diff options
Diffstat (limited to 'src/audio_core/hle')
-rw-r--r-- | src/audio_core/hle/common.h | 11 | ||||
-rw-r--r-- | src/audio_core/hle/dsp.cpp | 128 | ||||
-rw-r--r-- | src/audio_core/hle/dsp.h | 40 | ||||
-rw-r--r-- | src/audio_core/hle/filter.h | 1 | ||||
-rw-r--r-- | src/audio_core/hle/mixers.cpp | 201 | ||||
-rw-r--r-- | src/audio_core/hle/mixers.h | 63 | ||||
-rw-r--r-- | src/audio_core/hle/pipe.cpp | 41 | ||||
-rw-r--r-- | src/audio_core/hle/pipe.h | 16 | ||||
-rw-r--r-- | src/audio_core/hle/source.cpp | 320 | ||||
-rw-r--r-- | src/audio_core/hle/source.h | 144 |
10 files changed, 909 insertions, 56 deletions
diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h index 37d441eb2..596b67eaf 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. @@ -26,7 +27,7 @@ using QuadFrame32 = std::array<std::array<s32, 4>, AudioCore::samples_per_fram */ template<typename FrameT, typename FilterT> void FilterFrame(FrameT& frame, FilterT& filter) { - std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const typename FrameT::value_type& sample) { + std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const auto& sample) { return filter.ProcessSample(sample); }); } diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp index c89356edc..0640e1eff 100644 --- a/src/audio_core/hle/dsp.cpp +++ b/src/audio_core/hle/dsp.cpp @@ -2,40 +2,138 @@ // Licensed under GPLv2 or any later version // Refer to the license.txt file included. +#include <array> +#include <memory> + #include "audio_core/hle/dsp.h" +#include "audio_core/hle/mixers.h" #include "audio_core/hle/pipe.h" +#include "audio_core/hle/source.h" +#include "audio_core/sink.h" +#include "audio_core/time_stretch.h" namespace DSP { namespace HLE { -SharedMemory g_region0; -SharedMemory g_region1; +// Region management + +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()]; +} + +// 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) +}; +static Mixers mixers; + +static StereoFrame16 GenerateCurrentFrame() { + SharedMemory& read = ReadRegion(); + SharedMemory& write = WriteRegion(); + + std::array<QuadFrame32, 3> intermediate_mixes = {}; + + // 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]); + 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); + + StereoFrame16 output_frame = mixers.GetOutput(); + + // Write current output frame to the shared memory region + for (size_t samplei = 0; samplei < output_frame.size(); samplei++) { + for (size_t channeli = 0; channeli < output_frame[0].size(); channeli++) { + write.final_samples.pcm16[samplei][channeli] = s16_le(output_frame[samplei][channeli]); + } + } + + return output_frame; +} + +// Audio output + +static std::unique_ptr<AudioCore::Sink> sink; +static AudioCore::TimeStretcher time_stretcher; + +static void OutputCurrentFrame(const StereoFrame16& frame) { + time_stretcher.AddSamples(&frame[0][0], frame.size()); + sink->EnqueueSamples(time_stretcher.Process(sink->SamplesInQueue())); +} + +// Public Interface void Init() { DSP::HLE::ResetPipes(); + + for (auto& source : sources) { + source.Reset(); + } + + mixers.Reset(); + + time_stretcher.Reset(); + if (sink) { + time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate()); + } } void Shutdown() { + time_stretcher.Flush(); + while (true) { + std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue()); + if (residual_audio.empty()) + break; + sink->EnqueueSamples(residual_audio); + } } bool Tick() { - return true; -} + StereoFrame16 current_frame = {}; -SharedMemory& CurrentRegion() { - // The region with the higher frame counter is chosen unless there is wraparound. + // TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to shared memory region) + current_frame = GenerateCurrentFrame(); - if (g_region0.frame_counter == 0xFFFFu && g_region1.frame_counter != 0xFFFEu) { - // Wraparound has occured. - return g_region1; - } + OutputCurrentFrame(current_frame); - if (g_region1.frame_counter == 0xFFFFu && g_region0.frame_counter != 0xFFFEu) { - // Wraparound has occured. - return g_region0; - } + return