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Merge pull request #925 from bunnei/audren

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Implement audren audio output
This commit is contained in:
bunnei
2018-08-05 23:35:22 -04:00
committed by GitHub
parent c8e5c74092 b46df98e93
commit bb21c2198a
19 changed files with 653 additions and 295 deletions
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4
src/audio_core/CMakeLists.txt
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@@ -1,9 +1,13 @@
add_library(audio_core STATIC
audio_out.cpp
audio_out.h
audio_renderer.cpp
audio_renderer.h
buffer.h
cubeb_sink.cpp
cubeb_sink.h
codec.cpp
codec.h
null_sink.h
stream.cpp
stream.h

10
src/audio_core/audio_out.cpp
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@@ -27,16 +27,16 @@ static Stream::Format ChannelsToStreamFormat(u32 num_channels) {
return {};
}
StreamPtr AudioOut::OpenStream(u32 sample_rate, u32 num_channels,
StreamPtr AudioOut::OpenStream(u32 sample_rate, u32 num_channels, std::string&& name,
Stream::ReleaseCallback&& release_callback) {
if (!sink) {
const SinkDetails& sink_details = GetSinkDetails(Settings::values.sink_id);
sink = sink_details.factory(Settings::values.audio_device_id);
}
return std::make_shared<Stream>(sample_rate, ChannelsToStreamFormat(num_channels),
std::move(release_callback),
sink->AcquireSinkStream(sample_rate, num_channels));
return std::make_shared<Stream>(
sample_rate, ChannelsToStreamFormat(num_channels), std::move(release_callback),
sink->AcquireSinkStream(sample_rate, num_channels, name), std::move(name));
}
std::vector<Buffer::Tag> AudioOut::GetTagsAndReleaseBuffers(StreamPtr stream, size_t max_count) {
@@ -51,7 +51,7 @@ void AudioOut::StopStream(StreamPtr stream) {
stream->Stop();
}
bool AudioOut::QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<u8>&& data) {
bool AudioOut::QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data) {
return stream->QueueBuffer(std::make_shared<Buffer>(tag, std::move(data)));
}

5
src/audio_core/audio_out.h
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@@ -5,6 +5,7 @@
#pragma once
#include <memory>
#include <string>
#include <vector>
#include "audio_core/buffer.h"
@@ -20,7 +21,7 @@ namespace AudioCore {
class AudioOut {
public:
/// Opens a new audio stream
StreamPtr OpenStream(u32 sample_rate, u32 num_channels,
StreamPtr OpenStream(u32 sample_rate, u32 num_channels, std::string&& name,
Stream::ReleaseCallback&& release_callback);
/// Returns a vector of recently released buffers specified by tag for the specified stream
@@ -33,7 +34,7 @@ public:
void StopStream(StreamPtr stream);
/// Queues a buffer into the specified audio stream, returns true on success
bool QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<u8>&& data);
bool QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data);
private:
SinkPtr sink;

