Spaces:
Running
Running
junzhaosun
commited on
Commit
·
21147ce
1
Parent(s):
3bb42a7
fixed bugs
Browse files- app.py +50 -4
- requirements.txt +3 -0
app.py
CHANGED
|
@@ -1,6 +1,50 @@
|
|
| 1 |
#!/usr/local/bin/python3
|
| 2 |
#-*- coding:utf-8 -*-
|
| 3 |
import gradio as gr
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 4 |
|
| 5 |
title = "OpenAI Whisper Large v2"
|
| 6 |
|
|
@@ -40,15 +84,17 @@ examples = [
|
|
| 40 |
["examples/see_in_eyes.wav", None],
|
| 41 |
]
|
| 42 |
|
| 43 |
-
gr.
|
| 44 |
-
|
| 45 |
inputs=[
|
| 46 |
gr.Audio(label="上传语音", source="upload", type="numpy"),
|
| 47 |
gr.Audio(label="录制语音", source="microphone", type="numpy"),
|
| 48 |
],
|
| 49 |
-
outputs=
|
|
|
|
|
|
|
| 50 |
title=title,
|
| 51 |
description=description,
|
| 52 |
article=article,
|
| 53 |
-
examples=examples
|
| 54 |
).launch()
|
|
|
|
| 1 |
#!/usr/local/bin/python3
|
| 2 |
#-*- coding:utf-8 -*-
|
| 3 |
import gradio as gr
|
| 4 |
+
import librosa
|
| 5 |
+
import torch
|
| 6 |
+
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
|
| 7 |
+
|
| 8 |
+
checkpoint = "openai/whisper-large-v2"
|
| 9 |
+
processor = AutoProcessor.from_pretrained(checkpoint)
|
| 10 |
+
model = AutoModelForSpeechSeq2Seq.from_pretrained(checkpoint)
|
| 11 |
+
|
| 12 |
+
|
| 13 |
+
def process_audio(sampling_rate, waveform):
|
| 14 |
+
# convert from int16 to floating point
|
| 15 |
+
waveform = waveform / 32678.0
|
| 16 |
+
|
| 17 |
+
# convert to mono if stereo
|
| 18 |
+
if len(waveform.shape) > 1:
|
| 19 |
+
waveform = librosa.to_mono(waveform.T)
|
| 20 |
+
|
| 21 |
+
# resample to 16 kHz if necessary
|
| 22 |
+
if sampling_rate != 16000:
|
| 23 |
+
waveform = librosa.resample(waveform, orig_sr=sampling_rate, target_sr=16000)
|
| 24 |
+
|
| 25 |
+
# limit to 30 seconds
|
| 26 |
+
waveform = waveform[:16000*30]
|
| 27 |
+
|
| 28 |
+
# make PyTorch tensor
|
| 29 |
+
waveform = torch.tensor(waveform)
|
| 30 |
+
return waveform
|
| 31 |
+
|
| 32 |
+
|
| 33 |
+
def predict(audio, mic_audio=None):
|
| 34 |
+
# audio = tuple (sample_rate, frames) or (sample_rate, (frames, channels))
|
| 35 |
+
if mic_audio is not None:
|
| 36 |
+
sampling_rate, waveform = mic_audio
|
| 37 |
+
elif audio is not None:
|
| 38 |
+
sampling_rate, waveform = audio
|
| 39 |
+
else:
|
| 40 |
+
return "(please provide audio)"
|
| 41 |
+
|
| 42 |
+
waveform = process_audio(sampling_rate, waveform)
|
| 43 |
+
inputs = processor(audio=waveform, sampling_rate=16000, return_tensors="pt")
|
| 44 |
+
predicted_ids = model.generate(**inputs, max_length=400)
|
| 45 |
+
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)
|
| 46 |
+
return transcription[0]
|
| 47 |
+
|
| 48 |
|
| 49 |
title = "OpenAI Whisper Large v2"
|
| 50 |
|
|
|
|
| 84 |
["examples/see_in_eyes.wav", None],
|
| 85 |
]
|
| 86 |
|
| 87 |
+
gr.Interface(
|
| 88 |
+
fn=predict,
|
| 89 |
inputs=[
|
| 90 |
gr.Audio(label="上传语音", source="upload", type="numpy"),
|
| 91 |
gr.Audio(label="录制语音", source="microphone", type="numpy"),
|
| 92 |
],
|
| 93 |
+
outputs=[
|
| 94 |
+
gr.Text(label="识别出的文字"),
|
| 95 |
+
],
|
| 96 |
title=title,
|
| 97 |
description=description,
|
| 98 |
article=article,
|
| 99 |
+
examples=examples,
|
| 100 |
).launch()
|
requirements.txt
CHANGED
|
@@ -0,0 +1,3 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
| 1 |
+
git+https://github.com/huggingface/transformers.git
|
| 2 |
+
torch
|
| 3 |
+
librosa
|