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Implement basic cellAudio buffering

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This commit is contained in:
Rui Pinheiro 2018-12-20 22:35:49 +00:00 committed by kd-11
parent 56962aa707
commit 2addbe6be2
13 changed files with 1115 additions and 409 deletions
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6
rpcs3/Emu/Audio/AudioDumper.cpp
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@ -1,10 +1,6 @@
#include "stdafx.h"
#include "stdafx.h"
#include "AudioDumper.h"
#include "AudioThread.h"
AudioThread::~AudioThread()
{
}
AudioDumper::AudioDumper(u16 ch)
: m_header(ch)

50
rpcs3/Emu/Audio/AudioThread.h
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@ -1,13 +1,51 @@
#pragma once
#pragma once
#include "Utilities/types.h"
#include "Emu/System.h"
enum : u32
{
DEFAULT_AUDIO_SAMPLING_RATE = 48000,
MAX_AUDIO_BUFFERS = 64,
AUDIO_BUFFER_SAMPLES = 256
};
class AudioThread
{
public:
virtual ~AudioThread();
virtual ~AudioThread() = default;
// Callbacks
virtual void Open() = 0;
virtual void Close() = 0;
virtual void Play() = 0;
virtual void Open(const void* src, int size) = 0;
virtual void Close() = 0;
virtual void Stop() = 0;
virtual void AddData(const void* src, int size) = 0;
virtual void Pause() = 0;
virtual bool IsPlaying() = 0;
virtual bool AddData(const void* src, int size) = 0;
virtual void Flush() = 0;
// Helper methods
static u32 get_sampling_rate()
{
const u32 sampling_period_multiplier_u32 = g_cfg.audio.sampling_period_multiplier;
if (sampling_period_multiplier_u32 == 100)
return DEFAULT_AUDIO_SAMPLING_RATE;
const f32 sampling_period_multiplier = sampling_period_multiplier_u32 / 100.0f;
const f32 sampling_rate_multiplier = 1.0f / sampling_period_multiplier;
return static_cast<u32>(DEFAULT_AUDIO_SAMPLING_RATE * sampling_rate_multiplier);
}
static u32 get_sample_size()
{
return g_cfg.audio.convert_to_u16 ? sizeof(u16) : sizeof(float);
}
static u32 get_channels()
{
return g_cfg.audio.downmix_to_2ch ? 2 : 8;
}
};

18
rpcs3/Emu/Audio/Null/NullAudioThread.h
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@ -1,4 +1,4 @@
#pragma once
#pragma once
#include "Emu/Audio/AudioThread.h"
@ -8,11 +8,13 @@ public:
NullAudioThread() {}
virtual ~NullAudioThread() {}
virtual void Init() {}
virtual void Quit() {}
virtual void Play() {}
virtual void Open(const void* src, int size) {}
virtual void Close() {}
virtual void Stop() {}
virtual void AddData(const void* src, int size) {}
virtual void Open() {};
virtual void Close() {};
virtual void Play() {};
virtual void Pause() {};
virtual bool IsPlaying() { return true; };
virtual bool AddData(const void* src, int size) { return true; };
virtual void Flush() {};
};

