diff options
Diffstat (limited to 'src/audio_core')
| -rw-r--r-- | src/audio_core/CMakeLists.txt | 7 | ||||
| -rw-r--r-- | src/audio_core/audio_core.cpp | 13 | ||||
| -rw-r--r-- | src/audio_core/audio_core.h | 2 | ||||
| -rw-r--r-- | src/audio_core/hle/common.h | 9 | ||||
| -rw-r--r-- | src/audio_core/hle/dsp.cpp | 44 | ||||
| -rw-r--r-- | src/audio_core/hle/dsp.h | 21 | ||||
| -rw-r--r-- | src/audio_core/hle/pipe.cpp | 32 | ||||
| -rw-r--r-- | src/audio_core/hle/pipe.h | 4 | ||||
| -rw-r--r-- | src/audio_core/interpolate.cpp | 85 | ||||
| -rw-r--r-- | src/audio_core/interpolate.h | 41 |
10 files changed, 202 insertions, 56 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt index 869da5e83..a965af291 100644 --- a/src/audio_core/CMakeLists.txt +++ b/src/audio_core/CMakeLists.txt @@ -4,6 +4,7 @@ set(SRCS hle/dsp.cpp hle/filter.cpp hle/pipe.cpp + interpolate.cpp ) set(HEADERS @@ -13,9 +14,13 @@ set(HEADERS hle/dsp.h hle/filter.h hle/pipe.h + interpolate.h sink.h ) +include_directories(../../externals/soundtouch/include) + create_directory_groups(${SRCS} ${HEADERS}) -add_library(audio_core STATIC ${SRCS} ${HEADERS})
\ No newline at end of file +add_library(audio_core STATIC ${SRCS} ${HEADERS}) +target_link_libraries(audio_core SoundTouch) diff --git a/src/audio_core/audio_core.cpp b/src/audio_core/audio_core.cpp index 894f46990..cbe869a04 100644 --- a/src/audio_core/audio_core.cpp +++ b/src/audio_core/audio_core.cpp @@ -4,6 +4,7 @@ #include "audio_core/audio_core.h" #include "audio_core/hle/dsp.h" +#include "audio_core/hle/pipe.h" #include "core/core_timing.h" #include "core/hle/kernel/vm_manager.h" @@ -17,10 +18,10 @@ static constexpr u64 audio_frame_ticks = 1310252ull; ///< Units: ARM11 cycles static void AudioTickCallback(u64 /*userdata*/, int cycles_late) { if (DSP::HLE::Tick()) { - // HACK: We're not signaling the interrups when they should be, but just firing them all off together. - // It should be only (interrupt_id = 2, channel_id = 2) that's signalled here. - // TODO(merry): Understand when the other interrupts are fired. - DSP_DSP::SignalAllInterrupts(); + // TODO(merry): Signal all the other interrupts as appropriate. + DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Audio); + // HACK(merry): Added to prevent regressions. Will remove soon. + DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Binary); } // Reschedule recurrent event @@ -37,10 +38,10 @@ void Init() { /// Add DSP address spaces to Process's address space. void AddAddressSpace(Kernel::VMManager& address_space) { - auto r0_vma = address_space.MapBackingMemory(DSP::HLE::region0_base, reinterpret_cast<u8*>(&DSP::HLE::g_region0), 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_region1), 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); } diff --git a/src/audio_core/audio_core.h b/src/audio_core/audio_core.h index 64c330914..b349895ea 100644 --- a/src/audio_core/audio_core.h +++ b/src/audio_core/audio_core.h @@ -10,8 +10,6 @@ class VMManager; namespace AudioCore { -constexpr int num_sources = 24; -constexpr int samples_per_frame = 160; ///< Samples per audio frame at native sample rate constexpr int native_sample_rate = 32728; ///< 32kHz /// Initialise Audio Core diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h index 37d441eb2..7910f42ae 100644 --- a/src/audio_core/hle/common.h +++ b/src/audio_core/hle/common.h @@ -7,18 +7,19 @@ #include <algorithm> #include <array> -#include "audio_core/audio_core.h" - #include "common/common_types.h" namespace DSP { namespace HLE { +constexpr int num_sources = 24; +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>, AudioCore::samples_per_frame>; +using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>; /// The DSP is quadraphonic internally. -using QuadFrame32 = std::array<std::array<s32, 4>, AudioCore::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. diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp index c89356edc..5759a5b9e 100644 --- a/src/audio_core/hle/dsp.cpp +++ b/src/audio_core/hle/dsp.cpp @@ -8,8 +8,32 @@ namespace DSP { namespace HLE { -SharedMemory g_region0; -SharedMemory g_region1; +std::array<SharedMemory, 2> g_regions; + +static size_t CurrentRegionIndex() { + // The region with the higher frame counter is chosen unless there is wraparound. + // This function only returns a 0 or 1. + + if (g_regions[0].frame_counter == 0xFFFFu && g_regions[1].frame_counter != 0xFFFEu) { + // Wraparound has occured. + return 1; + } + + if (g_regions[1].frame_counter == 0xFFFFu && g_regions[0].frame_counter != 0xFFFEu) { + // Wraparound has occured. + return 0; + } + + return (g_regions[0].frame_counter > g_regions[1].frame_counter) ? 