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Caution Review failedThe pull request is closed. WalkthroughThe changes introduce a new utility function Changes
Sequence Diagram(s)sequenceDiagram
participant Client
participant WebSocket
participant TranscribeService
Client->>WebSocket: Send audio data
WebSocket->>TranscribeService: Receive audio data
TranscribeService->>TranscribeService: _has_speech(data, sample_rate)
TranscribeService-->>WebSocket: Speech detected (true/false)
WebSocket-->>Client: Response with speech detection result
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Actionable comments posted: 0
🧹 Outside diff range and nitpick comments (2)
backend/routers/transcribe_v2.py (2)
76-88: LGTM! Consider removing or conditionally enabling the debug print statement.The
_has_speechfunction is well-implemented and correctly handles different sample rates. It efficiently processes audio data in chunks and uses the WebRTC VAD to detect speech.Consider removing the print statement on line 84 or making it conditional based on a debug flag:
- print(f"{list(sample)} {has_speech} {sample_rate}") + if DEBUG: + print(f"{list(sample)} {has_speech} {sample_rate}")This will help reduce noise in the logs during normal operation while still allowing for debugging when needed.
328-330: LGTM! Consider adding a comment explaining the opus codec condition.The refactoring improves code organization by using the new
_has_speechfunction. This change enhances readability and maintainability.Consider adding a comment explaining why the speech detection is skipped for the opus codec:
- if include_speech_profile and codec != 'opus': # don't do for opus 1.0.4 for now + # Skip speech detection for opus codec (version 1.0.4) as it's handled differently + if include_speech_profile and codec != 'opus': has_speech = _has_speech(data, sample_rate) if not has_speech: continueThis will help future developers understand the reasoning behind this condition.
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- backend/routers/transcribe_v2.py (2 hunks)
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backend/routers/transcribe_v2.py (1)
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1-424: Overall, the changes look good and align with the PR objectives.The introduction of the
_has_speechfunction and its integration into thereceive_audiofunction improve the code organization and should enhance the speech detection capabilities. The changes are well-implemented and maintain the existing logic for different codecs and speech profiles.A few minor suggestions were made to improve code clarity and logging:
- Consider adding debug flags for print statements in the
_has_speechfunction.- Add a comment explaining the opus codec condition in the
receive_audiofunction.These changes should help with fixing mic records as intended in the PR objectives.
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91-91:
asynciomay be undefined, or defined from star imports(F405)
<!-- This is an auto-generated comment: release notes by coderabbit.ai --> ## Summary by CodeRabbit - **New Features** - Introduced a utility function for detecting speech in audio data, improving the accuracy of audio processing. - **Bug Fixes** - Simplified speech detection logic to enhance performance and reduce redundancy. - **Chores** - Minor formatting adjustments for improved code clarity. <!-- end of auto-generated comment: release notes by coderabbit.ai -->
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Summary by CodeRabbit
New Features
Bug Fixes
Chores