Files
archived-pcsx2/pcsx2/Host/AudioStream.cpp

908 lines
28 KiB
C++

// SPDX-FileCopyrightText: 2002-2026 PCSX2 Dev Team
// SPDX-License-Identifier: GPL-3.0+
#include "Host/AudioStream.h"
#include "FreeSurroundDecoder.h"
#include "Host.h"
#include "GS/GSVector.h"
#include "common/Assertions.h"
#include "common/BitUtils.h"
#include "common/Console.h"
#include "common/Error.h"
#include "common/Pcsx2Defs.h"
#include "common/SettingsWrapper.h"
#include "common/SmallString.h"
#include "common/Timer.h"
#include "SoundTouch.h"
#include <algorithm>
#include <bit>
#include <cmath>
#include <cstring>
#include <limits>
//#define LOG_UNDERRUN(...) DEV_LOG(__VA_ARGS__)
#define LOG_UNDERRUN(...) (void)0
static constexpr bool LOG_TIMESTRETCH_STATS = false;
static constexpr const std::array<std::pair<u8, u8>, static_cast<size_t>(AudioExpansionMode::Count)>
s_expansion_channel_count = {{
{u8(2), u8(2)}, // Disabled
{u8(3), u8(3)}, // StereoLFE
{u8(5), u8(4)}, // Quadraphonic
{u8(5), u8(5)}, // QuadraphonicLFE
{u8(6), u8(6)}, // Surround51
{u8(8), u8(8)}, // Surround71
}};
AudioStream::DeviceInfo::DeviceInfo(std::string name_, std::string display_name_, u32 minimum_latency_)
: name(std::move(name_))
, display_name(std::move(display_name_))
, minimum_latency_frames(minimum_latency_)
{
}
AudioStream::DeviceInfo::~DeviceInfo() = default;
AudioStream::AudioStream(u32 sample_rate, const AudioStreamParameters& parameters)
: m_sample_rate(sample_rate)
, m_parameters(parameters)
, m_internal_channels(s_expansion_channel_count[static_cast<size_t>(parameters.expansion_mode)].first)
, m_output_channels(s_expansion_channel_count[static_cast<size_t>(parameters.expansion_mode)].second)
{
}
AudioStream::~AudioStream()
{
DestroyBuffer();
}
std::unique_ptr<AudioStream> AudioStream::CreateNullStream(u32 sample_rate, u32 buffer_ms)
{
// no point stretching with no output
AudioStreamParameters params;
params.expansion_mode = AudioExpansionMode::Disabled;
params.buffer_ms = static_cast<u16>(buffer_ms);
std::unique_ptr<AudioStream> stream(new AudioStream(sample_rate, params));
stream->BaseInitialize(&StereoSampleReaderImpl, false);
stream->SetOutputVolume(0);
return stream;
}
std::vector<std::pair<std::string, std::string>> AudioStream::GetDriverNames(AudioBackend backend)
{
std::vector<std::pair<std::string, std::string>> ret;
switch (backend)
{
case AudioBackend::Cubeb:
ret = GetCubebDriverNames();
break;
default:
break;
}
return ret;
}
std::vector<AudioStream::DeviceInfo> AudioStream::GetOutputDevices(AudioBackend backend, const char* driver)
{
std::vector<AudioStream::DeviceInfo> ret;
switch (backend)
{
case AudioBackend::Cubeb:
ret = GetCubebOutputDevices(driver);
break;
default:
break;
}
return ret;
}
std::unique_ptr<AudioStream> AudioStream::CreateStream(AudioBackend backend, u32 sample_rate, const AudioStreamParameters& parameters,
const char* driver_name, const char* device_name, bool stretch_enabled, Error* error)
{
INFO_LOG("Creating {} audio stream, sample rate = {}, expansion = {}, buffer = {}, latency = {}, stretching {}, driver = {}, device = {}",
GetBackendName(backend), sample_rate, GetExpansionModeName(parameters.expansion_mode), parameters.buffer_ms, parameters.output_latency_ms,
stretch_enabled ? "enabled" : "disabled", driver_name, device_name);
switch (backend)
{
case AudioBackend::Cubeb:
return CreateCubebAudioStream(sample_rate, parameters, driver_name, device_name, stretch_enabled, error);
case AudioBackend::SDL:
return CreateSDLAudioStream(sample_rate, parameters, stretch_enabled, error);
case AudioBackend::Null:
return CreateNullStream(sample_rate, parameters.buffer_ms);
default:
Error::SetStringView(error, "Unknown audio backend.");
return nullptr;
}
}
u32 AudioStream::GetAlignedBufferSize(u32 size)
{
static_assert(std::has_single_bit(CHUNK_SIZE));
return Common::AlignUpPow2(size, CHUNK_SIZE);
}
u32 AudioStream::GetBufferSizeForMS(u32 sample_rate, u32 ms)
{
return GetAlignedBufferSize((ms * sample_rate) / 1000u);
}
u32 AudioStream::GetMSForBufferSize(u32 sample_rate, u32 buffer_size)
{
buffer_size = GetAlignedBufferSize(buffer_size);
return (buffer_size * 1000u) / sample_rate;
}
static constexpr const std::array s_backend_names = {
"Null",
"Cubeb",
"SDL",
};
static constexpr const std::array s_backend_display_names = {
TRANSLATE_NOOP("AudioStream", "Null (No Output)"),
TRANSLATE_NOOP("AudioStream", "Cubeb"),
TRANSLATE_NOOP("AudioStream", "SDL"),
};
std::optional<AudioBackend> AudioStream::ParseBackendName(const char* str)
{
int index = 0;
for (const char* name : s_backend_names)
{
if (std::strcmp(name, str) == 0)
return static_cast<AudioBackend>(index);
index++;
}
return std::nullopt;
}
const char* AudioStream::GetBackendName(AudioBackend backend)
{
return s_backend_names[static_cast<int>(backend)];
}
const char* AudioStream::GetBackendDisplayName(AudioBackend backend)
{
return Host::TranslateToCString("AudioStream", s_backend_display_names[static_cast<int>(backend)]);
}
static constexpr const std::array s_expansion_mode_names = {
"Disabled",
"StereoLFE",
"Quadraphonic",
"QuadraphonicLFE",
"Surround51",
"Surround71",
};
static constexpr const std::array s_expansion_mode_display_names = {
TRANSLATE_NOOP("AudioStream", "Disabled (Stereo)"),
TRANSLATE_NOOP("AudioStream", "Stereo with LFE"),
TRANSLATE_NOOP("AudioStream", "Quadraphonic"),
TRANSLATE_NOOP("AudioStream", "Quadraphonic with LFE"),
TRANSLATE_NOOP("AudioStream", "5.1 Surround"),
TRANSLATE_NOOP("AudioStream", "7.1 Surround"),
};
const char* AudioStream::GetExpansionModeName(AudioExpansionMode mode)
{
return (static_cast<u32>(mode) < s_expansion_mode_names.size()) ? s_expansion_mode_names[static_cast<u32>(mode)] : "";
}
const char* AudioStream::GetExpansionModeDisplayName(AudioExpansionMode mode)
{
return (static_cast<u32>(mode) < s_expansion_mode_display_names.size()) ?
Host::TranslateToCString("AudioStream", s_expansion_mode_display_names[static_cast<u32>(mode)]) :
"";
}
std::optional<AudioExpansionMode> AudioStream::ParseExpansionMode(const char* name)
{
for (u8 i = 0; i < static_cast<u8>(AudioExpansionMode::Count); i++)
{
if (std::strcmp(name, s_expansion_mode_names[i]) == 0)
return static_cast<AudioExpansionMode>(i);
}
return std::nullopt;
}
u32 AudioStream::GetBufferedFramesRelaxed() const
{
const u32 rpos = m_rpos.load(std::memory_order_relaxed);
const u32 wpos = m_wpos.load(std::memory_order_relaxed);
return (wpos + m_buffer_size - rpos) % m_buffer_size;
}
void AudioStream::ReadFrames(SampleType* samples, u32 num_frames)
{
const u32 available_frames = GetBufferedFramesRelaxed();
u32 frames_to_read = num_frames;
u32 silence_frames = 0;
if (m_filling)
{
u32 toFill = m_buffer_size / (IsStretchEnabled() ? 32 : 400);
toFill = GetAlignedBufferSize(toFill);
if (available_frames < toFill)
{
silence_frames = num_frames;
frames_to_read = 0;
}
else
{
m_filling = false;
LOG_UNDERRUN("Underrun compensation done ({} frames buffered)", toFill);
}
}
