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Sources: Run clang-format on everything.

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This commit is contained in:
Emmanuel Gil Peyrot
2016-09-18 09:38:01 +09:00
parent fe948af095
commit dc8479928c
386 changed files with 19560 additions and 18080 deletions
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18
src/audio_core/audio_core.cpp
View File

@@ -42,10 +42,18 @@ void Init() {
}
void AddAddressSpace(Kernel::VMManager& address_space) {
auto r0_vma = address_space.MapBackingMemory(DSP::HLE::region0_base, reinterpret_cast<u8*>(&DSP::HLE::g_regions[0]), sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO).MoveFrom();
auto r0_vma = address_space
.MapBackingMemory(DSP::HLE::region0_base,
reinterpret_cast<u8*>(&DSP::HLE::g_regions[0]),
sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO)
.MoveFrom();
address_space.Reprotect(r0_vma, Kernel::VMAPermission::ReadWrite);
auto r1_vma = address_space.MapBackingMemory(DSP::HLE::region1_base, reinterpret_cast<u8*>(&DSP::HLE::g_regions[1]), sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO).MoveFrom();
auto r1_vma = address_space
.MapBackingMemory(DSP::HLE::region1_base,
reinterpret_cast<u8*>(&DSP::HLE::g_regions[1]),
sizeof(DSP::HLE::SharedMemory), Kernel::MemoryState::IO)
.MoveFrom();
address_space.Reprotect(r1_vma, Kernel::VMAPermission::ReadWrite);
}
@@ -58,9 +66,9 @@ void SelectSink(std::string sink_id) {
return;
}
auto iter = std::find_if(g_sink_details.begin(), g_sink_details.end(), [sink_id](const auto& sink_detail) {
return sink_detail.id == sink_id;
});
auto iter =
std::find_if(g_sink_details.begin(), g_sink_details.end(),
[sink_id](const auto& sink_detail) { return sink_detail.id == sink_id; });
if (iter == g_sink_details.end()) {
LOG_ERROR(Audio, "AudioCore::SelectSink given invalid sink_id");

2
src/audio_core/audio_core.h
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@@ -12,7 +12,7 @@ class VMManager;
namespace AudioCore {
constexpr int native_sample_rate = 32728; ///< 32kHz
constexpr int native_sample_rate = 32728; ///< 32kHz
/// Initialise Audio Core
void Init();

25
src/audio_core/codec.cpp
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@@ -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;
}
};

10
src/audio_core/codec.h
View File

@@ -29,7 +29,8 @@ struct ADPCMState {
* @param state ADPCM state, this is updated with new state
* @return Decoded stereo signed PCM16 data, sample_count in length
*/
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);
/**
* @param num_channels Number of channels
@@ -37,7 +38,8 @@ StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, cons
* @param sample_count Length of buffer in terms of number of samples
* @return Decoded stereo signed PCM16 data, sample_count in length
*/
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);
/**
* @param num_channels Number of channels
@@ -45,6 +47,6 @@ StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, con
* @param sample_count Length of buffer in terms of number of samples
* @return Decoded stereo signed PCM16 data, sample_count in length
*/
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);
};

11
src/audio_core/hle/common.h
View File

@@ -13,23 +13,22 @@ namespace DSP {
namespace HLE {
constexpr int num_sources = 24;
constexpr int samples_per_frame = 160; ///< Samples per audio frame at native sample rate
constexpr int samples_per_frame = 160; ///< Samples per audio frame at native sample rate
/// The final output to the speakers is stereo. Preprocessing output in Source is also stereo.
using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>;
/// The DSP is quadraphonic internally.
using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
/**
* This performs the filter operation defined by FilterT::ProcessSample on the frame in-place.
* FilterT::ProcessSample is called sequentially on the samples.
*/
template<typename FrameT, typename FilterT>
template <typename FrameT, typename FilterT>
void FilterFrame(FrameT& frame, FilterT& filter) {
std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const auto& sample) {
return filter.ProcessSample(sample);
});
std::transform(frame.begin(), frame.end(), frame.begin(),
[&filter](const auto& sample) { return filter.ProcessSample(sample); });
}
} // namespace HLE

17
src/audio_core/hle/dsp.cpp
View File

@@ -47,11 +47,9 @@ static SharedMemory& WriteRegion() {
// Audio processing and mixing
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(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)};
static Mixers mixers;
static StereoFrame16 GenerateCurrentFrame() {
@@ -62,14 +60,16 @@ static StereoFrame16 GenerateCurrentFrame() {
// Generate intermediate mixes
for (size_t i = 0; i < num_sources; i++) {
write.source_statuses.status[i] = sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
write.source_statuses.status[i] =
sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
for (size_t mix = 0; mix < 3; mix++) {
sources[i].MixInto(intermediate_mixes[mix], mix);
}
}
// Generate final mix
write.dsp_status = mixers.Tick(read.dsp_configuration, read.intermediate_mix_samples, write.intermediate_mix_samples, intermediate_mixes);
write.dsp_status = mixers.Tick(read.dsp_configuration, read.intermediate_mix_samples,
write.intermediate_mix_samples, intermediate_mixes);
StereoFrame16 output_frame = mixers.GetOutput();
@@ -152,7 +152,8 @@ void Shutdown() {
bool Tick() {
StereoFrame16 current_frame = {};
// TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to shared memory region)
// TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to
// shared memory region)
current_frame = GenerateCurrentFrame();
OutputCurrentFrame(current_frame);

