DSP/HLE: Implement Source processing
This commit is contained in:
MerryMage
parent
c1f0044a4b
commit
b242bdf945
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@ -4,6 +4,7 @@ set(SRCS
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hle/dsp.cpp
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hle/filter.cpp
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hle/pipe.cpp
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hle/source.cpp
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interpolate.cpp
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sink_details.cpp
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)
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@ -15,6 +16,7 @@ set(HEADERS
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hle/dsp.h
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hle/filter.h
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hle/pipe.h
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hle/source.h
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interpolate.h
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null_sink.h
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sink.h
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@ -27,7 +27,7 @@ using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
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*/
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template<typename FrameT, typename FilterT>
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void FilterFrame(FrameT& frame, FilterT& filter) {
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std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const typename FrameT::value_type& sample) {
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std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const auto& sample) {
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return filter.ProcessSample(sample);
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});
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}
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@ -2,10 +2,12 @@
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <array>
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#include <memory>
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#include "audio_core/hle/dsp.h"
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#include "audio_core/hle/pipe.h"
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#include "audio_core/hle/source.h"
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#include "audio_core/sink.h"
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namespace DSP {
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@ -38,16 +40,38 @@ static SharedMemory& WriteRegion() {
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return g_regions[1 - CurrentRegionIndex()];
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}
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static std::array<Source, num_sources> sources = {
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Source(0), Source(1), Source(2), Source(3), Source(4), Source(5),
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Source(6), Source(7), Source(8), Source(9), Source(10), Source(11),
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Source(12), Source(13), Source(14), Source(15), Source(16), Source(17),
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Source(18), Source(19), Source(20), Source(21), Source(22), Source(23)
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};
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static std::unique_ptr<AudioCore::Sink> sink;
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void Init() {
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DSP::HLE::ResetPipes();
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for (auto& source : sources) {
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source.Reset();
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}
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}
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void Shutdown() {
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}
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bool Tick() {
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SharedMemory& read = ReadRegion();
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SharedMemory& write = WriteRegion();
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std::array<QuadFrame32, 3> intermediate_mixes = {};
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for (size_t i = 0; i < num_sources; i++) {
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write.source_statuses.status[i] = sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
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for (size_t mix = 0; mix < 3; mix++) {
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sources[i].MixInto(intermediate_mixes[mix], mix);
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}
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}
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return true;
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}
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@ -169,9 +169,9 @@ struct SourceConfiguration {
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float_le rate_multiplier;
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enum class InterpolationMode : u8 {
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None = 0,
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Polyphase = 0,
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Linear = 1,
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Polyphase = 2
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None = 2
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};
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InterpolationMode interpolation_mode;
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@ -318,10 +318,10 @@ ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
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struct SourceStatus {
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struct Status {
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u8 is_enabled; ///< Is this channel enabled? (Doesn't have to be playing anything.)
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u8 previous_buffer_id_dirty; ///< Non-zero when previous_buffer_id changes
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u8 current_buffer_id_dirty; ///< Non-zero when current_buffer_id changes
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u16_le sync; ///< Is set by the DSP to the value of SourceConfiguration::sync
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u32_dsp buffer_position; ///< Number of samples into the current buffer
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u16_le previous_buffer_id; ///< Updated when a buffer finishes playing
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u16_le current_buffer_id; ///< Updated when a buffer finishes playing
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INSERT_PADDING_DSPWORDS(1);
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};
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@ -16,6 +16,7 @@ namespace HLE {
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/// Preprocessing filters. There is an independent set of filters for each Source.
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class SourceFilters final {
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public:
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SourceFilters() { Reset(); }
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/// Reset internal state.
