csukuangfj
commited on
Commit
·
a97e72d
1
Parent(s):
588da9c
minor fixes.
Browse files- app.py +89 -36
- model.py +159 -22
- offline_asr.py +40 -32
app.py
CHANGED
@@ -19,6 +19,7 @@
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# References:
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# https://gradio.app/docs/#dropdown
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import os
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import time
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from datetime import datetime
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@@ -26,43 +27,43 @@ from datetime import datetime
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import gradio as gr
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import torchaudio
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-
from model import
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get_gigaspeech_pre_trained_model,
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sample_rate,
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get_wenetspeech_pre_trained_model,
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)
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-
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"Chinese": get_wenetspeech_pre_trained_model(),
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"English": get_gigaspeech_pre_trained_model(),
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}
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def convert_to_wav(in_filename: str) -> str:
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"""Convert the input audio file to a wave file"""
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out_filename = in_filename + ".wav"
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-
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_ = os.system(f"ffmpeg -hide_banner -i '{in_filename}' '{out_filename}'")
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return out_filename
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-
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-
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def process(in_filename: str, language: str) -> str:
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print("in_filename", in_filename)
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print("language", language)
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filename = convert_to_wav(in_filename)
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now = datetime.now()
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date_time = now.strftime("%Y-%m-%d %H:%M:%S.%f")
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-
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start = time.time()
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wave, wave_sample_rate = torchaudio.load(filename)
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if wave_sample_rate != sample_rate:
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-
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f"Expected sample rate: {sample_rate}. Given: {wave_sample_rate}. "
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f"Resampling to {sample_rate}."
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)
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@@ -74,7 +75,11 @@ def process(in_filename: str, language: str) -> str:
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)
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wave = wave[0] # use only the first channel.
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hyp =
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date_time = now.strftime("%Y-%m-%d %H:%M:%S.%f")
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end = time.time()
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@@ -82,11 +87,10 @@ def process(in_filename: str, language: str) -> str:
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duration = wave.shape[0] / sample_rate
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rtf = (end - start) / duration
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-
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-
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-
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print(hyp)
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return hyp
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@@ -103,51 +107,100 @@ See more information by visiting the following links:
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- <https://github.com/lhotse-speech/lhotse>
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"""
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with demo:
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gr.Markdown(title)
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-
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-
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label="Language",
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choices=language_choices,
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)
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with gr.Tabs():
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with gr.TabItem("Upload from disk"):
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uploaded_file = gr.
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source="upload", # Choose between "microphone", "upload"
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type="filepath",
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optional=False,
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label="Upload from disk",
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)
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upload_button = gr.Button("Submit for recognition")
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uploaded_output = gr.
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label="Recognized speech from uploaded file"
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)
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with gr.TabItem("Record from microphone"):
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microphone = gr.
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source="microphone", # Choose between "microphone", "upload"
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type="filepath",
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optional=False,
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label="Record from microphone",
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)
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recorded_output = gr.outputs.Textbox(
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label="Recognized speech from recordings"
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)
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record_button = gr.Button("Submit for recognition")
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upload_button.click(
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process,
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inputs=[
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outputs=uploaded_output,
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)
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record_button.click(
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process,
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inputs=[
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outputs=recorded_output,
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)
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if __name__ == "__main__":
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demo.launch()
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# References:
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# https://gradio.app/docs/#dropdown
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+
import logging
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import os
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import time
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from datetime import datetime
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import gradio as gr
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import torchaudio
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from model import get_pretrained_model, language_to_models, sample_rate
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languages = sorted(language_to_models.keys())
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def convert_to_wav(in_filename: str) -> str:
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"""Convert the input audio file to a wave file"""
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out_filename = in_filename + ".wav"
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logging.info(f"Converting '{in_filename}' to '{out_filename}'")
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_ = os.system(f"ffmpeg -hide_banner -i '{in_filename}' '{out_filename}'")
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return out_filename
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def process(
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in_filename: str,
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language: str,
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repo_id: str,
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decoding_method: str,
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num_active_paths: int,
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) -> str:
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logging.info(f"in_filename: {in_filename}")
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logging.info(f"language: {language}")
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logging.info(f"repo_id: {repo_id}")
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logging.info(f"decoding_method: {decoding_method}")
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logging.info(f"num_active_paths: {num_active_paths}")
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filename = convert_to_wav(in_filename)
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now = datetime.now()
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date_time = now.strftime("%Y-%m-%d %H:%M:%S.%f")
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logging.info(f"Started at {date_time}")
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start = time.time()
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wave, wave_sample_rate = torchaudio.load(filename)
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if wave_sample_rate != sample_rate:
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logging.info(
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f"Expected sample rate: {sample_rate}. Given: {wave_sample_rate}. "
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f"Resampling to {sample_rate}."
