mirror of
https://github.com/dolphin-emu/dolphin.git
synced 2024-11-15 13:57:57 -07:00
397 lines
12 KiB
C++
397 lines
12 KiB
C++
// Copyright 2008 Dolphin Emulator Project
|
|
// Licensed under GPLv2+
|
|
// Refer to the license.txt file included.
|
|
|
|
#include "AudioCommon/Mixer.h"
|
|
|
|
#include <cmath>
|
|
#include <cstring>
|
|
|
|
#include "Common/CommonTypes.h"
|
|
#include "Common/Logging/Log.h"
|
|
#include "Common/MathUtil.h"
|
|
#include "Common/Swap.h"
|
|
#include "Core/ConfigManager.h"
|
|
|
|
CMixer::CMixer(unsigned int BackendSampleRate) : m_sampleRate(BackendSampleRate)
|
|
{
|
|
INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
|
|
|
|
m_sound_touch.setChannels(2);
|
|
m_sound_touch.setSampleRate(BackendSampleRate);
|
|
m_sound_touch.setPitch(1.0);
|
|
m_sound_touch.setTempo(1.0);
|
|
m_sound_touch.setSetting(SETTING_USE_QUICKSEEK, 0);
|
|
m_sound_touch.setSetting(SETTING_SEQUENCE_MS, 62);
|
|
m_sound_touch.setSetting(SETTING_SEEKWINDOW_MS, 28);
|
|
m_sound_touch.setSetting(SETTING_OVERLAP_MS, 8);
|
|
}
|
|
|
|
CMixer::~CMixer()
|
|
{
|
|
}
|
|
|
|
// Executed from sound stream thread
|
|
unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
|
|
bool consider_framelimit)
|
|
{
|
|
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 emulationspeed = SConfig::GetInstance().m_EmulationSpeed;
|
|
float aid_sample_rate = static_cast<float>(m_input_sample_rate);
|
|
if (consider_framelimit && emulationspeed > 0.0f)
|
|
{
|
|
float numLeft = static_cast<float>(((indexW - indexR) & INDEX_MASK) / 2);
|
|
|
|
u32 low_waterwark = m_input_sample_rate * SConfig::GetInstance().iTimingVariance / 1000;
|
|
low_waterwark = std::min(low_waterwark, MAX_SAMPLES / 2);
|
|
|
|
m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
|
|
float offset = (m_numLeftI - low_waterwark) * 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();
|
|
|
|
// 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 = Common::swap16(m_buffer[indexR & INDEX_MASK]); // current
|
|
s16 l2 = Common::swap16(m_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] = MathUtil::Clamp(sampleL, -32767, 32767);
|
|
|
|
s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]); // current
|
|
s16 r2 = Common::swap16(m_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] = MathUtil::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] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
|
|
s[1] = Common::swap16(m_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 = MathUtil::Clamp(s[0] + samples[currentSample + 0], -32767, 32767);
|
|
int sampleL = MathUtil::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 CMixer::Mix(short* samples, unsigned int num_samples)
|
|
{
|
|
if (!samples)
|
|
return 0;
|
|
|
|
memset(samples, 0, num_samples * 2 * sizeof(short));
|
|
|
|
if (SConfig::GetInstance().m_audio_stretch)
|
|
{
|
|
unsigned int available_samples =
|
|
std::min(m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples());
|
|
|
|
m_stretch_buffer.fill(0);
|
|
|
|
m_dma_mixer.Mix(m_stretch_buffer.data(), available_samples, false);
|
|
m_streaming_mixer.Mix(m_stretch_buffer.data(), available_samples, false);
|
|
m_wiimote_speaker_mixer.Mix(m_stretch_buffer.data(), available_samples, false);
|
|
|
|
if (!m_is_stretching)
|
|
{
|
|
m_sound_touch.clear();
|
|
m_is_stretching = true;
|
|
}
|
|
StretchAudio(m_stretch_buffer.data(), available_samples, samples, num_samples);
|
|
}
|
|
else
|
|
{
|
|
m_dma_mixer.Mix(samples, num_samples, true);
|
|
m_streaming_mixer.Mix(samples, num_samples, true);
|
|
m_wiimote_speaker_mixer.Mix(samples, num_samples, true);
|
|
m_is_stretching = false;
|
|
}
|
|
|
|
return num_samples;
|
|
}
|
|
|
|
void CMixer::StretchAudio(const short* in, unsigned int num_in, short* out, unsigned int num_out)
|
|
{
|
|
const double time_delta = static_cast<double>(num_out) / m_sampleRate; // seconds
|
|
|
|
// We were given actual_samples number of samples, and num_samples were requested from us.