234
src/audio_core/audio_renderer.cpp Normal file
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@@ -0,0 +1,234 @@
// Copyright 2018 yuzu Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include "audio_core/audio_renderer.h"
#include "common/assert.h"
#include "common/logging/log.h"
#include "core/memory.h"
namespace AudioCore {
constexpr u32 STREAM_SAMPLE_RATE{48000};
constexpr u32 STREAM_NUM_CHANNELS{2};
AudioRenderer::AudioRenderer(AudioRendererParameter params,
Kernel::SharedPtr<Kernel::Event> buffer_event)
: worker_params{params}, buffer_event{buffer_event}, voices(params.voice_count) {
audio_core = std::make_unique<AudioCore::AudioOut>();
stream = audio_core->OpenStream(STREAM_SAMPLE_RATE, STREAM_NUM_CHANNELS, "AudioRenderer",
[=]() { buffer_event->Signal(); });
audio_core->StartStream(stream);
QueueMixedBuffer(0);
QueueMixedBuffer(1);
QueueMixedBuffer(2);
}
std::vector<u8> AudioRenderer::UpdateAudioRenderer(const std::vector<u8>& input_params) {
// Copy UpdateDataHeader struct
UpdateDataHeader config{};
std::memcpy(&config, input_params.data(), sizeof(UpdateDataHeader));
u32 memory_pool_count = worker_params.effect_count + (worker_params.voice_count * 4);
// Copy MemoryPoolInfo structs
std::vector<MemoryPoolInfo> mem_pool_info(memory_pool_count);
std::memcpy(mem_pool_info.data(),
input_params.data() + sizeof(UpdateDataHeader) + config.behavior_size,
memory_pool_count * sizeof(MemoryPoolInfo));
// Copy VoiceInfo structs
size_t offset{sizeof(UpdateDataHeader) + config.behavior_size + config.memory_pools_size +
config.voice_resource_size};
for (auto& voice : voices) {
std::memcpy(&voice.Info(), input_params.data() + offset, sizeof(VoiceInfo));
offset += sizeof(VoiceInfo);
}
// Update voices
for (auto& voice : voices) {
voice.UpdateState();
if (!voice.GetInfo().is_in_use) {
continue;
}
if (voice.GetInfo().is_new) {
voice.SetWaveIndex(voice.GetInfo().wave_buffer_head);
}
}
// Update memory pool state
std::vector<MemoryPoolEntry> memory_pool(memory_pool_count);
for (size_t index = 0; index < memory_pool.size(); ++index) {
if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestAttach) {
memory_pool[index].state = MemoryPoolStates::Attached;
} else if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestDetach) {
memory_pool[index].state = MemoryPoolStates::Detached;
}
}
// Release previous buffers and queue next ones for playback
ReleaseAndQueueBuffers();
// Copy output header
UpdateDataHeader response_data{worker_params};
std::vector<u8> output_params(response_data.total_size);
std::memcpy(output_params.data(), &response_data, sizeof(UpdateDataHeader));
// Copy output memory pool entries
std::memcpy(output_params.data() + sizeof(UpdateDataHeader), memory_pool.data(),
response_data.memory_pools_size);
// Copy output voice status
size_t voice_out_status_offset{sizeof(UpdateDataHeader) + response_data.memory_pools_size};
for (const auto& voice : voices) {
std::memcpy(output_params.data() + voice_out_status_offset, &voice.GetOutStatus(),
sizeof(VoiceOutStatus));
voice_out_status_offset += sizeof(VoiceOutStatus);
}
return output_params;
}
void AudioRenderer::VoiceState::SetWaveIndex(size_t index) {
wave_index = index & 3;
is_refresh_pending = true;
}
std::vector<s16> AudioRenderer::VoiceState::DequeueSamples(size_t sample_count) {
if (!IsPlaying()) {
return {};
}
if (is_refresh_pending) {
RefreshBuffer();
}
const size_t max_size{samples.size() - offset};
const size_t dequeue_offset{offset};
size_t size{sample_count * STREAM_NUM_CHANNELS};
if (size > max_size) {
size = max_size;
}
out_status.played_sample_count += size / STREAM_NUM_CHANNELS;
offset += size;
const auto& wave_buffer{info.wave_buffer[wave_index]};
if (offset == samples.size()) {
offset = 0;
if (!wave_buffer.is_looping) {
SetWaveIndex(wave_index + 1);
}
out_status.wave_buffer_consumed++;
if (wave_buffer.end_of_stream) {
info.play_state = PlayState::Paused;
}
}
return {samples.begin() + dequeue_offset, samples.begin() + dequeue_offset + size};
}
void AudioRenderer::VoiceState::UpdateState() {
if (is_in_use && !info.is_in_use) {
// No longer in use, reset state
is_refresh_pending = true;
wave_index = 0;
offset = 0;
out_status = {};
}
is_in_use = info.is_in_use;
}
void AudioRenderer::VoiceState::RefreshBuffer() {
std::vector<s16> new_samples(info.wave_buffer[wave_index].buffer_sz / sizeof(s16));
Memory::ReadBlock(info.wave_buffer[wave_index].buffer_addr, new_samples.data(),
info.wave_buffer[wave_index].buffer_sz);
switch (static_cast<Codec::PcmFormat>(info.sample_format)) {
case Codec::PcmFormat::Int16: {
// PCM16 is played as-is
break;
}
case Codec::PcmFormat::Adpcm: {
// Decode ADPCM to PCM16
Codec::ADPCM_Coeff coeffs;
Memory::ReadBlock(info.additional_params_addr, coeffs.data(), sizeof(Codec::ADPCM_Coeff));
new_samples = Codec::DecodeADPCM(reinterpret_cast<u8*>(new_samples.data()),
new_samples.size() * sizeof(s16), coeffs, adpcm_state);
break;
}
default:
LOG_CRITICAL(Audio, "Unimplemented sample_format={}", info.sample_format);
UNREACHABLE();
break;
}
switch (info.channel_count) {
case 1:
// 1 channel is upsampled to 2 channel
samples.resize(new_samples.size() * 2);
for (size_t index = 0; index < new_samples.size(); ++index) {
samples[index * 2] = new_samples[index];
samples[index * 2 + 1] = new_samples[index];
}
break;
case 2: {
// 2 channel is played as is
samples = std::move(new_samples);
break;
}
default:
LOG_CRITICAL(Audio, "Unimplemented channel_count={}", info.channel_count);
UNREACHABLE();
break;
}
is_refresh_pending = false;
}
static constexpr s16 ClampToS16(s32 value) {
return static_cast<s16>(std::clamp(value, -32768, 32767));
}
void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
constexpr size_t BUFFER_SIZE{512};
std::vector<s16> buffer(BUFFER_SIZE * stream->GetNumChannels());
for (auto& voice : voices) {
if (!voice.IsPlaying()) {
continue;
}
size_t offset{};
s64 samples_remaining{BUFFER_SIZE};
while (samples_remaining > 0) {
const std::vector<s16> samples{voice.DequeueSamples(samples_remaining)};
if (samples.empty()) {
break;
}
samples_remaining -= samples.size();
for (const auto& sample : samples) {
const s32 buffer_sample{buffer[offset]};
buffer[offset++] =
ClampToS16(buffer_sample + static_cast<s32>(sample * voice.GetInfo().volume));
}
}
}
audio_core->QueueBuffer(stream, tag, std::move(buffer));
}
void AudioRenderer::ReleaseAndQueueBuffers() {
const auto released_buffers{audio_core->GetTagsAndReleaseBuffers(stream, 2)};
for (const auto& tag : released_buffers) {
QueueMixedBuffer(tag);
}
}
} // namespace AudioCore