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)
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)
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wave = wave[0] # use only the first channel.
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hyp = get_pretrained_model(repo_id).decode_waves(
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[wave],
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decoding_method=decoding_method,
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num_active_paths=num_active_paths,
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)[0]
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date_time = now.strftime("%Y-%m-%d %H:%M:%S.%f")
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end = time.time()
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duration = wave.shape[0] / sample_rate
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rtf = (end - start) / duration
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logging.info(f"Finished at {date_time} s. Elapsed: {end - start: .3f} s")
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logging.info(f"Duration {duration: .3f} s")
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logging.info(f"RTF {rtf: .3f}")
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logging.info(f"hyp:\n{hyp}")
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return hyp
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- <https://github.com/lhotse-speech/lhotse>
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"""
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def update_model_dropdown(language: str):
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if language in language_to_models:
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choices = language_to_models[language]
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return gr.Dropdown.update(choices=choices, value=choices[0])
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raise ValueError(f"Unsupported language: {language}")
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demo = gr.Blocks()
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with demo:
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gr.Markdown(title)
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language_choices = list(language_to_models.keys())
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language_radio = gr.Radio(
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label="Language",
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choices=language_choices,
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value=language_choices[0],
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)
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model_dropdown = gr.Dropdown(
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choices=language_to_models[language_choices[0]],
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label="Select a model",
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value=language_to_models[language_choices[0]][0],
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)
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language_radio.change(
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update_model_dropdown,
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inputs=language_radio,
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outputs=model_dropdown,
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)
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decoding_method_radio = gr.Radio(
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label="Decoding method",
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choices=["greedy_search", "modified_beam_search"],
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value="greedy_search",
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)
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num_active_paths_slider = gr.Slider(
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minimum=1,
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value=4,
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step=1,
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label="Number of active paths for modified_beam_search",
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)
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with gr.Tabs():
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with gr.TabItem("Upload from disk"):
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uploaded_file = gr.Audio(
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source="upload", # Choose between "microphone", "upload"
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type="filepath",
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optional=False,
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label="Upload from disk",
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)
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upload_button = gr.Button("Submit for recognition")
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uploaded_output = gr.Textbox(label="Recognized speech from uploaded file")
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with gr.TabItem("Record from microphone"):
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microphone = gr.Audio(
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source="microphone", # Choose between "microphone", "upload"
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type="filepath",