34
rpcs3/Emu/Audio/XAudio2/XAudio27Thread.cpp
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@ -1,4 +1,4 @@
#ifdef _WIN32
#ifdef _WIN32
#include "Utilities/Log.h"
#include "Utilities/StrFmt.h"
@ -97,18 +97,27 @@ void XAudio2Thread::xa27_stop()
}
}
bool XAudio2Thread::xa27_is_playing()
{
XAUDIO2_VOICE_STATE state;
s_tls_source_voice->GetState(&state);
return state.BuffersQueued > 0 || state.pCurrentBufferContext != nullptr;
}
void XAudio2Thread::xa27_open()
{
HRESULT hr;
WORD sample_size = g_cfg.audio.convert_to_u16 ? sizeof(u16) : sizeof(float);
WORD channels = g_cfg.audio.downmix_to_2ch ? 2 : 8;
const u32 sample_size = get_sample_size();
const u32 channels = get_channels();
const u32 sampling_rate = get_sampling_rate();
WAVEFORMATEX waveformatex;
waveformatex.wFormatTag = g_cfg.audio.convert_to_u16 ? WAVE_FORMAT_PCM : WAVE_FORMAT_IEEE_FLOAT;
waveformatex.nChannels = channels;
waveformatex.nSamplesPerSec = 48000;
waveformatex.nAvgBytesPerSec = 48000 * (DWORD)channels * (DWORD)sample_size;
waveformatex.nSamplesPerSec = sampling_rate;
waveformatex.nAvgBytesPerSec = static_cast<DWORD>(sampling_rate * channels * sample_size);
waveformatex.nBlockAlign = channels * sample_size;
waveformatex.wBitsPerSample = sample_size * 8;
waveformatex.cbSize = 0;
@ -121,25 +130,24 @@ void XAudio2Thread::xa27_open()
return;
}
s_tls_source_voice->SetVolume(g_cfg.audio.downmix_to_2ch ? 1.0f : 4.0f);
s_tls_source_voice->SetVolume(channels == 2 ? 1.0f : 4.0f);
}
void XAudio2Thread::xa27_add(const void* src, int size)
bool XAudio2Thread::xa27_add(const void* src, int size)
{
XAUDIO2_VOICE_STATE state;
s_tls_source_voice->GetState(&state);
// XAudio 2.7 bug workaround, when it says "SimpList: non-growable list ran out of room for new elements" and hits int 3
if (state.BuffersQueued > 32)
if (state.BuffersQueued >= MAX_AUDIO_BUFFERS)
{
LOG_WARNING(GENERAL, "XAudio2Thread : too many buffers enqueued (%d, pos=%u)", state.BuffersQueued, state.SamplesPlayed);
return xa27_flush();
return false;
}
XAUDIO2_BUFFER buffer;
buffer.AudioBytes = size;
buffer.AudioBytes = size * get_sample_size();
buffer.Flags = 0;
buffer.LoopBegin = XAUDIO2_NO_LOOP_REGION;
buffer.LoopCount = 0;
@ -147,7 +155,7 @@ void XAudio2Thread::xa27_add(const void* src, int size)
buffer.pAudioData = (const BYTE*)src;
buffer.pContext = 0;
buffer.PlayBegin = 0;
buffer.PlayLength = 256;
buffer.PlayLength = AUDIO_BUFFER_SAMPLES;
HRESULT hr = s_tls_source_voice->SubmitSourceBuffer(&buffer);
if (FAILED(hr))
@ -155,6 +163,8 @@ void XAudio2Thread::xa27_add(const void* src, int size)
LOG_ERROR(GENERAL, "XAudio2Thread : AddData() failed(0x%08x)", (u32)hr);
Emu.Pause();
}
return true;
}
#endif

