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-rw-r--r--src/audio_core/codec.cpp2
-rw-r--r--src/audio_core/hle/dsp.cpp3
-rw-r--r--src/audio_core/hle/dsp.h72
-rw-r--r--src/audio_core/hle/filter.cpp2
-rw-r--r--src/audio_core/hle/mixers.cpp5
-rw-r--r--src/audio_core/hle/pipe.cpp10
-rw-r--r--src/audio_core/hle/pipe.h14
-rw-r--r--src/audio_core/hle/source.cpp44
-rw-r--r--src/audio_core/interpolate.cpp20
-rw-r--r--src/audio_core/interpolate.h4
-rw-r--r--src/audio_core/null_sink.h3
-rw-r--r--src/audio_core/sink.h4
-rw-r--r--src/audio_core/sink_details.h3
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;
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-24 21:26:43 +0100