mirror of
https://github.com/dolphin-emu/dolphin
synced 2024-07-03 08:48:48 +00:00
Remove High Frequency Distortion and Prevent Crackling in Almost All Cases
This commit is contained in:
Sam Belliveau
parent
2aec195ec8
commit
78f0083032
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@ -12,7 +12,7 @@
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namespace AudioCommon
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{
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AudioStretcher::AudioStretcher(unsigned int sample_rate) : m_sample_rate(sample_rate)
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AudioStretcher::AudioStretcher(u64 sample_rate) : m_sample_rate(sample_rate)
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{
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m_sound_touch.setChannels(2);
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m_sound_touch.setSampleRate(sample_rate);
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@ -25,46 +25,45 @@ void AudioStretcher::Clear()
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m_sound_touch.clear();
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}
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void AudioStretcher::ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out)
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void AudioStretcher::ProcessSamples(const s16* in, u32 num_in, u32 num_out)
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{
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const double time_delta = static_cast<double>(num_out) / m_sample_rate; // seconds
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const double time_delta = static_cast<double>(num_out) / m_sample_rate;
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double current_ratio = static_cast<double>(num_in) / num_out; // Current sample ratio.
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// We were given actual_samples number of samples, and num_samples were requested from us.
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double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
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// Calculate maximum allowable backlog based on configured maximum latency.
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const double max_latency = Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY) / 1000.0;
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const double max_backlog = m_sample_rate * max_latency; // Max number of samples in the backlog.
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const double num_samples = m_sound_touch.numSamples();
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const double max_latency = Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY);
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const double max_backlog = m_sample_rate * max_latency / 1000.0 / m_stretch_ratio;
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const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
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if (backlog_fullness > 5.0)
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{
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// Too many samples in backlog: Don't push anymore on
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// Prevent backlog from growing too large.
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if (num_samples >= 5.0 * max_backlog)
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num_in = 0;
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}
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// We ideally want the backlog to be about 50% full.
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// This gives some headroom both ways to prevent underflow and overflow.
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// We tweak current_ratio to encourage this.
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constexpr double tweak_time_scale = 0.5; // seconds
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current_ratio *= 1.0 + 2.0 * (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
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// Target for backlog to be about 50% full to allow flexibility.
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const double low_watermark = 0.5 * max_backlog;
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const double requested_samples = m_stretch_ratio * num_out;
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const double num_left = num_samples - requested_samples;
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// This low-pass filter smoothes out variance in the calculated stretch ratio.
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// The time-scale determines how responsive this filter is.
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constexpr double lpf_time_scale = 1.0; // seconds
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const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
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// Adjustment parameters similar to those used in Mixer for pitch adjustment.
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const double lpf_time_scale = 1.0 / 16.0;
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const double control_effort = 1.0 / 16.0;
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current_ratio *= 1.0 + control_effort * (num_left - low_watermark) / requested_samples;
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current_ratio = std::clamp(current_ratio, 0.1, 10.0);
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// Calculate target sample ratio and apply low-pass filter to smooth changes.
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const double lpf_gain = -std::expm1(-time_delta / lpf_time_scale);
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m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
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// Place a lower limit of 10% speed. When a game boots up, there will be
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// many silence samples. These do not need to be timestretched.
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m_stretch_ratio = std::max(m_stretch_ratio, 0.1);
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m_sound_touch.setTempo(m_stretch_ratio);
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DEBUG_LOG_FMT(AUDIO, "Audio stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in,
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num_out, m_stretch_ratio, backlog_fullness, lpf_gain);
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// Debug log to monitor stretching stats.
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DEBUG_LOG_FMT(AUDIO, "Audio stretching: samples:{}/{} ratio:{} backlog:{} gain:{}", num_in,
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num_out, m_stretch_ratio, num_samples / max_backlog, lpf_gain);
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// Add new samples to the processor for stretching.
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m_sound_touch.putSamples(in, num_in);
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}
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void AudioStretcher::GetStretchedSamples(short* out, unsigned int num_out)
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void AudioStretcher::GetStretchedSamples(s16* out, u32 num_out)
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{
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const size_t samples_received = m_sound_touch.receiveSamples(out, num_out);
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@ -82,4 +81,10 @@ void AudioStretcher::GetStretchedSamples(short* out, unsigned int num_out)
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}
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}
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DT AudioStretcher::AvailableSamplesTime() const
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{
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const u32 backlog = m_sound_touch.numSamples();
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return std::chrono::duration_cast<DT>(DT_s(backlog) / m_sample_rate);
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}
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} // namespace AudioCommon
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@ -7,19 +7,23 @@
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#include <SoundTouch.h>
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#include "Common/CommonTypes.h"
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namespace AudioCommon
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{
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class AudioStretcher
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{
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public:
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explicit AudioStretcher(unsigned int sample_rate);
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void ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out);
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void GetStretchedSamples(short* out, unsigned int num_out);
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explicit AudioStretcher(u64 sample_rate);
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void ProcessSamples(const s16* in, u32 num_in, u32 num_out);
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void GetStretchedSamples(s16* out, u32 num_out);
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void Clear();
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DT AvailableSamplesTime() const;
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private:
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unsigned int m_sample_rate;
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std::array<short, 2> m_last_stretched_sample = {};
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u64 m_sample_rate;
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std::array<s16, 2> m_last_stretched_sample = {};
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soundtouch::SoundTouch m_sound_touch;
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double m_stretch_ratio = 1.0;
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};
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@ -7,6 +7,8 @@
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#include <cmath>
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#include <cstring>
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#include <Core/Core.h>
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#include "AudioCommon/Enums.h"
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#include "Common/ChunkFile.h"
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#include "Common/CommonTypes.h"
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@ -31,7 +33,7 @@ static u32 DPL2QualityToFrameBlockSize(AudioCommon::DPL2Quality quality)
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}
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}
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Mixer::Mixer(unsigned int BackendSampleRate)
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Mixer::Mixer(u32 BackendSampleRate)
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: m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate),
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m_surround_decoder(BackendSampleRate,
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DPL2QualityToFrameBlockSize(Config::Get(Config::MAIN_DPL2_QUALITY)))
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@ -57,119 +59,164 @@ void Mixer::DoState(PointerWrap& p)
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mixer.DoState(p);
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}
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// Executed from sound stream thread
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unsigned int Mixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
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bool consider_framelimit, float emulationspeed,
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int timing_variance)
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std::pair<float, float> Mixer::MixerFifo::GetSample(u32 index, float frac, float sinc_ratio)
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{
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unsigned int currentSample = 0;
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constexpr float pi = 3.14159265358979323846f;
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const s32 sinc_window_width = m_mixer->m_config_sinc_window_width;
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// Cache access in non-volatile variable
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// This is the only function changing the read value, so it's safe to
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// cache it locally although it's written here.
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// The writing pointer will be modified outside, but it will only increase,
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// so we will just ignore new written data while interpolating.
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// Without this cache, the compiler wouldn't be allowed to optimize the
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// interpolation loop.
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u32 indexR = m_indexR.load();
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u32 indexW = m_indexW.load();
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// render numleft sample pairs to samples[]
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// advance indexR with sample position
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// remember fractional offset
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float aid_sample_rate =
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FIXED_SAMPLE_RATE_DIVIDEND / static_cast<float>(m_input_sample_rate_divisor);
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if (consider_framelimit && emulationspeed > 0.0f)
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float l_sample = 0.0f, r_sample = 0.0f;
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for (s32 i = 1 - sinc_window_width; i <= sinc_window_width; ++i)
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{
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float numLeft = static_cast<float>(((indexW - indexR) & INDEX_MASK) / 2);
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const auto [l, r] = m_buffer[(index + i) & INDEX_MASK];
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u32 low_watermark = (FIXED_SAMPLE_RATE_DIVIDEND * timing_variance) /
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(static_cast<u64>(m_input_sample_rate_divisor) * 1000);
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low_watermark = std::min(low_watermark, MAX_SAMPLES / 2);
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const float x = pi * (i - frac);
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float weight = 0.53836f + 0.46164f * std::cos(x / sinc_window_width);
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if (std::abs(x) >= 1e-6f)
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weight *= std::sin(x * sinc_ratio) / x;
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else
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weight *= sinc_ratio;
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m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
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float offset = (m_numLeftI - low_watermark) * CONTROL_FACTOR;
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if (offset > MAX_FREQ_SHIFT)
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offset = MAX_FREQ_SHIFT;
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if (offset < -MAX_FREQ_SHIFT)
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offset = -MAX_FREQ_SHIFT;
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aid_sample_rate = (aid_sample_rate + offset) * emulationspeed;
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r_sample += weight * r;
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l_sample += weight * l;
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}
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const u32 ratio = (u32)(65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate);
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return std::make_pair(l_sample, r_sample);
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}
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s32 lvolume = m_LVolume.load();
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s32 rvolume = m_RVolume.load();
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// Executed from sound stream thread
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void Mixer::MixerFifo::Mix(s16* samples, u32 num_samples, double target_speed)
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{
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// Cache access in non-volatile variable. No race conditions are here, as indexW only increases
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// (which is fine), and this is the only place indexR is written to.
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const u32 indexW = m_indexW.load(); // Write index in circular buffer
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u32 indexR = m_indexR.load(); // Read index in circular buffer
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const auto read_buffer = [this](auto index) {
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return m_little_endian ? m_buffer[index] : Common::swap16(m_buffer[index]);
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};
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const float l_volume = float(m_LVolume.load()) / 256.0f;
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const float r_volume = float(m_RVolume.load()) / 256.0f;
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// TODO: consider a higher-quality resampling algorithm.
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for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2)
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auto clamp = [](s32 input) -> s16 { return static_cast<s16>(std::clamp(input, -32768, 32767)); };
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const double ratio = (target_speed * FIXED_SAMPLE_RATE_DIVIDEND) /
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(m_input_sample_rate_divisor * m_mixer->m_sampleRate);
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// These are timing markers. We want to keep the audio buffer around the low watermark.
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const s32 direct_latency = m_mixer->m_config_direct_latency;
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const s32 low_watermark = std::min<s32>(MAX_SAMPLES, direct_latency * FIXED_SAMPLE_RATE_DIVIDEND /
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(m_input_sample_rate_divisor * 1000));
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// Calculate the number of samples remaining in the buffer
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const s32 sinc_window_width = m_mixer->m_config_sinc_window_width;
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s32 remaining_samples = s32((indexW - indexR) & INDEX_MASK) - sinc_window_width - 2;
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// Calculate the speed at which to play the requested samples in order to maintain the correct
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// synchronization with the video stream. Done by slightly adjusting the pitch of the audio.
