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fix(tts/kokoro-ane): quiet onset on long utterances — use fp32 KokoroProsody_v2 (#947) - #963

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fix/947-kokoro-quiet-onset
Sep 25, 2026
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fix/947-kokoro-quiet-onset

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@Alex-Wengg Alex-Wengg commented Sep 25, 2026 •

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Fixes #947.

The shipped fp16 KokoroProsody miscomputes F0/N over the first 1-3 s of the utterance on the Core ML CPU and ANE paths once T_a reaches about 400 frames (~10 s of audio). The opening words then come out ~12-15 dB quiet. Not iOS-specific: it reproduces on macOS too. All variants are affected (en and ja share weights; zh too).

  • Prosody stage is now KokoroProsody_v2.mlmodelc (fp32 compute, same fp16 I/O). It's renamed rather than overwritten, so cached clients re-download.
  • New unit tests pin the stage bundle names to ModelNames.KokoroAne.requiredCoreMLModels.

Verification

  • Reporter's sentence with af_heart: onset -38.2 dB → -26.5 dB (body -24 dB); ASR WER 7.7% → 3.8%.
  • Prosody vs PyTorch across T=20..1980: 0/99 broken lengths (v1 broken at 57/99).
  • Fresh HF download: sha of all 3 bundles matches the build; ja and zh synthesize.

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🤖 Generated with Claude Code

The shipped fp16 KokoroProsody miscomputes F0/N over the first ~1-3 s of
the utterance for many T_a >= 400 frames (~10 s of audio) on the Core ML
CPU and ANE paths (GPU fp16 and fp32 are exact). The corrupted F0 is flat
~120 Hz and N is compressed, so the opening words come out ~12-15 dB quiet
(af_heart 'Self-attention' -38 dB vs -24 dB body). Not length-monotonic:
57/99 lengths in T=20..1980 broken for en/ja (identical weights), 56/99 zh.

Isolation: same phonemes into PyTorch vs Core ML; swapping torch F0/N into
the Core ML chain restores the onset; LSTM alone, palettization and each
sub-op exposed as an output are all clean, so this is a fused fp16 kernel
issue inside the upsampling AdainResBlk1d.

KokoroProsody_v2 keeps the fp16 I/O and int8 palettization, switches
compute to fp32: 0/99 broken lengths, worst F0 MAE 0.3 Hz, +~3.7 ms per
call at T=450. Renamed (not overwritten) so cached clients re-download.

Co-Authored-By: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
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Supertonic3 Smoke Test ✅

Check Result
Build ✅
Model download (incl. VectorEstimatorVariants/ int4 buckets) ✅
Model load ✅
Synthesis pipeline (--ve-variant int4) ✅
Output WAV ✅ (364.7 KB)

Runtime: 0m21s

Note: CI VMs lack a physical Neural Engine; the ANE-bucketed VectorEstimator falls back to CPU here. This validates download + variant resolution + synthesis, not ANE residency/perf.

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Parakeet EOU Benchmark Results ✅

Status: Benchmark passed
Chunk Size: 320ms
Files Tested: 100/100

Performance Metrics

Metric Value Description
WER (Avg) 7.03% Average Word Error Rate
WER (Med) 4.17% Median Word Error Rate
RTFx 6.40x Real-time factor (higher = faster)
Total Audio 470.6s Total audio duration processed
Total Time 74.3s Total processing time

Streaming Metrics

Metric Value Description
Avg Chunk Time 0.074s Average chunk processing time
Max Chunk Time 0.149s Maximum chunk processing time
EOU Detections 0 Total End-of-Utterance detections

Test runtime: 1m43s • 09/25/2026, 03:08 PM EST

RTFx = Real-Time Factor (higher is better) • Processing includes: Model inference, audio preprocessing, state management, and file I/O

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VAD Benchmark Results

Performance Comparison

Dataset Accuracy Precision Recall F1-Score RTFx Files
MUSAN 94.0% 89.3% 100.0% 94.3% 649.9x faster 50
VOiCES 94.0% 89.3% 100.0% 94.3% 621.5x faster 50

Dataset Details

  • MUSAN: Music, Speech, and Noise dataset - standard VAD evaluation
  • VOiCES: Voices Obscured in Complex Environmental Settings - tests robustness in real-world conditions

✅: Average F1-Score above 70%

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Speaker Diarization Benchmark Results

Speaker Diarization Performance

Evaluating "who spoke when" detection accuracy

Metric Value Target Status Description
DER 15.1% <30% ✅ Diarization Error Rate (lower is better)
JER 24.9% <25% ✅ Jaccard Error Rate
RTFx 17.53x >1.0x ✅ Real-Time Factor (higher is faster)

Diarization Pipeline Timing Breakdown

Time spent in each stage of speaker diarization

Stage Time (s) % Description
Model Download 11.320 18.9 Fetching diarization models
Model Compile 4.851 8.1 CoreML compilation
Audio Load 0.041 0.1 Loading audio file
Segmentation 17.945 30.0 Detecting speech regions
Embedding 29.908 50.0 Extracting speaker voices
Clustering 11.963 20.0 Grouping same speakers
Total 59.860 100 Full pipeline

Speaker Diarization Research Comparison

Research baselines typically achieve 18-30% DER on standard datasets

Method DER Notes
FluidAudio 15.1% On-device CoreML
Research baseline 18-30% Standard dataset performance

Note: RTFx shown above is from GitHub Actions runner. On Apple Silicon with ANE:

  • M2 MacBook Air (2022): Runs at 150 RTFx real-time
  • Performance scales with Apple Neural Engine capabilities

🎯 Speaker Diarization Test • AMI Corpus ES2004a • 1049.0s meeting audio • 59.8s diarization time • Test runtime: 3m 23s • 09/25/2026, 03:09 PM EST

