Merge pull request #1734 from MerryMage/dsp-hle-source

DSP/HLE: Implement Source processing

1 files changed, 320 insertions, 0 deletions

diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
new file mode 100644
index 000000000..daaf6e3f3
--- /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