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High Quality Resampler, using FIR filter
For more information: https://docs.google.com/document/d/1tBEgsJh7QiwNwepXI0eobfK3U8LkJButSyeuFt1degM/edit?usp=sharing removed: SSE includes (not used) added: 16bit -> float -> 16bit functions added: linear interpolator and high-quality (windowed-sinc) interpolator functions (including Resampler class) added: dithering changed: renamed variables and reformatted a few things to fit with style guide (braces, #include->const) changed: use s16, u16, s32, u32 etc changed: store samples and do all computations as floats changed: volume from 0 - 255
This commit is contained in:
parent
1a3bc8f286
commit
e864521182
@ -15,35 +15,89 @@
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// UGLINESS
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#include "Core/PowerPC/PowerPC.h"
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#if _M_SSE >= 0x301 && !(defined __GNUC__ && !defined __SSSE3__)
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#include <tmmintrin.h>
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#ifndef M_PI
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#define M_PI 3.14159265358979323846
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#endif
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// Executed from sound stream thread
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unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples, bool consider_framelimit)
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const float CMixer::LOW_WATERMARK = 1280;
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const float CMixer::MAX_FREQ_SHIFT = 200;
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const float CMixer::CONTROL_FACTOR = 0.2f;
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const float CMixer::CONTROL_AVG = 32;
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const double CMixer::Resampler::LOWPASS_ROLLOFF = 0.9;
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const double CMixer::Resampler::KAISER_BETA = 6.0;
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const double CMixer::Resampler::BESSEL_EPSILON = 1e-21;
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void CMixer::LinearMixerFifo::Interpolate(u32 left_input_index, float* left_output, float* right_output)
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{
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unsigned int currentSample = 0;
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*left_output = (1 - m_fraction) * m_float_buffer[left_input_index & INDEX_MASK]
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+ m_fraction * m_float_buffer[(left_input_index + 2) & INDEX_MASK];
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*right_output = (1 - m_fraction) * m_float_buffer[(left_input_index + 1) & INDEX_MASK]
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+ m_fraction * m_float_buffer[(left_input_index + 3) & INDEX_MASK];
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}
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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 = Common::AtomicLoad(m_indexR);
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u32 indexW = Common::AtomicLoad(m_indexW);
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//see https://ccrma.stanford.edu/~jos/resample/Implementation.html
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void CMixer::WindowedSincMixerFifo::Interpolate(u32 left_input_index, float* left_output, float* right_output)
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{
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double left_temp = 0, right_temp = 0;
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float numLeft = (float)(((indexW - indexR) & INDEX_MASK) / 2);
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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) offset = MAX_FREQ_SHIFT;
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if (offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;
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// left wing of filter
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double left_wing_fraction = (m_fraction * Resampler::SAMPLES_PER_CROSSING);
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u32 left_wing_index = (u32) left_wing_fraction;
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left_wing_fraction -= left_wing_index;
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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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const Resampler& resampler = m_mixer->m_resampler;
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u32 current_index = left_input_index;
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while (left_wing_index < resampler.m_lowpass_filter.size())
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{
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double impulse = resampler.m_lowpass_filter[left_wing_index];
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impulse += resampler.m_lowpass_delta[left_wing_index] * left_wing_fraction;
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left_temp += (float) m_float_buffer[current_index & INDEX_MASK] * impulse;
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right_temp += (float) m_float_buffer[(current_index + 1) & INDEX_MASK] * impulse;
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left_wing_index += Resampler::SAMPLES_PER_CROSSING;
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current_index -= 2;
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}
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// right wing of filter
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double right_wing_fraction = (1 - m_fraction) * Resampler::SAMPLES_PER_CROSSING;
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u32 right_wing_index = ((u32) right_wing_fraction) % Resampler::SAMPLES_PER_CROSSING;
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right_wing_fraction -= right_wing_index;
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// we already used read_index for left wing
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current_index = left_input_index + 2;
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while (right_wing_index < resampler.m_lowpass_filter.size())
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{
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double impulse = resampler.m_lowpass_filter[right_wing_index];
