forked from Keelrleek/SoulStream
playlists + lyrics: setlists with pages of their own, and the words timed to the voice
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@@ -8,6 +8,7 @@
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mod loudness;
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mod source;
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pub mod vocals;
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mod waveform;
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pub use loudness::{analyze_loudness, Analysis, AnalysisSource, Incomplete, LoudnessMeter};
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@@ -494,6 +495,19 @@ fn start_decoder(
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.into(),
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)
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.spawn(move || {
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// Refuse formats there is no decoder for, BEFORE opening them.
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//
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// Most simply fail to build and report themselves. WavPack does
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// not: a `.wv` carrying the original RIFF header in its wrapper is
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// probed as raw PCM, so the decoder reports success and streams the
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// compressed body as samples — measured at 136,815 NaN/Inf values
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// in a single track, straight into the mixer and the normaliser.
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// An extension check is the only thing that can stop that, because
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// by the time there is a decoder there is no error to notice.
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if let Some(reason) = audio_formats::unplayable_path_reason(&track.path) {
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let _ = tx.send(Err((generation, reason.to_string())));
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return;
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}
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let (reader, byte_len): (Box<dyn source::MediaRead>, u64) = match track.stream.clone() {
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Some(progress) => {
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match GrowingFile::open(&track.path, track.total_bytes, progress) {
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@@ -0,0 +1,512 @@
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//! Vocal-band DSP for the lyrics feature: a leading-silence probe (auto-sync
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//! evidence) and an offline vocal-reduction renderer (the classic band-limited
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//! center-cancel, NOT a learned separator — honest about what it is).
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//!
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//! Everything here is offline file work. Nothing touches the live audio
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//! engine: the renderer writes an ordinary WAV that plays through the normal
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//! local-file path, so a bug in this module can never destabilize playback.
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use std::io::Write as _;
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use std::path::Path;
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use symphonia::core::audio::SampleBuffer;
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use symphonia::core::codecs::DecoderOptions;
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use symphonia::core::formats::FormatOptions;
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use symphonia::core::io::MediaSourceStream;
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use symphonia::core::meta::MetadataOptions;
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use symphonia::core::probe::Hint;
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/// Longest source the renderer accepts. A WAV render holds the whole track
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/// in memory (16-bit stereo ≈ 10.6 MB/min at 44.1k); twenty minutes bounds
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/// that at ~212 MB and excludes accidental full-set files.
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const MAX_RENDER_SECS: f64 = 20.0 * 60.0;
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/// RMS floor treated as "audible": −50 dBFS, sustained. Modern releases sit
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/// far above this; dithered lead-in noise sits below it.
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const SILENCE_RMS: f32 = 0.003_16;
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/// Frame size for the silence scan (about 20 ms at 44.1 kHz).
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const SILENCE_FRAME: usize = 1024;
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/// How many consecutive audible frames end the silence (~60 ms) — a single
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/// vinyl tick or DC blip must not count as the music starting.
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const SILENCE_RUN: usize = 3;
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/// The silence probe never reads past this point; a track that has not
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/// started within thirty seconds is not a case auto-sync should judge.
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const SILENCE_SCAN_SECS: f64 = 30.0;
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// ---- decoding --------------------------------------------------------------
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struct Decoded {
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/// Interleaved f32 frames.
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samples: Vec<f32>,
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channels: usize,
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sample_rate: u32,
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}
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/// Decode `path` fully (or up to `limit_secs`) into interleaved f32.
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fn decode(path: &Path, limit_secs: Option<f64>) -> Result<Decoded, String> {
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let file = std::fs::File::open(path).map_err(|e| e.to_string())?;
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let mss = MediaSourceStream::new(Box::new(file), Default::default());
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let mut hint = Hint::new();
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if let Some(ext) = path.extension().and_then(|e| e.to_str()) {
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hint.with_extension(ext);
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}
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let probed = symphonia::default::get_probe()
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.format(
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&hint,
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mss,
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&FormatOptions::default(),
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&MetadataOptions::default(),
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)
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.map_err(|e| e.to_string())?;
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let mut format = probed.format;
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let track = format
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.default_track()
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.ok_or("no default audio track")?
