mirror of
https://github.com/Monadical-SAS/reflector.git
synced 2025-12-20 20:29:06 +00:00
* 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
62 lines
2.0 KiB
Python
62 lines
2.0 KiB
Python
import pytest
|
|
|
|
|
|
@pytest.mark.asyncio
|
|
@pytest.mark.parametrize("enable_diarization", [False, True])
|
|
async def test_basic_process(
|
|
dummy_transcript,
|
|
dummy_llm,
|
|
dummy_processors,
|
|
enable_diarization,
|
|
dummy_diarization,
|
|
):
|
|
# goal is to start the server, and send rtc audio to it
|
|
# validate the events received
|
|
from pathlib import Path
|
|
|
|
from reflector.settings import settings
|
|
from reflector.tools.process import process_audio_file
|
|
|
|
# LLM_BACKEND no longer exists in settings
|
|
# settings.LLM_BACKEND = "test"
|
|
settings.TRANSCRIPT_BACKEND = "whisper"
|
|
|
|
# event callback
|
|
marks = {}
|
|
|
|
async def event_callback(event):
|
|
if event.processor not in marks:
|
|
marks[event.processor] = 0
|
|
marks[event.processor] += 1
|
|
|
|
# invoke the process and capture events
|
|
path = Path(__file__).parent / "records" / "test_mathieu_hello.wav"
|
|
|
|
if enable_diarization:
|
|
# Test with diarization - may fail if pyannote.audio is not installed
|
|
try:
|
|
await process_audio_file(
|
|
path.as_posix(), event_callback, enable_diarization=True
|
|
)
|
|
except SystemExit:
|
|
pytest.skip("pyannote.audio not installed - skipping diarization test")
|
|
else:
|
|
# Test without diarization - should always work
|
|
await process_audio_file(
|
|
path.as_posix(), event_callback, enable_diarization=False
|
|
)
|
|
|
|
print(f"Diarization: {enable_diarization}, Marks: {marks}")
|
|
|
|
# validate the events
|
|
# Each processor should be called for each audio segment processed
|
|
# The final processors (Topic, Title, Summary) should be called once at the end
|
|
assert marks["TranscriptLinerProcessor"] > 0
|
|
assert marks["TranscriptTranslatorPassthroughProcessor"] > 0
|
|
assert marks["TranscriptTopicDetectorProcessor"] == 1
|
|
assert marks["TranscriptFinalSummaryProcessor"] == 1
|
|
assert marks["TranscriptFinalTitleProcessor"] == 1
|
|
|
|
if enable_diarization:
|
|
assert marks["TestAudioDiarizationProcessor"] == 1
|