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impulse += resampler.m_lowpass_delta[right_wing_index] * right_wing_fraction;
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left_temp += (float) m_float_buffer[current_index & INDEX_MASK] * impulse;
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right_temp += (float) m_float_buffer[(current_index + 1) & INDEX_MASK] * impulse;
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right_wing_index += Resampler::SAMPLES_PER_CROSSING;
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current_index += 2;
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}
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*left_output = (float) left_temp;
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*right_output = (float) right_temp;
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}
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void CMixer::MixerFifo::Mix(std::vector<float>& samples, u32 numSamples, bool consider_framelimit)
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{
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u32 current_sample = 0;
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// Cache access in non-volatile variable so interpolation loop can be optimized
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u32 read_index = Common::AtomicLoad(m_read_index);
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const u32 write_index = Common::AtomicLoad(m_write_index);
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// Sync input rate by fifo size
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float num_left = (float) (((write_index - read_index) & INDEX_MASK) / 2);
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m_num_left_i = (num_left + m_num_left_i * (CONTROL_AVG - 1)) / CONTROL_AVG;
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float offset = (m_num_left_i - LOW_WATERMARK) * CONTROL_FACTOR;
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MathUtil::Clamp(&offset, -MAX_FREQ_SHIFT, MAX_FREQ_SHIFT);
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// adjust framerate with framelimit
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u32 framelimit = SConfig::GetInstance().m_Framelimit;
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float aid_sample_rate = m_input_sample_rate + offset;
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if (consider_framelimit && framelimit > 1)
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@ -51,134 +105,146 @@ unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples, boo
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aid_sample_rate = aid_sample_rate * (framelimit - 1) * 5 / VideoInterface::TargetRefreshRate;
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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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// ratio = 1 / upscale_factor = stepsize for each sample
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// e.g. going from 32khz to 48khz is 1 / (3 / 2) = 2 / 3
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// note because of syncing and framelimit, ratio will rarely be exactly 2 / 3
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float ratio = aid_sample_rate / (float) m_mixer->m_sample_rate;
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s32 lvolume = m_LVolume;
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s32 rvolume = m_RVolume;
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float l_volume = (float) m_lvolume / 255.f;
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float r_volume = (float) m_rvolume / 255.f;
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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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// for each output sample pair (left and right),
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// linear interpolate between current and next sample
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// increment output sample position
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// increment input sample position by ratio, store fraction
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// QUESTION: do we need to check for NUM_CROSSINGS samples before we interpolate?
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// seems to work fine as is
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for (; current_sample < numSamples * 2 && ((write_index - read_index) & INDEX_MASK) > 0; current_sample += 2)
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{
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u32 indexR2 = indexR + 2; //next sample
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float l_output, r_output;
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s16 l1 = Common::swap16(m_buffer[indexR & INDEX_MASK]); //current
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s16 l2 = Common::swap16(m_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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MathUtil::Clamp(&sampleL, -32767, 32767);
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samples[currentSample + 1] = sampleL;
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Interpolate(read_index, &l_output, &r_output);
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s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]); //current
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s16 r2 = Common::swap16(m_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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MathUtil::Clamp(&sampleR, -32767, 32767);
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samples[currentSample] = sampleR;
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samples[current_sample + 1] += l_volume * l_output;
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samples[current_sample] += r_volume * r_output;
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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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m_fraction += ratio;
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read_index += 2 * (s32) m_fraction;
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m_fraction = m_fraction - (s32) m_fraction;
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}
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// Padding
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short s[2];
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s[0] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