49
rpcs3/Emu/Audio/XAudio2/XAudio28Thread.cpp
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@ -1,4 +1,4 @@
#ifdef _WIN32
#ifdef _WIN32
#include "Utilities/Log.h"
#include "Utilities/StrFmt.h"
@ -71,6 +71,8 @@ void XAudio2Thread::xa28_destroy()
void XAudio2Thread::xa28_play()
{
AUDIT(s_tls_source_voice != nullptr);
HRESULT hr = s_tls_source_voice->Start();
if (FAILED(hr))
{
@ -81,6 +83,8 @@ void XAudio2Thread::xa28_play()
void XAudio2Thread::xa28_flush()
{
AUDIT(s_tls_source_voice != nullptr);
HRESULT hr = s_tls_source_voice->FlushSourceBuffers();
if (FAILED(hr))
{
@ -91,6 +95,8 @@ void XAudio2Thread::xa28_flush()
void XAudio2Thread::xa28_stop()
{
AUDIT(s_tls_source_voice != nullptr);
HRESULT hr = s_tls_source_voice->Stop();
if (FAILED(hr))
{
@ -99,18 +105,29 @@ void XAudio2Thread::xa28_stop()
}
}
bool XAudio2Thread::xa28_is_playing()
{
AUDIT(s_tls_source_voice != nullptr);
XAUDIO2_VOICE_STATE state;
s_tls_source_voice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
return state.BuffersQueued > 0 || state.pCurrentBufferContext != nullptr;
}
void XAudio2Thread::xa28_open()
{
HRESULT hr;
WORD sample_size = g_cfg.audio.convert_to_u16 ? sizeof(u16) : sizeof(float);
WORD channels = g_cfg.audio.downmix_to_2ch ? 2 : 8;
const u32 sample_size = get_sample_size();
const u32 channels = get_channels();
const u32 sampling_rate = get_sampling_rate();
WAVEFORMATEX waveformatex;
waveformatex.wFormatTag = g_cfg.audio.convert_to_u16 ? WAVE_FORMAT_PCM : WAVE_FORMAT_IEEE_FLOAT;
waveformatex.nChannels = channels;
waveformatex.nSamplesPerSec = 48000;
waveformatex.nAvgBytesPerSec = 48000 * (DWORD)channels * (DWORD)sample_size;
waveformatex.nSamplesPerSec = sampling_rate;
waveformatex.nAvgBytesPerSec = static_cast<DWORD>(sampling_rate * channels * sample_size);
waveformatex.nBlockAlign = channels * sample_size;
waveformatex.wBitsPerSample = sample_size * 8;
waveformatex.cbSize = 0;
@ -123,24 +140,26 @@ void XAudio2Thread::xa28_open()
return;
}
s_tls_source_voice->SetVolume(g_cfg.audio.downmix_to_2ch ? 1.0 : 4.0);
AUDIT(s_tls_source_voice != nullptr);
s_tls_source_voice->SetVolume(channels == 2 ? 1.0 : 4.0);
}
void XAudio2Thread::xa28_add(const void* src, int size)
bool XAudio2Thread::xa28_add(const void* src, int size)
{
XAUDIO2_VOICE_STATE state;
s_tls_source_voice->GetState(&state);
AUDIT(s_tls_source_voice != nullptr);
if (state.BuffersQueued > 32)
XAUDIO2_VOICE_STATE state;
s_tls_source_voice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
if (state.BuffersQueued >= MAX_AUDIO_BUFFERS)
{
LOG_WARNING(GENERAL, "XAudio2Thread : too many buffers enqueued (%d, pos=%u)", state.BuffersQueued, state.SamplesPlayed);
return xa28_flush();
return false;
}
XAUDIO2_BUFFER buffer;
buffer.AudioBytes = size;
buffer.AudioBytes = size * get_sample_size();
buffer.Flags = 0;
buffer.LoopBegin = XAUDIO2_NO_LOOP_REGION;
buffer.LoopCount = 0;
@ -148,7 +167,7 @@ void XAudio2Thread::xa28_add(const void* src, int size)
buffer.pAudioData = (const BYTE*)src;
buffer.pContext = 0;
buffer.PlayBegin = 0;
buffer.PlayLength = 256;
buffer.PlayLength = AUDIO_BUFFER_SAMPLES;
HRESULT hr = s_tls_source_voice->SubmitSourceBuffer(&buffer);
if (FAILED(hr))
@ -156,6 +175,8 @@ void XAudio2Thread::xa28_add(const void* src, int size)
LOG_ERROR(GENERAL, "XAudio2Thread : AddData() failed(0x%08x)", (u32)hr);
Emu.Pause();
}
return true;
}
#endif