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@ -0,0 +1,320 @@
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// Copyright 2016 Citra Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <algorithm>
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#include <array>
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#include "audio_core/codec.h"
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#include "audio_core/hle/common.h"
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#include "audio_core/hle/source.h"
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#include "audio_core/interpolate.h"
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#include "common/assert.h"
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#include "common/logging/log.h"
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#include "core/memory.h"
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namespace DSP {
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namespace HLE {
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SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
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ParseConfig(config, adpcm_coeffs);
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if (state.enabled) {
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GenerateFrame();
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}
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return GetCurrentStatus();
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}
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void Source::MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const {
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if (!state.enabled)
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return;
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const std::array<float, 4>& gains = state.gain.at(intermediate_mix_id);
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for (size_t samplei = 0; samplei < samples_per_frame; samplei++) {
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// Conversion from stereo (current_frame) to quadraphonic (dest) occurs here.
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dest[samplei][0] += static_cast<s32>(gains[0] * current_frame[samplei][0]);
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dest[samplei][1] += static_cast<s32>(gains[1] * current_frame[samplei][1]);
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dest[samplei][2] += static_cast<s32>(gains[2] * current_frame[samplei][0]);
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dest[samplei][3] += static_cast<s32>(gains[3] * current_frame[samplei][1]);
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}
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}
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void Source::Reset() {
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current_frame.fill({});
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state = {};
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}
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void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
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if (!config.dirty_raw) {
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return;
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}
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if (config.reset_flag) {
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config.reset_flag.Assign(0);
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Reset();
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LOG_TRACE(Audio_DSP, "source_id=%zu reset", source_id);
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}
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if (config.partial_reset_flag) {
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config.partial_reset_flag.Assign(0);
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state.input_queue = std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder>{};
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LOG_TRACE(Audio_DSP, "source_id=%zu partial_reset", source_id);
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}
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if (config.enable_dirty) {
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config.enable_dirty.Assign(0);
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state.enabled = config.enable != 0;
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LOG_TRACE(Audio_DSP, "source_id=%zu enable=%d", source_id, state.enabled);
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}
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if (config.sync_dirty) {
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config.sync_dirty.Assign(0);
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state.sync = config.sync;
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LOG_TRACE(Audio_DSP, "source_id=%zu sync=%u", source_id, state.sync);
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}
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if (config.rate_multiplier_dirty) {
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config.rate_multiplier_dirty.Assign(0);
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state.rate_multiplier = config.rate_multiplier;
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LOG_TRACE(Audio_DSP, "source_id=%zu rate=%f", source_id, state.rate_multiplier);
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if (state.rate_multiplier <= 0) {
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LOG_ERROR(Audio_DSP, "Was given an invalid rate multiplier: source_id=%zu rate=%f", source_id, state.rate_multiplier);
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state.rate_multiplier = 1.0f;
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// Note: Actual firmware starts producing garbage if this occurs.
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}
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}
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if (config.adpcm_coefficients_dirty) {
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config.adpcm_coefficients_dirty.Assign(0);
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std::transform(adpcm_coeffs, adpcm_coeffs + state.adpcm_coeffs.size(), state.adpcm_coeffs.begin(),
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[](const auto& coeff) { return static_cast<s16>(coeff); });
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LOG_TRACE(Audio_DSP, "source_id=%zu adpcm update", source_id);
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}
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if (config.gain_0_dirty) {
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config.gain_0_dirty.Assign(0);
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std::transform(config.gain[0], config.gain[0] + state.gain[0].size(), state.gain[0].begin(),
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[](const auto& coeff) { return static_cast<float>(coeff); });
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LOG_TRACE(Audio_DSP, "source_id=%zu gain 0 update", source_id);
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}
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if (config.gain_1_dirty) {
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config.gain_1_dirty.Assign(0);
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std::transform(config.gain[1], config.gain[1] + state.gain[1].size(), state.gain[1].begin(),
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[](const auto& coeff) { return static_cast<float>(coeff); });
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LOG_TRACE(Audio_DSP, "source_id=%zu gain 1 update", source_id);
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}
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if (config.gain_2_dirty) {