if (available_frames < frames_to_read)
{
silence_frames = frames_to_read - available_frames;
frames_to_read = available_frames;
m_filling = true;
if (IsStretchEnabled())
StretchUnderrun();
}
if (frames_to_read > 0)
{
u32 rpos = m_rpos.load(std::memory_order_acquire);
u32 end = m_buffer_size - rpos;
if (end > frames_to_read)
end = frames_to_read;
// towards the end of the buffer
if (end > 0)
{
m_sample_reader(samples, &m_buffer[rpos * m_internal_channels], end);
rpos += end;
rpos = (rpos == m_buffer_size) ? 0 : rpos;
}
// after wrapping around
const u32 start = frames_to_read - end;
if (start > 0)
{
m_sample_reader(&samples[end * m_output_channels], &m_buffer[0], start);
rpos = start;
}
m_rpos.store(rpos, std::memory_order_release);
}
if (silence_frames > 0)
{
if (frames_to_read > 0)
{
// 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>(frames_to_read) / static_cast<float>(num_frames)));
SampleType* resample_ptr =
static_cast<SampleType*>(alloca(frames_to_read * m_output_channels * sizeof(SampleType)));
std::memcpy(resample_ptr, samples, frames_to_read * m_output_channels * sizeof(SampleType));
SampleType* out_ptr = samples;
const u32 copy_stride = sizeof(SampleType) * m_output_channels;
u32 resample_subpos = 0;
for (u32 i = 0; i < num_frames; i++)
{
std::memcpy(out_ptr, resample_ptr, copy_stride);
out_ptr += m_output_channels;
resample_subpos += increment;
resample_ptr += (resample_subpos >> 16) * m_output_channels;
resample_subpos %= 65536u;
}
LOG_UNDERRUN("Audio buffer underflow, resampled {} frames to {}", frames_to_read, num_frames);
}
else
{
// no data, fall back to silence
std::memset(samples + (frames_to_read * m_output_channels), 0, silence_frames * m_output_channels * sizeof(s16));
}
}
if (m_volume != 100)
{
u32 num_samples = num_frames * m_output_channels;
const u32 aligned_samples = Common::AlignDownPow2(num_samples, 8);
num_samples -= aligned_samples;
const float volume_mult = static_cast<float>(m_volume) / 100.0f;
const GSVector4 volume_multv = GSVector4(volume_mult);
const SampleType* const aligned_samples_end = samples + aligned_samples;
for (; samples != aligned_samples_end; samples += 8)
{
GSVector4i iv = GSVector4i::load<false>(samples); // [0, 1, 2, 3, 4, 5, 6, 7]
GSVector4i iv1 = iv.upl16(iv); // [0, 0, 1, 1, 2, 2, 3, 3]
GSVector4i iv2 = iv.uph16(iv); // [4, 4, 5, 5, 6, 6, 7, 7]
iv1 = iv1.sra32<16>(); // [0, 1, 2, 3]
iv2 = iv2.sra32<16>(); // [4, 5, 6, 7]
GSVector4 fv1 = GSVector4(iv1); // [f0, f1, f2, f3]
GSVector4 fv2 = GSVector4(iv2); // [f4, f5, f6, f7]
fv1 = fv1 * volume_multv; // [f0, f1, f2, f3]
fv2 = fv2 * volume_multv; // [f4, f5, f6, f7]
iv1 = GSVector4i(fv1); // [0, 1, 2, 3]
iv2 = GSVector4i(fv2); // [4, 5, 6, 7]
iv = iv1.ps32(iv2); // [0, 1, 2, 3, 4, 5, 6, 7]
GSVector4i::store<false>(samples, iv);
}
while (num_samples > 0)
{
*samples = static_cast<s16>(std::clamp(static_cast<float>(*samples) * volume_mult, -32768.0f, 32767.0f));
samples++;
num_samples--;
}
}
}
void AudioStream::StereoSampleReaderImpl(SampleType* dest, const SampleType* src, u32 num_frames)
{
std::memcpy(dest, src, num_frames * 2 * sizeof(SampleType));
}
void AudioStream::InternalWriteFrames(const SampleType* data, u32 num_frames)
{
const u32 free = m_buffer_size - GetBufferedFramesRelaxed();
if (free <= num_frames)
{
if (IsStretchEnabled())
{
StretchOverrun();
}
else
{
LOG_UNDERRUN("Buffer overrun, chunk dropped");
return;
}
}
u32 wpos = m_wpos.load(std::memory_order_acquire);
// wrapping around the end of the buffer?