true; +} - return (g_region0.frame_counter > g_region1.frame_counter) ? g_region0 : g_region1; +void SetSink(std::unique_ptr<AudioCore::Sink> sink_) { + sink = std::move(sink_); + time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate()); } } // namespace HLE diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h index c15ef0b7a..9275cd7de 100644 --- a/src/audio_core/hle/dsp.h +++ b/src/audio_core/hle/dsp.h @@ -4,16 +4,22 @@ #pragma once +#include <array> #include <cstddef> +#include <memory> #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" #include "common/common_types.h" #include "common/swap.h" +namespace AudioCore { +class Sink; +} + namespace DSP { namespace HLE { @@ -27,13 +33,8 @@ namespace HLE { // double-buffer. The frame counter is located as the very last u16 of each region and is incremented // each audio tick. -struct SharedMemory; - constexpr VAddr region0_base = 0x1FF50000; -extern SharedMemory g_region0; - constexpr VAddr region1_base = 0x1FF70000; -extern SharedMemory g_region1; /** * The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from @@ -164,9 +165,9 @@ struct SourceConfiguration { float_le rate_multiplier; enum class InterpolationMode : u8 { - None = 0, + Polyphase = 0, Linear = 1, - Polyphase = 2 + None = 2 }; InterpolationMode interpolation_mode; @@ -305,7 +306,7 @@ 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); @@ -313,14 +314,14 @@ 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 previous_buffer_id_dirty; ///< Non-zero when previous_buffer_id changes + 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 previous_buffer_id; ///< Updated when a buffer finishes playing + u16_le current_buffer_id; ///< Updated when a buffer finishes playing INSERT_PADDING_DSPWORDS(1); }; - Status status[AudioCore::num_sources]; + Status status[num_sources]; }; ASSERT_DSP_STRUCT(SourceStatus::Status, 12); @@ -413,7 +414,7 @@ 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 +428,7 @@ 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[samples_per_frame][2]; }; ASSERT_DSP_STRUCT(FinalMixSamples, 640); @@ -437,7 +438,7 @@ 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; @@ -502,6 +503,8 @@ struct SharedMemory { }; 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"); @@ -535,8 +538,11 @@ void Shutdown(); */ bool Tick(); -/// Returns a mutable reference to the current region. Current region is selected based on the frame counter. -SharedMemory& CurrentRegion(); +/** + * Set the output sink. This must be called before calling Tick(). + * @param sink The sink to which audio will be output to. + */ +void SetSink(std::unique_ptr<AudioCore::Sink> sink); } // namespace HLE } // namespace DSP diff --git a/src/audio_core/hle/filter.h b/src/audio_core/hle/filter.h index 75738f600..43d2035cd 100644 --- a/src/audio_core/hle/filter.h +++ b/src/audio_core/hle/filter.h @@ -16,6 +16,7 @@ namespace HLE { /// Preprocessing filters. There is an independent set of filters for each Source. class SourceFilters final { +public: SourceFilters() { Reset(); } /// Reset internal state. diff --git a/src/audio_core/hle/mixers.cpp b/src/audio_core/hle/mixers.cpp new file mode 100644 index 000000000..18335f7f0 --- /dev/null +++ b/src/audio_core/hle/mixers.cpp @@ -0,0 +1,201 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include <cstddef> + +#include "audio_core/hle/common.h" +#include "audio_core/hle/dsp.h" +#include "audio_core/hle/mixers.h" + +#include "common/assert.h" +#include "common/logging/log.h" +#include "common/math_util.h" + +namespace DSP { +namespace HLE { + +void Mixers::Reset() { + current_frame.fill({}); + state = {}; +} + +DspStatus Mixers::Tick(DspConfiguration& config, + const IntermediateMixSamples& read_samples, + IntermediateMixSamples& write_samples, + const std::array<QuadFrame32, 3>& input) +{ + ParseConfig(config); + + AuxReturn(read_samples); + AuxSend(write_samples, input); + + MixCurrentFrame(); + + return GetCurrentStatus(); +} + +void Mixers::ParseConfig(DspConfiguration& config) { + if (!config.dirty_raw) { + return; + } + + if (config.mixer1_enabled_dirty) { + config.mixer1_enabled_dirty.Assign(0); + state.mixer1_enabled = config.mixer1_enabled != 0; + LOG_TRACE(Audio_DSP, "mixers mixer1_enabled = %hu", config.mixer1_enabled); + } + + if (config.mixer2_enabled_dirty) { + config.mixer2_enabled_dirty.Assign(0); + state.mixer2_enabled = config.mixer2_enabled != 0; + LOG_TRACE(Audio_DSP, "mixers mixer2_enabled = %hu", config.mixer2_enabled); + } + + if (config.volume_0_dirty) { + config.volume_0_dirty.Assign(0); + state.intermediate_mixer_volume[0] = config.volume[0]; + LOG_TRACE(Audio_DSP, "mixers volume[0] = %f", config.volume[0]); + } + + if (config.volume_1_dirty) { + config.volume_1_dirty.Assign(0); + state.intermediate_mixer_volume[1] = config.volume[1]; + LOG_TRACE(Audio_DSP, "mixers volume[1] = %f", config.volume[1]); + } + + if (config.volume_2_dirty) { + config.volume_2_dirty.Assign(0); + state.intermediate_mixer_volume[2] = config.volume[2]; + LOG_TRACE(Audio_DSP, "mixers volume[2] = %f", config.volume[2]); + } + + 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)); + } + + 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.) + LOG_TRACE(Audio_DSP, "mixers headphones_connected=%hu", config.headphones_connected); + } + + if (config.dirty_raw) { + LOG_DEBUG(Audio_DSP, "mixers remaining_dirty=%x", config.dirty_raw); + } + + config.dirty_raw = 0; +} + +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])) + }; +} + +void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples) { + // TODO(merry): Limiter. (Currently we're performing final mixing assuming a disabled limiter.) + + 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> { + // Downmix to mono + 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; + + case OutputFormat::Surround: + // 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> { + // 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; + } + + UNREACHABLE_MSG("Invalid output_format %zu", static_cast<size_t>(state.output_format)); +} + +void Mixers::AuxReturn(const IntermediateMixSamples& read_samples) { + // 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]; + } + } + } + + 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]; + } + } + } +} + +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]; + + if (state.mixer1_enabled) { + for (size_t sample = 0; sample < samples_per_frame; sample++) { + for (size_t channel = 0; channel < 4; channel++) { + write_samples.mix1.pcm32[channel][sample] = input[1][sample][channel]; + } + } + } else { + state.intermediate_mix_buffer[1] = input[1]; + } + + if (state.mixer2_enabled) { + for (size_t sample = 0; sample < samples_per_frame; sample++) { + for (size_t channel = 0; channel < 4; channel++) { + write_samples.mix2.pcm32[channel][sample] = input[2][sample][channel]; + } + } + } else { + state.intermediate_mix_buffer[2] = input[2]; + } +} + +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]); + } + + // TODO(merry): Compressor. (We currently assume a disabled compressor.) +} + +DspStatus Mixers::GetCurrentStatus() const { + DspStatus status; + status.unknown = 0; + status.dropped_frames = 0; + return status; +} + +} // namespace HLE +} // namespace DSP diff --git a/src/audio_core/hle/mixers.h b/src/audio_core/hle/mixers.h new file mode 100644 index 000000000..b52952eb5 --- /dev/null +++ b/src/audio_core/hle/mixers.h @@ -0,0 +1,63 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#pragma once + +#include <array> + +#include "audio_core/hle/common.h" +#include "audio_core/hle/dsp.h" + +namespace DSP { +namespace HLE { + +class Mixers final { +public: + Mixers() { + Reset(); + } + + void Reset(); + + DspStatus Tick(DspConfiguration& config, + const IntermediateMixSamples& read_samples, + IntermediateMixSamples& write_samples, + const std::array<QuadFrame32, 3>& input); + + StereoFrame16 GetOutput() const { + return current_frame; + } + +private: + StereoFrame16 current_frame = {}; + + using OutputFormat = DspConfiguration::OutputFormat; + + struct { + std::array<float, 3> intermediate_mixer_volume = {}; + + bool mixer1_enabled = false; + bool mixer2_enabled = false; + std::array<QuadFrame32, 3> intermediate_mix_buffer = {}; + + OutputFormat output_format = OutputFormat::Stereo; + + } state; + + /// INTERNAL: Update our internal state based on the current config. + void ParseConfig(DspConfiguration& config); + /// INTERNAL: Read samples from shared memory that have been modified by the ARM11. + void AuxReturn(const IntermediateMixSamples& read_samples); + /// INTERNAL: Write samples to shared memory for the ARM11 to modify. + 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. + void DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples); + /// INTERNAL: Generate DspStatus based on internal state. + DspStatus GetCurrentStatus() const; +}; + +} // namespace HLE +} // namespace DSP diff --git a/src/audio_core/hle/pipe.cpp b/src/audio_core/hle/pipe.cpp index 9381883b4..44dff1345 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) { @@ -27,17 +29,24 @@ 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)]; + if (length > UINT16_MAX) { // Can only read at most UINT16_MAX from the pipe + LOG_ERROR(Audio_DSP, "length of %u greater than max of %u", length, UINT16_MAX); + return {}; + } + + 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()); - 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()); } if (length == 0) @@ -49,16 +58,20 @@ 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); @@ -86,11 +99,13 @@ static void AudioPipeWriteStructAddresses() { }; // Begin with a u16 denoting the number of structs. - WriteU16(DspPipe::Audio, struct_addresses.size()); + WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size())); // Then write the struct addresses. 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 +160,7 @@ 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..b714c0496 100644 --- a/src/audio_core/hle/pipe.h +++ b/src/audio_core/hle/pipe.h @@ -19,15 +19,19 @@ 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. - * @param pipe_number The Pipe ID - * @param length How much data to request. - * @return The data read from the pipe. The size of this vector can be less than the length requested. + * 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 `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. + * @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. */ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length); diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp new file mode 100644 index 000000000..30552fe26 --- /dev/null +++ b/src/audio_core/hle/source.cpp @@ -0,0 +1,320 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include <algorithm> +#include <array> + +#include "audio_core/codec.h" +#include "audio_core/hle/common.h" +#include "audio_core/hle/source.h" +#include "audio_core/interpolate.h" + +#include "common/assert.h" +#include "common/logging/log.h" + +#include "core/memory.h" + +namespace DSP { +namespace HLE { + +SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) { + ParseConfig(config, adpcm_coeffs); + + if (state.enabled) { + GenerateFrame(); + } + + return GetCurrentStatus(); +} + +void Source::MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const { + if (!state.enabled) + return; + + const std::array<float, 4>& gains = state.gain.at(intermediate_mix_id); + for (size_t samplei = 0; samplei < samples_per_frame; samplei++) { + // Conversion from stereo (current_frame) to quadraphonic (dest) occurs here. + dest[samplei][0] += static_cast<s32>(gains[0] * current_frame[samplei][0]); + dest[samplei][1] += static_cast<s32>(gains[1] * current_frame[samplei][1]); + dest[samplei][2] += static_cast<s32>(gains[2] * current_frame[samplei][0]); + dest[samplei][3] += static_cast<s32>(gains[3] * current_frame[samplei][1]); + } +} + +void Source::Reset() { + current_frame.fill({}); + state = {}; +} + +void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) { + if (!config.dirty_raw) { + return; + } + + if (config.reset_flag) { + config.reset_flag.Assign(0); + Reset(); + LOG_TRACE(Audio_DSP, "source_id=%zu