Files
reflector/server/gpu/modal_deployments/README.md
Mathieu Virbel 3ea7f6b7b6 feat: pipeline improvement with file processing, parakeet, silero-vad (#540)
* 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
2025-08-20 20:07:19 -06:00

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Markdown

# Reflector GPU implementation - Transcription and LLM
This repository hold an API for the GPU implementation of the Reflector API service,
and use [Modal.com](https://modal.com)
- `reflector_diarizer.py` - Diarization API
- `reflector_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](https://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/transcriptions` endpoint (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-url` endpoint (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)
- `file` or `files[]` - audio file(s) to transcribe
- `model` - 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**
```json
{
"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:
```json
{
"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)
```json
{
"audio_file_url": "https://example.com/audio.mp3",
"model": "nvidia/parakeet-tdt-0.6b-v2",
"language": "en",
"timestamp_offset": 0.0
}
```
**response**
```json
{
"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 file
- `language` - language code (e.g. `en`)
**response**
```
{
"text": "xxx",
"words": [
{"text": "xxx", "start": 0.0, "end": 1.0}
]
}
```