mirror of
https://github.com/deepset-ai/haystack.git
synced 2025-07-25 18:00:28 +00:00

* fix: fix audio tests + unbound some dependencies * fix: update for Python 3.8 * refactor: change numpy assertion * feat: add voice recog. support on audio tests * fix: fix var assignement * chore: dummy commit * fix: fix sndfile error * refactor: change skip reason * refactor: hardcode variable * refactor: unpin numpy * fix: pin numpy only for audio
181 lines
7.4 KiB
Python
181 lines
7.4 KiB
Python
import os
|
|
|
|
import pytest
|
|
import numpy as np
|
|
|
|
try:
|
|
import soundfile as sf
|
|
import ffmpeg
|
|
|
|
soundfile_not_found = False
|
|
except:
|
|
soundfile_not_found = True
|
|
|
|
from transformers import WhisperProcessor, WhisperForConditionalGeneration
|
|
|
|
from haystack.schema import Span, Answer, SpeechAnswer, Document, SpeechDocument
|
|
from haystack.nodes.audio import AnswerToSpeech, DocumentToSpeech
|
|
from haystack.nodes.audio._text_to_speech import TextToSpeech
|
|
|
|
from ..conftest import SAMPLES_PATH
|
|
|
|
|
|
class WhisperHelper:
|
|
def __init__(self, model):
|
|
self._processor = WhisperProcessor.from_pretrained(model)
|
|
self._model = WhisperForConditionalGeneration.from_pretrained(model)
|
|
self._model.config.forced_decoder_ids = None
|
|
|
|
def transcribe(self, media_file: str):
|
|
output, _ = (
|
|
ffmpeg.input(media_file)
|
|
.output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=16000)
|
|
.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
|
|
)
|
|
data = np.frombuffer(output, np.int16).flatten().astype(np.float32) / 32768.0
|
|
|
|
features = self._processor(data, sampling_rate=16000, return_tensors="pt").input_features
|
|
tokens = self._model.generate(features)
|
|
|
|
return self._processor.batch_decode(tokens, skip_special_tokens=True)
|
|
|
|
|
|
@pytest.fixture(scope="session", autouse=True)
|
|
def whisper_helper():
|
|
return WhisperHelper("openai/whisper-medium")
|
|
|
|
|
|
@pytest.mark.skipif(soundfile_not_found, reason="soundfile/ffmpeg not found")
|
|
class TestTextToSpeech:
|
|
def test_text_to_speech_audio_data(self, tmp_path, whisper_helper: WhisperHelper):
|
|
text2speech = TextToSpeech(
|
|
model_name_or_path="espnet/kan-bayashi_ljspeech_vits",
|
|
transformers_params={"seed": 4535, "always_fix_seed": True},
|
|
)
|
|
|
|
audio_data = text2speech.text_to_audio_data(text="answer")
|
|
|
|
sf.write(
|
|
data=audio_data,
|
|
file=str(tmp_path / "audio1.wav"),
|
|
format="wav",
|
|
subtype="PCM_16",
|
|
samplerate=text2speech.model.fs,
|
|
)
|
|
|
|
expedtec_doc = whisper_helper.transcribe(str(SAMPLES_PATH / "audio" / "answer.wav"))
|
|
generated_doc = whisper_helper.transcribe(str(tmp_path / "audio1.wav"))
|
|
|
|
assert expedtec_doc == generated_doc
|
|
|
|
def test_text_to_speech_audio_file(self, tmp_path, whisper_helper: WhisperHelper):
|
|
text2speech = TextToSpeech(
|
|
model_name_or_path="espnet/kan-bayashi_ljspeech_vits",
|
|
transformers_params={"seed": 4535, "always_fix_seed": True},
|
|
)
|
|
|
|
audio_file = text2speech.text_to_audio_file(text="answer", generated_audio_dir=tmp_path / "test_audio")
|
|
assert os.path.exists(audio_file)
|
|
|
|
expected_doc = whisper_helper.transcribe(str(SAMPLES_PATH / "audio" / "answer.wav"))
|
|
generated_doc = whisper_helper.transcribe(str(audio_file))
|
|
|
|
assert expected_doc == generated_doc
|
|
|
|
def test_text_to_speech_compress_audio(self, tmp_path, whisper_helper: WhisperHelper):
|
|
text2speech = TextToSpeech(
|
|
model_name_or_path="espnet/kan-bayashi_ljspeech_vits",
|
|
transformers_params={"seed": 4535, "always_fix_seed": True},
|
|
)
|
|
expected_audio_file = SAMPLES_PATH / "audio" / "answer.wav"
|
|
audio_file = text2speech.text_to_audio_file(
|
|
text="answer", generated_audio_dir=tmp_path / "test_audio", audio_format="mp3"
|
|
)
|
|
