diff options
Diffstat (limited to 'src/audio_core/hle/source.cpp')
-rw-r--r-- | src/audio_core/hle/source.cpp | 320 |
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 |