reset", source_id); + } + + if (config.partial_reset_flag) { + config.partial_reset_flag.Assign(0); + state.input_queue = std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder>{}; + LOG_TRACE(Audio_DSP, "source_id=%zu partial_reset", source_id); + } + + if (config.enable_dirty) { + config.enable_dirty.Assign(0); + state.enabled = config.enable != 0; + LOG_TRACE(Audio_DSP, "source_id=%zu enable=%d", source_id, state.enabled); + } + + if (config.sync_dirty) { + config.sync_dirty.Assign(0); + state.sync = config.sync; + LOG_TRACE(Audio_DSP, "source_id=%zu sync=%u", source_id, state.sync); + } + + if (config.rate_multiplier_dirty) { + config.rate_multiplier_dirty.Assign(0); + state.rate_multiplier = config.rate_multiplier; + 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); + state.rate_multiplier = 1.0f; + // Note: Actual firmware starts producing garbage if this occurs. + } + } + + 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); }); + 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); }); + 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); }); + 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); }); + 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()); + } + + if (config.simple_filter_dirty) { + config.simple_filter_dirty.Assign(0); + state.filters.Configure(config.simple_filter); + LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update", source_id); + } + + if (config.biquad_filter_dirty) { + config.biquad_filter_dirty.Assign(0); + state.filters.Configure(config.biquad_filter); + LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update", source_id); + } + + 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)); + } + + 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)); + } + + 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)); + } + + 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); + } + + if (config.buffer_queue_dirty) { + config.buffer_queue_dirty.Assign(0); + 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); + } + } + config.buffers_dirty = 0; + } + + if (config.dirty_raw) { + LOG_DEBUG(Audio_DSP, "source_id=%zu remaining_dirty=%x", source_id, config.dirty_raw); + } + + config.dirty_raw = 0; +} + +void Source::GenerateFrame() { + current_frame.fill({}); + + if (state.current_buffer.empty() && !DequeueBuffer()) { + state.enabled = false; + state.buffer_update = true; + state.current_buffer_id = 0; + return; + } + + size_t frame_position = 0; + + state.current_sample_number = state.next_sample_number; + while (frame_position < current_frame.size()) { + if (state.current_buffer.empty() && !DequeueBuffer()) { + break; + } + + 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); + + frame_position += size_to_copy; + state.next_sample_number += static_cast<u32>(size_to_copy); + } + + state.filters.ProcessFrame(current_frame); +} + + +bool Source::DequeueBuffer() { + ASSERT_MSG(state.current_buffer.empty(), "Shouldn't dequeue; we still have data in current_buffer"); + + if (state.input_queue.empty()) + return false; + + const Buffer buf = state.input_queue.top(); + state.input_queue.pop(); + + if (buf.adpcm_dirty) { + state.adpcm_state.yn1 = buf.adpcm_yn[0]; + state.adpcm_state.yn2 = buf.adpcm_yn[1]; + } + + if (buf.is_looping) { + LOG_ERROR(Audio_DSP, "Looped buffers are unimplemented at the moment"); + } + + const u8* const memory = Memory::GetPhysicalPointer(buf.physical_address); + if (memory) { + const unsigned num_channels = buf.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1; + switch (buf.format) { + case Format::PCM8: + state.current_buffer = Codec::DecodePCM8(num_channels, memory, buf.length); + break; + case Format::PCM16: + state.current_buffer = Codec::DecodePCM16(num_channels, memory, buf.length); + break; + case Format::ADPCM: + DEBUG_ASSERT(num_channels == 1); + 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); + 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); + break; + case InterpolationMode::Linear: + 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); + break; + default: + UNIMPLEMENTED(); + break; + } + + state.current_sample_number = 0; + state.next_sample_number = 0; + state.current_buffer_id = buf.buffer_id; + 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()); + return true; +} + +SourceStatus::Status Source::GetCurrentStatus() { + SourceStatus::Status ret; + + // Applications depend on the correct emulation of + // current_buffer_id_dirty and current_buffer_id to synchronise + // audio with video. + ret.is_enabled = state.enabled; + ret.current_buffer_id_dirty = state.buffer_update ? 