1 files changed, 15 insertions, 10 deletions

diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
index 3e23323f1..c7efae753 100644
--- a/src/audio_core/codec.cpp
+++ b/src/audio_core/codec.cpp
@@ -15,22 +15,25 @@
 
 namespace Codec {
 
-StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, const std::array<s16, 16>& adpcm_coeff, ADPCMState& state) {
+StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count,
+                           const std::array<s16, 16>& adpcm_coeff, ADPCMState& state) {
     // GC-ADPCM with scale factor and variable coefficients.
     // Frames are 8 bytes long containing 14 samples each.
     // Samples are 4 bits (one nibble) long.
 
     constexpr size_t FRAME_LEN = 8;
     constexpr size_t SAMPLES_PER_FRAME = 14;
-    constexpr std::array<int, 16> SIGNED_NIBBLES {{ 0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1 }};
+    constexpr std::array<int, 16> SIGNED_NIBBLES{
+        {0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
 
-    const size_t ret_size = sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
+    const size_t ret_size =
+        sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
     StereoBuffer16 ret(ret_size);
 
-    int yn1 = state.yn1,
-        yn2 = state.yn2;
+    int yn1 = state.yn1, yn2 = state.yn2;
 
-    const size_t NUM_FRAMES = (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
+    const size_t NUM_FRAMES =
+        (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
     for (size_t framei = 0; framei < NUM_FRAMES; framei++) {
         const int frame_header = data[framei * FRAME_LEN];
         const int scale = 1 << (frame_header & 0xF);
@@ -43,7 +46,8 @@ StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, cons
         // Decodes an audio sample. One nibble produces one sample.
         const auto decode_sample = [&](const int nibble) -> s16 {
             const int xn = nibble * scale;
-            // We first transform everything into 11 bit fixed point, perform the second order digital filter, then transform back.
+            // We first transform everything into 11 bit fixed point, perform the second order
+            // digital filter, then transform back.
             // 0x400 == 0.5 in 11 bit fixed point.
             // Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
             int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
@@ -82,7 +86,8 @@ static s16 SignExtendS8(u8 x) {
     return static_cast<s16>(static_cast<s8>(x));
 }
 
-StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count) {
+StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data,
+                          const size_t sample_count) {
     ASSERT(num_channels == 1 || num_channels == 2);
 
     StereoBuffer16 ret(sample_count);
@@ -101,7 +106,8 @@ StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, con
     return ret;
 }
 
-StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, const size_t sample_count) {
+StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
+                           const size_t sample_count) {
     ASSERT(num_channels == 1 || num_channels == 2);
 
     StereoBuffer16 ret(sample_count);
@@ -118,5 +124,4 @@ StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, co
 
     return ret;
 }
-
 };