lac 0.1.0

//! Real-audio corpus tests: round-trip, compression ratio, FLAC comparison.
//!
//! These are integration tests — run with `cargo test --test corpus` to see
//! the printed ratio numbers, or `cargo test --test corpus -- --nocapture`
//! for verbose output during development.
//!
//! Each test is gated by the presence of its corpus file so the suite still
//! passes on a clean checkout without the WAV data.
//!
//! # What we measure
//!
//! - **Round-trip**: encode every frame of the input, decode back, verify
//!   sample-for-sample equality. Failure here means the codec is broken.
//! - **Compression ratio**: `encoded_size / raw_size` for the full file,
//!   framed at a realistic frame size. Ratios below 0.6 are "good" for a
//!   lossless codec; ratios above 0.8 suggest something is wrong with the
//!   LPC or Rice tuning.
//! - **FLAC comparison**: invoke `flac` CLI on the same input, compare
//!   compressed sizes. We expect LAC to be within 10-25% of FLAC size on
//!   most content; consistently much worse indicates a real codec gap (the
//!   Q15 coefficient-clamping limitation, most likely).
//! - **LAC encode wall-clock**: printed as `lac_enc_ms`. Not asserted —
//!   CI hardware variance makes any ceiling either useless or flaky. The
//!   number is visibility-only, meant to be correlated against
//!   `bench/compare-flac.sh` output for an engineer-side speed sanity
//!   check. Only the encode hot path is timed (decode is excluded: at
//!   microsecond scale, decode speed is dominated by allocator noise).

use std::path::{Path, PathBuf};
use std::process::Command;
use std::time::{Duration, Instant};

use hound::WavReader;
use lac::{decode_frame, encode_frame};

const CORPUS_DIR: &str = "corpus";

/// Encode-side frame length. 4096 is FLAC's default blocksize at
/// compression levels 1-8 for `≤ 48 kHz` content and sits inside LAC's
/// own archival-default band (README "Offline / archival"). Using the
/// same block size on both sides makes the FLAC comparison
/// apples-to-apples: neither codec is charged for a block-size
/// mismatch with the other.
///
/// Every partition order `0..=7` divides 4096, so the encoder stays on
/// the dense search path throughout.
const FRAME_SIZE: usize = 4096;

// ── WAV loading ─────────────────────────────────────────────────────────────

/// Load a WAV file as separated mono channels of i32 samples.
///
/// Samples are passed through at their native width — 16-bit values stay
/// 16-bit, inside the i32 carrier. LAC's only hard constraint is that
/// `|sample|` fits in 24 bits (ceiling for autocorrelation overflow
/// analysis); narrower inputs compress according to their actual
/// magnitudes.
///
/// The earlier version of this loader left-shifted to "promote" 16-bit to
/// 24-bit. That was wrong: Rice coding tracks residual magnitude, so a
/// 256× amplification costs 8 extra bits per residual and inflates the
/// output size proportionally.
fn load_wav_channels(path: &Path) -> Option<Vec<Vec<i32>>> {
    let mut reader = WavReader::open(path).ok()?;
    let spec = reader.spec();
    if spec.sample_format != hound::SampleFormat::Int {
        return None;
    }
    let ch = spec.channels as usize;

    // Sanity check: reject inputs whose values won't fit in 24 bits. In
    // practice the corpus is 16-bit or 24-bit integer PCM so this is just
    // a defensive guard.
    if spec.bits_per_sample > 24 {
        return None;
    }

    let mut channels: Vec<Vec<i32>> = (0..ch).map(|_| Vec::new()).collect();
    for (i, s) in reader.samples::<i32>().enumerate() {
        let s = s.ok()?;
        channels[i % ch].push(s);
    }
    Some(channels)
}

fn corpus_path(name: &str) -> PathBuf {
    Path::new(CORPUS_DIR).join(name)
}

/// Skip a test if its WAV file isn't present, so the suite stays green on
/// a clean checkout. Prints a one-line hint so the operator knows why.
macro_rules! require_corpus {
    ($path:expr) => {
        if !$path.exists() {
            eprintln!("skipping: corpus file not found: {}", $path.display());
            return;
        }
    };
}

// ── Per-frame round-trip harness ────────────────────────────────────────────

/// Aggregate per-file measurement: raw byte count, encoded byte count,
/// and wall-clock time spent inside `encode_frame`. The decode time is
/// not reported — it is an order of magnitude smaller than encode for
/// every content class in the corpus, so the ratio of interest is
/// encode-side.
struct Measurement {
    raw_bytes: usize,
    encoded_bytes: usize,
    encode_time: Duration,
}

impl Measurement {
    fn new() -> Self {
        Self {
            raw_bytes: 0,
            encoded_bytes: 0,
            encode_time: Duration::ZERO,
        }
    }

    fn add(&mut self, other: &Measurement) {
        self.raw_bytes += other.raw_bytes;
        self.encoded_bytes += other.encoded_bytes;
        self.encode_time += other.encode_time;
    }
}

