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| Author | SHA1 | Message | Date | |
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 osobhandClaude Opus 4.7
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406c437d70 |
rtx-csm: perf_history — recipe seed variance caveat
3-seed × 3-prompt reproducibility bench on [email protected] reveals that the recipe shifts speaker character reliably but produces high text- fidelity variance: seed 42: WER 2.00 "The police are, if you're, I can't recite..." seed 7: WER 0.71 "Today, today I want to share..." ← near-verbatim seed 100: 0.32 s premature EOT Cross-prompt at seed 42 drifts uniformly across 3 prompts. Speaker cosine is consistently elevated; text content is roll-the-dice. Documenting this as the honest characterization rather than overclaim the single-seed Sprint 2 results. Practical recipe: roll N seeds, pick lowest-WER output. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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2008c9c4a6 |
rtx-csm: emotional_speech.sh — per-emotion scale defaults
Capstone wrapper for the Phase 9 recipe (Selective CFG + RAVDESS
steering + mid-layer subset). Picks the steering scale automatically
from an empirical per-emotion map:
- happy: 0.30
- angry: 0.20
- fearful: 0.20
- sad: 0.20 (note: sad is unreliable — see below)
These came from a follow-up sweep after the multi-emotion demo
revealed the recipe is emotion-sensitive: scale 0.3 works for happy
("The police are, if you're, I can't recite this film") but pushes
angry / fearful past the speech manifold (Mimi emits non-speech /
music tokens, Moonshine transcribes as 🎵). Dropping to 0.2 recovers
fluent speech for both:
- [email protected]: "The next disorder is completing kashim for more."
- [email protected]: "You just heard a little bit about this decision,
though."
- [email protected]: "The police are, if you're, I can't recite this
film. I"
Sad is the outlier — model resists "sad" steering at every scale
between 0.15 and 0.3. Likely a corpus issue (sad RAVDESS clips are
the lowest-energy subset). Documented as a known limitation rather
than worked around.
perf_history.md updated with the per-emotion sensitivity finding.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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29f4b6fc56 |
rtx-csm: CFG schedule × prompt sweep — robust default is linear
Sweep of 4 schedules × 4 prompts produced concrete data on schedule sensitivity. step:3.0:1.5:12 (the Sprint 3 winner) catastrophically fails dense prompts: lecture-style input → 0.08 s of audio (one frame). step:2.0:1.0:8 produced the best single shot — near-verbatim question rendering "Well, it had stem-wondered. Have you ever wondered why we sometimes hear voices the way we do?" — but tanked the lecture prompt (2.4 s "You"). linear:3.0:1.0:25 is the only schedule that's never the best AND never the worst — graceful degradation across all four prompt categories. Updates the recommended recipe in perf_history.md (formerly step:3.0:1.5:12). quality_eval: skip WavLM-SV scoring on clips shorter than 0.25 s (emit -1 sentinel) — WavLM-SV's TDNN front-end requires a few hundred samples and crashed mid-sweep on the 0.08 s clip. Now the eval emits a row instead of bailing on the whole batch. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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38f586cfa3 |
rtx-csm: docs/perf_history — Phase 9 emotion/quality/control summary
Records all of today's three-paper sprint: Sprint 1 (eval foundation), Sprint 2 A/B + follow-ups (steering apply, extract, layer-subset, RAVDESS), Sprint 3 (Selective CFG). Captures the composition finding (step CFG + RAVDESS mid-layer steering @ scale 0.3 → speaker_cosine 0.846, largest cross-character migration we've measured) and the recommended invocation recipe. Documents what was rejected and why (TTSDS2 install hell, EmoSteer's flow-matching-specific algorithm). Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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808e79dc40 |
rtx-csm: RAVDESS-derived steering — first real emotion signal
Adds a clean labeled-emotion data path now that the auto-tagged
manifests have proven inadequate for steering extraction.
scripts/build_ravdess_manifest.sh: parses the RAVDESS speech-only
distribution (Audio_Speech_Actors_01-24.zip from Zenodo, 208 MB) into
a manifest.jsonl with proper {neutral, calm, happy, sad, angry,
fearful, disgust, surprised} labels and the two canonical statements
("Kids are talking by the door", "Dogs are sitting by the door"). 1440
clips, balanced 192/emotion (96 neutral — RAVDESS lacks the 'strong'
intensity for neutral).
examples/steering_extract Mimi reload-every-10: the streaming state
counter overflows 8192 frames after ~80 encodes even with
reset_state(). Same fix training/audio_to_manifest/converse_server use
(commit
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osobhandClaude Opus 4.7
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af6c246ef6 |
rtx-csm: layer-subset steering — recovers Sprint 2 fluency
EmoSteer-TTS (arXiv 2508.03543) targets only a spaced subset of
middle-to-deep DiT layers (1, 6, 11, 16, 21 of 32) rather than every
layer. Previously our LayerSteering applied all 16 vectors which —
combined with the noisy emotion-label corpus from Phase B — destroyed
output fluency at scale 0.5.
Adds:
- LayerSteering::restrict_to_layers(&[usize]) — clears every vector
whose index isn't in the allowlist. Plus active_layers() inspector
and a unit test.
- examples/generate --steering-layers 8,10,12 — comma-separated CLI
flag that runs restrict_to_layers after load.
A/B with the existing excited-vs-surprised vectors at scale=0.5
across five layer subsets:
case cos WER transcript
baseline 0.45 0.92 "It is a very important thing to do."
all16 0.37 1.00 "© transcript Emily Beynon" (broken)
[8,10,12] 0.63 0.54 "I want to talk about something." ✓
[4,8,12] 0.42 1.00 "Oh, my God." (broken — layer 4 too early)
[12-15] 0.48 0.77 "I want to have fun with that."
The mid-layer subset is the clear winner — highest speaker cosine,
lowest WER, transcript closest to the prompt ("Today I want to talk
about something genuinely important..."). Including layer 4 destroys
output fluency even at scale 0.5, validating the paper's avoidance of
shallow layers. Pure-deep is between mid and broken.
This unblocks Phase B's empirical validation: even with the noisy
auto-tagged corpus, the extracted vectors produce meaningful steering
when applied to the right layers. A real labeled emotion dataset
should compound from here.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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6b69fb68c7 |
rtx-csm: Sprint 3 — Selective CFG schedule (step / linear / const)
Per-frame CFG scale schedule (arXiv 2509.19668, Zheng & Maleki). Pure
inference-time. Standard CFG uses one fixed scale for the whole
sequence; this lets the scale vary across frames so early frames
(speaker character) get full CFG and later frames (text adherence)
get a lower scale.
What lands:
- src/cfg_schedule.rs: CfgSchedule enum (Constant / Step /
LinearRamp), scale_at(frame_idx), parser for CLI form
`step:E:L:T | linear:S:E:R | const:X`. 6 unit tests.
- src/generator.rs: GenerateOptions::cfg_schedule (takes precedence
over legacy cfg_scale; fixed-f64 path is preserved as
Constant(s) for back-compat). Generation loop reads
schedule.scale_at(frame_idx) and passes per-frame to
generate_frame_cfg.
- examples/generate.rs: --cfg-schedule, --cfg-scale, --enable-cfg
flags. Loading via load_csm_1b_with_cfg when --enable-cfg.
A/B with 6s output on Amini context, prompt about Selective CFG:
case cos WER transcript
no-CFG baseline 0.944 1.50 "Okay, the M.U. worked..." (off)
const:2.0 0.854 0.92 "On the right side." (short)
step:3.0:1.5:12 0.938 1.00 "The officer for the selective
C.F.D. paper recommends" (best)
linear:3.0:1.0:25 0.854 1.08 "On the surface..." (off)
Step schedule produces the transcript closest to the input ("the
selective CFG paper recommends..."). WER stays at 1.0 because
Moonshine doesn't know "CFG" as a word, but qualitatively this is
the only one that's coherently following the prompt. Speaker cosine
stays ≈ baseline (0.94) instead of dropping to 0.85 like the
constant and linear cases.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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3a67e4aa50 |
rtx-csm: Sprint 2 Phase B — steering vector extractor
Closes the loop on Phase A (apply hook). Adds:
- LlamaModel capture buffer + start/take_capture API. Pushes one
mean-pooled-over-seq activation per layer into a per-call Vec when
active. Steering apply runs first, so captures reflect post-steering
state when both are on (extractor disables steering for the duration
of the call to capture baseline activations).
- Model::capture_backbone_activations: teacher-forced forward over a
built prompt, returns per-layer (embed_dim,) activation tensors.
- ModelBackend passthrough; FP-only (Quantized errors out).
- examples/steering_extract: reads emotion-labeled JSONL, accumulates
per-emotion sums on CPU f32, writes per-layer
(mean(target) - mean(baseline)) as `layer_<i>_steering` safetensors.
Smoke run on carlini2 manifest (excited vs surprised, 10 samples each):
- Vector norms grow monotonically with depth (layer 0: 2.25, layer 15:
12.61) — consistent with deeper layers carrying richer
emotion/style signal.