138
src/audio_core/hle/dsp.h
View File

@@ -30,7 +30,8 @@ namespace HLE {
// 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
// double-buffer. The frame counter is located as the very last u16 of each region and is
// incremented
// each audio tick.
constexpr VAddr region0_base = 0x1FF50000;
@@ -56,6 +57,7 @@ struct u32_dsp {
void operator=(u32 new_value) {
storage = Convert(new_value);
}
private:
static constexpr u32 Convert(u32 value) {
return (value << 16) | (value >> 16);
@@ -89,11 +91,13 @@ static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivial
// #: This refers to the order in which they appear in the DspPipe::Audio DSP pipe.
// See also: DSP::HLE::PipeRead.
//
// Note that the above addresses do vary slightly between audio firmwares observed; the addresses are
// 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.
//
// Application requests the DSP service to convert DSP addresses into ARM11 virtual addresses using the
// 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:
// second_region_dsp_addr = first_region_dsp_addr | 0x10000
@@ -110,14 +114,17 @@ static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivial
// GCC versions < 5.0 do not implement std::is_trivially_copyable.
// Excluding MSVC because it has weird behaviour for std::is_trivially_copyable.
#if (__GNUC__ >= 5) || defined(__clang__)
#define ASSERT_DSP_STRUCT(name, size) \
static_assert(std::is_standard_layout<name>::value, "DSP structure " #name " doesn't use standard layout"); \
static_assert(std::is_trivially_copyable<name>::value, "DSP structure " #name " isn't trivially copyable"); \
static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
#define ASSERT_DSP_STRUCT(name, size) \
static_assert(std::is_standard_layout<name>::value, \
"DSP structure " #name " doesn't use standard layout"); \
static_assert(std::is_trivially_copyable<name>::value, \
"DSP structure " #name " isn't trivially copyable"); \
static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
#else
#define ASSERT_DSP_STRUCT(name, size) \
static_assert(std::is_standard_layout<name>::value, "DSP structure " #name " doesn't use standard layout"); \
static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
#define ASSERT_DSP_STRUCT(name, size) \
static_assert(std::is_standard_layout<name>::value, \
"DSP structure " #name " doesn't use standard layout"); \
static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
#endif
struct SourceConfiguration {
@@ -130,7 +137,8 @@ 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.
BitField<3, 1, u32_le>
partial_embedded_buffer_dirty; ///< Tends to be set when a looped buffer is queued.
BitField<4, 1, u32_le> partial_reset_flag;
BitField<16, 1, u32_le> enable_dirty;
@@ -138,7 +146,8 @@ 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.
BitField<21, 1, u32_le>
play_position_dirty; ///< Tends to also be set when embedded buffer is updated.
BitField<22, 1, u32_le> filters_enabled_dirty;
BitField<23, 1, u32_le> simple_filter_dirty;
BitField<24, 1, u32_le> biquad_filter_dirty;
@@ -164,11 +173,7 @@ 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
@@ -191,7 +196,8 @@ struct SourceConfiguration {
* This is a normalised biquad filter (second-order).
* 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
* Nintendo chose to negate the feedbackward coefficients. This differs from standard
* notation
* as in: https://ccrma.stanford.edu/~jos/filters/Direct_Form_I.html
* Values are signed fixed point with 14 fractional bits.
*/
@@ -239,23 +245,24 @@ struct SourceConfiguration {
/// Is a looping buffer.
u8 is_looping;
/// This value is shown in SourceStatus::previous_buffer_id when this buffer has finished.
/// 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
u16_le buffer_id;
INSERT_PADDING_DSPWORDS(1);
};
u16_le buffers_dirty; ///< Bitmap indicating which buffers are dirty (bit i -> buffers[i])
Buffer buffers[4]; ///< Queued Buffers
u16_le buffers_dirty; ///< Bitmap indicating which buffers are dirty (bit i -> buffers[i])
Buffer buffers[4]; ///< Queued Buffers
// Playback controls
u32_dsp loop_related;
u8 enable;
INSERT_PADDING_BYTES(1);
u16_le sync; ///< Application-side sync (See also: SourceStatus::sync)
u32_dsp play_position; ///< Position. (Units: number of samples)
u16_le sync; ///< Application-side sync (See also: SourceStatus::sync)
u32_dsp play_position; ///< Position. (Units: number of samples)
INSERT_PADDING_DSPWORDS(2);
// Embedded Buffer
@@ -268,16 +275,9 @@ 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;
@@ -299,10 +299,11 @@ struct SourceConfiguration {
union {
u16_le flags2_raw;
BitField<0, 1, u16_le> adpcm_dirty; ///< Has the ADPCM info above been changed?
BitField<1, 1, u16_le> is_looping; ///< Is this a looping buffer?
BitField<1, 1, u16_le> is_looping; ///< Is this a looping buffer?
};
/// Buffer id of embedded buffer (used as a buffer id in SourceStatus to reference this buffer).
/// Buffer id of embedded buffer (used as a buffer id in SourceStatus to reference this
/// buffer).
u16_le buffer_id;
};
@@ -313,11 +314,11 @@ ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
struct SourceStatus {
struct Status {
u8 is_enabled; ///< Is this channel enabled? (Doesn't have to be playing anything.)
u8 current_buffer_id_dirty; ///< Non-zero when current_buffer_id changes
u16_le sync; ///< Is set by the DSP to the value of SourceConfiguration::sync
u32_dsp buffer_position; ///< Number of samples into the current buffer
u16_le current_buffer_id; ///< Updated when a buffer finishes playing
u8 is_enabled; ///< Is this channel enabled? (Doesn't have to be playing anything.)
u8 current_buffer_id_dirty; ///< Non-zero when current_buffer_id changes
u16_le sync; ///< Is set by the DSP to the value of SourceConfiguration::sync
u32_dsp buffer_position; ///< Number of samples into the current buffer
u16_le current_buffer_id; ///< Updated when a buffer finishes playing
INSERT_PADDING_DSPWORDS(1);
};
@@ -347,16 +348,13 @@ struct DspConfiguration {
BitField<28, 1, u32_le> headphones_connected_dirty;
};
/// The DSP has three intermediate audio mixers. This controls the volume level (0.0-1.0) for each at the final mixer
/// The DSP has three intermediate audio mixers. This controls the volume level (0.0-1.0) for
/// 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;
@@ -388,8 +386,9 @@ 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
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
// Coefficients
s16_le g; ///< Fixed point with 7 fractional bits
@@ -506,21 +505,36 @@ ASSERT_DSP_STRUCT(SharedMemory, 0x8000);
extern std::array<SharedMemory, 2> g_regions;
// Structures must have an offset that is a multiple of two.
static_assert(offsetof(SharedMemory, frame_counter) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_statuses) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, adpcm_coefficients) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_configuration) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_status) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, final_samples) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, intermediate_mix_samples) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, compressor) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_debug) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown10) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown11) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown12) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown13) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown14) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, frame_counter) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_statuses) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, adpcm_coefficients) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_configuration) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_status) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, final_samples) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, intermediate_mix_samples) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, compressor) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_debug) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown10) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown11) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown12) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown13) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown14) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
#undef INSERT_PADDING_DSPWORDS
#undef ASSERT_DSP_STRUCT