206
src/audio_core/audio_renderer.h Normal file
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@@ -0,0 +1,206 @@
// Copyright 2018 yuzu Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <array>
#include <memory>
#include <vector>
#include "audio_core/audio_out.h"
#include "audio_core/codec.h"
#include "audio_core/stream.h"
#include "common/common_types.h"
#include "common/swap.h"
#include "core/hle/kernel/event.h"
namespace AudioCore {
enum class PlayState : u8 {
Started = 0,
Stopped = 1,
Paused = 2,
};
struct AudioRendererParameter {
u32_le sample_rate;
u32_le sample_count;
u32_le unknown_8;
u32_le unknown_c;
u32_le voice_count;
u32_le sink_count;
u32_le effect_count;
u32_le unknown_1c;
u8 unknown_20;
INSERT_PADDING_BYTES(3);
u32_le splitter_count;
u32_le unknown_2c;
INSERT_PADDING_WORDS(1);
u32_le revision;
};
static_assert(sizeof(AudioRendererParameter) == 52, "AudioRendererParameter is an invalid size");
enum class MemoryPoolStates : u32 { // Should be LE
Invalid = 0x0,
Unknown = 0x1,
RequestDetach = 0x2,
Detached = 0x3,
RequestAttach = 0x4,
Attached = 0x5,
Released = 0x6,
};
struct MemoryPoolEntry {
MemoryPoolStates state;
u32_le unknown_4;
u32_le unknown_8;
u32_le unknown_c;
};
static_assert(sizeof(MemoryPoolEntry) == 0x10, "MemoryPoolEntry has wrong size");
struct MemoryPoolInfo {
u64_le pool_address;
u64_le pool_size;
MemoryPoolStates pool_state;
INSERT_PADDING_WORDS(3); // Unknown
};
static_assert(sizeof(MemoryPoolInfo) == 0x20, "MemoryPoolInfo has wrong size");
struct BiquadFilter {
u8 enable;
INSERT_PADDING_BYTES(1);
std::array<s16_le, 3> numerator;
std::array<s16_le, 2> denominator;
};
static_assert(sizeof(BiquadFilter) == 0xc, "BiquadFilter has wrong size");
struct WaveBuffer {
u64_le buffer_addr;
u64_le buffer_sz;
s32_le start_sample_offset;
s32_le end_sample_offset;
u8 is_looping;
u8 end_of_stream;
u8 sent_to_server;
INSERT_PADDING_BYTES(5);
u64 context_addr;
u64 context_sz;
INSERT_PADDING_BYTES(8);
};
static_assert(sizeof(WaveBuffer) == 0x38, "WaveBuffer has wrong size");
struct VoiceInfo {
u32_le id;
u32_le node_id;
u8 is_new;
u8 is_in_use;
PlayState play_state;
u8 sample_format;
u32_le sample_rate;
u32_le priority;
u32_le sorting_order;
u32_le channel_count;
float_le pitch;
float_le volume;
std::array<BiquadFilter, 2> biquad_filter;
u32_le wave_buffer_count;
u32_le wave_buffer_head;
INSERT_PADDING_WORDS(1);
u64_le additional_params_addr;
u64_le additional_params_sz;
u32_le mix_id;
u32_le splitter_info_id;
std::array<WaveBuffer, 4> wave_buffer;
std::array<u32_le, 6> voice_channel_resource_ids;
INSERT_PADDING_BYTES(24);
};
static_assert(sizeof(VoiceInfo) == 0x170, "VoiceInfo is wrong size");
struct VoiceOutStatus {
u64_le played_sample_count;
u32_le wave_buffer_consumed;
u32_le voice_drops_count;
};
static_assert(sizeof(VoiceOutStatus) == 0x10, "VoiceOutStatus has wrong size");
struct UpdateDataHeader {
UpdateDataHeader() {}
explicit UpdateDataHeader(const AudioRendererParameter& config) {
revision = Common::MakeMagic('R', 'E', 'V', '4'); // 5.1.0 Revision
behavior_size = 0xb0;
memory_pools_size = (config.effect_count + (config.voice_count * 4)) * 0x10;
voices_size = config.voice_count * 0x10;
voice_resource_size = 0x0;
effects_size = config.effect_count * 0x10;
mixes_size = 0x0;
sinks_size = config.sink_count * 0x20;
performance_manager_size = 0x10;
total_size = sizeof(UpdateDataHeader) + behavior_size + memory_pools_size + voices_size +
effects_size + sinks_size + performance_manager_size;
}
u32_le revision;
u32_le behavior_size;
u32_le memory_pools_size;
u32_le voices_size;
u32_le voice_resource_size;
u32_le effects_size;
u32_le mixes_size;
u32_le sinks_size;
u32_le performance_manager_size;
INSERT_PADDING_WORDS(6);
u32_le total_size;
};
static_assert(sizeof(UpdateDataHeader) == 0x40, "UpdateDataHeader has wrong size");
class AudioRenderer {
public:
AudioRenderer(AudioRendererParameter params, Kernel::SharedPtr<Kernel::Event> buffer_event);
std::vector<u8> UpdateAudioRenderer(const std::vector<u8>& input_params);
void QueueMixedBuffer(Buffer::Tag tag);
void ReleaseAndQueueBuffers();
private:
class VoiceState {
public:
bool IsPlaying() const {
return is_in_use && info.play_state == PlayState::Started;
}
const VoiceOutStatus& GetOutStatus() const {
return out_status;
}
const VoiceInfo& GetInfo() const {
return info;
}
VoiceInfo& Info() {
return info;
}
void SetWaveIndex(size_t index);
std::vector<s16> DequeueSamples(size_t sample_count);
void UpdateState();
void RefreshBuffer();
private:
bool is_in_use{};
bool is_refresh_pending{};
size_t wave_index{};
size_t offset{};
Codec::ADPCMState adpcm_state{};
std::vector<s16> samples;
VoiceOutStatus out_status{};
VoiceInfo info{};
};
AudioRendererParameter worker_params;
Kernel::SharedPtr<Kernel::Event> buffer_event;
std::vector<VoiceState> voices;
std::unique_ptr<AudioCore::AudioOut> audio_core;
AudioCore::StreamPtr stream;
};
} // namespace AudioCore