/// Encode every `FRAME_SIZE`-sample chunk of `channel`, decode, assert
/// equality. The trailing partial chunk (if any) is encoded at whatever
/// partition_order divides its length; the encoder's search handles that
/// automatically. `encode_time` captures only the encode hot path —
/// decode, allocation of the returned `Vec`, and the round-trip assert
/// are excluded so the number is directly comparable to `flac`'s wall
/// clock at the same input.
fn roundtrip_channel(channel: &[i32], bytes_per_sample: usize) -> Measurement {
    let mut m = Measurement::new();
    for chunk in channel.chunks(FRAME_SIZE) {
        let t = Instant::now();
        let encoded = encode_frame(chunk);
        m.encode_time += t.elapsed();
        let decoded = decode_frame(&encoded).expect("decode_frame failed on own output");
        assert_eq!(
            decoded,
            chunk,
            "round-trip mismatch in frame of {} samples",
            chunk.len()
        );
        m.raw_bytes += chunk.len() * bytes_per_sample;
        m.encoded_bytes += encoded.len();
    }
    m
}

/// Aggregate round-trip over every channel in a WAV file. Returns a
/// `Measurement` whose `raw_bytes` uses the WAV's actual sample width,
/// so the ratio corresponds to the over-the-wire comparison a user
/// would do against the same file encoded with FLAC.
fn roundtrip_wav(path: &Path) -> Measurement {
    // Probe the spec once to know the bytes-per-sample for the raw-size
    // denominator. `load_wav_channels` filters unsupported formats, so the
    // spec read here is guaranteed to be a supported integer format.
    let spec = WavReader::open(path).expect("open for spec").spec();
    // Bits per sample rounds up to bytes: 16 → 2, 20 → 3, 24 → 3.
    let bytes_per_sample = spec.bits_per_sample.div_ceil(8) as usize;

    let channels = load_wav_channels(path).expect("load_wav_channels failed");
    let mut totals = Measurement::new();
    for ch in &channels {
        let m = roundtrip_channel(ch, bytes_per_sample);
        totals.add(&m);
    }
    totals
}

// ── FLAC comparison ─────────────────────────────────────────────────────────

/// Invoke the `flac` CLI to compress `path`, return the resulting byte
/// count. Returns `None` if the FLAC tool isn't installed.
fn flac_compress_size(path: &Path) -> Option<usize> {
    // `flac --stdout --silent <file>` writes compressed FLAC to stdout so
    // we never touch the filesystem for the output. `-o -` would do the
    // same but is not universally supported across FLAC versions.
    let out = Command::new("flac")
        .arg("--stdout")
        .arg("--silent")
        .arg("--best")
        .arg(path)
        .output()
        .ok()?;
    if !out.status.success() {
        return None;
    }
    Some(out.stdout.len())
}

// ── Tests ───────────────────────────────────────────────────────────────────

fn report_ratio(name: &str, m: &Measurement, flac_size: Option<usize>) {
    let ratio = m.encoded_bytes as f64 / m.raw_bytes as f64;
    let enc_ms = m.encode_time.as_secs_f64() * 1000.0;
    eprint!(
        "{name:40}  raw={:>10}  lac={:>10}  ratio={ratio:.3}  lac_enc_ms={enc_ms:>7.1}",
        m.raw_bytes, m.encoded_bytes,
    );
    if let Some(flac) = flac_size {
        let flac_ratio = flac as f64 / m.raw_bytes as f64;
        // `lac_vs_flac` > 1.0 means LAC is bigger than FLAC. Anything
        // above ~1.3 on typical content points at the Q15-clamping
        // limitation and motivates adding the coefficient-shift field.
        let lac_vs_flac = m.encoded_bytes as f64 / flac as f64;
        eprint!("  flac={flac:>10}  flac_ratio={flac_ratio:.3}  lac/flac={lac_vs_flac:.3}");
    }
    eprintln!();
}

// Music — solo piano from the Open Goldberg Variations project (Kimiko
// Ishizaka's recording of J.S. Bach's BWV 988, released CC0 by the
// project). 96 kHz / 24-bit / stereo studio masters — genuine lossless
// source, redistributable, covering three distinct pianistic content
// classes between them.

const GOLDBERG_PREFIX: &str =
    "Kimiko Ishizaka - J.S. Bach- -Open- Goldberg Variations, BWV 988 (Piano)";

#[test]
fn bach_aria() {
    // Slow, sustained, lyrical sarabande with long held notes and gentle
    // bass — LPC's best-case piano content (sustained harmonics,
    // minimal transients, smooth melodic motion). 300 s / 96 kHz / 24-bit
    // stereo.
    let path = corpus_path(&format!("{GOLDBERG_PREFIX} - 01 Aria.wav"));
    require_corpus!(path);
    let m = roundtrip_wav(&path);
    let flac = flac_compress_size(&path);
    report_ratio("bach_aria (solo piano, tonal)", &m, flac);
    let ratio = m.encoded_bytes as f64 / m.raw_bytes as f64;
    // Measured ~0.483 at FRAME_SIZE=4096 on 96 kHz / 24-bit stereo;
    // ceiling 0.503 keeps the ~2 pp regression budget used elsewhere.
    assert!(
        ratio < 0.503,
        "bach_aria ratio {} exceeds regression ceiling 0.503",
        ratio
    );
}

