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Initial commit

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added libraries:
opus
flac
libsndfile

updated:
libvorbis
libogg
openal

- Everything works as expected for now. Bare in mind libsndfile needed the check for whether or not it could find the xiph libraries removed in order for this to work.
This commit is contained in:
marauder2k7 2024-03-21 17:33:47 +00:00
parent 05a083ca6f
commit a745fc3757
1954 changed files with 431332 additions and 21037 deletions
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223
Engine/lib/openal-soft/alc/effects/pshifter.cpp
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@ -47,34 +47,36 @@ struct ContextBase;
namespace {
using uint = unsigned int;
using complex_d = std::complex<double>;
using complex_f = std::complex<float>;
#define STFT_SIZE 1024
#define STFT_HALF_SIZE (STFT_SIZE>>1)
#define OVERSAMP (1<<2)
constexpr size_t StftSize{1024};
constexpr size_t StftHalfSize{StftSize >> 1};
constexpr size_t OversampleFactor{8};
#define STFT_STEP (STFT_SIZE / OVERSAMP)
#define FIFO_LATENCY (STFT_STEP * (OVERSAMP-1))
static_assert(StftSize%OversampleFactor == 0, "Factor must be a clean divisor of the size");
constexpr size_t StftStep{StftSize / OversampleFactor};
/* Define a Hann window, used to filter the STFT input and output. */
std::array<double,STFT_SIZE> InitHannWindow()
{
std::array<double,STFT_SIZE> ret;
/* Create lookup table of the Hann window for the desired size, i.e. STFT_SIZE */
for(size_t i{0};i < STFT_SIZE>>1;i++)
struct Windower {
alignas(16) std::array<float,StftSize> mData;
Windower()
{
constexpr double scale{al::numbers::pi / double{STFT_SIZE}};
const double val{std::sin(static_cast<double>(i+1) * scale)};
ret[i] = ret[STFT_SIZE-1-i] = val * val;
/* Create lookup table of the Hann window for the desired size. */
for(size_t i{0};i < StftHalfSize;i++)
{
constexpr double scale{al::numbers::pi / double{StftSize}};
const double val{std::sin((static_cast<double>(i)+0.5) * scale)};
mData[i] = mData[StftSize-1-i] = static_cast<float>(val * val);
}
}
return ret;
}
alignas(16) const std::array<double,STFT_SIZE> HannWindow = InitHannWindow();
};
const Windower gWindow{};
struct FrequencyBin {
double Amplitude;
double FreqBin;
float Magnitude;
float FreqBin;
};
@ -83,27 +85,27 @@ struct PshifterState final : public EffectState {
size_t mCount;
size_t mPos;
uint mPitchShiftI;
double mPitchShift;
float mPitchShift;
/* Effects buffers */
std::array<double,STFT_SIZE> mFIFO;
std::array<double,STFT_HALF_SIZE+1> mLastPhase;
std::array<double,STFT_HALF_SIZE+1> mSumPhase;
std::array<double,STFT_SIZE> mOutputAccum;
std::array<float,StftSize> mFIFO;
std::array<float,StftHalfSize+1> mLastPhase;
std::array<float,StftHalfSize+1> mSumPhase;
std::array<float,StftSize> mOutputAccum;
std::array<complex_d,STFT_SIZE> mFftBuffer;
std::array<complex_f,StftSize> mFftBuffer;
std::array<FrequencyBin,STFT_HALF_SIZE+1> mAnalysisBuffer;
std::array<FrequencyBin,STFT_HALF_SIZE+1> mSynthesisBuffer;
std::array<FrequencyBin,StftHalfSize+1> mAnalysisBuffer;
std::array<FrequencyBin,StftHalfSize+1> mSynthesisBuffer;
alignas(16) FloatBufferLine mBufferOut;
/* Effect gains for each output channel */
float mCurrentGains[MAX_OUTPUT_CHANNELS];
float mTargetGains[MAX_OUTPUT_CHANNELS];
float mCurrentGains[MaxAmbiChannels];
float mTargetGains[MaxAmbiChannels];
void deviceUpdate(const DeviceBase *device, const Buffer &buffer) override;
void deviceUpdate(const DeviceBase *device, const BufferStorage *buffer) override;
void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