6
src/audio_core/hle/filter.cpp
View File

@@ -59,7 +59,8 @@ void SourceFilters::SimpleFilter::Reset() {
b0 = 1 << 15;
}
void SourceFilters::SimpleFilter::Configure(SourceConfiguration::Configuration::SimpleFilter config) {
void SourceFilters::SimpleFilter::Configure(
SourceConfiguration::Configuration::SimpleFilter config) {
a1 = config.a1;
b0 = config.b0;
}
@@ -88,7 +89,8 @@ void SourceFilters::BiquadFilter::Reset() {
b0 = 1 << 14;
}
void SourceFilters::BiquadFilter::Configure(SourceConfiguration::Configuration::BiquadFilter config) {
void SourceFilters::BiquadFilter::Configure(
SourceConfiguration::Configuration::BiquadFilter config) {
a1 = config.a1;
a2 = config.a2;
b0 = config.b0;

12
src/audio_core/hle/filter.h
View File

@@ -17,7 +17,9 @@ namespace HLE {
/// Preprocessing filters. There is an independent set of filters for each Source.
class SourceFilters final {
public:
SourceFilters() { Reset(); }
SourceFilters() {
Reset();
}
/// Reset internal state.
void Reset();
@@ -54,7 +56,9 @@ private:
bool biquad_filter_enabled;
struct SimpleFilter {
SimpleFilter() { Reset(); }
SimpleFilter() {
Reset();
}
/// Resets internal state.
void Reset();
@@ -80,7 +84,9 @@ private:
} simple_filter;
struct BiquadFilter {
BiquadFilter() { Reset(); }
BiquadFilter() {
Reset();
}
/// Resets internal state.
void Reset();