1 : 0; + state.buffer_update = false; + ret.current_buffer_id = state.current_buffer_id; + ret.buffer_position = state.current_sample_number; + ret.sync = state.sync; + + return ret; +} + +} // namespace HLE +} // namespace DSP diff --git a/src/audio_core/hle/source.h b/src/audio_core/hle/source.h new file mode 100644 index 000000000..7ee08d424 --- /dev/null +++ b/src/audio_core/hle/source.h @@ -0,0 +1,144 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#pragma once + +#include <array> +#include <queue> +#include <vector> + +#include "audio_core/codec.h" +#include "audio_core/hle/common.h" +#include "audio_core/hle/dsp.h" +#include "audio_core/hle/filter.h" +#include "audio_core/interpolate.h" + +#include "common/common_types.h" + +namespace DSP { +namespace HLE { + +/** + * This module performs: + * - Buffer management + * - Decoding of buffers + * - Buffer resampling and interpolation + * - Per-source filtering (SimpleFilter, BiquadFilter) + * - Per-source gain + * - Other per-source processing + */ +class Source final { +public: + explicit Source(size_t source_id_) : source_id(source_id_) { + Reset(); + } + + /// Resets internal state. + void Reset(); + + /** + * 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. + */ + 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. + * @param dest The QuadFrame32 to mix into. + * @param intermediate_mix_id The id of the intermediate mix whose gains we are using. + */ + void MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const; + +private: + const size_t source_id; + StereoFrame16 current_frame; + + using Format = SourceConfiguration::Configuration::Format; + using InterpolationMode = SourceConfiguration::Configuration::InterpolationMode; + using MonoOrStereo = SourceConfiguration::Configuration::MonoOrStereo; + + /// Internal representation of a buffer for our buffer queue + struct Buffer { + PAddr physical_address; + u32 length; + u8 adpcm_ps; + std::array<u16, 2> adpcm_yn; + bool adpcm_dirty; + bool is_looping; + u16 buffer_id; + + MonoOrStereo mono_or_stereo; + Format format; + + bool from_queue; + }; + + struct BufferOrder { + bool operator() (const Buffer& a, const Buffer& b) const { + // Lower buffer_id comes first. + return a.buffer_id > b.buffer_id; + } + }; + + struct { + + // State variables + + bool enabled = false; + u16 sync = 0; + + // Mixing + + std::array<std::array<float, 4>, 3> gain = {}; + + // Buffer queue + + std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder> input_queue; + MonoOrStereo mono_or_stereo = MonoOrStereo::Mono; + Format format = Format::ADPCM; + + // Current buffer + + u32 current_sample_number = 0; + u32 next_sample_number = 0; + std::vector<std::array<s16, 2>> current_buffer; + + // buffer_id state + + bool buffer_update = false; + u32 current_buffer_id = 0; + + // Decoding state + + std::array<s16, 16> adpcm_coeffs = {}; + Codec::ADPCMState adpcm_state = {}; + + // Resampling state + + float rate_multiplier = 1.0; + InterpolationMode interpolation_mode = InterpolationMode::Polyphase; + AudioInterp::State interp_state = {}; + + // Filter state + + SourceFilters filters; + + } state; + + // Internal functions + + /// INTERNAL: Update our internal state based on the current config. + 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. + bool DequeueBuffer(); + /// INTERNAL: Generates a SourceStatus::Status based on our internal state. + SourceStatus::Status GetCurrentStatus(); +}; + +} // namespace HLE +} // namespace DSP |