13 files changed, 110 insertions, 76 deletions

diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
index c7efae753..4edfe9be0 100644
--- a/src/audio_core/codec.cpp
+++ b/src/audio_core/codec.cpp
@@ -23,7 +23,7 @@ StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count,
 
     constexpr size_t FRAME_LEN = 8;
     constexpr size_t SAMPLES_PER_FRAME = 14;
-    constexpr std::array<int, 16> SIGNED_NIBBLES{
+    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 =
diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp
index 5c8afa111..aaa3a280f 100644
--- a/src/audio_core/hle/dsp.cpp
+++ b/src/audio_core/hle/dsp.cpp
@@ -49,7 +49,8 @@ static SharedMemory& WriteRegion() {
 static std::array<Source, num_sources> sources = {
     Source(0),  Source(1),  Source(2),  Source(3),  Source(4),  Source(5),  Source(6),  Source(7),
     Source(8),  Source(9),  Source(10), Source(11), Source(12), Source(13), Source(14), Source(15),
-    Source(16), Source(17), Source(18), Source(19), Source(20), Source(21), Source(22), Source(23)};
+    Source(16), Source(17), Source(18), Source(19), Source(20), Source(21), Source(22), Source(23),
+};
 static Mixers mixers;
 
 static StereoFrame16 GenerateCurrentFrame() {
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h
index 5b216eb87..f4c4b01e2 100644
--- a/src/audio_core/hle/dsp.h
+++ b/src/audio_core/hle/dsp.h
@@ -23,16 +23,15 @@ class Sink;
 namespace DSP {
 namespace HLE {
 
-// The application-accessible region of DSP memory consists of two parts.
-// Both are marked as IO and have Read/Write permissions.
+// The application-accessible region of DSP memory consists of two parts. Both are marked as IO and
+// have Read/Write permissions.
 //
 // First Region:  0x1FF50000 (Size: 0x8000)
 // Second Region: 0x1FF70000 (Size: 0x8000)
 //
 // The DSP reads from each region alternately based on the frame counter for each region much like a
 // double-buffer. The frame counter is located as the very last u16 of each region and is
-// incremented
-// each audio tick.
+// incremented each audio tick.
 
 constexpr VAddr region0_base = 0x1FF50000;
 constexpr VAddr region1_base = 0x1FF70000;
@@ -92,14 +91,12 @@ static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivial
 //    See also: DSP::HLE::PipeRead.
 //
 // Note that the above addresses do vary slightly between audio firmwares observed; the addresses
-// are
-// not fixed in stone. The addresses above are only an examplar; they're what this implementation
-// does and provides to applications.
+// are not fixed in stone. The addresses above are only an examplar; they're what this
+// implementation does and provides to applications.
 //
 // Application requests the DSP service to convert DSP addresses into ARM11 virtual addresses using
-// the
-// ConvertProcessAddressFromDspDram service call. Applications seem to derive the addresses for the
-// second region via:
+// the ConvertProcessAddressFromDspDram service call. Applications seem to derive the addresses for
+// the second region via:
 //     second_region_dsp_addr = first_region_dsp_addr | 0x10000
 //
 // Applications maintain most of its own audio state, the memory region is used mainly for
@@ -107,7 +104,7 @@ static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivial
 //
 // In the documentation below, filter and effect transfer functions are specified in the z domain.
 // (If you are more familiar with the Laplace transform, z = exp(sT). The z domain is the digital
-//  frequency domain, just like how the s domain is the analog frequency domain.)
+// frequency domain, just like how the s domain is the analog frequency domain.)
 
 #define INSERT_PADDING_DSPWORDS(num_words) INSERT_PADDING_BYTES(2 * (num_words))
 
@@ -137,8 +134,8 @@ struct SourceConfiguration {
             BitField<0, 1, u32_le> format_dirty;
             BitField<1, 1, u32_le> mono_or_stereo_dirty;
             BitField<2, 1, u32_le> adpcm_coefficients_dirty;
-            BitField<3, 1, u32_le>
-                partial_embedded_buffer_dirty; ///< Tends to be set when a looped buffer is queued.
+            /// Tends to be set when a looped buffer is queued.
+            BitField<3, 1, u32_le> partial_embedded_buffer_dirty;
             BitField<4, 1, u32_le> partial_reset_flag;
 
             BitField<16, 1, u32_le> enable_dirty;
@@ -146,8 +143,8 @@ struct SourceConfiguration {
             BitField<18, 1, u32_le> rate_multiplier_dirty;
             BitField<19, 1, u32_le> buffer_queue_dirty;
             BitField<20, 1, u32_le> loop_related_dirty;
-            BitField<21, 1, u32_le>
-                play_position_dirty; ///< Tends to also be set when embedded buffer is updated.
+            /// Tends to also be set when embedded buffer is updated.
+            BitField<21, 1, u32_le> play_position_dirty;
             BitField<22, 1, u32_le> filters_enabled_dirty;
             BitField<23, 1, u32_le> simple_filter_dirty;
             BitField<24, 1, u32_le> biquad_filter_dirty;
@@ -162,9 +159,9 @@ struct SourceConfiguration {
         // Gain control
 
