dolphin/Source/Core/AudioCommon/Mixer.cpp
CasualPokePlayer 4234b25682 Do not directly store input sample rate, rather just store a divisor for that sample rate, with it using a fixed dividend of 54000000 * 2.
This should reduce (but not completely eliminate) gradual audio desyncs in dumps. This also allows for accurate sample rates for the GameCube.
Completely eliminating gradual audio desyncs will require resampling to an integer sample rate, as nothing seems to support a non-integer sample rate.
2022-07-03 15:07:06 -07:00

452 lines
14 KiB
C++

// Copyright 2008 Dolphin Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "AudioCommon/Mixer.h"
#include <algorithm>
#include <cmath>
#include <cstring>
#include "AudioCommon/Enums.h"
#include "Common/ChunkFile.h"
#include "Common/CommonTypes.h"
#include "Common/Logging/Log.h"
#include "Common/Swap.h"
#include "Core/Config/MainSettings.h"
#include "Core/ConfigManager.h"
static u32 DPL2QualityToFrameBlockSize(AudioCommon::DPL2Quality quality)
{
switch (quality)
{
case AudioCommon::DPL2Quality::Lowest:
return 512;
case AudioCommon::DPL2Quality::Low:
return 1024;
case AudioCommon::DPL2Quality::Highest:
return 4096;
default:
return 2048;
}
}
Mixer::Mixer(unsigned int BackendSampleRate)
: m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate),
m_surround_decoder(BackendSampleRate,
DPL2QualityToFrameBlockSize(Config::Get(Config::MAIN_DPL2_QUALITY)))
{
m_config_changed_callback_id = Config::AddConfigChangedCallback([this] { RefreshConfig(); });
RefreshConfig();
INFO_LOG_FMT(AUDIO_INTERFACE, "Mixer is initialized");
}
Mixer::~Mixer()
{
Config::RemoveConfigChangedCallback(m_config_changed_callback_id);
}
void Mixer::DoState(PointerWrap& p)
{
m_dma_mixer.DoState(p);
m_streaming_mixer.DoState(p);
m_wiimote_speaker_mixer.DoState(p);
for (auto& mixer : m_gba_mixers)
mixer.DoState(p);
}
// Executed from sound stream thread
unsigned int Mixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
bool consider_framelimit, float emulationspeed,
int timing_variance)
{
unsigned int currentSample = 0;
// Cache access in non-volatile variable
// This is the only function changing the read value, so it's safe to
// cache it locally although it's written here.
// The writing pointer will be modified outside, but it will only increase,
// so we will just ignore new written data while interpolating.
// Without this cache, the compiler wouldn't be allowed to optimize the
// interpolation loop.
u32 indexR = m_indexR.load();
u32 indexW = m_indexW.load();
// render numleft sample pairs to samples[]
// advance indexR with sample position
// remember fractional offset
float aid_sample_rate =
FIXED_SAMPLE_RATE_DIVIDEND / static_cast<float>(m_input_sample_rate_divisor);
if (consider_framelimit && emulationspeed > 0.0f)
{
float numLeft = static_cast<float>(((indexW - indexR) & INDEX_MASK) / 2);
u32 low_watermark = (FIXED_SAMPLE_RATE_DIVIDEND * timing_variance) /
(static_cast<u64>(m_input_sample_rate_divisor) * 1000);
low_watermark = std::min(low_watermark, MAX_SAMPLES / 2);
m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
float offset = (m_numLeftI - low_watermark) * CONTROL_FACTOR;
if (offset > MAX_FREQ_SHIFT)
offset = MAX_FREQ_SHIFT;
if (offset < -MAX_FREQ_SHIFT)
offset = -MAX_FREQ_SHIFT;
aid_sample_rate = (aid_sample_rate + offset) * emulationspeed;
}
const u32 ratio = (u32)(65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate);
s32 lvolume = m_LVolume.load();
s32 rvolume = m_RVolume.load();
const auto read_buffer = [this](auto index) {
return m_little_endian ? m_buffer[index] : Common::swap16(m_buffer[index]);
};
// TODO: consider a higher-quality resampling algorithm.