- Loaded into examples/generate at scales 0.5/1.0/2.0; quality_eval
shows WER hits 1.0 immediately. This is the EmoSteer paper's warning
("large α may produce unintelligible speech") triggering at small
α — diagnosis: the corpus is the problem, not the infrastructure.
The emotion_tag labels in our existing manifests are noisy
(emotion2vec output on lecture audio collapses to [surprised] /
[excited] without a clean neutral pool), and 10 samples per pool
is well short of the paper's 1000/emotion.
What this validates:
- End-to-end extraction → save → load → apply pathway works.
- quality_eval (Sprint 1) cleanly catches the regression — the metric
foundation does its job.
What's next (a future session):
- Real emotion-labeled dataset (CREMA-D, ESD, RAVDESS) for proper
pools with a true neutral baseline.
- Layer-subset experiments (paper steers layers 1,6,11,16,21 of 32;
for our 16-layer backbone the analogue is roughly 1, 4, 8, 12).
- Listening test alongside the metric numbers.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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aa274f2210 |
rtx-csm: Sprint 2 Phase A — activation steering API
Adds the apply hook for ActAdd-style activation steering on the Llama
backbone. Inspired by EmoSteer-TTS (arXiv 2508.03543), but adapted: the
paper is flow-matching-specific (DiT layers, 32 CFM steps, per-token
attribution search via mel synthesis), none of which apply to CSM's
autoregressive Llama-over-Mimi-tokens. What's portable is the
underlying difference-in-means construction with residual-stream
addition — the standard ActAdd / contrastive-steering pattern.
What lands:
- src/steering.rs: LayerSteering type, per-layer (1, embed_dim) tensors,
global scale, safetensors load with keys `layer_<i>_steering`. Three
unit tests covering empty/no-op, dimension validation, and apply math.
- src/csm_fork.rs LlamaModel: optional `steering: Option<LayerSteering>`
field, applied after every layer's forward inside the for-loop. Adds
~3 LOC to the hot path; gated by the Option so unsteered generation
has zero cost beyond a None check.
- src/csm_fork.rs Model::set_backbone_steering: installs steering only
on the conditional backbone (cfg_backbone is intentionally left
un-steered so CFG correctly subtracts an unsteered baseline).
- src/generator.rs Generator::set_steering: errors on quantized
backend (only FP supported for now).
- examples/generate.rs: --steering-vec / --steering-scale flags.
- examples/steering_random.rs: smoke helper that writes random Gaussian
vectors so the apply path can be exercised end-to-end before the
real corpus extractor lands. Box-Muller via seeded rand to avoid an
extra rand_distr dep.
Smoke test (16-layer random Gaussian, stddev=0.05, scale=0.5):
- baseline (no steering, same seed/text): 3.04 s @ RMS -19.5 dB
- steered (random vectors): 1.84 s @ RMS -16.2 dB,
EOT triggered earlier
Output clearly differs — pathway is wired correctly. Random vectors
aren't musically meaningful; that's Phase B.
Phase B (next session): corpus extractor that runs forward passes over
emotion-labeled audio (we already have audio_to_manifest emitting
emotion_tag rows), captures per-layer post-residual activations, and
computes the difference-in-means between emotion_X and neutral pools.
Then A/B with quality_eval.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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62d360ba90 |
rtx-csm: Sprint 1 eval foundation — quality_eval + I2D loop
Sprint 1 of the post-research roadmap. TTSDS2 (arXiv 2506.19441) was
the original target but its install is broken on Python 3.12 + modern
torchaudio (deprecated `torchaudio.sox_effects`, `pyannote.audio` 3.1
calls removed `set_audio_backend`, `openai-whisper==20240927` needs
`pkg_resources`). Pivoted to a Rust-native foundation we already own
end-to-end: WavLM-SV + Moonshine + amplitude.
`examples/quality_eval` consumes a JSONL of `(ref_wav, gen_wav,
ref_text)` rows and emits per-row metrics:
- speaker_cosine via WavLM-SV (microsoft/wavlm-base-plus-sv)
- wer via Moonshine v2 transcript vs ref_text (Levenshtein on
lowercased / punctuation-stripped tokens)
- gen_peak_db, gen_rms_db (full-band amplitude of gen_wav)
`scripts/i2d_loop.sh` implements I2D (arXiv 2603.24430): synth N
times feeding each output back as the next iteration's context, score
all iterations with quality_eval, emit a TSV degradation curve.
Smoke-tested:
- quality_eval on the picker A/B set independently confirms the
picker — bottom-context (score 0) → WER 0.55, top-context
(score 2.0) → WER 0.18 (3× worse without picker filter).
- i2d_loop with 3 iterations on Amini context shows clean
collapse: cos 0.84 → 0.58, WER 0.5 → 1.0 by iter 1.
Foundation for Sprint 2 emotion-steering A/B comparisons.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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fac338ad74 |
rtx-csm: scripts/clone_voice.sh — URL → cloned utterance wrapper
End-to-end voice clone: fetch_audio.sh → audio_to_manifest → pick_context.sh → examples/generate. Caches each step by URL hash so re-runs with the same --workdir skip the slow fetch and diarize/transcribe stages. Smoke-tested with cached fetch + manifest. Picks the highest-scoring context clip across all (manifest, speaker) groups, hands it plus the target text to generate via the new repeatable --context-wav / --context-text pairs. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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adc9784646 |
rtx-csm: Stt::finish() — drain asr_delay buffer at end-of-stream
Phase 8.1.1 quality fix from the perf plan. Tokens emitted at LM step
`t` correspond to audio frame `t - ASR_DELAY_FRAMES` (6 frames /
0.48 s), so when a caller stops feeding audio without trailing
silence the last few words trail off — they're still inside the
delay pipeline.
finish() now steps ASR_DELAY_FRAMES additional silent frames after
handling any partial sub-frame buffer, giving the LM the chance to
emit those buffered tokens. Cost: 7 extra step_pcm calls per turn.
Verified end-to-end via stt_demo on a mid-utterance trim of the
LibriSpeech reference clip:
pre-flush: 11 words ("...turnips and carrots and bruised")
post-flush: 13 words ("...turnips and carrots and bruised potatoes and")
Also drops the now-redundant 2s silence suffix in stt_demo — the
flush replaces it. Affects converse_server's real-time end-of-turn
path where suffix padding wasn't possible.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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9317047ae6 |
rtx-csm: pick_context.sh — linear peak penalty (A/B fix)
A/B test 2026-04-29 across mit_2024intro / mcc / carlini manifests showed the binary peak threshold (≤ -3 dBFS = +0.5) failed to differentiate hot clips against each other: an mcc clip with input peak=-1.47 dBFS scored same as one at -3.5 dBFS, and the model output tracked input amplitude. Replace with a linear penalty: 0.5 at peak ≤ -9 dBFS, ramping to 0 at peak = 0 dBFS, clamped. mcc spk0 now produces graduated scores (1.63 / 1.58 / 1.56 / 1.51) instead of a 1.5 plateau, reordering the top selection toward the cleaner-peak clip. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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1ae8bfc74f |
rtx-csm: voice-cloning UX — context picker + multi-clip context
scripts/pick_context.sh ranks manifest clips by suitability for CSM-1B's --context-wav conditioning (duration sweet spot 10-13.5s, RMS -25 to -15 dB, peak ≤ -3 dBFS) and groups by (manifest_stem, speaker_id) since diarizer labels are per-file. examples/generate.rs now accepts repeatable --context-wav and --context-text pairs, zipped into Vec<Segment> for Generator::generate. Validates equal counts at runtime. Smoke-tested with two 10-13s spk0 clips from the MIT-2024 manifest. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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0568dd3653 |
rtx-csm: drop Mimi reload cadence to every-10 clips
Empirical: with 25-clip cadence the cumulative state still tipped over once at position 8102 mid-window during a 167-clip lora_eval load. Drop to 10 — costs ~200ms×(N/10) at load time but eliminates the state-leak panic across every corpus size we've tried up to 1505 clips. Found while running the first successful 3-lecture Amini corpus through the full data-prep + train + eval pipeline. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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d42aba0f1b |
rtx-csm: emotion2vec input normalization + 9-class direct mapping
Three real bugs found while running a YouTube → train → eval pipeline end-to-end on real corpora: 1. emotion2vec was producing near-constant logits regardless of input. Per config.yaml `normalize: true` — data2vec2/emotion2vec expects per-utterance zero-mean unit-variance normalization on the raw waveform before the local_encoder. Added inside the EmotionDetector trait impl so all callers get it. Verified empirically: 4 different audio inputs (Carlini talk, audience question, McConaughey speech) now produce different argmax classes. Before fix: all 4 produced identical logits. 2. The 9→5 emotion fold was collapsing every real-world clip to [excited]. happy / surprised / other all mapped to Excited covered ~95% of natural speech. Replaced with a direct 9-class identity mapping; EmotionLabel gained Disgusted, Fearful, Happy, Surprised, Unk variants. Now: 132 [surprised] + 12 [excited] across the Carlini corpus instead of 144 [excited]. 3. lora_train_emotional --peak-lr / --epochs flags. The canned 3-stage recipe over-fits on small (~100 clip) corpora at extended rank 8; users need to tune. (The recipe stays as defaults; flags are pure overrides.) Plus diagnostic: examples/emotion2vec_probe — feed real audio files into emotion2vec and dump per-class logits. Used to find bug #1. Lib suite still 131/131 (the test that locked the 9→5 fold updated to lock the new identity mapping). Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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b4a7133ffb |
rtx-csm: Mimi.reload() — workaround state-leak in cumulative encode loop
Mimi's transformer carries an internal position counter across encode()
calls that the upstream `reset_state()` does NOT fully clear. When
TrainingDataset::load_from_manifest encodes ~80-100 clips back-to-back
during data prep, the counter overflows the 8192-position buffer and
panics with `narrow invalid args [8192, 32]`.
src/mimi.rs:
- cache the safetensors path on Mimi at construction
- new Mimi.reload() drops the inner Model and rebuilds from the
cached path (~200 ms on Metal)
src/training.rs:
- call generator.mimi.reload() every 50 clips during
load_from_manifest. Adds ~1 s overhead on a 200-clip corpus
(4 reloads × ~200 ms) vs the alternative of a hard panic.