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s[1] = Common::swap16(m_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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// pad output if not enough input samples
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float s[2];
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s[0] = m_float_buffer[(read_index - 1) & INDEX_MASK] * r_volume;
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s[1] = m_float_buffer[(read_index - 2) & INDEX_MASK] * l_volume;
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for (; current_sample < numSamples * 2; current_sample += 2)
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{
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int sampleR = s[0] + samples[currentSample];
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MathUtil::Clamp(&sampleR, -32767, 32767);
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samples[currentSample] = sampleR;
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int sampleL = s[1] + samples[currentSample + 1];
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MathUtil::Clamp(&sampleL, -32767, 32767);
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samples[currentSample + 1] = sampleL;
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samples[current_sample] += s[0];
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samples[current_sample + 1] += s[1];
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}
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// Flush cached variable
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Common::AtomicStore(m_indexR, indexR);
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return numSamples;
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// update read index
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Common::AtomicStore(m_read_index, read_index);
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}
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unsigned int CMixer::Mix(short* samples, unsigned int num_samples, bool consider_framelimit)
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// we NEED dithering going from float -> 16bit
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void CMixer::TriangleDither(float* l_sample, float* r_sample)
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{
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float left_dither = DITHER_NOISE;
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float right_dither = DITHER_NOISE;
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*l_sample = (*l_sample) + left_dither - m_l_dither_prev;
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*r_sample = (*r_sample) + right_dither - m_r_dither_prev;
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m_l_dither_prev = left_dither;
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m_r_dither_prev = right_dither;
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}
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u32 CMixer::Mix(s16* samples, u32 num_samples, bool consider_framelimit)
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{
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if (!samples)
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return 0;
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std::lock_guard<std::mutex> lk(m_csMixing);
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memset(samples, 0, num_samples * 2 * sizeof(short));
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std::lock_guard<std::mutex> lk(m_cs_mixing);
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if (PowerPC::GetState() != PowerPC::CPU_RUNNING)
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{
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// Silence
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memset(samples, 0, num_samples * 2 * sizeof(s16));
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return num_samples;
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}
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m_dma_mixer.Mix(samples, num_samples, consider_framelimit);
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m_streaming_mixer.Mix(samples, num_samples, consider_framelimit);
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m_wiimote_speaker_mixer.Mix(samples, num_samples, consider_framelimit);
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// reset float output buffer
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m_output_buffer.resize(num_samples * 2);
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std::fill_n(m_output_buffer.begin(), num_samples * 2, 0.f);
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m_dma_mixer.Mix(m_output_buffer, num_samples, consider_framelimit);
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m_streaming_mixer.Mix(m_output_buffer, num_samples, consider_framelimit);
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m_wiimote_speaker_mixer.Mix(m_output_buffer, num_samples, consider_framelimit);
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// dither and clamp
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for (u32 i = 0; i < num_samples * 2; i += 2)
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{
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float l_output = m_output_buffer[i + 1];
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float r_output = m_output_buffer[i];
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TriangleDither(&m_output_buffer[i + 1], &m_output_buffer[i]);
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MathUtil::Clamp(&l_output, -1.f, 1.f);
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samples[i + 1] = FloatToSigned16(l_output);
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MathUtil::Clamp(&r_output, -1.f, 1.f);
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samples[i] = FloatToSigned16(r_output);
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}
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return num_samples;
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}
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void CMixer::MixerFifo::PushSamples(const short *samples, unsigned int num_samples)
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void CMixer::MixerFifo::PushSamples(const s16* samples, u32 num_samples)
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{
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// Cache access in non-volatile variable
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// indexR isn't allowed to cache in the audio throttling loop as it
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// needs to get updates to not deadlock.