         /**
-         * Gain is between 0.0-1.0. This determines how much will this source appear on
-         * each of the 12 channels that feed into the intermediate mixers.
-         * Each of the three intermediate mixers is fed two left and two right channels.
+         * Gain is between 0.0-1.0. This determines how much will this source appear on each of the
+         * 12 channels that feed into the intermediate mixers. Each of the three intermediate mixers
+         * is fed two left and two right channels.
          */
         float_le gain[3][4];
 
@@ -173,7 +170,11 @@ struct SourceConfiguration {
         /// Multiplier for sample rate. Resampling occurs with the selected interpolation method.
         float_le rate_multiplier;
 
-        enum class InterpolationMode : u8 { Polyphase = 0, Linear = 1, None = 2 };
+        enum class InterpolationMode : u8 {
+            Polyphase = 0,
+            Linear = 1,
+            None = 2,
+        };
 
         InterpolationMode interpolation_mode;
         INSERT_PADDING_BYTES(1); ///< Interpolation related
@@ -197,8 +198,7 @@ struct SourceConfiguration {
          * The transfer function of this filter is:
          *     H(z) = (b0 + b1 z^-1 + b2 z^-2) / (1 - a1 z^-1 - a2 z^-2)
          * Nintendo chose to negate the feedbackward coefficients. This differs from standard
-         * notation
-         * as in: https://ccrma.stanford.edu/~jos/filters/Direct_Form_I.html
+         * notation as in: https://ccrma.stanford.edu/~jos/filters/Direct_Form_I.html
          * Values are signed fixed point with 14 fractional bits.
          */
         struct BiquadFilter {
@@ -246,8 +246,8 @@ struct SourceConfiguration {
             u8 is_looping;
 
             /// This value is shown in SourceStatus::previous_buffer_id when this buffer has
-            /// finished.
-            /// This allows the emulated application to tell what buffer is currently playing
+            /// finished. This allows the emulated application to tell what buffer is currently
+            /// playing.
             u16_le buffer_id;
 
             INSERT_PADDING_DSPWORDS(1);
@@ -275,9 +275,16 @@ struct SourceConfiguration {
         /// Note a sample takes up different number of bytes in different buffer formats.
         u32_dsp length;
 
-        enum class MonoOrStereo : u16_le { Mono = 1, Stereo = 2 };
+        enum class MonoOrStereo : u16_le {
+            Mono = 1,
+            Stereo = 2,
+        };
 
-        enum class Format : u16_le { PCM8 = 0, PCM16 = 1, ADPCM = 2 };
+        enum class Format : u16_le {
+            PCM8 = 0,
+            PCM16 = 1,
+            ADPCM = 2,
+        };
 
         union {
             u16_le flags1_raw;
@@ -349,12 +356,16 @@ struct DspConfiguration {
     };
 
     /// The DSP has three intermediate audio mixers. This controls the volume level (0.0-1.0) for
-    /// each at the final mixer
+    /// each at the final mixer.
     float_le volume[3];
 
     INSERT_PADDING_DSPWORDS(3);
 
-    enum class OutputFormat : u16_le { Mono = 0, Stereo = 1, Surround = 2 };
+    enum class OutputFormat : u16_le {
+        Mono = 0,
+        Stereo = 1,
+        Surround = 2,
+    };
 
     OutputFormat output_format;
 
@@ -386,9 +397,10 @@ struct DspConfiguration {
         u16_le enable;
         INSERT_PADDING_DSPWORDS(1);
         u16_le outputs;
-        u32_dsp work_buffer_address; ///< The application allocates a block of memory for the DSP to
-                                     /// use as a work buffer.
-        u16_le frame_count;          ///< Frames to delay by
+        /// The application allocates a block of memory for the DSP to use as a work buffer.
+        u32_dsp work_buffer_address;
+        /// Frames to delay by
+        u16_le frame_count;
 
