Xargon/Duckstation
1
0
Fork 0
mirror of https://github.com/RetroDECK/Duckstation.git synced 2024-09-17 08:02:41 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity

AudioStream: Replace buffer queue with ring buffer

Browse source 
Should achieve a decent overall minimum latency reduction.
This commit is contained in:
Connor McLaughlin 2020-06-06 14:40:20 +10:00
parent 6acd8b27dd
commit 531c3ad5fa
16 changed files with 203 additions and 251 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

209
src/common/audio_stream.cpp
View file

@ -1,14 +1,16 @@
#include "audio_stream.h"
#include "assert.h"
#include "log.h"
#include <algorithm>
#include <cstring>
Log_SetChannel(AudioStream);
AudioStream::AudioStream() = default;
AudioStream::~AudioStream() = default;
bool AudioStream::Reconfigure(u32 output_sample_rate /*= DefaultOutputSampleRate*/, u32 channels /*= 1*/,
u32 buffer_size /*= DefaultBufferSize*/, u32 buffer_count /*= DefaultBufferCount*/)
u32 buffer_size /*= DefaultBufferSize*/)
{
if (IsDeviceOpen())
CloseDevice();
@ -16,13 +18,14 @@ bool AudioStream::Reconfigure(u32 output_sample_rate /*= DefaultOutputSampleRate
m_output_sample_rate = output_sample_rate;
m_channels = channels;
m_buffer_size = buffer_size;
AllocateBuffers(buffer_count);
m_output_paused = true;
if (!SetBufferSize(buffer_size))
return false;
if (!OpenDevice())
{
EmptyBuffers();
m_buffers.clear();
m_buffer_size = 0;
m_output_sample_rate = 0;
m_channels = 0;
@ -32,7 +35,7 @@ bool AudioStream::Reconfigure(u32 output_sample_rate /*= DefaultOutputSampleRate
return true;
}
void AudioStream::SetOutputVolume(s32 volume)
void AudioStream::SetOutputVolume(u32 volume)
{
std::unique_lock<std::mutex> lock(m_buffer_mutex);
m_output_volume = volume;
@ -58,7 +61,6 @@ void AudioStream::Shutdown()
CloseDevice();
EmptyBuffers();
m_buffers.clear();
m_buffer_size = 0;
m_output_sample_rate = 0;
m_channels = 0;
@ -69,182 +71,149 @@ void AudioStream::BeginWrite(SampleType** buffer_ptr, u32* num_frames)
{
m_buffer_mutex.lock();
EnsureBuffer();
EnsureBuffer(*num_frames * m_channels);
Buffer& buffer = m_buffers[m_first_free_buffer];
*buffer_ptr = buffer.data.data() + (buffer.write_position * m_channels);
*num_frames = m_buffer_size - buffer.write_position;
*buffer_ptr = m_buffer.GetWritePointer();
*num_frames = m_buffer.GetContiguousSpace() / m_channels;
}
void AudioStream::WriteFrames(const SampleType* frames, u32 num_frames)
{
u32 remaining_frames = num_frames;
const u32 num_samples = num_frames * m_channels;
std::unique_lock<std::mutex> lock(m_buffer_mutex);
while (remaining_frames > 0)
{
EnsureBuffer();
Buffer& buffer = m_buffers[m_first_free_buffer];
const u32 to_this_buffer = std::min(m_buffer_size - buffer.write_position, remaining_frames);
const u32 copy_count = to_this_buffer * m_channels;
std::memcpy(&buffer.data[buffer.write_position * m_channels], frames, copy_count * sizeof(SampleType));
frames += copy_count;
remaining_frames -= to_this_buffer;
buffer.write_position += to_this_buffer;
// End of the buffer?
if (buffer.write_position == m_buffer_size)
{
// Reset it back to the start, and enqueue it.
buffer.write_position = 0;
m_num_free_buffers--;
m_first_free_buffer = (m_first_free_buffer + 1) % m_buffers.size();
m_num_available_buffers++;
