/* 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 "BuildOptions.h" #include "Config/ConfigOptions.h" #include "Debug/DBGConsole.h" #include "HLEAudio/AudioBuffer.h" #include "System/Thread.h" #ifdef DAEDALUS_PSP #include "SysPSP/Utility/CacheUtil.h" #include #include #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 // static FILE * fh = nullptr; // if( !fh ) //{ // fh = fopen( "audio_in.raw", "wb" ); // } // fwrite( samples, sizeof( Sample ), num_samples, fh ); // fflush( fh ); // 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--; } // static FILE * fh = nullptr; // if( !fh ) //{ // fh = fopen( "audio_out.raw", "wb" ); // } // fwrite( samples, sizeof( Sample ), (num_samples-samples_required), fh ); // fflush( fh ); 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; }