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u32 indexW = Common::AtomicLoad(m_indexW);
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u32 current_write_index = Common::AtomicLoad(m_write_index);
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// Check if we have enough free space
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// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
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if (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
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if (num_samples * 2 + ((current_write_index - Common::AtomicLoad(m_read_index)) & INDEX_MASK) >= MAX_SAMPLES * 2)
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return;
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// AyuanX: Actual re-sampling work has been moved to sound thread
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// to alleviate the workload on main thread
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// and we simply store raw data here to make fast mem copy
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int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
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if (over_bytes > 0)
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// convert to float while copying to buffer
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for (u32 i = 0; i < num_samples * 2; ++i)
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{
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memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
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memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
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}
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else
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{
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memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
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m_float_buffer[(current_write_index + i) & INDEX_MASK] = Signed16ToFloat(Common::swap16(samples[i]));
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}
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Common::AtomicAdd(m_indexW, num_samples * 2);
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Common::AtomicAdd(m_write_index, num_samples * 2);
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return;
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}
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void CMixer::PushSamples(const short *samples, unsigned int num_samples)
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void CMixer::PushSamples(const s16* samples, u32 num_samples)
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{
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m_dma_mixer.PushSamples(samples, num_samples);
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if (m_log_dsp_audio)
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g_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples);
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}
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void CMixer::PushStreamingSamples(const short *samples, unsigned int num_samples)
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void CMixer::PushStreamingSamples(const s16* samples, u32 num_samples)
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{
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m_streaming_mixer.PushSamples(samples, num_samples);
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if (m_log_dtk_audio)
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g_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples);
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}
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void CMixer::PushWiimoteSpeakerSamples(const short *samples, unsigned int num_samples, unsigned int sample_rate)
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void CMixer::PushWiimoteSpeakerSamples(const s16* samples, u32 num_samples, u32 sample_rate)
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{
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short samples_stereo[MAX_SAMPLES * 2];
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s16 samples_stereo[MAX_SAMPLES * 2];
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if (num_samples < MAX_SAMPLES)
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{
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m_wiimote_speaker_mixer.SetInputSampleRate(sample_rate);