BufferAvailable();
}
}
EnsureBuffer(num_samples);
m_buffer.PushRange(frames, num_samples);
FramesAvailable();
}
void AudioStream::EndWrite(u32 num_frames)
{
Buffer& buffer = m_buffers[m_first_free_buffer];
DebugAssert((buffer.write_position + num_frames) <= m_buffer_size);
buffer.write_position += num_frames;
// End of the buffer?
if (buffer.write_position == m_buffer_size)
{
// Reset it back to the start, and enqueue it.
// Log_DevPrintf("Enqueue buffer %u", m_first_free_buffer);
buffer.write_position = 0;
m_num_free_buffers--;
m_first_free_buffer = (m_first_free_buffer + 1) % m_buffers.size();
m_num_available_buffers++;
BufferAvailable();
}
m_buffer.AdvanceTail(num_frames * m_channels);
FramesAvailable();
m_buffer_mutex.unlock();
}
float AudioStream::GetMinLatency(u32 sample_rate, u32 buffer_size, u32 buffer_count)
float AudioStream::GetMaxLatency(u32 sample_rate, u32 buffer_size)
{
return (static_cast<float>(buffer_size) / static_cast<float>(sample_rate));
}
float AudioStream::GetMaxLatency(u32 sample_rate, u32 buffer_size, u32 buffer_count)
bool AudioStream::SetBufferSize(u32 buffer_size)
{
return (static_cast<float>(buffer_size * (buffer_count - 1)) / static_cast<float>(sample_rate));
const u32 buffer_size_in_samples = buffer_size * m_channels;
const u32 max_samples = buffer_size_in_samples * 2u;
if (max_samples > m_buffer.GetCapacity())
return false;
m_buffer_size = buffer_size;
m_max_samples = max_samples;
return true;
}
u32 AudioStream::GetSamplesAvailable() const
{
// TODO: Use atomic loads
u32 available_buffers;
u32 available_samples;
{
std::unique_lock<std::mutex> lock(m_buffer_mutex);
available_buffers = m_num_available_buffers;
available_samples = m_buffer.GetSize();
}
return available_buffers * m_buffer_size;
return available_samples / m_channels;
}
u32 AudioStream::ReadSamples(SampleType* samples, u32 num_samples)
u32 AudioStream::GetSamplesAvailableLocked() const
{
u32 remaining_samples = num_samples;
std::unique_lock<std::mutex> lock(m_buffer_mutex);
return m_buffer.GetSize() / m_channels;
}
while (remaining_samples > 0 && m_num_available_buffers > 0)
void AudioStream::ReadFrames(SampleType* samples, u32 num_frames, bool apply_volume)
{
const u32 total_samples = num_frames * m_channels;
u32 samples_copied = 0;
{
Buffer& buffer = m_buffers[m_first_available_buffer];
const u32 from_this_buffer = std::min(m_buffer_size - buffer.read_position, remaining_samples);
const u32 copy_count = from_this_buffer * m_channels;
const SampleType* read_pointer = &buffer.data[buffer.read_position * m_channels];
for (u32 i = 0; i < copy_count; i++)
*(samples++) = ApplyVolume(*(read_pointer++), m_output_volume);
std::unique_lock<std::mutex> lock(m_buffer_mutex);
samples_copied = std::min(m_buffer.GetSize(), total_samples);
if (samples_copied > 0)
m_buffer.PopRange(samples, samples_copied);
remaining_samples -= from_this_buffer;
buffer.read_position += from_this_buffer;
m_buffer_draining_cv.notify_one();
}
if (buffer.read_position == m_buffer_size)
if (samples_copied < total_samples)
{
if (samples_copied > 0)
{
// Log_DevPrintf("Finish dequeing buffer %u", m_first_available_buffer);
// End of this buffer.
buffer.read_position = 0;
m_num_available_buffers--;
m_first_available_buffer = (m_first_available_buffer + 1) % m_buffers.size();