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config.gain_2_dirty.Assign(0);
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std::transform(config.gain[2], config.gain[2] + state.gain[2].size(), state.gain[2].begin(),
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[](const auto& coeff) { return static_cast<float>(coeff); });
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LOG_TRACE(Audio_DSP, "source_id=%zu gain 2 update", source_id);
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}
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if (config.filters_enabled_dirty) {
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config.filters_enabled_dirty.Assign(0);
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state.filters.Enable(config.simple_filter_enabled.ToBool(), config.biquad_filter_enabled.ToBool());
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LOG_TRACE(Audio_DSP, "source_id=%zu enable_simple=%hu enable_biquad=%hu",
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source_id, config.simple_filter_enabled.Value(), config.biquad_filter_enabled.Value());
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}
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if (config.simple_filter_dirty) {
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config.simple_filter_dirty.Assign(0);
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state.filters.Configure(config.simple_filter);
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LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update");
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}
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if (config.biquad_filter_dirty) {
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config.biquad_filter_dirty.Assign(0);
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state.filters.Configure(config.biquad_filter);
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LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update");
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}
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if (config.interpolation_dirty) {
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config.interpolation_dirty.Assign(0);
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state.interpolation_mode = config.interpolation_mode;
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LOG_TRACE(Audio_DSP, "source_id=%zu interpolation_mode=%zu", source_id, static_cast<size_t>(state.interpolation_mode));
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}
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if (config.format_dirty || config.embedded_buffer_dirty) {
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config.format_dirty.Assign(0);
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state.format = config.format;
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LOG_TRACE(Audio_DSP, "source_id=%zu format=%zu", source_id, static_cast<size_t>(state.format));
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}
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if (config.mono_or_stereo_dirty || config.embedded_buffer_dirty) {
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config.mono_or_stereo_dirty.Assign(0);
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state.mono_or_stereo = config.mono_or_stereo;
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LOG_TRACE(Audio_DSP, "source_id=%zu mono_or_stereo=%zu", source_id, static_cast<size_t>(state.mono_or_stereo));
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}
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if (config.embedded_buffer_dirty) {
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config.embedded_buffer_dirty.Assign(0);
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state.input_queue.emplace(Buffer{
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config.physical_address,
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config.length,
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static_cast<u8>(config.adpcm_ps),
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{ config.adpcm_yn[0], config.adpcm_yn[1] },
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config.adpcm_dirty.ToBool(),
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config.is_looping.ToBool(),
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config.buffer_id,
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state.mono_or_stereo,
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state.format,
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false
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});
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LOG_TRACE(Audio_DSP, "enqueuing embedded addr=0x%08x len=%u id=%hu", config.physical_address, config.length, config.buffer_id);
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}
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if (config.buffer_queue_dirty) {
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config.buffer_queue_dirty.Assign(0);
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for (size_t i = 0; i < 4; i++) {
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if (config.buffers_dirty & (1 << i)) {
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const auto& b = config.buffers[i];
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state.input_queue.emplace(Buffer{
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b.physical_address,
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b.length,
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static_cast<u8>(b.adpcm_ps),
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{ b.adpcm_yn[0], b.adpcm_yn[1] },
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b.adpcm_dirty != 0,
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b.is_looping != 0,
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b.buffer_id,
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state.mono_or_stereo,
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state.format,
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true
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});
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LOG_TRACE(Audio_DSP, "enqueuing queued %zu addr=0x%08x len=%u id=%hu", i, b.physical_address, b.length, b.buffer_id);
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}
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}
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config.buffers_dirty = 0;
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}
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if (config.dirty_raw) {
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LOG_DEBUG(Audio_DSP, "source_id=%zu remaining_dirty=%x", source_id, config.dirty_raw);
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}
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config.dirty_raw = 0;
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}
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void Source::GenerateFrame() {
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current_frame.fill({});
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if (state.current_buffer.empty() && !DequeueBuffer()) {
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state.enabled = false;
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state.buffer_update = true;
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state.current_buffer_id = 0;
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return;
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}
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size_t frame_position = 0;
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state.current_sample_number = state.next_sample_number;
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while (frame_position < current_frame.size()) {
