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soothe2-re/dsp/render48k.cpp
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// render48k.cpp — 48000/N=4096 internal-grid stereo/M/S 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)
// 2. resample 44100 -> 48000 (libsamplerate, SINC best)
// 3. M/S decode: mid = (L+R)/2, side = (L-R)/2
// 4. Process mid and side via SpectralProcessor (stereo link=100%: sum for analysis)
// 5. Apply balance: mid_reduction *= 1.0, side_reduction *= balance
// 6. M/S encode: L' = mid_out + side_out, R' = mid_out - side_out
// 7. resample 48000 -> 44100
// 8. write 24-bit output WAV (stereo)
//
// Usage:
// render48k <in.wav> <out.wav> mid_fc,mid_q,mid_sens[,scale] side_fc,side_q,side_sens[,scale]
// render48k <in.wav> <out.wav> fc,q,sens[,scale] (mono input, dual-mono processing)
// Env:
// RT_STEREO_LINK=1.0 (1.0 = sum channels for analysis, 0.0 = dual mono)
// RT_STEREO_BALANCE=0.284 (side reduction scale, 1.0 = equal, <1.0 = less on side)
// RT_DEPTH=0.864 (sens multiplier)
// RT_MODE=1.0 (0=soft, 1=hard)
// 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; // always output stereo interleaved
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; // mono -> stereo
} 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) {
// in is stereo interleaved: L0, R0, L1, R1, ...
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 mid_fc,mid_q,mid_sens[,scale] side_fc,side_q,side_sens[,scale]\n", argv[0]);
fprintf(stderr, " %s in.wav out.wav fc,q,sens[,scale] (mono input, dual-mono)\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> mid_bands, side_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;
if (mid_bands.empty()) mid_bands.push_back(b);
else side_bands.push_back(b);
}
if (mid_bands.empty()) mid_bands.push_back({1000.0f, 1.0f, 12.0f * depth});
if (side_bands.empty()) side_bands = mid_bands;
auto x48 = resample_stereo(x, sr, 48000);
if (x48.empty()) return 1;
size_t n = x48.size() / 2; // samples per channel
std::vector<float> mid(n), side(n);
// M/S decode (or mono duplication for mono input)
if (channels >= 2) {
for (size_t i = 0; i < n; i++) {
float L = x48[2*i];
float R = x48[2*i+1];
mid[i] = (L + R) * 0.5f;
side[i] = (L - R) * 0.5f;
}
} else {
for (size_t i = 0; i < n; i++) {
mid[i] = x48[i];
side[i] = x48[i];
}
}
// For stereo link=100%: sum mid+side for analysis
std::vector<float> analysis_buf(n);
if (stereo_link >= 1.0f) {
for (size_t i = 0; i < n; i++) analysis_buf[i] = mid[i] + side[i];
}
// Create processors
SpectralProcessor mid_sp(4096, 1024, 48000.0f);
SpectralProcessor side_sp(4096, 1024, 48000.0f);
mid_sp.setDetectorParams(mid_bands);
side_sp.setDetectorParams(side_bands);
// Process
std::vector<float> mid_out(n), side_out(n);
const size_t BLK = 1 << 16;
std::vector<float> inb(BLK), outb(BLK);
for (size_t s = 0; s < n; s += BLK) {
size_t blk = std::min(BLK, n - s);
memcpy(inb.data(), mid.data() + s, blk * sizeof(float));
for (size_t i = blk; i < BLK; i++) inb[i] = 0.0f;
mid_sp.processBlock(inb.data(), outb.data(), BLK, 1);
memcpy(mid_out.data() + s, outb.data(), blk * sizeof(float));
}
for (size_t s = 0; s < n; s += BLK) {
size_t blk = std::min(BLK, n - s);
memcpy(inb.data(), side.data() + s, blk * sizeof(float));
for (size_t i = blk; i < BLK; i++) inb[i] = 0.0f;
side_sp.processBlock(inb.data(), outb.data(), BLK, 1);
memcpy(side_out.data() + s, outb.data(), blk * sizeof(float));
}
// M/S encode with balance and mix
// mask is [0,1] where 1=no change, <1=reduction
// processed = input * mask
// output = input * (1 - mix) + processed * mix
// For side: apply balance to reduction amount
std::vector<float> Lout(n), Rout(n);
for (size_t i = 0; i < n; i++) {
float m_in = mid[i];
float s_in = side[i];
float m_proc = mid_out[i]; // m_in * mask_mid
float s_proc = side_out[i]; // s_in * mask_side
// Compute mask values (avoid division by zero)
float mask_mid = (std::abs(m_in) > 1e-12f) ? m_proc / m_in : 1.0f;
float mask_side = (std::abs(s_in) > 1e-12f) ? s_proc / s_in : 1.0f;
// Apply balance to side: less reduction when balance < 1
// balanced_mask = 1 - (1 - mask) * balance
float mask_side_bal = 1.0f - (1.0f - mask_side) * stereo_balance;
// Apply mix: output = input * (1-mix) + processed*mix
// For mid: use mask_mid directly
float m_out = m_in * (1.0f - mix) + m_in * mask_mid * mix;
// For side: use balanced mask
float s_out = s_in * (1.0f - mix) + s_in * mask_side_bal * mix;
// M/S encode
Lout[i] = m_out + s_out;
Rout[i] = m_out - s_out;
}
auto L44 = resample_mono(Lout, 48000, 44100);
auto R44 = resample_mono(Rout, 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), stereo_balance=%.3f depth=%.3f\n",
x.size(), channels, x48.size(), out_len, stereo_balance, depth);
return 0;
}
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