|
|
double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
|
|
|
|
const double max_latency = SConfig::GetInstance().m_audio_stretch_max_latency;
|
|
const double max_backlog = m_sampleRate * max_latency / 1000.0 / m_stretch_ratio;
|
|
const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
|
|
if (backlog_fullness > 5.0)
|
|
{
|
|
// Too many samples in backlog: Don't push anymore on
|
|
num_in = 0;
|
|
}
|
|
|
|
// We ideally want the backlog to be about 50% full.
|
|
// This gives some headroom both ways to prevent underflow and overflow.
|
|
// We tweak current_ratio to encourage this.
|
|
constexpr double tweak_time_scale = 0.5; // seconds
|
|
current_ratio *= 1.0 + 2.0 * (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
|
|
|
|
// This low-pass filter smoothes out variance in the calculated stretch ratio.
|
|
// The time-scale determines how responsive this filter is.
|
|
constexpr double lpf_time_scale = 1.0; // seconds
|
|
const double m_lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
|
|
m_stretch_ratio += m_lpf_gain * (current_ratio - m_stretch_ratio);
|
|
|
|
// Place a lower limit of 10% speed. When a game boots up, there will be
|
|
// many silence samples. These do not need to be timestretched.
|
|
m_stretch_ratio = std::max(m_stretch_ratio, 0.1);
|
|
m_sound_touch.setTempo(m_stretch_ratio);
|
|
|
|
DEBUG_LOG(AUDIO, "Audio stretching: samples:%u/%u ratio:%f backlog:%f gain: %f", num_in, num_out,
|
|
m_stretch_ratio, backlog_fullness, m_lpf_gain);
|
|
|
|
m_sound_touch.putSamples(in, num_in);
|
|
|
|
const size_t samples_received = m_sound_touch.receiveSamples(out, num_out);
|
|
|
|
if (samples_received != 0)
|
|
{
|
|
m_last_stretched_sample[0] = out[samples_received * 2 - 2];
|
|
m_last_stretched_sample[1] = out[samples_received * 2 - 1];
|
|
}
|
|
|
|
// Preform padding if we've run out of samples.
|
|
for (size_t i = samples_received; i < num_out; i++)
|
|
{
|
|
out[i * 2 + 0] = m_last_stretched_sample[0];
|
|
out[i * 2 + 1] = m_last_stretched_sample[1];
|
|
}
|
|
}
|
|
|
|
void CMixer::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 CMixer::PushSamples(const short* samples, unsigned int num_samples)
|
|
{
|
|
m_dma_mixer.PushSamples(samples, num_samples);
|
|
int sample_rate = m_dma_mixer.GetInputSampleRate();
|
|
if (m_log_dsp_audio)
|
|
m_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples, sample_rate);
|
|
}
|
|
|
|
void CMixer::PushStreamingSamples(const short* samples, unsigned int num_samples)
|
|
{
|
|
m_streaming_mixer.PushSamples(samples, num_samples);
|
|
int sample_rate = m_streaming_mixer.GetInputSampleRate();
|
|
if (m_log_dtk_audio)
|
|
m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples, sample_rate);
|
|
}
|
|
|
|
void CMixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
|
|
unsigned int sample_rate)
|
|
{
|
|
short samples_stereo[MAX_SAMPLES * 2];
|
|
|
|
if (num_samples < MAX_SAMPLES)
|
|
{
|
|
m_wiimote_speaker_mixer.SetInputSampleRate(sample_rate);
|
|
|
|
for (unsigned int i = 0; i < num_samples; ++i)
|
|
{
|
|
samples_stereo[i * 2] = Common::swap16(samples[i]);
|
|
