// render48k.cpp — 48000/N=4096 internal-grid stereo renderer
//
// Host audio is 44100; the plugin detector runs internally at 48000/N=4096 (the
// live IIR/warp/freq-axis tables are sized for that grid). This tool mirrors that:
//   1. read input WAV (44100 host samples, stereo or mono)
//   2. resample 44100 -> 48000  (libsamplerate, SINC best)
//   3. Process L and R channels (stereo link=100%: same processing for both)
//   4. Apply balance: scale reduction for R channel
//   5. Apply mix: wet-dry mix
//   6. resample 48000 -> 44100
//   7. write 24-bit output WAV (stereo)
//
// Usage:
//   render48k <in.wav> <out.wav> fc,q,sens[,scale] ...
// Env:
//   RT_STEREO_LINK=1.0   (1.0 = sum channels for analysis, 0.0 = dual mono)
//   RT_STEREO_BALANCE=0.284  (R channel reduction scale, 1.0 = equal, <1.0 = less on R)
//   RT_DEPTH=0.864       (sens multiplier)
//   RT_MIX=1.0           (0=dry, 1=full wet)

#include "spectral.hpp"
#include <cstdio>
#include <cstdlib>
#include <vector>
#include <cmath>
#include <cstring>
#include <samplerate.h>

static int g_in_ch = 1;

static bool load_wav_stereo(const char* path, std::vector<float>& out, int& sr, int& channels) {
    FILE* f = fopen(path, "rb");
    if (!f) return false;
    char hdr[44];
    if (fread(hdr, 1, 44, f) != 44) return false;
    sr = *(int*)(hdr + 24);
    int ch = *(short*)(hdr + 22);
    int bits = *(short*)(hdr + 34);
    int data = 0;
    int64_t pos = 12;
    fseek(f, 12, SEEK_SET);
    while (pos < 32 * 1024 * 1024) {
        char cid[4]; int csize;
        if (fread(cid, 1, 4, f) < 4 || fread(&csize, 4, 1, f) < 1) break;
        pos += 8;
        if (memcmp(cid, "data", 4) == 0) { data = csize; break; }
        pos += csize;
        int skip = csize;
        if (csize % 2) skip++;
        fseek(f, skip, SEEK_CUR);
    }
    if (!data) { fclose(f); return false; }
    int n = data / (ch * (bits / 8));
    g_in_ch = ch;
    channels = 2;
    out.resize(n * 2);
    if (bits == 16) {
        std::vector<short> raw(n * ch);
        fread(raw.data(), 2, n * ch, f);
        for (int i = 0; i < n; i++) {
            float v = 0.0f;
            if (ch == 1) {
                v = raw[i] / 32768.0f;
                out[2*i] = v; out[2*i+1] = v;
            } else {
                out[2*i] = raw[2*i] / 32768.0f;
                out[2*i+1] = raw[2*i+1] / 32768.0f;
            }
        }
    } else if (bits == 24) {
        std::vector<unsigned char> raw(n * ch * 3);
        fread(raw.data(), 1, n * ch * 3, f);
        for (int i = 0; i < n; i++) {
            auto read24 = [&](int idx) -> float {
                int32_t s = (raw[idx] | (raw[idx+1] << 8) | (raw[idx+2] << 16));
                if (s & 0x800000) s |= 0xFF000000;
                return s / 8388608.0f;
            };
            if (ch == 1) {
                float v = read24(i * 3);
                out[2*i] = v; out[2*i+1] = v;
            } else {
                out[2*i] = read24(2*i * 3);
                out[2*i+1] = read24((2*i+1) * 3);
            }
        }
    } else return false;
    fclose(f);
    return true;
}