65
src/audio_core/hle/mixers.cpp
View File

@@ -20,11 +20,9 @@ void Mixers::Reset() {
state = {};
}
DspStatus Mixers::Tick(DspConfiguration& config,
const IntermediateMixSamples& read_samples,
IntermediateMixSamples& write_samples,
const std::array<QuadFrame32, 3>& input)
{
DspStatus Mixers::Tick(DspConfiguration& config, const IntermediateMixSamples& read_samples,
IntermediateMixSamples& write_samples,
const std::array<QuadFrame32, 3>& input) {
ParseConfig(config);
AuxReturn(read_samples);
@@ -73,13 +71,15 @@ void Mixers::ParseConfig(DspConfiguration& config) {
if (config.output_format_dirty) {
config.output_format_dirty.Assign(0);
state.output_format = config.output_format;
LOG_TRACE(Audio_DSP, "mixers output_format = %zu", static_cast<size_t>(config.output_format));
LOG_TRACE(Audio_DSP, "mixers output_format = %zu",
static_cast<size_t>(config.output_format));
}
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.)
// (Note: Whether headphones are connected does affect coefficients used for surround
// sound.)
LOG_TRACE(Audio_DSP, "mixers headphones_connected=%hu", config.headphones_connected);
}
@@ -94,11 +94,10 @@ static s16 ClampToS16(s32 value) {
return static_cast<s16>(MathUtil::Clamp(value, -32768, 32767));
}
static std::array<s16, 2> AddAndClampToS16(const std::array<s16, 2>& a, const std::array<s16, 2>& b) {
return {
ClampToS16(static_cast<s32>(a[0]) + static_cast<s32>(b[0])),
ClampToS16(static_cast<s32>(a[1]) + static_cast<s32>(b[1]))
};
static std::array<s16, 2> AddAndClampToS16(const std::array<s16, 2>& a,
const std::array<s16, 2>& b) {
return {ClampToS16(static_cast<s32>(a[0]) + static_cast<s32>(b[0])),
ClampToS16(static_cast<s32>(a[1]) + static_cast<s32>(b[1]))};
}
void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples) {
@@ -106,27 +105,33 @@ void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& sample
switch (state.output_format) {
case OutputFormat::Mono:
std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
[gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> {
std::transform(
current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
[gain](const std::array<s16, 2>& accumulator,
const std::array<s32, 4>& sample) -> std::array<s16, 2> {
// Downmix to mono
s16 mono = ClampToS16(static_cast<s32>((gain * sample[0] + gain * sample[1] + gain * sample[2] + gain * sample[3]) / 2));
s16 mono = ClampToS16(static_cast<s32>(
(gain * sample[0] + gain * sample[1] + gain * sample[2] + gain * sample[3]) /
2));
// Mix into current frame
return AddAndClampToS16(accumulator, { mono, mono });
return AddAndClampToS16(accumulator, {mono, mono});
});
return;
case OutputFormat::Surround:
// TODO(merry): Implement surround sound.
// fallthrough
// TODO(merry): Implement surround sound.
// fallthrough
case OutputFormat::Stereo:
std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
[gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> {
std::transform(
current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
[gain](const std::array<s16, 2>& accumulator,
const std::array<s32, 4>& sample) -> std::array<s16, 2> {
// Downmix to stereo
s16 left = ClampToS16(static_cast<s32>(gain * sample[0] + gain * sample[2]));
s16 right = ClampToS16(static_cast<s32>(gain * sample[1] + gain * sample[3]));
// Mix into current frame
return AddAndClampToS16(accumulator, { left, right });
return AddAndClampToS16(accumulator, {left, right});
});
return;
}
@@ -135,12 +140,14 @@ void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& sample
}
void Mixers::AuxReturn(const IntermediateMixSamples& read_samples) {
// NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32.
// NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to
// QuadFrame32.
if (state.mixer1_enabled) {
for (size_t sample = 0; sample < samples_per_frame; sample++) {
for (size_t channel = 0; channel < 4; channel++) {
state.intermediate_mix_buffer[1][sample][channel] = read_samples.mix1.pcm32[channel][sample];
state.intermediate_mix_buffer[1][sample][channel] =
read_samples.mix1.pcm32[channel][sample];
}
}
}
@@ -148,14 +155,17 @@ void Mixers::AuxReturn(const IntermediateMixSamples& read_samples) {
if (state.mixer2_enabled) {
for (size_t sample = 0; sample < samples_per_frame; sample++) {
for (size_t channel = 0; channel < 4; channel++) {
state.intermediate_mix_buffer[2][sample][channel] = read_samples.mix2.pcm32[channel][sample];
state.intermediate_mix_buffer[2][sample][channel] =
read_samples.mix2.pcm32[channel][sample];
}
}
}
}
void Mixers::AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input) {
// NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32.
void Mixers::AuxSend(IntermediateMixSamples& write_samples,
const std::array<QuadFrame32, 3>& input) {
// NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to
// QuadFrame32.
state.intermediate_mix_buffer[0] = input[0];
@@ -184,7 +194,8 @@ void Mixers::MixCurrentFrame() {
current_frame.fill({});
for (size_t mix = 0; mix < 3; mix++) {
DownmixAndMixIntoCurrentFrame(state.intermediate_mixer_volume[mix], state.intermediate_mix_buffer[mix]);
DownmixAndMixIntoCurrentFrame(state.intermediate_mixer_volume[mix],
state.intermediate_mix_buffer[mix]);
}
// TODO(merry): Compressor. (We currently assume a disabled compressor.)