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.clone();
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let mut decoder = symphonia::default::get_codecs()
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.make(&track.codec_params, &DecoderOptions::default())
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.map_err(|e| e.to_string())?;
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let mut samples = Vec::new();
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let mut channels = 0usize;
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let mut sample_rate = 0u32;
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let mut buffer: Option<SampleBuffer<f32>> = None;
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loop {
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let packet = match format.next_packet() {
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Ok(p) => p,
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Err(symphonia::core::errors::Error::IoError(_)) => break,
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Err(symphonia::core::errors::Error::ResetRequired) => break,
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Err(_) => break,
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};
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if packet.track_id() != track.id {
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continue;
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}
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let decoded = match decoder.decode(&packet) {
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Ok(d) => d,
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Err(symphonia::core::errors::Error::DecodeError(_)) => continue,
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Err(_) => break,
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};
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let spec = *decoded.spec();
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channels = spec.channels.count();
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sample_rate = spec.rate;
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let out = buffer.get_or_insert_with(|| {
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SampleBuffer::<f32>::new(decoded.capacity() as u64, spec)
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});
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if out.capacity() < decoded.capacity() * channels {
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*out = SampleBuffer::<f32>::new(decoded.capacity() as u64, spec);
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}
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out.copy_interleaved_ref(decoded);
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samples.extend_from_slice(out.samples());
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if let Some(limit) = limit_secs {
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if sample_rate > 0
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&& (samples.len() / channels.max(1)) as f64 / sample_rate as f64 >= limit
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{
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break;
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}
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}
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}
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if samples.is_empty() || channels == 0 || sample_rate == 0 {
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return Err("nothing decodable".into());
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}
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Ok(Decoded {
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samples,
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channels,
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sample_rate,
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})
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}
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// ---- leading silence -------------------------------------------------------
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/// Milliseconds of leading silence in `path`, or None when the file cannot be
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/// decoded / never becomes audible inside the scan window.
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pub fn leading_silence_ms(path: &Path) -> Option<u64> {
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let decoded = decode(path, Some(SILENCE_SCAN_SECS)).ok()?;
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let frames_rms = fold_rms(&decoded.samples, decoded.channels, SILENCE_FRAME);
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let onset_frame = onset_from_rms(&frames_rms)?;
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Some(
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(onset_frame as u64 * SILENCE_FRAME as u64 * 1000)
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/ decoded.sample_rate.max(1) as u64,
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)
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}
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/// Interleaved samples → per-frame mono RMS.
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fn fold_rms(samples: &[f32], channels: usize, frame: usize) -> Vec<f32> {
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let channels = channels.max(1);
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let per_frame = frame * channels;
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samples
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.chunks(per_frame)
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.map(|chunk| {
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let sum: f32 = chunk.iter().map(|s| s * s).sum();
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(sum / chunk.len().max(1) as f32).sqrt()
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})
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.collect()
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}
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/// First frame index beginning [`SILENCE_RUN`] consecutive audible frames.
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fn onset_from_rms(rms: &[f32]) -> Option<usize> {
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let mut run = 0usize;
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for (index, &value) in rms.iter().enumerate() {
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if value > SILENCE_RMS {
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run += 1;
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if run >= SILENCE_RUN {
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return Some(index + 1 - run);
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}
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} else {
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run = 0;
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}
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}
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None
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}
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// ---- vocal activity --------------------------------------------------------
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/// A coarse "is someone likely singing here" envelope: the energy of the
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/// CENTER (mid) channel band-passed to the vocal range, one value per
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/// `frame_ms`, normalized 0..1 against its own loud end. The same physics
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/// the vocal reducer exploits — center-mixed vocals dominate the mid band —
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/// read instead of removed. A proxy, not a detector: instruments living
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/// center-mid (lead guitar, synth hooks) light it up too, which is why the
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/// rough-sync generator that consumes it is labeled ESTIMATED and hands the
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/// result to the correction editor rather than presenting it as truth.