m_num_free_buffers++;
m_buffer_available_cv.notify_one();
m_resample_buffer.resize(samples_copied);
std::memcpy(m_resample_buffer.data(), samples, sizeof(SampleType) * samples_copied);
// super basic resampler - spread the input samples evenly across the output samples. will sound like ass and have
// aliasing, but better than popping by inserting silence.
const u32 increment =
static_cast<u32>(65536.0f * (static_cast<float>(samples_copied / m_channels) / static_cast<float>(num_frames)));
SampleType* out_ptr = samples;
const SampleType* resample_ptr = m_resample_buffer.data();
const u32 copy_stride = sizeof(SampleType) * m_channels;
u32 resample_subpos = 0;
for (u32 i = 0; i < num_frames; i++)
{
std::memcpy(out_ptr, resample_ptr, copy_stride);
out_ptr += m_channels;
resample_subpos += increment;
resample_ptr += (resample_subpos >> 16) * m_channels;
resample_subpos %= 65536u;
}
Log_DevPrintf("Audio buffer underflow, resampled %u frames to %u", samples_copied / m_channels, num_frames);
}
else
{
// read nothing, so zero-fill
std::memset(samples, 0, sizeof(SampleType) * total_samples);
Log_DevPrintf("Audio buffer underflow with no samples, added %u frames silence", num_frames);
}
}
return num_samples - remaining_samples;
}
void AudioStream::AllocateBuffers(u32 buffer_count)
{
m_buffers.resize(buffer_count);
for (u32 i = 0; i < buffer_count; i++)
if (apply_volume && m_output_volume != FullVolume)
{
Buffer& buffer = m_buffers[i];
buffer.data.resize(m_buffer_size * m_channels);
buffer.read_position = 0;
buffer.write_position = 0;
SampleType* current_ptr = samples;
const SampleType* end_ptr = samples + (num_frames * m_channels);
while (current_ptr != end_ptr)
{
*current_ptr = ApplyVolume(*current_ptr, m_output_volume);
current_ptr++;
}
}
m_first_available_buffer = 0;
m_num_available_buffers = 0;
m_first_free_buffer = 0;
m_num_free_buffers = buffer_count;
}
void AudioStream::EnsureBuffer()
void AudioStream::EnsureBuffer(u32 size)
{
if (m_num_free_buffers > 0)
if (GetBufferSpace() >= size)
return;
if (m_sync)
{
std::unique_lock<std::mutex> lock(m_buffer_mutex, std::adopt_lock);
m_buffer_available_cv.wait(lock, [this]() { return m_num_free_buffers > 0; });
m_buffer_draining_cv.wait(lock, [this, size]() { return GetBufferSpace() >= size; });
lock.release();
}
else
{
DropBuffer();
m_buffer.Remove(size);
}
}
void AudioStream::DropBuffer()
void AudioStream::DropFrames(u32 count)
{
DebugAssert(m_num_available_buffers > 0);
// Log_DevPrintf("Dropping buffer %u", m_first_free_buffer);
// Out of space. We'll overwrite the oldest buffer with the new data.
// At the same time, we shift the available buffer forward one.
m_first_available_buffer = (m_first_available_buffer + 1) % m_buffers.size();
m_num_available_buffers--;
m_buffers[m_first_free_buffer].read_position = 0;
m_buffers[m_first_free_buffer].write_position = 0;
m_num_free_buffers++;
m_buffer.Remove(count);
}
void AudioStream::EmptyBuffers()
{
std::unique_lock<std::mutex> lock(m_buffer_mutex);
for (Buffer& buffer : m_buffers)
{
buffer.read_position = 0;
buffer.write_position = 0;
}
m_first_free_buffer = 0;
m_num_free_buffers = static_cast<u32>(m_buffers.size());
m_first_available_buffer = 0;
m_num_available_buffers = 0;
m_buffer.Clear();
}

