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Closes #120

Adds token-level timestamps to StreamingTranscriptionUpdate for token-level precision in diarization.

Changes

  • Add tokenTimings property to StreamingTranscriptionUpdate struct
  • Update StreamingAsrManager to pass tokenTimings from ASRResult
  • Change finish() method to return ASRResult instead of String
  • Update CLI commands to handle new finish() return type
  • Add test for StreamingTranscriptionUpdate with tokenTimings

This enables token-level precision for diarization by exposing individual token timestamps from the ASR model.

Generated with Claude Code

- Add tokenTimings property to StreamingTranscriptionUpdate struct
- Update StreamingAsrManager to pass tokenTimings from ASRResult
- Change finish() method to return ASRResult instead of String
- Update CLI commands to handle new finish() return type
- Add test for StreamingTranscriptionUpdate with tokenTimings

This enables token-level precision for diarization by exposing
individual token timestamps from the ASR model.

🤖 Generated with [Claude Code](https://claude.ai/code)

Co-authored-by: Brandon Weng <[email protected]>
@BrandonWeng BrandonWeng added enhancement New feature or request speech-to-text issues related to transcription/asr labels Oct 2, 2025
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claude bot commented Oct 2, 2025

Claude finished @BrandonWeng's task —— View job


PR Review Complete ✅

Reviewed PR #132: Add token-level timestamps to StreamingTranscriptionUpdate

Todo List:

  • Start pending review
  • Get PR diff information
  • Analyze code changes for issues
  • Add inline review comments
  • Submit comprehensive review

Review Summary

This PR successfully adds token-level timestamps to enable diarization precision. The implementation is solid with good backward compatibility and proper error handling.

Key Findings:

  • ✅ Maintains backward compatibility with optional tokenTimings
  • ✅ Proper API design - finish() now returns full ASRResult
  • ✅ Good test coverage for new functionality
  • 📝 Minor suggestion: Use more appropriate fallback confidence value (currently 1.0)
  • 📝 Consider extracting fallback ASRResult creation to helper method

Overall Assessment: Clean implementation that adds the requested functionality without breaking existing code. Recommend approval with minor suggestions addressed.

Full review submitted as non-blocking comments.

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Code Review Summary

This PR successfully adds token-level timestamps to StreamingTranscriptionUpdate for diarization precision. The implementation is well-structured and maintains backward compatibility. Here are my detailed findings:

✅ Strengths

  1. Backward Compatibility: Optional tokenTimings parameter ensures existing code continues to work
  2. Consistent API Design: The finish() method now returns the full ASRResult instead of just text, providing access to all timing information
  3. Proper Error Handling: Fallback logic when tokens aren't available
  4. Good Test Coverage: New test specifically validates token timing functionality

📝 Code Quality Issues

Sources/FluidAudio/ASR/Streaming/StreamingAsrManager.swift:179

  • Nit: Using confidence: 1.0 in the fallback case implies perfect confidence when we have no confidence information. Consider using 0.0 or a dedicated constant like UNKNOWN_CONFIDENCE.

Sources/FluidAudio/ASR/Streaming/StreamingAsrManager.swift:175-184

  • Nit: The fallback ASRResult creation could be extracted to a helper method for better readability and reusability.

🔧 API Design Considerations

Sources/FluidAudio/ASR/Streaming/StreamingAsrManager.swift:149

  • Good: The breaking change to finish() return type is justified since it provides more complete information for diarization use cases.

Sources/FluidAudio/ASR/Streaming/StreamingAsrManager.swift:586

  • Good: Making tokenTimings optional with a default nil value maintains compatibility while enabling the new functionality.

🧪 Testing

Tests/FluidAudioTests/StreamingAsrManagerTests.swift:161-183

  • Good: Comprehensive test for the new token timing functionality
  • Suggestion: Consider adding a test that verifies the finish() method returns token timings when available

📋 CLI Updates

Sources/FluidAudioCLI/Commands/ASR/TranscribeCommand.swift:333-334
Sources/FluidAudioCLI/Commands/MultiStreamCommand.swift:165-168

  • Good: Proper adaptation to the new finish() return type while maintaining existing behavior

Overall Assessment

This is a clean implementation that adds the requested functionality without breaking existing code. The changes are minimal, focused, and well-tested. The API design decision to return the full ASRResult from finish() is good as it provides access to all available information including token timings.

Recommendation: Approve with minor suggestions addressed.