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optional=False,
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label="Record from microphone",
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)
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record_button = gr.Button("Submit for recognition")
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recorded_output = gr.Textbox(label="Recognized speech from recordings")
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upload_button.click(
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process,
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inputs=[
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uploaded_file,
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language_radio,
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model_dropdown,
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decoding_method_radio,
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num_active_paths_slider,
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],
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outputs=uploaded_output,
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)
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record_button.click(
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process,
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inputs=[
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microphone,
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language_radio,
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model_dropdown,
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decoding_method_radio,
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num_active_paths_slider,
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],
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outputs=recorded_output,
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)
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gr.Markdown(description)
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if __name__ == "__main__":
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formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
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logging.basicConfig(format=formatter, level=logging.INFO)
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demo.launch()
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model.py
CHANGED
@@ -23,52 +23,189 @@ from offline_asr import OfflineAsr
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sample_rate = 16000
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@lru_cache(maxsize=
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def
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nn_model_filename = hf_hub_download(
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subfolder="exp",
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)
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bpe_model_filename = hf_hub_download(
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repo_id=
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filename=
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subfolder=
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)
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return OfflineAsr(
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nn_model_filename=nn_model_filename,
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bpe_model_filename=bpe_model_filename,
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token_filename=None,
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decoding_method="greedy_search",
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num_active_paths=4,
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sample_rate=sample_rate,
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device="cpu",
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)
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@lru_cache(maxsize=
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def
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-
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filename="cpu_jit_epoch_10_avg_2_torch_1.7.1.pt",
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subfolder="exp",
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)
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-
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)
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return OfflineAsr(
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nn_model_filename=nn_model_filename,
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bpe_model_filename=None,
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token_filename=token_filename,
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decoding_method="greedy_search",
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num_active_paths=4,
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sample_rate=sample_rate,
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device="cpu",
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)
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sample_rate = 16000
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@lru_cache(maxsize=30)
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def get_pretrained_model(repo_id: str) -> OfflineAsr:
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if repo_id in chinese_models:
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return chinese_models[repo_id](repo_id)