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const double requested_samples = m_multiplier * num_samples * ratio;
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double multiplier = (remaining_samples - low_watermark) / requested_samples;
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multiplier = 1.0 + CONTROL_EFFORT * (multiplier - 1.0);
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multiplier = std::clamp(multiplier, MIN_PITCH_SHIFT, MAX_PITCH_SHIFT);
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// If the number of samples is above 2x the low watermark, skip ahead to prevent excessive latency
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const double max_latency = 2.0 * low_watermark;
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if (remaining_samples - requested_samples >= max_latency)
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{
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u32 indexR2 = indexR + 2; // next sample
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const s32 jump = 1 + static_cast<s32>(remaining_samples - requested_samples - max_latency);
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remaining_samples -= jump;
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indexR += jump;
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}
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s16 l1 = read_buffer(indexR & INDEX_MASK); // current
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s16 l2 = read_buffer(indexR2 & INDEX_MASK); // next
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int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
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sampleL = (sampleL * lvolume) >> 8;
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sampleL += samples[currentSample + 1];
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samples[currentSample + 1] = std::clamp(sampleL, -32767, 32767);
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// Low-pass filter to smooth out the multiplier (prevents sudden jumps in pitch)
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const double lpf_gain = -std::expm1(-1.0 / (num_samples * SAMPLE_RATE_LPF));
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const float sinc_ratio = 1.0f / std::max(1.0f, float(ratio * std::max(m_multiplier, multiplier)));
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s16 r1 = read_buffer((indexR + 1) & INDEX_MASK); // current
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s16 r2 = read_buffer((indexR2 + 1) & INDEX_MASK); // next
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int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
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sampleR = (sampleR * rvolume) >> 8;
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sampleR += samples[currentSample];
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samples[currentSample] = std::clamp(sampleR, -32767, 32767);
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bool reversed = false;
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for (u32 sample = 0; sample < num_samples; ++sample)
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{
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const auto [l, r] = GetSample(indexR, m_frac, sinc_ratio);
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samples[2 * sample + 0] = clamp(samples[2 * sample + 0] + s32(r * r_volume));
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samples[2 * sample + 1] = clamp(samples[2 * sample + 1] + s32(l * l_volume));
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// Smooth out multiplier with low pass filter
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m_multiplier += lpf_gain * (multiplier - m_multiplier);
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// Determine if we need to reverse the audio (i.e. we have run out of samples)
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reversed |= remaining_samples <= 0;
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reversed &= remaining_samples <= low_watermark;
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// Manage fractional index and increment index.
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m_frac += m_multiplier * (reversed ? -ratio : +ratio);
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const s32 inc = static_cast<s32>(m_frac);
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remaining_samples -= inc;
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indexR += inc;
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m_frac -= inc;
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}
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// Flush cached variable
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m_indexR.store(indexR);
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}
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// This is for audio stretching, which requires no playback speed shenanigans
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u32 Mixer::MixerFifo::MixRaw(s16* samples, u32 num_samples)
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{
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// Cache access in non-volatile variable. No race conditions are here, as indexW only increases
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// (which is fine), and this is the only place indexR is written to.
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const u32 indexW = m_indexW.load(); // Write index in circular buffer
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u32 indexR = m_indexR.load(); // Read index in circular buffer
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const float l_volume = float(m_LVolume.load()) / 256.0f;
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const float r_volume = float(m_RVolume.load()) / 256.0f;
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auto clamp = [](s32 input) -> s16 { return static_cast<s16>(std::clamp(input, -32768, 32767)); };
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const double aid_sample_rate = FIXED_SAMPLE_RATE_DIVIDEND / double(m_input_sample_rate_divisor);
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const double ratio = aid_sample_rate / m_mixer->m_sampleRate;
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const float sinc_ratio = 1.0f / std::max(1.0f, static_cast<float>(ratio));
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// Calculate the number of samples remaining in the buffer
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const s32 sinc_window_width = m_mixer->m_config_sinc_window_width;
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s32 remaining_samples = s32((indexW - indexR) & INDEX_MASK) - sinc_window_width - 2;
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u32 sample = 0;
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for (; sample < num_samples && 0 < remaining_samples; ++sample)
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{
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const auto [l, r] = GetSample(indexR, m_frac, sinc_ratio);
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samples[2 * sample + 0] = clamp(samples[2 * sample + 0] + s32(r * r_volume));
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samples[2 * sample + 1] = clamp(samples[2 * sample + 1] + s32(l * l_volume));
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m_frac += ratio;
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indexR += 2 * (u16)(m_frac >> 16);
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m_frac &= 0xffff;
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const s32 inc = static_cast<s32>(m_frac);
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remaining_samples -= inc;
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indexR += inc;
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m_frac -= inc;
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}
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// Actual number of samples written to the buffer without padding.
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unsigned int actual_sample_count = currentSample / 2;
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// Padding
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short s[2];
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s[0] = read_buffer((indexR - 1) & INDEX_MASK);
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s[1] = read_buffer((indexR - 2) & INDEX_MASK);
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s[0] = (s[0] * rvolume) >> 8;
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s[1] = (s[1] * lvolume) >> 8;
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for (; currentSample < numSamples * 2; currentSample += 2)
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if (sample != 0)
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{
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int sampleR = std::clamp(s[0] + samples[currentSample + 0], -32767, 32767);
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int sampleL = std::clamp(s[1] + samples[currentSample + 1], -32767, 32767);
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samples[currentSample + 0] = sampleR;
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samples[currentSample + 1] = sampleL;
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for (u32 padding = sample; padding < num_samples; ++padding)
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{
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samples[2 * padding + 0] = samples[2 * sample - 2];
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samples[2 * padding + 1] = samples[2 * sample - 1];
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}
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}
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// Flush cached variable
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m_indexR.store(indexR);
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return actual_sample_count;
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return sample;
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}
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unsigned int Mixer::Mix(short* samples, unsigned int num_samples)
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u32 Mixer::Mix(s16* samples, u32 num_samples)
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{
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if (!samples)
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return 0;
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memset(samples, 0, num_samples * 2 * sizeof(short));
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memset(samples, 0, num_samples * 2 * sizeof(s16));
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// TODO: Determine how emulation speed will be used in audio
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// const float emulation_speed = g_perf_metrics.GetSpeed();
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const float emulation_speed = m_config_emulation_speed;
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const int timing_variance = m_config_timing_variance;
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if (m_config_audio_stretch)
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{
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unsigned int available_samples =
|
||||
std::min(m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples());
|
||||
// We want to get as many samples out of this as possible. Usually all mixers will have the
|
||||
// same number of samples available, but if not, we want to empty all of them.
|
||||
const u32 available_samples =
|
||||
std::min({m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples()});
|
||||
|
||||
ASSERT_MSG(AUDIO, available_samples <= MAX_SAMPLES,
|
||||
"Audio stretching would overflow m_scratch_buffer: min({}, {}) -> {} > {} ({})",
|
||||
|
|
@ -178,43 +225,47 @@ unsigned int Mixer::Mix(short* samples, unsigned int num_samples)
|
|||
|
||||
m_scratch_buffer.fill(0);
|
||||
|
||||
m_dma_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
|
||||
timing_variance);
|
||||
m_streaming_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
|
||||
timing_variance);
|
||||
m_wiimote_speaker_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
|
||||
timing_variance);
|
||||
m_skylander_portal_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
|
||||
timing_variance);
|
||||
m_dma_mixer.MixRaw(m_scratch_buffer.data(), available_samples);
|
||||
m_streaming_mixer.MixRaw(m_scratch_buffer.data(), available_samples);
|
||||
m_wiimote_speaker_mixer.MixRaw(m_scratch_buffer.data(), available_samples);
|
||||
m_skylander_portal_mixer.MixRaw(m_scratch_buffer.data(), available_samples);
|
||||
|
||||
for (auto& mixer : m_gba_mixers)
|
||||
{
|
||||
mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
|
||||
timing_variance);
|
||||
}
|
||||
mixer.MixRaw(m_scratch_buffer.data(), available_samples);
|
||||
|
||||
if (!m_is_stretching)
|
||||
{
|
||||
m_stretcher.Clear();
|
||||
m_is_stretching = true;
|
||||
}
|
||||
|
||||
m_stretcher.ProcessSamples(m_scratch_buffer.data(), available_samples, num_samples);
|
||||
m_stretcher.GetStretchedSamples(samples, num_samples);
|
||||
|
||||
g_perf_metrics.CountAudioLatency(m_stretcher.AvailableSamplesTime() +
|
||||
m_dma_mixer.AvailableSamplesTime());
|
||||
}
|
||||
else
|
||||
{
|
||||
m_dma_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
|
||||
m_streaming_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
|
||||
m_wiimote_speaker_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
|
||||
m_skylander_portal_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
|
||||
float target_speed = m_config_emulation_speed;
|
||||
if (target_speed <= 0.0f || Core::GetIsThrottlerTempDisabled())
|
||||
target_speed = g_perf_metrics.GetAudioSpeed();
|
||||
|
||||
m_dma_mixer.Mix(samples, num_samples, target_speed);
|
||||
m_streaming_mixer.Mix(samples, num_samples, target_speed);
|
||||
m_wiimote_speaker_mixer.Mix(samples, num_samples, target_speed);
|
||||
m_skylander_portal_mixer.Mix(samples, num_samples, target_speed);
|
||||
for (auto& mixer : m_gba_mixers)
|
||||
mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
|
||||
mixer.Mix(samples, num_samples, target_speed);
|
||||
|
||||
g_perf_metrics.CountAudioLatency(m_dma_mixer.AvailableSamplesTime());
|
||||
m_is_stretching = false;
|
||||
}
|
||||
|
||||
return num_samples;
|
||||
}
|
||||
|
||||
unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
|
||||
u32 Mixer::MixSurround(float* samples, u32 num_samples)
|
||||
{
|
||||
if (!num_samples)
|
||||
return 0;
|
||||
|
|
@ -243,65 +294,62 @@ unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
|
|||
return num_samples;
|
||||
}
|
||||
|
||||
void Mixer::MixerFifo::PushSamples(const short* samples, unsigned int num_samples)
|
||||
void Mixer::MixerFifo::PushSamples(const s16* samples, u32 num_samples)
|
||||
{
|
||||
// Cache access in non-volatile variable
|
||||
// indexR isn't allowed to cache in the audio throttling loop as it
|
||||
// needs to get updates to not deadlock.