         // Coefficients
         s16_le g; ///< Fixed point with 7 fractional bits
diff --git a/src/audio_core/hle/filter.cpp b/src/audio_core/hle/filter.cpp
index ab8814e59..da2a4684e 100644
--- a/src/audio_core/hle/filter.cpp
+++ b/src/audio_core/hle/filter.cpp
@@ -61,6 +61,7 @@ void SourceFilters::SimpleFilter::Reset() {
 
 void SourceFilters::SimpleFilter::Configure(
     SourceConfiguration::Configuration::SimpleFilter config) {
+
     a1 = config.a1;
     b0 = config.b0;
 }
@@ -91,6 +92,7 @@ void SourceFilters::BiquadFilter::Reset() {
 
 void SourceFilters::BiquadFilter::Configure(
     SourceConfiguration::Configuration::BiquadFilter config) {
+
     a1 = config.a1;
     a2 = config.a2;
     b0 = config.b0;
diff --git a/src/audio_core/hle/mixers.cpp b/src/audio_core/hle/mixers.cpp
index a661a7b27..126f328bc 100644
--- a/src/audio_core/hle/mixers.cpp
+++ b/src/audio_core/hle/mixers.cpp
@@ -77,9 +77,8 @@ void Mixers::ParseConfig(DspConfiguration& config) {
 
     if (config.headphones_connected_dirty) {
         config.headphones_connected_dirty.Assign(0);
-        // Do nothing.
-        // (Note: Whether headphones are connected does affect coefficients used for surround
-        // sound.)
+        // Do nothing. (Note: Whether headphones are connected does affect coefficients used for
+        // surround sound.)
         LOG_TRACE(Audio_DSP, "mixers headphones_connected=%hu", config.headphones_connected);
     }
 
diff --git a/src/audio_core/hle/pipe.cpp b/src/audio_core/hle/pipe.cpp
index fe67d2503..f2b6d6552 100644
--- a/src/audio_core/hle/pipe.cpp
+++ b/src/audio_core/hle/pipe.cpp
@@ -97,7 +97,8 @@ static void AudioPipeWriteStructAddresses() {
         0x8000 + offsetof(SharedMemory, unknown11) / 2,
         0x8000 + offsetof(SharedMemory, unknown12) / 2,
         0x8000 + offsetof(SharedMemory, unknown13) / 2,
-        0x8000 + offsetof(SharedMemory, unknown14) / 2};
+        0x8000 + offsetof(SharedMemory, unknown14) / 2,
+    };
 
     // Begin with a u16 denoting the number of structs.
     WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size()));
@@ -118,7 +119,12 @@ void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
             return;
         }
 
-        enum class StateChange { Initalize = 0, Shutdown = 1, Wakeup = 2, Sleep = 3 };
+        enum class StateChange {
+            Initalize = 0,
+            Shutdown = 1,
+            Wakeup = 2,
+            Sleep = 3,
+        };
 
         // The difference between Initialize and Wakeup is that Input state is maintained
         // when sleeping but isn't when turning it off and on again. (TODO: Implement this.)
diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h
index 73b857a90..6d7fd92ab 100644
--- a/src/audio_core/hle/pipe.h
+++ b/src/audio_core/hle/pipe.h
@@ -15,7 +15,12 @@ namespace HLE {
 /// Reset the pipes by setting pipe positions back to the beginning.
 void ResetPipes();
 
-enum class DspPipe { Debug = 0, Dma = 1, Audio = 2, Binary = 3 };
+enum class DspPipe {
+    Debug = 0,
+    Dma = 1,
+    Audio = 2,
+    Binary = 3,
+};
 constexpr size_t NUM_DSP_PIPE = 8;
 
 /**
@@ -46,7 +51,12 @@ size_t GetPipeReadableSize(DspPipe pipe_number);
  */
 void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer);
 
-enum class DspState { Off, On, Sleeping };
+enum class DspState {
+    Off,
+    On,
+    Sleeping,
+};
+
 /// Get the state of the DSP
 DspState GetDspState();
 
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
index fad0ce2ad..249acc449 100644
--- a/src/audio_core/hle/source.cpp
+++ b/src/audio_core/hle/source.cpp
@@ -163,16 +163,18 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config,
 
