mirror of
https://github.com/nzp-team/quakespasm.git
synced 2024-11-22 11:51:04 +00:00
527 lines
13 KiB
C
527 lines
13 KiB
C
/*
|
|
Copyright (C) 1996-2001 Id Software, Inc.
|
|
Copyright (C) 2010-2011 O. Sezer <sezero@users.sourceforge.net>
|
|
Copyright (C) 2010-2014 QuakeSpasm developers
|
|
|
|
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.
|
|
|
|
*/
|
|
// snd_mix.c -- portable code to mix sounds for snd_dma.c
|
|
|
|
#include "quakedef.h"
|
|
|
|
#define PAINTBUFFER_SIZE 2048
|
|
portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
|
|
int snd_scaletable[32][256];
|
|
int *snd_p, snd_linear_count;
|
|
short *snd_out;
|
|
|
|
static int snd_vol;
|
|
|
|
static void Snd_WriteLinearBlastStereo16 (void)
|
|
{
|
|
int i;
|
|
int val;
|
|
|
|
for (i = 0; i < snd_linear_count; i += 2)
|
|
{
|
|
val = snd_p[i] / 256;
|
|
if (val > 0x7fff)
|
|
snd_out[i] = 0x7fff;
|
|
else if (val < (short)0x8000)
|
|
snd_out[i] = (short)0x8000;
|
|
else
|
|
snd_out[i] = val;
|
|
|
|
val = snd_p[i+1] / 256;
|
|
if (val > 0x7fff)
|
|
snd_out[i+1] = 0x7fff;
|
|
else if (val < (short)0x8000)
|
|
snd_out[i+1] = (short)0x8000;
|
|
else
|
|
snd_out[i+1] = val;
|
|
}
|
|
}
|
|
|
|
static void S_TransferStereo16 (int endtime)
|
|
{
|
|
int lpos;
|
|
int lpaintedtime;
|
|
|
|
snd_p = (int *) paintbuffer;
|
|
lpaintedtime = paintedtime;
|
|
|
|
while (lpaintedtime < endtime)
|
|
{
|
|
// handle recirculating buffer issues
|
|
lpos = lpaintedtime & ((shm->samples >> 1) - 1);
|
|
|
|
snd_out = (short *)shm->buffer + (lpos << 1);
|
|
|
|
snd_linear_count = (shm->samples >> 1) - lpos;
|
|
if (lpaintedtime + snd_linear_count > endtime)
|
|
snd_linear_count = endtime - lpaintedtime;
|
|
|
|
snd_linear_count <<= 1;
|
|
|
|
// write a linear blast of samples
|
|
Snd_WriteLinearBlastStereo16 ();
|
|
|
|
snd_p += snd_linear_count;
|
|
lpaintedtime += (snd_linear_count >> 1);
|
|
}
|
|
}
|
|
|
|
static void S_TransferPaintBuffer (int endtime)
|
|
{
|
|
int out_idx, out_mask;
|
|
int count, step, val;
|
|
int *p;
|
|
|
|
if (shm->samplebits == 16 && shm->channels == 2)
|
|
{
|
|
S_TransferStereo16 (endtime);
|
|
return;
|
|
}
|
|
|
|
p = (int *) paintbuffer;
|
|
count = (endtime - paintedtime) * shm->channels;
|
|
out_mask = shm->samples - 1;
|
|
out_idx = paintedtime * shm->channels & out_mask;
|
|
step = 3 - shm->channels;
|
|
|
|
if (shm->samplebits == 16)
|
|
{
|
|
short *out = (short *)shm->buffer;
|
|
while (count--)
|
|
{
|
|
val = *p / 256;
|
|
p+= step;
|
|
if (val > 0x7fff)
|
|
val = 0x7fff;
|
|
else if (val < (short)0x8000)
|
|
val = (short)0x8000;
|
|
out[out_idx] = val;
|
|
out_idx = (out_idx + 1) & out_mask;
|
|
}
|
|
}
|
|
else if (shm->samplebits == 8 && !shm->signed8)
|
|
{
|
|
unsigned char *out = shm->buffer;
|
|
while (count--)
|
|
{
|
|
val = *p / 256;
|
|
p+= step;
|
|
if (val > 0x7fff)
|
|
val = 0x7fff;
|
|
else if (val < (short)0x8000)
|
|
val = (short)0x8000;
|
|
out[out_idx] = (val / 256) + 128;
|
|
out_idx = (out_idx + 1) & out_mask;
|
|
}
|
|
}
|
|
else if (shm->samplebits == 8) /* S8 format, e.g. with Amiga AHI */
|
|
{
|
|
signed char *out = (signed char *) shm->buffer;