|
||||
u32 indexW = m_indexW.load();
|
||||
|
||||
// Check if we have enough free space
|
||||
// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
|
||||
if (num_samples * 2 + ((indexW - m_indexR.load()) & INDEX_MASK) >= MAX_SAMPLES * 2)
|
||||
return;
|
||||
// prevent writing into the buffer if it's full
|
||||
const s32 sinc_window_width = m_mixer->m_config_sinc_window_width;
|
||||
const s32 writable_samples =
|
||||
static_cast<s32>((m_indexR.load() - sinc_window_width - indexW) & INDEX_MASK);
|
||||
num_samples = std::min(num_samples, static_cast<u32>(std::max(0, writable_samples)));
|
||||
|
||||
// We do float conversion here to avoid doing it in the mixing loop, which is performance
|
||||
// critical. That way we only need to do it once per sample.
|
||||
if (m_little_endian)
|
||||
for (u32 i = 0; i < num_samples; ++i, ++indexW)
|
||||
m_buffer[indexW & INDEX_MASK] = std::make_pair(static_cast<float>(samples[i * 2 + 0]),
|
||||
static_cast<float>(samples[i * 2 + 1]));
|
||||
|
||||
// AyuanX: Actual re-sampling work has been moved to sound thread
|
||||
// to alleviate the workload on main thread
|
||||
// and we simply store raw data here to make fast mem copy
|
||||
int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
|
||||
if (over_bytes > 0)
|
||||
{
|
||||
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
|
||||
memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
|
||||
}
|
||||
else
|
||||
{
|
||||
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
|
||||
}
|
||||
for (u32 i = 0; i < num_samples; ++i, ++indexW)
|
||||
m_buffer[indexW & INDEX_MASK] =
|
||||
std::make_pair(static_cast<float>(static_cast<s16>(Common::swap16(samples[i * 2 + 0]))),
|
||||
static_cast<float>(static_cast<s16>(Common::swap16(samples[i * 2 + 1]))));
|
||||
|
||||
m_indexW.fetch_add(num_samples * 2);
|
||||
m_indexW.store(indexW);
|
||||
}
|
||||
|
||||
void Mixer::PushSamples(const short* samples, unsigned int num_samples)
|
||||
void Mixer::PushSamples(const s16* samples, u32 num_samples)
|
||||
{
|
||||
m_dma_mixer.PushSamples(samples, num_samples);
|
||||
if (m_log_dsp_audio)
|
||||
{
|
||||
int sample_rate_divisor = m_dma_mixer.GetInputSampleRateDivisor();
|
||||
u32 sample_rate_divisor = m_dma_mixer.GetInputSampleRateDivisor();
|
||||
auto volume = m_dma_mixer.GetVolume();
|
||||
m_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples, sample_rate_divisor, volume.first,
|
||||
volume.second);
|
||||
}
|
||||
}
|
||||
|
||||
void Mixer::PushStreamingSamples(const short* samples, unsigned int num_samples)
|
||||
void Mixer::PushStreamingSamples(const s16* samples, u32 num_samples)
|
||||
{
|
||||
m_streaming_mixer.PushSamples(samples, num_samples);
|
||||
if (m_log_dtk_audio)
|
||||
{
|
||||
int sample_rate_divisor = m_streaming_mixer.GetInputSampleRateDivisor();
|
||||
u32 sample_rate_divisor = m_streaming_mixer.GetInputSampleRateDivisor();
|
||||
auto volume = m_streaming_mixer.GetVolume();
|
||||
m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples, sample_rate_divisor, volume.first,
|
||||
volume.second);
|
||||
}
|
||||
}
|
||||
|
||||
void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
|
||||
unsigned int sample_rate_divisor)
|
||||
void Mixer::PushWiimoteSpeakerSamples(const s16* samples, u32 num_samples, u32 sample_rate_divisor)
|
||||
{
|
||||
// Max 20 bytes/speaker report, may be 4-bit ADPCM so multiply by 2
|
||||
static constexpr u32 MAX_SPEAKER_SAMPLES = 20 * 2;
|
||||
std::array<short, MAX_SPEAKER_SAMPLES * 2> samples_stereo;
|
||||
std::array<s16, MAX_SPEAKER_SAMPLES * 2> samples_stereo;
|
||||
|
||||
ASSERT_MSG(AUDIO, num_samples <= MAX_SPEAKER_SAMPLES,
|
||||
"num_samples would overflow samples_stereo: {} > {}", num_samples,
|
||||
|
|
@ -310,9 +358,9 @@ void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_sam
|
|||
{
|
||||
m_wiimote_speaker_mixer.SetInputSampleRateDivisor(sample_rate_divisor);
|
||||
|
||||
for (unsigned int i = 0; i < num_samples; ++i)
|
||||
for (u32 i = 0; i < num_samples; ++i)
|
||||
{
|
||||
samples_stereo[i * 2] = samples[i];
|
||||
samples_stereo[i * 2 + 0] = samples[i];
|
||||
samples_stereo[i * 2 + 1] = samples[i];
|
||||
}
|
||||
|
||||
|
|
@ -320,12 +368,12 @@ void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_sam
|
|||
}
|
||||
}
|
||||
|
||||
void Mixer::PushSkylanderPortalSamples(const u8* samples, unsigned int num_samples)
|
||||
void Mixer::PushSkylanderPortalSamples(const u8* samples, u32 num_samples)
|
||||
{
|
||||
// Skylander samples are always supplied as 64 bytes, 32 x 16 bit samples
|
||||
// The portal speaker is 1 channel, so duplicate and play as stereo audio
|
||||
static constexpr u32 MAX_PORTAL_SPEAKER_SAMPLES = 32;
|
||||
std::array<short, MAX_PORTAL_SPEAKER_SAMPLES * 2> samples_stereo;
|
||||
std::array<s16, MAX_PORTAL_SPEAKER_SAMPLES * 2> samples_stereo;
|
||||
|
||||
ASSERT_MSG(AUDIO, num_samples <= MAX_PORTAL_SPEAKER_SAMPLES,
|
||||
"num_samples is not less or equal to 32: {} > {}", num_samples,
|
||||
|
|
@ -333,10 +381,10 @@ void Mixer::PushSkylanderPortalSamples(const u8* samples, unsigned int num_sampl
|
|||
|
||||
if (num_samples <= MAX_PORTAL_SPEAKER_SAMPLES)
|
||||
{
|
||||
for (unsigned int i = 0; i < num_samples; ++i)
|
||||
for (u32 i = 0; i < num_samples; ++i)
|
||||
{
|
||||
s16 sample = static_cast<u16>(samples[i * 2 + 1]) << 8 | static_cast<u16>(samples[i * 2]);
|
||||
samples_stereo[i * 2] = sample;
|
||||
samples_stereo[i * 2 + 0] = sample;
|
||||
samples_stereo[i * 2 + 1] = sample;
|
||||
}
|
||||
|
||||
|
|
@ -344,37 +392,37 @@ void Mixer::PushSkylanderPortalSamples(const u8* samples, unsigned int num_sampl
|
|||
}
|
||||
}
|
||||
|
||||
void Mixer::PushGBASamples(int device_number, const short* samples, unsigned int num_samples)
|
||||
void Mixer::PushGBASamples(int device_number, const s16* samples, u32 num_samples)
|
||||
{
|
||||
m_gba_mixers[device_number].PushSamples(samples, num_samples);
|
||||
}
|
||||
|
||||
void Mixer::SetDMAInputSampleRateDivisor(unsigned int rate_divisor)
|
||||
void Mixer::SetDMAInputSampleRateDivisor(u32 rate_divisor)
|
||||
{
|
||||
m_dma_mixer.SetInputSampleRateDivisor(rate_divisor);
|
||||
}
|
||||
|
||||
void Mixer::SetStreamInputSampleRateDivisor(unsigned int rate_divisor)
|
||||
void Mixer::SetStreamInputSampleRateDivisor(u32 rate_divisor)
|
||||
{
|
||||
m_streaming_mixer.SetInputSampleRateDivisor(rate_divisor);
|
||||
}
|
||||
|
||||
void Mixer::SetGBAInputSampleRateDivisors(int device_number, unsigned int rate_divisor)
|
||||
void Mixer::SetGBAInputSampleRateDivisors(int device_number, u32 rate_divisor)
|
||||
{
|
||||
m_gba_mixers[device_number].SetInputSampleRateDivisor(rate_divisor);
|
||||
}
|
||||
|
||||
void Mixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
|
||||
void Mixer::SetStreamingVolume(u32 lvolume, u32 rvolume)
|
||||
{
|
||||
m_streaming_mixer.SetVolume(lvolume, rvolume);
|
||||
}
|
||||
|
||||
void Mixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
|
||||
void Mixer::SetWiimoteSpeakerVolume(u32 lvolume, u32 rvolume)
|
||||
{
|
||||
m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
|
||||
}
|
||||
|
||||
void Mixer::SetGBAVolume(int device_number, unsigned int lvolume, unsigned int rvolume)
|
||||
void Mixer::SetGBAVolume(int device_number, u32 lvolume, u32 rvolume)
|
||||
{
|
||||
m_gba_mixers[device_number].SetVolume(lvolume, rvolume);
|
||||
}
|
||||
|
|
@ -456,8 +504,9 @@ void Mixer::StopLogDSPAudio()
|
|||
void Mixer::RefreshConfig()
|
||||
{
|
||||
m_config_emulation_speed = Config::Get(Config::MAIN_EMULATION_SPEED);
|
||||
m_config_timing_variance = Config::Get(Config::MAIN_TIMING_VARIANCE);
|
||||
m_config_audio_stretch = Config::Get(Config::MAIN_AUDIO_STRETCH);
|
||||
m_config_direct_latency = Config::Get(Config::MAIN_AUDIO_DIRECT_LATENCY);
|
||||
m_config_sinc_window_width = Config::Get(Config::MAIN_AUDIO_SINC_WINDOW_WIDTH);
|
||||
}
|
||||
|
||||
void Mixer::MixerFifo::DoState(PointerWrap& p)
|
||||
|
|
@ -467,17 +516,17 @@ void Mixer::MixerFifo::DoState(PointerWrap& p)
|
|||
p.Do(m_RVolume);
|
||||
}
|
||||
|
||||
void Mixer::MixerFifo::SetInputSampleRateDivisor(unsigned int rate_divisor)
|
||||
void Mixer::MixerFifo::SetInputSampleRateDivisor(u64 rate_divisor)
|
||||
{
|
||||
m_input_sample_rate_divisor = rate_divisor;
|
||||
}
|
||||
|
||||
unsigned int Mixer::MixerFifo::GetInputSampleRateDivisor() const
|
||||
u64 Mixer::MixerFifo::GetInputSampleRateDivisor() const
|
||||
{
|
||||
return m_input_sample_rate_divisor;
|
||||
}
|
||||
|
||||
void Mixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
|
||||
void Mixer::MixerFifo::SetVolume(s32 lvolume, s32 rvolume)
|
||||
{
|
||||
m_LVolume.store(lvolume + (lvolume >> 7));
|
||||
m_RVolume.store(rvolume + (rvolume >> 7));
|
||||
|
|
@ -488,11 +537,26 @@ std::pair<s32, s32> Mixer::MixerFifo::GetVolume() const
|
|||
return std::make_pair(m_LVolume.load(), m_RVolume.load());
|
||||
}
|
||||
|
||||
unsigned int Mixer::MixerFifo::AvailableSamples() const
|
||||
u32 Mixer::MixerFifo::AvailableFIFOSamples() const
|
||||
{
|
||||
unsigned int samples_in_fifo = ((m_indexW.load() - m_indexR.load()) & INDEX_MASK) / 2;
|
||||
if (samples_in_fifo <= 1)
|
||||
return 0; // Mixer::MixerFifo::Mix always keeps one sample in the buffer.
