542 lines
13 KiB
C
542 lines
13 KiB
C
/*
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Copyright (C) 1996-2001 Id Software, Inc.
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Copyright (C) 2010-2011 O. Sezer <sezero@users.sourceforge.net>
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Copyright (C) 2010-2014 QuakeSpasm developers
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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.
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See the 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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// snd_mix.c -- portable code to mix sounds for snd_dma.c
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#include "q_defs.h"
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#define PAINTBUFFER_SIZE 2048
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portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
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int32_t snd_scaletable[32][256];
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int32_t *snd_p, snd_linear_count;
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int16_t *snd_out;
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static int32_t snd_vol;
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static void Snd_WriteLinearBlastStereo16(void)
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{
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int32_t i;
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int32_t val;
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for(i = 0; i < snd_linear_count; i += 2)
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{
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val = snd_p[i] / 256;
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if(val > 0x7fff)
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snd_out[i] = 0x7fff;
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else if(val < (int16_t)0x8000)
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snd_out[i] = (int16_t)0x8000;
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else
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snd_out[i] = val;
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val = snd_p[i + 1] / 256;
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if(val > 0x7fff)
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snd_out[i + 1] = 0x7fff;
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else if(val < (int16_t)0x8000)
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snd_out[i + 1] = (int16_t)0x8000;
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else
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snd_out[i + 1] = val;
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}
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}
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static void S_TransferStereo16(int32_t endtime)
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{
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int32_t lpos;
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int32_t lpaintedtime;
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snd_p = (int32_t *) paintbuffer;
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lpaintedtime = paintedtime;
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while(lpaintedtime < endtime)
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{
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// handle recirculating buffer issues
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lpos = lpaintedtime & ((shm->samples >> 1) - 1);
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snd_out = (int16_t *)shm->buffer + (lpos << 1);
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snd_linear_count = (shm->samples >> 1) - lpos;
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if(lpaintedtime + snd_linear_count > endtime)
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snd_linear_count = endtime - lpaintedtime;
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snd_linear_count <<= 1;
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// write a linear blast of samples
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Snd_WriteLinearBlastStereo16();
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snd_p += snd_linear_count;
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lpaintedtime += (snd_linear_count >> 1);
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}
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}
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static void S_TransferPaintBuffer(int32_t endtime)
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{
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int32_t out_idx, out_mask;
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int32_t count, step, val;
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int32_t *p;
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if(shm->samplebits == 16 && shm->channels == 2)
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{
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S_TransferStereo16(endtime);
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return;
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}
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p = (int32_t *) paintbuffer;
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count = (endtime - paintedtime) * shm->channels;
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out_mask = shm->samples - 1;
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out_idx = paintedtime * shm->channels & out_mask;
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step = 3 - shm->channels;
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if(shm->samplebits == 16)
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{
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int16_t *out = (int16_t *)shm->buffer;
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while(count--)
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{
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val = *p / 256;
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p += step;
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if(val > 0x7fff)
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val = 0x7fff;
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else if(val < (int16_t)0x8000)
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val = (int16_t)0x8000;
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out[out_idx] = val;
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out_idx = (out_idx + 1) & out_mask;
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}
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}
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else if(shm->samplebits == 8 && !shm->signed8)
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{
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uint8_t *out = shm->buffer;
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while(count--)
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{
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val = *p / 256;
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p += step;
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if(val > 0x7fff)
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val = 0x7fff;
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else if(val < (int16_t)0x8000)
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val = (int16_t)0x8000;
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out[out_idx] = (val / 256) + 128;
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out_idx = (out_idx + 1) & out_mask;
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}
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}
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else if(shm->samplebits == 8) /* S8 format, e.g. with Amiga AHI */
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{
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int8_t *out = (int8_t *) shm->buffer;
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while(count--)
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{
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val = *p / 256;
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p += step;
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if(val > 0x7fff)
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val = 0x7fff;
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else if(val < (int16_t)0x8000)
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val = (int16_t)0x8000;
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out[out_idx] = (val / 256);
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out_idx = (out_idx + 1) & out_mask;
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}
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}