|
|
while (count--)
|
|
{
|
|
val = *p / 256;
|
|
p+= step;
|
|
if (val > 0x7fff)
|
|
val = 0x7fff;
|
|
else if (val < (short)0x8000)
|
|
val = (short)0x8000;
|
|
out[out_idx] = (val / 256);
|
|
out_idx = (out_idx + 1) & out_mask;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
==============
|
|
S_MakeBlackmanWindowKernel
|
|
|
|
Makes a lowpass filter kernel, from equation 16-4 in
|
|
"The Scientist and Engineer's Guide to Digital Signal Processing"
|
|
|
|
M is the kernel size (not counting the center point), must be even
|
|
kernel has room for M+1 floats
|
|
f_c is the filter cutoff frequency, as a fraction of the samplerate
|
|
==============
|
|
*/
|
|
static void S_MakeBlackmanWindowKernel(float *kernel, int M, float f_c)
|
|
{
|
|
int i;
|
|
for (i = 0; i <= M; i++)
|
|
{
|
|
if (i == M/2)
|
|
{
|
|
kernel[i] = 2 * M_PI * f_c;
|
|
}
|
|
else
|
|
{
|
|
kernel[i] = ( sin(2 * M_PI * f_c * (i - M/2.0)) / (i - (M/2.0)) )
|
|
* (0.42 - 0.5*cos(2 * M_PI * i / (double)M)
|
|
+ 0.08*cos(4 * M_PI * i / (double)M) );
|
|
}
|
|
}
|
|
|
|
// normalize the kernel so all of the values sum to 1
|
|
{
|
|
float sum = 0;
|
|
for (i = 0; i <= M; i++)
|
|
{
|
|
sum += kernel[i];
|
|
}
|
|
|
|
for (i = 0; i <= M; i++)
|
|
{
|
|
kernel[i] /= sum;
|
|
}
|
|
}
|
|
}
|
|
|
|
typedef struct {
|
|
float *memory; // kernelsize floats
|
|
float *kernel; // kernelsize floats
|
|
int kernelsize; // M+1, rounded up to be a multiple of 16
|
|
int M; // M value used to make kernel, even
|
|
int parity; // 0-3
|
|
float f_c; // cutoff frequency, [0..1], fraction of sample rate
|
|
} filter_t;
|
|
|
|
static void S_UpdateFilter(filter_t *filter, int M, float f_c)
|
|
{
|
|
if (filter->f_c != f_c || filter->M != M)
|
|
{
|
|
if (filter->memory != NULL) free(filter->memory);
|
|
if (filter->kernel != NULL) free(filter->kernel);
|
|
|
|
filter->M = M;
|
|
filter->f_c = f_c;
|
|
|
|
filter->parity = 0;
|
|
// M + 1 rounded up to the next multiple of 16
|
|
filter->kernelsize = (M + 1) + 16 - ((M + 1) % 16);
|
|
filter->memory = (float *) calloc(filter->kernelsize, sizeof(float));
|
|
filter->kernel = (float *) calloc(filter->kernelsize, sizeof(float));
|
|
|
|
S_MakeBlackmanWindowKernel(filter->kernel, M, f_c);
|
|
}
|
|
}
|
|
|
|
/*
|
|
==============
|
|
S_ApplyFilter
|
|
|
|
Lowpass-filter the given buffer containing 44100Hz audio.
|
|
|
|
As an optimization, it decimates the audio to 11025Hz (setting every sample
|
|
position that's not a multiple of 4 to 0), then convoluting with the filter
|
|
kernel is 4x faster, because we can skip 3/4 of the input samples that are
|
|
known to be 0 and skip 3/4 of the filter kernel.
|
|
==============
|
|
*/
|
|
static void S_ApplyFilter(filter_t *filter, int *data, int stride, int count)
|
|
{
|
|
int i, j;
|
|
float *input;
|
|
const int kernelsize = filter->kernelsize;
|
|
const float *kernel = filter->kernel;
|
|
int parity;
|
|
|
|
input = (float *) malloc(sizeof(float) * (filter->kernelsize + count));
|
|
|
|
// set up the input buffer
|
|
// memory holds the previous filter->kernelsize samples of input.