for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2)
{
u32 indexR2 = indexR + 2; // next sample
s16 l1 = read_buffer(indexR & INDEX_MASK); // current
s16 l2 = read_buffer(indexR2 & INDEX_MASK); // next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
sampleL = (sampleL * lvolume) >> 8;
sampleL += samples[currentSample + 1];
samples[currentSample + 1] = std::clamp(sampleL, -32767, 32767);
s16 r1 = read_buffer((indexR + 1) & INDEX_MASK); // current
s16 r2 = read_buffer((indexR2 + 1) & INDEX_MASK); // next
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
sampleR = (sampleR * rvolume) >> 8;
sampleR += samples[currentSample];
samples[currentSample] = std::clamp(sampleR, -32767, 32767);
m_frac += ratio;
indexR += 2 * (u16)(m_frac >> 16);
m_frac &= 0xffff;
}
// Actual number of samples written to the buffer without padding.
unsigned int actual_sample_count = currentSample / 2;
// Padding
short s[2];
s[0] = read_buffer((indexR - 1) & INDEX_MASK);
s[1] = read_buffer((indexR - 2) & INDEX_MASK);
s[0] = (s[0] * rvolume) >> 8;
s[1] = (s[1] * lvolume) >> 8;
for (; currentSample < numSamples * 2; currentSample += 2)
{
int sampleR = std::clamp(s[0] + samples[currentSample + 0], -32767, 32767);
int sampleL = std::clamp(s[1] + samples[currentSample + 1], -32767, 32767);
samples[currentSample + 0] = sampleR;
samples[currentSample + 1] = sampleL;
}
// Flush cached variable
m_indexR.store(indexR);
return actual_sample_count;
}
unsigned int Mixer::Mix(short* samples, unsigned int num_samples)
{
if (!samples)
return 0;
memset(samples, 0, num_samples * 2 * sizeof(short));
const float emulation_speed = m_config_emulation_speed;
const int timing_variance = m_config_timing_variance;
if (m_config_audio_stretch)
{
unsigned int available_samples =
std::min(m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples());
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);
for (auto& mixer : m_gba_mixers)
{
mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
timing_variance);
}
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);
}
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);
for (auto& mixer : m_gba_mixers)
mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
m_is_stretching = false;
}
return num_samples;
}
unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
{
if (!num_samples)
return 0;
memset(samples, 0, num_samples * SURROUND_CHANNELS * sizeof(float));
size_t needed_frames = m_surround_decoder.QueryFramesNeededForSurroundOutput(num_samples);
// Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads
// and writes.
size_t available_frames = Mix(m_scratch_buffer.data(), static_cast<u32>(needed_frames));
if (available_frames != needed_frames)
{
ERROR_LOG_FMT(AUDIO, "Error decoding surround frames.");
return 0;
}
m_surround_decoder.PutFrames(m_scratch_buffer.data(), needed_frames);
m_surround_decoder.ReceiveFrames(samples, num_samples);
return num_samples;
}
void Mixer::MixerFifo::PushSamples(const short* samples, unsigned int 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;
// 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);
}
m_indexW.fetch_add(num_samples * 2);
}
void Mixer::PushSamples(const short* samples, unsigned int num_samples)
{
m_dma_mixer.PushSamples(samples, num_samples);
if (m_log_dsp_audio)
{
int 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)
{
m_streaming_mixer.PushSamples(samples, num_samples);
if (m_log_dtk_audio)
{
int 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)
{
short samples_stereo[MAX_SAMPLES * 2];