61
src/common/audio_stream.h
View file

@ -1,4 +1,5 @@
#pragma once
#include "fifo_queue.h"
#include "types.h"
#include <condition_variable>
#include <memory>
@ -12,11 +13,12 @@ class AudioStream
public:
using SampleType = s16;
enum
enum : u32
{
DefaultOutputSampleRate = 44100,
DefaultBufferSize = 2048,
DefaultBufferCount = 3,
MaxSamples = 32768,
FullVolume = 100
};
AudioStream();
@ -25,14 +27,14 @@ public:
u32 GetOutputSampleRate() const { return m_output_sample_rate; }
u32 GetChannels() const { return m_channels; }
u32 GetBufferSize() const { return m_buffer_size; }
u32 GetBufferCount() const { return static_cast<u32>(m_buffers.size()); }
s32 GetOutputVolume() const { return m_output_volume; }
bool IsSyncing() const { return m_sync; }
bool Reconfigure(u32 output_sample_rate = DefaultOutputSampleRate, u32 channels = 1,
u32 buffer_size = DefaultBufferSize, u32 buffer_count = DefaultBufferCount);
u32 buffer_size = DefaultBufferSize);
void SetSync(bool enable) { m_sync = enable; }
void SetOutputVolume(s32 volume);
virtual void SetOutputVolume(u32 volume);
void PauseOutput(bool paused);
void EmptyBuffers();
@ -48,58 +50,43 @@ public:
static std::unique_ptr<AudioStream> CreateCubebAudioStream();
// Latency computation - returns values in seconds
static float GetMinLatency(u32 sample_rate, u32 buffer_size, u32 buffer_count);
static float GetMaxLatency(u32 sample_rate, u32 buffer_size, u32 buffer_count);
static float GetMaxLatency(u32 sample_rate, u32 buffer_size);
protected:
virtual bool OpenDevice() = 0;
virtual void PauseDevice(bool paused) = 0;
virtual void CloseDevice() = 0;
virtual void BufferAvailable() = 0;
virtual void FramesAvailable() = 0;
ALWAYS_INLINE static SampleType ApplyVolume(SampleType sample, s32 volume)
ALWAYS_INLINE static SampleType ApplyVolume(SampleType sample, u32 volume)
{
return s16((s32(sample) * volume) / 100);
return s16((s32(sample) * s32(volume)) / 100);
}
bool SetBufferSize(u32 buffer_size);
bool IsDeviceOpen() const { return (m_output_sample_rate > 0); }
u32 GetSamplesAvailable() const;
u32 ReadSamples(SampleType* samples, u32 num_samples);
void DropBuffer();
u32 GetSamplesAvailableLocked() const;
void ReadFrames(SampleType* samples, u32 num_frames, bool apply_volume);
void DropFrames(u32 count);
u32 m_output_sample_rate = 0;
u32 m_channels = 0;
u32 m_buffer_size = 0;
private:
struct Buffer
{
std::vector<SampleType> data;
u32 write_position;
u32 read_position;
};
// volume, 0-100
u32 m_output_volume = FullVolume;
void AllocateBuffers(u32 buffer_count);
void EnsureBuffer();
private:
ALWAYS_INLINE u32 GetBufferSpace() const { return (m_max_samples - m_buffer.GetSize()); }
void EnsureBuffer(u32 size);
std::vector<Buffer> m_buffers;
HeapFIFOQueue<SampleType, MaxSamples> m_buffer;
mutable std::mutex m_buffer_mutex;
// For input.
u32 m_first_free_buffer = 0;
u32 m_num_free_buffers = 0;
// For output.
u32 m_num_available_buffers = 0;
u32 m_first_available_buffer = 0;
// TODO: Switch to semaphore
std::condition_variable m_buffer_available_cv;
// volume, 0-100
s32 m_output_volume = 100;
std::condition_variable m_buffer_draining_cv;
std::vector<SampleType> m_resample_buffer;
u32 m_max_samples = 0;
bool m_output_paused = true;
bool m_sync = true;

33
src/common/cubeb_audio_stream.cpp
View file

@ -56,9 +56,21 @@ bool CubebAudioStream::OpenDevice()
Log_InfoPrintf("Minimum latency in frames: %u", latency_frames);
if (latency_frames > m_buffer_size)
Log_WarningPrintf("Minimum latency is above buffer size: %u vs %u", latency_frames, m_buffer_size);
{
Log_WarningPrintf("Minimum latency is above buffer size: %u vs %u, adjusting to compensate.", latency_frames,
m_buffer_size);
if (!SetBufferSize(latency_frames))
{
Log_ErrorPrintf("Failed to set new buffer size of %u frames", latency_frames);
DestroyContext();
return false;
}
}
else
{
latency_frames = m_buffer_size;
}
char stream_name[32];
std::snprintf(stream_name, sizeof(stream_name), "AudioStream_%p", this);
@ -72,6 +84,7 @@ bool CubebAudioStream::OpenDevice()
return false;
}
cubeb_stream_set_volume(m_cubeb_stream, static_cast<float>(m_output_volume) / 100.0f);
return true;
}
@ -106,15 +119,13 @@ long CubebAudioStream::DataCallback(cubeb_stream* stm, void* user_ptr, const voi
{
CubebAudioStream* const this_ptr = static_cast<CubebAudioStream*>(user_ptr);
const u32 read_frames =
this_ptr->ReadSamples(reinterpret_cast<SampleType*>(output_buffer), static_cast<u32>(nframes));
const u32 silence_frames = static_cast<u32>(nframes) - read_frames;
if (silence_frames > 0)
if (this_ptr->m_output_volume_changed.load())
{
std::memset(reinterpret_cast<SampleType*>(output_buffer) + (read_frames * this_ptr->m_channels), 0,
silence_frames * this_ptr->m_channels * sizeof(SampleType));
this_ptr->m_output_volume_changed.store(false);
cubeb_stream_set_volume(this_ptr->m_cubeb_stream, static_cast<float>(this_ptr->m_output_volume) / 100.0f);
}
this_ptr->ReadFrames(reinterpret_cast<SampleType*>(output_buffer), static_cast<u32>(nframes), false);
return nframes;
}
@ -125,7 +136,7 @@ void CubebAudioStream::StateCallback(cubeb_stream* stream, void* user_ptr, cubeb
this_ptr->m_paused = (state != CUBEB_STATE_STARTED);
}
void CubebAudioStream::BufferAvailable() {}
void CubebAudioStream::FramesAvailable() {}
void CubebAudioStream::DestroyContext()
{
@ -138,6 +149,12 @@ void CubebAudioStream::DestroyContext()
#endif
}
void CubebAudioStream::SetOutputVolume(u32 volume)
{
AudioStream::SetOutputVolume(volume);
m_output_volume_changed.store(true);
}
std::unique_ptr<AudioStream> AudioStream::CreateCubebAudioStream()
{
return std::make_unique<CubebAudioStream>();

