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fix(tts/kokoro-ane): restore long-text chunking in synthesizeDetailed(text:) (#940) - #965

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fix/940-kokoro-chunking
Sep 25, 2026
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fix/940-kokoro-chunking

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Fixes #940.

#790 rewrote synthesizeDetailed(text:) and dropped the #717 PhonemeChunker call. Text past 510 phonemes throws phonemeSequenceTooLong again.

  • The resolved phonemes are chunked again (after normalization and G2P), and the per-chunk results are joined. synthesizeFromPhonemes stays strict.
  • The cap is now counted in Unicode scalars, via a new countsUnicodeScalars mode on PhonemeChunker. The vocab encoder counts scalars, so French nasal vowels (ɑ̃ = two scalars) could overflow a Character-counted chunk. StyleTTS2 and Inflect are unchanged.
  • KokoroAneSynthesisResult.concatenating also joins inputIds and predictedDurations, which were added after feat(tts): auto-chunk long text in KokoroAne high-level synthesize (#712) #717. Each chunk keeps its own BOS/EOS, so ids and durations stay index-aligned for word timing. normalizedText and phonemes report the full input.

Verification

  • 916-char English paragraph (am_michael): phonemeSequenceTooLong(936) → 58.7 s, ASR WER 0%.
  • 766-char French paragraph: phonemeSequenceTooLong(681) → 35.2 s.
  • New unit tests: scalar-mode chunking with nasal vowels; result concatenation order, alignment and sums.

🤖 Generated with Claude Code

…(text:) (#940)

#790 rewrote synthesizeDetailed(text:) for NeMo normalization and dropped
the #717 PhonemeChunker call, so text past 510 phonemes threw
phonemeSequenceTooLong again (#712 regression).

- Chunk the resolved phonemes again (after normalization / G2P) and join
  the per-chunk results. synthesizeFromPhonemes stays strict.
- Count the cap in Unicode scalars (new PhonemeChunker
  countsUnicodeScalars mode): the vocab encoder counts scalars, so French
  nasal vowels (U+0303) would overflow a Character-counted chunk.
  StyleTTS2 / Inflect keep Character counting.
- KokoroAneSynthesisResult.concatenating joins samples, inputIds and
  predictedDurations (fields added after #717; one BOS/EOS per chunk, so
  ids and durations stay index-aligned), sums counts and timings.
  normalizedText / phonemes report the full input.

E2E: 916-char English paragraph (am_michael): phonemeSequenceTooLong(936)
-> 58.7 s, ASR WER 0. 766-char French paragraph:
phonemeSequenceTooLong(681) -> 35.2 s.

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

Check Result
Build ✅
Model download ✅
Model load ✅
Synthesis pipeline ✅
Output WAV ✅ (168.8 KB)

Runtime: 0m22s

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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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: 0m30s

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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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 17.69x >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 13.478 22.7 Fetching diarization models
Model Compile 5.776 9.7 CoreML compilation
Audio Load 0.121 0.2 Loading audio file
Segmentation 17.785 30.0 Detecting speech regions
Embedding 29.642 50.0 Extracting speaker voices
Clustering 11.857 20.0 Grouping same speakers
Total 59.325 100 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 • 59.3s diarization time • Test runtime: 4m 11s • 09/25/2026, 04:25 PM EST

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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 8.19x Real-time factor (higher = faster)
Total Audio 470.6s Total audio duration processed
Total Time 59.3s Total processing time

Streaming Metrics

Metric Value Description
Avg Chunk Time 0.059s Average chunk processing time
Max Chunk Time 0.119s Maximum chunk processing time
EOU Detections 0 Total End-of-Utterance detections

Test runtime: 1m23s • 09/25/2026, 04:29 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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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 12.6x >1.0x ✅
Speakers 4/4 - -

Sortformer High-Latency • ES2004a • Runtime: 4m 33s • 2026-09-25T20:30:01.896Z

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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 10.95x >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 19.768 20.6 Fetching diarization models
Model Compile 8.472 8.8 CoreML compilation
Audio Load 0.066 0.1 Loading audio file
Segmentation 26.320 27.5 VAD + speech detection
Embedding 95.582 99.7 Speaker embedding extraction
Clustering (VBx) 0.109 0.1 Hungarian algorithm + VBx clustering
Total 95.839 100 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 • 122.0s processing • Test runtime: 2m 14s • 09/25/2026, 04:30 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% 4.19x ✅
test-other 1.19% 0.00% 3.11x ✅

Parakeet v2 (English-optimized)

Dataset WER Avg WER Med RTFx Status
test-clean 0.80% 0.00% 4.92x ✅
test-other 1.00% 0.00% 3.64x ✅

Streaming (v3)

Metric Value Description
WER 0.00% Word Error Rate in streaming mode
RTFx 0.46x Streaming real-time factor
Avg Chunk Time 1.897s Average time to process each chunk
Max Chunk Time 2.187s Maximum chunk processing time
First Token 2.338s 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 0.59x Streaming real-time factor
Avg Chunk Time 1.531s Average time to process each chunk
Max Chunk Time 2.051s Maximum chunk processing time
First Token 1.546s 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: 8m55s • 09/25/2026, 04:31 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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VAD Benchmark Results

Performance Comparison

Dataset Accuracy Precision Recall F1-Score RTFx Files
MUSAN 94.0% 89.3% 100.0% 94.3% 441.6x faster 50
VOiCES 94.0% 89.3% 100.0% 94.3% 553.7x 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%

@Alex-Wengg
Alex-Wengg merged commit b58efcf into main Sep 25, 2026
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@Alex-Wengg
Alex-Wengg deleted the fix/940-kokoro-chunking branch September 25, 2026 20:40
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KokoroAne synthesizeDetailed(text:) no longer chunks long phoneme input since #790 (regresses #712 / #717)

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