#[test]
fn bach_variatio_4_fughetta() {
    // Short fugal variation — tight polyphonic counterpoint with
    // interleaved voices. Fast melodic runs stress Rice k-selection on
    // richer residual statistics than the Aria. Cheapest music test in
    // the suite at ~68 s.
    let path = corpus_path(&format!("{GOLDBERG_PREFIX} - 05 Variatio 4 a 1 Clav..wav"));
    require_corpus!(path);
    let m = roundtrip_wav(&path);
    let flac = flac_compress_size(&path);
    report_ratio("bach_variatio_4 (fugal)", &m, flac);
    let ratio = m.encoded_bytes as f64 / m.raw_bytes as f64;
    // Measured ~0.514; ceiling 0.534 keeps the ~2 pp regression budget.
    assert!(
        ratio < 0.534,
        "bach_variatio_4 ratio {} exceeds regression ceiling 0.534",
        ratio
    );
}

#[test]
fn bach_variatio_16_ouverture() {
    // French-overture style: dotted rhythms, strong attacks, ornamented
    // melodic runs. The transient-heavy content class — residual
    // statistics shift mid-frame as runs punctuate sustained harmonies,
    // exercising partition_order search.
    let path = corpus_path(&format!(
        "{GOLDBERG_PREFIX} - 17 Variatio 16 a 1 Clav. Ouverture.wav"
    ));
    require_corpus!(path);
    let m = roundtrip_wav(&path);
    let flac = flac_compress_size(&path);
    report_ratio("bach_variatio_16 (ouverture)", &m, flac);
    let ratio = m.encoded_bytes as f64 / m.raw_bytes as f64;
    // Measured ~0.512; ceiling 0.532 keeps the ~2 pp regression budget.
    assert!(
        ratio < 0.532,
        "bach_variatio_16 ratio {} exceeds regression ceiling 0.532",
        ratio
    );
}

// Speech — AMI meeting corpus. Clean headset mic gives near-ideal speech
// conditions; residuals should be near-Laplacian, which is exactly what
// Rice coding is optimal for. Expect the best ratios here.

#[test]
fn ami_headset_speech() {
    let path = corpus_path("ES2002a.Headset-0.wav");
    require_corpus!(path);
    let m = roundtrip_wav(&path);
    let flac = flac_compress_size(&path);
    report_ratio("ami_headset_speech", &m, flac);
    let ratio = m.encoded_bytes as f64 / m.raw_bytes as f64;
    // Measured ~0.178 at FRAME_SIZE=4096; ceiling 0.195 gives ~2 pp budget.
    assert!(
        ratio < 0.195,
        "headset ratio {} exceeds regression ceiling 0.195",
        ratio
    );
}

#[test]
fn ami_array_speech() {
    // Tabletop array mic: distant speech with room acoustics. Less
    // predictable than a headset mic — a useful stress case for LPC.
    let path = corpus_path("ES2002a.Array1-01.wav");
    require_corpus!(path);
    let m = roundtrip_wav(&path);
    let flac = flac_compress_size(&path);
    report_ratio("ami_array_speech", &m, flac);
    let ratio = m.encoded_bytes as f64 / m.raw_bytes as f64;
    // Measured ~0.375 at FRAME_SIZE=4096; ceiling 0.395 gives ~2 pp budget.
    assert!(
        ratio < 0.395,
        "array speech ratio {} exceeds regression ceiling 0.395",
        ratio
    );
}

#[test]
fn ami_mixed_meeting() {
    // Mixed headset: multiple simultaneous speakers. Highest spectral
    // complexity in the corpus.
    let path = corpus_path("ES2002a.Mix-Headset.wav");
    require_corpus!(path);
    let m = roundtrip_wav(&path);
    let flac = flac_compress_size(&path);
    report_ratio("ami_mixed_meeting", &m, flac);
    let ratio = m.encoded_bytes as f64 / m.raw_bytes as f64;
    // Measured ~0.292 at FRAME_SIZE=4096; ceiling 0.312 gives ~2 pp budget.
    assert!(
        ratio < 0.312,
        "mixed meeting ratio {} exceeds regression ceiling 0.312",
        ratio
    );
}

// The sparse-vs-exhaustive encoder differential lives in
// `src/frame.rs::tests::sparse_vs_exhaustive_on_headset_speech` — it
// needs `encode_frame_with_grid`, which is `pub(crate)` rather than
// part of the semver surface. Same fixture file, same assertions;
// moved into the crate-private test module when the grid entry-point
// was demoted from `pub #[doc(hidden)]`.