const EffectTarget target) override;
void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
@ -112,21 +114,21 @@ struct PshifterState final : public EffectState {
DEF_NEWDEL(PshifterState)
};
void PshifterState::deviceUpdate(const DeviceBase*, const Buffer&)
void PshifterState::deviceUpdate(const DeviceBase*, const BufferStorage*)
{
/* (Re-)initializing parameters and clear the buffers. */
mCount = 0;
mPos = FIFO_LATENCY;
mPos = StftSize - StftStep;
mPitchShiftI = MixerFracOne;
mPitchShift = 1.0;
mPitchShift = 1.0f;
std::fill(mFIFO.begin(), mFIFO.end(), 0.0);
std::fill(mLastPhase.begin(), mLastPhase.end(), 0.0);
std::fill(mSumPhase.begin(), mSumPhase.end(), 0.0);
std::fill(mOutputAccum.begin(), mOutputAccum.end(), 0.0);
std::fill(mFftBuffer.begin(), mFftBuffer.end(), complex_d{});
std::fill(mAnalysisBuffer.begin(), mAnalysisBuffer.end(), FrequencyBin{});
std::fill(mSynthesisBuffer.begin(), mSynthesisBuffer.end(), FrequencyBin{});
mFIFO.fill(0.0f);
mLastPhase.fill(0.0f);
mSumPhase.fill(0.0f);
mOutputAccum.fill(0.0f);
mFftBuffer.fill(complex_f{});
mAnalysisBuffer.fill(FrequencyBin{});
mSynthesisBuffer.fill(FrequencyBin{});
std::fill(std::begin(mCurrentGains), std::end(mCurrentGains), 0.0f);
std::fill(std::begin(mTargetGains), std::end(mTargetGains), 0.0f);
@ -137,16 +139,17 @@ void PshifterState::update(const ContextBase*, const EffectSlot *slot,
{
const int tune{props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune};
const float pitch{std::pow(2.0f, static_cast<float>(tune) / 1200.0f)};
mPitchShiftI = fastf2u(pitch*MixerFracOne);
mPitchShift = mPitchShiftI * double{1.0/MixerFracOne};
mPitchShiftI = clampu(fastf2u(pitch*MixerFracOne), MixerFracHalf, MixerFracOne*2);
mPitchShift = static_cast<float>(mPitchShiftI) * float{1.0f/MixerFracOne};
const auto coeffs = CalcDirectionCoeffs({0.0f, 0.0f, -1.0f}, 0.0f);
static constexpr auto coeffs = CalcDirectionCoeffs({0.0f, 0.0f, -1.0f});
mOutTarget = target.Main->Buffer;
ComputePanGains(target.Main, coeffs.data(), slot->Gain, mTargetGains);
}
void PshifterState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
void PshifterState::process(const size_t samplesToDo,
const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
/* Pitch shifter engine based on the work of Stephan Bernsee.
* http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
@ -155,103 +158,133 @@ void PshifterState::process(const size_t samplesToDo, const al::span<const Float
/* Cycle offset per update expected of each frequency bin (bin 0 is none,
* bin 1 is x1, bin 2 is x2, etc).
*/
constexpr double expected_cycles{al::numbers::pi*2.0 / OVERSAMP};
constexpr float expected_cycles{al::numbers::pi_v<float>*2.0f / OversampleFactor};
for(size_t base{0u};base < samplesToDo;)
{
const size_t todo{minz(STFT_STEP-mCount, samplesToDo-base)};
const size_t todo{minz(StftStep-mCount, samplesToDo-base)};
/* Retrieve the output samples from the FIFO and fill in the new input
* samples.
*/
auto fifo_iter = mFIFO.begin()+mPos + mCount;
std::transform(fifo_iter, fifo_iter+todo, mBufferOut.begin()+base,
[](double d) noexcept -> float { return static_cast<float>(d); });
std::copy_n(fifo_iter, todo, mBufferOut.begin()+base);
std::copy_n(samplesIn[0].begin()+base, todo, fifo_iter);
mCount += todo;
base += todo;
/* Check whether FIFO buffer is filled with new samples. */
if(mCount < STFT_STEP) break;
if(mCount < StftStep) break;
mCount = 0;
mPos = (mPos+STFT_STEP) & (mFIFO.size()-1);
mPos = (mPos+StftStep) & (mFIFO.size()-1);
/* Time-domain signal windowing, store in FftBuffer, and apply a
* forward FFT to get the frequency-domain signal.
