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Merge pull request #3880 from superctr/zsg2_cutoff

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zsg2.cpp: minor improvements [nw]
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R. Belmont 2018-08-21 17:04:05 -04:00 committed by GitHub
commit baedd0ca81
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1 changed files with 8 additions and 5 deletions
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13
src/devices/sound/zsg2.cpp
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@ -85,7 +85,7 @@
TODO: TODO:
- Filter and ramping behavior might not be perfect. - Filter and ramping behavior might not be perfect.
- sometimes clicking / popping noises. e.g. raycris song #12 in the sound test. - sometimes clicking / popping noises
- memory reads out of range sometimes - memory reads out of range sometimes
*/ */
@ -272,10 +272,10 @@ void zsg2_device::filter_samples(zchan *ch)
for (int i = 0; i < 4; i++) for (int i = 0; i < 4; i++)
{ {
ch->samples[i+1] = raw_samples[i];
ch->emphasis_filter_state += raw_samples[i]-((ch->emphasis_filter_state+EMPHASIS_ROUNDING)>>EMPHASIS_FILTER_SHIFT); ch->emphasis_filter_state += raw_samples[i]-((ch->emphasis_filter_state+EMPHASIS_ROUNDING)>>EMPHASIS_FILTER_SHIFT);
ch->samples[i+1] = ch->emphasis_filter_state >> EMPHASIS_OUTPUT_SHIFT;
int32_t sample = ch->emphasis_filter_state >> EMPHASIS_OUTPUT_SHIFT;
ch->samples[i+1] = std::min<int32_t>(std::max<int32_t>(sample, -32768), 32767);
} }
} }
@ -285,7 +285,6 @@ void zsg2_device::filter_samples(zchan *ch)
void zsg2_device::sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **outputs, int samples) void zsg2_device::sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **outputs, int samples)
{ {
// DSP is programmed to expect 24-bit samples! So we're not limiting to 16-bit here
for (int i = 0; i < samples; i++) for (int i = 0; i < samples; i++)
{ {
int32_t mix[4] = {}; int32_t mix[4] = {};
@ -328,6 +327,10 @@ void zsg2_device::sound_stream_update(sound_stream &stream, stream_sample_t **in
elem.output_filter_state += (sample - (elem.output_filter_state>>16)) * elem.output_cutoff; elem.output_filter_state += (sample - (elem.output_filter_state>>16)) * elem.output_cutoff;
sample = elem.output_filter_state >> 16; sample = elem.output_filter_state >> 16;
// To prevent DC bias, we need to slowly discharge the filter when the output filter cutoff is 0
if(!elem.output_cutoff)
elem.output_filter_state >>= 1;
sample = (sample * elem.vol)>>16; sample = (sample * elem.vol)>>16;
for(int output=0; output<4; output++) for(int output=0; output<4; output++)