     if (config.embedded_buffer_dirty) {
         config.embedded_buffer_dirty.Assign(0);
-        state.input_queue.emplace(Buffer{config.physical_address,
-                                         config.length,
-                                         static_cast<u8>(config.adpcm_ps),
-                                         {config.adpcm_yn[0], config.adpcm_yn[1]},
-                                         config.adpcm_dirty.ToBool(),
-                                         config.is_looping.ToBool(),
-                                         config.buffer_id,
-                                         state.mono_or_stereo,
-                                         state.format,
-                                         false});
+        state.input_queue.emplace(Buffer{
+            config.physical_address,
+            config.length,
+            static_cast<u8>(config.adpcm_ps),
+            {config.adpcm_yn[0], config.adpcm_yn[1]},
+            config.adpcm_dirty.ToBool(),
+            config.is_looping.ToBool(),
+            config.buffer_id,
+            state.mono_or_stereo,
+            state.format,
+            false,
+        });
         LOG_TRACE(Audio_DSP, "enqueuing embedded addr=0x%08x len=%u id=%hu",
                   config.physical_address, config.length, config.buffer_id);
     }
@@ -182,16 +184,18 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config,
         for (size_t i = 0; i < 4; i++) {
             if (config.buffers_dirty & (1 << i)) {
                 const auto& b = config.buffers[i];
-                state.input_queue.emplace(Buffer{b.physical_address,
-                                                 b.length,
-                                                 static_cast<u8>(b.adpcm_ps),
-                                                 {b.adpcm_yn[0], b.adpcm_yn[1]},
-                                                 b.adpcm_dirty != 0,
-                                                 b.is_looping != 0,
-                                                 b.buffer_id,
-                                                 state.mono_or_stereo,
-                                                 state.format,
-                                                 true});
+                state.input_queue.emplace(Buffer{
+                    b.physical_address,
+                    b.length,
+                    static_cast<u8>(b.adpcm_ps),
+                    {b.adpcm_yn[0], b.adpcm_yn[1]},
+                    b.adpcm_dirty != 0,
+                    b.is_looping != 0,
+                    b.buffer_id,
+                    state.mono_or_stereo,
+                    state.format,
+                    true,
+                });
                 LOG_TRACE(Audio_DSP, "enqueuing queued %zu addr=0x%08x len=%u id=%hu", i,
                           b.physical_address, b.length, b.buffer_id);
             }
diff --git a/src/audio_core/interpolate.cpp b/src/audio_core/interpolate.cpp
index 7751c545d..cb1c58a67 100644
--- a/src/audio_core/interpolate.cpp
+++ b/src/audio_core/interpolate.cpp
@@ -71,15 +71,17 @@ StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multip
 
 StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
     // Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
-    return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0,
-                                                             const auto& x1, const auto& x2) {
-        // This is a saturated subtraction. (Verified by black-box fuzzing.)
-        s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
-        s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
-
-        return std::array<s16, 2>{static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
-                                  static_cast<s16>(x0[1] + fraction * delta1 / scale_factor)};
-    });
+    return StepOverSamples(state, input, rate_multiplier,
+                           [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
+                               // This is a saturated subtraction. (Verified by black-box fuzzing.)
+                               s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
+                               s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
+
+                               return std::array<s16, 2>{
+                                   static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
+                                   static_cast<s16>(x0[1] + fraction * delta1 / scale_factor),
+                               };
+                           });
 }
 
 } // namespace AudioInterp
diff --git a/src/audio_core/interpolate.h b/src/audio_core/interpolate.h
index 99e5b9657..2d2e60311 100644
--- a/src/audio_core/interpolate.h
+++ b/src/audio_core/interpolate.h
@@ -25,7 +25,7 @@ struct State {
  * @param input Input buffer.
  * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
  *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
- * performs upsampling.
+ *                        performs upsampling.
  * @return The resampled audio buffer.
  */
 StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
@@ -35,7 +35,7 @@ StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multip
  * @param input Input buffer.
  * @param rate_multiplier Stretch factor. Must be a positive non-zero value.
  *                        rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
- * performs upsampling.
+ *                        performs upsampling.
  * @return The resampled audio buffer.
  */
 StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);
diff --git a/src/audio_core/null_sink.h b/src/audio_core/null_sink.h
index b82cd3b9a..9931c4778 100644
--- a/src/audio_core/null_sink.h
+++ b/src/audio_core/null_sink.h
@@ -19,8 +19,7 @@ public:
         return native_sample_rate;
     }
 
-    void EnqueueSamples(const s16*, size_t) override {
-    }
+    void EnqueueSamples(const s16*, size_t) override {}
 
     size_t SamplesInQueue() const override {
         return 0;
diff --git a/src/audio_core/sink.h b/src/audio_core/sink.h
index c938e87d2..f5ce55a6b 100644
--- a/src/audio_core/sink.h
+++ b/src/audio_core/sink.h
@@ -12,8 +12,8 @@ namespace AudioCore {
 
 /**
  * This class is an interface for an audio sink. An audio sink accepts samples in stereo signed
- * PCM16 format to be output.
- * Sinks *do not* handle resampling and expect the correct sample rate. They are dumb outputs.
+ * PCM16 format to be output. Sinks *do not* handle resampling and expect the correct sample rate.
+ * They are dumb outputs.
  */
 class Sink {
 public:
diff --git a/src/audio_core/sink_details.h b/src/audio_core/sink_details.h
index 34110c97a..4b30cf835 100644
--- a/src/audio_core/sink_details.h
+++ b/src/audio_core/sink_details.h
@@ -14,8 +14,7 @@ class Sink;
 
 struct SinkDetails {
     SinkDetails(const char* id_, std::function<std::unique_ptr<Sink>()> factory_)
-        : id(id_), factory(factory_) {
-    }
+        : id(id_), factory(factory_) {}
 
     /// Name for this sink.
     const char* id;