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for (unsigned int i = 0; i < num_samples; ++i)
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for (u32 i = 0; i < num_samples; ++i)
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{
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samples_stereo[i * 2] = Common::swap16(samples[i]);
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samples_stereo[i * 2 + 1] = Common::swap16(samples[i]);
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@ -188,33 +254,90 @@ void CMixer::PushWiimoteSpeakerSamples(const short *samples, unsigned int num_sa
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}
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}
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void CMixer::SetDMAInputSampleRate(unsigned int rate)
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void CMixer::SetDMAInputSampleRate(u32 rate)
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{
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m_dma_mixer.SetInputSampleRate(rate);
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}
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void CMixer::SetStreamInputSampleRate(unsigned int rate)
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void CMixer::SetStreamInputSampleRate(u32 rate)
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{
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m_streaming_mixer.SetInputSampleRate(rate);
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}
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void CMixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
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void CMixer::SetStreamingVolume(u32 lvolume, u32 rvolume)
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{
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m_streaming_mixer.SetVolume(lvolume, rvolume);
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}
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void CMixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
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void CMixer::SetWiimoteSpeakerVolume(u32 lvolume, u32 rvolume)
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{
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m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
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}
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void CMixer::MixerFifo::SetInputSampleRate(unsigned int rate)
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void CMixer::MixerFifo::SetInputSampleRate(u32 rate)
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{
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m_input_sample_rate = rate;
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}
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void CMixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
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void CMixer::MixerFifo::SetVolume(u32 lvolume, u32 rvolume)
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{
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m_LVolume = lvolume + (lvolume >> 7);
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m_RVolume = rvolume + (rvolume >> 7);
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m_lvolume = lvolume;
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m_rvolume = rvolume;
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}
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void CMixer::MixerFifo::GetVolume(u32* lvolume, u32* rvolume) const
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{
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*lvolume = m_lvolume;
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*rvolume = m_rvolume;
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}
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// I_0(x) = summation((((x/2)^k) / k!)^2) for k from 0 to Infinity
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double CMixer::Resampler::ModBessel0th(const double x)
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{
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double sum = 1;
|
||||
s32 factorial_store = 1;
|
||||
double half_x = x / 2.f;
|
||||
double previous = 1;
|
||||
do
|
||||
{
|
||||
double temp = half_x / (double) factorial_store;
|
||||
temp *= temp;
|
||||
previous *= temp;
|
||||
sum += previous;
|
||||
factorial_store++;
|
||||
} while (previous >= BESSEL_EPSILON * sum);
|
||||
return sum;
|
||||
}
|
||||
|
||||
// one wing of FIR by using sinc * Kaiser window
