tcd/README.md

tcd - Transcode Detector
=========================

`tcd` analyses an audio file's frequency spectrum to determine whether it is a
genuine native encode or a *transcode* (a lossy → lossless re-encode). It can
also detect *upscaling* (a lossy file that has been re-encoded at a higher
bitrate by the same lossy codec, e.g. 128 → 320 kbps MP3).

---

Obligatory AI-slop disclaimer
-----------------------------
`tcd` is 98% vibe-coded (a.k.a. "ai slop"). If that's a problem for you, please
kindly just use a different tool. There is also absolutely *NO* guarantee this
will work reliably, be useful in any way, or even make any sense whatsoever.

Careful! Dragons ahead!
-----------------------
`tcd` *can* and absolutely *will* delete your data. Don't blindly use -a, and
please read at least the --help information and *understand* what -a does.

---

How it works (the short version)
--------------------------------

tcd decodes your audio file, converts it to the frequency domain (like a
graph showing how much energy exists at each frequency), then measures several
properties of that frequency graph. Each property is a clue about whether the
audio was produced by a lossy encoder. Combined, these clues give a verdict.

---

What tcd displays and what each number means
---------------------------------------------

Here is example output from an MP3 file:

```
File:         ./08 You Got Me.mp3
Format:       mp3
Bitrate:      320 kbps
Sample rate:  44100 Hz
Channels:     2
Windows:      13550
Peak:         47.5 dBFS
Cutoff:       20790 / 22050 Hz = 94.3%
Steepness:    20209 Hz
Roughness:    0.323
Band ratio:   0.555
Noise floor:  -56.4 dB
Verdict:      NATIVE
Verdict Info: bandwidth used=94.3% (20790/22050 Hz), expected≥90% for 320 kbps
```

Each metric is explained below.

---

### Cutoff (20790 / 22050 Hz = 94.3%)

**What it is:** The highest frequency where the audio still has measurable
energy. Everything above this point is silence or noise.

**The Nyquist ceiling:** Digital audio is made of snapshots (samples). For CD
quality (44100 snapshots per second), there is a hard limit: you cannot store
a frequency higher than half the snapshot rate = **22050 Hz**. This is called
the *Nyquist frequency*. It is a physical ceiling - higher frequencies simply
cannot exist.

**How lossy encoding changes it:** MP3 and other lossy codecs deliberately cut
off high frequencies to save space. The cutoff gets lower as the bitrate drops:

| Bitrate | Typical cutoff | Audio quality impact |
|---------|---------------|----------------------|
| 320 kbps | ≥20000 Hz (≥90%) | Keeps almost all audible high end |
| 256 kbps | ≥19000 Hz (≥86%) | Still very clean |
| 192 kbps | ~17500 Hz (~79%) | Moderate high-end roll-off |
| 128 kbps | ~16000 Hz (~73%) | Noticeable treble loss |
| 96 kbps | ~13000 Hz (~59%) | Significant high-end missing |
| 64 kbps | ~11000 Hz (~50%) | Sounds dull, heavily filtered |

**What 94.3% means for the example file:** 20790 / 22050 = 94.3%. The cutoff is very
close to the theoretical maximum. This is what we expect from a 320 kbps
encode. If this same file showed 54% (~12000 Hz), it would mean the treble
was chopped off by an aggressive low-bitrate encoder, and someone just
re-encoded it at 320 kbps - the cutoff is permanent and cannot be restored.
That would be an **UPSCALED** file.

---

### Steepness

**What it measures:** How abruptly the sound drops off *at* the cutoff point.
tcd measures this as the frequency gap between the −20 dB point (still loud)
and the −60 dB cutoff (essentially silent). A narrow gap = a sharp drop.

**The analogy:** Imagine the frequency graph as a mountain ridge. A lossless
recording rolls off like a natural hillside - gradual, smooth, taking
thousands of Hz to go from loud to silent. A lossy encoder's lowpass filter
creates a cliff - a near-vertical drop from audible signal to nothing.

