Update app.py
Browse files
app.py
CHANGED
@@ -1,21 +1,10 @@
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import torch
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import gradio as gr
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import yt_dlp as youtube_dl
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import numpy as np
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from datasets import Dataset, Audio
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from scipy.io import wavfile
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from transformers import pipeline
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from
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import tempfile
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import os
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import time
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import demucs.api
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MODEL_NAME = "openai/whisper-large-v3"
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DEMUCS_MODEL_NAME = "htdemucs_ft"
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BATCH_SIZE = 8
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FILE_LIMIT_MB = 1000
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YT_LENGTH_LIMIT_S = 3600 # limit to 1 hour YouTube files
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device = 0 if torch.cuda.is_available() else "cpu"
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@@ -26,181 +15,30 @@ pipe = pipeline(
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device=device,
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)
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def separate_vocal(path):
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origin, separated = separator.separate_audio_file(path)
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demucs.api.save_audio(separated["vocals"], path, samplerate=separator.samplerate)
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return path
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def transcribe(inputs_path, task, use_demucs, dataset_name, oauth_token: gr.OAuthToken | None, progress=gr.Progress()):
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if inputs_path is None:
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raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting your request.")
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if dataset_name is None:
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raise gr.Error("No dataset name submitted! Please submit a dataset name. Should be in the format : <user>/<dataset_name> or <org>/<dataset_name>. Also accepts <dataset_name>, which will default to the namespace of the logged-in user.")
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if oauth_token is None:
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gr.Warning("Make sure to click and login before using this demo.")
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return [["transcripts will appear here"]], ""
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total_step = 4
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current_step = 0
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current_step += 1
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progress((current_step, total_step), desc="Transcribe using Whisper.")
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sampling_rate, inputs = wavfile.read(inputs_path)
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out = pipe(inputs_path, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=True)
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text = out["text"]
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print(f"Separating vocals #{i}")
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path = separate_vocal(path)
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audios.append(path)
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transcripts.append(chunk["text"])
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dataset = Dataset.from_dict({"audio": audios, "text": transcripts}).cast_column("audio", Audio())
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current_step += 1
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progress((current_step, total_step), desc="Push dataset.")
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dataset.push_to_hub(dataset_name, token=oauth_token.token if oauth_token else oauth_token)
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return [[transcript] for transcript in transcripts], text
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def _return_yt_html_embed(yt_url):
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video_id = yt_url.split("?v=")[-1]
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HTML_str = (
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f'<center> <iframe width="500" height="320" src="https://www.youtube.com/embed/{video_id}"> </iframe>'
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" </center>"
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)
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return HTML_str
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def download_yt_audio(yt_url, filename):
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info_loader = youtube_dl.YoutubeDL()
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try:
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info = info_loader.extract_info(yt_url, download=False)
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except youtube_dl.utils.DownloadError as err:
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raise gr.Error(str(err))
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file_length = info["duration_string"]
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file_h_m_s = file_length.split(":")
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file_h_m_s = [int(sub_length) for sub_length in file_h_m_s]
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if len(file_h_m_s) == 1:
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file_h_m_s.insert(0, 0)
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if len(file_h_m_s) == 2:
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file_h_m_s.insert(0, 0)
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file_length_s = file_h_m_s[0] * 3600 + file_h_m_s[1] * 60 + file_h_m_s[2]
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if file_length_s > YT_LENGTH_LIMIT_S:
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yt_length_limit_hms = time.strftime("%HH:%MM:%SS", time.gmtime(YT_LENGTH_LIMIT_S))
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file_length_hms = time.strftime("%HH:%MM:%SS", time.gmtime(file_length_s))
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raise gr.Error(f"Maximum YouTube length is {yt_length_limit_hms}, got {file_length_hms} YouTube video.")
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ydl_opts = {"outtmpl": filename, "format": "worstvideo[ext=mp4]+bestaudio[ext=m4a]/best[ext=mp4]/best"}
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with youtube_dl.YoutubeDL(ydl_opts) as ydl:
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try:
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ydl.download([yt_url])
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except youtube_dl.utils.ExtractorError as err:
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raise gr.Error(str(err))
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def yt_transcribe(yt_url, task, use_demucs, dataset_name, oauth_token: gr.OAuthToken | None, max_filesize=75.0, dataset_sampling_rate=24000, progress=gr.Progress()):
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if yt_url is None:
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raise gr.Error("No youtube link submitted! Please put a working link.")
