docs: add iPhone 17 Pro ANE TTS demos to Video Demos

[Alex-Wengg]

Summary

Add three @sach1n demo videos to the Video Demos table in the README, showcasing CoreML TTS models running on the Apple Neural Engine on iPhone 17 Pro.

New entries:

• MagpieTTS Multilingual on iPhone 17 Pro — NVIDIA's MagpieTTS Multilingual converted to CoreML and running on iPhone 17 Pro
• Kyutai PocketTTS on ANE — Kyutai labs PocketTTS in CoreML running on the ANE, fast and background-capable on iOS
• Supertonic-3 on iPhone 17 Pro ANE — Supertonic-3 running on iPhone 17 Pro via ANE/CoreML — 2 minutes of audio generated in 3 seconds, low RAM, background-capable

Test plan

• README renders correctly (markdown table syntax preserved)
• All three URLs are valid and point to the referenced posts

[github-actions]

PocketTTS Smoke Test ✅

Check	Result
Build	✅
Model download	✅
Model load	✅
Synthesis pipeline	✅
Output WAV	✅ (157.5 KB)

_{Runtime: 0m19s}

_{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.}

[github-actions]

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	13.77x	>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	10.016	13.1	Fetching diarization models
Model Compile	4.292	5.6	CoreML compilation
Audio Load	0.035	0.0	Loading audio file
Segmentation	21.380	28.1	VAD + speech detection
Embedding	75.973	99.7	Speaker embedding extraction
Clustering (VBx)	0.097	0.1	Hungarian algorithm + VBx clustering
Total	76.183	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 38s • 05/19/2026, 01:57 AM EST}

[github-actions]

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	8.53x	Real-time factor (higher = faster)
Total Audio	470.6s	Total audio duration processed
Total Time	58.5s	Total processing time

Streaming Metrics

Metric	Value	Description
Avg Chunk Time	0.058s	Average chunk processing time
Max Chunk Time	0.117s	Maximum chunk processing time
EOU Detections	0	Total End-of-Utterance detections

_{Test runtime: 1m18s • 05/19/2026, 01:59 AM EST}

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

[github-actions]

Qwen3-ASR int8 Smoke Test ✅

Check	Result
Build	✅
Model download	✅
Model load	✅
Transcription pipeline	✅
Decoder size	571 MB (vs 1.1 GB f32)

Performance Metrics

Metric	CI Value	Expected on Apple Silicon
Median RTFx	0.04x	~2.5x
Overall RTFx	0.04x	~2.5x

_{Runtime: 4m35s}

_{Note: CI VM lacks physical GPU — CoreML MLState (macOS 15) KV cache produces degraded results on virtualized runners. On Apple Silicon: ~1.3% WER / 2.5x RTFx.}

[github-actions]

ASR Benchmark Results ✅

Status: All benchmarks passed

Parakeet v3 (multilingual)

Dataset	WER Avg	WER Med	RTFx	Status
test-clean	0.57%	0.00%	3.12x	✅
test-other	1.40%	0.00%	2.95x	✅

Parakeet v2 (English-optimized)

Dataset	WER Avg	WER Med	RTFx	Status
test-clean	0.80%	0.00%	3.04x	✅
test-other	1.16%	0.00%	3.08x	✅

Streaming (v3)

Metric	Value	Description
WER	0.00%	Word Error Rate in streaming mode
RTFx	0.37x	Streaming real-time factor
Avg Chunk Time	2.342s	Average time to process each chunk
Max Chunk Time	3.306s	Maximum chunk processing time
First Token	3.069s	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	0.54x	Streaming real-time factor
Avg Chunk Time	1.783s	Average time to process each chunk
Max Chunk Time	2.454s	Maximum chunk processing time
First Token	1.923s	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: 8m53s • 05/19/2026, 02:07 AM 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}

[github-actions]

VAD Benchmark Results

Performance Comparison

Dataset	Accuracy	Precision	Recall	F1-Score	RTFx	Files
MUSAN	92.0%	86.2%	100.0%	92.6%	375.5x faster	50
VOiCES	92.0%	86.2%	100.0%	92.6%	408.1x 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%

[github-actions]

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	7.7x	>1.0x	✅
Speakers	4/4	-	-

_{Sortformer High-Latency • ES2004a • Runtime: 4m 31s • 2026-05-19T06:12:39.175Z}

[github-actions]

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	18.11x	>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	9.128	15.7	Fetching diarization models
Model Compile	3.912	6.7	CoreML compilation
Audio Load	0.059	0.1	Loading audio file
Segmentation	17.370	30.0	Detecting speech regions
Embedding	28.951	50.0	Extracting speaker voices
Clustering	11.580	20.0	Grouping same speakers
Total	57.959	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 • 57.9s diarization time • Test runtime: 3m 12s • 05/19/2026, 02:16 AM EST}