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docs(ane-profiler): explain why convolutions map to the ANE and attention doesn't - #887

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docs(ane-profiler): explain why convolutions map to the ANE and attention doesn't#887
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The ANE Profiler doc records the measured ANE/GPU/CPU splits and lists the gotchas that knock work off the ANE, but never explains why the splits fall the way they do.

Adds a closing section, "Why convolutions love the ANE and attention doesn't", tying the symptoms to the architectural cause:

  • The ANE is a weight-stationary convolution engine: fixed weights in the MAC array, channels-major 4D layout, static tiles.
  • Attention is the worst case for that design: activation×activation matmuls (nothing to keep stationary), Kᵀ transposes as real memory-bound ops, softmax stalling the pipeline, S×S matrices that don't tile, and dynamic KV-cache state.
  • Each point is anchored to an existing repo example (Kokoro vocoder 99% ANE, PocketTTS flowlm rank-5 scatter rejection, Parakeet v3 encoder deliberately shipping on GPU at +8% RTFx).
  • Notes the mirror image on NVIDIA tensor cores (GEMM engines; small-channel speech convs utilize them poorly), and the practical rule: for transformer graphs the ANE is a power/residency play earned via graph surgery; for conv graphs it's free performance.

Docs-only change; no code.

🤖 Generated with Claude Code

…esn't

The profiler doc records the ANE/GPU/CPU splits and lists the gotchas, but
never explains the architectural cause. Adds a closing section tying the
symptoms together: the ANE is a weight-stationary convolution engine, and
attention is activation-x-activation matmuls + transposes + softmax + dynamic
KV state — each mapped to the specific gotcha and repo example it produces
(Kokoro vocoder 99% ANE, PocketTTS flowlm scatter rejection, Parakeet v3
encoder shipping on GPU). Also notes the mirror-image on NVIDIA tensor cores
and the practical rule: ANE for transformer graphs is a power/residency play
earned by graph surgery; for conv graphs it's free performance.
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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 10.84x Real-time factor (higher = faster)
Total Audio 470.6s Total audio duration processed
Total Time 45.7s Total processing time

Streaming Metrics

Metric Value Description
Avg Chunk Time 0.046s Average chunk processing time
Max Chunk Time 0.091s Maximum chunk processing time
EOU Detections 0 Total End-of-Utterance detections

Test runtime: 1m32s • 09/03/2026, 12:01 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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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 25.15x >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 14.792 35.4 Fetching diarization models
Model Compile 6.339 15.2 CoreML compilation
Audio Load 0.138 0.3 Loading audio file
Segmentation 12.511 30.0 Detecting speech regions
Embedding 20.851 50.0 Extracting speaker voices
Clustering 8.340 20.0 Grouping same speakers
Total 41.730 100 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 • 41.7s diarization time • Test runtime: 3m 23s • 09/03/2026, 12:02 PM EST

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PocketTTS Smoke Test ✅

Check Result
Build
Model download
Model load
Synthesis pipeline
Output WAV ✅ (157.5 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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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 18.7x >1.0x
Speakers 4/4 - -

Sortformer High-Latency • ES2004a • Runtime: 6m 19s • 2026-09-03T16:05:35.451Z

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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% 727.2x faster 50
VOiCES 94.0% 89.3% 100.0% 94.3% 748.0x 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: 1m14s

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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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 10.73x >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 25.212 25.8 Fetching diarization models
Model Compile 10.805 11.1 CoreML compilation
Audio Load 0.069 0.1 Loading audio file
Segmentation 30.078 30.8 VAD + speech detection
Embedding 97.520 99.7 Speaker embedding extraction
Clustering (VBx) 0.126 0.1 Hungarian algorithm + VBx clustering
Total 97.779 100 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 • 127.7s processing • Test runtime: 2m 20s • 09/03/2026, 12:14 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% 4.21x
test-other 1.19% 0.00% 2.68x

Parakeet v2 (English-optimized)

Dataset WER Avg WER Med RTFx Status
test-clean 0.80% 0.00% 5.88x
test-other 1.62% 0.00% 3.60x

Streaming (v3)

Metric Value Description
WER 0.00% Word Error Rate in streaming mode
RTFx 0.43x Streaming real-time factor
Avg Chunk Time 2.088s Average time to process each chunk
Max Chunk Time 3.033s Maximum chunk processing time
First Token 2.543s 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 0.60x Streaming real-time factor
Avg Chunk Time 1.532s Average time to process each chunk
Max Chunk Time 1.700s Maximum chunk processing time
First Token 1.485s 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: 15m26s • 09/03/2026, 12:23 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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