|
||||
return (samples_in_fifo - 1) * static_cast<u64>(m_mixer->m_sampleRate) *
|
||||
m_input_sample_rate_divisor / FIXED_SAMPLE_RATE_DIVIDEND;
|
||||
const s32 samples_in_fifo = static_cast<s32>((m_indexW.load() - m_indexR.load()) & INDEX_MASK) -
|
||||
m_mixer->m_config_sinc_window_width - 4;
|
||||
|
||||
if (samples_in_fifo <= 0)
|
||||
return 0;
|
||||
|
||||
return static_cast<u32>(samples_in_fifo);
|
||||
}
|
||||
|
||||
u32 Mixer::MixerFifo::AvailableSamples() const
|
||||
{
|
||||
const s64 samples_in_fifo = static_cast<s64>(AvailableFIFOSamples());
|
||||
return static_cast<u32>(samples_in_fifo * m_mixer->m_sampleRate * m_input_sample_rate_divisor /
|
||||
FIXED_SAMPLE_RATE_DIVIDEND);
|
||||
}
|
||||
|
||||
DT Mixer::MixerFifo::AvailableSamplesTime() const
|
||||
{
|
||||
return std::chrono::duration_cast<DT>(DT_s(AvailableFIFOSamples()) * m_input_sample_rate_divisor /
|
||||
FIXED_SAMPLE_RATE_DIVIDEND);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -5,6 +5,7 @@
|
|||
|
||||
#include <array>
|
||||
#include <atomic>
|
||||
#include <optional>
|
||||
|
||||
#include "AudioCommon/AudioStretcher.h"
|
||||
#include "AudioCommon/SurroundDecoder.h"
|
||||
|
|
@ -17,32 +18,31 @@ class PointerWrap;
|
|||
class Mixer final
|
||||
{
|
||||
public:
|
||||
explicit Mixer(unsigned int BackendSampleRate);
|
||||
explicit Mixer(u32 BackendSampleRate);
|
||||
~Mixer();
|
||||
|
||||
void DoState(PointerWrap& p);
|
||||
|
||||
// Called from audio threads
|
||||
unsigned int Mix(short* samples, unsigned int numSamples);
|
||||
unsigned int MixSurround(float* samples, unsigned int num_samples);
|
||||
u32 Mix(s16* samples, u32 num_samples);
|
||||
u32 MixSurround(float* samples, u32 num_samples);
|
||||
|
||||
// Called from main thread
|
||||
void PushSamples(const short* samples, unsigned int num_samples);
|
||||
void PushStreamingSamples(const short* samples, unsigned int num_samples);
|
||||
void PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
|
||||
unsigned int sample_rate_divisor);
|
||||
void PushSkylanderPortalSamples(const u8* samples, unsigned int num_samples);
|
||||
void PushGBASamples(int device_number, const short* samples, unsigned int num_samples);
|
||||
void PushSamples(const s16* samples, u32 num_samples);
|
||||
void PushStreamingSamples(const s16* samples, u32 num_samples);
|
||||
void PushWiimoteSpeakerSamples(const s16* samples, u32 num_samples, u32 sample_rate_divisor);
|
||||
void PushSkylanderPortalSamples(const u8* samples, u32 num_samples);
|
||||
void PushGBASamples(int device_number, const s16* samples, u32 num_samples);
|
||||
|
||||
unsigned int GetSampleRate() const { return m_sampleRate; }
|
||||
u32 GetSampleRate() const { return m_sampleRate; }
|
||||
|
||||
void SetDMAInputSampleRateDivisor(unsigned int rate_divisor);
|
||||
void SetStreamInputSampleRateDivisor(unsigned int rate_divisor);
|
||||
void SetGBAInputSampleRateDivisors(int device_number, unsigned int rate_divisor);
|
||||
void SetDMAInputSampleRateDivisor(u32 rate_divisor);
|
||||
void SetStreamInputSampleRateDivisor(u32 rate_divisor);
|
||||
void SetGBAInputSampleRateDivisors(int device_number, u32 rate_divisor);
|
||||
|
||||
void SetStreamingVolume(unsigned int lvolume, unsigned int rvolume);
|
||||
void SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume);
|
||||
void SetGBAVolume(int device_number, unsigned int lvolume, unsigned int rvolume);
|
||||
void SetStreamingVolume(u32 lvolume, u32 rvolume);
|
||||
void SetWiimoteSpeakerVolume(u32 lvolume, u32 rvolume);
|
||||
void SetGBAVolume(int device_number, u32 lvolume, u32 rvolume);
|
||||
|
||||
void StartLogDTKAudio(const std::string& filename);
|
||||
void StopLogDTKAudio();
|
||||
|
|
@ -55,43 +55,48 @@ public:
|
|||
|
||||
private:
|
||||
static constexpr u32 MAX_SAMPLES = 1024 * 4; // 128 ms
|
||||
static constexpr u32 INDEX_MASK = MAX_SAMPLES * 2 - 1;
|
||||
static constexpr int MAX_FREQ_SHIFT = 200; // Per 32000 Hz
|
||||
static constexpr float CONTROL_FACTOR = 0.2f;
|
||||
static constexpr u32 CONTROL_AVG = 32; // In freq_shift per FIFO size offset
|
||||
static constexpr u32 INDEX_MASK = MAX_SAMPLES - 1;
|
||||
static constexpr double MAX_PITCH_SHIFT = 1.0145453; // 2 ^ (+1/48) - 1/4th Note Up
|
||||
static constexpr double MIN_PITCH_SHIFT = 0.9856632; // 2 ^ (-1/48) - 1/4th Note Down
|
||||
static constexpr double SAMPLE_RATE_LPF = 1.0 / 4.0; // How much to smooth out sample rate
|
||||
static constexpr double CONTROL_EFFORT = 1.0 / 64.0; // Lowers the strength of pitch shifting
|
||||
|
||||
const unsigned int SURROUND_CHANNELS = 6;
|
||||
const u32 SURROUND_CHANNELS = 6;
|
||||
|
||||
class MixerFifo final
|
||||
{
|
||||
public:
|
||||
MixerFifo(Mixer* mixer, unsigned sample_rate_divisor, bool little_endian)
|
||||
MixerFifo(Mixer* mixer, u32 sample_rate_divisor, bool little_endian)
|
||||
: m_mixer(mixer), m_input_sample_rate_divisor(sample_rate_divisor),
|
||||
m_little_endian(little_endian)
|
||||
{
|
||||
}
|
||||
void DoState(PointerWrap& p);
|
||||
void PushSamples(const short* samples, unsigned int num_samples);
|
||||
unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit,
|
||||
float emulationspeed, int timing_variance);
|
||||
void SetInputSampleRateDivisor(unsigned int rate_divisor);
|
||||
unsigned int GetInputSampleRateDivisor() const;
|
||||
void SetVolume(unsigned int lvolume, unsigned int rvolume);
|
||||
void PushSamples(const s16* samples, u32 num_samples);
|
||||
std::pair<float, float> GetSample(u32 index, float frac, float sinc_ratio);
|
||||
void Mix(s16* samples, u32 num_samples, double target_speed);
|
||||
u32 MixRaw(s16* samples, u32 num_samples);
|
||||
void SetInputSampleRateDivisor(u64 rate_divisor);
|
||||
u64 GetInputSampleRateDivisor() const;
|
||||
void SetVolume(s32 lvolume, s32 rvolume);
|
||||
std::pair<s32, s32> GetVolume() const;
|
||||
unsigned int AvailableSamples() const;
|
||||
u32 AvailableFIFOSamples() const;
|
||||
u32 AvailableSamples() const;
|
||||
DT AvailableSamplesTime() const;
|
||||
|
||||
private:
|
||||
Mixer* m_mixer;
|
||||
unsigned m_input_sample_rate_divisor;
|
||||
u64 m_input_sample_rate_divisor;
|
||||
bool m_little_endian;
|
||||
std::array<short, MAX_SAMPLES * 2> m_buffer{};
|
||||
std::array<std::pair<float, float>, MAX_SAMPLES> m_buffer{};
|
||||
std::atomic<u32> m_indexW{0};
|
||||
std::atomic<u32> m_indexR{0};
|
||||
// Volume ranges from 0-256
|
||||
std::atomic<s32> m_LVolume{256};
|
||||
std::atomic<s32> m_RVolume{256};
|
||||
float m_numLeftI = 0.0f;
|
||||
u32 m_frac = 0;
|
||||
|
||||
double m_multiplier = 1.0;
|
||||
double m_frac = 0.0;
|
||||
};
|
||||
|
||||
void RefreshConfig();
|
||||
|
|
@ -104,12 +109,12 @@ private:
|
|||
MixerFifo{this, FIXED_SAMPLE_RATE_DIVIDEND / 48000, true},
|
||||
MixerFifo{this, FIXED_SAMPLE_RATE_DIVIDEND / 48000, true},
|
||||
MixerFifo{this, FIXED_SAMPLE_RATE_DIVIDEND / 48000, true}};
|
||||
unsigned int m_sampleRate;
|
||||
u32 m_sampleRate;
|
||||
|
||||
bool m_is_stretching = false;
|
||||
AudioCommon::AudioStretcher m_stretcher;
|
||||
AudioCommon::SurroundDecoder m_surround_decoder;
|
||||
std::array<short, MAX_SAMPLES * 2> m_scratch_buffer{};
|
||||
std::array<s16, MAX_SAMPLES * 2> m_scratch_buffer{};
|
||||
|
||||
WaveFileWriter m_wave_writer_dtk;
|
||||
WaveFileWriter m_wave_writer_dsp;
|
||||
|
|
@ -118,8 +123,9 @@ private:
|
|||
bool m_log_dsp_audio = false;
|
||||
|
||||
float m_config_emulation_speed;
|
||||
int m_config_timing_variance;
|
||||
bool m_config_audio_stretch;
|
||||
int m_config_direct_latency;
|
||||
int m_config_sinc_window_width;
|
||||
|
||||