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elif repo_id in english_models:
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return english_models[repo_id](repo_id)
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elif repo_id in chinese_english_mixed_models:
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return chinese_english_mixed_models[repo_id](repo_id)
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else:
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raise ValueError(f"Unsupported repo_id: {repo_id}")
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def _get_nn_model_filename(
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repo_id: str,
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filename: str,
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subfolder: str = "exp",
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) -> str:
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nn_model_filename = hf_hub_download(
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repo_id=repo_id,
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filename=filename,
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subfolder=subfolder,
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)
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return nn_model_filename
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+
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def _get_bpe_model_filename(
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repo_id: str,
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filename: str = "bpe.model",
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subfolder: str = "data/lang_bpe_500",
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) -> str:
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bpe_model_filename = hf_hub_download(
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repo_id=repo_id,
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filename=filename,
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subfolder=subfolder,
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)
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return bpe_model_filename
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+
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def _get_token_filename(
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repo_id: str,
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filename: str = "tokens.txt",
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subfolder: str = "data/lang_char",
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) -> str:
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token_filename = hf_hub_download(
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repo_id=repo_id,
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filename=filename,
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subfolder=subfolder,
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)
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return token_filename
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+
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+
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@lru_cache(maxsize=10)
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def _get_aishell2_pretrained_model(repo_id: str) -> OfflineAsr:
|
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assert repo_id in [
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+
# context-size 1
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81 |
+
"yuekai/icefall-asr-aishell2-pruned-transducer-stateless5-A-2022-07-12", # noqa
|
82 |
+
# context-size 2
|
83 |
+
"yuekai/icefall-asr-aishell2-pruned-transducer-stateless5-B-2022-07-12", # noqa
|
84 |
+
]
|
85 |
+
|
86 |
+
nn_model_filename = _get_nn_model_filename(
|
87 |
+
repo_id=repo_id,
|
88 |
+
filename="cpu_jit.pt",
|
89 |
+
)
|
90 |
+
token_filename = _get_token_filename(repo_id=repo_id)
|
91 |
+
|
92 |
+
return OfflineAsr(
|
93 |
+
nn_model_filename=nn_model_filename,
|
94 |
+
bpe_model_filename=None,
|
95 |
+
token_filename=token_filename,
|
96 |
+
sample_rate=sample_rate,
|
97 |
+
device="cpu",
|
98 |
+
)
|
99 |
+
|
100 |
+
|
101 |
+
@lru_cache(maxsize=10)
|
102 |
+
def _get_gigaspeech_pre_trained_model(repo_id: str) -> OfflineAsr:
|
103 |
+
assert repo_id in [
|
104 |
+
"wgb14/icefall-asr-gigaspeech-pruned-transducer-stateless2",
|
105 |
+
]
|
106 |
+
|
107 |
+
nn_model_filename = _get_nn_model_filename(
|
108 |
+
# It is converted from https://huggingface.co/wgb14/icefall-asr-gigaspeech-pruned-transducer-stateless2 # noqa
|
109 |
+
repo_id="csukuangfj/icefall-asr-gigaspeech-pruned-transducer-stateless2", # noqa
|
110 |
+
filename="cpu_jit-epoch-29-avg-11-torch-1.10.0.pt",
|
111 |
)
|
112 |
+
bpe_model_filename = _get_bpe_model_filename(repo_id=repo_id)
|
113 |
|
114 |
return OfflineAsr(
|
115 |
nn_model_filename=nn_model_filename,
|
116 |
bpe_model_filename=bpe_model_filename,
|
117 |
token_filename=None,
|
|
|
|
|
118 |
sample_rate=sample_rate,
|
119 |
device="cpu",
|
120 |
)
|
121 |
|
122 |
|
123 |
+
@lru_cache(maxsize=10)
|
124 |
+
def _get_librispeech_pre_trained_model(repo_id: str) -> OfflineAsr:
|
125 |
+
assert repo_id in [
|
126 |
+
"csukuangfj/icefall-asr-librispeech-pruned-transducer-stateless3-2022-05-13", # noqa
|
127 |
+
]
|
128 |
+
|
129 |
+
nn_model_filename = _get_nn_model_filename(
|
130 |
+
repo_id=repo_id,
|
131 |
+
filename="cpu_jit.pt",
|
132 |
+
)
|
133 |
+