|
||||
void CMixer::Resampler::PopulateFilterCoeff()
|
||||
{
|
||||
// Generate sinc table
|
||||
m_lowpass_filter[0] = LOWPASS_ROLLOFF;
|
||||
for (u32 i = 1; i < m_lowpass_filter.size(); ++i)
|
||||
{
|
||||
double temp = M_PI * (double) i / SAMPLES_PER_CROSSING;
|
||||
m_lowpass_filter[i] = sin(temp * LOWPASS_ROLLOFF) / temp;
|
||||
}
|
||||
|
||||
// use a Kaiser window
|
||||
// https://ccrma.stanford.edu/~jos/sasp/Kaiser_Window.html
|
||||
//
|
||||
double I0_beta = 1.0 / ModBessel0th(KAISER_BETA);
|
||||
double inside = 1.0 / (m_lowpass_filter.size() - 1);
|
||||
for (u32 i = 1; i < m_lowpass_filter.size(); ++i)
|
||||
{
|
||||
double temp = (double) i * inside;
|
||||
temp = 1.0 - temp * temp;
|
||||
temp = (temp < 0) ? 0 : temp;
|
||||
m_lowpass_filter[i] *= ModBessel0th(KAISER_BETA * sqrt(temp)) * I0_beta;
|
||||
}
|
||||
|
||||
// store deltas in delta table for faster lookup to interpolate impulse
|
||||
for (u32 i = 0; i < m_lowpass_filter.size() - 1; ++i)
|
||||
{
|
||||
m_lowpass_delta[i] = m_lowpass_filter[i + 1] - m_lowpass_filter[i];
|
||||
}
|
||||
m_lowpass_delta.back() = -1 * m_lowpass_filter.back();
|
||||
|
||||
}
|
@ -4,50 +4,56 @@
|
||||
|
||||
#pragma once
|
||||
|
||||
#include <array>
|
||||
#include <mutex>
|
||||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
#include "AudioCommon/WaveFile.h"
|
||||
|
||||
// 16 bit Stereo
|
||||
#define MAX_SAMPLES (1024 * 2) // 64ms
|
||||
#define INDEX_MASK (MAX_SAMPLES * 2 - 1)
|
||||
|
||||
#define LOW_WATERMARK 1280 // 40 ms
|
||||
#define MAX_FREQ_SHIFT 200 // per 32000 Hz
|
||||
#define CONTROL_FACTOR 0.2f // in freq_shift per fifo size offset
|
||||
#define CONTROL_AVG 32
|
||||
|
||||
class CMixer {
|
||||
// Dither define
|
||||
#define DITHER_NOISE (rand() / (float) RAND_MAX - 0.5f)
|
||||
|
||||
class CMixer
|
||||
{
|
||||
public:
|
||||
CMixer(unsigned int BackendSampleRate)
|
||||
CMixer(u32 BackendSampleRate)
|
||||
: m_dma_mixer(this, 32000)
|
||||
, m_streaming_mixer(this, 48000)
|
||||
, m_wiimote_speaker_mixer(this, 3000)
|
||||
, m_sampleRate(BackendSampleRate)
|
||||
, m_sample_rate(BackendSampleRate)
|
||||
, m_log_dtk_audio(0)
|
||||
, m_log_dsp_audio(0)
|
||||
, m_speed(0)
|
||||
, m_l_dither_prev(0)
|
||||
, m_r_dither_prev(0)
|
||||
{
|
||||
INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
|
||||
m_output_buffer.reserve(MAX_SAMPLES * 2);
|
||||
}
|
||||
|
||||
static const u32 MAX_SAMPLES = 2048;
|
||||
static const u32 INDEX_MASK = MAX_SAMPLES * 2 - 1;
|
||||
static const float LOW_WATERMARK;
|
||||
static const float MAX_FREQ_SHIFT;
|
||||
static const float CONTROL_FACTOR;
|
||||
static const float CONTROL_AVG;
|
||||
|
||||
virtual ~CMixer() {}
|
||||
|
||||
// Called from audio threads
|
||||
virtual unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit = true);
|
||||
u32 Mix(s16* samples, u32 numSamples, bool consider_framelimit = true);
|
||||
|
||||
// Called from main thread
|
||||
virtual void PushSamples(const short* samples, unsigned int num_samples);
|
||||
virtual void PushStreamingSamples(const short* samples, unsigned int num_samples);
|
||||
virtual void PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples, unsigned int sample_rate);
|
||||
unsigned int GetSampleRate() const { return m_sampleRate; }
|
||||
virtual void PushSamples(const s16* samples, u32 num_samples);
|
||||
virtual void PushStreamingSamples(const s16* samples, u32 num_samples);
|
||||
virtual void PushWiimoteSpeakerSamples(const s16* samples, u32 num_samples, u32 sample_rate);
|
||||
u32 GetSampleRate() const { return m_sample_rate; }
|
||||
|
||||
void SetDMAInputSampleRate(unsigned int rate);
|
||||
void SetStreamInputSampleRate(unsigned int rate);
|
||||
void SetStreamingVolume(unsigned int lvolume, unsigned int rvolume);
|
||||
void SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume);
|
||||
void SetDMAInputSampleRate(u32 rate);
|
||||
void SetStreamInputSampleRate(u32 rate);
|
||||
void SetStreamingVolume(u32 lvolume, u32 rvolume);
|
||||
void SetWiimoteSpeakerVolume(u32 lvolume, u32 rvolume);
|
||||
|
||||
virtual void StartLogDTKAudio(const std::string& filename)
|
||||
{
|
||||
@ -107,46 +113,98 @@ public:
|
||||
}
|
||||
}
|
||||
|
||||
std::mutex& MixerCritical() { return m_csMixing; }
|
||||
std::mutex& MixerCritical() { return m_cs_mixing; }
|
||||
|
||||
float GetCurrentSpeed() const { return m_speed; }
|
||||
void UpdateSpeed(volatile float val) { m_speed = val; }
|
||||
|
||||
protected:
|
||||
class MixerFifo {
|
||||
class MixerFifo
|
||||
{
|
||||
public:
|
||||
MixerFifo(CMixer *mixer, unsigned sample_rate)
|
||||