if ((m_buffer_size - wpos) <= num_frames)
{
// needs to be written in two parts
const u32 end = m_buffer_size - wpos;
const u32 start = num_frames - end;
// start is zero when this chunk reaches exactly the end
std::memcpy(&m_buffer[wpos * m_internal_channels], data, end * m_internal_channels * sizeof(SampleType));
if (start > 0)
std::memcpy(&m_buffer[0], data + end * m_internal_channels, start * m_internal_channels * sizeof(SampleType));
wpos = start;
}
else
{
// no split
std::memcpy(&m_buffer[wpos * m_internal_channels], data, num_frames * m_internal_channels * sizeof(SampleType));
wpos += num_frames;
}
m_wpos.store(wpos, std::memory_order_release);
}
void AudioStream::BaseInitialize(SampleReader sample_reader, bool stretch_enabled)
{
m_stretch_enabled = stretch_enabled;
m_paused = false;
m_sample_reader = sample_reader;
AllocateBuffer();
ExpandAllocate();
StretchAllocate();
}
void AudioStream::AllocateBuffer()
{
// use a larger buffer when time stretching, since we need more input
const u32 multiplier = IsStretchEnabled() ? 16 : 1;
m_buffer_size = GetAlignedBufferSize(((m_parameters.buffer_ms * multiplier) * m_sample_rate) / 1000);
m_target_buffer_size = GetAlignedBufferSize((m_sample_rate * m_parameters.buffer_ms) / 1000u);
m_buffer = std::make_unique<s16[]>(m_buffer_size * m_internal_channels);
m_staging_buffer = std::make_unique<s16[]>(CHUNK_SIZE * m_internal_channels);
m_float_buffer = std::make_unique<float[]>(CHUNK_SIZE * m_internal_channels);
if (IsExpansionEnabled())
m_expand_buffer = std::make_unique<float[]>(m_parameters.expand_block_size * NUM_INPUT_CHANNELS);
DEV_LOG(
"Allocated buffer of {} frames for buffer of {} ms [expansion {} (block size {}), stretch {}, target size {}].",
m_buffer_size, m_parameters.buffer_ms, GetExpansionModeName(m_parameters.expansion_mode),
m_parameters.expand_block_size, m_stretch_enabled ? "enabled" : "disabled", m_target_buffer_size);
}
void AudioStream::DestroyBuffer()
{
m_expand_buffer.reset();
m_staging_buffer.reset();
m_float_buffer.reset();
m_buffer.reset();
m_buffer_size = 0;
m_wpos.store(0, std::memory_order_release);
m_rpos.store(0, std::memory_order_release);
}
void AudioStream::EmptyBuffer()
{
if (IsExpansionEnabled())
{
m_expander->Flush();
m_expand_output_buffer = nullptr;
m_expand_buffer_pos = 0;
}
if (IsStretchEnabled())
{
m_soundtouch->clear();
if (IsStretchEnabled())
m_soundtouch->setTempo(m_nominal_rate);
}
m_wpos.store(m_rpos.load(std::memory_order_acquire), std::memory_order_release);
}
void AudioStream::SetNominalRate(float tempo)
{
m_nominal_rate = tempo;
}
void AudioStream::UpdateTargetTempo(float tempo)
{
if (!IsStretchEnabled())
return;
// undo sqrt()
if (tempo)
tempo *= tempo;
m_average_position = AVERAGING_WINDOW;
m_average_available = AVERAGING_WINDOW;
std::fill_n(m_average_fullness.data(), AVERAGING_WINDOW, tempo);
m_soundtouch->setTempo(tempo);
m_stretch_reset = 0;
m_stretch_inactive = false;
m_stretch_ok_count = 0;
m_dynamic_target_usage = static_cast<float>(m_target_buffer_size) * m_nominal_rate;
}
void AudioStream::SetStretchEnabled(bool enabled)
{
if (m_stretch_enabled == enabled)
return;
// can't resize the buffers while paused
const bool paused = m_paused;
if (!paused)
SetPaused(true);
DestroyBuffer();
StretchDestroy();
m_stretch_enabled = enabled;
AllocateBuffer();
StretchAllocate();
if (!paused)
SetPaused(false);
}
void AudioStream::SetPaused(bool paused)
{
m_paused = paused;
}
void AudioStream::SetOutputVolume(u32 volume)
{
m_volume = volume;
}
void AudioStream::BeginWrite(SampleType** buffer_ptr, u32* num_frames)
{
*buffer_ptr = &m_staging_buffer[m_staging_buffer_pos];
*num_frames = CHUNK_SIZE - (m_staging_buffer_pos / NUM_INPUT_CHANNELS);
}
void AudioStream::WriteFrame(const SampleType* frame)