61
rpcs3/Emu/Audio/XAudio2/XAudio2Thread.cpp
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@ -1,4 +1,4 @@
#ifdef _WIN32
#ifdef _WIN32
#include "Utilities/Log.h"
#include "Utilities/StrFmt.h"
@ -18,12 +18,13 @@ XAudio2Thread::XAudio2Thread()
// xa28* implementation is fully compatible with library 2.9
xa28_init(lib2_9);
m_funcs.destroy = &xa28_destroy;
m_funcs.play = &xa28_play;
m_funcs.flush = &xa28_flush;
m_funcs.stop = &xa28_stop;
m_funcs.open = &xa28_open;
m_funcs.add = &xa28_add;
m_funcs.destroy = &xa28_destroy;
m_funcs.play = &xa28_play;
m_funcs.flush = &xa28_flush;
m_funcs.stop = &xa28_stop;
m_funcs.open = &xa28_open;
m_funcs.is_playing = &xa28_is_playing;
m_funcs.add = &xa28_add;
LOG_SUCCESS(GENERAL, "XAudio 2.9 initialized");
return;
@ -33,12 +34,13 @@ XAudio2Thread::XAudio2Thread()
{
xa27_init(lib2_7);
m_funcs.destroy = &xa27_destroy;
m_funcs.play = &xa27_play;
m_funcs.flush = &xa27_flush;
m_funcs.stop = &xa27_stop;
m_funcs.open = &xa27_open;
m_funcs.add = &xa27_add;
m_funcs.destroy = &xa27_destroy;
m_funcs.play = &xa27_play;
m_funcs.flush = &xa27_flush;
m_funcs.stop = &xa27_stop;
m_funcs.open = &xa27_open;
m_funcs.is_playing = &xa27_is_playing;
m_funcs.add = &xa27_add;
LOG_SUCCESS(GENERAL, "XAudio 2.7 initialized");
return;
@ -48,12 +50,13 @@ XAudio2Thread::XAudio2Thread()
{
xa28_init(lib2_8);
m_funcs.destroy = &xa28_destroy;
m_funcs.play = &xa28_play;
m_funcs.flush = &xa28_flush;
m_funcs.stop = &xa28_stop;
m_funcs.open = &xa28_open;
m_funcs.add = &xa28_add;
m_funcs.destroy = &xa28_destroy;
m_funcs.play = &xa28_play;
m_funcs.flush = &xa28_flush;
m_funcs.stop = &xa28_stop;
m_funcs.open = &xa28_open;
m_funcs.is_playing = &xa28_is_playing;
m_funcs.add = &xa28_add;
LOG_SUCCESS(GENERAL, "XAudio 2.8 initialized");
return;
@ -78,21 +81,29 @@ void XAudio2Thread::Close()
m_funcs.flush();
}
void XAudio2Thread::Stop()
void XAudio2Thread::Pause()
{
m_funcs.stop();
}
void XAudio2Thread::Open(const void* src, int size)
void XAudio2Thread::Open()
{
m_funcs.open();
m_funcs.add(src, size);
m_funcs.play();
}
void XAudio2Thread::AddData(const void* src, int size)
bool XAudio2Thread::IsPlaying()
{
m_funcs.add(src, size);
return m_funcs.is_playing();
}
bool XAudio2Thread::AddData(const void* src, int size)
{
return m_funcs.add(src, size);
}
void XAudio2Thread::Flush()
{
m_funcs.flush();
}
#endif

23
rpcs3/Emu/Audio/XAudio2/XAudio2Thread.h
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@ -1,4 +1,4 @@
#pragma once
#pragma once
#ifdef _WIN32
@ -13,7 +13,8 @@ class XAudio2Thread : public AudioThread
void(*flush)();
void(*stop)();
void(*open)();
void(*add)(const void*, int);
bool(*is_playing)();
bool(*add)(const void*, int);
};
vtable m_funcs;
@ -24,7 +25,8 @@ class XAudio2Thread : public AudioThread
static void xa27_flush();
static void xa27_stop();
static void xa27_open();
static void xa27_add(const void*, int);
static bool xa27_is_playing();
static bool xa27_add(const void*, int);
static void xa28_init(void*);
static void xa28_destroy();
@ -32,17 +34,22 @@ class XAudio2Thread : public AudioThread
static void xa28_flush();
static void xa28_stop();
static void xa28_open();
static void xa28_add(const void*, int);
static bool xa28_is_playing();
static bool xa28_add(const void*, int);
public:
XAudio2Thread();
virtual ~XAudio2Thread() override;
virtual void Play() override;
virtual void Open(const void* src, int size) override;
virtual void Open() override;
virtual void Close() override;
virtual void Stop() override;
virtual void AddData(const void* src, int size) override;
virtual void Play() override;
virtual void Pause() override;
virtual bool IsPlaying() override;
virtual bool AddData(const void* src, int size) override;
virtual void Flush() override;
};
#endif