13
src/audio_core/buffer.h
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@@ -18,11 +18,16 @@ class Buffer {
public:
using Tag = u64;
Buffer(Tag tag, std::vector<u8>&& data) : tag{tag}, data{std::move(data)} {}
Buffer(Tag tag, std::vector<s16>&& samples) : tag{tag}, samples{std::move(samples)} {}
/// Returns the raw audio data for the buffer
const std::vector<u8>& GetData() const {
return data;
std::vector<s16>& Samples() {
return samples;
}
/// Returns the raw audio data for the buffer
const std::vector<s16>& GetSamples() const {
return samples;
}
/// Returns the buffer tag, this is provided by the game to the audout service
@@ -32,7 +37,7 @@ public:
private:
Tag tag;
std::vector<u8> data;
std::vector<s16> samples;
};
using BufferPtr = std::shared_ptr<Buffer>;

77
src/audio_core/codec.cpp Normal file
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@@ -0,0 +1,77 @@
// Copyright 2018 yuzu Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <algorithm>
#include "audio_core/codec.h"
namespace AudioCore::Codec {
std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
ADPCMState& state) {
// GC-ADPCM with scale factor and variable coefficients.
// Frames are 8 bytes long containing 14 samples each.
// Samples are 4 bits (one nibble) long.
constexpr size_t FRAME_LEN = 8;
constexpr size_t SAMPLES_PER_FRAME = 14;
constexpr std::array<int, 16> SIGNED_NIBBLES = {
{0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
const size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
const size_t ret_size =
sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
std::vector<s16> ret(ret_size);
int yn1 = state.yn1, yn2 = state.yn2;
const size_t NUM_FRAMES =
(sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
for (size_t framei = 0; framei < NUM_FRAMES; framei++) {
const int frame_header = data[framei * FRAME_LEN];
const int scale = 1 << (frame_header & 0xF);
const int idx = (frame_header >> 4) & 0x7;
// 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

44
src/audio_core/codec.h Normal file
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@@ -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

56
src/audio_core/cubeb_sink.cpp
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@@ -13,20 +13,30 @@ namespace AudioCore {
class SinkStreamImpl final : public SinkStream {
public:
SinkStreamImpl(cubeb* ctx, cubeb_devid output_device) : ctx{ctx} {
cubeb_stream_params params;
params.rate = 48000;
params.channels = GetNumChannels();
params.format = CUBEB_SAMPLE_S16NE;
params.layout = CUBEB_LAYOUT_STEREO;
SinkStreamImpl(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
const std::string& name)
: ctx{ctx}, num_channels{num_channels_} {
u32 minimum_latency = 0;
if (num_channels == 6) {
// 6-channel audio does not seem to work with cubeb + SDL, so we downsample this to 2
// channel for now
is_6_channel = true;
num_channels = 2;
}
cubeb_stream_params params{};
params.rate = sample_rate;
params.channels = num_channels;
params.format = CUBEB_SAMPLE_S16NE;
params.layout = num_channels == 1 ? CUBEB_LAYOUT_MONO : CUBEB_LAYOUT_STEREO;
u32 minimum_latency{};
if (cubeb_get_min_latency(ctx, &params, &minimum_latency) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error getting minimum latency");
}
if (cubeb_stream_init(ctx, &stream_backend, "yuzu Audio Output", nullptr, nullptr,
output_device, &params, std::max(512u, minimum_latency),
if (cubeb_stream_init(ctx, &stream_backend, name.c_str(), nullptr, nullptr, output_device,
&params, std::max(512u, minimum_latency),
&SinkStreamImpl::DataCallback, &SinkStreamImpl::StateCallback,
this) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream");
@@ -51,33 +61,29 @@ public:
cubeb_stream_destroy(stream_backend);
}
void EnqueueSamples(u32 num_channels, const s16* samples, size_t sample_count) override {
void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) override {
if (!ctx) {
return;
}
queue.reserve(queue.size() + sample_count * GetNumChannels());
queue.reserve(queue.size() + samples.size() * GetNumChannels());
if (num_channels == 2) {
// Copy as-is
std::copy(samples, samples + sample_count * GetNumChannels(),
std::back_inserter(queue));
} else if (num_channels == 6) {
if (is_6_channel) {
// Downsample 6 channels to 2
const size_t sample_count_copy_size = sample_count * num_channels * 2;
const size_t sample_count_copy_size = samples.size() * 2;
queue.reserve(sample_count_copy_size);
for (size_t i = 0; i < sample_count * num_channels; i += num_channels) {
for (size_t i = 0; i < samples.size(); i += num_channels) {
queue.push_back(samples[i]);
queue.push_back(samples[i + 1]);
}
} else {
ASSERT_MSG(false, "Unimplemented");
// Copy as-is
std::copy(samples.begin(), samples.end(), std::back_inserter(queue));
}
}
u32 GetNumChannels() const {
// Only support 2-channel stereo output for now
return 2;
return num_channels;
}
private:
@@ -85,6 +91,8 @@ private:
cubeb* ctx{};
cubeb_stream* stream_backend{};
u32 num_channels{};
bool is_6_channel{};
std::vector<s16> queue;
@@ -129,8 +137,10 @@ CubebSink::~CubebSink() {
cubeb_destroy(ctx);
}
SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels) {
sink_streams.push_back(std::make_unique<SinkStreamImpl>(ctx, output_device));
SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels,
const std::string& name) {
sink_streams.push_back(
std::make_unique<SinkStreamImpl>(ctx, sample_rate, num_channels, output_device, name));
return *sink_streams.back();
}