{
pxAssert((CHUNK_SIZE - (m_staging_buffer_pos / NUM_INPUT_CHANNELS)) > 0);
std::memcpy(&m_staging_buffer[m_staging_buffer_pos], frame, sizeof(SampleType) * NUM_INPUT_CHANNELS);
EndWrite(1);
}
static void S16ChunkToFloat(const s16* src, float* dst, u32 num_samples)
{
constexpr GSVector4 S16_TO_FLOAT_V = GSVector4::cxpr(1.0f / 32767.0f);
const u32 iterations = (num_samples + 7) / 8;
for (u32 i = 0; i < iterations; i++)
{
const GSVector4i sv = GSVector4i::load<false>(src);
src += 8;
GSVector4i iv1 = sv.upl16(sv); // [0, 0, 1, 1, 2, 2, 3, 3]
GSVector4i iv2 = sv.uph16(sv); // [4, 4, 5, 5, 6, 6, 7, 7]
iv1 = iv1.sra32<16>(); // [0, 1, 2, 3]
iv2 = iv2.sra32<16>(); // [4, 5, 6, 7]
GSVector4 fv1 = GSVector4(iv1); // [f0, f1, f2, f3]
GSVector4 fv2 = GSVector4(iv2); // [f4, f5, f6, f7]
fv1 = fv1 * S16_TO_FLOAT_V;
fv2 = fv2 * S16_TO_FLOAT_V;
GSVector4::store<false>(dst + 0, fv1);
GSVector4::store<false>(dst + 4, fv2);
dst += 8;
}
}
static void FloatChunkToS16(s16* dst, const float* src, u32 num_samples)
{
const GSVector4 FLOAT_TO_S16_V = GSVector4::cxpr(32767.0f);
const u32 iterations = (num_samples + 7) / 8;
for (u32 i = 0; i < iterations; i++)
{
GSVector4 fv1 = GSVector4::load<false>(src + 0);
GSVector4 fv2 = GSVector4::load<false>(src + 4);
src += 8;
fv1 = fv1 * FLOAT_TO_S16_V;
fv2 = fv2 * FLOAT_TO_S16_V;
GSVector4i iv1 = GSVector4i(fv1);
GSVector4i iv2 = GSVector4i(fv2);
const GSVector4i iv = iv1.ps32(iv2);
GSVector4i::store<false>(dst, iv);
dst += 8;
}
}
void AudioStream::ExpandAllocate()
{
pxAssert(!m_expander);
if (m_parameters.expansion_mode == AudioExpansionMode::Disabled)
return;
static constexpr std::array<std::pair<FreeSurroundDecoder::ChannelSetup, bool>,
static_cast<size_t>(AudioExpansionMode::Count)>
channel_setup_mapping = {{
{FreeSurroundDecoder::ChannelSetup::Stereo, false}, // Disabled
{FreeSurroundDecoder::ChannelSetup::Stereo, true}, // StereoLFE
{FreeSurroundDecoder::ChannelSetup::Surround41, false}, // Quadraphonic
{FreeSurroundDecoder::ChannelSetup::Surround41, true}, // QuadraphonicLFE
{FreeSurroundDecoder::ChannelSetup::Surround51, true}, // Surround51
{FreeSurroundDecoder::ChannelSetup::Surround71, true}, // Surround71
}};
const auto [fs_setup, fs_lfe] = channel_setup_mapping[static_cast<size_t>(m_parameters.expansion_mode)];
m_expander = std::make_unique<FreeSurroundDecoder>(fs_setup, m_parameters.expand_block_size);
m_expander->SetBassRedirection(fs_lfe);
m_expander->SetCircularWrap(m_parameters.expand_circular_wrap);
m_expander->SetShift(m_parameters.expand_shift);
m_expander->SetDepth(m_parameters.expand_depth);
m_expander->SetFocus(m_parameters.expand_focus);
m_expander->SetCenterImage(m_parameters.expand_center_image);
m_expander->SetFrontSeparation(m_parameters.expand_front_separation);
m_expander->SetRearSeparation(m_parameters.expand_rear_separation);
m_expander->SetLowCutoff(static_cast<float>(m_parameters.expand_low_cutoff) / m_sample_rate * 2);
m_expander->SetHighCutoff(static_cast<float>(m_parameters.expand_high_cutoff) / m_sample_rate * 2);
}
void AudioStream::EndWrite(u32 num_frames)
{
// don't bother committing anything when muted
if (m_volume == 0)
return;
m_staging_buffer_pos += num_frames * NUM_INPUT_CHANNELS;
pxAssert(m_staging_buffer_pos <= (CHUNK_SIZE * NUM_INPUT_CHANNELS));
if ((m_staging_buffer_pos / NUM_INPUT_CHANNELS) < CHUNK_SIZE)
return;
m_staging_buffer_pos = 0;
WriteChunk(m_staging_buffer.get());
}
void AudioStream::WriteChunk(const SampleType* chunk)
{
if (!IsExpansionEnabled() && !IsStretchEnabled())
{
InternalWriteFrames(chunk, CHUNK_SIZE);
return;
}
if (IsExpansionEnabled())
{
// StretchWriteBlock() overwrites the staging buffer on output, so we need to copy into the expand buffer first.
