* feat: improve pipeline threading, and transcriber (parakeet and silero vad) * refactor: remove whisperx, implement parakeet * refactor: make audio_chunker more smart and wait for speech, instead of fixed frame * refactor: make audio merge to always downscale the audio to 16k for transcription * refactor: make the audio transcript modal accepting batches * refactor: improve type safety and remove prometheus metrics - Add DiarizationSegment TypedDict for proper diarization typing - Replace List/Optional with modern Python list/| None syntax - Remove all Prometheus metrics from TranscriptDiarizationAssemblerProcessor - Add comprehensive file processing pipeline with parallel execution - Update processor imports and type annotations throughout - Implement optimized file pipeline as default in process.py tool * refactor: convert FileDiarizationProcessor I/O types to BaseModel Update FileDiarizationInput and FileDiarizationOutput to inherit from BaseModel instead of plain classes, following the standard pattern used by other processors in the codebase. * test: add tests for file transcript and diarization with pytest-recording * build: add pytest-recording * feat: add local pyannote for testing * fix: replace PyAV AudioResampler with torchaudio for reliable audio processing - Replace problematic PyAV AudioResampler that was causing ValueError: [Errno 22] Invalid argument - Use torchaudio.functional.resample for robust sample rate conversion - Optimize processing: skip conversion for already 16kHz mono audio - Add direct WAV writing with Python wave module for better performance - Consolidate duplicate downsample checks for cleaner code - Maintain list[av.AudioFrame] input interface - Required for Silero VAD which needs 16kHz mono audio * fix: replace PyAV AudioResampler with torchaudio solution - Resolves ValueError: [Errno 22] Invalid argument in AudioMergeProcessor - Replaces problematic PyAV AudioResampler with torchaudio.functional.resample - Optimizes processing to skip unnecessary conversions when audio is already 16kHz mono - Uses direct WAV writing with Python's wave module for better performance - Fixes test_basic_process to disable diarization (pyannote dependency not installed) - Updates test expectations to match actual processor behavior - Removes unused pydub dependency from pyproject.toml - Adds comprehensive TEST_ANALYSIS.md documenting test suite status * feat: add parameterized test for both diarization modes - Adds @pytest.mark.parametrize to test_basic_process with enable_diarization=[False, True] - Test with diarization=False always passes (tests core AudioMergeProcessor functionality) - Test with diarization=True gracefully skips when pyannote.audio is not installed - Provides comprehensive test coverage for both pipeline configurations * fix: resolve pipeline property naming conflict in AudioDiarizationPyannoteProcessor - Renames 'pipeline' property to 'diarization_pipeline' to avoid conflict with base Processor.pipeline attribute - Fixes AttributeError: 'property 'pipeline' object has no setter' when set_pipeline() is called - Updates property usage in _diarize method to use new name - Now correctly supports pipeline initialization for diarization processing * fix: add local for pyannote * test: add diarization test * fix: resample on audio merge now working * fix: correctly restore timestamp * fix: display exception in a threaded processor if that happen * Update pyproject.toml * ci: remove option * ci: update astral-sh/setup-uv * test: add monadical url for pytest-recording * refactor: remove previous version * build: move faster whisper to local dep * test: fix missing import * refactor: improve main_file_pipeline organization and error handling - Move all imports to the top of the file - Create unified EmptyPipeline class to replace duplicate mock pipeline code - Remove timeout and fallback logic - let processors handle their own retries - Fix error handling to raise any exception from parallel tasks - Add proper type hints and validation for captured results * fix: wrong function * fix: remove task_done * feat: add configurable file processing timeouts for modal processors - Add TRANSCRIPT_FILE_TIMEOUT setting (default: 600s) for file transcription - Add DIARIZATION_FILE_TIMEOUT setting (default: 600s) for file diarization - Replace hardcoded timeout=600 with configurable settings in modal processors - Allows customization of timeout values via environment variables * fix: use logger * fix: worker process meetings now use file pipeline * fix: topic not gathered * refactor: remove prepare(), pipeline now work * refactor: implement many review from Igor * test: add test for test_pipeline_main_file * refactor: remove doc * doc: add doc * ci: update build to use native arm64 builder * fix: merge fixes * refactor: changes from Igor review + add test (not by default) to test gpu modal part * ci: update to our own runner linux-amd64 * ci: try using suggested mode=min * fix: update diarizer for latest modal, and use volume * fix: modal file extension detection * fix: put the diarizer as A100
Reflector GPU implementation - Transcription and LLM
This repository hold an API for the GPU implementation of the Reflector API service, and use Modal.com
reflector_diarizer.py- Diarization APIreflector_transcriber.py- Transcription API (Whisper)reflector_transcriber_parakeet.py- Transcription API (NVIDIA Parakeet)reflector_translator.py- Translation API
Modal.com deployment
Create a modal secret, and name it reflector-gpu.