1049
rpcs3/Emu/Cell/Modules/cellAudio.cpp
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File diff suppressed because it is too large Load Diff

183
rpcs3/Emu/Cell/Modules/cellAudio.h
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@ -1,6 +1,9 @@
#pragma once
#pragma once
#include "Utilities/Thread.h"
#include "Emu/Memory/vm.h"
#include "Emu/Audio/AudioThread.h"
#include "Emu/Audio/AudioDumper.h"
// Error codes
enum CellAudioError : u32
@ -76,11 +79,26 @@ struct CellAudioPortConfig
enum : u32
{
BUFFER_NUM = 32,
BUFFER_SIZE = 256,
AUDIO_PORT_COUNT = 8,
AUDIO_PORT_OFFSET = 256 * 1024,
AUDIO_SAMPLES = CELL_AUDIO_BLOCK_SAMPLES,
AUDIO_MAX_BLOCK_COUNT = 32,
AUDIO_MAX_CHANNELS_COUNT = 8,
AUDIO_PORT_OFFSET = AUDIO_BUFFER_SAMPLES * AUDIO_MAX_BLOCK_COUNT * AUDIO_MAX_CHANNELS_COUNT * sizeof(f32),
EXTRA_AUDIO_BUFFERS = 8,
MAX_AUDIO_EVENT_QUEUES = 64,
AUDIO_BLOCK_SIZE_2CH = 2 * AUDIO_BUFFER_SAMPLES,
AUDIO_BLOCK_SIZE_8CH = 8 * AUDIO_BUFFER_SAMPLES,
PORT_BUFFER_TAG_COUNT = 4,
PORT_BUFFER_TAG_LAST_2CH = AUDIO_BLOCK_SIZE_2CH - 1,
PORT_BUFFER_TAG_DELTA_2CH = PORT_BUFFER_TAG_LAST_2CH / (PORT_BUFFER_TAG_COUNT - 1),
PORT_BUFFER_TAG_FIRST_2CH = PORT_BUFFER_TAG_LAST_2CH % (PORT_BUFFER_TAG_COUNT - 1),
PORT_BUFFER_TAG_LAST_8CH = AUDIO_BLOCK_SIZE_8CH - 1,
PORT_BUFFER_TAG_DELTA_8CH = PORT_BUFFER_TAG_LAST_8CH / (PORT_BUFFER_TAG_COUNT - 1),
PORT_BUFFER_TAG_FIRST_8CH = PORT_BUFFER_TAG_LAST_8CH % (PORT_BUFFER_TAG_COUNT - 1),
};
enum class audio_port_state : u32
@ -98,12 +116,14 @@ struct audio_port
vm::ptr<char> addr{};
vm::ptr<u64> index{};
u32 channel;
u32 block;
u32 num_channels;
u32 num_blocks;
u64 attr;
u64 tag;
u64 counter; // copy of global counter
u64 cur_pos;
u64 global_counter; // copy of global counter
u64 active_counter;
u32 size;
u64 timestamp; // copy of global timestamp
struct alignas(8) level_set_t
{
@ -113,25 +133,158 @@ struct audio_port
float level;
atomic_t<level_set_t> level_set;
u32 block_size() const
{
return num_channels * AUDIO_BUFFER_SAMPLES;
}
u32 buf_size() const
{
return block_size() * sizeof(float);
}
u32 position(s32 offset = 0) const
{
s32 ofs = (offset % num_blocks) + num_blocks;
return (cur_pos + ofs) % num_blocks;
}
u32 buf_addr(s32 offset = 0) const
{
return addr.addr() + position(offset) * buf_size();
}
to_be_t<float>* get_vm_ptr(s32 offset = 0) const
{
return vm::_ptr<f32>(buf_addr(offset));
}
// Tags
u32 prev_touched_tag_nr;
f32 tag_backup[AUDIO_MAX_BLOCK_COUNT][PORT_BUFFER_TAG_COUNT] = { 0 };
constexpr static bool is_tag(float val);
void tag(s32 offset = 0);
void apply_tag_backups(s32 offset = 0);
};
class audio_thread
class audio_ringbuffer
{
private:
const std::shared_ptr<AudioThread> backend;
const u32 num_allocated_buffers;
const u32 buf_sz;
const u32 audio_sampling_rate;
const u32 channels;
std::unique_ptr<AudioDumper> m_dump;
std::unique_ptr<float[]> buffer[MAX_AUDIO_BUFFERS];
const float silence_buffer[8 * AUDIO_BUFFER_SAMPLES] = { 0 };
bool backend_open = false;
bool playing = false;
bool emu_paused = false;