bpe_model_filename = _get_bpe_model_filename(repo_id=repo_id)
|
134 |
+
|
135 |
+
return OfflineAsr(
|
136 |
+
nn_model_filename=nn_model_filename,
|
137 |
+
bpe_model_filename=bpe_model_filename,
|
138 |
+
token_filename=None,
|
139 |
+
sample_rate=sample_rate,
|
140 |
+
device="cpu",
|
141 |
+
)
|
142 |
+
|
143 |
+
|
144 |
+
@lru_cache(maxsize=10)
|
145 |
+
def _get_wenetspeech_pre_trained_model(repo_id: str):
|
146 |
+
assert repo_id in [
|
147 |
+
"luomingshuang/icefall_asr_wenetspeech_pruned_transducer_stateless2",
|
148 |
+
]
|
149 |
+
|
150 |
+
nn_model_filename = _get_nn_model_filename(
|
151 |
+
repo_id=repo_id,
|
152 |
filename="cpu_jit_epoch_10_avg_2_torch_1.7.1.pt",
|
|
|
153 |
)
|
154 |
+
token_filename = _get_token_filename(repo_id=repo_id)
|
155 |
|
156 |
+
return OfflineAsr(
|
157 |
+
nn_model_filename=nn_model_filename,
|
158 |
+
bpe_model_filename=None,
|
159 |
+
token_filename=token_filename,
|
160 |
+
sample_rate=sample_rate,
|
161 |
+
device="cpu",
|
162 |
+
)
|
163 |
+
|
164 |
+
|
165 |
+
@lru_cache(maxsize=10)
|
166 |
+
def _get_tal_csasr_pre_trained_model(repo_id: str):
|
167 |
+
assert repo_id in [
|
168 |
+
"luomingshuang/icefall_asr_tal-csasr_pruned_transducer_stateless5",
|
169 |
+
]
|
170 |
+
|
171 |
+
nn_model_filename = _get_nn_model_filename(
|
172 |
+
repo_id=repo_id,
|
173 |
+
filename="cpu_jit.pt",
|
174 |
)
|
175 |
+
token_filename = _get_token_filename(repo_id=repo_id)
|
176 |
|
177 |
return OfflineAsr(
|
178 |
nn_model_filename=nn_model_filename,
|
179 |
bpe_model_filename=None,
|
180 |
token_filename=token_filename,
|
|
|
|
|
181 |
sample_rate=sample_rate,
|
182 |
device="cpu",
|
183 |
)
|
184 |
+
|
185 |
+
|
186 |
+
chinese_models = {
|
187 |
+
"yuekai/icefall-asr-aishell2-pruned-transducer-stateless5-A-2022-07-12": _get_aishell2_pretrained_model, # noqa
|
188 |
+
"yuekai/icefall-asr-aishell2-pruned-transducer-stateless5-B-2022-07-12": _get_aishell2_pretrained_model, # noqa
|
189 |
+
"luomingshuang/icefall_asr_wenetspeech_pruned_transducer_stateless2": _get_wenetspeech_pre_trained_model, # noqa
|
190 |
+
}
|
191 |
+
|
192 |
+
english_models = {
|
193 |
+
"wgb14/icefall-asr-gigaspeech-pruned-transducer-stateless2": _get_gigaspeech_pre_trained_model, # noqa
|
194 |
+
"csukuangfj/icefall-asr-librispeech-pruned-transducer-stateless3-2022-05-13": _get_librispeech_pre_trained_model, # noqa
|
195 |
+
}
|
196 |
+
|
197 |
+
chinese_english_mixed_models = {
|
198 |
+
"luomingshuang/icefall_asr_tal-csasr_pruned_transducer_stateless5": _get_tal_csasr_pre_trained_model, # noqa
|
199 |
+
}
|
200 |
+
|
201 |
+
all_models = {
|
202 |
+
**chinese_models,
|
203 |
+
**english_models,
|
204 |
+
**chinese_english_mixed_models,
|
205 |
+
}
|
206 |
+
|
207 |
+
language_to_models = {
|
208 |
+
"Chinese": sorted(chinese_models.keys()),
|
209 |
+
"English": sorted(english_models.keys()),
|
210 |
+
"Chinese+English": sorted(chinese_english_mixed_models.keys()),
|
211 |
+
}
|
offline_asr.py
CHANGED
@@ -206,10 +206,10 @@ class OfflineAsr(object):
|
|
206 |
def __init__(
|
207 |
self,
|
208 |
nn_model_filename: str,
|
209 |
-
bpe_model_filename: Optional[str],
|
210 |
-
token_filename: Optional[str],
|
211 |
-
decoding_method: str,
|
212 |
-
num_active_paths: int,
|
213 |
sample_rate: int = 16000,
|
214 |
device: Union[str, torch.device] = "cpu",
|
215 |
):
|
@@ -223,14 +223,6 @@ class OfflineAsr(object):
|
|
223 |
token_filename:
|
224 |
Path to tokens.txt. If it is None, you have to provide
|
225 |
`bpe_model_filename`.
|
226 |
-
decoding_method:
|
227 |
-
The decoding method to use. Currently, only greedy_search and
|
228 |
-
modified_beam_search are implemented.
|
229 |
-
num_active_paths:
|
230 |
-
Used only when decoding_method is modified_beam_search.
|
231 |
-
It specifies number of active paths for each utterance. Due to
|
232 |
-
merging paths with identical token sequences, the actual number
|
233 |
-
may be less than "num_active_paths".
|
234 |
sample_rate:
|
235 |
Expected sample rate of the feature extractor.
|
236 |
device:
|
@@ -246,6 +238,7 @@ class OfflineAsr(object):
|
|
246 |
self.sp = spm.SentencePieceProcessor()
|
247 |
self.sp.load(bpe_model_filename)
|
248 |
else:
|
|
|
249 |
self.token_table = k2.SymbolTable.from_file(token_filename)
|
250 |
|
251 |
self.feature_extractor = self._build_feature_extractor(
|
@@ -253,24 +246,6 @@ class OfflineAsr(object):
|
|
253 |
device=device,
|
254 |
)
|
255 |
|
256 |
-
assert decoding_method in (
|
257 |
-
"greedy_search",
|
258 |
-
"modified_beam_search",
|
259 |
-
), decoding_method
|
260 |
-
if decoding_method == "greedy_search":
|
261 |
-
nn_and_decoding_func = run_model_and_do_greedy_search
|
262 |
-
elif decoding_method == "modified_beam_search":
|
263 |
-
nn_and_decoding_func = functools.partial(
|
264 |
-
run_model_and_do_modified_beam_search,
|
265 |
-
num_active_paths=num_active_paths,
|
266 |
-
)
|
267 |
-
else:
|
268 |
-
raise ValueError(
|
269 |
-
f"Unsupported decoding_method: {decoding_method} "
|
270 |
-
"Please use greedy_search or modified_beam_search"
|
271 |
-
)
|
272 |
-
|
273 |
-
self.nn_and_decoding_func = nn_and_decoding_func
|
274 |
self.device = device
|
275 |
|
276 |
def _build_feature_extractor(
|
@@ -299,7 +274,12 @@ class OfflineAsr(object):
|
|
299 |
|
300 |
return fbank
|
301 |
|
302 |
-
def decode_waves(
|
|
|
|
|
|
|
|
|
|
|
303 |
"""
|
304 |
Args:
|
305 |
waves:
|
@@ -313,20 +293,48 @@ class OfflineAsr(object):
|
|
313 |
then the given waves have to contain samples in this range.
|
314 |
|
315 |
All models trained in icefall use the normalized range [-1, 1].
