cathar 0.5.3

//! Enhancement: voice isolation, de-essing, breath removal, bandwidth extension.

use crate::util::hann_window;
use crate::{NoisePrint, resample};
use realfft::RealFftPlanner;

/// Isolate speech from background using energy-based VAD + spectral gating.
pub fn voice_isolate(
    signal: &[f32],
    sample_rate: u32,
    noise_print: Option<&NoisePrint>,
) -> Vec<f32> {
    let fft_size = 2048;
    let hop_size = 512;
    let n = signal.len();
    if n < fft_size {
        return signal.to_vec();
    }

    let frame_len = sample_rate as usize / 50;
    let vad_frames = n / frame_len;
    let mut frame_energies: Vec<f32> = Vec::with_capacity(vad_frames);
    for fi in 0..vad_frames {
        let start = fi * frame_len;
        let end = (start + frame_len).min(n);
        let energy: f32 =
            signal[start..end].iter().map(|s| s * s).sum::<f32>() / (end - start) as f32;
        frame_energies.push(energy);
    }
    if frame_energies.is_empty() {
        return signal.to_vec();
    }

    let mut sorted = frame_energies.clone();
    sorted.sort_by(|a, b| a.partial_cmp(b).unwrap_or(std::cmp::Ordering::Equal));
    let noise_floor = sorted[(sorted.len() / 5).min(sorted.len() - 1)];
    let threshold = noise_floor * 4.0;

    let mut is_voice = vec![false; vad_frames + 1];
    for (fi, &energy) in frame_energies.iter().enumerate() {
        is_voice[fi] = energy > threshold;
    }

    let min_voice_frames = (sample_rate as f32 * 0.05 / frame_len as f32).max(1.0) as usize;
    let max_gap_frames = (sample_rate as f32 * 0.12 / frame_len as f32).max(1.0) as usize;

    let mut smoothed = is_voice.clone();
    for fi in 0..vad_frames {
        if !is_voice[fi] {
            let voice_before = (1..=max_gap_frames).any(|j| fi >= j && is_voice[fi - j]);
            let voice_after = (1..=max_gap_frames).any(|j| fi + j < vad_frames && is_voice[fi + j]);
            if voice_before && voice_after {
                smoothed[fi] = true;
            }
        }
    }

    for fi in 0..vad_frames {
        if smoothed[fi] {
            let run = (fi..vad_frames).take_while(|&j| j < vad_frames && smoothed[j]).count();
            if run < min_voice_frames {
                for s in smoothed.iter_mut().take((fi + run).min(vad_frames)).skip(fi) {
                    *s = false;
                }
            }
        }
    }

    let mut planner = RealFftPlanner::<f32>::new();
    let r2c = planner.plan_fft_forward(fft_size);
    let c2r = planner.plan_fft_inverse(fft_size);
    let hann = hann_window(fft_size);
    let scale = 1.0f32 / (fft_size as f32);
    let n_bins = fft_size / 2 + 1;
    let frames = n / hop_size;
    let mut output = vec![0.0f32; n + fft_size];
    let mut in_buf = r2c.make_input_vec();
    let mut out_buf = r2c.make_output_vec();
    let noise_spec = noise_print.map(|np| &np.spectrum);

    for fi in 0..frames {
        let offset = fi * hop_size;
        if offset + fft_size > n {
            break;
        }
        let center_sample = offset + fft_size / 2;
        let vad_idx = center_sample / frame_len;
        let voice_present = smoothed.get(vad_idx).copied().unwrap_or(false);

        for i in 0..fft_size {
            in_buf[i] = signal[offset + i] * hann[i];
        }
        r2c.process(&mut in_buf, &mut out_buf).unwrap();

        if !voice_present {
            if let Some(ns) = noise_spec {
                for k in 0..n_bins {
                    let mag =
                        (out_buf[k].re * out_buf[k].re + out_buf[k].im * out_buf[k].im).sqrt();
                    let gate = (ns[k] * 0.3 / mag.max(1e-10)).min(1.0);
                    out_buf[k].re *= gate;
                    out_buf[k].im *= gate;
                }
            } else {
                for val in out_buf.iter_mut().take(n_bins) {
                    val.re *= 0.01;
                    val.im *= 0.01;
                }
            }
        }

        c2r.process(&mut out_buf, &mut in_buf).unwrap();
        for i in 0..fft_size {
            output[offset + i] += in_buf[i] * hann[i] * scale;
        }
    }
    output.truncate(n);
    output
}