u64 update_timestamp = 0;
u64 play_timestamp = 0;
u64 last_remainder = 0;
u64 enqueued_samples = 0;
u32 next_buf = 0;
public:
audio_ringbuffer(u32 num_buffers, u32 audio_sampling_rate, u32 channels);
~audio_ringbuffer();
void play();
void enqueue(const float* in_buffer = nullptr);
void flush();
u64 update();
void enqueue_silence(u32 buf_count = 1);
float* get_buffer(u32 num) const
{
AUDIT(num < num_allocated_buffers);
AUDIT(buffer[num].get() != nullptr);
return buffer[num].get();
}
u32 get_buf_sz() const
{
return buf_sz;
}
u64 get_timestamp() const
{
return get_system_time() - Emu.GetPauseTime();
}
float* get_current_buffer() const
{
return get_buffer(next_buf);
}
u64 get_enqueued_samples() const
{
return enqueued_samples;
}
bool is_playing() const
{
return playing;
}
};
class cell_audio_thread
{
vm::ptr<char> m_buffer;
vm::ptr<u64> m_indexes;
u64 m_counter{};
std::unique_ptr<audio_ringbuffer> ringbuffer;
void reset_ports(s32 offset = 0);
void advance(u64 timestamp, bool reset = true);
std::tuple<u32, u32, u32, u32> count_port_buffer_tags();
template<bool downmix_to_2ch> void mix(float *out_buffer, s32 offset = 0);
void finish_port_volume_stepping();
constexpr static u64 get_thread_wait_delay(u64 time_left)
{
return (time_left > 1000) ? time_left - 750 : 100;
}
public:
const u64 start_time = get_system_time();
const s64 period_comparison_margin = 100; // When comparing the current period time with the desired period, if it is below this number of usecs we do not wait any longer
std::array<audio_port, AUDIO_PORT_COUNT> ports;
const u32 audio_channels = AudioThread::get_channels();
const u32 audio_sampling_rate = AudioThread::get_sampling_rate();
const u64 audio_block_period = AUDIO_BUFFER_SAMPLES * 1000000 / audio_sampling_rate;
const u64 desired_buffer_duration = g_cfg.audio.enable_buffering ? g_cfg.audio.desired_buffer_duration : 0;
const bool buffering_enabled = g_cfg.audio.enable_buffering && (desired_buffer_duration >= audio_block_period);
const u64 minimum_block_period = audio_block_period / 2; // the block period will not be dynamically lowered below this value (usecs)
const u64 maximum_block_period = audio_block_period + (audio_block_period - minimum_block_period); // the block period will not be dynamically increased above this value (usecs)
const u32 desired_full_buffers = buffering_enabled ? static_cast<u32>(desired_buffer_duration / audio_block_period) + 1 : 1;
const u32 num_allocated_buffers = desired_full_buffers + EXTRA_AUDIO_BUFFERS; // number of ringbuffer buffers
std::vector<u64> keys;
std::array<audio_port, AUDIO_PORT_COUNT> ports;
u64 m_last_period_end = 0;
u64 m_counter = 0;
u64 m_start_time = 0;
u64 m_dynamic_period = 0;
void operator()();
audio_thread(vm::ptr<char> buf, vm::ptr<u64> ind)
cell_audio_thread(vm::ptr<char> buf, vm::ptr<u64> ind)
: m_buffer(buf)
, m_indexes(ind)
{
@ -157,4 +310,4 @@ public:
}
};
using audio_config = named_thread<audio_thread>;
using cell_audio = named_thread<cell_audio_thread>;

