Merge pull request #4310 from ogniK5377/apollo-1-prod

audio_core: Apollo Part 1, AudioRenderer refactor

2 files changed, 35 insertions, 0 deletions

diff --git a/src/audio_core/algorithm/interpolate.cpp b/src/audio_core/algorithm/interpolate.cpp
index 49ab9d3e1..689a54508 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 ab1a31754..d534077af 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