6
src/common/cubeb_audio_stream.h
View file

@ -1,6 +1,7 @@
#pragma once
#include "common/audio_stream.h"
#include "cubeb/cubeb.h"
#include <atomic>
#include <cstdint>
class CubebAudioStream final : public AudioStream
@ -9,13 +10,15 @@ public:
CubebAudioStream();
~CubebAudioStream();
void SetOutputVolume(u32 volume) override;
protected:
bool IsOpen() const { return m_cubeb_stream != nullptr; }
bool OpenDevice() override;
void PauseDevice(bool paused) override;
void CloseDevice() override;
void BufferAvailable() override;
void FramesAvailable() override;
void DestroyContext();
@ -26,6 +29,7 @@ protected:
cubeb* m_cubeb_context = nullptr;
cubeb_stream* m_cubeb_stream = nullptr;
bool m_paused = true;
std::atomic_bool m_output_volume_changed{ false };
#ifdef WIN32
bool m_com_initialized_by_us = false;

13
src/common/fifo_queue.h
View file

@ -17,9 +17,10 @@ class FIFOQueue
public:
const T* GetDataPointer() const { return m_ptr; }
T* GetDataPointer() { return m_ptr; }
const T* GetFrontPointer() const { return &m_ptr[m_head]; }
T* GetFrontPointer() { return &m_ptr[m_head]; }
const T* GetReadPointer() const { return &m_ptr[m_head]; }
T* GetReadPointer() { return &m_ptr[m_head]; }
constexpr u32 GetCapacity() const { return CAPACITY; }
T* GetWritePointer() { return &m_ptr[m_tail]; }
u32 GetSize() const { return m_size; }
u32 GetSpace() const { return CAPACITY - m_size; }
u32 GetContiguousSpace() const { return (m_tail >= m_head) ? (CAPACITY - m_tail) : (m_head - m_tail); }
@ -148,6 +149,14 @@ public:
}
}
void AdvanceTail(u32 count)
{
DebugAssert((m_size + count) < CAPACITY);
DebugAssert((m_tail + count) <= CAPACITY);
m_tail = (m_tail + count) % CAPACITY;
m_size += count;
}
protected:
FIFOQueue() = default;

4
src/common/null_audio_stream.cpp
View file

@ -13,10 +13,10 @@ void NullAudioStream::PauseDevice(bool paused) {}
void NullAudioStream::CloseDevice() {}
void NullAudioStream::BufferAvailable()
void NullAudioStream::FramesAvailable()
{
// drop any buffer as soon as they're available
DropBuffer();
DropFrames(GetSamplesAvailableLocked());
}
std::unique_ptr<AudioStream> AudioStream::CreateNullAudioStream()

2
src/common/null_audio_stream.h
View file

@ -11,5 +11,5 @@ protected:
bool OpenDevice() override;
void PauseDevice(bool paused) override;
void CloseDevice() override;
void BufferAvailable() override;
void FramesAvailable() override;
};