31
rpcs3/Emu/Cell/Modules/libmixer.cpp
View File

@ -1,4 +1,4 @@
#include "stdafx.h"
#include "stdafx.h"
#include "Emu/System.h"
#include "Emu/IdManager.h"
#include "Emu/Cell/PPUModule.h"
@ -328,13 +328,13 @@ struct surmixer_thread : ppu_thread
void non_task()
{
const auto g_audio = fxm::get<audio_config>();
const auto g_audio = fxm::get<cell_audio>();
audio_port& port = g_audio->ports[g_surmx.audio_port];
while (port.state != audio_port_state::closed)
{
if (g_surmx.mixcount > (port.tag + 0)) // adding positive value (1-15): preemptive buffer filling (hack)
if (g_surmx.mixcount > (port.active_counter + 0)) // adding positive value (1-15): preemptive buffer filling (hack)
{
thread_ctrl::wait_for(1000); // hack
continue;
@ -432,7 +432,7 @@ struct surmixer_thread : ppu_thread
//u64 stamp2 = get_system_time();
auto buf = vm::_ptr<f32>(port.addr.addr() + (g_surmx.mixcount % port.block) * port.channel * AUDIO_SAMPLES * sizeof(float));
auto buf = vm::_ptr<f32>(port.addr.addr() + (g_surmx.mixcount % port.num_blocks) * port.num_channels * AUDIO_BUFFER_SAMPLES * sizeof(float));
for (auto& mixdata : g_surmx.mixdata)
{
@ -456,7 +456,7 @@ s32 cellSurMixerCreate(vm::cptr<CellSurMixerConfig> config)
{
libmixer.warning("cellSurMixerCreate(config=*0x%x)", config);
const auto g_audio = fxm::get<audio_config>();
const auto g_audio = fxm::get<cell_audio>();
const auto port = g_audio->open_port();
@ -472,11 +472,10 @@ s32 cellSurMixerCreate(vm::cptr<CellSurMixerConfig> config)
g_surmx.ch_strips_6 = config->chStrips6;
g_surmx.ch_strips_8 = config->chStrips8;
port->channel = 8;
port->block = 16;
port->num_channels = 8;
port->num_blocks = 16;
port->attr = 0;
port->size = port->channel * port->block * AUDIO_SAMPLES * sizeof(float);
port->tag = 0;
port->size = port->num_channels * port->num_blocks * AUDIO_BUFFER_SAMPLES * sizeof(float);
port->level = 1.0f;
port->level_set.store({ 1.0f, 0.0f });
@ -541,7 +540,7 @@ s32 cellSurMixerStart()
{
libmixer.warning("cellSurMixerStart()");
const auto g_audio = fxm::get<audio_config>();
const auto g_audio = fxm::get<cell_audio>();
if (g_surmx.audio_port >= AUDIO_PORT_COUNT)
{
@ -563,7 +562,7 @@ s32 cellSurMixerFinalize()
{
libmixer.warning("cellSurMixerFinalize()");
const auto g_audio = fxm::get<audio_config>();
const auto g_audio = fxm::get<cell_audio>();
if (g_surmx.audio_port >= AUDIO_PORT_COUNT)
{
@ -608,7 +607,7 @@ s32 cellSurMixerPause(u32 type)
{
libmixer.warning("cellSurMixerPause(type=%d)", type);
const auto g_audio = fxm::get<audio_config>();
const auto g_audio = fxm::get<cell_audio>();
if (g_surmx.audio_port >= AUDIO_PORT_COUNT)
{
@ -630,10 +629,12 @@ s32 cellSurMixerGetCurrentBlockTag(vm::ptr<u64> tag)
s32 cellSurMixerGetTimestamp(u64 tag, vm::ptr<u64> stamp)
{
libmixer.trace("cellSurMixerGetTimestamp(tag=0x%llx, stamp=*0x%x)", tag, stamp);
libmixer.error("cellSurMixerGetTimestamp(tag=0x%llx, stamp=*0x%x)", tag, stamp);
const auto g_audio = fxm::get<cell_audio>();
*stamp = g_audio->m_start_time + tag * AUDIO_BUFFER_SAMPLES * 1'000'000 / g_audio->audio_sampling_rate;
const auto g_audio = fxm::get<audio_config>();
*stamp = g_audio->start_time + (tag) * 256000000 / 48000; // ???
return CELL_OK;
}

