dolphin/Source/Core/AudioCommon/Mixer.cpp

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// Copyright 2008 Dolphin Emulator Project
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// Licensed under GPLv2+
// Refer to the license.txt file included.
#include "AudioCommon/Mixer.h"
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#include <cmath>
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#include <cstring>
#include "AudioCommon/DPL2Decoder.h"
#include "Common/CommonTypes.h"
#include "Common/Logging/Log.h"
#include "Common/MathUtil.h"
#include "Common/Swap.h"
#include "Core/ConfigManager.h"
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Mixer::Mixer(unsigned int BackendSampleRate)
: m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate)
{
INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
DPL2Reset();
}
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Mixer::~Mixer()
{
}
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// Executed from sound stream thread
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unsigned int Mixer::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;
}
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unsigned int Mixer::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());
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m_scratch_buffer.fill(0);
m_dma_mixer.Mix(m_scratch_buffer.data(), available_samples, false);
m_streaming_mixer.Mix(m_scratch_buffer.data(), available_samples, false);
m_wiimote_speaker_mixer.Mix(m_scratch_buffer.data(), available_samples, false);
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);
m_streaming_mixer.Mix(samples, num_samples, true);
m_wiimote_speaker_mixer.Mix(samples, num_samples, true);
m_is_stretching = false;
}
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return num_samples;
}
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unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
{
if (!num_samples)
return 0;
memset(samples, 0, num_samples * 6 * sizeof(float));
// Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads and
// writes.
unsigned int available_samples = Mix(m_scratch_buffer.data(), num_samples);
for (size_t i = 0; i < static_cast<size_t>(available_samples) * 2; ++i)
{
m_float_conversion_buffer[i] =
m_scratch_buffer[i] / static_cast<float>(std::numeric_limits<short>::max());
}
DPL2Decode(m_float_conversion_buffer.data(), available_samples, samples);
return available_samples;
}
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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);
}
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void Mixer::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);
}
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void Mixer::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);
}
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void Mixer::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);
}
}
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void Mixer::SetDMAInputSampleRate(unsigned int rate)
{
m_dma_mixer.SetInputSampleRate(rate);
}
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void Mixer::SetStreamInputSampleRate(unsigned int rate)
{
m_streaming_mixer.SetInputSampleRate(rate);
}
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void Mixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
{
m_streaming_mixer.SetVolume(lvolume, rvolume);
}
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void Mixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
{
m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
}
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void Mixer::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");
}
}
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void Mixer::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");
}
}
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void Mixer::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");
}
}
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void Mixer::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");
}
}
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void Mixer::MixerFifo::SetInputSampleRate(unsigned int rate)
{
m_input_sample_rate = rate;
}
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unsigned int Mixer::MixerFifo::GetInputSampleRate() const
{
return m_input_sample_rate;
}
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void Mixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
{
m_LVolume.store(lvolume + (lvolume >> 7));
m_RVolume.store(rvolume + (rvolume >> 7));
}
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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)
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return 0; // Mixer::MixerFifo::Mix always keeps one sample in the buffer.
return (samples_in_fifo - 1) * m_mixer->m_sampleRate / m_input_sample_rate;
}