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Ensure low pass audio filter is applied correctly when mono output is selected

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This commit is contained in:
jdgleaver 2018-11-26 10:55:08 +00:00
parent f4007769c2
commit b61ff4888c
1 changed files with 68 additions and 31 deletions
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99
pico/sound/sound.c
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@ -32,6 +32,69 @@ extern int *sn76496_regs;
static int32_t lpf_lp;
static int32_t lpf_rp;
static void low_pass_filter_stereo(int *buf32, int length)
{
int samples = length;
int *out32 = buf32;
// Restore previous samples
int32_t lpf_l = lpf_lp;
int32_t lpf_r = lpf_rp;
// Single-pole low-pass filter (6 dB/octave)
int32_t factor_a = PicoIn.sndFilterRange;
int32_t factor_b = 0x10000 - factor_a;
do
{
// Apply low-pass filter
lpf_l = (lpf_l * factor_a) + (out32[0] * factor_b);
lpf_r = (lpf_r * factor_a) + (out32[1] * factor_b);
// 16.16 fixed point
lpf_l >>= 16;
lpf_r >>= 16;
// Update sound buffer
*out32++ = lpf_l;
*out32++ = lpf_r;
}
while (--samples);
// Save last samples for next frame
lpf_lp = lpf_l;
lpf_rp = lpf_r;
}
static void low_pass_filter_mono(int *buf32, int length)
{
int samples = length;
int *out32 = buf32;
// Restore previous sample
int32_t lpf_l = lpf_lp;
// Single-pole low-pass filter (6 dB/octave)
int32_t factor_a = PicoIn.sndFilterRange;
int32_t factor_b = 0x10000 - factor_a;
do
{
// Apply low-pass filter
lpf_l = (lpf_l * factor_a) + (out32[0] * factor_b);
// 16.16 fixed point
lpf_l >>= 16;
// Update sound buffer
*out32++ = lpf_l;
}
while (--samples);
// Save last sample for next frame
lpf_lp = lpf_l;
}
void (*low_pass_filter)(int *buf32, int length) = low_pass_filter_stereo;
static void dac_recalculate(void)
{
int lines = Pico.m.pal ? 313 : 262;
@ -139,6 +202,9 @@ void PsndRerate(int preserve_state)
// set mixer
PsndMix_32_to_16l = (PicoIn.opt & POPT_EN_STEREO) ? mix_32_to_16l_stereo : mix_32_to_16_mono;
// set low pass filter
low_pass_filter = (PicoIn.opt & POPT_EN_STEREO) ? low_pass_filter_stereo : low_pass_filter_mono;
if (PicoIn.AHW & PAHW_PICO)
PicoReratePico();
}
@ -326,37 +392,8 @@ static int PsndRender(int offset, int length)
p32x_pwm_update(buf32, length, stereo);
// Apply low pass filter, if required
if (PicoIn.sndFilter == 1)
{
int samples = length;
int *out32 = buf32;
// Restore previous samples
int32_t lpf_l = lpf_lp;
int32_t lpf_r = lpf_rp;
// Single-pole low-pass filter (6 dB/octave)
int32_t factor_a = PicoIn.sndFilterRange;
int32_t factor_b = 0x10000 - factor_a;
do
{
// Apply low-pass filter
lpf_l = (lpf_l * factor_a) + (out32[0] * factor_b);
lpf_r = (lpf_r * factor_a) + (out32[1] * factor_b);
// 16.16 fixed point
lpf_l >>= 16;
lpf_r >>= 16;
// Update sound buffer
*out32++ = lpf_l;
*out32++ = lpf_r;
}
while (--samples);
// Save last samples for next frame
lpf_lp = lpf_l;
lpf_rp = lpf_r;
if (PicoIn.sndFilter == 1) {
low_pass_filter(buf32, length);
}
// convert + limit to normal 16bit output