*/
for(size_t src{mPos}, k{0u};src < STFT_SIZE;++src,++k)
mFftBuffer[k] = mFIFO[src] * HannWindow[k];
for(size_t src{0u}, k{STFT_SIZE-mPos};src < mPos;++src,++k)
mFftBuffer[k] = mFIFO[src] * HannWindow[k];
forward_fft(mFftBuffer);
for(size_t src{mPos}, k{0u};src < StftSize;++src,++k)
mFftBuffer[k] = mFIFO[src] * gWindow.mData[k];
for(size_t src{0u}, k{StftSize-mPos};src < mPos;++src,++k)
mFftBuffer[k] = mFIFO[src] * gWindow.mData[k];
forward_fft(al::as_span(mFftBuffer));
/* Analyze the obtained data. Since the real FFT is symmetric, only
* STFT_HALF_SIZE+1 samples are needed.
* StftHalfSize+1 samples are needed.
*/
for(size_t k{0u};k < STFT_HALF_SIZE+1;k++)
for(size_t k{0u};k < StftHalfSize+1;k++)
{
const double amplitude{std::abs(mFftBuffer[k])};
const double phase{std::arg(mFftBuffer[k])};
const float magnitude{std::abs(mFftBuffer[k])};
const float phase{std::arg(mFftBuffer[k])};
/* Compute phase difference and subtract expected phase difference */
double tmp{(phase - mLastPhase[k]) - static_cast<double>(k)*expected_cycles};
/* Map delta phase into +/- Pi interval */
int qpd{double2int(tmp / al::numbers::pi)};
tmp -= al::numbers::pi * (qpd + (qpd%2));
/* Get deviation from bin frequency from the +/- Pi interval */
tmp /= expected_cycles;
/* Compute the k-th partials' true frequency and store the
* amplitude and frequency bin in the analysis buffer.
/* Compute the phase difference from the last update and subtract
* the expected phase difference for this bin.
*
* When oversampling, the expected per-update offset increments by
* 1/OversampleFactor for every frequency bin. So, the offset wraps
* every 'OversampleFactor' bin.
*/
mAnalysisBuffer[k].Amplitude = amplitude;
mAnalysisBuffer[k].FreqBin = static_cast<double>(k) + tmp;
/* Store the actual phase[k] for the next frame. */
const auto bin_offset = static_cast<float>(k % OversampleFactor);
float tmp{(phase - mLastPhase[k]) - bin_offset*expected_cycles};
/* Store the actual phase for the next update. */
mLastPhase[k] = phase;
/* Normalize from pi, and wrap the delta between -1 and +1. */
tmp *= al::numbers::inv_pi_v<float>;
int qpd{float2int(tmp)};
tmp -= static_cast<float>(qpd + (qpd%2));
/* Get deviation from bin frequency (-0.5 to +0.5), and account for
* oversampling.
*/
tmp *= 0.5f * OversampleFactor;
/* Compute the k-th partials' frequency bin target and store the
* magnitude and frequency bin in the analysis buffer. We don't
* need the "true frequency" since it's a linear relationship with
* the bin.
*/
mAnalysisBuffer[k].Magnitude = magnitude;
mAnalysisBuffer[k].FreqBin = static_cast<float>(k) + tmp;
}
/* Shift the frequency bins according to the pitch adjustment,
* accumulating the amplitudes of overlapping frequency bins.
* accumulating the magnitudes of overlapping frequency bins.
*/
std::fill(mSynthesisBuffer.begin(), mSynthesisBuffer.end(), FrequencyBin{});
const size_t bin_count{minz(STFT_HALF_SIZE+1,
(((STFT_HALF_SIZE+1)<<MixerFracBits) - (MixerFracOne>>1) - 1)/mPitchShiftI + 1)};
constexpr size_t bin_limit{((StftHalfSize+1)<<MixerFracBits) - MixerFracHalf - 1};
const size_t bin_count{minz(StftHalfSize+1, bin_limit/mPitchShiftI + 1)};
for(size_t k{0u};k < bin_count;k++)
{
const size_t j{(k*mPitchShiftI + (MixerFracOne>>1)) >> MixerFracBits};
mSynthesisBuffer[j].Amplitude += mAnalysisBuffer[k].Amplitude;
mSynthesisBuffer[j].FreqBin = mAnalysisBuffer[k].FreqBin * mPitchShift;
const size_t j{(k*mPitchShiftI + MixerFracHalf) >> MixerFracBits};
/* If more than two bins end up together, use the target frequency
* bin for the one with the dominant magnitude. There might be a
* better way to handle this, but it's better than last-index-wins.