static bool save_wav24_stereo(const char* path, const std::vector<float>& L, const std::vector<float>& R, int sr) {
    FILE* f = fopen(path, "wb");
    if (!f) return false;
    int ch = 2, bits = 24;
    size_t n = std::min(L.size(), R.size());
    int data = (int)(n * ch * (bits / 8));
    char hdr[44]; memset(hdr, 0, 44);
    memcpy(hdr, "RIFF", 4); *(int*)(hdr + 4) = 36 + data;
    memcpy(hdr + 8, "WAVE", 4); memcpy(hdr + 12, "fmt ", 4);
    *(int*)(hdr + 16) = 16; *(short*)(hdr + 20) = 1; *(short*)(hdr + 22) = (short)ch;
    *(int*)(hdr + 24) = sr; *(int*)(hdr + 28) = sr * ch * (bits / 8);
    *(short*)(hdr + 32) = (short)ch; *(short*)(hdr + 34) = (short)bits;
    memcpy(hdr + 36, "data", 4); *(int*)(hdr + 40) = data;
    fwrite(hdr, 1, 44, f);
    for (size_t i = 0; i < n; i++) {
        int32_t vl = (int32_t)(std::max(-1.0f, std::min(1.0f, L[i])) * 8388607.0f);
        int32_t vr = (int32_t)(std::max(-1.0f, std::min(1.0f, R[i])) * 8388607.0f);
        unsigned char b0 = vl & 0xFF, b1 = (vl >> 8) & 0xFF, b2 = (vl >> 16) & 0xFF;
        fwrite(&b0, 1, 1, f); fwrite(&b1, 1, 1, f); fwrite(&b2, 1, 1, f);
        b0 = vr & 0xFF; b1 = (vr >> 8) & 0xFF; b2 = (vr >> 16) & 0xFF;
        fwrite(&b0, 1, 1, f); fwrite(&b1, 1, 1, f); fwrite(&b2, 1, 1, f);
    }
    fclose(f);
    return true;
}

static std::vector<float> resample_mono(const std::vector<float>& in, int src_sr, int dst_sr) {
    double frac = (double)dst_sr / src_sr;
    int out_len = (int)(in.size() * frac) + 16;
    std::vector<float> buf(out_len);
    SRC_DATA sd;
    sd.data_in = in.data(); sd.input_frames = (long)in.size();
    sd.data_out = buf.data(); sd.output_frames = out_len;
    sd.src_ratio = frac; sd.end_of_input = 1;
    int err = src_simple(&sd, SRC_SINC_BEST_QUALITY, 1);
    if (err != 0) { fprintf(stderr, "resample err %d\n", err); return {}; }
    buf.resize(sd.output_frames_gen);
    return buf;
}

static std::vector<float> resample_stereo(const std::vector<float>& in, int src_sr, int dst_sr) {
    size_t n = in.size() / 2;
    double frac = (double)dst_sr / src_sr;
    int out_len = (int)(n * frac) + 16;
    std::vector<float> buf(out_len * 2);
    SRC_DATA sd;
    sd.data_in = in.data(); sd.input_frames = (long)n;
    sd.data_out = buf.data(); sd.output_frames = out_len;
    sd.src_ratio = frac; sd.end_of_input = 1;
    int err = src_simple(&sd, SRC_SINC_BEST_QUALITY, 2);
    if (err != 0) { fprintf(stderr, "resample err %d\n", err); return {}; }
    buf.resize(sd.output_frames_gen * 2);
    return buf;
}

int main(int argc, char** argv) {
    if (argc < 3) {
        fprintf(stderr, "usage: %s in.wav out.wav fc,q,sens[,scale] ...\n", argv[0]);
        return 1;
    }
    std::vector<float> x; int sr, channels;
    if (!load_wav_stereo(argv[1], x, sr, channels)) { fprintf(stderr, "cannot load %s\n", argv[1]); return 1; }

    // Parse stereo parameters from env
    float stereo_link = getenv("RT_STEREO_LINK") ? atof(getenv("RT_STEREO_LINK")) : 1.0f;
    float stereo_balance = getenv("RT_STEREO_BALANCE") ? atof(getenv("RT_STEREO_BALANCE")) : 0.284f;
    float depth = getenv("RT_DEPTH") ? atof(getenv("RT_DEPTH")) : 1.0f;
    float mix = getenv("RT_MIX") ? atof(getenv("RT_MIX")) : 1.0f;

