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feat(speaker): CAM++ speaker-embedding backend (CoreML)#652

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feat(speaker): CAM++ speaker-embedding backend (CoreML)#652
Alex-Wengg wants to merge 3 commits into
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feat/campplus-coreml

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Summary

Adds CAM++ (FunASR, ~7.2M) as a CoreML speaker-embedding extractor for speaker verification / diarization clustering. Model: FluidInference/campplus-coreml.

Pipeline

waveform → [Preprocessor fp32/CPU] → fbank [1,T,80]
        → [CAM++ fp16, RangeDim] → [1,192] → L2 normalize
        → cosine similarity for verification / clustering

CAM++ uses a dynamic time dim (RangeDim) so it runs on variable-length audio without padding (padding would corrupt the statistics-pooled embedding). The ANE compiler rejects RangeDim, so it runs on CPU/GPU — fine for a 7.2M model.

Changes

  • ModelNames: campPlus Repo + CampPlus registry
  • Sources/FluidAudio/Speaker/: CampPlusModels (download/load), CampPlusEmbedder (audio → 192-d embedding + cosine)
  • CLI: campplus-embed <a.wav> [b.wav] (embedding, or speaker-verification cosine)

Verification

End-to-end on M5 Pro:

  • Same speaker: cosine 0.74 → "same speaker"
  • Different speakers: cosine 0.35 → "different"

Clear separation. CoreML↔torch embedding cosine 0.9997–0.99999.

Notes

  • Overlaps FluidAudio's existing Diarizer embedding extractor — lands as an alternative.
  • A full speaker-verification EER (CN-Celeb trials) is future work; this PR validates functional speaker discrimination.

CoreML CAM++ from FluidInference/campplus-coreml: 2-stage (fbank80 preprocessor
fp32/CPU -> CAM++ RangeDim -> 192-d L2-normalized embedding). Dynamic time dim
(RangeDim, CPU/GPU; tiny model) avoids padding corrupting the stats-pooled embedding.

- ModelNames: campPlus Repo + CampPlus registry
- Speaker/: CampPlusModels, CampPlusEmbedder (audio -> embedding, cosine)
- CLI: campplus-embed (one file -> embedding; two -> speaker-verification cosine)
Verified: same-speaker cosine 0.74 vs different 0.35 on the example pairs.
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PocketTTS Smoke Test ✅

Check Result
Build
Model download
Model load
Synthesis pipeline
Output WAV ✅ (146.3 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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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 8.92x Real-time factor (higher = faster)
Total Audio 470.6s Total audio duration processed
Total Time 54.6s Total processing time

Streaming Metrics

Metric Value Description
Avg Chunk Time 0.055s Average chunk processing time
Max Chunk Time 0.109s Maximum chunk processing time
EOU Detections 0 Total End-of-Utterance detections

Test runtime: 1m20s • 05/31/2026, 10:47 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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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 0.06x ~2.5x
Overall RTFx 0.06x ~2.5x

Runtime: 4m0s

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.

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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% 4.72x
test-other 1.59% 0.00% 3.13x

Parakeet v2 (English-optimized)

Dataset WER Avg WER Med RTFx Status
test-clean 0.80% 0.00% 3.84x
test-other 1.00% 0.00% 2.07x

Streaming (v3)

Metric Value Description
WER 0.00% Word Error Rate in streaming mode
RTFx 0.53x Streaming real-time factor
Avg Chunk Time 1.660s Average time to process each chunk
Max Chunk Time 2.049s Maximum chunk processing time
First Token 1.991s 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 0.38x Streaming real-time factor
Avg Chunk Time 2.493s Average time to process each chunk
Max Chunk Time 3.799s Maximum chunk processing time
First Token 2.498s 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: 8m16s • 05/31/2026, 10:57 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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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 13.86x >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 15.831 20.9 Fetching diarization models
Model Compile 6.785 9.0 CoreML compilation
Audio Load 0.027 0.0 Loading audio file
Segmentation 21.119 27.9 VAD + speech detection
Embedding 75.525 99.7 Speaker embedding extraction
Clustering (VBx) 0.091 0.1 Hungarian algorithm + VBx clustering
Total 75.720 100 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 • 96.7s processing • Test runtime: 1m 44s • 05/31/2026, 10:44 PM EST

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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 13.5x >1.0x
Speakers 4/4 - -

Sortformer High-Latency • ES2004a • Runtime: 2m 26s • 2026-06-01T02:54:36.599Z

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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 19.91x >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 12.220 23.2 Fetching diarization models
Model Compile 5.237 9.9 CoreML compilation
Audio Load 0.121 0.2 Loading audio file
Segmentation 15.801 30.0 Detecting speech regions
Embedding 26.335 50.0 Extracting speaker voices
Clustering 10.534 20.0 Grouping same speakers
Total 52.714 100 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 • 52.7s diarization time • Test runtime: 3m 19s • 05/31/2026, 10:49 PM EST

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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% 604.4x faster 50
VOiCES 92.0% 86.2% 100.0% 92.6% 570.7x 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%

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