It should contain an REFLECTOR_APIKEY environment variable with a value.
The deployment is done using Modal.com service.
$ modal deploy reflector_transcriber.py
...
└── 🔨 Created web => https://xxxx--reflector-transcriber-web.modal.run
$ modal deploy reflector_transcriber_parakeet.py
...
└── 🔨 Created web => https://xxxx--reflector-transcriber-parakeet-web.modal.run
$ modal deploy reflector_llm.py
...
└── 🔨 Created web => https://xxxx--reflector-llm-web.modal.run
Then in your reflector api configuration .env, you can set these keys:
TRANSCRIPT_BACKEND=modal
TRANSCRIPT_URL=https://xxxx--reflector-transcriber-web.modal.run
TRANSCRIPT_MODAL_API_KEY=REFLECTOR_APIKEY
DIARIZATION_BACKEND=modal
DIARIZATION_URL=https://xxxx--reflector-diarizer-web.modal.run
DIARIZATION_MODAL_API_KEY=REFLECTOR_APIKEY
TRANSLATION_BACKEND=modal
TRANSLATION_URL=https://xxxx--reflector-translator-web.modal.run
TRANSLATION_MODAL_API_KEY=REFLECTOR_APIKEY
API
Authentication must be passed with the Authorization header, using the bearer scheme.
Authorization: bearer <REFLECTOR_APIKEY>
LLM
POST /llm
request
{
"prompt": "xxx"
}
response
{
"text": "xxx completed"
}
Transcription
Parakeet Transcriber (reflector_transcriber_parakeet.py)
NVIDIA Parakeet is a state-of-the-art ASR model optimized for real-time transcription with superior word-level timestamps.
GPU Configuration:
-
A10G GPU - Used for
/v1/audio/transcriptionsendpoint (small files, live transcription)- Higher concurrency (max_inputs=10)
- Optimized for multiple small audio files
- Supports batch processing for efficiency
-
L40S GPU - Used for
/v1/audio/transcriptions-from-urlendpoint (large files)- Lower concurrency but more powerful processing
- Optimized for single large audio files
- VAD-based chunking for long-form audio
/v1/audio/transcriptions - Small file transcription
request (multipart/form-data)
fileorfiles[]- audio file(s) to transcribemodel- model name (default:nvidia/parakeet-tdt-0.6b-v2)language- language code (default:en)batch- whether to use batch processing for multiple files (default:true)
response
{
"text": "transcribed text",
"words": [
{"word": "hello", "start": 0.0, "end": 0.5},
{"word": "world", "start": 0.5, "end": 1.0}
],
"filename": "audio.mp3"
}
For multiple files with batch=true:
{
"results": [
{
"filename": "audio1.mp3",
"text": "transcribed text",
"words": [...]
},
{
"filename": "audio2.mp3",
"text": "transcribed text",
"words": [...]
}
]
}
/v1/audio/transcriptions-from-url - Large file transcription
request (application/json)
{
"audio_file_url": "https://example.com/audio.mp3",
"model": "nvidia/parakeet-tdt-0.6b-v2",
"language": "en",
"timestamp_offset": 0.0
}
response
{
"text": "transcribed text from large file",
"words": [
{"word": "hello", "start": 0.0, "end": 0.5},
{"word": "world", "start": 0.5, "end": 1.0}
]
}
Supported file types: mp3, mp4, mpeg, mpga, m4a, wav, webm
Whisper Transcriber (reflector_transcriber.py)
POST /transcribe
request (multipart/form-data)
file- audio filelanguage- language code (e.g.en)
response
{
"text": "xxx",
"words": [
{"text": "xxx", "start": 0.0, "end": 1.0}
]
}