9
src/audio_core/hle/mixers.h
View File

@@ -20,10 +20,8 @@ public:
void Reset();
DspStatus Tick(DspConfiguration& config,
const IntermediateMixSamples& read_samples,
IntermediateMixSamples& write_samples,
const std::array<QuadFrame32, 3>& input);
DspStatus Tick(DspConfiguration& config, const IntermediateMixSamples& read_samples,
IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input);
StereoFrame16 GetOutput() const {
return current_frame;
@@ -53,7 +51,8 @@ private:
void AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input);
/// INTERNAL: Mix current_frame.
void MixCurrentFrame();
/// INTERNAL: Downmix from quadraphonic to stereo based on status.output_format and accumulate into current_frame.
/// INTERNAL: Downmix from quadraphonic to stereo based on status.output_format and accumulate
/// into current_frame.
void DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples);
/// INTERNAL: Generate DspStatus based on internal state.
DspStatus GetCurrentStatus() const;

26
src/audio_core/hle/pipe.cpp
View File

@@ -44,8 +44,10 @@ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
std::vector<u8>& data = pipe_data[pipe_index];
if (length > data.size()) {
LOG_WARNING(Audio_DSP, "pipe_number = %zu is out of data, application requested read of %u but %zu remain",
pipe_index, length, data.size());
LOG_WARNING(
Audio_DSP,
"pipe_number = %zu is out of data, application requested read of %u but %zu remain",
pipe_index, length, data.size());
length = static_cast<u32>(data.size());
}
@@ -95,8 +97,7 @@ 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()));
@@ -112,16 +113,12 @@ void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
switch (pipe_number) {
case DspPipe::Audio: {
if (buffer.size() != 4) {
LOG_ERROR(Audio_DSP, "DspPipe::Audio: Unexpected buffer length %zu was written", buffer.size());
LOG_ERROR(Audio_DSP, "DspPipe::Audio: Unexpected buffer length %zu was written",
buffer.size());
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.)
@@ -152,7 +149,9 @@ void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
dsp_state = DspState::Sleeping;
break;
default:
LOG_ERROR(Audio_DSP, "Application has requested unknown state transition of DSP hardware %hhu", buffer[0]);
LOG_ERROR(Audio_DSP,
"Application has requested unknown state transition of DSP hardware %hhu",
buffer[0]);
dsp_state = DspState::Off;
break;
}
@@ -160,7 +159,8 @@ void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
return;
}
default:
LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented", static_cast<size_t>(pipe_number));
LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented",
static_cast<size_t>(pipe_number));
UNIMPLEMENTED();
return;
}

19
src/audio_core/hle/pipe.h
View File

@@ -15,20 +15,17 @@ 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;
/**
* Reads `length` bytes from the DSP pipe identified with `pipe_number`.
* @note Can read up to the maximum value of a u16 in bytes (65,535).
* @note IF an error is encoutered with either an invalid `pipe_number` or `length` value, an empty vector will be returned.
* @note IF an error is encoutered with either an invalid `pipe_number` or `length` value, an empty
* vector will be returned.
* @note IF `length` is set to 0, an empty vector will be returned.
* @note IF `length` is greater than the amount of data available, this function will only read the available amount.
* @note IF `length` is greater than the amount of data available, this function will only read the
* available amount.
* @param pipe_number a `DspPipe`
* @param length the number of bytes to read. The max is 65,535 (max of u16).
* @returns a vector of bytes from the specified pipe. On error, will be empty.
@@ -49,11 +46,7 @@ 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();

