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fix(tts/supertonic): lower inter-chunk silence 0.3s→0.05s + add --silence flag#737

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fix(tts/supertonic): lower inter-chunk silence 0.3s→0.05s + add --silence flag#737
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fix/supertonic-paragraph-pauses

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Problem (#736)

Supertonic-3 reads short sentences naturally and fast, but full paragraphs get unintended pauses between chunks. Root cause: the 70-char chunk cap (#669) splits a paragraph into several chunks, and a fixed 0.3 s of silence is concatenated at every seam. That pad stacks on top of the model's own trailing sentence silence, inflating natural ~0.5–1.0 s sentence pauses to ~1.1–1.2 s.

Change

  • Supertonic3Constants.defaultSilenceDuration 0.3 → 0.05
  • Expose silenceDuration through the CLI --silence flag (the tts command was calling synthesize without it, so CLI users were pinned to the default)

Measured

Same 4-sentence (351-char) paragraph, ~8 chunks:

--silence total duration largest internal gaps
0.3 (old default) 25.44 s 0.9–1.2 s (robotic)
0.05 (new default) 23.69 s 0.5–1.0 s (natural sentence pauses)
0.0 23.34 s 0.5–1.0 s

Dropping the pad removes ~2.1 s of dead air across the paragraph; remaining gaps land at sentence/comma boundaries (the chunker splits there first), so there are no mid-word cuts.

Not addressed (intrinsic to the base model)

The 70-char cap can't be raised without a WER cost — the base Supertone supertonic-3 weights degrade past ~90 chars (relative-position attention trained at sentence scale: ~0% WER ≤70 chars → ~6% by ~100 chars). Supertonic-3 is a per-sentence/streaming TTS by design; for seamless long-form, PocketTTS is the better fit. This PR only fixes the avoidable silence artifact.

🤖 Generated with Claude Code

…ence flag

The 70-char chunk cap (#669) splits a paragraph into several chunks, and the
0.3s silence padded at each seam stacks on top of the model's own trailing
sentence silence, inflating natural ~0.5–1.0s pauses to ~1.1–1.2s — the
"unintended pauses" reported in #736. Measured on a 4-sentence paragraph,
0.3s→0.0s removed ~2.1s of dead air; 0.05s keeps seams from butting tokens
together while letting the model's intrinsic sentence prosody through.

- defaultSilenceDuration 0.3 → 0.05
- expose silenceDuration via CLI `--silence` (was hardcoded to the default)
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PocketTTS Smoke Test ✅

Check Result
Build
Model download
Model load
Synthesis pipeline
Output WAV ✅ (146.3 KB)

Runtime: 1m9s

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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VAD Benchmark Results

Performance Comparison

Dataset Accuracy Precision Recall F1-Score RTFx Files
MUSAN 92.0% 86.2% 100.0% 92.6% 732.6x faster 50
VOiCES 92.0% 86.2% 100.0% 92.6% 713.5x 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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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: 0m19s

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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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 6.7x >1.0x
Speakers 4/4 - -

Sortformer High-Latency • ES2004a • Runtime: 5m 28s • 2026-06-24T18:02:19.007Z

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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 15.86x >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 11.281 17.0 Fetching diarization models
Model Compile 4.835 7.3 CoreML compilation
Audio Load 0.077 0.1 Loading audio file
Segmentation 19.839 30.0 Detecting speech regions
Embedding 33.066 50.0 Extracting speaker voices
Clustering 13.226 20.0 Grouping same speakers
Total 66.176 100 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 • 66.1s diarization time • Test runtime: 3m 11s • 06/24/2026, 02:03 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 5.01x Real-time factor (higher = faster)
Total Audio 470.6s Total audio duration processed
Total Time 95.1s Total processing time

Streaming Metrics

Metric Value Description
Avg Chunk Time 0.095s Average chunk processing time
Max Chunk Time 0.190s Maximum chunk processing time
EOU Detections 0 Total End-of-Utterance detections

Test runtime: 2m31s • 06/24/2026, 02:04 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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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 13.72x >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 13.041 17.0 Fetching diarization models
Model Compile 5.589 7.3 CoreML compilation
Audio Load 0.034 0.0 Loading audio file
Segmentation 21.123 27.6 VAD + speech detection
Embedding 76.303 99.7 Speaker embedding extraction
Clustering (VBx) 0.095 0.1 Hungarian algorithm + VBx clustering
Total 76.505 100 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 • 97.5s processing • Test runtime: 1m 43s • 06/24/2026, 02:06 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% 4.02x
test-other 1.19% 0.00% 3.60x

Parakeet v2 (English-optimized)

Dataset WER Avg WER Med RTFx Status
test-clean 0.80% 0.00% 5.60x
test-other 1.00% 0.00% 3.58x

Streaming (v3)

Metric Value Description
WER 0.00% Word Error Rate in streaming mode
RTFx 0.65x Streaming real-time factor
Avg Chunk Time 1.397s Average time to process each chunk
Max Chunk Time 1.768s Maximum chunk processing time
First Token 1.718s 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 0.64x Streaming real-time factor
Avg Chunk Time 1.398s Average time to process each chunk
Max Chunk Time 1.634s Maximum chunk processing time
First Token 1.393s 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: 7m54s • 06/24/2026, 02:08 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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