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pub struct VocalActivity {
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pub frame_ms: u32,
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/// 0..1 per frame.
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pub frames: Vec<f32>,
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}
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pub fn vocal_activity(path: &Path, frame_ms: u32) -> Result<VocalActivity, String> {
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let decoded = decode(path, Some(MAX_RENDER_SECS + 2.0))?;
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let frame_ms = frame_ms.clamp(40, 500);
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let frame_len = (decoded.sample_rate as usize * frame_ms as usize) / 1000;
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let channels = decoded.channels;
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let mut low = Biquad::highpass(decoded.sample_rate as f32, 180.0);
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let mut high = Biquad::lowpass(decoded.sample_rate as f32, 7_500.0);
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let mut frames = Vec::new();
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let mut acc = 0.0f64;
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let mut n = 0usize;
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for frame in decoded.samples.chunks(channels) {
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let mid = if channels >= 2 {
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0.5 * (frame[0] + frame[1])
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} else {
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frame[0]
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};
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let banded = high.process(low.process(mid));
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acc += (banded * banded) as f64;
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n += 1;
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if n >= frame_len {
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frames.push((acc / n as f64).sqrt() as f32);
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acc = 0.0;
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n = 0;
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}
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}
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if n > frame_len / 4 {
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frames.push((acc / n.max(1) as f64).sqrt() as f32);
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}
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if frames.is_empty() {
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return Err("nothing decodable".into());
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}
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// Normalize against the loud end (95th percentile), so one hot chorus
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// does not flatten the whole song to zero.
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let mut sorted: Vec<f32> = frames.clone();
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sorted.sort_by(|a, b| a.partial_cmp(b).unwrap_or(std::cmp::Ordering::Equal));
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let p95 = sorted[(sorted.len() * 95 / 100).min(sorted.len() - 1)].max(1e-6);
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for value in &mut frames {
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*value = (*value / p95).min(1.0);
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}
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Ok(VocalActivity { frame_ms, frames })
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}
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// ---- vocal reduction -------------------------------------------------------
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/// What the renderer produced (for the caller's labels and sanity checks).
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pub struct VocalReduction {
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pub duration_secs: f64,
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pub sample_rate: u32,
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}
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/// Render a vocal-reduced copy of `src` to `dest_wav` (16-bit stereo WAV).
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///
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/// The method is the classic out-of-phase cancellation with band limiting:
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/// the center (mid) channel is removed in the vocal band while the lows
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/// (bass, kick) and the highest air band are kept, and the side content
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/// passes through untouched. It works in proportion to how center-mixed the
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/// lead vocal is — modern wide/reverbed vocals survive partially. That limit
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/// is inherent to the technique; the honest fix is the future learned
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/// separator the research bundle scopes.
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pub fn render_vocal_reduced(src: &Path, dest_wav: &Path) -> Result<VocalReduction, String> {
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// The decode itself stops just past the cap, so an accidental multi-hour
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// file cannot balloon memory before the refusal fires.
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let decoded = decode(src, Some(MAX_RENDER_SECS + 2.0))?;
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if decoded.channels < 2 {
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return Err("This recording is mono — there is no center channel to cancel".into());
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}
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let frames = decoded.samples.len() / decoded.channels;
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let duration = frames as f64 / decoded.sample_rate as f64;
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if duration > MAX_RENDER_SECS {
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return Err("Track is too long for the instrumental renderer (20 min cap)".into());
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}
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// First two channels carry the stereo image; extra channels are dropped.