119
src/audio_core/hle/source.cpp
View File

@@ -18,7 +18,8 @@
namespace DSP {
namespace HLE {
SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config,
const s16_le (&adpcm_coeffs)[16]) {
ParseConfig(config, adpcm_coeffs);
if (state.enabled) {
@@ -47,7 +48,8 @@ void Source::Reset() {
state = {};
}
void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
void Source::ParseConfig(SourceConfiguration::Configuration& config,
const s16_le (&adpcm_coeffs)[16]) {
if (!config.dirty_raw) {
return;
}
@@ -82,7 +84,8 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_l
LOG_TRACE(Audio_DSP, "source_id=%zu rate=%f", source_id, state.rate_multiplier);
if (state.rate_multiplier <= 0) {
LOG_ERROR(Audio_DSP, "Was given an invalid rate multiplier: source_id=%zu rate=%f", source_id, state.rate_multiplier);
LOG_ERROR(Audio_DSP, "Was given an invalid rate multiplier: source_id=%zu rate=%f",
source_id, state.rate_multiplier);
state.rate_multiplier = 1.0f;
// Note: Actual firmware starts producing garbage if this occurs.
}
@@ -90,37 +93,39 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_l
if (config.adpcm_coefficients_dirty) {
config.adpcm_coefficients_dirty.Assign(0);
std::transform(adpcm_coeffs, adpcm_coeffs + state.adpcm_coeffs.size(), state.adpcm_coeffs.begin(),
[](const auto& coeff) { return static_cast<s16>(coeff); });
std::transform(adpcm_coeffs, adpcm_coeffs + state.adpcm_coeffs.size(),
state.adpcm_coeffs.begin(),
[](const auto& coeff) { return static_cast<s16>(coeff); });
LOG_TRACE(Audio_DSP, "source_id=%zu adpcm update", source_id);
}
if (config.gain_0_dirty) {
config.gain_0_dirty.Assign(0);
std::transform(config.gain[0], config.gain[0] + state.gain[0].size(), state.gain[0].begin(),
[](const auto& coeff) { return static_cast<float>(coeff); });
[](const auto& coeff) { return static_cast<float>(coeff); });
LOG_TRACE(Audio_DSP, "source_id=%zu gain 0 update", source_id);
}
if (config.gain_1_dirty) {
config.gain_1_dirty.Assign(0);
std::transform(config.gain[1], config.gain[1] + state.gain[1].size(), state.gain[1].begin(),
[](const auto& coeff) { return static_cast<float>(coeff); });
[](const auto& coeff) { return static_cast<float>(coeff); });
LOG_TRACE(Audio_DSP, "source_id=%zu gain 1 update", source_id);
}
if (config.gain_2_dirty) {
config.gain_2_dirty.Assign(0);
std::transform(config.gain[2], config.gain[2] + state.gain[2].size(), state.gain[2].begin(),
[](const auto& coeff) { return static_cast<float>(coeff); });
[](const auto& coeff) { return static_cast<float>(coeff); });
LOG_TRACE(Audio_DSP, "source_id=%zu gain 2 update", source_id);
}
if (config.filters_enabled_dirty) {
config.filters_enabled_dirty.Assign(0);
state.filters.Enable(config.simple_filter_enabled.ToBool(), config.biquad_filter_enabled.ToBool());
LOG_TRACE(Audio_DSP, "source_id=%zu enable_simple=%hu enable_biquad=%hu",
source_id, config.simple_filter_enabled.Value(), config.biquad_filter_enabled.Value());
state.filters.Enable(config.simple_filter_enabled.ToBool(),
config.biquad_filter_enabled.ToBool());
LOG_TRACE(Audio_DSP, "source_id=%zu enable_simple=%hu enable_biquad=%hu", source_id,
config.simple_filter_enabled.Value(), config.biquad_filter_enabled.Value());
}
if (config.simple_filter_dirty) {
@@ -138,36 +143,38 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_l
if (config.interpolation_dirty) {
config.interpolation_dirty.Assign(0);
state.interpolation_mode = config.interpolation_mode;
LOG_TRACE(Audio_DSP, "source_id=%zu interpolation_mode=%zu", source_id, static_cast<size_t>(state.interpolation_mode));
LOG_TRACE(Audio_DSP, "source_id=%zu interpolation_mode=%zu", source_id,
static_cast<size_t>(state.interpolation_mode));
}
if (config.format_dirty || config.embedded_buffer_dirty) {
config.format_dirty.Assign(0);
state.format = config.format;
LOG_TRACE(Audio_DSP, "source_id=%zu format=%zu", source_id, static_cast<size_t>(state.format));
LOG_TRACE(Audio_DSP, "source_id=%zu format=%zu", source_id,
static_cast<size_t>(state.format));
}
if (config.mono_or_stereo_dirty || config.embedded_buffer_dirty) {
config.mono_or_stereo_dirty.Assign(0);
state.mono_or_stereo = config.mono_or_stereo;
LOG_TRACE(Audio_DSP, "source_id=%zu mono_or_stereo=%zu", source_id, static_cast<size_t>(state.mono_or_stereo));
LOG_TRACE(Audio_DSP, "source_id=%zu mono_or_stereo=%zu", source_id,
static_cast<size_t>(state.mono_or_stereo));
}
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
});
LOG_TRACE(Audio_DSP, "enqueuing embedded addr=0x%08x len=%u id=%hu", config.physical_address, config.length, config.buffer_id);
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);
}
if (config.buffer_queue_dirty) {
@@ -175,19 +182,18 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_l
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
});
LOG_TRACE(Audio_DSP, "enqueuing queued %zu addr=0x%08x len=%u id=%hu", i, b.physical_address, b.length, b.buffer_id);
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);
}
}
config.buffers_dirty = 0;
@@ -218,10 +224,13 @@ void Source::GenerateFrame() {
break;
}
const size_t size_to_copy = std::min(state.current_buffer.size(), current_frame.size() - frame_position);
const size_t size_to_copy =
std::min(state.current_buffer.size(), current_frame.size() - frame_position);
std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy, current_frame.begin() + frame_position);
state.current_buffer.erase(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy);
std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy,
current_frame.begin() + frame_position);
state.current_buffer.erase(state.current_buffer.begin(),
state.current_buffer.begin() + size_to_copy);
frame_position += size_to_copy;
state.next_sample_number += static_cast<u32>(size_to_copy);
@@ -230,9 +239,9 @@ void Source::GenerateFrame() {
state.filters.ProcessFrame(current_frame);
}
bool Source::DequeueBuffer() {
ASSERT_MSG(state.current_buffer.empty(), "Shouldn't dequeue; we still have data in current_buffer");
ASSERT_MSG(state.current_buffer.empty(),
"Shouldn't dequeue; we still have data in current_buffer");
if (state.input_queue.empty())
return false;
@@ -261,29 +270,34 @@ bool Source::DequeueBuffer() {
break;
case Format::ADPCM:
DEBUG_ASSERT(num_channels == 1);
state.current_buffer = Codec::DecodeADPCM(memory, buf.length, state.adpcm_coeffs, state.adpcm_state);
state.current_buffer =
Codec::DecodeADPCM(memory, buf.length, state.adpcm_coeffs, state.adpcm_state);
break;
default:
UNIMPLEMENTED();
break;
}
} else {
LOG_WARNING(Audio_DSP, "source_id=%zu buffer_id=%hu length=%u: Invalid physical address 0x%08X",
source_id, buf.buffer_id, buf.length, buf.physical_address);
LOG_WARNING(Audio_DSP,
"source_id=%zu buffer_id=%hu length=%u: Invalid physical address 0x%08X",
source_id, buf.buffer_id, buf.length, buf.physical_address);
state.current_buffer.clear();
return true;
}
switch (state.interpolation_mode) {
case InterpolationMode::None:
state.current_buffer = AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
state.current_buffer =
AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
break;
case InterpolationMode::Linear:
state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
state.current_buffer =
AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
break;
case InterpolationMode::Polyphase:
// TODO(merry): Implement polyphase interpolation
state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
state.current_buffer =
AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
break;
default:
UNIMPLEMENTED();
@@ -296,7 +310,8 @@ bool Source::DequeueBuffer() {
state.buffer_update = buf.from_queue;
LOG_TRACE(Audio_DSP, "source_id=%zu buffer_id=%hu from_queue=%s current_buffer.size()=%zu",
source_id, buf.buffer_id, buf.from_queue ? "true" : "false", state.current_buffer.size());
source_id, buf.buffer_id, buf.from_queue ? "true" : "false",
state.current_buffer.size());
return true;
}