**What the number means:** Steepness is the width (in Hz) of that drop zone.
The smaller the number, the sharper the cliff:

| Steepness | What it looks like | Likely origin |
|-----------|-------------------|---------------|
| <500 Hz   | Brick-wall drop    | Lossy encoder (MP3, AAC) |
| 500–2000 Hz | Fairly sharp    | Could be lossy or aggressive production filter |
| 2000–5000 Hz | Moderate       | Might be natural |
| >5000 Hz  | Gentle slope       | Natural acoustic roll-off (lossless) |

To understand steepness, imagine a guitar string being plucked. The sound
naturally fades across many frequencies - the harmonics near the top end of
your hearing get quieter and quieter over a broad range. This is a gentle
slope. Now imagine someone put a pair of scissors on the frequency spectrum
and cut everything above a certain note. That sharp edge - the difference
between "still audible" and "completely gone" in just a few hundred Hz - is
what lossy compression does. The steepness number tells you how sharp that
scissor cut was.

---

### Roughness (0.323)

**What it measures:** How "bumpy" or "irregular" the spectrum looks just
before the cutoff point.

**The analogy:** Lossy encoding introduces quantization noise - tiny
rounding errors that are unevenly distributed across frequencies. In the
frequency graph, this looks like a jagged, bumpy line instead of a smooth
one. Think of it like a dirt road vs a paved highway: lossless audio is
smooth, lossy audio is bumpy. Double-encoded audio (a transcode) is even
bumpier because the errors from two encodings stack on top of each other.

**What the number means:**

| Roughness | What it looks like | Likely origin |
|-----------|-------------------|---------------|
| <0.15     | Very smooth       | Natural/lossless |
| 0.15–0.30 | Slightly bumpy    | Could be lossy single encode |
| 0.30–0.50 | Clearly bumpy     | Lossy single encode, or borderline transcode |
| >0.50     | Very jagged       | Almost certainly a transcode |

---

### Band ratio (0.555)

**What it measures:** How much high-frequency energy (16–20 kHz) remains
compared to mid-high energy (12–16 kHz).

**Why it matters:** Human hearing is least sensitive above 16 kHz. Lossy
encoders exploit this by spending almost no bits on those frequencies. The
result is that the 16–20 kHz region is much quieter than the 12–16 kHz region.
In native recordings, this drop is modest; in lossy/transcoded material, it
is severe.

**What the number means:** Band ratio = energy in 16–20 kHz band ÷ energy in
12–16 kHz band. A ratio of 1.0 means both bands are equally loud. A ratio of
0.5 means the top band is half as loud.

| Band ratio | What it means |
|------------|---------------|
| >0.85      | Healthy high end - likely native lossless |
| 0.70–0.85  | Mild roll-off - could be lossy or natural |
| 0.50–0.70  | Significant high-end loss - likely lossy |
| <0.50      | Severe high-end loss - almost certainly lossy or transcoded |

---

### Noise floor (-56.4 dB)

**What it measures:** The average noise level in the highest quarter of the
frequency range (roughly 16500–22050 Hz).

**The analogy:** Imagine listening in a quiet room - the background hiss is
very low. Now imagine that same room with a fan running - the background
noise rises. A lossy encoder introduces quantization noise that raises the
"background hiss" in the high frequencies.

**What the number means:** This is measured in decibels (dB). More negative =
quieter (better). Less negative = noisier (worse):

| Noise floor | What it means |
|-------------|---------------|
| −90 to −110 dB | Very clean - native lossless |
| −70 to −90 dB | Moderate - could be lossy or quiet lossless |
| −50 to −70 dB | Noisy - likely lossy |
| >−50 dB      | Very noisy - almost certainly lossy or transcoded |

---

How tcd combines these clues into a verdict
---------------------------------------------

tcd does not rely on any single metric. It combines them in stages, like a
detective building a case.

### Scenario 1: The input file is lossy (MP3, AAC, etc.)

The file already claims to be lossy. The question is: was it *originally*
encoded at the stated bitrate, or was it re-encoded from a lower bitrate?