0 : 1; +} + +static SharedMemory& ReadRegion() { + return g_regions[CurrentRegionIndex()]; +} + +static SharedMemory& WriteRegion() { + return g_regions[1 - CurrentRegionIndex()]; +} void Init() { DSP::HLE::ResetPipes(); @@ -22,21 +46,5 @@ bool Tick() { return true; } -SharedMemory& CurrentRegion() { - // The region with the higher frame counter is chosen unless there is wraparound. - - if (g_region0.frame_counter == 0xFFFFu && g_region1.frame_counter != 0xFFFEu) { - // Wraparound has occured. - return g_region1; - } - - if (g_region1.frame_counter == 0xFFFFu && g_region0.frame_counter != 0xFFFEu) { - // Wraparound has occured. - return g_region0; - } - - return (g_region0.frame_counter > g_region1.frame_counter) ? g_region0 : g_region1; -} - } // namespace HLE } // namespace DSP diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h index c15ef0b7a..f0f125284 100644 --- a/src/audio_core/hle/dsp.h +++ b/src/audio_core/hle/dsp.h @@ -4,10 +4,11 @@ #pragma once +#include <array> #include <cstddef> #include <type_traits> -#include "audio_core/audio_core.h" +#include "audio_core/hle/common.h" #include "common/bit_field.h" #include "common/common_funcs.h" @@ -30,10 +31,9 @@ namespace HLE { struct SharedMemory; constexpr VAddr region0_base = 0x1FF50000; -extern SharedMemory g_region0; - constexpr VAddr region1_base = 0x1FF70000; -extern SharedMemory g_region1; + +extern std::array<SharedMemory, 2> g_regions; /** * The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from @@ -305,7 +305,7 @@ struct SourceConfiguration { u16_le buffer_id; }; - Configuration config[AudioCore::num_sources]; + Configuration config[num_sources]; }; ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192); ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20); @@ -320,7 +320,7 @@ struct SourceStatus { INSERT_PADDING_DSPWORDS(1); }; - Status status[AudioCore::num_sources]; + Status status[num_sources]; }; ASSERT_DSP_STRUCT(SourceStatus::Status, 12); @@ -413,7 +413,7 @@ ASSERT_DSP_STRUCT(DspConfiguration::ReverbEffect, 52); struct AdpcmCoefficients { /// Coefficients are signed fixed point with 11 fractional bits. /// Each source has 16 coefficients associated with it. - s16_le coeff[AudioCore::num_sources][16]; + s16_le coeff[num_sources][16]; }; ASSERT_DSP_STRUCT(AdpcmCoefficients, 768); @@ -427,7 +427,7 @@ ASSERT_DSP_STRUCT(DspStatus, 32); /// Final mixed output in PCM16 stereo format, what you hear out of the speakers. /// When the application writes to this region it has no effect. struct FinalMixSamples { - s16_le pcm16[2 * AudioCore::samples_per_frame]; + s16_le pcm16[2 * samples_per_frame]; }; ASSERT_DSP_STRUCT(FinalMixSamples, 640); @@ -437,7 +437,7 @@ ASSERT_DSP_STRUCT(FinalMixSamples, 640); /// Values that exceed s16 range will be clipped by the DSP after further processing. struct IntermediateMixSamples { struct Samples { - s32_le pcm32[4][AudioCore::samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian. + s32_le pcm32[4][samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian. }; Samples mix1; @@ -535,8 +535,5 @@ void Shutdown(); */ bool Tick(); -/// Returns a mutable reference to the current region. Current region is selected based on the frame counter. -SharedMemory& CurrentRegion(); - } // namespace HLE } // namespace DSP diff --git a/src/audio_core/hle/pipe.cpp b/src/audio_core/hle/pipe.cpp index 9381883b4..03280780f 100644 --- a/src/audio_core/hle/pipe.cpp +++ b/src/audio_core/hle/pipe.cpp @@ -12,12 +12,14 @@ #include "common/common_types.h" #include "common/logging/log.h" +#include "core/hle/service/dsp_dsp.h" + namespace DSP { namespace HLE { static DspState dsp_state = DspState::Off; -static std::array<std::vector<u8>, static_cast<size_t>(DspPipe::DspPipe_MAX)> pipe_data; +static std::array<std::vector<u8>, NUM_DSP_PIPE> pipe_data; void ResetPipes() { for (auto& data : pipe_data) { @@ -27,16 +29,18 @@ void ResetPipes() { } std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) { - if (pipe_number >= DspPipe::DspPipe_MAX) { - LOG_ERROR(Audio_DSP, "pipe_number = %u invalid", pipe_number); + const size_t pipe_index = static_cast<size_t>(pipe_number); + + if (pipe_index >= NUM_DSP_PIPE) { + LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index); return {}; } - std::vector<u8>& data = pipe_data[static_cast<size_t>(pipe_number)]; + std::vector<u8>& data = pipe_data[pipe_index]; if (length > data.size()) { - LOG_WARNING(Audio_DSP, "pipe_number = %u is out of data, application requested read of %u but %zu remain", - pipe_number, 