14
src/core/host_interface.cpp
View file

@ -61,20 +61,18 @@ void HostInterface::Shutdown() {}
void HostInterface::CreateAudioStream()
{
Log_InfoPrintf("Creating '%s' audio stream, sample rate = %u, channels = %u, buffer size = %u, buffer count = %u",
Log_InfoPrintf("Creating '%s' audio stream, sample rate = %u, channels = %u, buffer size = %u",
Settings::GetAudioBackendName(m_settings.audio_backend), AUDIO_SAMPLE_RATE, AUDIO_CHANNELS,
m_settings.audio_buffer_size, m_settings.audio_buffer_count);
m_settings.audio_buffer_size);
m_audio_stream = CreateAudioStream(m_settings.audio_backend);
if (!m_audio_stream || !m_audio_stream->Reconfigure(AUDIO_SAMPLE_RATE, AUDIO_CHANNELS, m_settings.audio_buffer_size,
m_settings.audio_buffer_count))
if (!m_audio_stream || !m_audio_stream->Reconfigure(AUDIO_SAMPLE_RATE, AUDIO_CHANNELS, m_settings.audio_buffer_size))
{
ReportFormattedError("Failed to create or configure audio stream, falling back to null output.");
m_audio_stream.reset();
m_audio_stream = AudioStream::CreateNullAudioStream();
m_audio_stream->Reconfigure(AUDIO_SAMPLE_RATE, AUDIO_CHANNELS, m_settings.audio_buffer_size,
m_settings.audio_buffer_count);
m_audio_stream->Reconfigure(AUDIO_SAMPLE_RATE, AUDIO_CHANNELS, m_settings.audio_buffer_size);
}
m_audio_stream->SetOutputVolume(m_settings.audio_output_muted ? 0 : m_settings.audio_output_volume);
@ -1011,7 +1009,6 @@ void HostInterface::SetDefaultSettings(SettingsInterface& si)
si.SetStringValue("Audio", "Backend", Settings::GetAudioBackendName(AudioBackend::Cubeb));
si.SetIntValue("Audio", "OutputVolume", 100);
si.SetIntValue("Audio", "BufferSize", DEFAULT_AUDIO_BUFFER_SIZE);
si.SetIntValue("Audio", "BufferCount", DEFAULT_AUDIO_BUFFER_COUNT);
si.SetIntValue("Audio", "OutputMuted", false);
si.SetBoolValue("Audio", "Sync", true);
si.SetBoolValue("Audio", "DumpOnBoot", false);
@ -1072,8 +1069,7 @@ void HostInterface::UpdateSettings(SettingsInterface& si)
}
if (m_settings.audio_backend != old_settings.audio_backend ||
m_settings.audio_buffer_size != old_settings.audio_buffer_size ||
m_settings.audio_buffer_count != old_settings.audio_buffer_count)
m_settings.audio_buffer_size != old_settings.audio_buffer_size)
{
if (m_settings.audio_backend != old_settings.audio_backend)
ReportFormattedMessage("Switching to %s audio backend.",

3
src/core/host_interface.h
View file

@ -38,8 +38,7 @@ public:
{
AUDIO_SAMPLE_RATE = 44100,
AUDIO_CHANNELS = 2,
DEFAULT_AUDIO_BUFFER_SIZE = 2048,
DEFAULT_AUDIO_BUFFER_COUNT = 4
DEFAULT_AUDIO_BUFFER_SIZE = 2048
};
struct SaveStateInfo

2
src/core/settings.cpp
View file

@ -61,7 +61,6 @@ void Settings::Load(SettingsInterface& si)
ParseAudioBackend(si.GetStringValue("Audio", "Backend", "Cubeb").c_str()).value_or(AudioBackend::Cubeb);
audio_output_volume = si.GetIntValue("Audio", "OutputVolume", 100);
audio_buffer_size = si.GetIntValue("Audio", "BufferSize", HostInterface::DEFAULT_AUDIO_BUFFER_SIZE);
audio_buffer_count = si.GetIntValue("Audio", "BufferCount", HostInterface::DEFAULT_AUDIO_BUFFER_COUNT);
audio_output_muted = si.GetBoolValue("Audio", "OutputMuted", false);
audio_sync_enabled = si.GetBoolValue("Audio", "Sync", true);
audio_dump_on_boot = si.GetBoolValue("Audio", "DumpOnBoot", false);
@ -151,7 +150,6 @@ void Settings::Save(SettingsInterface& si) const
si.SetStringValue("Audio", "Backend", GetAudioBackendName(audio_backend));
si.SetIntValue("Audio", "OutputVolume", audio_output_volume);
si.SetIntValue("Audio", "BufferSize", audio_buffer_size);
si.SetIntValue("Audio", "BufferCount", audio_buffer_count);
si.SetBoolValue("Audio", "OutputMuted", audio_output_muted);
si.SetBoolValue("Audio", "Sync", audio_sync_enabled);
si.SetBoolValue("Audio", "DumpOnBoot", audio_dump_on_boot);

1
src/core/settings.h
View file

@ -71,7 +71,6 @@ struct Settings
AudioBackend audio_backend = AudioBackend::Cubeb;
s32 audio_output_volume = 100;
u32 audio_buffer_size = 2048;
u32 audio_buffer_count = 4;
bool audio_output_muted = false;
bool audio_sync_enabled = true;
bool audio_dump_on_boot = true;

