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| author | David Marcec <dmarcecguzman@gmail.com> | 2020-07-12 21:59:14 +1000 |
|---|---|---|
| committer | David Marcec <dmarcecguzman@gmail.com> | 2020-07-25 12:39:34 +1000 |
| commit | 380658c21d39cf05ac765a9284da246388cca2a4 (patch) | |
| tree | 1416cd7e9aee96ec40675078d16a8240d410d04b /src/audio_core/algorithm | |
| parent | c73701edeae64e19097ecbfaa6860c86e2a4675b (diff) | |
audio_core: Apollo Part 1, AudioRenderer refactor
Diffstat (limited to 'src/audio_core/algorithm')
| -rw-r--r-- | src/audio_core/algorithm/interpolate.cpp | 32 | ||||
| -rw-r--r-- | src/audio_core/algorithm/interpolate.h | 3 |
2 files changed, 35 insertions, 0 deletions
diff --git a/src/audio_core/algorithm/interpolate.cpp b/src/audio_core/algorithm/interpolate.cpp index 49ab9d3e15..689a54508a 100644 --- a/src/audio_core/algorithm/interpolate.cpp +++ b/src/audio_core/algorithm/interpolate.cpp @@ -197,4 +197,36 @@ std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input, return output; } +void Resample(s32* output, const s32* input, s32 pitch, s32& fraction, std::size_t sample_count) { + const std::array<s16, 512>& lut = [pitch] { + if (pitch > 0xaaaa) { + return curve_lut0; + } + if (pitch <= 0x8000) { + return curve_lut1; + } + return curve_lut2; + }(); + + std::size_t index{}; + + for (std::size_t i = 0; i < sample_count; i++) { + const std::size_t lut_index{(static_cast<std::size_t>(fraction) >> 8) * 4}; + const auto l0 = lut[lut_index + 0]; + const auto l1 = lut[lut_index + 1]; + const auto l2 = lut[lut_index + 2]; + const auto l3 = lut[lut_index + 3]; + + const auto s0 = static_cast<s32>(input[index]); + const auto s1 = static_cast<s32>(input[index + 1]); + const auto s2 = static_cast<s32>(input[index + 2]); + const auto s3 = static_cast<s32>(input[index + 3]); + + output[i] = (l0 * s0 + l1 * s1 + l2 * s2 + l3 * s3) >> 15; + fraction += pitch; + index += (fraction >> 15); + fraction &= 0x7fff; + } +} + } // namespace AudioCore diff --git a/src/audio_core/algorithm/interpolate.h b/src/audio_core/algorithm/interpolate.h index ab1a31754e..d534077afe 100644 --- a/src/audio_core/algorithm/interpolate.h +++ b/src/audio_core/algorithm/interpolate.h @@ -38,4 +38,7 @@ inline std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> return Interpolate(state, std::move(input), ratio); } +/// Nintendo Switchs DSP resampling algorithm. Based on a single channel +void Resample(s32* output, const s32* input, s32 pitch, s32& fraction, std::size_t sample_count); + } // namespace AudioCore |