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import os |
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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 huggingface_hub import whoami |
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from transformers import pipeline |
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from transformers.pipelines.audio_utils import ffmpeg_read |
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import time |
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import demucs.api |
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def hello(profile: gr.OAuthProfile | None) -> str: |
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if profile is None: |
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return "I don't know you." |
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return f"Hello {profile.name}" |
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def list_organizations(oauth_token: gr.OAuthToken | None) -> str: |
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if oauth_token is None: |
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return "Please log in to list organizations." |
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org_names = [org["name"] for org in whoami(oauth_token.token)["orgs"]] |
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return f"You belong to {', '.join(org_names)}." |
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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 |
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device = 0 if torch.cuda.is_available() else "cpu" |
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pipe = pipeline( |
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task="automatic-speech-recognition", |
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model=MODEL_NAME, |
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chunk_length_s=30, |
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device=device, |
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) |
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separator = demucs.api.Separator(model = DEMUCS_MODEL_NAME, ) |
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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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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, 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, 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 [[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, |
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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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def naive_postprocess_whisper_chunks(chunks, audio_array, sampling_rate, stop_chars = ".!:;?", min_duration = 5): |
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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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current_dur += end-begin |
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text = "".join([text, ch["text"]]) |
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chunk_to_concat.append(audio_array[begin:end]) |
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new_chunks.append({ |
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"text": text.strip(), |
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"audio": np.concatenate(chunk_to_concat), |
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}) |
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print(f"LENGTH CHUNK #{len(new_chunks)}: {current_dur/sampling_rate}s") |
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return new_chunks |
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css = """ |
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#intro{ |
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max-width: 100%; |
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text-align: center; |
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margin: 0 auto; |
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} |
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""" |
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with gr.Blocks(css=css) as demo: |
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with gr.Row(): |
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gr.LoginButton() |
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gr.LogoutButton() |
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with gr.Tab("YouTube"): |
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gr.Markdown("Create your own TTS dataset using Youtube", elem_id="intro") |
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gr.Markdown( |
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"This demo allows use to create a text-to-speech dataset from an input audio snippet and push it to hub to keep track of it." |
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f"Demo uses the checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) and π€ Transformers to automatically transcribe audio files" |
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" of arbitrary length. It then merge chunks of audio and push it to the hub." |
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) |
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with gr.Row(): |
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with gr.Column(): |
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audio_youtube = gr.Textbox(lines=1, placeholder="Paste the URL to a YouTube video here", label="YouTube URL") |
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task_youtube = gr.Radio(["transcribe", "translate"], label="Task", value="transcribe") |
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cleaning_youtube = gr.Radio(["no-post-processing", "separate-audio"], label="Audio separation and cleaning (takes longer - use it if your samples are not cleaned (background noise and music))", value="separate-audio") |
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textbox_youtube = gr.Textbox(lines=1, placeholder="Place your new dataset name here. 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.", label="Dataset name") |
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with gr.Row(): |
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clear_youtube = gr.ClearButton([audio_youtube, task_youtube, cleaning_youtube, textbox_youtube]) |
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submit_youtube = gr.Button("Submit") |
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with gr.Column(): |
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html_youtube = gr.HTML() |
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dataset_youtube = gr.Dataset(label="Transcribed samples.",components=["text"], headers=["Transcripts"], samples=[["transcripts will appear here"]]) |
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transcript_youtube = gr.Textbox(label="Transcription") |
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with gr.Tab("Microphone or Audio file"): |
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gr.Markdown("Create your own TTS dataset using your own recordings", elem_id="intro") |
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gr.Markdown( |
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"This demo allows use to create a text-to-speech dataset from an input audio snippet and push it to hub to keep track of it." |
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f"Demo uses the checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) and π€ Transformers to automatically transcribe audio files" |
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" of arbitrary length. It then merge chunks of audio and push it to the hub." |
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) |
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with gr.Row(): |
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with gr.Column(): |
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audio_file = gr.Audio(type="filepath") |
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task_file = gr.Radio(["transcribe", "translate"], label="Task", value="transcribe") |
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cleaning_file = gr.Radio(["no-post-processing", "separate-audio"], label="Audio separation and cleaning (takes longer - use it if your samples are not cleaned (background noise and music))", value="separate-audio") |
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textbox_file = gr.Textbox(lines=1, placeholder="Place your new dataset name here. 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.", label="Dataset name") |
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with gr.Row(): |
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clear_file = gr.ClearButton([audio_file, task_file, cleaning_file, textbox_file]) |
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submit_file = gr.Button("Submit") |
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with gr.Column(): |
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dataset_file = gr.Dataset(label="Transcribed samples.", components=["text"], headers=["Transcripts"], samples=[["transcripts will appear here"]]) |
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transcript_file = gr.Textbox(label="Transcription") |
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submit_file.click(transcribe, inputs=[audio_file, task_file, cleaning_file, textbox_file], outputs=[dataset_file, transcript_file]) |
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submit_youtube.click(yt_transcribe, inputs=[audio_youtube, task_youtube, cleaning_youtube, textbox_youtube], outputs=[html_youtube, dataset_youtube, transcript_youtube]) |
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demo.launch(debug=True) |
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