**The check:** tcd compares the cutoff against what is expected for that
bitrate:

| Stated bitrate | Expected cutoff |
|----------------|-----------------|
| <192 kbps      | ≥75% of Nyquist |
| 192–255 kbps   | ≥85% of Nyquist |
| ≥256 kbps      | ≥90% of Nyquist |

If the actual cutoff is **more than 8 percentage points lower** than expected,
the file is **UPSCALED**. For example, a file claiming 320 kbps (expecting
≥90%) but showing a cutoff of 70% (≈15400 Hz) would be flagged as upscaled
from ~96 kbps.

Otherwise it is **NATIVE** - a genuine single encode at this bitrate.

### Scenario 2: The input file is lossless (FLAC, WAV, ALAC, etc.)

The file claims to be lossless. The question is: was it actually created by
decoding a lossy file and re-encoding to lossless?

tcd uses a **two-layer** check:

**Layer 1 - Cutoff + Steepness (primary):**

| Cutoff range | Max steepness allowed | If exceeded → |
|-------------|----------------------|---------------|
| <50% of Nyquist | 4000 Hz | TRANSCODE |
| 50–70% | 3000 Hz | TRANSCODE |
| 70–80% | 2000 Hz | TRANSCODE |
| 80–90% | 1200 Hz | TRANSCODE |
| ≥90% | 500 Hz | TRANSCODE |

This works because lossy cutoffs are always sharp (low steepness). A lossless
recording that happens to have a low cutoff (e.g., a muddy recording with
little treble) would still have a *gradual* roll-off (high steepness) - you
need both a low cutoff **and** a sharp drop to convict.

**Layer 2 - Roughness + Band ratio (secondary):**

If Layer 1 did not trigger but the cutoff is above 80%, tcd checks roughness
and band ratio. This catches transcodes where the cutoff happens to be high
enough to pass Layer 1 but the spectrum is still bumpy and depleted in the
top band:

| Roughness | Band ratio | If matched → |
|-----------|------------|--------------|
| >0.40     | any        | TRANSCODE |
| >0.30     | <0.90      | TRANSCODE |
| >0.20     | <0.85      | TRANSCODE |

If neither layer triggers, the file is **GENUINE** (native lossless).

---

### The verdicts at a glance

| Verdict | Input codec | What it means |
|---------|------------|---------------|
| **NATIVE** | lossy | Encoded once at the stated bitrate - genuine |
| **UPSCALED** | lossy | Originally encoded at a lower bitrate, then re-encoded higher |
| **GENUINE** | lossless | Appears to be native lossless - no evidence of lossy origin |
| **TRANSCODE** | lossless | Originated from a lossy source, decoded to lossless |
| **SILENT** | any | No detectable audio content |

---

Confidence score
----------------

A continuous **confidence** (0–100 %) is computed using the same metrics with
a sliding scale, providing a graded measure of how certain the tool is about
its verdict.

---

Auto-remove mode (`-a`)
------------------------

When `-a` is passed, any file that is not classified as NATIVE or GENUINE is
automatically deleted after analysis. This is useful for batch cleanup of
corrupt or transcoded libraries. Careful - this will eat data.

---

Why the method is (somewhat!) scientifically reliable
------------------------------------------------------

### 1. Lossy encoding leaves a permanent spectral fingerprint

Every lossy audio codec discards information. The most obvious form is a
**lowpass filter** - once applied, the frequencies above the cutoff are gone
forever. Decoding back to PCM and re-encoding to lossless cannot restore them.
This means a "lossless" FLAC file made from an MP3 will contain the MP3's
permanent spectral cutoff.

### 2. Steepness catches the filter shape

Lossy encoders use sharp digital filters (brick-wall style) that drop from
audible to silent in a few hundred Hz. Natural acoustic sources (voice,
instruments, room ambience) roll off gradually over many kHz. A drop steeper
than 2 kHz at any cutoff position is extremely unlikely to occur naturally.

