Add a minimal, non-breaking hook to WhisperLive that allows external
projects to post-process transcription segments before they are sent
to the client.
Changes:
- ServeClientBase: add segment_post_processor attribute (default None)
- ServeClientBase.send_transcription_to_client: apply post_processor
per-segment with error handling (falls back to original segment)
- TranscriptionServer: add segment_post_processor parameter to run()
and wire it to each client on creation
This enables downstream projects to plug in custom processing
(e.g. formatting, PII redaction, diarization tagging) without
modifying WhisperLive core code.
- New word_timestamps option (default False) in client handshake
- When enabled, each segment includes 'words' array with per-word
start/end times and probability scores
- Wired through entire pipeline: client → server → backend → transcribe()
- Words include timestamp_offset for accurate absolute times
- REST API already supported word timestamps; now WebSocket does too
- Added 9 unit tests for word timestamp extraction and formatting
- New whisper_live/diarization.py: SpeakerDiarizer with online clustering
- Uses pyannote.audio speaker embeddings (optional dependency)
- Cosine similarity threshold for speaker matching (default 0.55)
- Running average embedding update for speaker stability
- Configurable max_speakers limit (default 10)
- Client options: enable_diarization, max_speakers
- Segments include 'speaker' field when diarization is active
- Graceful fallback: logs warning if pyannote not installed
- Added 12 unit tests (mock-based, no GPU required)
- New whisper_live/metrics.py with Counter, Gauge, Histogram metrics
- Track connections (opened/closed/rejected), transcription latency,
audio processed, segments emitted, REST requests, and errors
- All metric helpers are no-ops when prometheus_client not installed
- --metrics_port CLI flag to expose /metrics endpoint (0 = disabled)
- Metrics integrated into server.py, base.py at key instrumentation points
- 17 new tests in tests/test_metrics.py (178 total passing)
- MAX_BUFFER_DURATION_S (45): max audio buffer before trimming
- BUFFER_TRIM_DURATION_S (30): duration to discard on trim
- CLIP_THRESHOLD_DURATION_S (25): stale audio clip threshold
- CLIP_TAIL_DURATION_S (5): audio tail to keep after clipping
- All values can now be overridden by subclasses
- Add MAX_TRANSCRIPT_LENGTH (500) and MAX_TRANSLATION_QUEUE_SIZE (100)
class constants to ServeClientBase
- Trim transcript and text lists after each update_segments() call
- Create translation queue with maxsize to prevent unbounded growth
- Added tests for _trim_transcript()
Translate from any language to any language with alirezamsh/small100
running in a thread and reading from a queue shared with transcription thread.
Signed-off-by: makaveli10 <vineet.suryan@collabora.com>
makaveli10