3
src/audio_core/cubeb_sink.h
View File

@@ -18,7 +18,8 @@ public:
explicit CubebSink(std::string device_id);
~CubebSink() override;
SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels) override;
SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels,
const std::string& name) override;
private:
cubeb* ctx{};

6
src/audio_core/null_sink.h
View File

@@ -13,14 +13,14 @@ public:
explicit NullSink(std::string){};
~NullSink() override = default;
SinkStream& AcquireSinkStream(u32 /*sample_rate*/, u32 /*num_channels*/) override {
SinkStream& AcquireSinkStream(u32 /*sample_rate*/, u32 /*num_channels*/,
const std::string& /*name*/) override {
return null_sink_stream;
}
private:
struct NullSinkStreamImpl final : SinkStream {
void EnqueueSamples(u32 /*num_channels*/, const s16* /*samples*/,
size_t /*sample_count*/) override {}
void EnqueueSamples(u32 /*num_channels*/, const std::vector<s16>& /*samples*/) override {}
} null_sink_stream;
};

4
src/audio_core/sink.h
View File

@@ -5,6 +5,7 @@
#pragma once
#include <memory>
#include <string>
#include "audio_core/sink_stream.h"
#include "common/common_types.h"
@@ -21,7 +22,8 @@ constexpr char auto_device_name[] = "auto";
class Sink {
public:
virtual ~Sink() = default;
virtual SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels) = 0;
virtual SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels,
const std::string& name) = 0;
};
using SinkPtr = std::unique_ptr<Sink>;

4
src/audio_core/sink_stream.h
View File

@@ -5,6 +5,7 @@
#pragma once
#include <memory>
#include <vector>
#include "common/common_types.h"
@@ -22,9 +23,8 @@ public:
* Feed stereo samples to sink.
* @param num_channels Number of channels used.
* @param samples Samples in interleaved stereo PCM16 format.
* @param sample_count Number of samples.
*/
virtual void EnqueueSamples(u32 num_channels, const s16* samples, size_t sample_count) = 0;
virtual void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) = 0;
};
using SinkStreamPtr = std::unique_ptr<SinkStream>;

26
src/audio_core/stream.cpp
View File

@@ -32,17 +32,13 @@ u32 Stream::GetNumChannels() const {
return {};
}
u32 Stream::GetSampleSize() const {
return GetNumChannels() * 2;
}
Stream::Stream(u32 sample_rate, Format format, ReleaseCallback&& release_callback,
SinkStream& sink_stream)
SinkStream& sink_stream, std::string&& name_)
: sample_rate{sample_rate}, format{format}, release_callback{std::move(release_callback)},
sink_stream{sink_stream} {
sink_stream{sink_stream}, name{std::move(name_)} {
release_event = CoreTiming::RegisterEvent(
"Stream::Release", [this](u64 userdata, int cycles_late) { ReleaseActiveBuffer(); });
name, [this](u64 userdata, int cycles_late) { ReleaseActiveBuffer(); });
}
void Stream::Play() {
@@ -55,17 +51,15 @@ void Stream::Stop() {
}
s64 Stream::GetBufferReleaseCycles(const Buffer& buffer) const {
const size_t num_samples{buffer.GetData().size() / GetSampleSize()};
const size_t num_samples{buffer.GetSamples().size() / GetNumChannels()};
return CoreTiming::usToCycles((static_cast<u64>(num_samples) * 1000000) / sample_rate);
}
static std::vector<s16> GetVolumeAdjustedSamples(const std::vector<u8>& data) {
std::vector<s16> samples(data.size() / sizeof(s16));
std::memcpy(samples.data(), data.data(), data.size());
static void VolumeAdjustSamples(std::vector<s16>& samples) {
const float volume{std::clamp(Settings::values.volume, 0.0f, 1.0f)};
if (volume == 1.0f) {
return samples;
return;
}
// Implementation of a volume slider with a dynamic range of 60 dB
@@ -73,8 +67,6 @@ static std::vector<s16> GetVolumeAdjustedSamples(const std::vector<u8>& data) {
for (auto& sample : samples) {
sample = static_cast<s16>(sample * volume_scale_factor);
}
return samples;
}
void Stream::PlayNextBuffer() {
@@ -96,14 +88,14 @@ void Stream::PlayNextBuffer() {
active_buffer = queued_buffers.front();
queued_buffers.pop();
const size_t sample_count{active_buffer->GetData().size() / GetSampleSize()};
sink_stream.EnqueueSamples(
GetNumChannels(), GetVolumeAdjustedSamples(active_buffer->GetData()).data(), sample_count);
VolumeAdjustSamples(active_buffer->Samples());
sink_stream.EnqueueSamples(GetNumChannels(), active_buffer->GetSamples());
CoreTiming::ScheduleEventThreadsafe(GetBufferReleaseCycles(*active_buffer), release_event, {});
}
void Stream::ReleaseActiveBuffer() {
ASSERT(active_buffer);
released_buffers.push(std::move(active_buffer));
release_callback();
PlayNextBuffer();

