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<std::array<s32, 4>, samples_per_frame>;
*/
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 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 <array>
#include <memory>
#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<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 std::unique_ptr<AudioCore::Sink> sink;
void Init() {
DSP::HLE::ResetPipes();
+ for (auto& source : sources) {
+ source.Reset();
+ }
}
void Shutdown() {
}
bool Tick() {
+ SharedMemory& read = ReadRegion();
+ SharedMemory& write = WriteRegion();
+
+ std::array<QuadFrame32, 3> 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 <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");
+ }
+
+ 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<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 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 <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