mirror of https://git.suyu.dev/suyu/suyu
commit
bb21c2198a
@ -0,0 +1,234 @@
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// Copyright 2018 yuzu 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 "audio_core/audio_renderer.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 AudioCore {
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constexpr u32 STREAM_SAMPLE_RATE{48000};
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constexpr u32 STREAM_NUM_CHANNELS{2};
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AudioRenderer::AudioRenderer(AudioRendererParameter params,
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Kernel::SharedPtr<Kernel::Event> buffer_event)
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: worker_params{params}, buffer_event{buffer_event}, voices(params.voice_count) {
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audio_core = std::make_unique<AudioCore::AudioOut>();
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stream = audio_core->OpenStream(STREAM_SAMPLE_RATE, STREAM_NUM_CHANNELS, "AudioRenderer",
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[=]() { buffer_event->Signal(); });
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audio_core->StartStream(stream);
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QueueMixedBuffer(0);
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QueueMixedBuffer(1);
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QueueMixedBuffer(2);
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}
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std::vector<u8> AudioRenderer::UpdateAudioRenderer(const std::vector<u8>& input_params) {
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// Copy UpdateDataHeader struct
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UpdateDataHeader config{};
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std::memcpy(&config, input_params.data(), sizeof(UpdateDataHeader));
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u32 memory_pool_count = worker_params.effect_count + (worker_params.voice_count * 4);
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// Copy MemoryPoolInfo structs
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std::vector<MemoryPoolInfo> mem_pool_info(memory_pool_count);
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std::memcpy(mem_pool_info.data(),
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input_params.data() + sizeof(UpdateDataHeader) + config.behavior_size,
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memory_pool_count * sizeof(MemoryPoolInfo));
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// Copy VoiceInfo structs
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size_t offset{sizeof(UpdateDataHeader) + config.behavior_size + config.memory_pools_size +
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config.voice_resource_size};
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for (auto& voice : voices) {
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std::memcpy(&voice.Info(), input_params.data() + offset, sizeof(VoiceInfo));
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offset += sizeof(VoiceInfo);
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}
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// Update voices
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for (auto& voice : voices) {
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voice.UpdateState();
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if (!voice.GetInfo().is_in_use) {
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continue;
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}
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if (voice.GetInfo().is_new) {
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voice.SetWaveIndex(voice.GetInfo().wave_buffer_head);
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}
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}
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// Update memory pool state
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std::vector<MemoryPoolEntry> memory_pool(memory_pool_count);
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for (size_t index = 0; index < memory_pool.size(); ++index) {
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if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestAttach) {
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memory_pool[index].state = MemoryPoolStates::Attached;
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} else if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestDetach) {
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memory_pool[index].state = MemoryPoolStates::Detached;
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}
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}
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// Release previous buffers and queue next ones for playback
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ReleaseAndQueueBuffers();
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// Copy output header
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UpdateDataHeader response_data{worker_params};
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std::vector<u8> output_params(response_data.total_size);
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std::memcpy(output_params.data(), &response_data, sizeof(UpdateDataHeader));
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// Copy output memory pool entries
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std::memcpy(output_params.data() + sizeof(UpdateDataHeader), memory_pool.data(),
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response_data.memory_pools_size);
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// Copy output voice status
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size_t voice_out_status_offset{sizeof(UpdateDataHeader) + response_data.memory_pools_size};
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for (const auto& voice : voices) {
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std::memcpy(output_params.data() + voice_out_status_offset, &voice.GetOutStatus(),
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sizeof(VoiceOutStatus));
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voice_out_status_offset += sizeof(VoiceOutStatus);
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}
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return output_params;
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}
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void AudioRenderer::VoiceState::SetWaveIndex(size_t index) {
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wave_index = index & 3;
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is_refresh_pending = true;
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}
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std::vector<s16> AudioRenderer::VoiceState::DequeueSamples(size_t sample_count) {
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if (!IsPlaying()) {
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return {};
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}
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if (is_refresh_pending) {
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RefreshBuffer();
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}
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const size_t max_size{samples.size() - offset};
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const size_t dequeue_offset{offset};
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size_t size{sample_count * STREAM_NUM_CHANNELS};
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if (size > max_size) {
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size = max_size;
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}
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out_status.played_sample_count += size / STREAM_NUM_CHANNELS;
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offset += size;
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const auto& wave_buffer{info.wave_buffer[wave_index]};
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if (offset == samples.size()) {
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offset = 0;
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if (!wave_buffer.is_looping) {
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SetWaveIndex(wave_index + 1);
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}
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out_status.wave_buffer_consumed++;
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if (wave_buffer.end_of_stream) {
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info.play_state = PlayState::Paused;
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}
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}
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return {samples.begin() + dequeue_offset, samples.begin() + dequeue_offset + size};
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}
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void AudioRenderer::VoiceState::UpdateState() {
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if (is_in_use && !info.is_in_use) {
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// No longer in use, reset state
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is_refresh_pending = true;
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wave_index = 0;
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offset = 0;
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out_status = {};
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}
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is_in_use = info.is_in_use;
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}
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void AudioRenderer::VoiceState::RefreshBuffer() {
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std::vector<s16> new_samples(info.wave_buffer[wave_index].buffer_sz / sizeof(s16));
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Memory::ReadBlock(info.wave_buffer[wave_index].buffer_addr, new_samples.data(),
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info.wave_buffer[wave_index].buffer_sz);