7
src/audio_core/stream.h
View File

@@ -6,6 +6,7 @@
#include <functional>
#include <memory>
#include <string>
#include <vector>
#include <queue>
@@ -33,7 +34,7 @@ public:
using ReleaseCallback = std::function<void()>;
Stream(u32 sample_rate, Format format, ReleaseCallback&& release_callback,
SinkStream& sink_stream);
SinkStream& sink_stream, std::string&& name_);
/// Plays the audio stream
void Play();
@@ -68,9 +69,6 @@ public:
/// Gets the number of channels
u32 GetNumChannels() const;
/// Gets the sample size in bytes
u32 GetSampleSize() const;
private:
/// Current state of the stream
enum class State {
@@ -96,6 +94,7 @@ private:
std::queue<BufferPtr> queued_buffers; ///< Buffers queued to be played in the stream
std::queue<BufferPtr> released_buffers; ///< Buffers recently released from the stream
SinkStream& sink_stream; ///< Output sink for the stream
std::string name; ///< Name of the stream, must be unique
};
using StreamPtr = std::shared_ptr<Stream>;

12
src/core/hle/service/audio/audout_u.cpp
View File

@@ -4,6 +4,8 @@
#include <array>
#include <vector>
#include "audio_core/codec.h"
#include "common/logging/log.h"
#include "core/core.h"
#include "core/hle/ipc_helpers.h"
@@ -48,7 +50,7 @@ public:
buffer_event = Kernel::Event::Create(Kernel::ResetType::Sticky, "IAudioOutBufferReleased");
stream = audio_core.OpenStream(audio_params.sample_rate, audio_params.channel_count,
[=]() { buffer_event->Signal(); });
"IAudioOut", [=]() { buffer_event->Signal(); });
}
private:
@@ -111,10 +113,10 @@ private:
std::memcpy(&audio_buffer, input_buffer.data(), sizeof(AudioBuffer));
const u64 tag{rp.Pop<u64>()};
std::vector<u8> data(audio_buffer.buffer_size);
Memory::ReadBlock(audio_buffer.buffer, data.data(), data.size());
std::vector<s16> samples(audio_buffer.buffer_size / sizeof(s16));
Memory::ReadBlock(audio_buffer.buffer, samples.data(), audio_buffer.buffer_size);
if (!audio_core.QueueBuffer(stream, tag, std::move(data))) {
if (!audio_core.QueueBuffer(stream, tag, std::move(samples))) {
IPC::ResponseBuilder rb{ctx, 2};
rb.Push(ResultCode(ErrorModule::Audio, ErrCodes::BufferCountExceeded));
}
@@ -200,7 +202,7 @@ void AudOutU::OpenAudioOutImpl(Kernel::HLERequestContext& ctx) {
rb.Push(RESULT_SUCCESS);
rb.Push<u32>(DefaultSampleRate);
rb.Push<u32>(params.channel_count);
rb.Push<u32>(static_cast<u32>(PcmFormat::Int16));
rb.Push<u32>(static_cast<u32>(AudioCore::Codec::PcmFormat::Int16));
rb.Push<u32>(static_cast<u32>(AudioState::Stopped));
rb.PushIpcInterface<Audio::IAudioOut>(audio_out_interface);
}

10
src/core/hle/service/audio/audout_u.h
View File

@@ -38,16 +38,6 @@ private:
void ListAudioOutsImpl(Kernel::HLERequestContext& ctx);
void OpenAudioOutImpl(Kernel::HLERequestContext& ctx);
enum class PcmFormat : u32 {
Invalid = 0,
Int8 = 1,
Int16 = 2,
Int24 = 3,
Int32 = 4,
PcmFloat = 5,
Adpcm = 6,
};
};
} // namespace Service::Audio

