added test inference script
Browse files- inference.py +80 -0
inference.py
ADDED
|
@@ -0,0 +1,80 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 1 |
+
from transformers import Wav2Vec2Processor, AutoConfig
|
| 2 |
+
import onnxruntime as rt
|
| 3 |
+
import torch
|
| 4 |
+
import torch.nn.functional as F
|
| 5 |
+
import numpy as np
|
| 6 |
+
import os
|
| 7 |
+
import torchaudio
|
| 8 |
+
import soundfile as sf
|
| 9 |
+
|
| 10 |
+
|
| 11 |
+
class EndOfSpeechDetection:
|
| 12 |
+
processor: Wav2Vec2Processor
|
| 13 |
+
config: AutoConfig
|
| 14 |
+
session: rt.InferenceSession
|
| 15 |
+
|
| 16 |
+
def load_model(self, path, use_gpu=False):
|
| 17 |
+
processor = Wav2Vec2Processor.from_pretrained(path)
|
| 18 |
+
config = AutoConfig.from_pretrained(path)
|
| 19 |
+
|
| 20 |
+
sess_options = rt.SessionOptions()
|
| 21 |
+
sess_options.graph_optimization_level = rt.GraphOptimizationLevel.ORT_ENABLE_ALL
|
| 22 |
+
|
| 23 |
+
providers = ["ROCMExecutionProvider"] if use_gpu else ["CPUExecutionProvider"]
|
| 24 |
+
session = rt.InferenceSession(
|
| 25 |
+
os.path.join(path, "model.onnx"), sess_options, providers=providers
|
| 26 |
+
)
|
| 27 |
+
return processor, config, session
|
| 28 |
+
|
| 29 |
+
def predict(self, segment, file_type="pcm"):
|
| 30 |
+
if file_type == "pcm":
|
| 31 |
+
# pcm files
|
| 32 |
+
speech_array = np.memmap(segment, dtype="float32", mode="r").astype(
|
| 33 |
+
np.float32
|
| 34 |
+
)
|
| 35 |
+
else:
|
| 36 |
+
# wave files
|
| 37 |
+
speech_array, _ = torchaudio.load(segment)
|
| 38 |
+
speech_array = speech_array[0].numpy()
|
| 39 |
+
|
| 40 |
+
features = self.processor(
|
| 41 |
+
speech_array, sampling_rate=16000, return_tensors="pt", padding=True
|
| 42 |
+
)
|
| 43 |
+
input_values = features.input_values
|
| 44 |
+
outputs = self.session.run(
|
| 45 |
+
[self.session.get_outputs()[-1].name],
|
| 46 |
+
{self.session.get_inputs()[-1].name: input_values.detach().cpu().numpy()},
|
| 47 |
+
)[0]
|
| 48 |
+
softmax_output = F.softmax(torch.tensor(outputs), dim=1)
|
| 49 |
+
|
| 50 |
+
both_classes_with_prob = {
|
| 51 |
+
self.config.id2label[i]: softmax_output[0][i].item()
|
| 52 |
+
for i in range(len(softmax_output[0]))
|
| 53 |
+
}
|
| 54 |
+
|
| 55 |
+
return both_classes_with_prob
|
| 56 |
+
|
| 57 |
+
|
| 58 |
+
if __name__ == "__main__":
|
| 59 |
+
eos = EndOfSpeechDetection()
|
| 60 |
+
eos.processor, eos.config, eos.session = eos.load_model("eos-model-onnx")
|
| 61 |
+
|
| 62 |
+
audio_file = "5sec_audio.wav"
|
| 63 |
+
audio, sr = torchaudio.load(audio_file)
|
| 64 |
+
audio = audio[0].numpy()
|
| 65 |
+
audio_len = len(audio)
|
| 66 |
+
segment_len = 700 * sr // 1000
|
| 67 |
+
segments = []
|
| 68 |
+
for i in range(0, audio_len, segment_len):
|
| 69 |
+
if i + segment_len < audio_len:
|
| 70 |
+
segment = audio[i : i + segment_len]
|
| 71 |
+
else:
|
| 72 |
+
segment = audio[i:]
|
| 73 |
+
|
| 74 |
+
segments.append(segment)
|
| 75 |
+
|
| 76 |
+
if not os.path.exists("segments"):
|
| 77 |
+
os.makedirs("segments")
|
| 78 |
+
for i, segment in enumerate(segments):
|
| 79 |
+
sf.write(f"segments/segment_{i}.wav", segment, sr)
|
| 80 |
+
print(eos.predict(f"segments/segment_{i}.wav", file_type="wav"))
|