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switch (static_cast<Codec::PcmFormat>(info.sample_format)) {
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case Codec::PcmFormat::Int16: {
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// PCM16 is played as-is
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break;
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}
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case Codec::PcmFormat::Adpcm: {
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// Decode ADPCM to PCM16
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Codec::ADPCM_Coeff coeffs;
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Memory::ReadBlock(info.additional_params_addr, coeffs.data(), sizeof(Codec::ADPCM_Coeff));
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new_samples = Codec::DecodeADPCM(reinterpret_cast<u8*>(new_samples.data()),
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new_samples.size() * sizeof(s16), coeffs, adpcm_state);
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break;
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}
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default:
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LOG_CRITICAL(Audio, "Unimplemented sample_format={}", info.sample_format);
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UNREACHABLE();
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break;
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}
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switch (info.channel_count) {
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case 1:
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// 1 channel is upsampled to 2 channel
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samples.resize(new_samples.size() * 2);
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for (size_t index = 0; index < new_samples.size(); ++index) {
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samples[index * 2] = new_samples[index];
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samples[index * 2 + 1] = new_samples[index];
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}
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break;
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case 2: {
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// 2 channel is played as is
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samples = std::move(new_samples);
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break;
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}
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default:
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LOG_CRITICAL(Audio, "Unimplemented channel_count={}", info.channel_count);
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UNREACHABLE();
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break;
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}
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is_refresh_pending = false;
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}
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static constexpr s16 ClampToS16(s32 value) {
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return static_cast<s16>(std::clamp(value, -32768, 32767));
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}
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void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
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constexpr size_t BUFFER_SIZE{512};
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std::vector<s16> buffer(BUFFER_SIZE * stream->GetNumChannels());
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for (auto& voice : voices) {
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if (!voice.IsPlaying()) {
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continue;
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}
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size_t offset{};
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s64 samples_remaining{BUFFER_SIZE};
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while (samples_remaining > 0) {
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const std::vector<s16> samples{voice.DequeueSamples(samples_remaining)};
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if (samples.empty()) {
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break;
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}
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samples_remaining -= samples.size();
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for (const auto& sample : samples) {
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const s32 buffer_sample{buffer[offset]};
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buffer[offset++] =
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ClampToS16(buffer_sample + static_cast<s32>(sample * voice.GetInfo().volume));
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}
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}
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}
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audio_core->QueueBuffer(stream, tag, std::move(buffer));
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}
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void AudioRenderer::ReleaseAndQueueBuffers() {
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const auto released_buffers{audio_core->GetTagsAndReleaseBuffers(stream, 2)};
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for (const auto& tag : released_buffers) {
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QueueMixedBuffer(tag);
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}
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}
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} // namespace AudioCore
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// Copyright 2018 yuzu 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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#pragma once
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#include <array>
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#include <memory>
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#include <vector>
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#include "audio_core/audio_out.h"
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#include "audio_core/codec.h"
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#include "audio_core/stream.h"
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#include "common/common_types.h"
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#include "common/swap.h"
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#include "core/hle/kernel/event.h"
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namespace AudioCore {
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enum class PlayState : u8 {
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Started = 0,
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Stopped = 1,
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Paused = 2,
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};
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struct AudioRendererParameter {
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u32_le sample_rate;
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u32_le sample_count;
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u32_le unknown_8;
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u32_le unknown_c;
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u32_le voice_count;
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u32_le sink_count;
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u32_le effect_count;
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u32_le unknown_1c;
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u8 unknown_20;
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INSERT_PADDING_BYTES(3);
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u32_le splitter_count;
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u32_le unknown_2c;
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INSERT_PADDING_WORDS(1);
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u32_le revision;
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};
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static_assert(sizeof(AudioRendererParameter) == 52, "AudioRendererParameter is an invalid size");
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enum class MemoryPoolStates : u32 { // Should be LE
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Invalid = 0x0,
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Unknown = 0x1,
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RequestDetach = 0x2,
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Detached = 0x3,
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RequestAttach = 0x4,
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Attached = 0x5,
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Released = 0x6,
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};
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struct MemoryPoolEntry {
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MemoryPoolStates state;
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u32_le unknown_4;
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u32_le unknown_8;
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u32_le unknown_c;
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};
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static_assert(sizeof(MemoryPoolEntry) == 0x10, "MemoryPoolEntry has wrong size");
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struct MemoryPoolInfo {
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u64_le pool_address;
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u64_le pool_size;
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MemoryPoolStates pool_state;
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INSERT_PADDING_WORDS(3); // Unknown
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};
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static_assert(sizeof(MemoryPoolInfo) == 0x20, "MemoryPoolInfo has wrong size");
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struct BiquadFilter {