// ── De-esser ────────────────────────────────────────────────────────────────

/// Reduce sibilance (harsh "s", "sh", "ch" sounds) using HF compression.
pub fn deesser(
    signal: &[f32],
    sample_rate: u32,
    crossover_freq: f32,
    threshold_db: f32,
    ratio: f32,
) -> Vec<f32> {
    let fft_size = 2048;
    let hop_size = 256;
    let n = signal.len();
    if n < fft_size {
        return signal.to_vec();
    }
    let mut planner = RealFftPlanner::<f32>::new();
    let r2c = planner.plan_fft_forward(fft_size);
    let c2r = planner.plan_fft_inverse(fft_size);
    let hann = hann_window(fft_size);
    let scale = 1.0f32 / (fft_size as f32);
    let n_bins = fft_size / 2 + 1;
    let nyquist = sample_rate as f32 / 2.0;
    let crossover_bin = ((crossover_freq / nyquist) * (n_bins - 1) as f32).round() as usize;
    let frames = n / hop_size;
    let threshold_linear = 10.0f32.powf(threshold_db / 20.0);

    let mut output = vec![0.0f32; n + fft_size];
    let mut in_buf = r2c.make_input_vec();
    let mut out_buf = r2c.make_output_vec();

    for fi in 0..frames {
        let offset = fi * hop_size;
        if offset + fft_size > n {
            break;
        }
        for i in 0..fft_size {
            in_buf[i] = signal[offset + i] * hann[i];
        }
        r2c.process(&mut in_buf, &mut out_buf).unwrap();

        let mut broadband_power = 0.0f32;
        let mut hf_power = 0.0f32;
        for (k, val) in out_buf.iter().enumerate().take(n_bins) {
            let power = val.re * val.re + val.im * val.im;
            broadband_power += power;
            if k >= crossover_bin {
                hf_power += power;
            }
        }

        let broadband_rms = broadband_power.sqrt();
        let hf_rms = hf_power.sqrt();
        let hf_ratio = if broadband_rms > 1e-10 { hf_rms / broadband_rms } else { 0.0 };

        if hf_ratio > threshold_linear {
            let overshoot = hf_ratio / threshold_linear;
            let gain_reduction = 1.0 / (1.0 + (overshoot - 1.0) * ratio);
            for (k, val) in out_buf.iter_mut().enumerate().take(n_bins) {
                let frac = (k.min(crossover_bin) as f32 / crossover_bin.max(1) as f32).min(1.0);
                let bin_gain = gain_reduction + (1.0 - gain_reduction) * (1.0 - frac);
                val.re *= bin_gain;
                val.im *= bin_gain;
            }
        }

        c2r.process(&mut out_buf, &mut in_buf).unwrap();
        for i in 0..fft_size {
            output[offset + i] += in_buf[i] * hann[i] * scale;
        }
    }
    output.truncate(n);
    output
}

/// Multiband, adaptive de-esser. The region above `crossover_freq` is split into
/// `bands` sub-bands; each tracks its own short-term level (an exponential moving
/// average) and is compressed by `ratio` only when its instantaneous level rises
/// `threshold_db` above that adaptive average. Adapting *per band* and *over
/// time* catches sibilance concentrated in part of the band and follows a
/// speaker's changing level, where a single fixed-threshold band over- or
/// under-reacts. Falls back to a single band when `bands <= 1`.
pub fn deess_multiband(
    signal: &[f32],
    sample_rate: u32,
    crossover_freq: f32,
    threshold_db: f32,
    ratio: f32,
    bands: usize,
) -> Vec<f32> {
    let fft_size = 2048;
    let hop_size = 256;
    let n = signal.len();
    if n < fft_size {
        return signal.to_vec();
    }
    let bands = bands.max(1);
    let mut planner = RealFftPlanner::<f32>::new();
    let r2c = planner.plan_fft_forward(fft_size);
    let c2r = planner.plan_fft_inverse(fft_size);
    let hann = hann_window(fft_size);
    let scale = 1.0f32 / fft_size as f32;
    let n_bins = fft_size / 2 + 1;
    let nyquist = sample_rate as f32 / 2.0;
    let crossover_bin =
        (((crossover_freq / nyquist) * (n_bins - 1) as f32).round() as usize).min(n_bins - 1);
    let frames = n / hop_size;
    // `threshold_db` is how far above the running average triggers compression.
    let threshold_linear = 10.0f32.powf(threshold_db / 20.0);
    let span = n_bins.saturating_sub(crossover_bin).max(1);
    let band_width = span.div_ceil(bands);
    // Adaptive per-band level; EMA smoothing — slow enough that a sibilant
    // transient stands above the running average, fast enough to follow speech.
    let mut avg = vec![0.0f32; bands];
    let smoothing = 0.85f32;