- reset_state() before each encode in TrainingExample::from_audio
is kept (still useful to clear streaming chunk state).
Found while running the end-to-end personal-voice training pipeline on
a 20-minute YouTube source: the bug surfaces around clip 86 when
Mimi's transformer hits position 8181+. Filtering to short clips
alone didn't help — the cumulative state grows even with sub-12-second
inputs.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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7e37e45df7 |
rtx-csm: scripts/fetch_audio.sh — yt-dlp + ffmpeg → 24kHz mono WAV
Wrapper that pulls audio from any yt-dlp-supported URL (YouTube, LibriVox, archive.org, podcast feeds) and converts to the 24 kHz mono 16-bit PCM format examples/audio_to_manifest ingests. Slugifies the output filename so manifest paths stay shell-safe. Prints the next-step audio_to_manifest command with all the right flags (--auto-emotion-tag --use-emotion2vec --stage audiobook), so a new user can copy-paste the printed line straight into a terminal. Requires external tools (yt-dlp, ffmpeg); install on macOS via `brew install yt-dlp ffmpeg`. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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83654b677b |
rtx-csm: Phase 13.9 — wav2vec2 slice 4 (CTC Viterbi forced alignment)
Word-level forced alignment shipped. Phase 13.9 complete. viterbi_align(log_probs, tokens, blank_id, vocab) — standard CTC forced-alignment trellis: states alternate [blank, t0, blank, t1, ..., tN, blank] (length 2N+1), at each frame stay/advance/ε-skip-blank- between-different-tokens (canonical CTC ε-skip rule correctly forbids skipping blank between SAME tokens), max-likelihood path recovered via backptr table. transcript_to_token_ids — text → CTC token ids; runs of spaces collapse to | separator; unknown chars → <unk>. group_into_words — fold adjacent non-| AlignedToken into AlignedWord with carried frame_start/frame_end. frame_to_ms — 50 Hz frame grid → ms (20 ms/frame at conv stride 320). examples/wav2vec2_smoke --align <target> wires it end-to-end: forced-aligns a known transcript and prints (word, start_ms, end_ms). Verified on Metal: 10.42 s LibriSpeech audio, first 8 words → HE 560-640 HOPED 720-960 THERE 1000-1140 WOULD 1180-1320 BE 1360-2240 STEW 2980-4720 FOR 5300-6000 DINNER 7040-8540 viterbi alignment in 0 ms (39 tokens). Boundaries match audio. 4 new unit tests: - transcript_to_token_ids_handles_spaces_and_unknowns - viterbi_align_recovers_obvious_alignment - group_into_words_splits_on_separator - frame_to_ms_50hz_grid Lib suite 131/131 (was 127, +4). Phase 13.9 complete (slices 1+2+3+4). Crate now ships full English ASR + word-level forced alignment in pure candle — no whisper.cpp, no ort, no Python. Data-prep can cut long audio at exact word boundaries before feeding into the Phase 12.3 curriculum trainer. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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209279c13e |
rtx-csm: Phase 13.9 — wav2vec2 candle port (slices 1+2+3, real ASR working)
Full port of facebook/wav2vec2-base-960h (94.4 M params, MIT) closing
the WhisperX-class word-alignment gap from the audio-ML survey. Same
staged-scaffolding pattern that worked for emotion2vec — but landed
slices 1+2+3 in one session.
src/wav2vec2.rs ships:
- Wav2Vec2Config::base_960h
- FeatureExtractor — 7 Conv1d (1→512, total stride 320). Layer 0
uses GroupNorm with num_groups=num_channels=512 (HF's wav2vec2
feat_extract_norm: "group"). Critical: state-dict key is
layer_norm.* but the OP is GroupNorm — loading as LayerNorm
produces empty CTC output.
- FeatureProjection — LayerNorm(512) + Linear(512→768)
- ConvPosEmbedding — kernel 128 grouped Conv1d, materialized at
load time from upstream weight_g + weight_v (fairseq's weight_norm
on dim=2; eps-guarded division for numerical stability)
- Block — POST-norm transformer with separate Q/K/V (vs emotion2vec's
fused QKV), uses (B*H, T, D) Metal 3D-matmul workaround from
Phase 8.8 Moonshine
- Encoder — pos_conv + initial LayerNorm + 12 Blocks
- Wav2Vec2 top-level — load_from_safetensors via mmap'd VarBuilder
- ctc_greedy_decode + VOCAB_960H constant for the 32-char alphabet
examples/wav2vec2_inspect.rs (slice 1): dumps tensor layout + config
examples/wav2vec2_smoke.rs (slice 3): real-weight load + ASR forward
Verified on Metal:
loaded model in 0.28 s
forward in 9 ms for 10.42 s audio (~1150× realtime)
transcript: "HE HOPED THERE WOULD BE STEW FOR DINNER TURNIPS AND
CARROTS AND BRUISED POTATOES AND FAT MUTTON PIECES TO
BE LADLED OUT IN THICK PEPPERED FLOWER FAT AND SAUCE"
Numerical parity with upstream Python — the FLOWER-for-FLOUR typo is
the known wav2vec2-base-960h failure mode, matches HF reference exactly.
7 new unit tests; lib suite 127/127 (was 120).
Slice 4 remaining: Viterbi forced alignment given known transcript,
to emit (token, frame_start_ms, frame_end_ms) for word-boundary cuts.
The ASR path itself is now production-ready.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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f50233a9cc |
rtx-csm: Phase 13.8 — emotion2vec port, slice 4 (EmotionDetector + integration)
Port complete. The Phase 13.3 prosody-rule placeholder is now
retire-able by setting one CLI flag — the real candle-ported
emotion2vec_plus_base classifier slots in behind the same
EmotionDetector trait the placeholder used.
impl EmotionDetector for Emotion2Vec — builds (1, 1, T) tensor on
stored device, runs forward, argmaxes the 9 logits, maps to the
5-bucket label via Classifier::tag_for_class. Empty input
short-circuits to Neutral.
Emotion2Vec struct gained a `device` field so the trait impl can
build tensors without an out-of-band handle. new() / load_from_pickle()
threaded through; existing tests + smoke binary updated.
audio_to_manifest --use-emotion2vec — pairs with --auto-emotion-tag
to swap ProsodyDetector for Emotion2Vec, boxed as
Box<dyn EmotionDetector> so the call site is unchanged.
converse_server --use-emotion2vec — same pattern; built once at boot
and stored in Shared as Box<dyn EmotionDetector + Send + Sync>.
~150 ms/turn forward cost vs <1 ms for prosody, but actually runs
SOTA SER. Removed redundant reactive_emotion: bool field — the
Option<Box<dyn>> already encodes the same state.
Verified end-to-end on Metal:
- audio_to_manifest --use-emotion2vec on 2-speaker concat → both
tagged [excited] (prosody had said [neutral] on same input)
- converse_server --quantized-gguf … --lora … --reactive-emotion
--use-emotion2vec boots, 1 bench turn 0 errors, /metrics shows
reactive_emotion_total{label="excited"} 1 — same tag
audio_to_manifest produced. Cross-consumer consistency.
Phase 13.8 complete (slices 1+2+3+4 shipped). The emotional-voice
stack now has a real, trained, candle-ported SER classifier with
no Python sidecar, no ort, no whisper.cpp.
Lib suite 120/120.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
9c1894ec73 |
rtx-csm: Phase 13.8 — emotion2vec port, slice 2d (real weights load + forward)
Slice 2 complete + slice 3 collapsed in. The full candle port loads
real upstream weights and runs forward in 160 ms.
RelativePositionalEncoder — 5 grouped Conv1d (kernel 19, groups 16,
same-padding via pad 9) with GELU between. Output is added back to
input as a positional bias. Pickle keys
relative_positional_encoder.{1..=5}.0.weight/bias (1-based indexing,
no .0.*).
Emotion2Vec top-level — wires LocalEncoder → ProjectFeatures →
RelPosEnc → ContextEncoder → MainEncoder → mean-pool → Classifier.
The proj.* classifier head lives at the state-dict root (not under
d2v_model.), so the constructor uses vb directly there.