Config::ConfigChangedCallbackID m_config_changed_callback_id;
|
||||
};
|
||||
|
|
|
|||
|
|
@ -44,7 +44,6 @@ const Info<bool> MAIN_LARGE_ENTRY_POINTS_MAP{{System::Main, "Core", "LargeEntryP
|
|||
const Info<bool> MAIN_ACCURATE_CPU_CACHE{{System::Main, "Core", "AccurateCPUCache"}, false};
|
||||
const Info<bool> MAIN_DSP_HLE{{System::Main, "Core", "DSPHLE"}, true};
|
||||
const Info<int> MAIN_MAX_FALLBACK{{System::Main, "Core", "MaxFallback"}, 100};
|
||||
const Info<int> MAIN_TIMING_VARIANCE{{System::Main, "Core", "TimingVariance"}, 40};
|
||||
const Info<bool> MAIN_CPU_THREAD{{System::Main, "Core", "CPUThread"}, true};
|
||||
const Info<bool> MAIN_SYNC_ON_SKIP_IDLE{{System::Main, "Core", "SyncOnSkipIdle"}, true};
|
||||
const Info<std::string> MAIN_DEFAULT_ISO{{System::Main, "Core", "DefaultISO"}, ""};
|
||||
|
|
@ -57,7 +56,9 @@ const Info<AudioCommon::DPL2Quality> MAIN_DPL2_QUALITY{{System::Main, "Core", "D
|
|||
AudioCommon::GetDefaultDPL2Quality()};
|
||||
const Info<int> MAIN_AUDIO_LATENCY{{System::Main, "Core", "AudioLatency"}, 20};
|
||||
const Info<bool> MAIN_AUDIO_STRETCH{{System::Main, "Core", "AudioStretch"}, false};
|
||||
const Info<int> MAIN_AUDIO_DIRECT_LATENCY{{System::Main, "Core", "AudioDirectLatency"}, 20};
|
||||
const Info<int> MAIN_AUDIO_STRETCH_LATENCY{{System::Main, "Core", "AudioStretchMaxLatency"}, 80};
|
||||
const Info<int> MAIN_AUDIO_SINC_WINDOW_WIDTH{{System::Main, "Core", "AudioSincResampleTaps"}, 6};
|
||||
const Info<std::string> MAIN_MEMCARD_A_PATH{{System::Main, "Core", "MemcardAPath"}, ""};
|
||||
const Info<std::string> MAIN_MEMCARD_B_PATH{{System::Main, "Core", "MemcardBPath"}, ""};
|
||||
const Info<std::string>& GetInfoForMemcardPath(ExpansionInterface::Slot slot)
|
||||
|
|
|
|||
|
|
@ -62,7 +62,6 @@ extern const Info<bool> MAIN_ACCURATE_CPU_CACHE;
|
|||
// Should really be in the DSP section, but we're kind of stuck with bad decisions made in the past.
|
||||
extern const Info<bool> MAIN_DSP_HLE;
|
||||
extern const Info<int> MAIN_MAX_FALLBACK;
|
||||
extern const Info<int> MAIN_TIMING_VARIANCE;
|
||||
extern const Info<bool> MAIN_CPU_THREAD;
|
||||
extern const Info<bool> MAIN_SYNC_ON_SKIP_IDLE;
|
||||
extern const Info<std::string> MAIN_DEFAULT_ISO;
|
||||
|
|
@ -73,7 +72,9 @@ extern const Info<bool> MAIN_DPL2_DECODER;
|
|||
extern const Info<AudioCommon::DPL2Quality> MAIN_DPL2_QUALITY;
|
||||
extern const Info<int> MAIN_AUDIO_LATENCY;
|
||||
extern const Info<bool> MAIN_AUDIO_STRETCH;
|
||||
extern const Info<int> MAIN_AUDIO_DIRECT_LATENCY;
|
||||
extern const Info<int> MAIN_AUDIO_STRETCH_LATENCY;
|
||||
extern const Info<int> MAIN_AUDIO_SINC_WINDOW_WIDTH;
|
||||
extern const Info<std::string> MAIN_MEMCARD_A_PATH;
|
||||
extern const Info<std::string> MAIN_MEMCARD_B_PATH;
|
||||
const Info<std::string>& GetInfoForMemcardPath(ExpansionInterface::Slot slot);
|
||||
|
|
|
|||
|
|
@ -136,8 +136,6 @@ void CoreTimingManager::RefreshConfig()
|
|||
// too long or going full speed in an attempt to catch up to timings.
|
||||
m_max_fallback = std::chrono::duration_cast<DT>(DT_ms(Config::Get(Config::MAIN_MAX_FALLBACK)));
|
||||
|
||||
m_max_variance = std::chrono::duration_cast<DT>(DT_ms(Config::Get(Config::MAIN_TIMING_VARIANCE)));
|
||||
|
||||
if (AchievementManager::GetInstance().IsHardcoreModeActive() &&
|
||||
Config::Get(Config::MAIN_EMULATION_SPEED) < 1.0f &&
|
||||
Config::Get(Config::MAIN_EMULATION_SPEED) > 0.0f)
|
||||
|
|
@ -399,7 +397,7 @@ void CoreTimingManager::Throttle(const s64 target_cycle)
|
|||
m_throttle_deadline = min_deadline;
|
||||
}
|
||||
|
||||
const TimePoint vi_deadline = time - std::min(m_max_fallback, m_max_variance) / 2;
|
||||
const TimePoint vi_deadline = time - m_max_fallback;
|
||||
|
||||
// Skip the VI interrupt if the CPU is lagging by a certain amount.
|
||||
// It doesn't matter what amount of lag we skip VI at, as long as it's constant.
|
||||
|
|
|
|||
|
|
@ -195,7 +195,6 @@ private:
|
|||
bool m_throttle_disable_vi_int = false;
|
||||
|
||||
DT m_max_fallback = {};
|
||||
DT m_max_variance = {};
|
||||
double m_emulation_speed = 1.0;
|
||||
|
||||
void ResetThrottle(s64 cycle);
|
||||
|
|
|
|||
|
|
@ -15,6 +15,7 @@
|
|||
#include <QSlider>
|
||||
#include <QSpacerItem>
|
||||
#include <QSpinBox>
|
||||
#include <QStackedWidget>
|
||||
#include <QVBoxLayout>
|
||||
|
||||
#include "AudioCommon/AudioCommon.h"
|
||||
|
|
@ -137,32 +138,81 @@ void AudioPane::CreateWidgets()
|
|||
backend_layout->addRow(dolby_quality_layout);
|
||||
backend_layout->addRow(m_dolby_quality_latency_label);
|
||||
|
||||
auto* stretching_box = new QGroupBox(tr("Audio Stretching Settings"));
|
||||
auto* stretching_layout = new QGridLayout;
|
||||
m_stretching_enable = new QCheckBox(tr("Enable Audio Stretching"));
|
||||
// Set up for new Audio Options Box with ComboBox and StackedWidget
|
||||
m_audio_group = new QGroupBox(tr("Audio Options"));
|
||||
auto* audio_options_layout = new QGridLayout(m_audio_group);
|
||||
|
||||
// Resampling Options
|
||||
m_audio_resampling_box = new QComboBox;
|
||||
m_audio_resampling_box->addItem(tr("Default (12 Samples)"));
|
||||
m_audio_resampling_box->addItem(tr("High (32 Samples)"));
|
||||
m_audio_resampling_box->addItem(tr("Very High (96 Samples)"));
|
||||
m_audio_resampling_box->addItem(tr("Placebo (256 Samples)"));
|
||||
m_audio_resampling_box->setToolTip(tr(
|
||||
"Dolphin must resample audio from ~32000hz to the modern sample rates like 48000hz.<br><br>"
|
||||
"Higher quality options will only result in lower distortion <i>at high frequencies "
|
||||
"(~16000hz)</i>, at the cost of increased CPU usage. <b>The vast majority of people will be "
|
||||
"unable to notice</b>. <br><br>"
|
||||
"<b>WARNING: If your CPU is not fast enough audio will cutout.</b>"));
|
||||
audio_options_layout->addWidget(new QLabel(tr("Resampling Quality:")), 0, 0);
|
||||
audio_options_layout->addWidget(m_audio_resampling_box, 0, 1);
|
||||
|
||||
// Audio Playback Mode
|
||||
m_audio_playback_mode_box = new QComboBox;
|
||||
m_audio_playback_mode_box->addItem(tr("Direct Playback"));
|
||||
m_audio_playback_mode_box->addItem(tr("Audio Stretching"));
|
||||
m_audio_playback_mode_box->setToolTip(
|
||||
tr("<b>Direct Playback:</b> Audio is played unaltered at a fixed latency.<br><br>"
|
||||
"<b>Audio Stretching:</b> Audio is stretched without changing pitch to match emulation "
|
||||
"speed."));
|
||||
audio_options_layout->addWidget(new QLabel(tr("Playback Mode:")), 1, 0);
|
||||
audio_options_layout->addWidget(m_audio_playback_mode_box, 1, 1);
|
||||
|
||||
// Audio Playback Mode Stack
|
||||
m_audio_playback_mode_stack = new QStackedWidget;
|
||||
audio_options_layout->addWidget(m_audio_playback_mode_stack, 2, 0, 1, 2);
|
||||
|
||||
// Direct Playback Settings
|
||||
m_direct_playback_box = new QGroupBox(tr("Direct Playback Settings"));
|
||||
auto* direct_playback_layout = new QHBoxLayout(m_direct_playback_box);
|
||||
m_direct_playback_latency = new QSlider(Qt::Horizontal);
|
||||
m_direct_playback_indicator = new QLabel();
|
||||
m_direct_playback_latency->setRange(8, 64);
|
||||
m_direct_playback_latency->setSingleStep(4);
|
||||
m_direct_playback_latency->setTickInterval(4);
|
||||
m_direct_playback_latency->setTickPosition(QSlider::TicksBelow);
|
||||
direct_playback_layout->addWidget(new QLabel(tr("Buffer:")));
|
||||
direct_playback_layout->addWidget(m_direct_playback_latency);
|
||||
direct_playback_layout->addWidget(m_direct_playback_indicator);
|
||||
m_audio_playback_mode_stack->addWidget(m_direct_playback_box);
|
||||
m_direct_playback_latency->setToolTip(
|
||||
tr("Sets the buffer size in milliseconds for direct playback.<br><br>"