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u8 enable;
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INSERT_PADDING_BYTES(1);
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std::array<s16_le, 3> numerator;
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std::array<s16_le, 2> denominator;
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};
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static_assert(sizeof(BiquadFilter) == 0xc, "BiquadFilter has wrong size");
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struct WaveBuffer {
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u64_le buffer_addr;
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u64_le buffer_sz;
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s32_le start_sample_offset;
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s32_le end_sample_offset;
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u8 is_looping;
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u8 end_of_stream;
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u8 sent_to_server;
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INSERT_PADDING_BYTES(5);
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u64 context_addr;
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u64 context_sz;
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INSERT_PADDING_BYTES(8);
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};
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static_assert(sizeof(WaveBuffer) == 0x38, "WaveBuffer has wrong size");
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struct VoiceInfo {
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u32_le id;
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u32_le node_id;
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u8 is_new;
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u8 is_in_use;
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PlayState play_state;
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u8 sample_format;
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u32_le sample_rate;
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u32_le priority;
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u32_le sorting_order;
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u32_le channel_count;
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float_le pitch;
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float_le volume;
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std::array<BiquadFilter, 2> biquad_filter;
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u32_le wave_buffer_count;
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u32_le wave_buffer_head;
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INSERT_PADDING_WORDS(1);
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u64_le additional_params_addr;
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u64_le additional_params_sz;
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u32_le mix_id;
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u32_le splitter_info_id;
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std::array<WaveBuffer, 4> wave_buffer;
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std::array<u32_le, 6> voice_channel_resource_ids;
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INSERT_PADDING_BYTES(24);
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};
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static_assert(sizeof(VoiceInfo) == 0x170, "VoiceInfo is wrong size");
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struct VoiceOutStatus {
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u64_le played_sample_count;
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u32_le wave_buffer_consumed;
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u32_le voice_drops_count;
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};
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static_assert(sizeof(VoiceOutStatus) == 0x10, "VoiceOutStatus has wrong size");
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struct UpdateDataHeader {
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UpdateDataHeader() {}
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explicit UpdateDataHeader(const AudioRendererParameter& config) {
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revision = Common::MakeMagic('R', 'E', 'V', '4'); // 5.1.0 Revision
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behavior_size = 0xb0;
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memory_pools_size = (config.effect_count + (config.voice_count * 4)) * 0x10;
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voices_size = config.voice_count * 0x10;
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voice_resource_size = 0x0;
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effects_size = config.effect_count * 0x10;
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mixes_size = 0x0;
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sinks_size = config.sink_count * 0x20;
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performance_manager_size = 0x10;
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total_size = sizeof(UpdateDataHeader) + behavior_size + memory_pools_size + voices_size +
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effects_size + sinks_size + performance_manager_size;
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}
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u32_le revision;
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u32_le behavior_size;
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u32_le memory_pools_size;
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u32_le voices_size;
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u32_le voice_resource_size;
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u32_le effects_size;
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u32_le mixes_size;
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u32_le sinks_size;
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u32_le performance_manager_size;
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INSERT_PADDING_WORDS(6);
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u32_le total_size;
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};
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static_assert(sizeof(UpdateDataHeader) == 0x40, "UpdateDataHeader has wrong size");
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class AudioRenderer {
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public:
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AudioRenderer(AudioRendererParameter params, Kernel::SharedPtr<Kernel::Event> buffer_event);
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std::vector<u8> UpdateAudioRenderer(const std::vector<u8>& input_params);
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void QueueMixedBuffer(Buffer::Tag tag);
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void ReleaseAndQueueBuffers();
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private:
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class VoiceState {
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public:
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bool IsPlaying() const {
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return is_in_use && info.play_state == PlayState::Started;
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}
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const VoiceOutStatus& GetOutStatus() const {
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return out_status;
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}
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const VoiceInfo& GetInfo() const {
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return info;
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}
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VoiceInfo& Info() {
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return info;
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}
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void SetWaveIndex(size_t index);
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std::vector<s16> DequeueSamples(size_t sample_count);
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void UpdateState();
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void RefreshBuffer();
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private:
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bool is_in_use{};
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bool is_refresh_pending{};
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size_t wave_index{};
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size_t offset{};
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Codec::ADPCMState adpcm_state{};
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std::vector<s16> samples;
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VoiceOutStatus out_status{};
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VoiceInfo info{};
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};
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AudioRendererParameter worker_params;
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Kernel::SharedPtr<Kernel::Event> buffer_event;
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std::vector<VoiceState> voices;
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std::unique_ptr<AudioCore::AudioOut> audio_core;
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AudioCore::StreamPtr stream;
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};
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} // namespace AudioCore
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@ -0,0 +1,77 @@
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// Copyright 2018 yuzu 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 "audio_core/codec.h"
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namespace AudioCore::Codec {
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std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
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ADPCMState& state) {
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// GC-ADPCM with scale factor and variable coefficients.