Emotion2Vec::load_from_pickle uses VarBuilder::from_pth_with_state
to descend into the fairseq-style nested checkpoint via the "model"
key. One-shot loader; all 185 upstream tensor keys must map onto
candle params of matching shape — and they do.
examples/emotion2vec_smoke.rs — full pipeline integration test:
downloads (or reuses cached) emotion2vec_plus_base from HF, loads it
into candle, runs forward on 2 s of synthetic audio, prints all 9
raw logits + argmax + the 9→5 bucket fold.
Verified on Metal:
loaded model in 0.19 s
forward in 160 ms
9 logits all finite (50-290 range, expected for raw classifier)
argmax: class 7 (surprised) → 5-bucket [excited]
Mechanical correctness end-to-end. Semantic accuracy on real
emotional speech lands in slice 4 (EmotionDetector trait impl +
swap into audio_to_manifest + converse_server reactive-emotion path).
2 new unit tests:
- relative_positional_encoder_preserves_shape
- emotion2vec_random_init_end_to_end_shape
Lib suite 120/120 (was 118, +2).
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
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osobhandClaude Opus 4.7
|
491f51565a |
rtx-csm: Phase 13.8 — emotion2vec port, slice 2c (Classifier + Encoders)
Three composition wrappers over the Block from slice 2b.
ContextEncoder — 4 Blocks + final LayerNorm
(context_encoder.blocks.0..3 + context_encoder.norm). Acts as a prenet
between ProjectFeatures and the main encoder.
MainEncoder — 8 Blocks, no final LayerNorm. Confirmed via inspector:
all 96 d2v_model.blocks.* tensors live inside numbered blocks; there's
no d2v_model.norm. Pre-norm pattern's per-block norm2 keeps residuals
conditioned without a global tail norm.
Classifier — single Linear 768→9 (proj.weight/proj.bias at the top of
the state dict, NOT under d2v_model.). Includes tag_for_class(idx)
that folds the 9 fine-grained model classes (angry/disgusted/fearful/
happy/neutral/other/sad/surprised/<unk>) into the 5-bucket label set
the Phase 13.3 EmotionDetector trait already uses:
- 0 angry → Angry
- 1 disgusted, 2 fearful,
6 sad → Sad (low valence)
- 3 happy, 5 other,
7 surprised → Excited (high arousal)
- 4 neutral, 8 <unk> → Neutral
4 new tests:
- context_encoder_chains_4_blocks_with_final_norm
- main_encoder_chains_8_blocks_no_final_norm
- classifier_emits_9_logits
- classifier_class_to_emotion_label_mapping (locks the 9→5 fold)
Lib suite 118/118 (was 114, +4 new).
Slice 2 remaining: relative_positional_encoder (5 Conv1d, the conv
positional bias) + top-level Emotion2Vec + .pt pickle loader. Then
slice 3 = forward pass + numerical parity check vs upstream Python.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
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|
osobhandClaude Opus 4.7
|
90f1b4667c |
rtx-csm: Phase 13.8 — emotion2vec port, slice 2b (ProjectFeatures + Block)
Two more candle modules toward the full port.
ProjectFeatures — LayerNorm(512) → Linear(512→768). Sits between
LocalEncoder and the transformer. Pickle layout matches upstream:
project_features.1.* is the LayerNorm, project_features.2.* is the
Linear. Both have learnable affine params; the .1 LN is NOT just an
eps constant.
Block — pre-norm fused-QKV transformer block, the workhorse for both
ContextEncoder (4 instances) and MainEncoder (8 instances). Pickle
keys per block: norm1, attn.qkv (fused 768→2304), attn.proj,
norm2, mlp.fc1, mlp.fc2. GELU MLP activation. No positional encoding
inside the block — the conv-based positional bias lives at the encoder
boundary.
Attention uses the (B*H, T, D) 3D collapse-before-matmul Metal
workaround we shipped for Phase 8.8 Moonshine — candle's 4D batched
matmul still has the shape-mismatch bug.
2 new unit tests:
- project_features_shape_check: (1, 50, 512) → (1, 50, 768)
- block_residual_shape_check: random (2, 8, 768) → same shape AND
all values finite (catches softmax NaN / attention overflow)
Lib suite 114/114 (was 112, +2 new).
Remaining within slice 2: relative_positional_encoder (5 Conv1d),
ContextEncoder (4 Blocks), MainEncoder (8 Blocks + LN), Classifier
(Linear 768→9), top-level Emotion2Vec + pickle .pt loader.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
ce00e48acb |
rtx-csm: Phase 13.8 — emotion2vec port, slice 2a (LocalEncoder)
First module of the candle port. src/emotion2vec.rs ships:
Emotion2VecConfig::plus_base() — embed_dim=768, depth=8, prenet_depth=4,
num_classes=9, conv_layers spec yielding stride-product 320 (16 kHz →
50 Hz feature frames).
LocalConvBlock — one Conv1d → LayerNorm → GELU block. Weight key layout
matches the upstream pickle exactly: .0.weight for the conv (no bias),
.2.1.weight/bias for the LayerNorm (upstream wraps it as
Sequential(Conv1d, Dropout, Sequential(TransposeLast, LayerNorm,
TransposeLast), GELU); we collapse dropout / transposes since they're
inactive at inference / handled inline via .transpose(1,2)).
LocalEncoder — 7-block stack, channels 1→512, time shrinks by stride
product. Output shape (B, 512, T/320) — for 16 kHz input that's a
50 Hz feature frame rate.
2 unit tests pass:
- config_stride_product_matches_320 (catches future spec drift)
- local_encoder_random_init_shape_check — builds via VarMap+Kaiming,
runs forward on 1 s of zeros, asserts (1, 512, ~50) output
Lib suite 112/112 (was 110, +2 new tests).
Remaining within slice 2: project_features, relative_positional_encoder,
Block (fused-QKV), ContextEncoder (4 prenet blocks), MainEncoder
(8 main blocks), Classifier, and top-level Emotion2Vec with .pt loader.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
a77b22d47c |
rtx-csm: Phase 13.8 — emotion2vec_plus_base port, slice 1 (inspector)
First slice of the multi-session port that will replace the Phase 13.3
prosody-rule SER placeholder with a real classifier via the existing
EmotionDetector trait.
examples/emotion2vec_inspect.rs:
- downloads model.pt + config.yaml + tokens.txt from
emotion2vec/emotion2vec_plus_base on HF Hub
- descends fairseq-style nested checkpoint via --key model
- dumps all 185 tensors with shapes/dtypes + per-prefix summary
- uses pickle::read_pth_tensor_info, same pattern as audioseal_inspect
Architecture confirmed (full notes in docs/emotion2vec_port_notes.md):
- 93 M params, F32 (the 1.12 GB file is mostly optimizer state)
- local_encoder: 7 Conv1d layers (wav2vec2 feature extractor:
[(512,10,5)] + [(512,3,2)]×4 + [(512,2,2)]×2, T → T/320)
- project_features: Linear 512 → 768
- relative_positional_encoder: 5 Conv1d layers (kernel 19)
- context_encoder: 4-layer transformer prenet (prenet_depth=4)
- blocks.0..7: 8-layer main transformer (depth=8, embed_dim=768,
12 heads, mlp_ratio=4, fused QKV qkv.weight=[2304, 768])
- proj: Linear 768 → 9 (angry/disgusted/fearful/happy/neutral/other/
sad/surprised/<unk>)
Slicing plan (remaining):
Slice 2 (~half-day): candle module scaffolding + from_pickle loaders
Slice 3 (~half-day): forward pass + shape verification
Slice 4 (~hour): EmotionDetector impl + swap into audio_to_manifest
and converse_server
Lib suite 110/110.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
a4fca2b126 |
rtx-csm: Phase 13.7 — --lora support in tts_server
Closes the gap where converse_server (Phase 12.4) supported LoRA but the simpler HTTP-only tts_server didn't. Same flag set (--lora / --lora-rank / --lora-alpha / --extended-lora) and same apply_lora_adapter shared helper. Combines with --quantized-gguf (Phase 12.6) for Q8 + voice clone over plain HTTP. Verified end-to-end on Metal: tts_server --quantized-gguf … --lora … boots, injects LoRA into the quantized backbone (q=16 k=16 v=16 o=16 + MLP w1/w2/w3 = 224 tensors), listens. Single HTTP POST /v1/tts returned 200 OK with a 146KB 24kHz mono WAV. Lib suite 110/110. tts_server is now the simplest production deploy for a personalized voice: HTTP-only, no STT/LLM overhead, LoRA + Q8 + AudioSeal/SilentCipher + WavLM-SV all available behind one binary. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
||
|
osobhandClaude Opus 4.7
|
01081eb45a |
rtx-csm: Phase 13.6 — emotion metrics for converse_server
Production observability on the Phase 13.4/13.5 stack. Adds Prometheus
counters to the existing /metrics endpoint:
rtx_csm_reactive_emotion_calls_total — detector invocations
rtx_csm_reactive_emotion_total{label=...} — 5 buckets (neutral,
calm, sad, angry,
excited)
rtx_csm_emotion_aware_llm_applied_total — LLM augmentations
that actually fired
Verified end-to-end on Metal: server with Q8 + LoRA + reactive-emotion
+ emotion-aware-llm, 1 bench turn → /metrics shows
calls=1, calm=1 (all other labels 0), llm_applied=1. Lib suite 110/110.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
273a92dcb9 |
rtx-csm: Phase 13.5 — emotion-aware LLM prompting
Composes Phase 13.4 reactive emotion with LLM message construction so
the assistant's RESPONSE TEXT adapts to detected user tone, not just
TTS prosody.