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 = data.size(); } @@ -49,16 +53,20 @@ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) { } size_t GetPipeReadableSize(DspPipe pipe_number) { - if (pipe_number >= DspPipe::DspPipe_MAX) { - LOG_ERROR(Audio_DSP, "pipe_number = %u invalid", pipe_number); + const size_t pipe_index = static_cast<size_t>(pipe_number); + + if (pipe_index >= NUM_DSP_PIPE) { + LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index); return 0; } - return pipe_data[static_cast<size_t>(pipe_number)].size(); + return pipe_data[pipe_index].size(); } static void WriteU16(DspPipe pipe_number, u16 value) { - std::vector<u8>& data = pipe_data[static_cast<size_t>(pipe_number)]; + const size_t pipe_index = static_cast<size_t>(pipe_number); + + std::vector<u8>& data = pipe_data.at(pipe_index); // Little endian data.emplace_back(value & 0xFF); data.emplace_back(value >> 8); @@ -91,6 +99,8 @@ static void AudioPipeWriteStructAddresses() { for (u16 addr : struct_addresses) { WriteU16(DspPipe::Audio, addr); } + // Signal that we have data on this pipe. + DSP_DSP::SignalPipeInterrupt(DspPipe::Audio); } void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) { @@ -145,7 +155,7 @@ void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) { return; } default: - LOG_CRITICAL(Audio_DSP, "pipe_number = %u unimplemented", pipe_number); + LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented", static_cast<size_t>(pipe_number)); UNIMPLEMENTED(); return; } diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h index 382d35e87..64d97f8ba 100644 --- a/src/audio_core/hle/pipe.h +++ b/src/audio_core/hle/pipe.h @@ -19,9 +19,9 @@ enum class DspPipe { Debug = 0, Dma = 1, Audio = 2, - Binary = 3, - DspPipe_MAX + Binary = 3 }; +constexpr size_t NUM_DSP_PIPE = 8; /** * Read a DSP pipe. diff --git a/src/audio_core/interpolate.cpp b/src/audio_core/interpolate.cpp new file mode 100644 index 000000000..fcd3aa066 --- /dev/null +++ b/src/audio_core/interpolate.cpp @@ -0,0 +1,85 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include "audio_core/interpolate.h" + +#include "common/assert.h" +#include "common/math_util.h" + +namespace AudioInterp { + +// Calculations are done in fixed point with 24 fractional bits. +// (This is not verified. This was chosen for minimal error.) +constexpr u64 scale_factor = 1 << 24; +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) { + ASSERT(rate_multiplier > 0); + + if (input.size() < 2) + return {}; + + StereoBuffer16 output; + output.reserve(static_cast<size_t>(input.size() / rate_multiplier)); + + u64 step_size = static_cast<u64>(rate_multiplier * scale_factor); + + u64 fposition = 0; + const u64 max_fposition = input.size() * scale_factor; + + while (fposition < 1 * scale_factor) { + u64 fraction = fposition & scale_mask; + + output.push_back(fn(fraction, state.xn2, state.xn1, input[0])); + + fposition += step_size; + } + + while (fposition < 2 * scale_factor) { + u64 fraction = fposition & scale_mask; + + output.push_back(fn(fraction, state.xn1, input[0], input[1])); + + fposition += step_size; + } + + while (fposition < max_fposition) { + u64 fraction = fposition & scale_mask; + + size_t index = static_cast<size_t>(fposition / scale_factor); + output.push_back(fn(fraction, input[index - 2], input[index - 1], input[index])); + + fposition += step_size; + } + + state.xn2 = input[input.size() - 2]; + state.xn1 = input[input.size() - 1]; + + return output; +} + +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; + }); +} + +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) + }; + }); +} + +} // namespace AudioInterp diff --git a/src/audio_core/interpolate.h b/src/audio_core/interpolate.h new file mode 100644 index 000000000..a4c0a453d --- /dev/null +++ b/src/audio_core/interpolate.h @@ -0,0 +1,41 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#pragma once + +#include <array> +#include <vector> + +#include "common/common_types.h" + +namespace AudioInterp { + +/// A variable length buffer of signed PCM16 stereo samples. +using StereoBuffer16 = std::vector<std::array<s16, 2>>; + +struct State { + // Two historical samples. + std::array<s16, 2> xn1 = {}; ///< x[n-1] + std::array<s16, 2> xn2 = {}; ///< x[n-2] +}; + +/** + * 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. + * @return The resampled audio buffer. + */ +StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier); + +/** + * 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. + * @return The resampled audio buffer. + */ +StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier); + +} // namespace AudioInterp |