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if (state.current_buffer.empty() && !DequeueBuffer()) {
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break;
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}
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const size_t size_to_copy = std::min(state.current_buffer.size(), current_frame.size() - frame_position);
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std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy, current_frame.begin() + frame_position);
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state.current_buffer.erase(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy);
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frame_position += size_to_copy;
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state.next_sample_number += static_cast<u32>(size_to_copy);
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}
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state.filters.ProcessFrame(current_frame);
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}
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bool Source::DequeueBuffer() {
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ASSERT_MSG(state.current_buffer.empty(), "Shouldn't dequeue; we still have data in current_buffer");
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if (state.input_queue.empty())
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return false;
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const Buffer buf = state.input_queue.top();
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state.input_queue.pop();
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if (buf.adpcm_dirty) {
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state.adpcm_state.yn1 = buf.adpcm_yn[0];
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state.adpcm_state.yn2 = buf.adpcm_yn[1];
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}
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if (buf.is_looping) {
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LOG_ERROR(Audio_DSP, "Looped buffers are unimplemented at the moment");
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}
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const u8* const memory = Memory::GetPhysicalPointer(buf.physical_address);
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if (memory) {
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const unsigned num_channels = buf.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1;
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switch (buf.format) {
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case Format::PCM8:
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state.current_buffer = Codec::DecodePCM8(num_channels, memory, buf.length);
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break;
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case Format::PCM16:
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state.current_buffer = Codec::DecodePCM16(num_channels, memory, buf.length);
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break;
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case Format::ADPCM:
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DEBUG_ASSERT(num_channels == 1);
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state.current_buffer = Codec::DecodeADPCM(memory, buf.length, state.adpcm_coeffs, state.adpcm_state);
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break;
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default:
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UNIMPLEMENTED();
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break;
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}
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} else {
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LOG_WARNING(Audio_DSP, "source_id=%zu buffer_id=%hu length=%u: Invalid physical address 0x%08X",
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source_id, buf.buffer_id, buf.length, buf.physical_address);
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state.current_buffer.clear();
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return true;
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}
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switch (state.interpolation_mode) {
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case InterpolationMode::None:
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state.current_buffer = AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
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break;
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case InterpolationMode::Linear:
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state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
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break;
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case InterpolationMode::Polyphase:
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// TODO(merry): Implement polyphase interpolation
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state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
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break;
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default:
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UNIMPLEMENTED();
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break;
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}
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state.current_sample_number = 0;
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state.next_sample_number = 0;
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state.current_buffer_id = buf.buffer_id;
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state.buffer_update = buf.from_queue;
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LOG_TRACE(Audio_DSP, "source_id=%zu buffer_id=%hu from_queue=%s current_buffer.size()=%zu",
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source_id, buf.buffer_id, buf.from_queue ? "true" : "false", state.current_buffer.size());
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return true;
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}
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SourceStatus::Status Source::GetCurrentStatus() {
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SourceStatus::Status ret;
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// Applications depend on the correct emulation of
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// current_buffer_id_dirty and current_buffer_id to synchronise
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// audio with video.
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ret.is_enabled = state.enabled;
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ret.current_buffer_id_dirty = state.buffer_update ? 1 : 0;
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state.buffer_update = false;
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ret.current_buffer_id = state.current_buffer_id;
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ret.buffer_position = state.current_sample_number;
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ret.sync = state.sync;
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return ret;
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}
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} // namespace HLE
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} // namespace DSP
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@ -0,0 +1,144 @@
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// Copyright 2016 Citra Emulator Project
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||||
// 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
|
||||
Reference in New Issue