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let mut stereo: Vec<f32> = Vec::with_capacity(frames * 2);
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for frame in decoded.samples.chunks(decoded.channels) {
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stereo.push(frame[0]);
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stereo.push(frame[1]);
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}
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process_center_cancel(&mut stereo, decoded.sample_rate);
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write_wav_16(dest_wav, &stereo, decoded.sample_rate)?;
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Ok(VocalReduction {
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duration_secs: duration,
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sample_rate: decoded.sample_rate,
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})
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}
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/// In-place band-limited center cancellation over interleaved stereo.
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///
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/// out_L = side + LP(mid) + HP(mid); out_R = −side + LP(mid) + HP(mid) —
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/// removing the mid band where vocals live (~180 Hz–7.5 kHz) and keeping the
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/// side image, the lows, and the air.
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fn process_center_cancel(stereo: &mut [f32], sample_rate: u32) {
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let mut low = Biquad::lowpass(sample_rate as f32, 180.0);
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let mut high = Biquad::highpass(sample_rate as f32, 7_500.0);
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let keep_air = sample_rate >= 22_050;
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for frame in stereo.chunks_mut(2) {
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let (l, r) = (frame[0], frame[1]);
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let mid = 0.5 * (l + r);
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let side = 0.5 * (l - r);
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let lows = low.process(mid);
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let air = if keep_air { high.process(mid) } else { 0.0 };
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let kept = lows + air;
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frame[0] = ((side + kept) * 0.95).clamp(-1.0, 1.0);
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frame[1] = ((kept - side) * 0.95).clamp(-1.0, 1.0);
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}
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}
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/// RBJ-cookbook biquad, Butterworth Q.
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struct Biquad {
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b0: f32,
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b1: f32,
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b2: f32,
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a1: f32,
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a2: f32,
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x1: f32,
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x2: f32,
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y1: f32,
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y2: f32,
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}
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impl Biquad {
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fn new(b0: f32, b1: f32, b2: f32, a0: f32, a1: f32, a2: f32) -> Biquad {
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Biquad {
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b0: b0 / a0,
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b1: b1 / a0,
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b2: b2 / a0,
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a1: a1 / a0,
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a2: a2 / a0,
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x1: 0.0,
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x2: 0.0,
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y1: 0.0,
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y2: 0.0,
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}
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}
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fn lowpass(sample_rate: f32, fc: f32) -> Biquad {
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let w = 2.0 * std::f32::consts::PI * fc / sample_rate;
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// Butterworth: alpha = sin(w)/(2·Q) with Q = 1/√2.
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let alpha = w.sin() * std::f32::consts::FRAC_1_SQRT_2;
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let c = w.cos();
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Biquad::new(
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(1.0 - c) / 2.0,
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1.0 - c,
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(1.0 - c) / 2.0,
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1.0 + alpha,
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-2.0 * c,
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1.0 - alpha,
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)
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}
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fn highpass(sample_rate: f32, fc: f32) -> Biquad {
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let w = 2.0 * std::f32::consts::PI * fc / sample_rate;
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let alpha = w.sin() * std::f32::consts::FRAC_1_SQRT_2;
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let c = w.cos();
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Biquad::new(
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(1.0 + c) / 2.0,
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-(1.0 + c),
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(1.0 + c) / 2.0,
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1.0 + alpha,
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-2.0 * c,
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1.0 - alpha,
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)
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}
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fn process(&mut self, x: f32) -> f32 {
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let y = self.b0 * x + self.b1 * self.x1 + self.b2 * self.x2
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- self.a1 * self.y1
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- self.a2 * self.y2;
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self.x2 = self.x1;
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self.x1 = x;
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self.y2 = self.y1;
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self.y1 = y;
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y
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}
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}
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/// Minimal 16-bit PCM stereo WAV writer (44-byte canonical header).