208
src/core/hle/service/audio/audren_u.cpp
View File

@@ -15,13 +15,10 @@
namespace Service::Audio {
/// TODO(bunnei): Find a proper value for the audio_ticks
constexpr u64 audio_ticks{static_cast<u64>(CoreTiming::BASE_CLOCK_RATE / 200)};
class IAudioRenderer final : public ServiceFramework<IAudioRenderer> {
public:
explicit IAudioRenderer(AudioRendererParameter audren_params)
: ServiceFramework("IAudioRenderer"), worker_params(audren_params) {
explicit IAudioRenderer(AudioCore::AudioRendererParameter audren_params)
: ServiceFramework("IAudioRenderer") {
static const FunctionInfo functions[] = {
{0, nullptr, "GetAudioRendererSampleRate"},
{1, nullptr, "GetAudioRendererSampleCount"},
@@ -39,21 +36,8 @@ public:
RegisterHandlers(functions);
system_event =
Kernel::Event::Create(Kernel::ResetType::OneShot, "IAudioRenderer:SystemEvent");
// Register event callback to update the Audio Buffer
audio_event = CoreTiming::RegisterEvent(
"IAudioRenderer::UpdateAudioCallback", [this](u64 userdata, int cycles_late) {
UpdateAudioCallback();
CoreTiming::ScheduleEvent(audio_ticks - cycles_late, audio_event);
});
// Start the audio event
CoreTiming::ScheduleEvent(audio_ticks, audio_event);
voice_status_list.resize(worker_params.voice_count);
}
~IAudioRenderer() {
CoreTiming::UnscheduleEvent(audio_event, 0);
Kernel::Event::Create(Kernel::ResetType::Sticky, "IAudioRenderer:SystemEvent");
renderer = std::make_unique<AudioCore::AudioRenderer>(audren_params, system_event);
}
private:
@@ -62,60 +46,9 @@ private:
}
void RequestUpdateAudioRenderer(Kernel::HLERequestContext& ctx) {
UpdateDataHeader config{};
auto buf = ctx.ReadBuffer();
std::memcpy(&config, buf.data(), sizeof(UpdateDataHeader));
u32 memory_pool_count = worker_params.effect_count + (worker_params.voice_count * 4);
std::vector<MemoryPoolInfo> mem_pool_info(memory_pool_count);
std::memcpy(mem_pool_info.data(),
buf.data() + sizeof(UpdateDataHeader) + config.behavior_size,
memory_pool_count * sizeof(MemoryPoolInfo));
std::vector<VoiceInfo> voice_info(worker_params.voice_count);
std::memcpy(voice_info.data(),
buf.data() + sizeof(UpdateDataHeader) + config.behavior_size +
config.memory_pools_size + config.voice_resource_size,
worker_params.voice_count * sizeof(VoiceInfo));
UpdateDataHeader response_data{worker_params};
ASSERT(ctx.GetWriteBufferSize() == response_data.total_size);
std::vector<u8> output(response_data.total_size);
std::memcpy(output.data(), &response_data, sizeof(UpdateDataHeader));
std::vector<MemoryPoolEntry> memory_pool(memory_pool_count);
for (unsigned i = 0; i < memory_pool.size(); i++) {
if (mem_pool_info[i].pool_state == MemoryPoolStates::RequestAttach)
memory_pool[i].state = MemoryPoolStates::Attached;
else if (mem_pool_info[i].pool_state == MemoryPoolStates::RequestDetach)
memory_pool[i].state = MemoryPoolStates::Detached;
}
std::memcpy(output.data() + sizeof(UpdateDataHeader), memory_pool.data(),
response_data.memory_pools_size);
for (unsigned i = 0; i < voice_info.size(); i++) {
if (voice_info[i].is_new) {
voice_status_list[i].played_sample_count = 0;
voice_status_list[i].wave_buffer_consumed = 0;
} else if (voice_info[i].play_state == (u8)PlayStates::Started) {
for (u32 buff_idx = 0; buff_idx < voice_info[i].wave_buffer_count; buff_idx++) {
voice_status_list[i].played_sample_count +=
(voice_info[i].wave_buffer[buff_idx].end_sample_offset -
voice_info[i].wave_buffer[buff_idx].start_sample_offset) /
2;
voice_status_list[i].wave_buffer_consumed++;
}
}
}
std::memcpy(output.data() + sizeof(UpdateDataHeader) + response_data.memory_pools_size,
voice_status_list.data(), response_data.voices_size);
ctx.WriteBuffer(output);
ctx.WriteBuffer(renderer->UpdateAudioRenderer(ctx.ReadBuffer()));
IPC::ResponseBuilder rb{ctx, 2};
rb.Push(RESULT_SUCCESS);
LOG_WARNING(Service_Audio, "(STUBBED) called");
}
@@ -136,8 +69,6 @@ private:
}
void QuerySystemEvent(Kernel::HLERequestContext& ctx) {
// system_event->Signal();
IPC::ResponseBuilder rb{ctx, 2, 1};
rb.Push(RESULT_SUCCESS);
rb.PushCopyObjects(system_event);
@@ -145,131 +76,8 @@ private:
LOG_WARNING(Service_Audio, "(STUBBED) called");
}
enum class MemoryPoolStates : u32 { // Should be LE
Invalid = 0x0,
Unknown = 0x1,
RequestDetach = 0x2,
Detached = 0x3,
RequestAttach = 0x4,
Attached = 0x5,
Released = 0x6,
};
enum class PlayStates : u8 {
Started = 0,
Stopped = 1,
};
struct MemoryPoolEntry {
MemoryPoolStates state;
u32_le unknown_4;