S16ChunkToFloat(chunk, m_expand_buffer.get() + m_expand_buffer_pos * NUM_INPUT_CHANNELS, CHUNK_SIZE * NUM_INPUT_CHANNELS);
// Output the corresponding block.
if (m_expand_output_buffer)
StretchWriteBlock(m_expand_output_buffer + m_expand_buffer_pos * m_internal_channels);
// Decode the next block if we buffered enough.
m_expand_buffer_pos += CHUNK_SIZE;
if (m_expand_buffer_pos == m_parameters.expand_block_size)
{
m_expand_buffer_pos = 0;
m_expand_output_buffer = m_expander->Decode(m_expand_buffer.get());
}
}
else
{
S16ChunkToFloat(chunk, m_float_buffer.get(), CHUNK_SIZE * NUM_INPUT_CHANNELS);
StretchWriteBlock(m_float_buffer.get());
}
}
template <class T>
__fi static bool IsInRange(const T& val, const T& min, const T& max)
{
return (min <= val && val <= max);
}
void AudioStream::StretchAllocate()
{
if (!IsStretchEnabled())
return;
m_soundtouch = std::make_unique<soundtouch::SoundTouch>();
m_soundtouch->setSampleRate(m_sample_rate);
m_soundtouch->setChannels(m_internal_channels);
m_soundtouch->setSetting(SETTING_USE_QUICKSEEK, m_parameters.stretch_use_quickseek);
m_soundtouch->setSetting(SETTING_USE_AA_FILTER, m_parameters.stretch_use_aa_filter);
m_soundtouch->setSetting(SETTING_SEQUENCE_MS, m_parameters.stretch_sequence_length_ms);
m_soundtouch->setSetting(SETTING_SEEKWINDOW_MS, m_parameters.stretch_seekwindow_ms);
m_soundtouch->setSetting(SETTING_OVERLAP_MS, m_parameters.stretch_overlap_ms);
m_soundtouch->setTempo(m_nominal_rate);
m_stretch_reset = STRETCH_RESET_THRESHOLD;
m_stretch_inactive = false;
m_stretch_ok_count = 0;
m_dynamic_target_usage = 0.0f;
m_average_position = 0;
m_average_available = 0;
m_staging_buffer_pos = 0;
}
void AudioStream::StretchDestroy()
{
m_soundtouch.reset();
}
void AudioStream::StretchWriteBlock(const float* block)
{
if (IsStretchEnabled())
{
m_soundtouch->putSamples(block, CHUNK_SIZE);
u32 tempProgress;
while (tempProgress = m_soundtouch->receiveSamples(m_float_buffer.get(), CHUNK_SIZE), tempProgress != 0)
{
FloatChunkToS16(m_staging_buffer.get(), m_float_buffer.get(), tempProgress * m_internal_channels);
InternalWriteFrames(m_staging_buffer.get(), tempProgress);
}
if (IsStretchEnabled())
UpdateStretchTempo();
}
else
{
FloatChunkToS16(m_staging_buffer.get(), block, CHUNK_SIZE * m_internal_channels);
InternalWriteFrames(m_staging_buffer.get(), CHUNK_SIZE);
}
}
float AudioStream::AddAndGetAverageTempo(float val)
{
if (m_stretch_reset >= STRETCH_RESET_THRESHOLD)
m_average_available = 0;
if (m_average_available < AVERAGING_BUFFER_SIZE)
m_average_available++;
m_average_fullness[m_average_position] = val;
m_average_position = (m_average_position + 1U) % AVERAGING_BUFFER_SIZE;
const u32 actual_window = std::min<u32>(m_average_available, AVERAGING_WINDOW);
const u32 first_index = (m_average_position - actual_window + AVERAGING_BUFFER_SIZE) % AVERAGING_BUFFER_SIZE;
float sum = 0;
for (u32 i = first_index; i < first_index + actual_window; i++)
sum += m_average_fullness[i % AVERAGING_BUFFER_SIZE];
sum = sum / actual_window;
return (sum != 0.0f) ? sum : 1.0f;
}
void AudioStream::UpdateStretchTempo()
{
static constexpr float MIN_TEMPO = 0.05f;
static constexpr float MAX_TEMPO = 50.0f;
// Which range we will run in 1:1 mode for.