*/
if(mAnalysisBuffer[k].Magnitude > mSynthesisBuffer[j].Magnitude)
mSynthesisBuffer[j].FreqBin = mAnalysisBuffer[k].FreqBin * mPitchShift;
mSynthesisBuffer[j].Magnitude += mAnalysisBuffer[k].Magnitude;
}
/* Reconstruct the frequency-domain signal from the adjusted frequency
* bins.
*/
for(size_t k{0u};k < STFT_HALF_SIZE+1;k++)
for(size_t k{0u};k < StftHalfSize+1;k++)
{
/* Calculate actual delta phase and accumulate it to get bin phase */
mSumPhase[k] += mSynthesisBuffer[k].FreqBin * expected_cycles;
/* Calculate the actual delta phase for this bin's target frequency
* bin, and accumulate it to get the actual bin phase.
*/
float tmp{mSumPhase[k] + mSynthesisBuffer[k].FreqBin*expected_cycles};
mFftBuffer[k] = std::polar(mSynthesisBuffer[k].Amplitude, mSumPhase[k]);
/* Wrap between -pi and +pi for the sum. If mSumPhase is left to
* grow indefinitely, it will lose precision and produce less exact
* phase over time.
*/
tmp *= al::numbers::inv_pi_v<float>;
int qpd{float2int(tmp)};
tmp -= static_cast<float>(qpd + (qpd%2));
mSumPhase[k] = tmp * al::numbers::pi_v<float>;
mFftBuffer[k] = std::polar(mSynthesisBuffer[k].Magnitude, mSumPhase[k]);
}
for(size_t k{STFT_HALF_SIZE+1};k < STFT_SIZE;++k)
mFftBuffer[k] = std::conj(mFftBuffer[STFT_SIZE-k]);
for(size_t k{StftHalfSize+1};k < StftSize;++k)
mFftBuffer[k] = std::conj(mFftBuffer[StftSize-k]);
/* Apply an inverse FFT to get the time-domain siganl, and accumulate
/* Apply an inverse FFT to get the time-domain signal, and accumulate
* for the output with windowing.
*/
inverse_fft(mFftBuffer);
for(size_t dst{mPos}, k{0u};dst < STFT_SIZE;++dst,++k)
mOutputAccum[dst] += HannWindow[k]*mFftBuffer[k].real() * (4.0/OVERSAMP/STFT_SIZE);
for(size_t dst{0u}, k{STFT_SIZE-mPos};dst < mPos;++dst,++k)
mOutputAccum[dst] += HannWindow[k]*mFftBuffer[k].real() * (4.0/OVERSAMP/STFT_SIZE);
inverse_fft(al::as_span(mFftBuffer));
static constexpr float scale{3.0f / OversampleFactor / StftSize};
for(size_t dst{mPos}, k{0u};dst < StftSize;++dst,++k)
mOutputAccum[dst] += gWindow.mData[k]*mFftBuffer[k].real() * scale;
for(size_t dst{0u}, k{StftSize-mPos};dst < mPos;++dst,++k)
mOutputAccum[dst] += gWindow.mData[k]*mFftBuffer[k].real() * scale;
/* Copy out the accumulated result, then clear for the next iteration. */
std::copy_n(mOutputAccum.begin() + mPos, STFT_STEP, mFIFO.begin() + mPos);
std::fill_n(mOutputAccum.begin() + mPos, STFT_STEP, 0.0);
std::copy_n(mOutputAccum.begin() + mPos, StftStep, mFIFO.begin() + mPos);
std::fill_n(mOutputAccum.begin() + mPos, StftStep, 0.0f);
}
/* Now, mix the processed sound data to the output. */