|
|
memcpy(input, filter->memory, filter->kernelsize * sizeof(float));
|
|
|
|
for (i=0; i<count; i++)
|
|
{
|
|
input[filter->kernelsize+i] = data[i * stride] / (32768.0 * 256.0);
|
|
}
|
|
|
|
// copy out the last filter->kernelsize samples to 'memory' for next time
|
|
memcpy(filter->memory, input + count, filter->kernelsize * sizeof(float));
|
|
|
|
// apply the filter
|
|
parity = filter->parity;
|
|
|
|
for (i=0; i<count; i++)
|
|
{
|
|
const float *input_plus_i = input + i;
|
|
float val[4] = {0, 0, 0, 0};
|
|
|
|
for (j = (4 - parity) % 4; j < kernelsize; j+=16)
|
|
{
|
|
val[0] += kernel[j] * input_plus_i[j];
|
|
val[1] += kernel[j+4] * input_plus_i[j+4];
|
|
val[2] += kernel[j+8] * input_plus_i[j+8];
|
|
val[3] += kernel[j+12] * input_plus_i[j+12];
|
|
}
|
|
|
|
// 4.0 factor is to increase volume by 12 dB; this is to make up the
|
|
// volume drop caused by the zero-filling this filter does.
|
|
data[i * stride] = (val[0] + val[1] + val[2] + val[3])
|
|
* (32768.0 * 256.0 * 4.0);
|
|
|
|
parity = (parity + 1) % 4;
|
|
}
|
|
|
|
filter->parity = parity;
|
|
|
|
free(input);
|
|
}
|
|
|
|
/*
|
|
==============
|
|
S_LowpassFilter
|
|
|
|
lowpass filters 24-bit integer samples in 'data' (stored in 32-bit ints).
|
|
assumes 44100Hz sample rate, and lowpasses at around 5kHz
|
|
memory should be a zero-filled filter_t struct
|
|
==============
|
|
*/
|
|
static void S_LowpassFilter(int *data, int stride, int count,
|
|
filter_t *memory)
|
|
{
|
|
int M;
|
|
float bw, f_c;
|
|
|
|
switch ((int)snd_filterquality.value)
|
|
{
|
|
case 1:
|
|
M = 126; bw = 0.900; break;
|
|
case 2:
|
|
M = 150; bw = 0.915; break;
|
|
case 3:
|
|
M = 174; bw = 0.930; break;
|
|
case 4:
|
|
M = 198; bw = 0.945; break;
|
|
case 5:
|
|
default:
|
|
M = 222; bw = 0.960; break;
|
|
}
|
|
|
|
f_c = (bw * 11025 / 2.0) / 44100.0;
|
|
|
|
S_UpdateFilter(memory, M, f_c);
|
|
S_ApplyFilter(memory, data, stride, count);
|
|
}
|
|
|
|
/*
|
|
===============================================================================
|
|
|
|
CHANNEL MIXING
|
|
|
|
===============================================================================
|
|
*/
|
|
|
|
static void SND_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int endtime, int paintbufferstart);
|
|
static void SND_PaintChannelFrom16 (channel_t *ch, sfxcache_t *sc, int endtime, int paintbufferstart);
|
|
|
|
void S_PaintChannels (int endtime)
|
|
{
|
|
int i;
|
|
int end, ltime, count;
|
|
channel_t *ch;
|
|
sfxcache_t *sc;
|
|
|
|
snd_vol = sfxvolume.value * 256;
|
|
|
|
while (paintedtime < endtime)
|
|
{
|
|
// if paintbuffer is smaller than DMA buffer
|
|
end = endtime;
|
|
if (endtime - paintedtime > PAINTBUFFER_SIZE)
|
|
end = paintedtime + PAINTBUFFER_SIZE;
|
|
|
|
// clear the paint buffer
|
|
memset(paintbuffer, 0, (end - paintedtime) * sizeof(portable_samplepair_t));
|
|
|
|
// paint in the channels.
|
|
ch = snd_channels;
|
|
for (i = 0; i < total_channels; i++, ch++)
|
|
{
|
|
if (!ch->sfx)
|
|
continue;
|
|
if (!ch->leftvol && !ch->rightvol)
|
|
continue;
|
|
sc = S_LoadSound (ch->sfx);
|
|
if (!sc)
|
|
continue;
|
|
|
|
ltime = paintedtime;
|
|
|
|
while (ltime < end)
|
|
{ // paint up to end
|
|
if (ch->end < end)
|
|
count = ch->end - ltime;
|
|
else
|
|
count = end - ltime;
|
|
|
|
if (count > 0)
|
|
{
|
|
// the last param to SND_PaintChannelFrom is the index
|
|
// to start painting to in the paintbuffer, usually 0.