if (num_samples < MAX_SAMPLES)
{
m_wiimote_speaker_mixer.SetInputSampleRateDivisor(sample_rate_divisor);
for (unsigned int i = 0; i < num_samples; ++i)
{
samples_stereo[i * 2] = samples[i];
samples_stereo[i * 2 + 1] = samples[i];
}
m_wiimote_speaker_mixer.PushSamples(samples_stereo, num_samples);
}
}
void Mixer::PushGBASamples(int device_number, const short* samples, unsigned int num_samples)
{
m_gba_mixers[device_number].PushSamples(samples, num_samples);
}
void Mixer::SetDMAInputSampleRateDivisor(unsigned int rate_divisor)
{
m_dma_mixer.SetInputSampleRateDivisor(rate_divisor);
}
void Mixer::SetStreamInputSampleRateDivisor(unsigned int rate_divisor)
{
m_streaming_mixer.SetInputSampleRateDivisor(rate_divisor);
}
void Mixer::SetGBAInputSampleRateDivisors(int device_number, unsigned int rate_divisor)
{
m_gba_mixers[device_number].SetInputSampleRateDivisor(rate_divisor);
}
void Mixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
{
m_streaming_mixer.SetVolume(lvolume, rvolume);
}
void Mixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
{
m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
}
void Mixer::SetGBAVolume(int device_number, unsigned int lvolume, unsigned int rvolume)
{
m_gba_mixers[device_number].SetVolume(lvolume, rvolume);
}
void Mixer::StartLogDTKAudio(const std::string& filename)
{
if (!m_log_dtk_audio)
{
bool success = m_wave_writer_dtk.Start(filename, m_streaming_mixer.GetInputSampleRateDivisor());
if (success)
{
m_log_dtk_audio = true;
m_wave_writer_dtk.SetSkipSilence(false);
NOTICE_LOG_FMT(AUDIO, "Starting DTK Audio logging");
}
else
{
m_wave_writer_dtk.Stop();
NOTICE_LOG_FMT(AUDIO, "Unable to start DTK Audio logging");
}
}
else
{
WARN_LOG_FMT(AUDIO, "DTK Audio logging has already been started");
}
}
void Mixer::StopLogDTKAudio()
{
if (m_log_dtk_audio)
{
m_log_dtk_audio = false;
m_wave_writer_dtk.Stop();
NOTICE_LOG_FMT(AUDIO, "Stopping DTK Audio logging");
}
else
{
WARN_LOG_FMT(AUDIO, "DTK Audio logging has already been stopped");
}
}
void Mixer::StartLogDSPAudio(const std::string& filename)
{
if (!m_log_dsp_audio)
{
bool success = m_wave_writer_dsp.Start(filename, m_dma_mixer.GetInputSampleRateDivisor());
if (success)
{
m_log_dsp_audio = true;
m_wave_writer_dsp.SetSkipSilence(false);
NOTICE_LOG_FMT(AUDIO, "Starting DSP Audio logging");
}
else
{
m_wave_writer_dsp.Stop();
NOTICE_LOG_FMT(AUDIO, "Unable to start DSP Audio logging");
}
}
else
{
WARN_LOG_FMT(AUDIO, "DSP Audio logging has already been started");
}
}
void Mixer::StopLogDSPAudio()
{
if (m_log_dsp_audio)
{
m_log_dsp_audio = false;
m_wave_writer_dsp.Stop();
NOTICE_LOG_FMT(AUDIO, "Stopping DSP Audio logging");
}
else
{
WARN_LOG_FMT(AUDIO, "DSP Audio logging has already been stopped");
}
}
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);
}
void Mixer::MixerFifo::DoState(PointerWrap& p)
{
p.Do(m_input_sample_rate_divisor);
p.Do(m_LVolume);
p.Do(m_RVolume);
}
void Mixer::MixerFifo::SetInputSampleRateDivisor(unsigned int rate_divisor)
{
m_input_sample_rate_divisor = rate_divisor;
}
unsigned int Mixer::MixerFifo::GetInputSampleRateDivisor() const
{
return m_input_sample_rate_divisor;
}
void Mixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
{
m_LVolume.store(lvolume + (lvolume >> 7));
m_RVolume.store(rvolume + (rvolume >> 7));
}
std::pair<s32, s32> Mixer::MixerFifo::GetVolume() const
{
return std::make_pair(m_LVolume.load(), m_RVolume.load());
}
unsigned int Mixer::MixerFifo::AvailableSamples() 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;
}