2
src/core/spu.cpp
View file

@ -674,7 +674,7 @@ void SPU::Execute(TickCount ticks)
{
AudioStream* const output_stream = m_system->GetHostInterface()->GetAudioStream();
s16* output_frame_start;
u32 output_frame_space;
u32 output_frame_space = remaining_frames;
output_stream->BeginWrite(&output_frame_start, &output_frame_space);
s16* output_frame = output_frame_start;

13
src/duckstation-qt/audiosettingswidget.cpp
View file

@ -14,13 +14,11 @@ AudioSettingsWidget::AudioSettingsWidget(QtHostInterface* host_interface, QWidge
&Settings::ParseAudioBackend, &Settings::GetAudioBackendName);
SettingWidgetBinder::BindWidgetToBoolSetting(m_host_interface, m_ui.syncToOutput, "Audio/Sync");
SettingWidgetBinder::BindWidgetToIntSetting(m_host_interface, m_ui.bufferSize, "Audio/BufferSize");
SettingWidgetBinder::BindWidgetToIntSetting(m_host_interface, m_ui.bufferCount, "Audio/BufferCount");
SettingWidgetBinder::BindWidgetToIntSetting(m_host_interface, m_ui.volume, "Audio/OutputVolume");
SettingWidgetBinder::BindWidgetToBoolSetting(m_host_interface, m_ui.muted, "Audio/OutputMuted");
SettingWidgetBinder::BindWidgetToBoolSetting(m_host_interface, m_ui.startDumpingOnBoot, "Audio/DumpOnBoot");
connect(m_ui.bufferSize, &QSlider::valueChanged, this, &AudioSettingsWidget::updateBufferingLabel);
connect(m_ui.bufferCount, &QSlider::valueChanged, this, &AudioSettingsWidget::updateBufferingLabel);
connect(m_ui.volume, &QSlider::valueChanged, this, &AudioSettingsWidget::updateVolumeLabel);
updateBufferingLabel();
@ -32,14 +30,9 @@ AudioSettingsWidget::~AudioSettingsWidget() = default;
void AudioSettingsWidget::updateBufferingLabel()
{
const u32 buffer_size = static_cast<u32>(m_ui.bufferSize->value());
const u32 buffer_count = static_cast<u32>(m_ui.bufferCount->value());
const float min_latency = AudioStream::GetMinLatency(HostInterface::AUDIO_SAMPLE_RATE, buffer_size, buffer_count);
const float max_latency = AudioStream::GetMaxLatency(HostInterface::AUDIO_SAMPLE_RATE, buffer_size, buffer_count);
m_ui.bufferingLabel->setText(tr("%1 samples, %2 buffers (min %3ms, max %4ms)")
.arg(buffer_size)
.arg(buffer_count)
.arg(min_latency * 1000.0f, 0, 'f', 2)
.arg(max_latency * 1000.0f, 0, 'f', 2));
const float max_latency = AudioStream::GetMaxLatency(HostInterface::AUDIO_SAMPLE_RATE, buffer_size);
m_ui.bufferingLabel->setText(
tr("Maximum latency: %1 frames (%2ms)").arg(buffer_size).arg(max_latency * 1000.0f, 0, 'f', 2));
}
void AudioSettingsWidget::updateVolumeLabel()

39
src/duckstation-qt/audiosettingswidget.ui
View file

@ -74,7 +74,7 @@
</property>
</widget>
</item>
<item row="3" column="0">
<item row="2" column="0">
<spacer name="horizontalSpacer">
<property name="orientation">
<enum>Qt::Horizontal</enum>
@ -87,59 +87,30 @@
</property>
</spacer>
</item>
<item row="3" column="1">
<item row="2" column="1">
<widget class="QLabel" name="bufferingLabel">
<property name="text">
<string>2048 samples, 4 buffers (min 0.00ms, max 0.00ms))</string>
<string>Maximum latency: 0 frames (0.00ms)</string>
</property>
<property name="alignment">
<set>Qt::AlignCenter</set>
</property>
</widget>
</item>
<item row="4" column="0" colspan="2">
<item row="3" column="0" colspan="2">
<widget class="QCheckBox" name="syncToOutput">
<property name="text">
<string>Sync To Output</string>
</property>
</widget>
</item>
<item row="5" column="0" colspan="2">
<item row="4" column="0" colspan="2">
<widget class="QCheckBox" name="startDumpingOnBoot">
<property name="text">
<string>Start Dumping On Boot</string>
</property>
</widget>
</item>
<item row="2" column="0">
<widget class="QLabel" name="label_4">
<property name="text">
<string>Buffer Count:</string>
</property>
</widget>
</item>
<item row="2" column="1">
<widget class="QSlider" name="bufferCount">
<property name="minimum">
<number>2</number>
</property>
<property name="maximum">
<number>8</number>
</property>
<property name="pageStep">
<number>1</number>
</property>
<property name="orientation">
<enum>Qt::Horizontal</enum>
</property>
<property name="tickPosition">
<enum>QSlider::TicksBothSides</enum>
</property>
<property name="tickInterval">
<number>1</number>
</property>
</widget>
</item>
</layout>
</widget>
</item>