|
|
if (sc->width == 1)
|
|
SND_PaintChannelFrom8(ch, sc, count, ltime - paintedtime);
|
|
else
|
|
SND_PaintChannelFrom16(ch, sc, count, ltime - paintedtime);
|
|
|
|
ltime += count;
|
|
}
|
|
|
|
// if at end of loop, restart
|
|
if (ltime >= ch->end)
|
|
{
|
|
if (sc->loopstart >= 0)
|
|
{
|
|
ch->pos = sc->loopstart;
|
|
ch->end = ltime + sc->length - ch->pos;
|
|
}
|
|
else
|
|
{ // channel just stopped
|
|
ch->sfx = NULL;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// clip each sample to 0dB, then reduce by 6dB (to leave some headroom for
|
|
// the lowpass filter and the music). the lowpass will smooth out the
|
|
// clipping
|
|
for (i=0; i<end-paintedtime; i++)
|
|
{
|
|
paintbuffer[i].left = CLAMP(-32768 * 256, paintbuffer[i].left, 32767 * 256) / 2;
|
|
paintbuffer[i].right = CLAMP(-32768 * 256, paintbuffer[i].right, 32767 * 256) / 2;
|
|
}
|
|
|
|
// apply a lowpass filter
|
|
if (sndspeed.value == 11025 && shm->speed == 44100)
|
|
{
|
|
static filter_t memory_l, memory_r;
|
|
S_LowpassFilter((int *)paintbuffer, 2, end - paintedtime, &memory_l);
|
|
S_LowpassFilter(((int *)paintbuffer) + 1, 2, end - paintedtime, &memory_r);
|
|
}
|
|
|
|
// paint in the music
|
|
if (s_rawend >= paintedtime)
|
|
{ // copy from the streaming sound source
|
|
int s;
|
|
int stop;
|
|
|
|
stop = (end < s_rawend) ? end : s_rawend;
|
|
|
|
for (i = paintedtime; i < stop; i++)
|
|
{
|
|
s = i & (MAX_RAW_SAMPLES - 1);
|
|
// lower music by 6db to match sfx
|
|
paintbuffer[i - paintedtime].left += s_rawsamples[s].left / 2;
|
|
paintbuffer[i - paintedtime].right += s_rawsamples[s].right / 2;
|
|
}
|
|
// if (i != end)
|
|
// Con_Printf ("partial stream\n");
|
|
// else
|
|
// Con_Printf ("full stream\n");
|
|
}
|
|
|
|
// transfer out according to DMA format
|
|
S_TransferPaintBuffer(end);
|
|
paintedtime = end;
|
|
}
|
|
}
|
|
|
|
void SND_InitScaletable (void)
|
|
{
|
|
int i, j;
|
|
int scale;
|
|
|
|
for (i = 0; i < 32; i++)
|
|
{
|
|
scale = i * 8 * 256 * sfxvolume.value;
|
|
for (j = 0; j < 256; j++)
|
|
{
|
|
/* When compiling with gcc-4.1.0 at optimisations O1 and
|
|
higher, the tricky signed char type conversion is not
|
|
guaranteed. Therefore we explicity calculate the signed
|
|
value from the index as required. From Kevin Shanahan.
|
|
See: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=26719
|
|
*/
|
|
// snd_scaletable[i][j] = ((signed char)j) * scale;
|
|
snd_scaletable[i][j] = ((j < 128) ? j : j - 256) * scale;
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
static void SND_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int count, int paintbufferstart)
|
|
{
|
|
int data;
|
|
int *lscale, *rscale;
|
|
unsigned char *sfx;
|
|
int i;
|
|
|
|
if (ch->leftvol > 255)
|
|
ch->leftvol = 255;
|
|
if (ch->rightvol > 255)
|
|
ch->rightvol = 255;
|
|
|
|
lscale = snd_scaletable[ch->leftvol >> 3];
|
|
rscale = snd_scaletable[ch->rightvol >> 3];
|
|
sfx = (unsigned char *)sc->data + ch->pos;
|
|
|
|
for (i = 0; i < count; i++)
|
|
{
|
|
data = sfx[i];
|
|
paintbuffer[paintbufferstart + i].left += lscale[data];
|
|
paintbuffer[paintbufferstart + i].right += rscale[data];
|
|
}
|
|
|
|
ch->pos += count;
|
|
}
|
|
|
|
static void SND_PaintChannelFrom16 (channel_t *ch, sfxcache_t *sc, int count, int paintbufferstart)
|
|
{
|
|
int data;
|
|
int left, right;
|
|
int leftvol, rightvol;
|
|
signed short *sfx;
|
|
int i;
|
|
|
|
leftvol = ch->leftvol * snd_vol;
|
|
rightvol = ch->rightvol * snd_vol;
|
|
leftvol /= 256;
|
|
rightvol /= 256;
|
|
sfx = (signed short *)sc->data + ch->pos;
|
|
|
|
for (i = 0; i < count; i++)
|
|
{
|
|
data = sfx[i];
|
|
// this was causing integer overflow as observed in quakespasm
|
|
// with the warpspasm mod moved >>8 to left/right volume above.
|
|
// 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;
|
|
}
|
|
|