From b242bdf9458642201bab4f1f884556ef73051554 Mon Sep 17 00:00:00 2001 From: MerryMage Date: Mon, 25 Apr 2016 08:54:57 +0100 Subject: [PATCH] DSP/HLE: Implement Source processing --- src/audio_core/CMakeLists.txt | 2 + src/audio_core/hle/common.h | 2 +- src/audio_core/hle/dsp.cpp | 24 +++ src/audio_core/hle/dsp.h | 8 +- src/audio_core/hle/filter.h | 1 + src/audio_core/hle/source.cpp | 320 ++++++++++++++++++++++++++++++++++ src/audio_core/hle/source.h | 144 +++++++++++++++ 7 files changed, 496 insertions(+), 5 deletions(-) create mode 100644 src/audio_core/hle/source.cpp create mode 100644 src/audio_core/hle/source.h diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt index 5a2747e78d..4cd7aba672 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 + hle/source.cpp interpolate.cpp sink_details.cpp ) @@ -15,6 +16,7 @@ set(HEADERS hle/dsp.h hle/filter.h hle/pipe.h + hle/source.h interpolate.h null_sink.h sink.h diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h index 7910f42ae2..596b67eafc 100644 --- a/src/audio_core/hle/common.h +++ b/src/audio_core/hle/common.h @@ -27,7 +27,7 @@ using QuadFrame32 = std::array, samples_per_frame>; */ template 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 4d44bd2d93..0cdbdb06ab 100644 --- a/src/audio_core/hle/dsp.cpp +++ b/src/audio_core/hle/dsp.cpp @@ -2,10 +2,12 @@ // Licensed under GPLv2 or any later version // Refer to the license.txt file included. +#include #include #include "audio_core/hle/dsp.h" #include "audio_core/hle/pipe.h" +#include "audio_core/hle/source.h" #include "audio_core/sink.h" namespace DSP { @@ -38,16 +40,38 @@ static SharedMemory& WriteRegion() { return g_regions[1 - CurrentRegionIndex()]; } +static std::array 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 std::unique_ptr sink; void Init() { DSP::HLE::ResetPipes(); + for (auto& source : sources) { + source.Reset(); + } } void Shutdown() { } bool Tick() { + SharedMemory& read = ReadRegion(); + SharedMemory& write = WriteRegion(); + + std::array 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); + } + } + return true; } diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h index 4f2410c27a..4459a5668d 100644 --- a/src/audio_core/hle/dsp.h +++ b/src/audio_core/hle/dsp.h @@ -169,9 +169,9 @@ struct SourceConfiguration { float_le rate_multiplier; enum class InterpolationMode : u8 { - None = 0, + Polyphase = 0, Linear = 1, - Polyphase = 2 + None = 2 }; InterpolationMode interpolation_mode; @@ -318,10 +318,10 @@ 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); }; diff --git a/src/audio_core/hle/filter.h b/src/audio_core/hle/filter.h index 75738f600e..43d2035cd1 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/source.cpp b/src/audio_core/hle/source.cpp new file mode 100644 index 0000000000..daaf6e3f3a --- /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 +#include + +#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& 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(gains[0] * current_frame[samplei][0]); + dest[samplei][1] += static_cast(gains[1] * current_frame[samplei][1]); + dest[samplei][2] += static_cast(gains[2] * current_frame[samplei][0]); + dest[samplei][3] += static_cast(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, 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(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(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(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(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"); + } + + 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"); + } + + 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(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(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(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(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(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(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 0000000000..7ee08d424e --- /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 +#include +#include + +#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 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, 3> gain = {}; + + // Buffer queue + + std::priority_queue, 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> current_buffer; + + // buffer_id state + + bool buffer_update = false; + u32 current_buffer_id = 0; + + // Decoding state + + std::array 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