samples_stereo[i * 2 + 1] = Common::swap16(samples[i]);
|
|
}
|
|
|
|
m_wiimote_speaker_mixer.PushSamples(samples_stereo, num_samples);
|
|
}
|
|
}
|
|
|
|
void CMixer::SetDMAInputSampleRate(unsigned int rate)
|
|
{
|
|
m_dma_mixer.SetInputSampleRate(rate);
|
|
}
|
|
|
|
void CMixer::SetStreamInputSampleRate(unsigned int rate)
|
|
{
|
|
m_streaming_mixer.SetInputSampleRate(rate);
|
|
}
|
|
|
|
void CMixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
|
|
{
|
|
m_streaming_mixer.SetVolume(lvolume, rvolume);
|
|
}
|
|
|
|
void CMixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
|
|
{
|
|
m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
|
|
}
|
|
|
|
void CMixer::StartLogDTKAudio(const std::string& filename)
|
|
{
|
|
if (!m_log_dtk_audio)
|
|
{
|
|
bool success = m_wave_writer_dtk.Start(filename, m_streaming_mixer.GetInputSampleRate());
|
|
if (success)
|
|
{
|
|
m_log_dtk_audio = true;
|
|
m_wave_writer_dtk.SetSkipSilence(false);
|
|
NOTICE_LOG(AUDIO, "Starting DTK Audio logging");
|
|
}
|
|
else
|
|
{
|
|
m_wave_writer_dtk.Stop();
|
|
NOTICE_LOG(AUDIO, "Unable to start DTK Audio logging");
|
|
}
|
|
}
|
|
else
|
|
{
|
|
WARN_LOG(AUDIO, "DTK Audio logging has already been started");
|
|
}
|
|
}
|
|
|
|
void CMixer::StopLogDTKAudio()
|
|
{
|
|
if (m_log_dtk_audio)
|
|
{
|
|
m_log_dtk_audio = false;
|
|
m_wave_writer_dtk.Stop();
|
|
NOTICE_LOG(AUDIO, "Stopping DTK Audio logging");
|
|
}
|
|
else
|
|
{
|
|
WARN_LOG(AUDIO, "DTK Audio logging has already been stopped");
|
|
}
|
|
}
|
|
|
|
void CMixer::StartLogDSPAudio(const std::string& filename)
|
|
{
|
|
if (!m_log_dsp_audio)
|
|
{
|
|
bool success = m_wave_writer_dsp.Start(filename, m_dma_mixer.GetInputSampleRate());
|
|
if (success)
|
|
{
|
|
m_log_dsp_audio = true;
|
|
m_wave_writer_dsp.SetSkipSilence(false);
|
|
NOTICE_LOG(AUDIO, "Starting DSP Audio logging");
|
|
}
|
|
else
|
|
{
|
|
m_wave_writer_dsp.Stop();
|
|
NOTICE_LOG(AUDIO, "Unable to start DSP Audio logging");
|
|
}
|
|
}
|
|
else
|
|
{
|
|
WARN_LOG(AUDIO, "DSP Audio logging has already been started");
|
|
}
|
|
}
|
|
|
|
void CMixer::StopLogDSPAudio()
|
|
{
|
|
if (m_log_dsp_audio)
|
|
{
|
|
m_log_dsp_audio = false;
|
|
m_wave_writer_dsp.Stop();
|
|
NOTICE_LOG(AUDIO, "Stopping DSP Audio logging");
|
|
}
|
|
else
|
|
{
|
|
WARN_LOG(AUDIO, "DSP Audio logging has already been stopped");
|
|
}
|
|
}
|
|
|
|
void CMixer::MixerFifo::SetInputSampleRate(unsigned int rate)
|
|
{
|
|
m_input_sample_rate = rate;
|
|
}
|
|
|
|
unsigned int CMixer::MixerFifo::GetInputSampleRate() const
|
|
{
|
|
return m_input_sample_rate;
|
|
}
|
|
|
|
void CMixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
|
|
{
|
|
m_LVolume.store(lvolume + (lvolume >> 7));
|
|
m_RVolume.store(rvolume + (rvolume >> 7));
|
|
}
|
|
|
|
unsigned int CMixer::MixerFifo::AvailableSamples() const
|
|
{
|
|
unsigned int samples_in_fifo = ((m_indexW.load() - m_indexR.load()) & INDEX_MASK) / 2;
|
|
if (samples_in_fifo <= 1)
|
|
return 0; // CMixer::MixerFifo::Mix always keeps one sample in the buffer.
|
|
return (samples_in_fifo - 1) * m_mixer->m_sampleRate / m_input_sample_rate;
|
|
}
|