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// Frames are 8 bytes long containing 14 samples each.
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// Samples are 4 bits (one nibble) long.
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constexpr size_t FRAME_LEN = 8;
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constexpr size_t SAMPLES_PER_FRAME = 14;
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constexpr std::array<int, 16> SIGNED_NIBBLES = {
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{0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
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|
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const size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
|
||||
const size_t ret_size =
|
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sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
|
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std::vector<s16> ret(ret_size);
|
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|
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int yn1 = state.yn1, yn2 = state.yn2;
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const size_t NUM_FRAMES =
|
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(sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
|
||||
for (size_t framei = 0; framei < NUM_FRAMES; framei++) {
|
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const int frame_header = data[framei * FRAME_LEN];
|
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const int scale = 1 << (frame_header & 0xF);
|
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const int idx = (frame_header >> 4) & 0x7;
|
||||
|
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// Coefficients are fixed point with 11 bits fractional part.
|
||||
const int coef1 = coeff[idx * 2 + 0];
|
||||
const int coef2 = coeff[idx * 2 + 1];
|
||||
|
||||
// Decodes an audio sample. One nibble produces one sample.
|
||||
const auto decode_sample = [&](const int nibble) -> s16 {
|
||||
const int xn = nibble * scale;
|
||||
// We first transform everything into 11 bit fixed point, perform the second order
|
||||
// digital filter, then transform back.
|
||||
// 0x400 == 0.5 in 11 bit fixed point.
|
||||
// Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
|
||||
int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
|
||||
// Clamp to output range.
|
||||
val = std::clamp<s32>(val, -32768, 32767);
|
||||
// Advance output feedback.
|
||||
yn2 = yn1;
|
||||
yn1 = val;
|
||||
return static_cast<s16>(val);
|
||||
};
|
||||
|
||||
size_t outputi = framei * SAMPLES_PER_FRAME;
|
||||
size_t datai = framei * FRAME_LEN + 1;
|
||||
for (size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
|
||||
const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
|
||||
ret[outputi] = sample1;
|
||||
outputi++;
|
||||
|
||||
const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]);
|
||||
ret[outputi] = sample2;
|
||||
outputi++;
|
||||
|
||||
datai++;
|
||||
}
|
||||
}
|
||||
|
||||
state.yn1 = yn1;
|
||||
state.yn2 = yn2;
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
} // namespace AudioCore::Codec
|
@ -0,0 +1,44 @@
|
||||
// Copyright 2018 yuzu Emulator Project
|
||||
// Licensed under GPLv2 or any later version
|
||||
// Refer to the license.txt file included.
|
||||
|
||||
#pragma once
|
||||
|
||||
#include <array>
|
||||
#include <vector>
|
||||
|
||||
#include "common/common_types.h"
|
||||
|
||||
namespace AudioCore::Codec {
|
||||
|
||||
enum class PcmFormat : u32 {
|
||||
Invalid = 0,
|
||||
Int8 = 1,
|
||||
Int16 = 2,
|
||||
Int24 = 3,
|
||||
Int32 = 4,
|
||||
PcmFloat = 5,
|
||||
Adpcm = 6,
|
||||
};
|
||||
|
||||
/// See: Codec::DecodeADPCM
|
||||
struct ADPCMState {
|
||||
// Two historical samples from previous processed buffer,
|
||||
// required for ADPCM decoding
|
||||
s16 yn1; ///< y[n-1]
|
||||
s16 yn2; ///< y[n-2]
|
||||
};
|
||||
|
||||
using ADPCM_Coeff = std::array<s16, 16>;
|
||||
|
||||
/**
|
||||
* @param data Pointer to buffer that contains ADPCM data to decode
|
||||
* @param size Size of buffer in bytes
|
||||
* @param coeff ADPCM coefficients
|
||||
* @param state ADPCM state, this is updated with new state
|
||||
* @return Decoded stereo signed PCM16 data, sample_count in length
|
||||
*/
|
||||
std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
|
||||
ADPCMState& state);
|
||||
|
||||
}; // namespace AudioCore::Codec
|
Loading…
Reference in New Issue