ResampleLQ.cc

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#include "ResampleLQ.hh"
#include "ResampledSoundDevice.hh"
#include "likely.hh"
#include "ranges.hh"
#include "xrange.hh"
#include <cassert>
#include <cstring>
#include <memory>
#include <vector>
 
namespace openmsx {
 
// 16-byte aligned buffer of ints (shared among all instances of this resampler)
static std::vector<float> bufferStorage; // (possibly) unaligned storage
static unsigned bufferSize = 0; // usable buffer size (aligned portion)
static float* aBuffer = nullptr; // pointer to aligned sub-buffer
 
 
template<unsigned CHANNELS>
std::unique_ptr<ResampleLQ<CHANNELS>> ResampleLQ<CHANNELS>::create(
                ResampledSoundDevice& input, const DynamicClock& hostClock)
{
        std::unique_ptr<ResampleLQ<CHANNELS>> result;
        if (input.getEmuClock().getPeriod() >= hostClock.getPeriod()) {
                result = std::make_unique<ResampleLQUp  <CHANNELS>>(input, hostClock);
        } else {
                result = std::make_unique<ResampleLQDown<CHANNELS>>(input, hostClock);
        }
        return result;
}
 
template<unsigned CHANNELS>
ResampleLQ<CHANNELS>::ResampleLQ(
                ResampledSoundDevice& input_, const DynamicClock& hostClock_)
        : ResampleAlgo(input_)
        , hostClock(hostClock_)
        , step([&]{ // calculate 'getEmuClock().getFreq() / hostClock.getFreq()', but with less rounding errors
                        uint64_t emuPeriod = input_.getEmuClock().getPeriod().length(); // unknown units
                        uint64_t hostPeriod = hostClock.getPeriod().length(); // unknown units, but same as above
                        assert(unsigned( emuPeriod) ==  emuPeriod);
                        assert(unsigned(hostPeriod) == hostPeriod);
                        return FP::roundRatioDown(hostPeriod, emuPeriod);
                }())
{
        ranges::fill(lastInput, 0.0f);
}
 
[[nodiscard]] static bool isSilent(float x)
{
        constexpr float threshold = 1.0f / 32768;
        return std::abs(x) < threshold;
}
 
template<unsigned CHANNELS>
bool ResampleLQ<CHANNELS>::fetchData(EmuTime::param time, unsigned& valid)
{
        auto& emuClk = getEmuClock();
        unsigned emuNum = emuClk.getTicksTill(time);
        valid = 2 + emuNum;
 
        unsigned required = emuNum + 4;
        if (unlikely(required > bufferSize)) {
                // grow buffer (3 extra to be able to align)
                bufferStorage.resize(required + 3);
                // align at 16-byte boundary
                auto p = reinterpret_cast<uintptr_t>(bufferStorage.data());
                aBuffer = reinterpret_cast<float*>((p + 15) & ~15);
                // calculate actual usable size (the aligned portion)
                bufferSize = (bufferStorage.data() + bufferStorage.size()) - aBuffer;
                assert(bufferSize >= required);
        }
 
        emuClk += emuNum;
 
        auto* buffer = &aBuffer[4 - 2 * CHANNELS];
        assert((uintptr_t(&buffer[2 * CHANNELS]) & 15) == 0);
 
        if (!input.generateInput(&buffer[2 * CHANNELS], emuNum)) {
                // New input is all zero
                if (ranges::all_of(lastInput, [](auto& l) { return isSilent(l); })) {
                        // Old input was also all zero, then the resampled
                        // output will be all zero as well.
                        return false;
                }
                memset(&buffer[CHANNELS], 0, emuNum * CHANNELS * sizeof(float));
        }
        for (auto j : xrange(2 * CHANNELS)) {
                buffer[j] = lastInput[j];
                lastInput[j] = buffer[emuNum * CHANNELS + j];
        }
        return true;
}
 
 
template<unsigned CHANNELS>
ResampleLQUp<CHANNELS>::ResampleLQUp(
                ResampledSoundDevice& input_, const DynamicClock& hostClock_)
        : ResampleLQ<CHANNELS>(input_, hostClock_)
{
        assert(input_.getEmuClock().getFreq() <= hostClock_.getFreq()); // only upsampling
}
 
template<unsigned CHANNELS>
bool ResampleLQUp<CHANNELS>::generateOutputImpl(
        float* __restrict dataOut, unsigned hostNum, EmuTime::param time)
{
        auto& emuClk = this->getEmuClock();
        EmuTime host1 = this->hostClock.getFastAdd(1);
        assert(host1 > emuClk.getTime());
        FP pos;
        emuClk.getTicksTill(host1, pos);
        assert(pos.toInt() < 2);
 
        unsigned valid; // only indices smaller than this number are valid
        if (!this->fetchData(time, valid)) return false;
 
        // this is currently only used to upsample cassette player sound,
        // sound quality is not so important here, so use 0-th order
        // interpolation (instead of 1st-order).
        auto* buffer = &aBuffer[4 - 2 * CHANNELS];
        for (auto i : xrange(hostNum)) {
                unsigned p = pos.toInt();
                assert(p < valid);
                for (auto j : xrange(CHANNELS)) {
                        dataOut[i * CHANNELS + j] = buffer[p * CHANNELS + j];
                }
                pos += this->step;
        }
 
        return true;
}
 
 
template<unsigned CHANNELS>
ResampleLQDown<CHANNELS>::ResampleLQDown(
                ResampledSoundDevice& input_, const DynamicClock& hostClock_)
        : ResampleLQ<CHANNELS>(input_, hostClock_)
{
        assert(input_.getEmuClock().getFreq() >= hostClock_.getFreq()); // can only do downsampling
}
 
template<unsigned CHANNELS>
bool ResampleLQDown<CHANNELS>::generateOutputImpl(
        float* __restrict dataOut, unsigned hostNum, EmuTime::param time)
{
        auto& emuClk = this->getEmuClock();
        EmuTime host1 = this->hostClock.getFastAdd(1);
        assert(host1 > emuClk.getTime());
        FP pos;
        emuClk.getTicksTill(host1, pos);
 
        unsigned valid;
        if (!this->fetchData(time, valid)) return false;
 
        auto* buffer = &aBuffer[4 - 2 * CHANNELS];
        for (auto i : xrange(hostNum)) {
                unsigned p = pos.toInt();
                assert((p + 1) < valid);
                FP fract = pos.fract();
                for (auto j : xrange(CHANNELS)) {
                        auto s0 = buffer[(p + 0) * CHANNELS + j];
                        auto s1 = buffer[(p + 1) * CHANNELS + j];
                        auto out = s0 + (fract.toFloat() * (s1 - s0));
                        dataOut[i * CHANNELS + j] = out;
                }
                pos += this->step;
        }
        return true;
}
 
 
// Force template instantiation.
template class ResampleLQ<1>;
template class ResampleLQ<2>;
 
} // namespace openmsx