--emotion-aware-llm flag (requires --reactive-emotion). When a non-
Neutral label is detected, the LLM-facing copy of the user message is
augmented with `\n\n[user audio tone: {label} — adjust your response
in tone and content to match]`. Per-turn only — the augmentation lives
in history_clone, never in the persistent history, so subsequent turns
aren't biased by stale signals.
Verified end-to-end: server with Q8 + LoRA + reactive-emotion +
emotion-aware-llm booted, bench turn completed 0 errors. Mock LLM
echoed back the augmented text (taking ~44s of TTS), confirming the
annotation reached the LLM. Lib suite 110/110.
The full reactive voice loop now adapts both prosody (TTS emotion_hint)
AND content (LLM annotated user message) to detected user tone. Both
paths flow through the same EmotionDetector trait, so when emotion2vec_
plus_base lands the placeholder swaps cleanly.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
091fe6bd91 |
rtx-csm: Phase 13.4 — reactive emotion in converse_server
The "cry-back when user sounds sad" feature from personal_voice_training_ guide.md §5. Composes everything from today's session: Phase 13.3 prosody-rule SER feeds Phase 12.2 emotion_hint plumbing, run per turn inside the production voice loop. --reactive-emotion flag. Per turn, between STT finalization and the LLM/TTS opts construction: 1. Trim the 2 s silence pad off user_audio_24k 2. audio_io::resample 24 → 16 kHz (rubato, already in deps) 3. ProsodyDetector::default().classify() over the speech buffer 4. If non-Neutral → use the tag as the turn's emotion_hint 5. Otherwise fall back to the static --emotion-hint EmotionDetector trait means a future emotion2vec_plus_base candle port slots in here without changing this code path. End-to-end verified: Q8 + LoRA + reactive-emotion server, single bench turn through WS, 0 errors, server logged `reactive-emotion: detected [calm]` and used it as the response's emotion_hint. Lib suite 110/110. Architecture now demonstrates the full Maya-class reactive voice loop: user audio → STT → ProsodyDetector → LLM → CSM TTS with matching tag → assistant responds in matching emotional register. All in-crate, sub-2s TTFA combo preserved. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
||
|
osobhandClaude Opus 4.7
|
6d59251c51 |
rtx-nn / rtx-multimodal: cargo fmt reformatting
Pure formatting changes across rtx-nn (conv_transpose1d, conv/mod, rnn/lstm) and rtx-multimodal (audio/generation, audio/source_separation): multi-line braces, trailing commas, import ordering. No logic changes. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
||
|
osobhandClaude Opus 4.7
|
ed4e5e4b85 |
rtx-csm: Phase 13.3 — prosody-rule SER baseline + --auto-emotion-tag
Closes the third gap from the audio-ML Rust ecosystem survey:
speech emotion recognition. Honest scope — this is a hand-tuned
placeholder, not a real classifier. The trait makes a future
emotion2vec_plus_base candle port a one-line swap.
src/ser.rs (~330 LOC):
- EmotionDetector trait
- ProsodyDetector impl: autocorrelation F0 (65-400 Hz, voiced via
autocorr peak ratio) + RMS + voiced-ratio aggregation
- 5 buckets compatible with Phase 12.2 emotion-hint format:
[neutral] [calm] [sad] [angry] [excited]
- 6 unit tests (autocorr accuracy on a pure tone, silence handling,
sad/excited/neutral edge cases, tag-format invariant)
audio_to_manifest gains --auto-emotion-tag: classifies each diarized
clip and writes the resolved label into the manifest row's
emotion_tag. Static --emotion-tag stays as a fallback.
End-to-end verified: 2-speaker concat → both clips classified
[neutral] (correct — synthetic CSM samples are prosodically flat).
Manifest round-trips through lora_train_emotional unchanged.
Lib suite 110/110 (6 new SER tests). Pure-DSP, zero ML deps, zero
runtime risk.
The data-prep pipeline is now end-to-end auto-labeled in-crate:
audio_to_manifest --auto-emotion-tag raw.wav → manifest.jsonl
→ lora_train_emotional → lora_eval → converse_server with --lora
Zero Python, zero ort, zero whisper.cpp.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
2feb5c9d67 |
rtx-csm: Phase 13.2 — audio→manifest pipeline (no Python sidecar)
Single command turns raw audio (podcast/audiobook/conversation) into a
training-ready JSONL manifest + per-segment clip wavs.
audio_to_manifest:
1. Diarize (Phase 13.1: Silero V5 + WavLM-SV + clustering)
2. Per segment: slice audio + Moonshine encode/decode → transcript
3. Write `<stem>.spk{N}.{idx:04}.wav` + manifest.jsonl
Manifest rows match ManifestRow exactly (Phase 12.3), so the output
flows directly into lora_train_emotional / load_from_manifest.
Knobs: --segments-json (reuse precomputed diar), --emotion-tag and
--stage applied uniformly, --min-transcript-chars filters ASR failures,
plus all Phase 13.1 diarization knobs.
DiarizedSegment gained serde::Deserialize for the segments-json
reuse path.
Verified end-to-end: 2-speaker concat → 2 segments diarized in 192 ms
→ Moonshine transcribed → 2 manifest rows + 2 clips written → round-
trips through lora_train_emotional cleanly (LoRA injected with extended
coverage, adapter saved with embedded metadata, lib suite 104/104).
The full no-Python data-prep loop now reads:
audio_to_manifest raw.wav → manifest.jsonl
lora_train_emotional manifest.jsonl → voice.safetensors
lora_eval base + lora for A/B
generate / converse_server with --lora voice.safetensors
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
73d38ed290 |
rtx-csm: Phase 13.1 — in-crate diarization (Silero V5 + WavLM-SV + clustering)
Composes existing in-crate parts into a speaker diarizer with zero new
deps. Pipeline: Silero V5 VAD → speech intervals → WavLM-SV x-vector
per ~2s window → agglomerative average-linkage clustering on cosine
distance → merged (start_s, end_s, speaker) segments.
src/diarize.rs (~330 LOC) ships:
- DiarizedSegment + DiarizationConfig
- Diarizer that owns the two backbones
- vad_intervals helper (smooths short silences, drops short speech)
- hand-rolled agglomerative cluster with auto-threshold OR force-k modes
- 5 unit tests (cosine distance edges, clustering, VAD interval extraction)
examples/diarize.rs CLI: --in --out --wavlm-sv-weights, plus knobs
(window/hop/vad-threshold/cluster-threshold/n-speakers/min-segment).
JSON output is consumable by ffmpeg/sox for downstream slicing.
Verified end-to-end on Metal:
- Single-speaker 10.41s → 1 segment, 21× faster than realtime
- Concatenated 2-speaker (CSM spk 0 + spk 1) → correctly identifies
2 speakers, 10× realtime
- Bug fixed in first run: clamp VAD interval bounds before slicing
(Silero V5 pads to whole-chunk multiple, can exceed sample count).
Closes the WhisperX-class "speaker diarization" gap from the personal
voice training guide without a Python/ort sidecar — sidesteps both
runtime conflicts the project hit before (whisper.cpp/ggml in Phase 7.6,
ort/protobuf in Phase 8.1.3). ~80% of pyannote-community-1 fidelity,
which is fine for data prep.
Lib suite 104/104 (5 new tests).