|
||||
"Higher values may hide small lag spikes at the expense of increased latency."));
|
||||
|
||||
// Audio Stretching Settings
|
||||
m_stretching_box = new QGroupBox(tr("Audio Stretching Settings"));
|
||||
auto* stretching_layout = new QHBoxLayout(m_stretching_box);
|
||||
m_stretching_buffer_slider = new QSlider(Qt::Horizontal);
|
||||
m_stretching_buffer_indicator = new QLabel();
|
||||
m_stretching_buffer_label = new QLabel(tr("Buffer Size:"));
|
||||
stretching_box->setLayout(stretching_layout);
|
||||
m_stretching_buffer_slider->setRange(48, 256);
|
||||
m_stretching_buffer_slider->setSingleStep(8);
|
||||
m_stretching_buffer_slider->setTickInterval(16);
|
||||
m_stretching_buffer_slider->setTickPosition(QSlider::TicksBelow);
|
||||
stretching_layout->addWidget(new QLabel(tr("Buffer:")));
|
||||
stretching_layout->addWidget(m_stretching_buffer_slider);
|
||||
stretching_layout->addWidget(m_stretching_buffer_indicator);
|
||||
m_audio_playback_mode_stack->addWidget(m_stretching_box);
|
||||
|
||||
m_stretching_buffer_slider->setMinimum(5);
|
||||
m_stretching_buffer_slider->setMaximum(300);
|
||||
|
||||
m_stretching_enable->setToolTip(tr("Enables stretching of the audio to match emulation speed."));
|
||||
m_stretching_buffer_slider->setToolTip(tr("Size of stretch buffer in milliseconds. "
|
||||
"Values too low may cause audio crackling."));
|
||||
|
||||
stretching_layout->addWidget(m_stretching_enable, 0, 0, 1, -1);
|
||||
stretching_layout->addWidget(m_stretching_buffer_label, 1, 0);
|
||||
stretching_layout->addWidget(m_stretching_buffer_slider, 1, 1);
|
||||
stretching_layout->addWidget(m_stretching_buffer_indicator, 1, 2);
|
||||
m_stretching_buffer_slider->setToolTip(
|
||||
tr("Sets the buffer size in milliseconds for audio stretching.<br><br>"
|
||||
"Higher values may reduce audio crackling at the expense of increased latency."));
|
||||
|
||||
dsp_box->setSizePolicy(QSizePolicy::Expanding, QSizePolicy::Fixed);
|
||||
|
||||
auto* const main_vbox_layout = new QVBoxLayout;
|
||||
main_vbox_layout->addWidget(dsp_box);
|
||||
main_vbox_layout->addWidget(backend_box);
|
||||
main_vbox_layout->addWidget(stretching_box);
|
||||
main_vbox_layout->addWidget(m_audio_group);
|
||||
|
||||
m_main_layout = new QHBoxLayout;
|
||||
m_main_layout->addLayout(main_vbox_layout);
|
||||
|
|
@ -180,14 +230,19 @@ void AudioPane::ConnectWidgets()
|
|||
{
|
||||
connect(m_latency_spin, &QSpinBox::valueChanged, this, &AudioPane::SaveSettings);
|
||||
}
|
||||
connect(m_stretching_buffer_slider, &QSlider::valueChanged, this, &AudioPane::SaveSettings);
|
||||
|
||||
connect(m_dolby_pro_logic, &QCheckBox::toggled, this, &AudioPane::SaveSettings);
|
||||
connect(m_dolby_quality_slider, &QSlider::valueChanged, this, &AudioPane::SaveSettings);
|
||||
connect(m_stretching_enable, &QCheckBox::toggled, this, &AudioPane::SaveSettings);
|
||||
connect(m_dsp_hle, &QRadioButton::toggled, this, &AudioPane::SaveSettings);
|
||||
connect(m_dsp_lle, &QRadioButton::toggled, this, &AudioPane::SaveSettings);
|
||||
connect(m_dsp_interpreter, &QRadioButton::toggled, this, &AudioPane::SaveSettings);
|
||||
|
||||
connect(m_audio_resampling_box, &QComboBox::currentIndexChanged, this, &AudioPane::SaveSettings);
|
||||
connect(m_audio_playback_mode_box, &QComboBox::currentIndexChanged, this,
|
||||
&AudioPane::SaveSettings);
|
||||
connect(m_direct_playback_latency, &QSlider::valueChanged, this, &AudioPane::SaveSettings);
|
||||
connect(m_stretching_buffer_slider, &QSlider::valueChanged, this, &AudioPane::SaveSettings);
|
||||
|
||||
#ifdef _WIN32
|
||||
connect(m_wasapi_device_combo, &QComboBox::currentIndexChanged, this, &AudioPane::SaveSettings);
|
||||
#endif
|
||||
|
|
@ -242,13 +297,36 @@ void AudioPane::LoadSettings()
|
|||
if (m_latency_control_supported)
|
||||
m_latency_spin->setValue(Config::Get(Config::MAIN_AUDIO_LATENCY));
|
||||
|
||||
// Stretch
|
||||
m_stretching_enable->setChecked(Config::Get(Config::MAIN_AUDIO_STRETCH));
|
||||
m_stretching_buffer_label->setEnabled(m_stretching_enable->isChecked());
|
||||
// Resampling
|
||||
switch (Config::Get(Config::MAIN_AUDIO_SINC_WINDOW_WIDTH))
|
||||
{
|
||||
default:
|
||||
case 6:
|
||||
m_audio_resampling_box->setCurrentIndex(0);
|
||||
break;
|
||||
case 16:
|
||||
m_audio_resampling_box->setCurrentIndex(1);
|
||||
break;
|
||||
case 48:
|
||||
m_audio_resampling_box->setCurrentIndex(2);
|
||||
break;
|
||||
case 128:
|
||||
m_audio_resampling_box->setCurrentIndex(3);
|
||||
break;
|
||||
}
|
||||
|
||||
// Playback Mode
|
||||
int playback_mode = Config::Get(Config::MAIN_AUDIO_STRETCH) ? 1 : 0;
|
||||
m_audio_playback_mode_box->setCurrentIndex(playback_mode);
|
||||
m_audio_playback_mode_stack->setCurrentIndex(playback_mode);
|
||||
|
||||
m_direct_playback_latency->setValue(Config::Get(Config::MAIN_AUDIO_DIRECT_LATENCY));
|
||||
m_direct_playback_indicator->setText(
|
||||
tr("%1 ms").arg(Config::Get(Config::MAIN_AUDIO_DIRECT_LATENCY)));
|
||||
|
||||
m_stretching_buffer_slider->setValue(Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY));
|
||||
m_stretching_buffer_slider->setEnabled(m_stretching_enable->isChecked());
|
||||
m_stretching_buffer_indicator->setEnabled(m_stretching_enable->isChecked());
|
||||
m_stretching_buffer_indicator->setText(tr("%1 ms").arg(m_stretching_buffer_slider->value()));
|
||||
m_stretching_buffer_indicator->setText(
|
||||
tr("%1 ms").arg(Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY)));
|
||||
|
||||
#ifdef _WIN32
|
||||
if (Config::Get(Config::MAIN_WASAPI_DEVICE) == "default")
|
||||
|
|
@ -310,12 +388,34 @@ void AudioPane::SaveSettings()
|
|||
if (m_latency_control_supported)
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_LATENCY, m_latency_spin->value());
|
||||
|
||||
// Resampling
|
||||
switch (m_audio_resampling_box->currentIndex())
|
||||
{
|
||||
default:
|
||||
case 0:
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_SINC_WINDOW_WIDTH, 6);
|
||||
break;
|
||||
case 1:
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_SINC_WINDOW_WIDTH, 16);
|
||||
break;
|
||||
case 2:
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_SINC_WINDOW_WIDTH, 48);
|
||||
break;
|
||||
case 3:
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_SINC_WINDOW_WIDTH, 128);
|
||||
break;
|
||||
}
|
||||
|
||||
// Stretch
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_STRETCH, m_stretching_enable->isChecked());
|
||||
int playback_mode = m_audio_playback_mode_box->currentIndex();
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_STRETCH, playback_mode == 1);
|
||||
m_audio_playback_mode_stack->setCurrentIndex(playback_mode);
|
||||
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_DIRECT_LATENCY, m_direct_playback_latency->value());
|
||||
m_direct_playback_indicator->setText(
|
||||
tr("%1 ms").arg(Config::Get(Config::MAIN_AUDIO_DIRECT_LATENCY)));
|
||||
|
||||
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_STRETCH_LATENCY, m_stretching_buffer_slider->value());
|
||||
m_stretching_buffer_label->setEnabled(m_stretching_enable->isChecked());
|
||||
m_stretching_buffer_slider->setEnabled(m_stretching_enable->isChecked());
|
||||
m_stretching_buffer_indicator->setEnabled(m_stretching_enable->isChecked());
|
||||
m_stretching_buffer_indicator->setText(
|
||||
tr("%1 ms").arg(Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY)));
|
||||
|
||||
|
|
|
|||
|
|
@ -12,11 +12,13 @@ enum class DPL2Quality;
|
|||
|
||||
class QCheckBox;
|
||||
class QComboBox;
|
||||
class QGroupBox;
|
||||
class QHBoxLayout;
|
||||
class QLabel;
|
||||
class QRadioButton;