static constexpr float INACTIVE_GOOD_FACTOR = 1.04f;
static constexpr float INACTIVE_BAD_FACTOR = 1.2f;
static constexpr u32 INACTIVE_MIN_OK_COUNT = 50;
static constexpr u32 COMPENSATION_DIVIDER = 100;
float base_target_usage = static_cast<float>(m_target_buffer_size) * m_nominal_rate;
// state vars
if (m_stretch_reset >= STRETCH_RESET_THRESHOLD)
{
LOG_UNDERRUN("___ Stretcher is being reset.");
m_stretch_inactive = false;
m_stretch_ok_count = 0;
m_dynamic_target_usage = base_target_usage;
}
const u32 ibuffer_usage = GetBufferedFramesRelaxed();
float buffer_usage = static_cast<float>(ibuffer_usage);
float tempo = buffer_usage / m_dynamic_target_usage;
tempo = AddAndGetAverageTempo(tempo);
// Dampening when we get close to target.
if (tempo < 2.0f)
tempo = std::sqrt(tempo);
tempo = std::clamp(tempo, MIN_TEMPO, MAX_TEMPO);
if (tempo < 1.0f)
base_target_usage /= std::sqrt(tempo);
m_dynamic_target_usage +=
static_cast<float>(base_target_usage / tempo - m_dynamic_target_usage) / static_cast<float>(COMPENSATION_DIVIDER);
if (IsInRange(tempo, 0.9f, 1.1f) &&
IsInRange(m_dynamic_target_usage, base_target_usage * 0.9f, base_target_usage * 1.1f))
{
m_dynamic_target_usage = base_target_usage;
}
if (!m_stretch_inactive)
{
if (IsInRange(tempo, 1.0f / INACTIVE_GOOD_FACTOR, INACTIVE_GOOD_FACTOR))
m_stretch_ok_count++;
else
m_stretch_ok_count = 0;
if (m_stretch_ok_count >= INACTIVE_MIN_OK_COUNT)
{
LOG_UNDERRUN("=== Stretcher is now inactive.");
m_stretch_inactive = true;
}
}
else if (!IsInRange(tempo, 1.0f / INACTIVE_BAD_FACTOR, INACTIVE_BAD_FACTOR))
{
LOG_UNDERRUN("~~~ Stretcher is now active @ tempo {}.", tempo);
m_stretch_inactive = false;
m_stretch_ok_count = 0;
}
if (m_stretch_inactive)
tempo = m_nominal_rate;
if constexpr (LOG_TIMESTRETCH_STATS)
{
static int iterations = 0;
static u64 last_log_time = 0;
const u64 now = Common::Timer::GetCurrentValue();
if (Common::Timer::ConvertValueToSeconds(now - last_log_time) > 1.0f)
{
DEV_LOG("buffers: {:4} ms ({:3.0f}%), tempo: {}, comp: {:2.3f}, iters: {}, reset:{}",
(ibuffer_usage * 1000u) / m_sample_rate, 100.0f * buffer_usage / base_target_usage, tempo,
m_dynamic_target_usage / base_target_usage, iterations, m_stretch_reset);
last_log_time = now;
iterations = 0;
}
iterations++;
}
m_soundtouch->setTempo(tempo);
if (m_stretch_reset >= STRETCH_RESET_THRESHOLD)
m_stretch_reset = 0;
}
void AudioStream::StretchUnderrun()
{
// Didn't produce enough frames in time.
m_stretch_reset++;
}
void AudioStream::StretchOverrun()
{
// Produced more frames than can fit in the buffer.
m_stretch_reset++;
// Drop two packets to give the time stretcher a bit more time to slow things down.