- **MP3 (ISO/IEC 11172-3):** typical roll-off of several hundred Hz to ~2 kHz
- **AAC (ISO/IEC 13818-7):** sharper, often 200–800 Hz
- **Vorbis:** variable but always steeper than natural

### 3. Roughness detects double-encoding noise

When audio is lossy-encoded, quantization noise is shaped to be masked by the
music. Re-encoding adds a *second* layer of noise-shaping, creating
irregularities in the spectrum that are statistically unlikely in a single
encode.

### 4. Band ratio exploits the Fletcher–Munson curves

Human hearing is least sensitive above 16 kHz. Lossy encoders exploit this by
spending almost no bits there. In native recordings, the 16–20 kHz region is
typically only 2–6 dB quieter than 12–16 kHz (band ratio 0.5–1.0). In
transcoded material it is often 10–20 dB quieter (band ratio < 0.3).

### 5. Multiple independent metrics prevent false positives

No single metric is perfectly reliable. A low cutoff could occur in a genuine
recording that used an aggressive lowpass during mastering. By requiring
**both** a low cutoff **and** a sharp roll-off (primary), or **both** high
roughness **and** a low band ratio (secondary), the tool avoids false
positives.

### 6. The upscaling detector is conservative

For lossy files, an 8-percentage-point margin is subtracted before flagging a
file as upscaled. This accounts for encoder variability and prevents false
positives on legitimate encodes that simply use a conservative lowpass.

---

Usage
-----

```
tcd [options] <audio-file>

 -t, --threshold PCT  Overall detection sensitivity (1-99). Controls all
            decision thresholds: transition bandwidth, roughness,
            and band ratio. Maps to -40 dB cutoff level (1, least
            sensitive) through -80 dB (99, most sensitive).
            [default: 50]. 50 is neutral; lower = fewer detections,
            higher = more detections. Adjust in small steps.
 -f, --fft-size N   FFT size (power of 2) [default: 4096]
 -d, --duration SEC  Max seconds to analyze [default: 60]
 -r, --recursive    Recurse into subdirectories
 -F, --full      Analyze entire file (overrides --duration)
 -v, --verbose     Verbose output
 -s, --visual     Graphical spectrum visualization (TUI)
 -V          Alias for -s
 -a, --auto-remove   Automatically remove non-native files
 -h, --help      Show this help
```

Exit codes:

| Code | Meaning              |
|------|-----------------------------------|
| 0  | NATIVE or GENUINE (file is clean) |
| 1  | UPSCALED or TRANSCODE detected  |
| 2  | SILENT (no detectable content)  |

---

Limitations
-----------

- **Very short files** (< `fft_size` samples) cannot be analysed. Use `-f`
 to reduce the FFT size.
- **Already-lowpass-filtered material** (e.g. deliberate 15 kHz LPF during
 mastering) may trigger false positives. The confidence score helps assess
 borderline cases.
- **High-bitrate lossy encodes** (320 kbps MP3, 256 kbps AAC) have cutoffs
 very close to Nyquist and may not be distinguishable from lossless by
 cutoff alone. The tool relies on roughness and band ratio in this regime.
- **Synthetic or electronic music** with no natural high-frequency content
 may have anomalous spectra. Use the visual mode (`-s`) to inspect the
 spectrum manually.

---

References
----------

- ISO/IEC 11172-3:1993 - Coding of moving pictures and associated audio for
 digital storage media at up to about 1.5 Mbit/s, Part 3: Audio (MPEG-1
 Audio Layer III, "MP3").
- ISO/IEC 13818-7:2006 - Generic coding of moving pictures and associated
 audio information, Part 7: Advanced Audio Coding (AAC).
- Zwicker, E. & Fastl, H. - *Psychoacoustics: Facts and Models*, Springer,
 1999 (Fletcher–Munson equal-loudness contours).
- Lerch, A. - *An Introduction to Audio Content Analysis*, Wiley, 2012
 (spectral features for audio forensics).