17
src/audio_core/hle/source.h
View File

@@ -40,13 +40,17 @@ public:
/**
* This is called once every audio frame. This performs per-source processing every frame.
* @param config The new configuration we've got for this Source from the application.
* @param adpcm_coeffs ADPCM coefficients to use if config tells us to use them (may contain invalid values otherwise).
* @return The current status of this Source. This is given back to the emulated application via SharedMemory.
* @param adpcm_coeffs ADPCM coefficients to use if config tells us to use them (may contain
* invalid values otherwise).
* @return The current status of this Source. This is given back to the emulated application via
* SharedMemory.
*/
SourceStatus::Status Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
SourceStatus::Status Tick(SourceConfiguration::Configuration& config,
const s16_le (&adpcm_coeffs)[16]);
/**
* Mix this source's output into dest, using the gains for the `intermediate_mix_id`-th intermediate mixer.
* Mix this source's output into dest, using the gains for the `intermediate_mix_id`-th
* intermediate mixer.
* @param dest The QuadFrame32 to mix into.
* @param intermediate_mix_id The id of the intermediate mix whose gains we are using.
*/
@@ -77,7 +81,7 @@ private:
};
struct BufferOrder {
bool operator() (const Buffer& a, const Buffer& b) const {
bool operator()(const Buffer& a, const Buffer& b) const {
// Lower buffer_id comes first.
return a.buffer_id > b.buffer_id;
}
@@ -134,7 +138,8 @@ private:
void ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
/// INTERNAL: Generate the current audio output for this frame based on our internal state.
void GenerateFrame();
/// INTERNAL: Dequeues a buffer and does preprocessing on it (decoding, resampling). Puts it into current_buffer.
/// INTERNAL: Dequeues a buffer and does preprocessing on it (decoding, resampling). Puts it
/// into current_buffer.
bool DequeueBuffer();
/// INTERNAL: Generates a SourceStatus::Status based on our internal state.
SourceStatus::Status GetCurrentStatus();

18
src/audio_core/interpolate.cpp
View File

@@ -17,7 +17,8 @@ constexpr u64 scale_mask = scale_factor - 1;
/// Here we step over the input in steps of rate_multiplier, until we consume all of the input.
/// Three adjacent samples are passed to fn each step.
template <typename Function>
static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input, float rate_multiplier, Function fn) {
static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input,
float rate_multiplier, Function fn) {
ASSERT(rate_multiplier > 0);
if (input.size() < 2)
@@ -63,22 +64,21 @@ static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input,
}
StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
return x0;
});
return StepOverSamples(
state, input, rate_multiplier,
[](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
}
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) {
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 std::array<s16, 2>{static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
static_cast<s16>(x0[1] + fraction * delta1 / scale_factor)};
});
}

6
src/audio_core/interpolate.h
View File

@@ -24,7 +24,8 @@ struct State {
* No interpolation. This is equivalent to a zero-order hold. There is a two-sample predelay.
* @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.
* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
* performs upsampling.
* @return The resampled audio buffer.
*/
StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
@@ -33,7 +34,8 @@ StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multip
* Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
* @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.
* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
* performs upsampling.
* @return The resampled audio buffer.
*/
StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);

3
src/audio_core/null_sink.h
View File

@@ -19,7 +19,8 @@ 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;

