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

server/gpu/modal_deployments/README.md

@@ -4,7 +4,8 @@ This repository hold an API for the GPU implementation of the Reflector API serv
 and use [Modal.com](https://modal.com)
 
 - `reflector_diarizer.py` - Diarization API
-- `reflector_transcriber.py` - Transcription API
+- `reflector_transcriber.py` - Transcription API (Whisper)
+- `reflector_transcriber_parakeet.py` - Transcription API (NVIDIA Parakeet)
 - `reflector_translator.py` - Translation API
 
 ## Modal.com deployment
@@ -19,6 +20,10 @@ $ 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
@@ -68,6 +73,86 @@ Authorization: bearer <REFLECTOR_APIKEY>
 
 ### 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)