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Offline VBx Pipeline Results

Speaker Diarization Performance (VBx Batch Mode)

Optimal clustering with Hungarian algorithm for maximum accuracy

Metric Value Target Status Description
DER 10.4% <20% ✅ Diarization Error Rate (lower is better)
RTFx 9.11x >1.0x ✅ Real-Time Factor (higher is faster)

Offline VBx Pipeline Timing Breakdown

Time spent in each stage of batch diarization

Stage Time (s) % Description
Model Download 19.915 17.3 Fetching diarization models
Model Compile 8.535 7.4 CoreML compilation
Audio Load 0.078 0.1 Loading audio file
Segmentation 30.537 26.5 VAD + speech detection
Embedding 114.826 99.7 Speaker embedding extraction
Clustering (VBx) 0.132 0.1 Hungarian algorithm + VBx clustering
Total 115.128 100 Full VBx pipeline

Speaker Diarization Research Comparison

Offline VBx achieves competitive accuracy with batch processing

Method DER Mode Description
FluidAudio (Offline) 10.4% VBx Batch On-device CoreML with optimal clustering
FluidAudio (Streaming) 17.7% Chunk-based First-occurrence speaker mapping
Research baseline 18-30% Various Standard dataset performance

Pipeline Details:

  • Mode: Offline VBx with Hungarian algorithm for optimal speaker-to-cluster assignment
  • Segmentation: VAD-based voice activity detection
  • Embeddings: WeSpeaker-compatible speaker embeddings
  • Clustering: PowerSet with VBx refinement
  • Accuracy: Higher than streaming due to optimal post-hoc mapping

🎯 Offline VBx Test • AMI Corpus ES2004a • 1049.0s meeting audio • 145.5s processing • Test runtime: 2m 32s • 09/25/2026, 03:15 PM EST

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ASR Benchmark Results ✅

Status: All benchmarks passed

Parakeet v3 (multilingual)

Dataset WER Avg WER Med RTFx Status
test-clean 0.57% 0.00% 2.74x ✅
test-other 1.19% 0.00% 1.87x ✅

Parakeet v2 (English-optimized)

Dataset WER Avg WER Med RTFx Status
test-clean 0.80% 0.00% 2.70x ✅
test-other 1.00% 0.00% 2.57x ✅

Streaming (v3)

Metric Value Description
WER 0.00% Word Error Rate in streaming mode
RTFx 0.32x Streaming real-time factor
Avg Chunk Time 2.721s Average time to process each chunk
Max Chunk Time 3.658s Maximum chunk processing time
First Token 3.481s Latency to first transcription token
Total Chunks 31 Number of chunks processed

Streaming (v2)

Metric Value Description
WER 0.00% Word Error Rate in streaming mode
RTFx 0.48x Streaming real-time factor
Avg Chunk Time 1.916s Average time to process each chunk
Max Chunk Time 2.433s Maximum chunk processing time
First Token 1.984s Latency to first transcription token
Total Chunks 31 Number of chunks processed

Streaming tests use 5 files with 0.5s chunks to simulate real-time audio streaming

25 files per dataset • Test runtime: 12m12s • 09/25/2026, 03:21 PM EST

RTFx = Real-Time Factor (higher is better) • Calculated as: Total audio duration ÷ Total processing time
Processing time includes: Model inference on Apple Neural Engine, audio preprocessing, state resets between files, token-to-text conversion, and file I/O
Example: RTFx of 2.0x means 10 seconds of audio processed in 5 seconds (2x faster than real-time)

Expected RTFx Performance on Physical M1 Hardware:

• M1 Mac: ~28x (clean), ~25x (other)
• CI shows ~0.5-3x due to virtualization limitations

Testing methodology follows HuggingFace Open ASR Leaderboard

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PocketTTS Smoke Test ✅

Check Result
Build ✅
Model download ✅
Model load ✅
Synthesis pipeline ✅
Output WAV ✅ (183.8 KB)

Runtime: 0m21s

Note: PocketTTS uses CoreML MLState (macOS 15) KV cache + Mimi streaming state. CI VM lacks physical GPU — audio quality and performance may differ from Apple Silicon.

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Alex-Wengg merged commit 8c1bd91 into main Sep 25, 2026
18 of 19 checks passed
@Alex-Wengg
Alex-Wengg deleted the fix/947-kokoro-quiet-onset branch September 25, 2026 19:36
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Sortformer High-Latency Benchmark Results

ES2004a Performance (30.4s latency config)

Metric Value Target Status
DER 30.3% <35% ✅
Miss Rate 28.2% - -
False Alarm 0.9% - -
Speaker Error 1.2% - -
RTFx 12.7x >1.0x ✅
Speakers 4/4 - -

Sortformer High-Latency • ES2004a • Runtime: 4m 8s • 2026-09-25T19:39:15.607Z

Alex-Wengg added a commit that referenced this pull request Sep 25, 2026
Adds a short convention under **HUGGINGFACE UPLOADS** in `CLAUDE.md` so
every model fix can be traced from the issue to the code to the HF
commit, and back.

- Fixed models ship as `_v2` next to the original.
- HF commit messages name the issue and link the PRs.
- mobius and FluidAudio PRs link the issue, each other, and the HF
commit.
- A `## Changelog` row goes on the HF model card.

First applied to #947 / #963 / FluidInference/mobius#108. The
[kokoro-82m-coreml
card](https://huggingface.co/FluidInference/kokoro-82m-coreml) now has
the table, backfilled for the #836, #852, #914 and #926 uploads.

🤖 Generated with [Claude Code](https://claude.com/claude-code)

Co-authored-by: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
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kokoro iOS 27 the sound is very quiet at the beginning.

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