46
src/frontend-common/sdl_audio_stream.cpp
View file

@ -1,7 +1,7 @@
#include "sdl_audio_stream.h"
#include "sdl_initializer.h"
#include "common/assert.h"
#include "common/log.h"
#include "sdl_initializer.h"
#include <SDL.h>
Log_SetChannel(SDLAudioStream);
@ -9,7 +9,7 @@ SDLAudioStream::SDLAudioStream() = default;
SDLAudioStream::~SDLAudioStream()
{
if (m_is_open)
if (IsOpen())
SDLAudioStream::CloseDevice();
}
@ -20,7 +20,7 @@ std::unique_ptr<SDLAudioStream> SDLAudioStream::Create()
bool SDLAudioStream::OpenDevice()
{
DebugAssert(!m_is_open);
DebugAssert(!IsOpen());
FrontendCommon::EnsureSDLInitialized();
@ -38,41 +38,49 @@ bool SDLAudioStream::OpenDevice()
spec.callback = AudioCallback;
spec.userdata = static_cast<void*>(this);
if (SDL_OpenAudio(&spec, nullptr) < 0)
SDL_AudioSpec obtained_spec = {};
m_device_id = SDL_OpenAudioDevice(nullptr, 0, &spec, &obtained_spec, SDL_AUDIO_ALLOW_SAMPLES_CHANGE);
if (m_device_id == 0)
{
Log_ErrorPrintf("SDL_OpenAudio failed");
Log_ErrorPrintf("SDL_OpenAudioDevice() failed: %s", SDL_GetError());
SDL_QuitSubSystem(SDL_INIT_AUDIO);
return false;
}
m_is_open = true;
if (obtained_spec.samples > spec.samples)
{
Log_WarningPrintf("Requested buffer size %u, got buffer size %u. Adjusting to compensate.", spec.samples,
obtained_spec.samples);
if (!SetBufferSize(obtained_spec.samples))
{
Log_ErrorPrintf("Failed to set new buffer size of %u", obtained_spec.samples);
CloseDevice();
return false;
}
}
return true;
}
void SDLAudioStream::PauseDevice(bool paused)
{
SDL_PauseAudio(paused ? 1 : 0);
SDL_PauseAudioDevice(m_device_id, paused ? 1 : 0);
}
void SDLAudioStream::CloseDevice()
{
DebugAssert(m_is_open);
SDL_CloseAudio();
SDL_CloseAudioDevice(m_device_id);
SDL_QuitSubSystem(SDL_INIT_AUDIO);
m_is_open = false;
m_device_id = 0;
}
void SDLAudioStream::AudioCallback(void* userdata, uint8_t* stream, int len)
{
SDLAudioStream* const this_ptr = static_cast<SDLAudioStream*>(userdata);
const u32 num_samples = len / sizeof(SampleType) / this_ptr->m_channels;
const u32 read_samples = this_ptr->ReadSamples(reinterpret_cast<SampleType*>(stream), num_samples);
const u32 silence_samples = num_samples - read_samples;
if (silence_samples > 0)
{
std::memset(reinterpret_cast<SampleType*>(stream) + (read_samples * this_ptr->m_channels), 0,
silence_samples * this_ptr->m_channels * sizeof(SampleType));
}
const u32 num_frames = len / sizeof(SampleType) / this_ptr->m_channels;
this_ptr->ReadFrames(reinterpret_cast<SampleType*>(stream), num_frames, false);
}
void SDLAudioStream::BufferAvailable() {}
void SDLAudioStream::FramesAvailable() {}

6
src/frontend-common/sdl_audio_stream.h
View file

@ -11,12 +11,14 @@ public:
static std::unique_ptr<SDLAudioStream> Create();
protected:
ALWAYS_INLINE bool IsOpen() const { return (m_device_id != 0); }
bool OpenDevice() override;
void PauseDevice(bool paused) override;
void CloseDevice() override;
void BufferAvailable() override;
void FramesAvailable() override;
static void AudioCallback(void* userdata, uint8_t* stream, int len);
bool m_is_open = false;
u32 m_device_id = 0;
};