    // Parse bands
    std::vector<DetectorBand> bands;
    for (int i = 3; i < argc; i++) {
        if (!strchr(argv[i], ',')) continue;
        float fc, q, sens, scl = 1.0f;
        if (sscanf(argv[i], "%f,%f,%f,%f", &fc, &q, &sens, &scl) < 3) continue;
        DetectorBand b; b.fc = fc; b.q = q; b.sens = sens * depth; b.level_scale = scl;
        bands.push_back(b);
    }
    if (bands.empty()) bands.push_back({1000.0f, 1.0f, 12.0f * depth});

    auto x48 = resample_stereo(x, sr, 48000);
    if (x48.empty()) return 1;

    size_t n = x48.size() / 2;

    // Stereo processing per soothe2 manual:
    // "With the stereo link at 100%, Soothe will sum the channels for analysis
    //  and apply the same processing to both channels."
    // Use separate processors for L and R to avoid stateful interference.

    SpectralProcessor procL(4096, 1024, 48000.0f);
    SpectralProcessor procR(4096, 1024, 48000.0f);
    procL.setDetectorParams(bands);
    procR.setDetectorParams(bands);

    std::vector<float> L_out(n), R_out(n);
    const size_t BLK = 1 << 16;
    std::vector<float> inb(BLK), outb(BLK);

    // Process L channel
    for (size_t s = 0; s < n; s += BLK) {
        size_t blk = std::min(BLK, n - s);
        memcpy(inb.data(), x48.data() + 2*s, blk * sizeof(float));
        for (size_t i = blk; i < BLK; i++) inb[i] = 0.0f;
        procL.processBlock(inb.data(), outb.data(), BLK, 1);
        for (size_t i = 0; i < blk; i++) L_out[s+i] = outb[i];
    }

    // Process R channel
    for (size_t s = 0; s < n; s += BLK) {
        size_t blk = std::min(BLK, n - s);
        memcpy(inb.data(), x48.data() + 2*s + 1, blk * sizeof(float));
        for (size_t i = blk; i < BLK; i++) inb[i] = 0.0f;
        procR.processBlock(inb.data(), outb.data(), BLK, 1);
        for (size_t i = 0; i < blk; i++) R_out[s+i] = outb[i];
    }

    // Apply balance and mix
    std::vector<float> L_final(n), R_final(n);
    for (size_t i = 0; i < n; i++) {
        float L = x48[2*i];
        float R = x48[2*i+1];
        float L_proc = L_out[i];
        float R_proc = R_out[i];

        float mask_L = (std::abs(L) > 1e-12f) ? L_proc / L : 1.0f;
        float mask_R = (std::abs(R) > 1e-12f) ? R_proc / R : 1.0f;

        // Apply balance: scale reduction for R channel
        float mask_R_bal = 1.0f - (1.0f - mask_R) * stereo_balance;

        // Apply mix
        L_final[i] = L * (1.0f - mix) + L * mask_L * mix;
        R_final[i] = R * (1.0f - mix) + R * mask_R_bal * mix;
    }

    auto L44 = resample_mono(L_final, 48000, 44100);
    auto R44 = resample_mono(R_final, 48000, 44100);
    size_t out_len = std::min(L44.size(), R44.size());
    out_len = std::min(out_len, x.size() / std::max(channels, 1));
    L44.resize(out_len);
    R44.resize(out_len);
    save_wav24_stereo(argv[2], L44, R44, 44100);
    printf("render48k: %zu hostsamps ch=%d -> %zu (48k) -> %zu (out), balance=%.3f depth=%.3f\n",
           x.size(), channels, x48.size(), out_len, stereo_balance, depth);
    return 0;
}