    let mut output = vec![0.0f32; n + fft_size];
    let mut wsum = vec![0.0f32; n + fft_size];
    let mut in_buf = r2c.make_input_vec();
    let mut out_buf = r2c.make_output_vec();

    for fi in 0..frames {
        let offset = fi * hop_size;
        if offset + fft_size > n {
            break;
        }
        for i in 0..fft_size {
            in_buf[i] = signal[offset + i] * hann[i];
        }
        r2c.process(&mut in_buf, &mut out_buf).unwrap();

        for (band, level) in avg.iter_mut().enumerate() {
            let lo = crossover_bin + band * band_width;
            let hi = (lo + band_width).min(n_bins);
            if lo >= hi {
                continue;
            }
            let power: f32 = out_buf[lo..hi].iter().map(|v| v.re * v.re + v.im * v.im).sum();
            let rms = power.sqrt();
            if *level <= 0.0 {
                *level = rms; // seed
            }
            let over = rms / (*level * threshold_linear).max(1e-10);
            if over > 1.0 {
                let gr = 1.0 / (1.0 + (over - 1.0) * ratio);
                for v in out_buf[lo..hi].iter_mut() {
                    v.re *= gr;
                    v.im *= gr;
                }
            }
            // Track the pre-compression level so transient sibilance reads as "over".
            *level = smoothing * *level + (1.0 - smoothing) * rms;
        }

        c2r.process(&mut out_buf, &mut in_buf).unwrap();
        for i in 0..fft_size {
            output[offset + i] += in_buf[i] * hann[i] * scale;
            wsum[offset + i] += hann[i] * hann[i];
        }
    }
    for i in 0..n {
        if wsum[i] > 1e-6 {
            output[i] /= wsum[i];
        }
    }
    output.truncate(n);
    output
}

// ── Breath removal ──────────────────────────────────────────────────────────

/// Detect and attenuate breath sounds between speech segments.
pub fn breath_remove(signal: &[f32], sample_rate: u32) -> Vec<f32> {
    let frame_len = sample_rate as usize / 50;
    let n = signal.len();
    if n < frame_len * 10 {
        return signal.to_vec();
    }
    let vad_frames = n / frame_len;

    let mut frame_energies: Vec<f32> = Vec::with_capacity(vad_frames);
    for fi in 0..vad_frames {
        let start = fi * frame_len;
        let end = (start + frame_len).min(n);
        let energy: f32 =
            signal[start..end].iter().map(|s| s * s).sum::<f32>() / (end - start) as f32;
        frame_energies.push(energy);
    }
    if frame_energies.is_empty() {
        return signal.to_vec();
    }

    let mut sorted = frame_energies.clone();
    sorted.sort_by(|a, b| a.partial_cmp(b).unwrap_or(std::cmp::Ordering::Equal));
    let noise_floor = sorted[(sorted.len() / 5).min(sorted.len() - 1)];
    let threshold = noise_floor * 4.0;

    let mut is_voice = vec![false; vad_frames];
    for (fi, &energy) in frame_energies.iter().enumerate() {
        is_voice[fi] = energy > threshold;
    }

    let breath_margin_frames = (sample_rate as f32 * 0.15 / frame_len as f32) as usize;
    let mut breath_mask = vec![false; n];
    for fi in 1..vad_frames {
        if is_voice[fi] && !is_voice[fi - 1] {
            let start_frame = fi.saturating_sub(breath_margin_frames);
            for bf in start_frame..fi {
                let bs = bf * frame_len;
                let be = ((bf + 1) * frame_len).min(n);
                for b in breath_mask.iter_mut().take(be).skip(bs) {
                    *b = true;
                }
            }
        }
    }

    let hpf_cutoff = 200.0;
    let hpf_coeff = (-2.0 * std::f32::consts::PI * hpf_cutoff / sample_rate as f32).exp();
    let mut output = signal.to_vec();
    let (mut prev_in, mut prev_out) = (0.0f32, 0.0f32);
    let dry_wet = 0.6;

    for (i, s) in signal.iter().enumerate() {
        if breath_mask[i] {
            let hp = hpf_coeff * prev_out + hpf_coeff * (*s - prev_in);
            prev_in = *s;
            prev_out = hp;
            output[i] = *s * (1.0 - dry_wet) + hp * dry_wet;
        }
    }
    output
}