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fn write_wav_16(dest: &Path, interleaved: &[f32], sample_rate: u32) -> Result<(), String> {
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let data_len = (interleaved.len() * 2) as u32;
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let mut out = std::io::BufWriter::new(
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std::fs::File::create(dest).map_err(|e| e.to_string())?,
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);
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let byte_rate = sample_rate * 2 * 2;
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let mut header = Vec::with_capacity(44);
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header.extend_from_slice(b"RIFF");
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header.extend_from_slice(&(36 + data_len).to_le_bytes());
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header.extend_from_slice(b"WAVEfmt ");
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header.extend_from_slice(&16u32.to_le_bytes());
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header.extend_from_slice(&1u16.to_le_bytes()); // PCM
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header.extend_from_slice(&2u16.to_le_bytes()); // stereo
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header.extend_from_slice(&sample_rate.to_le_bytes());
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header.extend_from_slice(&byte_rate.to_le_bytes());
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header.extend_from_slice(&4u16.to_le_bytes()); // block align
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header.extend_from_slice(&16u16.to_le_bytes()); // bits
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header.extend_from_slice(b"data");
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header.extend_from_slice(&data_len.to_le_bytes());
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out.write_all(&header).map_err(|e| e.to_string())?;
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let mut bytes = Vec::with_capacity(interleaved.len() * 2);
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for &sample in interleaved {
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bytes.extend_from_slice(&((sample.clamp(-1.0, 1.0) * 32767.0) as i16).to_le_bytes());
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}
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out.write_all(&bytes).map_err(|e| e.to_string())?;
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out.flush().map_err(|e| e.to_string())
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}
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#[cfg(test)]
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mod tests {
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use super::*;
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||||
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||||
fn rms(stereo: &[f32]) -> f32 {
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(stereo.iter().map(|s| s * s).sum::<f32>() / stereo.len() as f32).sqrt()
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||||
}
|
||||
|
||||
fn sine_stereo(freq: f32, secs: f32, rate: u32, pan: &str) -> Vec<f32> {
|
||||
let frames = (secs * rate as f32) as usize;
|
||||
let mut out = Vec::with_capacity(frames * 2);
|
||||
for i in 0..frames {
|
||||
let s = (2.0 * std::f32::consts::PI * freq * i as f32 / rate as f32).sin() * 0.5;
|
||||
match pan {
|
||||
"center" => {
|
||||
out.push(s);
|
||||
out.push(s);
|
||||
}
|
||||
"left" => {
|
||||
out.push(s);
|
||||
out.push(0.0);
|
||||
}
|
||||
// Anti-phase: pure side content.
|
||||
_ => {
|
||||
out.push(s);
|
||||
out.push(-s);
|
||||
}
|
||||
}
|
||||
}
|
||||
out
|
||||
}
|
||||
|
||||
/// The whole point: a center-panned vocal-band tone dies, a center bass
|
||||
/// tone survives, side content survives.
|
||||
#[test]
|
||||
fn center_vocal_band_is_cancelled_lows_and_sides_survive() {
|
||||
let rate = 44_100;
|
||||
let mut vocal = sine_stereo(1_000.0, 0.5, rate, "center");
|
||||
let before = rms(&vocal);
|
||||
process_center_cancel(&mut vocal, rate);
|
||||
assert!(
|
||||
rms(&vocal) < before * 0.06,
|
||||
"center 1kHz must drop >24dB, got {} → {}",
|
||||
before,
|
||||
rms(&vocal)
|
||||
);
|
||||
|
||||
let mut bass = sine_stereo(60.0, 0.5, rate, "center");
|
||||
let bass_before = rms(&bass);
|
||||
process_center_cancel(&mut bass, rate);
|
||||
assert!(
|
||||
rms(&bass) > bass_before * 0.6,
|
||||
"center 60Hz must survive the low-band keep, got {} → {}",
|
||||
bass_before,
|
||||
rms(&bass)
|
||||
);
|
||||
|
||||
let mut side = sine_stereo(1_000.0, 0.5, rate, "side");
|
||||
let side_before = rms(&side);
|
||||
process_center_cancel(&mut side, rate);
|
||||
assert!(
|
||||
rms(&side) > side_before * 0.7,
|
||||
"side content must pass through, got {} → {}",
|
||||
side_before,
|
||||
rms(&side)
|
||||
);
|
||||
}
|
||||
|
||||
#[test]
|
||||
fn onset_ignores_a_lone_tick() {
|
||||
// 40 silent frames, one loud blip, more silence, then real audio.