u32_le unknown_8;
u32_le unknown_c;
};
static_assert(sizeof(MemoryPoolEntry) == 0x10, "MemoryPoolEntry has wrong size");
struct MemoryPoolInfo {
u64_le pool_address;
u64_le pool_size;
MemoryPoolStates pool_state;
INSERT_PADDING_WORDS(3); // Unknown
};
static_assert(sizeof(MemoryPoolInfo) == 0x20, "MemoryPoolInfo has wrong size");
struct UpdateDataHeader {
UpdateDataHeader() {}
explicit UpdateDataHeader(const AudioRendererParameter& config) {
revision = Common::MakeMagic('R', 'E', 'V', '4'); // 5.1.0 Revision
behavior_size = 0xb0;
memory_pools_size = (config.effect_count + (config.voice_count * 4)) * 0x10;
voices_size = config.voice_count * 0x10;
voice_resource_size = 0x0;
effects_size = config.effect_count * 0x10;
mixes_size = 0x0;
sinks_size = config.sink_count * 0x20;
performance_manager_size = 0x10;
total_size = sizeof(UpdateDataHeader) + behavior_size + memory_pools_size +
voices_size + effects_size + sinks_size + performance_manager_size;
}
u32_le revision;
u32_le behavior_size;
u32_le memory_pools_size;
u32_le voices_size;
u32_le voice_resource_size;
u32_le effects_size;
u32_le mixes_size;
u32_le sinks_size;
u32_le performance_manager_size;
INSERT_PADDING_WORDS(6);
u32_le total_size;
};
static_assert(sizeof(UpdateDataHeader) == 0x40, "UpdateDataHeader has wrong size");
struct BiquadFilter {
u8 enable;
INSERT_PADDING_BYTES(1);
s16_le numerator[3];
s16_le denominator[2];
};
static_assert(sizeof(BiquadFilter) == 0xc, "BiquadFilter has wrong size");
struct WaveBuffer {
u64_le buffer_addr;
u64_le buffer_sz;
s32_le start_sample_offset;
s32_le end_sample_offset;
u8 loop;
u8 end_of_stream;
u8 sent_to_server;
INSERT_PADDING_BYTES(5);
u64 context_addr;
u64 context_sz;
INSERT_PADDING_BYTES(8);
};
static_assert(sizeof(WaveBuffer) == 0x38, "WaveBuffer has wrong size");
struct VoiceInfo {
u32_le id;
u32_le node_id;
u8 is_new;
u8 is_in_use;
u8 play_state;
u8 sample_format;
u32_le sample_rate;
u32_le priority;
u32_le sorting_order;
u32_le channel_count;
float_le pitch;
float_le volume;
BiquadFilter biquad_filter[2];
u32_le wave_buffer_count;
u16_le wave_buffer_head;
INSERT_PADDING_BYTES(6);
u64_le additional_params_addr;
u64_le additional_params_sz;
u32_le mix_id;
u32_le splitter_info_id;
WaveBuffer wave_buffer[4];
u32_le voice_channel_resource_ids[6];
INSERT_PADDING_BYTES(24);
};
static_assert(sizeof(VoiceInfo) == 0x170, "VoiceInfo is wrong size");
struct VoiceOutStatus {
u64_le played_sample_count;
u32_le wave_buffer_consumed;
INSERT_PADDING_WORDS(1);
};
static_assert(sizeof(VoiceOutStatus) == 0x10, "VoiceOutStatus has wrong size");
/// This is used to trigger the audio event callback.
CoreTiming::EventType* audio_event;
Kernel::SharedPtr<Kernel::Event> system_event;
AudioRendererParameter worker_params;
std::vector<VoiceOutStatus> voice_status_list;
std::unique_ptr<AudioCore::AudioRenderer> renderer;
};
class IAudioDevice final : public ServiceFramework<IAudioDevice> {
@@ -368,7 +176,7 @@ AudRenU::AudRenU() : ServiceFramework("audren:u") {
void AudRenU::OpenAudioRenderer(Kernel::HLERequestContext& ctx) {
IPC::RequestParser rp{ctx};
auto params = rp.PopRaw<AudioRendererParameter>();
auto params = rp.PopRaw<AudioCore::AudioRendererParameter>();
IPC::ResponseBuilder rb{ctx, 2, 0, 1};
rb.Push(RESULT_SUCCESS);
@@ -379,7 +187,7 @@ void AudRenU::OpenAudioRenderer(Kernel::HLERequestContext& ctx) {
void AudRenU::GetAudioRendererWorkBufferSize(Kernel::HLERequestContext& ctx) {
IPC::RequestParser rp{ctx};
auto params = rp.PopRaw<AudioRendererParameter>();
auto params = rp.PopRaw<AudioCore::AudioRendererParameter>();
u64 buffer_sz = Common::AlignUp(4 * params.unknown_8, 0x40);
buffer_sz += params.unknown_c * 1024;

19
src/core/hle/service/audio/audren_u.h
View File

@@ -4,6 +4,7 @@
#pragma once
#include "audio_core/audio_renderer.h"
#include "core/hle/service/service.h"
namespace Kernel {
@@ -12,24 +13,6 @@ class HLERequestContext;
namespace Service::Audio {
struct AudioRendererParameter {
u32_le sample_rate;
u32_le sample_count;
u32_le unknown_8;
u32_le unknown_c;
u32_le voice_count;
u32_le sink_count;
u32_le effect_count;
u32_le unknown_1c;
u8 unknown_20;
INSERT_PADDING_BYTES(3);
u32_le splitter_count;
u32_le unknown_2c;
INSERT_PADDING_WORDS(1);
u32_le revision;
};
static_assert(sizeof(AudioRendererParameter) == 52, "AudioRendererParameter is an invalid size");
class AudRenU final : public ServiceFramework<AudRenU> {
public:
explicit AudRenU();