audio.cpp

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#include "audio.h"
 
namespace fl {
 
const AudioSample::VectorPCM &AudioSample::pcm() const {
    if (isValid()) {
        return mImpl->pcm();
    }
    static VectorPCM empty;
    return empty;
}
 
AudioSample &AudioSample::operator=(const AudioSample &other) {
    mImpl = other.mImpl;
    return *this;
}
 
size_t AudioSample::size() const {
    if (isValid()) {
        return mImpl->pcm().size();
    }
    return 0;
}
 
const int16_t &AudioSample::at(size_t i) const {
    if (i < size()) {
        return pcm()[i];
    }
    return empty()[0];
}
 
const int16_t &AudioSample::operator[](size_t i) const { return at(i); }
 
bool AudioSample::operator==(const AudioSample &other) const {
    if (mImpl == other.mImpl) {
        return true;
    }
    if (mImpl == nullptr || other.mImpl == nullptr) {
        return false;
    }
    if (mImpl->pcm().size() != other.mImpl->pcm().size()) {
        return false;
    }
    for (size_t i = 0; i < mImpl->pcm().size(); ++i) {
        if (mImpl->pcm()[i] != other.mImpl->pcm()[i]) {
            return false;
        }
    }
    return true;
}
 
bool AudioSample::operator!=(const AudioSample &other) const {
    return !(*this == other);
}
 
const AudioSample::VectorPCM &AudioSample::empty() {
    static int16_t empty_data[1] = {0};
    static VectorPCM empty(empty_data);
    return empty;
}
 
float AudioSample::zcf() const { return mImpl->zcf(); }
 
float AudioSample::rms() const {
    if (!isValid()) {
        return 0.0f;
    }
    uint64_t sum_sq = 0;
    const int N = size();
    for (int i = 0; i < N; ++i) {
        int32_t x32 = int32_t(pcm()[i]);
        sum_sq += x32 * x32;
    }
    float rms = sqrtf(float(sum_sq) / N);
    return rms;
}
 
SoundLevelMeter::SoundLevelMeter(double spl_floor, double smoothing_alpha)
    : spl_floor_(spl_floor), smoothing_alpha_(smoothing_alpha),
      dbfs_floor_global_(INFINITY_DOUBLE), offset_(0.0), current_dbfs_(0.0),
      current_spl_(spl_floor) {}
 
void SoundLevelMeter::processBlock(const int16_t *samples, size_t count) {
    // 1) compute block power → dBFS
    double sum_sq = 0.0;
    for (size_t i = 0; i < count; ++i) {
        double s = samples[i] / 32768.0; // normalize to ±1
        sum_sq += s * s;
    }
    double p = sum_sq / count; // mean power
    double dbfs = 10.0 * log10(p + 1e-12);
    current_dbfs_ = dbfs;
 
    // 2) update global floor (with optional smoothing)
    if (dbfs < dbfs_floor_global_) {
        if (smoothing_alpha_ <= 0.0) {
            dbfs_floor_global_ = dbfs;
        } else {
            dbfs_floor_global_ = smoothing_alpha_ * dbfs +
                                 (1.0 - smoothing_alpha_) * dbfs_floor_global_;
        }
        offset_ = spl_floor_ - dbfs_floor_global_;
    }
 
    // 3) estimate SPL
    current_spl_ = dbfs + offset_;
}
 
static FFT gFlexFFT;
 
void AudioSample::fft(FFTBins *out) {
    fl::Slice<const int16_t> sample = pcm();
    FFT_Args args;
    args.samples = sample.size();
    args.bands = out->size();
    args.fmin = FFT_Args::DefaultMinFrequency();
    args.fmax = FFT_Args::DefaultMaxFrequency();
    args.sample_rate =
        FFT_Args::DefaultSampleRate(); // TODO: get sample rate from AudioSample
    gFlexFFT.run(sample, out, args);
}
 
} // namespace fl