assert os.path.exists(audio_file)
|
|
assert audio_file.suffix == ".mp3"
|
|
|
|
expected_doc = whisper_helper.transcribe(str(expected_audio_file))
|
|
generated_doc = whisper_helper.transcribe(str(audio_file))
|
|
|
|
assert expected_doc == generated_doc
|
|
|
|
def test_text_to_speech_naming_function(self, tmp_path, whisper_helper: WhisperHelper):
|
|
text2speech = TextToSpeech(
|
|
model_name_or_path="espnet/kan-bayashi_ljspeech_vits",
|
|
transformers_params={"seed": 4535, "always_fix_seed": True},
|
|
)
|
|
expected_audio_file = SAMPLES_PATH / "audio" / "answer.wav"
|
|
audio_file = text2speech.text_to_audio_file(
|
|
text="answer", generated_audio_dir=tmp_path / "test_audio", audio_naming_function=lambda text: text
|
|
)
|
|
assert os.path.exists(audio_file)
|
|
assert audio_file.name == expected_audio_file.name
|
|
|
|
expected_doc = whisper_helper.transcribe(str(expected_audio_file))
|
|
generated_doc = whisper_helper.transcribe(str(audio_file))
|
|
|
|
assert expected_doc == generated_doc
|
|
|
|
def test_answer_to_speech(self, tmp_path, whisper_helper: WhisperHelper):
|
|
text_answer = Answer(
|
|
answer="answer",
|
|
type="extractive",
|
|
context="the context for this answer is here",
|
|
offsets_in_document=[Span(31, 37)],
|
|
offsets_in_context=[Span(21, 27)],
|
|
meta={"some_meta": "some_value"},
|
|
)
|
|
expected_audio_answer = SAMPLES_PATH / "audio" / "answer.wav"
|
|
expected_audio_context = SAMPLES_PATH / "audio" / "the context for this answer is here.wav"
|
|
|
|
answer2speech = AnswerToSpeech(
|
|
generated_audio_dir=tmp_path / "test_audio",
|
|
audio_params={"audio_naming_function": lambda text: text},
|
|
transformers_params={"seed": 4535, "always_fix_seed": True},
|
|
)
|
|
results, _ = answer2speech.run(answers=[text_answer])
|
|
|
|
audio_answer: SpeechAnswer = results["answers"][0]
|
|
assert isinstance(audio_answer, SpeechAnswer)
|
|
assert audio_answer.type == "generative"
|
|
assert audio_answer.answer_audio.name == expected_audio_answer.name
|
|
assert audio_answer.context_audio.name == expected_audio_context.name
|
|
assert audio_answer.answer == "answer"
|
|
assert audio_answer.context == "the context for this answer is here"
|
|
assert audio_answer.offsets_in_document == [Span(31, 37)]
|
|
assert audio_answer.offsets_in_context == [Span(21, 27)]
|
|
assert audio_answer.meta["some_meta"] == "some_value"
|
|
assert audio_answer.meta["audio_format"] == "wav"
|
|
|
|
expected_doc = whisper_helper.transcribe(str(expected_audio_answer))
|
|
generated_doc = whisper_helper.transcribe(str(audio_answer.answer_audio))
|
|
|
|
assert expected_doc == generated_doc
|
|
|
|
def test_document_to_speech(self, tmp_path, whisper_helper: WhisperHelper):
|
|
text_doc = Document(
|
|
content="this is the content of the document", content_type="text", meta={"name": "test_document.txt"}
|
|
)
|
|
expected_audio_content = SAMPLES_PATH / "audio" / "this is the content of the document.wav"
|
|
|
|
doc2speech = DocumentToSpeech(
|
|
generated_audio_dir=tmp_path / "test_audio",
|
|
audio_params={"audio_naming_function": lambda text: text},
|
|
transformers_params={"seed": 4535, "always_fix_seed": True},
|
|
)
|
|
|
|
results, _ = doc2speech.run(documents=[text_doc])
|
|
|
|
audio_doc: SpeechDocument = results["documents"][0]
|
|
assert isinstance(audio_doc, SpeechDocument)
|
|
assert audio_doc.content_type == "audio"
|
|
assert audio_doc.content_audio.name == expected_audio_content.name
|
|
assert audio_doc.content == "this is the content of the document"
|
|
assert audio_doc.meta["name"] == "test_document.txt"
|
|
assert audio_doc.meta["audio_format"] == "wav"
|
|
|
|
expected_doc = whisper_helper.transcribe(str(expected_audio_content))
|
|
generated_doc = whisper_helper.transcribe(str(audio_doc.content_audio))
|
|
|
|
assert expected_doc == generated_doc
|