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-rw-r--r--src/audio_core/hle/source.cpp320
1 files changed, 320 insertions, 0 deletions
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
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+++ b/src/audio_core/hle/source.cpp
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+// 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