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}
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/*
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==============
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S_MakeBlackmanWindowKernel
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Makes a lowpass filter kernel, from equation 16-4 in
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"The Scientist and Engineer's Guide to Digital Signal Processing"
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M is the kernel size (not counting the center point), must be even
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kernel has room for M+1 floats
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f_c is the filter cutoff frequency, as a fraction of the samplerate
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==============
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*/
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static void S_MakeBlackmanWindowKernel(float *kernel, int32_t M, float f_c)
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{
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int32_t i;
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for(i = 0; i <= M; i++)
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{
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if(i == M / 2)
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{
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kernel[i] = 2 * PI * f_c;
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}
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else
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{
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kernel[i] = (sin(2 * PI * f_c * (i - M / 2.0)) / (i - (M / 2.0))) *
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(0.42 - 0.5 * cos(2 * PI * i / (double)M) +
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0.08 * cos(4 * PI * i / (double)M));
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}
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}
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// normalize the kernel so all of the values sum to 1
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{
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float sum = 0;
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for(i = 0; i <= M; i++)
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{
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sum += kernel[i];
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}
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for(i = 0; i <= M; i++)
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{
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kernel[i] /= sum;
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}
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}
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}
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typedef struct
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{
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float *memory; // kernelsize floats
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float *kernel; // kernelsize floats
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int32_t kernelsize; // M+1, rounded up to be a multiple of 16
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int32_t M; // M value used to make kernel, even
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int32_t parity; // 0-3
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float f_c; // cutoff frequency, [0..1], fraction of sample rate
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} filter_t;
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static void S_UpdateFilter(filter_t *filter, int32_t M, float f_c)
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{
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if(filter->f_c != f_c || filter->M != M)
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{
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if(filter->memory != NULL) free(filter->memory);
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if(filter->kernel != NULL) free(filter->kernel);
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filter->M = M;
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filter->f_c = f_c;
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filter->parity = 0;
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// M + 1 rounded up to the next multiple of 16
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filter->kernelsize = (M + 1) + 16 - ((M + 1) % 16);
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filter->memory = (float *) calloc(filter->kernelsize, sizeof(float));
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filter->kernel = (float *) calloc(filter->kernelsize, sizeof(float));
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S_MakeBlackmanWindowKernel(filter->kernel, M, f_c);
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}
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}
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/*
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==============
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S_ApplyFilter
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Lowpass-filter the given buffer containing 44100Hz audio.
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As an optimization, it decimates the audio to 11025Hz (setting every sample
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position that's not a multiple of 4 to 0), then convoluting with the filter
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kernel is 4x faster, because we can skip 3/4 of the input samples that are
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known to be 0 and skip 3/4 of the filter kernel.
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==============
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*/
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static void S_ApplyFilter(filter_t *filter, int32_t *data, int32_t stride, int32_t count)
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{
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int32_t i, j;
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float *input;
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const int32_t kernelsize = filter->kernelsize;
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const float *kernel = filter->kernel;
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int32_t parity;
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input = (float *) malloc(sizeof(float) * (filter->kernelsize + count));
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// set up the input buffer
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// memory holds the previous filter->kernelsize samples of input.
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memcpy(input, filter->memory, filter->kernelsize * sizeof(float));
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for(i = 0; i < count; i++)
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{
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input[filter->kernelsize + i] = data[i * stride] / (32768.0 * 256.0);
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}
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// copy out the last filter->kernelsize samples to 'memory' for next time
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memcpy(filter->memory, input + count, filter->kernelsize * sizeof(float));
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// apply the filter
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parity = filter->parity;
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for(i = 0; i < count; i++)
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{
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const float *input_plus_i = input + i;
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float val[4] = {0, 0, 0, 0};
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for(j = (4 - parity) % 4; j < kernelsize; j += 16)
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{
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val[0] += kernel[j] * input_plus_i[j];
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val[1] += kernel[j + 4] * input_plus_i[j + 4];
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val[2] += kernel[j + 8] * input_plus_i[j + 8];
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val[3] += kernel[j + 12] * input_plus_i[j + 12];
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}
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// 4.0 factor is to increase volume by 12 dB; this is to make up the
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// volume drop caused by the zero-filling this filter does.