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if dataset_name is None:
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raise gr.Error("No dataset name submitted! Please submit a dataset name. Should be in the format : <user>/<dataset_name> or <org>/<dataset_name>. Also accepts <dataset_name>, which will default to the namespace of the logged-in user.")
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total_step = 5
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current_step = 0
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html_embed_str = _return_yt_html_embed(yt_url)
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if oauth_token is None:
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gr.Warning("Make sure to click and login before using this demo.")
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return html_embed_str, [["transcripts will appear here"]], ""
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current_step += 1
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progress((current_step, total_step), desc="Load video.")
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with tempfile.TemporaryDirectory() as tmpdirname:
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filepath = os.path.join(tmpdirname, "video.mp4")
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download_yt_audio(yt_url, filepath)
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with open(filepath, "rb") as f:
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inputs_path = f.read()
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inputs = ffmpeg_read(inputs_path, pipe.feature_extractor.sampling_rate)
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inputs = {"array": inputs, "sampling_rate": pipe.feature_extractor.sampling_rate}
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current_step += 1
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progress((current_step, total_step), desc="Transcribe using Whisper.")
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out = pipe(inputs, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=True)
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text = out["text"]
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inputs = ffmpeg_read(inputs_path, dataset_sampling_rate)
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current_step += 1
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progress((current_step, total_step), desc="Merge chunks.")
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chunks = naive_postprocess_whisper_chunks(out["chunks"], inputs, dataset_sampling_rate)
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current_step += 1
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progress((current_step, total_step), desc="Create dataset.")
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transcripts = []
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audios = []
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with tempfile.TemporaryDirectory() as tmpdirname:
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for i, chunk in enumerate(progress.tqdm(chunks, desc="Creating dataset (and clean audio if asked for).")):
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arr = chunk["audio"]
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path = os.path.join(tmpdirname, f"{i}.wav")
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wavfile.write(path, dataset_sampling_rate, arr)
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if use_demucs == "separate-audio":
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print(f"Separating vocals #{i}")
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path = separate_vocal(path)
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audios.append(path)
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transcripts.append(chunk["text"])
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dataset = Dataset.from_dict({"audio": audios, "text": transcripts}).cast_column("audio", Audio())
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current_step += 1
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progress((current_step, total_step), desc="Push dataset.")
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dataset.push_to_hub(dataset_name, token=oauth_token.token if oauth_token else oauth_token)
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return html_embed_str, [[transcript] for transcript in transcripts], text
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min_duration = int(min_duration * sampling_rate)
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new_chunks = []
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while chunks:
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current_chunk = chunks.pop(0)
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begin, end = current_chunk["timestamp"]
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begin, end = int(begin * sampling_rate), int(end * sampling_rate)
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current_dur = end - begin
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text = current_chunk["text"]
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chunk_to_concat = [audio_array[begin:end]]
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while chunks and (text[-1] not in stop_chars or (current_dur < min_duration)):
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ch = chunks.pop(0)
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begin, end = ch["timestamp"]
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begin, end = int(begin * sampling_rate), int(end * sampling_rate)
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import torch
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import gradio as gr
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from transformers import pipeline
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from scipy.io import wavfile
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MODEL_NAME = "openai/whisper-large-v3"
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BATCH_SIZE = 8
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device = 0 if torch.cuda.is_available() else "cpu"
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device=device,
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def transcribe_simple(inputs_path, task):
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if inputs_path is None:
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raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting your request.")
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sampling_rate, inputs = wavfile.read(inputs_path)
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out = pipe(inputs_path, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=True)
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text = out["text"]
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return [[transcript] for transcript in text.split(".") if transcript], text
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with gr.Blocks() as demo:
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with gr.Row():
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with gr.Column():
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audio_input = gr.Audio(source="upload", type="filepath", label="Upload Audio")
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task_input = gr.Dropdown(choices=["transcribe", "translate"], value="transcribe", label="Task")
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submit_button = gr.Button("Transcribe")
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with gr.Column():
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output_text = gr.Dataframe(label="Transcripts")
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output_full_text = gr.Textbox(label="Full Text")
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submit_button.click(
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transcribe_simple,
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inputs=[audio_input, task_input],
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outputs=[output_text, output_full_text],
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demo.launch()
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