples);
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while (samples_required > 0) {
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// Check if empty
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if (read_ptr == write_ptr)
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break;
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*out_ptr++ = *read_ptr++;
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if (read_ptr >= mBufferEnd)
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read_ptr = mBufferBegin;
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samples_required--;
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}
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// static FILE * fh = nullptr;
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// if( !fh )
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//{
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// fh = fopen( "audio_out.raw", "wb" );
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// }
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// fwrite( samples, sizeof( Sample ), (num_samples-samples_required), fh );
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// fflush( fh );
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mReadPtr = read_ptr; // No need to invalidate, as this is uncached
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// clear the Cache
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#ifdef DAEDALUS_PSP
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// sceKernelDcacheWritebackInvalidateAll();
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#endif
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//
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// If there weren't enough samples, zero out the buffer
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// FIXME(strmnnrmn): Unnecessary on OSX...
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//
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if (samples_required > 0) {
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// DBGConsole_Msg( 0, "Buffer underflow (%d samples)\n", samples_required );
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// printf( "Buffer underflow (%d samples)\n", samples_required );
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memset(out_ptr, 0, samples_required * sizeof(Sample));
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}
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// Return the number of samples written
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return num_samples - samples_required;
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}
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