MixerFifo(CMixer* mixer, u32 sample_rate)
|
||||
: m_mixer(mixer)
|
||||
, m_input_sample_rate(sample_rate)
|
||||
, m_indexW(0)
|
||||
, m_indexR(0)
|
||||
, m_LVolume(256)
|
||||
, m_RVolume(256)
|
||||
, m_numLeftI(0.0f)
|
||||
, m_frac(0)
|
||||
, m_write_index(0)
|
||||
, m_read_index(0)
|
||||
, m_lvolume(255)
|
||||
, m_rvolume(255)
|
||||
, m_num_left_i(0.0f)
|
||||
, m_fraction(0.0f)
|
||||
{
|
||||
memset(m_buffer, 0, sizeof(m_buffer));
|
||||
srand((u32) time(nullptr));
|
||||
}
|
||||
void PushSamples(const short* samples, unsigned int num_samples);
|
||||
unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit = true);
|
||||
void SetInputSampleRate(unsigned int rate);
|
||||
void SetVolume(unsigned int lvolume, unsigned int rvolume);
|
||||
private:
|
||||
virtual void Interpolate(u32 left_input_index, float* left_output, float* right_output) = 0;
|
||||
void PushSamples(const s16* samples, u32 num_samples);
|
||||
void Mix(std::vector<float>& samples, u32 numSamples, bool consider_framelimit = true);
|
||||
void SetInputSampleRate(u32 rate);
|
||||
void SetVolume(u32 lvolume, u32 rvolume);
|
||||
void GetVolume(u32* lvolume, u32* rvolume) const;
|
||||
|
||||
protected:
|
||||
CMixer* m_mixer;
|
||||
unsigned m_input_sample_rate;
|
||||
short m_buffer[MAX_SAMPLES * 2];
|
||||
volatile u32 m_indexW;
|
||||
volatile u32 m_indexR;
|
||||
// Volume ranges from 0-256
|
||||
volatile s32 m_LVolume;
|
||||
volatile s32 m_RVolume;
|
||||
float m_numLeftI;
|
||||
u32 m_frac;
|
||||
u32 m_input_sample_rate;
|
||||
|
||||
std::array<float, MAX_SAMPLES * 2> m_float_buffer;
|
||||
|
||||
volatile u32 m_write_index;
|
||||
volatile u32 m_read_index;
|
||||
|
||||
// Volume ranges from 0-255
|
||||
volatile u32 m_lvolume;
|
||||
volatile u32 m_rvolume;
|
||||
|
||||
float m_num_left_i;
|
||||
float m_fraction;
|
||||
};
|
||||
MixerFifo m_dma_mixer;
|
||||
MixerFifo m_streaming_mixer;
|
||||
MixerFifo m_wiimote_speaker_mixer;
|
||||
unsigned int m_sampleRate;
|
||||
|
||||
class LinearMixerFifo : public MixerFifo
|
||||
{
|
||||
public:
|
||||
LinearMixerFifo(CMixer* mixer, u32 sample_rate) : MixerFifo(mixer, sample_rate) {}
|
||||
void Interpolate(u32 left_input_index, float* left_output, float* right_output) override;
|
||||
};
|
||||
|
||||
class WindowedSincMixerFifo : public MixerFifo
|
||||
{
|
||||
public:
|
||||
WindowedSincMixerFifo(CMixer* mixer, u32 sample_rate) : MixerFifo(mixer, sample_rate) {}
|
||||
void Interpolate(u32 left_input_index, float* left_output, float* right_output) override;
|
||||
};
|
||||
|
||||
class Resampler
|
||||
{
|
||||
static const double LOWPASS_ROLLOFF;
|
||||
static const double KAISER_BETA;
|
||||
static const double BESSEL_EPSILON; // acceptable delta for Kaiser Window calculation
|
||||
|
||||
void PopulateFilterCoeff();
|
||||
double ModBessel0th(const double x);
|
||||
public:
|
||||
|
||||
static const u32 SAMPLES_PER_CROSSING = 4096;
|
||||
static const u32 NUM_CROSSINGS = 35;
|
||||
static const u32 WING_SIZE = SAMPLES_PER_CROSSING * (NUM_CROSSINGS - 1) / 2;
|
||||
|
||||
Resampler()
|
||||
{
|
||||
PopulateFilterCoeff();
|
||||
}
|
||||
|
||||
std::array<double, WING_SIZE> m_lowpass_filter;
|
||||
std::array<double, WING_SIZE> m_lowpass_delta;
|
||||
};
|
||||
|
||||
Resampler m_resampler;
|
||||
|
||||
WindowedSincMixerFifo m_dma_mixer;
|
||||
WindowedSincMixerFifo m_streaming_mixer;
|
||||
|
||||
// Linear interpolation seems to be the best for Wiimote 3khz -> 48khz, for now.
|
||||
// TODO: figure out why and make it work with the above FIR
|
||||
LinearMixerFifo m_wiimote_speaker_mixer;
|
||||
|
||||
u32 m_sample_rate;
|
||||
|
||||
WaveFileWriter g_wave_writer_dtk;
|
||||
WaveFileWriter g_wave_writer_dsp;
|
||||
@ -154,7 +212,26 @@ protected:
|
||||
bool m_log_dtk_audio;
|
||||
bool m_log_dsp_audio;
|
||||
|
||||
std::mutex m_csMixing;
|
||||
std::mutex m_cs_mixing;
|
||||
|
||||
volatile float m_speed; // Current rate of the emulation (1.0 = 100% speed)
|
||||
|
||||
private:
|
||||
// converts [-32768, 32767] -> [-1.0, 1.0]
|
||||
static inline float Signed16ToFloat(const s16 s)
|
||||
{
|
||||
return (s > 0) ? (float) (s / (float) 0x7fff) : (float) (s / (float) 0x8000);
|
||||
}
|
||||
|
||||
// converts [-1.0, 1.0] -> [-32768, 32767]
|
||||
static inline s16 FloatToSigned16(const float f)
|
||||
{
|
||||
return (f > 0) ? (s16) (f * 0x7fff) : (s16) (f * 0x8000);
|
||||
}
|
||||
|
||||
void TriangleDither(float* l_sample, float* r_sample);
|
||||
|
||||
std::vector<float> m_output_buffer;
|
||||
float m_l_dither_prev;
|
||||
float m_r_dither_prev;
|
||||
};
|
Loading…
Reference in New Issue
Block a user