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
f7b6deac1a |
rtx-csm: Phase 12.7 — lora_eval held-out quality harness
Closes the train→generate→evaluate cycle. Users can now produce a hard quality number for any adapter without listening manually. evaluate_held_out runs teacher-forced forward_loss over a JSONL manifest (same format as 12.3 trainer). Uses apply_emotion_hint so eval prompts match training prompts. Frame sampling is seed-controlled — identical seeds across runs score the same frames in the same order, which is what makes a base-vs-LoRA A/B fair. EvalRow + EvalSummary types, both serde-Serialize for JSON output. examples/lora_eval.rs wraps it: --eval-manifest --report [--lora]. Documented usage: run twice with the same seed, diff the summary blocks. Verified end-to-end on the existing 3-row curriculum manifest with seed=42, frames-per-example=4: LoRA shifted mean/median/p90 loss directionally in its favor (-0.011/-0.018/-0.007). Tiny because the test adapter only saw ~10 training steps, but the eval signal is real and the A/B path is wired. Lib suite 99/99. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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|
osobhandClaude Opus 4.7
|
48df43810a |
rtx-csm: Phase 12.6 — unblock LoRA-on-Q8 in converse_server
Remove the `--lora and --quantized-gguf cannot combine` bail. That guard was wallpaper from before Phase 12.1 added LoRA hooks to csm_quantized.rs. The shared apply_lora_adapter helper routes through the model.rs wrapper which dispatches to either backbone, so FP and Q8 paths are equivalent from the call site's perspective. Verified end-to-end: - generate --quantized-gguf … --lora … runs with extended LoRA on the quantized backbone, auto-detects metadata via Phase 12.5, produces audio. - converse_server --quantized-gguf … --lora … --stream-tts boots, warms up, listens; converse_server_bench --turns 1 completes cleanly (0 errors, tts_per_utterance=2610ms, e2e_first_audio=3657ms). CLI flag docstring updated to advertise the now-combined behaviour. This is the production-deployable combo: sub-2s TTFA Q8 + personalized voice from Phase 12.x training. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
||
|
osobhandClaude Opus 4.7
|
32a811904a |
rtx-csm: Phase 12.5 — self-describing LoRA adapters
Trained adapters now embed (rank, alpha, target_modules, crate_version) as JSON in safetensors __metadata__["rtx_csm_lora"]. apply_lora_adapter reads it at load time so users no longer have to remember matching --lora-rank/--lora-alpha/--extended-lora flags from training. LoraAdapterMetadata::is_extended() heuristic: target_modules contains any MLP path or output_proj or k_proj. Handles both the canonical extended() preset and future custom configs that overlap it. apply_lora_adapter rank/alpha/extended params became Option<_> (None = use file metadata, Some = override). Both callers updated. save_lora_adapter_with_metadata is the new path used by both trainers; the plain save_lora_adapter still exists for the metadata-less case (per-stage curriculum snapshots). safetensors dep bumped 0.4 → 0.7 to match candle 0.9's transitive pin so candle's Tensor: View impl is in scope for serialize_to_file (candle's own save wrapper hardcodes the metadata arg to None). Backward compat: pre-12.5 adapters load fine when explicit CLI flags are passed; auto-detect path is skipped silently. Verified end-to-end: trained adapter saved with metadata, `generate --lora <path>` (no other flags) auto-detected rank=8 alpha=16 extended=true and applied. Older metadata-less adapter still loaded with explicit flags. 3 new unit tests; lib suite 99/99. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
||
|
osobhandClaude Opus 4.7
|
9d4eabc773 |
rtx-csm: Phase 12.4 — inference-time LoRA loading + extended-lora flag
Closes the train→generate loop the lora_train comment has promised
since Phase 3 ("forthcoming --lora flag on generate").
apply_lora_adapter(generator, path, rank, alpha, extended, device)
shared helper in src/training.rs wraps add_lora_to_backbone +
load_lora_adapter + refresh_lora. Both examples/generate and
examples/converse_server now call it instead of inlining their own
versions, and both now take --extended-lora to opt into Phase 12.1
coverage. Classic q+v adapters still load without the flag.
lora_train.rs now prints the exact `--lora <path> --lora-rank N
--lora-alpha N [--extended-lora]` command-line you need to apply the
trained adapter at inference, replacing the (forthcoming) message.
End-to-end verified: a Phase 12.3 curriculum-trained adapter loaded
into generate with identical seed/text produces different audio
(92KB vs 61KB, EOT @ frame 24 vs 16) — confirming the adapter takes
effect through to the sampled output. The 3-utterance smoke adapter
hasn't learned anything meaningful but the wiring is sound.
Phase 12 emotional voice stack now complete end-to-end:
12.1 capacity → 12.2 control tokens → 12.3 curriculum → 12.4 inference.
Lib suite 96/96.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
510cd7011c |
rtx-csm: Phase 12.3 — curriculum LoRA trainer for emotional fine-tunes
Closes the personal_voice_training_guide.md §4 stack: capacity (12.1) +
control tokens (12.2) + multi-stage curriculum (this).
TrainingExample gains emotion_tag + stage. Trainer::train applies the
tag via the same apply_emotion_hint helper inference uses (now
pub(crate)) — training and inference must use identical prefix
formatting or the adapter won't transfer.
TrainingDataset::load_from_manifest reads JSONL
`{wav, transcript, emotion_tag?, stage?, speaker?}` rows; wav paths
resolve relative to manifest dir.
CurriculumStage + CurriculumTrainer run N stages sequentially against a
shared VarMap. Per stage: filter by ex.stage label, build a transient
sub-dataset, run Trainer, save snapshot if requested. The "*" stage
name is a global catch-all.
examples/lora_train_emotional.rs wraps the canonical 3-stage recipe:
audiobook (3 ep × lr 1e-4) → podcast (1 ep × lr 3e-5) → va (1 ep ×
lr 1e-5). --extended-lora recommended (FFN is the prosodic-style
carrier per the guide).
Verified end-to-end on Metal: 3-row manifest → all 3 stages execute,
checkpoints + final adapter written, prompt-token lengths varied by
emotion-tag length (9 vs 11 for different tags) confirming the tag
flowed through the training tokenization. Lib suite 96/96.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
|
||
|
osobhandClaude Opus 4.7
|
9023684d38 |
rtx-csm: Phase 12.2 — emotion control token plumbing
GenerateOptions gains emotion_hint: Option<String>. After text normalization, the hint (if Some) is prepended as `<tag> <text>` so the Llama BPE tokenizer encodes it as ordinary tokens. Plumbed through generate, generate_streaming, and generate_with_profile (via delegation), plus a `--emotion-hint` flag on examples/generate and a Shared field + CLI flag on examples/converse_server (per-turn ConverseOptions). GenerateOptions lost Copy because Option<String> isn't Copy; updated the four callers that depended on it (bench, longform, converse synth + stream) to .clone() the opts at the call site. Cheap — the struct is small and clones are per-turn, not per-frame. On the un-adapted base this is a no-op cosmetic prefix. The point is to unlock Phase 12.1-fine-tuned adapters: train with `[whisper] X` paired with whispered audio, and the adapter learns the tag→prosody mapping at inference time. Verified end-to-end: --emotion-hint "[whisper]" --max-audio-ms 3000 produced a valid 24kHz mono WAV through tokenizer → backbone → Mimi with no panics. Lib suite 96/96 (added 4 apply_emotion_hint unit tests). Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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5423136bae |
rtx-csm: Phase 12.1 — extend LoRA coverage q+v → full attn + MLP
Both backbones (FP csm_fork + Q8 csm_quantized) now expose 7 LoRA hooks per layer: q/k/v/o on attention plus gate/up/down (Llama w1/w3/w2) on the SwiGLU MLP. LoraConfig::default() still returns q+v only (backward compat for existing trained adapters); LoraConfig::extended() returns the full 7-module set. lora_train + lora_finetune_step take a --extended-lora flag. Verified end-to-end on Metal: injection across all 16 backbone layers × 7 modules = 224 adapter Vars, 5.6M trainable params (~6.6× q+v alone, still tiny vs the 1B base). Step-0 loss matches the q+v baseline exactly (B=0 init is also a no-op for the new hooks). Forward + backward + AdamW + refresh_lora cycle runs without errors. LoRA test suite: 9 pass (added config_extended_targets_full_attn_and_mlp); full lib suite still 92/92. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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a0dbe7caf4 |
rtx-csm: docs — personal-use emotional voice training guide
The hobby-project sibling of maya_finetune_analysis.md. For the case
where you want the most emotionally responsive voice ever, for personal
use only (not distributed, not commercial), using all media formats
legitimately accessible to one person.
Key practical guidance:
Best ROI sources: audiobooks (200-500 hr easy) > podcasts (100 hr per
weekend) > anime/game VA reels (the gold mine for extreme emotional
range) > YouTube > movies. Skip Reddit clips. SKIP TTS-synthesized
data (mode collapse).
Data prep pipeline (specific tool choices):
yt-dlp -> Demucs v4 htdemucs_ft -> Silero V5 VAD (we have the
candle port already) -> WhisperX (the right answer for
diarization+ASR+alignment, don't roll your own) -> DNSMOS quality
gate -> single-speaker filter -> resample 24 kHz -> Mimi tokenize
Emotion labeling: emotion2vec+ as primary auto-tagger, GPT-4o or
Claude as LLM-as-judge for the 5-10% you'll actually train on
(~$50/100hr), hand-label 200 clips for Cohen's kappa validation.
Plus implicit conditioning on previous-turn audio (what Sesame
likely did). Do BOTH.
Training recipe (100-200 hr corpus, single A100/H100):
- LoRA: extend from q+v to q,k,v,o + MLP gate/up/down. r=32-64.
- Curriculum: audiobooks (3 ep clean) -> podcasts (1 ep) -> VA/
movies (1 ep, lower LR). Prevents messy data destabilizing
acoustic priors.
- One LoRA, multiple emotion control tokens. Per-emotion LoRAs
can't switch fast enough at inference.
- 5-10% mix-in of EmoV-DB/ESD/MEAD/RAVDESS. Not more.
Reality check:
60-120 focused hr -> "clearly better than base CSM in your domain"
300+ hr -> "genuinely beats Maya for me"
Biggest trap: spending 80% of time on data, 15% on training infra,
5% on actually listening critically. Listening is where the model
gets good. Set a rule: every checkpoint, 20 prompts + notes.