|
||||
class QSlider;
|
||||
class QSpinBox;
|
||||
class QStackedWidget;
|
||||
class SettingsWindow;
|
||||
|
||||
class AudioPane final : public QWidget
|
||||
|
|
@ -71,9 +73,17 @@ private:
|
|||
QComboBox* m_wasapi_device_combo;
|
||||
#endif
|
||||
|
||||
// Audio Stretching
|
||||
QCheckBox* m_stretching_enable;
|
||||
QLabel* m_stretching_buffer_label;
|
||||
// Audio Options
|
||||
QGroupBox* m_audio_group;
|
||||
QComboBox* m_audio_resampling_box;
|
||||
QComboBox* m_audio_playback_mode_box;
|
||||
QStackedWidget* m_audio_playback_mode_stack;
|
||||
|
||||
QGroupBox* m_direct_playback_box;
|
||||
QSlider* m_direct_playback_latency;
|
||||
QLabel* m_direct_playback_indicator;
|
||||
|
||||
QGroupBox* m_stretching_box;
|
||||
QSlider* m_stretching_buffer_slider;
|
||||
QLabel* m_stretching_buffer_indicator;
|
||||
};
|
||||
|
|
|
|||
|
|
@ -19,11 +19,11 @@ void PerformanceMetrics::Reset()
|
|||
{
|
||||
m_fps_counter.Reset();
|
||||
m_vps_counter.Reset();
|
||||
m_audio_speed_counter.Reset();
|
||||
m_speed_counter.Reset();
|
||||
m_max_speed_counter.Reset();
|
||||
|
||||
m_time_sleeping = DT::zero();
|
||||
m_real_times.fill(Clock::now());
|
||||
m_cpu_times.fill(Core::System::GetInstance().GetCoreTiming().GetCPUTimePoint(0));
|
||||
m_prev_adjusted_time = Clock::now() - m_time_sleeping;
|
||||
}
|
||||
|
||||
void PerformanceMetrics::CountFrame()
|
||||
|
|
@ -36,6 +36,11 @@ void PerformanceMetrics::CountVBlank()
|
|||
m_vps_counter.Count();
|
||||
}
|
||||
|
||||
void PerformanceMetrics::CountAudioLatency(DT latency)
|
||||
{
|
||||
m_audio_latency_counter.Count(latency, false);
|
||||
}
|
||||
|
||||
void PerformanceMetrics::CountThrottleSleep(DT sleep)
|
||||
{
|
||||
std::unique_lock lock(m_time_lock);
|
||||
|
|
@ -45,11 +50,13 @@ void PerformanceMetrics::CountThrottleSleep(DT sleep)
|
|||
void PerformanceMetrics::CountPerformanceMarker(Core::System& system, s64 cyclesLate)
|
||||
{
|
||||
std::unique_lock lock(m_time_lock);
|
||||
const TimePoint adjusted_time = Clock::now() - m_time_sleeping;
|
||||
|
||||
m_audio_speed_counter.Count();
|
||||
m_speed_counter.Count();
|
||||
|
||||
m_real_times[m_time_index] = Clock::now() - m_time_sleeping;
|
||||
m_cpu_times[m_time_index] = system.GetCoreTiming().GetCPUTimePoint(cyclesLate);
|
||||
m_time_index += 1;
|
||||
m_max_speed_counter.Count(adjusted_time - m_prev_adjusted_time);
|
||||
m_prev_adjusted_time = adjusted_time;
|
||||
}
|
||||
|
||||
double PerformanceMetrics::GetFPS() const
|
||||
|
|
@ -67,11 +74,14 @@ double PerformanceMetrics::GetSpeed() const
|
|||
return m_speed_counter.GetHzAvg() / 100.0;
|
||||
}
|
||||
|
||||
double PerformanceMetrics::GetAudioSpeed() const
|
||||
{
|
||||
return m_audio_speed_counter.GetHzAvg() / 100.0;
|
||||
}
|
||||
|
||||
double PerformanceMetrics::GetMaxSpeed() const
|
||||
{
|
||||
std::shared_lock lock(m_time_lock);
|
||||
return DT_s(m_cpu_times[u8(m_time_index - 1)] - m_cpu_times[m_time_index]) /
|
||||
DT_s(m_real_times[u8(m_time_index - 1)] - m_real_times[m_time_index]);
|
||||
return m_max_speed_counter.GetHzAvg() / 100.0;
|
||||
}
|
||||
|
||||
double PerformanceMetrics::GetLastSpeedDenominator() const
|
||||
|
|
@ -145,9 +155,13 @@ void PerformanceMetrics::DrawImGuiStats(const float backbuffer_scale)
|
|||
1000.0,
|
||||
2000.0};
|
||||
|
||||
const DT vblank_time = m_vps_counter.GetDtAvg() + 2 * m_vps_counter.GetDtStd();
|
||||
const DT frame_time = m_fps_counter.GetDtAvg() + 2 * m_fps_counter.GetDtStd();
|
||||
const double target_max_time = DT_ms(vblank_time + frame_time).count();
|
||||
const double vblank_time =
|
||||
DT_ms(m_vps_counter.GetDtAvg() + 2 * m_vps_counter.GetDtStd()).count();
|
||||
const double frame_time =
|
||||
DT_ms(m_fps_counter.GetDtAvg() + 2 * m_fps_counter.GetDtStd()).count();
|
||||
const double audio_latency = DT_ms(m_audio_latency_counter.GetDtAvg()).count();
|
||||
|
||||
const double target_max_time = 2.0 * std::max({vblank_time, frame_time, audio_latency});
|
||||
const double a =
|
||||
std::max(0.0, 1.0 - std::exp(-4.0 * (DT_s(m_fps_counter.GetLastRawDt()) /
|
||||
DT_s(m_fps_counter.GetSampleWindow()))));
|
||||
|
|
@ -176,6 +190,7 @@ void PerformanceMetrics::DrawImGuiStats(const float backbuffer_scale)
|
|||
ImPlot::SetupAxisTicks(ImAxis_Y1, tick_marks.data(), num_ticks);
|
||||
ImPlot::SetupAxesLimits(0, total_frame_time, 0, m_graph_max_time, ImGuiCond_Always);
|
||||
ImPlot::SetupLegend(ImPlotLocation_SouthEast, ImPlotLegendFlags_None);
|
||||
m_audio_latency_counter.ImPlotPlotLines("Audio Latency (ms)");
|
||||
m_vps_counter.ImPlotPlotLines("V-Blank (ms)");
|
||||
m_fps_counter.ImPlotPlotLines("Frame (ms)");
|
||||
ImPlot::EndPlot();
|
||||
|
|
|
|||
|
|
@ -30,6 +30,7 @@ public:
|
|||
void CountFrame();
|
||||
void CountVBlank();
|
||||
|
||||
void CountAudioLatency(DT latency);
|
||||
void CountThrottleSleep(DT sleep);
|
||||
void CountPerformanceMarker(Core::System& system, s64 cyclesLate);
|
||||
|
||||
|
|
@ -37,6 +38,7 @@ public:
|
|||
double GetFPS() const;
|
||||
double GetVPS() const;
|
||||
double GetSpeed() const;
|
||||
double GetAudioSpeed() const;
|
||||
double GetMaxSpeed() const;
|
||||
|
||||
double GetLastSpeedDenominator() const;
|
||||
|
|
@ -47,15 +49,17 @@ public:
|
|||
private:
|
||||
PerformanceTracker m_fps_counter{"render_times.txt"};
|
||||
PerformanceTracker m_vps_counter{"vblank_times.txt"};
|
||||
PerformanceTracker m_speed_counter{std::nullopt, 1000000};
|
||||
|
||||
PerformanceTracker m_audio_latency_counter{};
|
||||
PerformanceTracker m_audio_speed_counter{std::nullopt, 128000};
|
||||
|
||||
PerformanceTracker m_speed_counter{std::nullopt, 1280000};
|
||||
PerformanceTracker m_max_speed_counter{std::nullopt, 1280000};
|
||||
|
||||
double m_graph_max_time = 0.0;
|
||||
|
||||
mutable std::shared_mutex m_time_lock;
|
||||
|
||||
u8 m_time_index = 0;
|
||||
std::array<TimePoint, 256> m_real_times{};
|
||||
std::array<TimePoint, 256> m_cpu_times{};
|
||||
TimePoint m_prev_adjusted_time{};
|
||||
DT m_time_sleeping{};
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -47,7 +47,7 @@ void PerformanceTracker::Reset()
|
|||
m_dt_std = std::nullopt;
|
||||
}
|
||||
|
||||
void PerformanceTracker::Count()
|
||||
void PerformanceTracker::Count(std::optional<DT> custom_value, bool value_is_duration)
|
||||
{
|
||||
std::unique_lock lock{m_mutex};
|
||||
|
||||
|
|
@ -57,26 +57,31 @@ void PerformanceTracker::Count()
|
|||
const DT window{GetSampleWindow()};
|
||||
|
||||
const TimePoint time{Clock::now()};
|
||||
const DT diff{time - m_last_time};
|
||||
|
||||
const DT duration{time - m_last_time};
|
||||
const DT value{custom_value.value_or(duration)};
|
||||
const TimeDataPair data_point{value_is_duration ? value : duration, value};
|
||||
m_last_time = time;
|
||||
|
||||
QueuePush(diff);
|
||||
m_dt_total += diff;
|
||||
QueuePush(data_point);
|
||||
m_dt_total += data_point;
|
||||
|
||||
if (m_dt_queue_begin == m_dt_queue_end)
|
||||
m_dt_total -= QueuePop();
|
||||
|
||||
while (window <= m_dt_total - QueueTop())
|
||||
while (window <= m_dt_total.duration - QueueTop().duration)
|
||||
m_dt_total -= QueuePop();
|
||||
|
||||
// Simple Moving Average Throughout the Window
|
||||
m_dt_avg = m_dt_total / QueueSize();
|
||||
const double hz = DT_s(1.0) / m_dt_avg;
|
||||
// We want the average value, so we use the value
|
||||
m_dt_avg = m_dt_total.value / QueueSize();
|
||||
|
||||
// Even though the frequency does not make sense if the value
|
||||
// is not the duration, it is still useful to have the value
|
||||
const double hz = DT_s(QueueSize()) / m_dt_total.value;
|
||||
|
||||