/// Restore high-frequency content lost to compression or low sample rates.
///
/// Uses spectral band replication: the spectral envelope from the upper octave
/// of the source signal is transposed into the missing high band, shaped, and
/// mixed back. No ML — pure DSP, zero weights.
pub fn bandwidth_extend(signal: &[f32], sample_rate: u32, target_rate: u32) -> Vec<f32> {
    if target_rate <= sample_rate {
        return signal.to_vec();
    }

    // ── 1. Resample to target rate (shared Kaiser-windowed sinc) ──
    let resampled = resample(signal, sample_rate, target_rate);

    // ── 2. SBR: replicate low-band spectrum shape into high band ──
    let fft_size = 4096;
    let hop_size = fft_size / 4;
    let n = resampled.len();
    if n < fft_size {
        return resampled;
    }

    let old_nyquist = sample_rate as f32 / 2.0;
    let new_nyquist = target_rate as f32 / 2.0;
    let n_bins = fft_size / 2 + 1;
    let old_nyquist_bin = ((old_nyquist / new_nyquist) * (n_bins - 1) as f32).round() as usize;
    let source_band_start = (old_nyquist_bin as f32 * 0.6).round() as usize;
    let source_band_width = old_nyquist_bin - source_band_start;

    let mut planner = RealFftPlanner::<f32>::new();
    let r2c = planner.plan_fft_forward(fft_size);
    let c2r = planner.plan_fft_inverse(fft_size);
    let hann = hann_window(fft_size);
    let scale = 1.0f32 / (fft_size as f32);
    let frames = n / hop_size;

    let mut output = vec![0.0f32; n + fft_size];
    let mut in_buf = r2c.make_input_vec();
    let mut out_buf = r2c.make_output_vec();

    for fi in 0..frames {
        let offset = fi * hop_size;
        if offset + fft_size > n {
            break;
        }
        for i in 0..fft_size {
            in_buf[i] = resampled[offset + i] * hann[i];
        }
        r2c.process(&mut in_buf, &mut out_buf).unwrap();

        // Estimate spectral envelope from source band
        let mut envelope = vec![0.0f32; source_band_width];
        for (k, env) in envelope.iter_mut().enumerate() {
            let src_k = source_band_start + k;
            *env = (out_buf[src_k].re * out_buf[src_k].re + out_buf[src_k].im * out_buf[src_k].im)
                .sqrt();
        }

        // Replicate into high band with gentle rolloff
        for tile in 0..4 {
            let target_start = old_nyquist_bin + tile * source_band_width;
            if target_start >= n_bins - 1 {
                break;
            }
            let rolloff = 1.0 - tile as f32 * 0.3;
            if rolloff <= 0.0 {
                break;
            }
            for (k, env_val) in envelope.iter().enumerate() {
                let tgt = target_start + k;
                if tgt >= n_bins - 1 {
                    break;
                }
                let existing =
                    (out_buf[tgt].re * out_buf[tgt].re + out_buf[tgt].im * out_buf[tgt].im).sqrt();
                if existing < 1e-6 {
                    let src_k = source_band_start + k;
                    let phase =
                        out_buf[src_k].im.atan2(out_buf[src_k].re) + std::f32::consts::PI * 0.25;
                    let sbr_amp = env_val * rolloff * 0.6;
                    let freq_rolloff =
                        (-(tgt as f32 - old_nyquist_bin as f32) / 300.0).exp().max(0.02);
                    let amp = sbr_amp * freq_rolloff;
                    out_buf[tgt].re += amp * phase.cos();
                    out_buf[tgt].im += amp * phase.sin();
                }
            }
        }

        c2r.process(&mut out_buf, &mut in_buf).unwrap();
        for i in 0..fft_size {
            output[offset + i] += in_buf[i] * hann[i] * scale;
        }
    }

    output.truncate(n);
    output
}