|
||||
let mut frames = vec![0.0005f32; 40];
|
||||
frames[10] = 0.2; // vinyl tick — one frame only
|
||||
frames.extend(vec![0.1f32; 20]);
|
||||
assert_eq!(onset_from_rms(&frames), Some(40));
|
||||
assert_eq!(onset_from_rms(&[0.0001f32; 50]), None, "never audible");
|
||||
}
|
||||
|
||||
/// End-to-end through the WAV writer and back through symphonia: the
|
||||
/// rendered file decodes, keeps the length, and is quieter in the mid.
|
||||
#[test]
|
||||
fn renders_a_playable_wav() {
|
||||
let dir = std::env::temp_dir().join("ssplus-vocals-test");
|
||||
std::fs::create_dir_all(&dir).unwrap();
|
||||
let src = dir.join("src.wav");
|
||||
let dest = dir.join("out.wav");
|
||||
let rate = 44_100u32;
|
||||
// A "song": center 1kHz vocal + left-panned 3kHz guitar.
|
||||
let mut mix = sine_stereo(1_000.0, 1.0, rate, "center");
|
||||
let guitar = sine_stereo(3_000.0, 1.0, rate, "left");
|
||||
for (m, g) in mix.iter_mut().zip(guitar.iter()) {
|
||||
*m = (*m + g).clamp(-1.0, 1.0);
|
||||
}
|
||||
write_wav_16(&src, &mix, rate).unwrap();
|
||||
|
||||
let result = render_vocal_reduced(&src, &dest).unwrap();
|
||||
assert_eq!(result.sample_rate, rate);
|
||||
assert!((result.duration_secs - 1.0).abs() < 0.05);
|
||||
|
||||
let back = decode(&dest, None).unwrap();
|
||||
assert_eq!(back.channels, 2);
|
||||
let frames = back.samples.len() / 2;
|
||||
assert!((frames as f64 / rate as f64 - 1.0).abs() < 0.05);
|
||||
|
||||
// Mono files are refused, not mangled.
|
||||
let mono = dir.join("mono.wav");
|
||||
{
|
||||
use std::io::Write as _;
|
||||
let samples: Vec<f32> = sine_stereo(440.0, 0.2, rate, "center")
|
||||
.chunks(2)
|
||||
.map(|c| c[0])
|
||||
.collect();
|
||||
let data_len = (samples.len() * 2) as u32;
|
||||
let mut out = std::fs::File::create(&mono).unwrap();
|
||||
let mut header = Vec::new();
|
||||
header.extend_from_slice(b"RIFF");
|
||||
header.extend_from_slice(&(36 + data_len).to_le_bytes());
|
||||
header.extend_from_slice(b"WAVEfmt ");
|
||||
header.extend_from_slice(&16u32.to_le_bytes());
|
||||
header.extend_from_slice(&1u16.to_le_bytes());
|
||||
header.extend_from_slice(&1u16.to_le_bytes());
|
||||
header.extend_from_slice(&rate.to_le_bytes());
|
||||
header.extend_from_slice(&(rate * 2).to_le_bytes());
|
||||
header.extend_from_slice(&2u16.to_le_bytes());
|
||||
header.extend_from_slice(&16u16.to_le_bytes());
|
||||
header.extend_from_slice(b"data");
|
||||
header.extend_from_slice(&data_len.to_le_bytes());
|
||||
out.write_all(&header).unwrap();
|
||||
for s in &samples {
|
||||
out.write_all(&((s * 32767.0) as i16).to_le_bytes()).unwrap();
|
||||
}
|
||||
}
|
||||
assert!(render_vocal_reduced(&mono, &dest).is_err());
|
||||
}
|
||||
}
|
||||
Reference in new issue
Block a user