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data[i * stride] = (val[0] + val[1] + val[2] + val[3])
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* (32768.0 * 256.0 * 4.0);
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parity = (parity + 1) % 4;
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}
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filter->parity = parity;
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free(input);
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}
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/*
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==============
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S_LowpassFilter
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lowpass filters 24-bit integer samples in 'data' (stored in 32-bit ints).
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assumes 44100Hz sample rate, and lowpasses at around 5kHz
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memory should be a zero-filled filter_t struct
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==============
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*/
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static void S_LowpassFilter(int32_t *data, int32_t stride, int32_t count,
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filter_t *memory)
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{
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int32_t M;
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float bw, f_c;
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switch((int32_t)snd_filterquality.value)
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{
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case 1:
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M = 126;
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bw = 0.900;
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break;
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case 2:
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M = 150;
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bw = 0.915;
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break;
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case 3:
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M = 174;
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bw = 0.930;
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break;
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case 4:
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M = 198;
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bw = 0.945;
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break;
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case 5:
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default:
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M = 222;
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bw = 0.960;
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break;
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}
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f_c = (bw * 11025 / 2.0) / 44100.0;
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S_UpdateFilter(memory, M, f_c);
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S_ApplyFilter(memory, data, stride, count);
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}
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/*
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===============================================================================
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CHANNEL MIXING
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===============================================================================
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*/
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static void SND_PaintChannelFrom8(channel_t *ch, sfxcache_t *sc, int32_t endtime, int32_t paintbufferstart);
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static void SND_PaintChannelFrom16(channel_t *ch, sfxcache_t *sc, int32_t endtime, int32_t paintbufferstart);
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void S_PaintChannels(int32_t endtime)
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{
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int32_t i;
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int32_t end, ltime, count;
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channel_t *ch;
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sfxcache_t *sc;
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snd_vol = sfxvolume.value * 256;
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while(paintedtime < endtime)
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{
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// if paintbuffer is smaller than DMA buffer
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end = endtime;
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if(endtime - paintedtime > PAINTBUFFER_SIZE)
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end = paintedtime + PAINTBUFFER_SIZE;
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// clear the paint buffer
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memset(paintbuffer, 0, (end - paintedtime) * sizeof(portable_samplepair_t));
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// paint in the channels.
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ch = snd_channels;
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for(i = 0; i < total_channels; i++, ch++)
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{
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if(!ch->sfx)
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continue;
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if(!ch->leftvol && !ch->rightvol)
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continue;
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sc = S_LoadSound(ch->sfx);
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if(!sc)
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continue;
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ltime = paintedtime;
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while(ltime < end)
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{
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// paint up to end
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if(ch->end < end)
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count = ch->end - ltime;
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else
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count = end - ltime;
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if(count > 0)
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{
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// the last param to SND_PaintChannelFrom is the index
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// to start painting to in the paintbuffer, usually 0.