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
316 |
Returns:
|
317 |
Return a list of decoded results. `ans[i]` contains the decoded
|
318 |
results for `wavs[i]`.
|
319 |
"""
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
320 |
waves = [w.to(self.device) for w in waves]
|
321 |
features = self.feature_extractor(waves)
|
322 |
|
323 |
-
tokens =
|
324 |
|
325 |
if hasattr(self, "sp"):
|
326 |
results = self.sp.decode(tokens)
|
327 |
else:
|
328 |
results = [[self.token_table[i] for i in hyp] for hyp in tokens]
|
|
|
329 |
results = ["".join(r) for r in results]
|
|
|
330 |
|
331 |
return results
|
332 |
|
|
|
206 |
def __init__(
|
207 |
self,
|
208 |
nn_model_filename: str,
|
209 |
+
bpe_model_filename: Optional[str] = None,
|
210 |
+
token_filename: Optional[str] = None,
|
211 |
+
decoding_method: str = "greedy_search",
|
212 |
+
num_active_paths: int = 4,
|
213 |
sample_rate: int = 16000,
|
214 |
device: Union[str, torch.device] = "cpu",
|
215 |
):
|
|
|
223 |
token_filename:
|
224 |
Path to tokens.txt. If it is None, you have to provide
|
225 |
`bpe_model_filename`.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
226 |
sample_rate:
|
227 |
Expected sample rate of the feature extractor.
|
228 |
device:
|
|
|
238 |
self.sp = spm.SentencePieceProcessor()
|
239 |
self.sp.load(bpe_model_filename)
|
240 |
else:
|
241 |
+
assert token_filename is not None, token_filename
|
242 |
self.token_table = k2.SymbolTable.from_file(token_filename)
|
243 |
|
244 |
self.feature_extractor = self._build_feature_extractor(
|
|
|
246 |
device=device,
|
247 |
)
|
248 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
249 |
self.device = device
|
250 |
|
251 |
def _build_feature_extractor(
|
|
|
274 |
|
275 |
return fbank
|
276 |
|
277 |
+
def decode_waves(
|
278 |
+
self,
|
279 |
+
waves: List[torch.Tensor],
|
280 |
+
decoding_method: str,
|
281 |
+
num_active_paths: int,
|
282 |
+
) -> List[List[str]]:
|
283 |
"""
|
284 |
Args:
|
285 |
waves:
|
|
|
293 |
then the given waves have to contain samples in this range.
|
294 |
|
295 |
All models trained in icefall use the normalized range [-1, 1].
|
296 |
+
decoding_method:
|
297 |
+
The decoding method to use. Currently, only greedy_search and
|
298 |
+
modified_beam_search are implemented.
|
299 |
+
num_active_paths:
|
300 |
+
Used only when decoding_method is modified_beam_search.
|
301 |
+
It specifies number of active paths for each utterance. Due to
|
302 |
+
merging paths with identical token sequences, the actual number
|
303 |
+
may be less than "num_active_paths".
|
304 |
Returns:
|
305 |
Return a list of decoded results. `ans[i]` contains the decoded
|
306 |
results for `wavs[i]`.
|
307 |
"""
|
308 |
+
assert decoding_method in (
|
309 |
+
"greedy_search",
|
310 |
+
"modified_beam_search",
|
311 |
+
), decoding_method
|
312 |
+
|
313 |
+
if decoding_method == "greedy_search":
|
314 |
+
nn_and_decoding_func = run_model_and_do_greedy_search
|
315 |
+
elif decoding_method == "modified_beam_search":
|
316 |
+
nn_and_decoding_func = functools.partial(
|
317 |
+
run_model_and_do_modified_beam_search,
|
318 |
+
num_active_paths=num_active_paths,
|
319 |
+
)
|
320 |
+
else:
|
321 |
+
raise ValueError(
|
322 |
+
f"Unsupported decoding_method: {decoding_method} "
|
323 |
+
"Please use greedy_search or modified_beam_search"
|
324 |
+
)
|
325 |
+
|
326 |
waves = [w.to(self.device) for w in waves]
|
327 |
features = self.feature_extractor(waves)
|
328 |
|
329 |
+
tokens = nn_and_decoding_func(self.model, features)
|
330 |
|
331 |
if hasattr(self, "sp"):
|
332 |
results = self.sp.decode(tokens)
|
333 |
else:
|
334 |
results = [[self.token_table[i] for i in hyp] for hyp in tokens]
|
335 |
+
blank = chr(0x2581)
|
336 |
results = ["".join(r) for r in results]
|
337 |
+
results = [r.replace(blank, " ") for r in results]
|
338 |
|
339 |
return results
|
340 |
|