22
src/audio_core/sdl2_sink.cpp
View File

@@ -10,9 +10,9 @@
#include "audio_core/audio_core.h"
#include "audio_core/sdl2_sink.h"
#include <numeric>
#include "common/assert.h"
#include "common/logging/log.h"
#include <numeric>
namespace AudioCore {
@@ -45,7 +45,8 @@ SDL2Sink::SDL2Sink() : impl(std::make_unique<Impl>()) {
SDL_AudioSpec obtained_audiospec;
SDL_zero(obtained_audiospec);
impl->audio_device_id = SDL_OpenAudioDevice(nullptr, false, &desired_audiospec, &obtained_audiospec, 0);
impl->audio_device_id =
SDL_OpenAudioDevice(nullptr, false, &desired_audiospec, &obtained_audiospec, 0);
if (impl->audio_device_id <= 0) {
LOG_CRITICAL(Audio_Sink, "SDL_OpenAudioDevice failed");
return;
@@ -86,11 +87,12 @@ size_t SDL2Sink::SamplesInQueue() const {
SDL_LockAudioDevice(impl->audio_device_id);
size_t total_size = std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<size_t>(0),
[](size_t sum, const auto& buffer) {
// Division by two because each stereo sample is made of two s16.
return sum + buffer.size() / 2;
});
size_t total_size = std::accumulate(impl->queue.begin(), impl->queue.end(),
static_cast<size_t>(0), [](size_t sum, const auto& buffer) {
// Division by two because each stereo sample is made of
// two s16.
return sum + buffer.size() / 2;
});
SDL_UnlockAudioDevice(impl->audio_device_id);
@@ -100,7 +102,8 @@ size_t SDL2Sink::SamplesInQueue() const {
void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) {
Impl* impl = reinterpret_cast<Impl*>(impl_);
size_t remaining_size = static_cast<size_t>(buffer_size_in_bytes) / sizeof(s16); // Keep track of size in 16-bit increments.
size_t remaining_size = static_cast<size_t>(buffer_size_in_bytes) /
sizeof(s16); // Keep track of size in 16-bit increments.
while (remaining_size > 0 && !impl->queue.empty()) {
if (impl->queue.front().size() <= remaining_size) {
@@ -111,7 +114,8 @@ void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes)
} else {
memcpy(buffer, impl->queue.front().data(), remaining_size * sizeof(s16));
buffer += remaining_size * sizeof(s16);
impl->queue.front().erase(impl->queue.front().begin(), impl->queue.front().begin() + remaining_size);
impl->queue.front().erase(impl->queue.front().begin(),
impl->queue.front().begin() + remaining_size);
remaining_size = 0;
}
}

6
src/audio_core/sink.h
View File

@@ -11,14 +11,16 @@
namespace AudioCore {
/**
* This class is an interface for an audio sink. An audio sink accepts samples in stereo signed PCM16 format to be output.
* 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.
*/
class Sink {
public:
virtual ~Sink() = default;
/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units: samples/sec)
/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units:
/// samples/sec)
virtual unsigned int GetNativeSampleRate() const = 0;
/**

4
src/audio_core/sink_details.cpp
View File

@@ -17,9 +17,9 @@ namespace AudioCore {
// g_sink_details is ordered in terms of desirability, with the best choice at the top.
const std::vector<SinkDetails> g_sink_details = {
#ifdef HAVE_SDL2
{ "sdl2", []() { return std::make_unique<SDL2Sink>(); } },
{"sdl2", []() { return std::make_unique<SDL2Sink>(); }},
#endif
{ "null", []() { return std::make_unique<NullSink>(); } },
{"null", []() { return std::make_unique<NullSink>(); }},
};
} // namespace AudioCore

3
src/audio_core/sink_details.h
View File

@@ -14,7 +14,8 @@ 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;

10
src/audio_core/time_stretch.cpp
View File

@@ -26,8 +26,8 @@ static double ClampRatio(double ratio) {
return MathUtil::Clamp(ratio, MIN_RATIO, MAX_RATIO);
}
constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
constexpr size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples
constexpr double SMOOTHING_FACTOR = 0.007;
@@ -48,7 +48,8 @@ std::vector<s16> TimeStretcher::Process(size_t samples_in_queue) {
double ratio = CalculateCurrentRatio();
ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue);
impl->smoothed_ratio = (1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
impl->smoothed_ratio =
(1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio);
// SoundTouch's tempo definition the inverse of our ratio definition.
@@ -100,7 +101,8 @@ double TimeStretcher::CalculateCurrentRatio() {
const steady_clock::time_point now = steady_clock::now();
const std::chrono::duration<double> duration = now - impl->frame_timer;
const double expected_time = static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
const double expected_time =
static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
const double actual_time = duration.count();
double ratio;

6
src/audio_core/time_stretch.h
View File

@@ -37,7 +37,8 @@ public:
/**
* Does audio stretching and produces the time-stretched samples.
* Timer calculations use sample_delay to determine how much of a margin we have.
* @param sample_delay How many samples are buffered downstream of this module and haven't been played yet.
* @param sample_delay How many samples are buffered downstream of this module and haven't been
* played yet.
* @return Samples to play in interleaved stereo PCM16 format.
*/
std::vector<s16> Process(size_t sample_delay);
@@ -48,7 +49,8 @@ private:
/// INTERNAL: ratio = wallclock time / emulated time
double CalculateCurrentRatio();
/// INTERNAL: If we have too many or too few samples downstream, nudge ratio in the appropriate direction.
/// INTERNAL: If we have too many or too few samples downstream, nudge ratio in the appropriate
/// direction.
double CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const;
/// INTERNAL: Gets the time-stretched samples from SoundTouch.
std::vector<s16> GetSamples();