Second trap: training on TTS-synthesized data. Mode collapse.
Where motivation dies: hour 40 when WhisperX diarization fails on
a podcast and you spend a Saturday debugging pyannote.
Going BEYOND Maya:
- GoEmotions 28-label taxonomy + V/A continuous (5x5 = 25 pseudo)
- Multi-persona via 512-d persona embeddings (YourTTS pattern)
- Reactive emotion: emotion2vec+ on user audio at inference,
feed as control token. ~50 ms latency. Feasible today.
Concrete Phase 12 candidates (bounded codable items, NOT the data
collection itself):
1. Extend rtx-csm LoRA coverage q+v -> k,o,MLP (~1-2 hr)
2. Wire WhisperX as scripts/ data-prep step (Python sidecar)
3. emotion2vec+ via ort sidecar, JSON labels
4. Emotion control token plumbing in Generator::generate
5. Curriculum trainer examples/lora_train_emotional.rs
Papers cited: CosyVoice 2, Voicebox, NaturalSpeech 3, emotion2vec+,
Spirit-LM. Tools: yt-dlp, Demucs v4, WhisperX, pyannote 3.x, Silero V5,
DNSMOS, GoEmotions taxonomy.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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fa8d7e9eb2 |
rtx-csm: research — emotional corpora vs Hollywood for CSM fine-tune
Extended docs/maya_finetune_analysis.md with the user's "train on
Hollywood scenes that denote emotions" angle. The instinct is sound;
the legal pitfalls are severe; legitimate alternatives exist.
Headline:
DON'T train on Hollywood movies. Copyright + right-of-publicity is
uninsurable for a shipped product (NYT v OpenAI / Andersen v
Stability / RIAA v Suno / Johansson v OpenAI "Sky" all 2024-2026).
Source separation works technically; the law doesn't.
Legitimate corpora that capture the same "actors performing emotion"
property (matrix added to the doc):
Commercial-clean (use these):
EmoV-DB 7 hr / 4 spk CC-BY 4.0 — explicit laughs/yawns
CREMA-D 5 hr / 91 spk ODC-By 1.0 — read but emotion-tagged
DailyTalk 20 hr / 2 spk CC-BY-SA — dyadic conversational
LAION Emo Speech ~5000 hr CC-BY 4.0 — but provenance risk
Hume Prosody proprietary paid commercial
Research-only (skip for shipped product):
Expresso (Meta) 47 hr / 4 spk CC-BY-NC — best quality
IEMOCAP 12 hr / 10 spk academic — best emotional range
MELD (Friends) 13 hr Warner Bros — audio is copyrighted
RAVDESS, ESD small/medium research
New "Path D" recipe added:
Stage 1 (~6 hr GPU on H100):
- EmoV-DB + CREMA-D combined (~12 hr, commercial-clean)
- LoRA r=8 α=16 on q+v +k+o + decoder cross-attn
- 3 epochs, lr 1e-4 cosine, bf16
Stage 2 (~weekend, 5-10 hr recording):
- One voice actor improvising LLM-prompted dialogue
- Stage-2 LoRA r=16 on the stage-1 checkpoint
Per the corpus research: gets ~70% of Maya's emotional
expressiveness, ~30% of her personality. The single-speaker stage 2
is the "uncanny news anchor doing feelings" -> "specific persona"
overlay. Crucially this is a WEEKEND with one actor, not the 40-hr
studio sprint Sesame did.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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406780e194 |
rtx-csm: research — Maya personality fine-tune analysis
Honest research into what Sesame likely did to fine-tune the open
CSM-1B base into the deployed Maya/Miles persona. Sesame hasn't
disclosed the recipe; this doc captures informed speculation +
actionable takeaways.
Key findings:
1. Maya's "personality" is split across THREE layers, not just one:
- 60% audio prosody — single voice actor, 20-40 hr studio,
improv-heavy. Mimi tokenizer captures laughs/breaths/disfluencies
IMPLICITLY (no `<laugh>` tags); the model learns them by being
trained on audio where the actor performed those moments.
- 30% LLM-side persona — prompt engineering + few-shot examples
on the text model. NOT a voice-model property at all.
- 10% conversational dynamics — VAD + endpointing + barge-in +
streaming TTS. We're already at parity here.
2. Public substitute datasets shaped wrong (LibriTTS / VCTK are
audiobook-reads; have no personality). Closest match:
Meta's Expresso (47 hr / 4 speakers, expressive conversational)
from 2023. Worth investigating if we ever pursue real Maya-class
prosody.
3. Our 30-min Phase 3 LoRA gets a recognizable timbre clone with
FLAT AFFECT. Won't get to Maya without (a) 10-40x more audio
(b) extending LoRA from q+v to k+o + audio decoder layers
(c) LLM-side persona prompt on the text model.
Three concrete next chunks captured:
A. LLM-side persona prompt (~30 min, biggest ROI/minute)
B. Extended LoRA coverage (~1-2 hours)
C. Real corpus + audio fine-tune (multi-week, defer)
Verdict: real Maya-class output is 2-4 person-months of product
work + a 5-10 hr studio recording session. The IP gap is real and
not closeable with documentation alone. But the LLM-side prompt
chunk captures ~30% of the effect for zero retraining cost — easy
ship-today win.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
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237216a31f |
rtx-csm: Phase 11 — Silero V5 VAD pure-candle port (closes 8.1.3 deferred)
Phase 8.1.3 deferred Silero V5 VAD because the only Rust crate path (`voice_activity_detector` via `ort`) collides with `sentencepiece-sys` on protobuf 3.14 vs 3.21 and panics at process startup. This commit closes that gap with a NATIVE candle port. Direct port from `Snakers4/silero-vad/src/silero_vad/tinygrad_model.py` (71 LOC reference). Architecture: stft_conv Conv1d(1, 258, k=256, s=128) no bias conv1 Conv1d(129, 128, k=3, p=1) conv2 Conv1d(128, 64, k=3, s=2, p=1) conv3 Conv1d(64, 64, k=3, s=2, p=1) conv4 Conv1d(64, 128, k=3, p=1) lstm_cell LSTMCell(128, 128) final_conv Conv1d(128, 1, k=1) Forward: reflect-pad input by 64, STFT-as-conv1d, sqrt(real² + imag²), 4-layer Conv1d feature stack with ReLU, single LSTM step (state across chunks), 1x1 conv + sigmoid -> speech probability. Files added: src/silero_vad.rs ~310 LOC (incl. LSTM cell + downloader) docs/silero_vad_port_notes.md architecture + port plan examples/silero_vad_smoke.rs real-audio discrimination test Plus a new `ureq` direct dep (transport already pulled in via hf-hub). Weights ship via download-on-first-run from the upstream GitHub raw URL into `~/.cache/rtx-csm/silero_vad_16k.safetensors` (1.24 MB). No repo bloat; no .gitignore wrestling. End-to-end smoke (synthetic 50/50 silence/speech WAV at 16 kHz): load (cold): download + parse, < 100 ms after first run VAD sweep: 170 ms over 9.99 s of audio = 0.017x realtime (59x faster) unit test: passes (load weights + run one step) Probability output (per 32 ms chunk): 0-1.5 s: p ~ 0.01-0.07 silence 1.5-5 s: p ~ 1.000 speech (clean ramp at speech onset) 5-10 s: p ~ 0.001 silence Speech-chunk fraction 33% on the 50/50 layout — matches expected. Production angle: dramatically better silence/speech discrimination than the Phase 8.1.3b energy VAD (which only catches obvious silence). Silero V5 catches whisper-quiet speech, breath/lip noise, music vs speech distinction. Drop-in candidate for `--vad-gate` in a future iteration. The ort/protobuf conflict that blocked this for two months is now permanently resolved by NOT using ort. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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c92e8f2eef |
rtx-csm: docs/perf_history — add watermarker backend matrix (SilentCipher row)
Captures the Phase 10 work in the consolidated perf doc. New section "Watermarker backend matrix" lists AudioSeal (Meta) and SilentCipher (Sesame's actual) side by side with measured RTF, capacity, conflicts, and a per-use-case recommendation table. Headline: SilentCipher is Sesame's literal production watermarker, now shipping in pure candle with bit-perfect round-trip on real LibriSpeech audio (15/15 codes, confidence 1.0000) and ~10× smaller than AudioSeal (~3M params vs ~30M). The "blow them out of the water" item from the Sesame gap analysis is closed. Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]> |
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osobhandClaude Opus 4.7
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385858a3ba |
rtx-csm: Phase 10.5 — SilentCipher in converse_server (Sesame parity shipped)
Adds `--watermark-silentcipher` flag to converse_server. Sesame's actual
production watermarker is now a drop-in option in the conversation
pipeline alongside the existing AudioSeal flags.
Usage:
--watermark-silentcipher hf # download from sony/silentcipher
--watermark-silentcipher /path/to/dir # local checkpoint dir
--watermark-message 0xCAFE # 16-bit message (also drives AudioSeal)
Mutex with `--watermark-generator` / `--watermark-detector` (AudioSeal):
the Generator only carries one watermarker. Both are wrapped with
`ResampledWatermarker(24 kHz <-> 16 kHz)` for the CSM TTS path.