const u32 discard = CHUNK_SIZE * 2;
m_rpos.store((m_rpos.load(std::memory_order_acquire) + discard) % m_buffer_size, std::memory_order_release);
}
void AudioStreamParameters::LoadSave(SettingsWrapper& wrap, const char* section)
{
wrap.EnumEntry(section, "ExpansionMode", expansion_mode, &AudioStream::ParseExpansionMode, &AudioStream::GetExpansionModeName, DEFAULT_EXPANSION_MODE);
minimal_output_latency = wrap.EntryBitBool(section, "OutputLatencyMinimal", DEFAULT_OUTPUT_LATENCY_MINIMAL);
buffer_ms = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "BufferMS", buffer_ms, DEFAULT_BUFFER_MS), 0, std::numeric_limits<u16>::max()));
output_latency_ms = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "OutputLatencyMS", output_latency_ms, DEFAULT_OUTPUT_LATENCY_MS), 0, std::numeric_limits<u16>::max()));
stretch_sequence_length_ms = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "StretchSequenceLengthMS", DEFAULT_STRETCH_SEQUENCE_LENGTH), 0, std::numeric_limits<u16>::max()));
stretch_seekwindow_ms = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "StretchSeekWindowMS", DEFAULT_STRETCH_SEEKWINDOW), 0, std::numeric_limits<u16>::max()));
stretch_overlap_ms = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "StretchOverlapMS", DEFAULT_STRETCH_OVERLAP), 0, std::numeric_limits<u16>::max()));
stretch_use_quickseek = wrap.EntryBitBool(section, "StretchUseQuickSeek", DEFAULT_STRETCH_USE_QUICKSEEK);
stretch_use_aa_filter = wrap.EntryBitBool(section, "StretchUseAAFilter", DEFAULT_STRETCH_USE_AA_FILTER);
expand_block_size = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "ExpandBlockSize", DEFAULT_EXPAND_BLOCK_SIZE), 0, std::numeric_limits<u16>::max()));
wrap.Entry(section, "ExpandCircularWrap", expand_circular_wrap, DEFAULT_EXPAND_CIRCULAR_WRAP);
wrap.Entry(section, "ExpandShift", expand_shift, DEFAULT_EXPAND_SHIFT);
wrap.Entry(section, "ExpandDepth", expand_depth, DEFAULT_EXPAND_DEPTH);
wrap.Entry(section, "ExpandFocus", expand_focus, DEFAULT_EXPAND_FOCUS);
wrap.Entry(section, "ExpandCenterImage", expand_center_image, DEFAULT_EXPAND_CENTER_IMAGE);
wrap.Entry(section, "ExpandFrontSeparation", expand_front_separation, DEFAULT_EXPAND_FRONT_SEPARATION);
wrap.Entry(section, "ExpandRearSeparation", expand_rear_separation, DEFAULT_EXPAND_REAR_SEPARATION);
expand_low_cutoff = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "ExpandLowCutoff", DEFAULT_EXPAND_LOW_CUTOFF), 0, std::numeric_limits<u8>::max()));
expand_high_cutoff = static_cast<u16>(std::clamp<int>(wrap.EntryBitfield(section, "ExpandHighCutoff", DEFAULT_EXPAND_HIGH_CUTOFF), 0, std::numeric_limits<u8>::max()));
// Clamping of values.
if (wrap.IsLoading())
{
stretch_sequence_length_ms = std::clamp<u16>(stretch_sequence_length_ms, 20, 100);
stretch_seekwindow_ms = std::clamp<u16>(stretch_seekwindow_ms, 10, 30);
stretch_overlap_ms = std::clamp<u16>(stretch_overlap_ms, 5, 15);
expand_block_size = std::clamp<u16>(std::has_single_bit(expand_block_size) ? expand_block_size : std::bit_ceil(expand_block_size), 128, 8192);
expand_circular_wrap = std::clamp(expand_circular_wrap, 0.0f, 360.0f);
expand_shift = std::clamp(expand_shift, -1.0f, 1.0f);
expand_depth = std::clamp(expand_depth, 0.0f, 5.0f);
expand_focus = std::clamp(expand_focus, -1.0f, 1.0f);
expand_center_image = std::clamp(expand_center_image, 0.0f, 1.0f);
expand_front_separation = std::clamp(expand_front_separation, 0.0f, 10.0f);
expand_rear_separation = std::clamp(expand_rear_separation, 0.0f, 10.0f);
expand_low_cutoff = std::min<u8>(expand_low_cutoff, 100);
expand_high_cutoff = std::min<u8>(expand_high_cutoff, 100);
}
}
bool AudioStreamParameters::operator!=(const AudioStreamParameters& rhs) const
{
return (std::memcmp(this, &rhs, sizeof(*this)) != 0);
}
bool AudioStreamParameters::operator==(const AudioStreamParameters& rhs) const
{
return (std::memcmp(this, &rhs, sizeof(*this)) == 0);
}