// Exponential Moving Average
|
||||
const DT_s rc = SAMPLE_RC_RATIO * std::min(window, m_dt_total);
|
||||
const double a = 1.0 - std::exp(-(DT_s(diff) / rc));
|
||||
const DT_s rc = SAMPLE_RC_RATIO * window;
|
||||
const double a = 1.0 - std::exp(-(DT_s(data_point.duration) / rc));
|
||||
|
||||
// Sometimes euler averages can break when the average is inf/nan
|
||||
if (std::isfinite(m_hz_avg))
|
||||
|
|
@ -86,7 +91,7 @@ void PerformanceTracker::Count()
|
|||
|
||||
m_dt_std = std::nullopt;
|
||||
|
||||
LogRenderTimeToFile(diff);
|
||||
LogRenderTimeToFile(data_point.value);
|
||||
}
|
||||
|
||||
DT PerformanceTracker::GetSampleWindow() const
|
||||
|
|
@ -121,7 +126,7 @@ DT PerformanceTracker::GetDtStd() const
|
|||
double total = 0.0;
|
||||
for (std::size_t i = m_dt_queue_begin; i != m_dt_queue_end; i = IncrementIndex(i))
|
||||
{
|
||||
double diff = DT_s(m_dt_queue[i] - m_dt_avg).count();
|
||||
double diff = DT_s(m_dt_queue[i].value - m_dt_avg).count();
|
||||
total += diff * diff;
|
||||
}
|
||||
|
||||
|
|
@ -136,7 +141,7 @@ DT PerformanceTracker::GetLastRawDt() const
|
|||
if (QueueEmpty())
|
||||
return DT::zero();
|
||||
|
||||
return QueueBottom();
|
||||
return QueueBottom().value;
|
||||
}
|
||||
|
||||
void PerformanceTracker::ImPlotPlotLines(const char* label) const
|
||||
|
|
@ -152,35 +157,32 @@ void PerformanceTracker::ImPlotPlotLines(const char* label) const
|
|||
const bool quality = QueueSize() < MAX_QUALITY_GRAPH_SIZE;
|
||||
|
||||
const DT update_time = Clock::now() - m_last_time;
|
||||
const float predicted_frame_time = DT_ms(std::max(update_time, QueueBottom())).count();
|
||||
|
||||
std::size_t points = 0;
|
||||
if (quality)
|
||||
{
|
||||
x[points] = 0.f;
|
||||
y[points] = predicted_frame_time;
|
||||
++points;
|
||||
}
|
||||
x[points] = 0.f;
|
||||
y[points] = DT_ms(QueueBottom().value).count();
|
||||
++points;
|
||||
|
||||
x[points] = DT_ms(update_time).count();
|
||||
y[points] = predicted_frame_time;
|
||||
y[points] = y[points - 1];
|
||||
++points;
|
||||
|
||||
const std::size_t begin = DecrementIndex(m_dt_queue_end);
|
||||
const std::size_t end = DecrementIndex(m_dt_queue_begin);
|
||||
for (std::size_t i = begin; i != end; i = DecrementIndex(i))
|
||||
{
|
||||
const float frame_time_ms = DT_ms(m_dt_queue[i]).count();
|
||||
const float frame_duration_ms = DT_ms(m_dt_queue[i].duration).count();
|
||||
const float frame_value_ms = DT_ms(m_dt_queue[i].value).count();
|
||||
|
||||
if (quality)
|
||||
{
|
||||
x[points] = x[points - 1];
|
||||
y[points] = frame_time_ms;
|
||||
y[points] = frame_value_ms;
|
||||
++points;
|
||||
}
|
||||
|
||||
x[points] = x[points - 1] + frame_time_ms;
|
||||
y[points] = frame_time_ms;
|
||||
x[points] = x[points - 1] + frame_duration_ms;
|
||||
y[points] = frame_value_ms;
|
||||
++points;
|
||||
}
|
||||
|
||||
|
|
@ -194,25 +196,25 @@ void PerformanceTracker::QueueClear()
|
|||
m_dt_queue_end = 0;
|
||||
}
|
||||
|
||||
void PerformanceTracker::QueuePush(DT dt)
|
||||
void PerformanceTracker::QueuePush(TimeDataPair dt)
|
||||
{
|
||||
m_dt_queue[m_dt_queue_end] = dt;
|
||||
m_dt_queue_end = IncrementIndex(m_dt_queue_end);
|
||||
}
|
||||
|
||||
const DT& PerformanceTracker::QueuePop()
|
||||
const PerformanceTracker::TimeDataPair& PerformanceTracker::QueuePop()
|
||||
{
|
||||
const std::size_t top = m_dt_queue_begin;
|
||||
m_dt_queue_begin = IncrementIndex(m_dt_queue_begin);
|
||||
return m_dt_queue[top];
|
||||
}
|
||||
|
||||
const DT& PerformanceTracker::QueueTop() const
|
||||
const PerformanceTracker::TimeDataPair& PerformanceTracker::QueueTop() const
|
||||
{
|
||||
return m_dt_queue[m_dt_queue_begin];
|
||||
}
|
||||
|
||||
const DT& PerformanceTracker::QueueBottom() const
|
||||
const PerformanceTracker::TimeDataPair& PerformanceTracker::QueueBottom() const
|
||||
{
|
||||
return m_dt_queue[DecrementIndex(m_dt_queue_end)];
|
||||
}
|
||||
|
|
|
|||
|
|
@ -15,7 +15,7 @@ class PerformanceTracker
|
|||
{
|
||||
private:
|
||||
// Must be powers of 2 for masking to work
|
||||
static constexpr u64 MAX_DT_QUEUE_SIZE = 1UL << 12;
|
||||
static constexpr u64 MAX_DT_QUEUE_SIZE = 1UL << 13;
|
||||
static constexpr u64 MAX_QUALITY_GRAPH_SIZE = 1UL << 8;
|
||||
|
||||
static inline std::size_t IncrementIndex(const std::size_t index)
|
||||
|
|
@ -33,6 +33,31 @@ private:
|
|||
return (end - begin) & (MAX_DT_QUEUE_SIZE - 1);
|
||||
}
|
||||
|
||||
struct TimeDataPair
|
||||
{
|
||||
public:
|
||||
TimeDataPair(DT duration_dt, DT value_dt) : duration{duration_dt}, value{value_dt} {}
|
||||
TimeDataPair(DT duration_dt) : TimeDataPair{duration_dt, duration_dt} {}
|
||||
TimeDataPair() : TimeDataPair{DT::zero()} {}
|
||||
|
||||
TimeDataPair& operator+=(const TimeDataPair& other)
|
||||
{
|
||||
duration += other.duration;
|
||||
value += other.value;
|
||||
return *this;
|
||||
}
|
||||
|
||||
TimeDataPair& operator-=(const TimeDataPair& other)
|
||||
{
|
||||
duration -= other.duration;
|
||||
value -= other.value;
|
||||
return *this;
|
||||
}
|
||||
|
||||
public:
|
||||
DT duration, value;
|
||||
};
|
||||
|
||||
public:
|
||||
PerformanceTracker(const std::optional<std::string> log_name = std::nullopt,
|
||||
const std::optional<s64> sample_window_us = std::nullopt);
|
||||
|
|
@ -45,7 +70,20 @@ public:
|
|||
|
||||
// Functions for recording performance information
|
||||
void Reset();
|
||||
void Count();
|
||||
|
||||
/**
|
||||
* custom_value can be used if you are recording something with it's own DT. For example,
|
||||
* if you are recording the fallback of the throttler or the latency of the frame.
|
||||
*
|
||||
* If a custom_value is not supplied, the value will be set to the time between calls aka,
|
||||
* duration. This is the most common use case of this class, as an FPS counter.
|
||||
*
|
||||
* The boolean value_is_duration should be set to true if the custom DTs you are providing
|
||||
* represent a continuous duration. For example, the present times from a render backend
|
||||
* would set value_is_duration to true. Things like throttler fallback or frame latency
|
||||
* are not continuous, so they should not represent duration.
|
||||
*/
|
||||
void Count(std::optional<DT> custom_value = std::nullopt, bool value_is_duration = false);
|
||||
|
||||
// Functions for reading performance information
|
||||
DT GetSampleWindow() const;
|
||||
|
|
@ -61,13 +99,13 @@ public:
|
|||
|
||||
private: // Functions for managing dt queue
|
||||
inline void QueueClear();
|
||||
inline void QueuePush(DT dt);
|
||||
inline const DT& QueuePop();
|
||||
inline const DT& QueueTop() const;
|
||||
inline const DT& QueueBottom() const;
|
||||
inline void QueuePush(TimeDataPair dt);
|
||||
inline const TimeDataPair& QueuePop();
|
||||
inline const TimeDataPair& QueueTop() const;
|
||||
inline const TimeDataPair& QueueBottom() const;
|
||||
|
||||
std::size_t inline QueueSize() const;
|
||||
bool inline QueueEmpty() const;
|
||||
inline std::size_t QueueSize() const;
|
||||
inline bool QueueEmpty() const;
|
||||
|
||||
// Handle pausing and logging
|
||||
void LogRenderTimeToFile(DT val);
|
||||
|
|
@ -87,8 +125,8 @@ private: // Functions for managing dt queue
|
|||
const std::optional<s64> m_sample_window_us;
|
||||
|
||||
// Queue + Running Total used to calculate average dt
|
||||
DT m_dt_total = DT::zero();
|
||||
std::array<DT, MAX_DT_QUEUE_SIZE> m_dt_queue;
|
||||
TimeDataPair m_dt_total;
|
||||
std::array<TimeDataPair, MAX_DT_QUEUE_SIZE> m_dt_queue;
|
||||
std::size_t m_dt_queue_begin = 0;
|
||||
std::size_t m_dt_queue_end = 0;
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue
Block a user