forked from ShuriZma/suyu
Merge pull request #1877 from heapo/audio_interp
Perf: Avoid (expensive) audio interpolation when sample rates already match
This commit is contained in:
commit
2c6679bb01
|
@ -54,8 +54,9 @@ std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
|
||||||
double l = 0.0;
|
double l = 0.0;
|
||||||
double r = 0.0;
|
double r = 0.0;
|
||||||
for (std::size_t j = 0; j < h.size(); j++) {
|
for (std::size_t j = 0; j < h.size(); j++) {
|
||||||
l += Lanczos(taps, pos + j - taps + 1) * h[j][0];
|
const double lanczos_calc = Lanczos(taps, pos + j - taps + 1);
|
||||||
r += Lanczos(taps, pos + j - taps + 1) * h[j][1];
|
l += lanczos_calc * h[j][0];
|
||||||
|
r += lanczos_calc * h[j][1];
|
||||||
}
|
}
|
||||||
output.emplace_back(static_cast<s16>(std::clamp(l, -32768.0, 32767.0)));
|
output.emplace_back(static_cast<s16>(std::clamp(l, -32768.0, 32767.0)));
|
||||||
output.emplace_back(static_cast<s16>(std::clamp(r, -32768.0, 32767.0)));
|
output.emplace_back(static_cast<s16>(std::clamp(r, -32768.0, 32767.0)));
|
||||||
|
|
|
@ -285,8 +285,11 @@ void AudioRenderer::VoiceState::RefreshBuffer() {
|
||||||
break;
|
break;
|
||||||
}
|
}
|
||||||
|
|
||||||
samples =
|
// Only interpolate when necessary, expensive.
|
||||||
Interpolate(interp_state, std::move(samples), GetInfo().sample_rate, STREAM_SAMPLE_RATE);
|
if (GetInfo().sample_rate != STREAM_SAMPLE_RATE) {
|
||||||
|
samples = Interpolate(interp_state, std::move(samples), GetInfo().sample_rate,
|
||||||
|
STREAM_SAMPLE_RATE);
|
||||||
|
}
|
||||||
|
|
||||||
is_refresh_pending = false;
|
is_refresh_pending = false;
|
||||||
}
|
}
|
||||||
|
|
Loading…
Reference in New Issue