453 lines
14 KiB
C++
453 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;
|
|
}
|