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if(sc->width == 1)
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SND_PaintChannelFrom8(ch, sc, count, ltime - paintedtime);
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else
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SND_PaintChannelFrom16(ch, sc, count, ltime - paintedtime);
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ltime += count;
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}
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// if at end of loop, restart
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if(ltime >= ch->end)
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{
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if(sc->loopstart >= 0)
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{
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ch->pos = sc->loopstart;
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ch->end = ltime + sc->length - ch->pos;
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}
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else
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{
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// channel just stopped
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ch->sfx = NULL;
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break;
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}
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}
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}
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}
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// clip each sample to 0dB, then reduce by 6dB (to leave some headroom for
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// the lowpass filter and the music). the lowpass will smooth out the
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// clipping
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for(i = 0; i < end - paintedtime; i++)
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{
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paintbuffer[i].left = CLAMP(-32768 * 256, paintbuffer[i].left, 32767 * 256) / 2;
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paintbuffer[i].right = CLAMP(-32768 * 256, paintbuffer[i].right, 32767 * 256) / 2;
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}
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// apply a lowpass filter
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if(sndspeed.value == 11025 && shm->speed == 44100)
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{
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static filter_t memory_l, memory_r;
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S_LowpassFilter((int32_t *)paintbuffer, 2, end - paintedtime, &memory_l);
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S_LowpassFilter(((int32_t *)paintbuffer) + 1, 2, end - paintedtime, &memory_r);
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}
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// paint in the music
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if(s_rawend >= paintedtime)
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{
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// copy from the streaming sound source
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int32_t s;
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int32_t stop;
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stop = (end < s_rawend) ? end : s_rawend;
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for(i = paintedtime; i < stop; i++)
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{
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s = i & (MAX_RAW_SAMPLES - 1);
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// lower music by 6db to match sfx
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paintbuffer[i - paintedtime].left += s_rawsamples[s].left / 2;
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paintbuffer[i - paintedtime].right += s_rawsamples[s].right / 2;
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}
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// if (i != end)
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// Con_Printf ("partial stream\n");
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// else
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// Con_Printf ("full stream\n");
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}
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// transfer out according to DMA format
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S_TransferPaintBuffer(end);
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paintedtime = end;
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}
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}
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void SND_InitScaletable(void)
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{
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int32_t i, j;
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int32_t scale;
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for(i = 0; i < 32; i++)
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{
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scale = i * 8 * 256 * sfxvolume.value;
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for(j = 0; j < 256; j++)
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{
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/* When compiling with gcc-4.1.0 at optimisations O1 and
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higher, the tricky int8_t type conversion is not
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guaranteed. Therefore we explicity calculate the signed
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value from the index as required. From Kevin Shanahan.
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See: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=26719
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*/
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// snd_scaletable[i][j] = ((int8_t)j) * scale;
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snd_scaletable[i][j] = ((j < 128) ? j : j - 256) * scale;
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}
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}
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}
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static void SND_PaintChannelFrom8(channel_t *ch, sfxcache_t *sc, int32_t count, int32_t paintbufferstart)
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{
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int32_t data;
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int32_t *lscale, *rscale;
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uint8_t *sfx;
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int32_t i;
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if(ch->leftvol > 255)
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ch->leftvol = 255;
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if(ch->rightvol > 255)
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ch->rightvol = 255;
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lscale = snd_scaletable[ch->leftvol >> 3];
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rscale = snd_scaletable[ch->rightvol >> 3];
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sfx = (uint8_t *)sc->data + ch->pos;
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for(i = 0; i < count; i++)
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{
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data = sfx[i];
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paintbuffer[paintbufferstart + i].left += lscale[data];
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paintbuffer[paintbufferstart + i].right += rscale[data];
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}
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ch->pos += count;
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}
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static void SND_PaintChannelFrom16(channel_t *ch, sfxcache_t *sc, int32_t count, int32_t paintbufferstart)
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{
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int32_t data;
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int32_t left, right;
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int32_t leftvol, rightvol;
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int16_t *sfx;
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int32_t i;
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leftvol = ch->leftvol * snd_vol;
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rightvol = ch->rightvol * snd_vol;
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leftvol /= 256;
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rightvol /= 256;
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sfx = (int16_t *)sc->data + ch->pos;
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for(i = 0; i < count; i++)
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{
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data = sfx[i];
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// this was causing integer overflow as observed in quakespasm
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// with the warpspasm mod moved >>8 to left/right volume above.
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// left = (data * leftvol) >> 8;
|
|
// right = (data * rightvol) >> 8;
|
|
left = data * leftvol;
|
|
right = data * rightvol;
|
|
paintbuffer[paintbufferstart + i].left += left;
|
|
paintbuffer[paintbufferstart + i].right += right;
|
|
}
|
|
|
|
ch->pos += count;
|
|
}
|
|
|