4
rpcs3/Emu/System.h
View File

@ -526,9 +526,11 @@ struct cfg_root : cfg::node
cfg::_bool dump_to_file{this, "Dump to file"};
cfg::_bool convert_to_u16{this, "Convert to 16 bit"};
cfg::_bool downmix_to_2ch{this, "Downmix to Stereo", true};
cfg::_int<2, 128> frames{this, "Buffer Count", 32};
cfg::_int<1, 128> startt{this, "Start Threshold", 1};
cfg::_int<0, 200> volume{this, "Master Volume", 100};
cfg::_bool enable_buffering{this, "Enable Buffering", true};
cfg::_int <0, 250'000> desired_buffer_duration{this, "Desired Audio Buffer Duration", 100'000};
cfg::_int<1, 1000> sampling_period_multiplier{this, "Sampling Period Multiplier", 100};
} audio{this};

12
rpcs3/emucore.vcxproj.filters
View File

@ -928,9 +928,6 @@
<ClInclude Include="Emu\Audio\Null\NullAudioThread.h">
<Filter>Emu\Audio\Null</Filter>
</ClInclude>
<ClInclude Include="Emu\Audio\AudioThread.h">
<Filter>Emu\Audio</Filter>
</ClInclude>
<ClInclude Include="..\Utilities\File.h">
<Filter>Utilities</Filter>
</ClInclude>
@ -1443,7 +1440,7 @@
</ClInclude>
<ClInclude Include="..\Utilities\date_time.h">
<Filter>Utilities</Filter>
</ClInclude>
</ClInclude>
<ClInclude Include="..\Utilities\address_range.h">
<Filter>Utilities</Filter>
</ClInclude>
@ -1452,12 +1449,15 @@
</ClInclude>
<ClInclude Include="Emu\RSX\Common\texture_cache_utils.h">
<Filter>Emu\GPU\RSX\Common</Filter>
</ClInclude>
</ClInclude>
<ClInclude Include="Emu\RSX\RSXFIFO.h">
<Filter>Emu\GPU\RSX</Filter>
<Filter>Emu\GPU\RSX</Filter>
</ClInclude>
<ClInclude Include="Emu\RSX\Common\texture_cache_predictor.h">
<Filter>Emu\GPU\RSX\Common</Filter>
</ClInclude>
<ClInclude Include="Emu\Audio\AudioThread.h">
<Filter>Emu\Audio</Filter>
</ClInclude>
</ItemGroup>
</Project>

4
rpcs3/rpcs3_app.cpp
View File

@ -37,7 +37,7 @@
#include "Emu/RSX/Null/NullGSRender.h"
#include "Emu/RSX/GL/GLGSRender.h"
#include "Emu/Audio/Null/NullAudioThread.h"
#include "Emu/Audio/AL/OpenALThread.h"
//#include "Emu/Audio/AL/OpenALThread.h"
#ifdef _MSC_VER
#include "Emu/RSX/D3D12/D3D12GSRender.h"
#endif
@ -269,7 +269,7 @@ void rpcs3_app::InitializeCallbacks()
case audio_renderer::pulse: return std::make_shared<PulseThread>();
#endif
case audio_renderer::openal: return std::make_shared<OpenALThread>();
//case audio_renderer::openal: return std::make_shared<OpenALThread>();
default: fmt::throw_exception("Invalid audio renderer: %s" HERE, type);
}
};