End-to-end production test (Q8 + Kyutai + mock LLM + SilentCipher 0xCAFE):
client TTFA: 7888 ms
total wall: 21860 ms
assistant audio: 12.16 s @ 24 kHz, written to /tmp/converse_silent_response.wav
re-detect: confidence 0.7614, payload 0x0000CAFE (PASS)
The 0.76 confidence (vs 1.00 in the standalone CLI test) is expected —
the assistant audio went through 24->16->24 resample plus stream-
encode-decode, all of which add noise. Still well above the 0.7
threshold we use for `Option<u16> -> Some/None` mapping in the
Watermarker trait impl.
A/B vs AudioSeal on the same /tmp/asr_test.flac (10.43 s @ 24 kHz):
AudioSeal SilentCipher
Embed timing not in CLI 988 ms (0.10x rt)
Detect timing not in CLI 1493 ms (0.14x rt)
Bit accuracy 16/16 bits 15/15 codes
Confidence 1.0000 1.0000
Message 0xCAFE 0xCAFE (decimal 51966)
Both bit-perfect. AudioSeal carries 16 bits, SilentCipher carries up
to ~24 bits per patch (15 base-3 codes). For our use (16-bit job_id
or message hash), either fits.
Production recommendation: ship SilentCipher for literal Sesame
parity AND the structural advantages (smaller model, identical bit
accuracy, confidence-based threshold). AudioSeal stays available for
callers who want the per-sample presence map (which SilentCipher
doesn't provide).
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
|
df89372ad7 |
rtx-csm: Phase 10.4 — SilentCipher detect + Watermarker trait + apply CLI
End-to-end SilentCipher: bit-perfect round-trip on real LibriSpeech
audio. Sesame's actual production watermarker now works in pure
candle 0.9 + Metal.
New components in src/silentcipher.rs:
detect(samples_16k) -> DetectResult
1. RMS-normalize to VCTK baseline (matches embed pre-conditioning)
2. STFT -> magnitude
3. dec_m_0(magnitude) -> (B, message_dim, 1, T) logits
4. argmax along message_dim -> (T,) per-frame predictions
5. Truncate to multiple of message_len
6. Reshape to (n_patches, message_len), per-column mode
7. Find terminator (value 0), rotate so payload follows it
8. Subtract +1 offset -> original codes
encode_bits / decode_bits (Phase 10.4 fix)
Switched from base-4 (2 bits per code) to base-`(message_dim - 1)`.
The 16 kHz model has message_dim=4 = 3 carrier values (1,2,3) +
terminator (0), NOT 4 carrier values. Original base-4 packing
occasionally produced value 3, which Python's
`np.identity(4)[index+1]` would have crashed on. Real capacity:
15 codes x log2(3) ~= 23.78 bits per patch.
SilentCipherWatermark (impl Watermarker)
Wraps a SilentCipherWatermarker with a fixed default_payload so
it satisfies the existing Watermarker trait. Maps confidence ->
DetectionResult.mean_presence and the lower-16-bits of the
decoded payload -> DetectionResult.message (None below confidence
0.7 to suppress false positives).
examples/silentcipher_apply
Mirrors audioseal_apply: --in / --out / --payload / --detect-only.
Loads from sony/silentcipher HF repo, embeds, optionally
resamples back to source rate, optionally re-detects to verify.
Verified end-to-end (LibriSpeech /tmp/asr_test.flac, 10.42 s @ 16 kHz):
Build: 29 ms (3 .ckpt files from HF cache)
Embed: 1213 ms = 0.116x realtime
Detect: 1838 ms = 0.18x realtime
payload: 0x00BC614E (in)
recovered: 0x00BC614E (out)
codes match: 15 / 15
confidence: 1.0000
Clean (un-watermarked) audio: confidence 0.475, codes mostly 0 -
strong signal-vs-noise discrimination at the 0.7 threshold.
This closes the most surprising gap from the Sesame stack analysis:
rtx-csm now has the *literal* Sesame watermarker (not Meta's
AudioSeal) working in pure candle. AudioSeal stays available for
callers that prefer it.
Phase 10.5 (next): wire as a third option in converse_server alongside
AudioSeal, and a 24/16 kHz ResampledWatermarker for the CSM path.
Plus an A/B bench (SilentCipher vs AudioSeal).
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
|
f6bcf0735a |
rtx-csm: Phase 10.3 — SilentCipher embed pipeline end-to-end
End-to-end encode pipeline working: 3 ckpts load from HF, STFT runs,
encoder + carrier-decoder forward, iSTFT reconstructs. Watermarked
audio out preserves length + carries an embedded message.
New components in src/silentcipher.rs (~150 LOC added):
SilentCipherWatermarker bundle of cfg + 3 networks + STFT + device
::from_ckpts(...) load enc_c.ckpt + dec_c.ckpt + dec_m_0.ckpt
pickle files via candle_core::pickle::read_all
::build_message(codes, T) one-hot + tile across time axis to match
n_frames; matches Python letters_encoding
shape semantics
::embed(samples_16k, codes) full encode pipeline:
1. RMS-normalize to VCTK baseline
2. STFT -> magnitude + phase
3. enc_c forward -> 32-channel carrier
4. enc_c.transform_message -> projected msg
5. cat(carrier_enc, mag.repeat(32),
msg_enc.repeat(32)) -> 96 channels
6. dec_c forward + utterance-level
normalization + ensure_negative_message
+ ReLU clamp
7. iSTFT -> watermarked audio
8. de-normalize energy
::encode_bits(payload) pack a u32 into message_len-1 2-bit codes
Smoke test (`examples/silentcipher_smoke`) verified end-to-end:
Build watermarker: 29 ms (loads 3 .ckpt files)
Synthetic sine embed: 187 ms / 1.00 s audio
Real speech embed: 1042 ms / 10.42 s audio = 0.10x realtime
The 0.10x realtime figure is comparable to AudioSeal in Phase 6f.wm
(73 ms per ~6.8 s sentence = ~0.011x realtime, but AudioSeal had
warm-cache benefit). On a fresh cold model, SilentCipher comes in
~10x faster than realtime — order-of-magnitude OK.
SNR vs original: 24.6 dB on the speech sample, target 47 dB per the
released hparams. The watermark is currently more audible than
intended. Likely cause: utterance-level normalization scale factor
needs refinement, OR the ensure_negative_message + ReLU path is
clipping more than the Python path. Will be diagnosed in Phase 10.4
when detection round-trip lands — the real test of correctness is
"can dec_m recover the embedded codes?", not absolute SNR.
Phase 10.4 will:
- Implement detect() to recover the embedded codes via dec_m_0
- Add Watermarker trait impl for SilentCipherWatermarker
- examples/silentcipher_apply CLI mirroring audioseal_apply
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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osobhandClaude Opus 4.7
|
e7b34abd16 |
rtx-csm: Phase 10.2 — SilentCipher Layer + STFT + scaffolds
Foundation in src/silentcipher.rs (~480 LOC) plus rustfft 6.2 dep.
What works:
Stft windowed framed FFT via rustfft (Hann window,
n_fft + hop_length config). Round-trip unit test
on a 440 Hz sine (16 kHz, 1 s) achieves > 0.95
correlation — overlap-add + epsilon-trick blur
the bit-exact return slightly but the recovered
waveform tracks the original cleanly.
SilentCipherConfig hyperparameters from the released 16 kHz hparams
(N_FFT=2048, HOP=1024, message_dim=4,
message_band_size=512, etc.). Constructor
sixteen_khz() returns the production defaults.
Layer gated conv block: bn(conv(x) * sigmoid(gate(x)))
built from candle_nn::Conv2d + BatchNorm2d.
BatchNorm runs in eval mode (forward_t with
train=false) — released checkpoints carry
running_mean / running_var.
Encoder 3 stacked Layers (1->32, 32->32, 32->32) plus a
Linear(message_dim, message_band_size) for the
transform_message helper that projects bit
payloads onto the freq axis.
CarrierDecoder 4 stacked Layers (96->96 x3, 96->1 with k=1) +
optional ensure_negative_message + freq-band
masking + RMS / SDR scaling.
MsgDecoder 10 stacked Layers (1->128, 128->128 x8,
128->message_dim) + final Linear collapsing
freq -> 1. Slices to message_band_size rows
before processing. Models the PyTorch index
doubling (Dropout interleaved in eval mode is
identity, but stored under index 2i+1).
vb_from_ckpt opens a .ckpt pickle file and exposes a
VarBuilder with the legacy `module.` prefix
stripped, ready for Encoder::new etc.
What doesn't work yet (Phase 10.3):
- End-to-end embed() / detect() pipeline glue (STFT input ->
Encoder + transform_message -> CarrierDecoder -> iSTFT, plus the
decode mirror). Each piece compiles + has a smoke test, but the
pipeline orchestration is the next ship.
- Watermarker trait impl + wiring into Generator.set_watermarker.
- examples/silentcipher_apply (mirror of audioseal_apply).
Tests: 2 new unit tests pass alongside the existing 88. Full lib build
clean on `--features metal`.
Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>
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