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github-actions bot commented Oct 2, 2025

VAD Benchmark Results

Performance Comparison

Dataset Accuracy Precision Recall F1-Score RTFx Files
MUSAN 92.0% 86.2% 100.0% 92.6% 391.5x faster 50
VOiCES 92.0% 86.2% 100.0% 92.6% 380.2x 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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github-actions bot commented Oct 2, 2025

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 17.19x >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 9.637 15.8 Fetching diarization models
Model Compile 4.130 6.8 CoreML compilation
Audio Load 0.068 0.1 Loading audio file
Segmentation 18.296 30.0 Detecting speech regions
Embedding 30.493 49.9 Extracting speaker voices
Clustering 12.197 20.0 Grouping same speakers
Total 61.048 100 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 • 61.0s diarization time • Test runtime: 1m 43s • 10/02/2025, 09:24 PM EST

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Comment on lines 172 to 173
audioSamples: [], // Not needed for final text conversion
processingTime: 0

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P1 Badge Final result never includes token timings

The new finish() API advertises “The complete ASR result with token timings”, but the implementation still discards all timing information. The manager only accumulates token IDs and, when building the final ASRResult, calls processTranscriptionResult with empty timestamps and confidences, so ASRResult.tokenTimings is always empty or nil even when the streaming updates contained per-token timings. Any consumer relying on finish() to provide token-level timestamps for diarization will never receive them. Consider retaining the token timestamps/confidences as they stream and pass them into processTranscriptionResult (or aggregate the interim token timings) before returning the final result.

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github-actions bot commented Oct 2, 2025

ASR Benchmark Results ✅

Status: All benchmarks passed

Parakeet v3 (multilingual)

Dataset WER Avg WER Med RTFx Status
test-clean 0.57% 0.00% 2.70x
test-other 1.59% 0.00% 1.41x

Parakeet v2 (English-optimized)

Dataset WER Avg WER Med RTFx Status
test-clean 0.40% 0.00% 1.90x
test-other 1.22% 0.00% 1.46x

Streaming (v3)

Metric Value Description
WER 0.00% Word Error Rate in streaming mode
RTFx 0.23x Streaming real-time factor
Avg Chunk Time 3.742s Average time to process each chunk
Max Chunk Time 5.235s Maximum chunk processing time
First Token 4.650s 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 0.21x Streaming real-time factor
Avg Chunk Time 4.356s Average time to process each chunk
Max Chunk Time 5.917s Maximum chunk processing time
First Token 4.035s 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: 11m18s • 10/02/2025, 09:35 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

@BrandonWeng
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@codex take another pass

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Comment on lines 315 to 378
)

// Update state
accumulatedTokens.append(contentsOf: tokens)
accumulateTokenMetadata(tokens: tokens, timestamps: timestamps, confidences: confidences)

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P1 Badge Offset streaming timestamps before collecting token metadata

The final tokenTimings returned by finish() will not represent real positions in the audio stream. Each call to transcribeStreamingChunk yields frame indices that are relative to the current sliding window (the call is made with contextFrameAdjustment: 0 and no global offset), yet those indices are appended verbatim via accumulateTokenMetadata. Because every window includes left context, tokens emitted after the first chunk retain timestamps near the beginning of their window rather than the absolute time in the session. When the accumulated timestamps are converted to TokenTimings, later tokens appear out of order and their start times repeat, which defeats the diarization precision this change is meant to provide. The timestamps should be shifted by the window’s absolute position (e.g. using nextWindowCenterStart or lastProcessedFrame) before being stored.

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@BrandonWeng BrandonWeng requested a review from Alex-Wengg October 3, 2025 20:32
@FluidInference FluidInference deleted a comment from claude bot Oct 3, 2025
let duration = TimeInterval(sampleCount) / TimeInterval(config.asrConfig.sampleRate)
return ASRResult(
text: finalTranscriptText(),
confidence: 1.0,
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why is this hardcoded to 1.0

maxTimestampFrame > 0
? (maxTimestampFrame + 1) * ASRConstants.samplesPerEncoderFrame : 0
let sampleCount = max(totalSamplesProcessed, derivedSampleCount)
let processingTime = sampleCount > 0 ? max(elapsedTime, minimumProcessingTime) : elapsedTime
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whats the difference between these two variables exactly

logger.warning(
"Final token \(label) count (\(values.count)) does not match token count (\(expectedCount)); \(omissionDetail)"
)
return []
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why are we returning [] for a mismatch count ?


#if DEBUG
extension StreamingAsrManager {
internal func setAsrManagerForTesting(_ manager: AsrManager?) {
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should this be async ?

@Alex-Wengg
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comments done

@BrandonWeng
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Not happy with the PR, going to rewrite

@BrandonWeng BrandonWeng closed this Oct 8, 2025
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StreamingASRManager Token level timestamps for diarization
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