import torch import time import gradio as gr import yt_dlp as youtube_dl from transformers import pipeline from transformers.pipelines.audio_utils import ffmpeg_read import tempfile import os BATCH_SIZE = 8 FILE_LIMIT_MB = 1000 YT_LENGTH_LIMIT_S = 3600 # limit to 1 hour YouTube files device = 0 if torch.cuda.is_available() else "cpu" def transcribe(model, audio, task): if audio is None: raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting your request.") pipe = pipeline( task="automatic-speech-recognition", model=model, chunk_length_s=30, device=device, ) text = pipe(audio, batch_size=BATCH_SIZE, generate_kwargs={"language": "latvian", "task": task}, return_timestamps=True)["text"] return text def _return_yt_html_embed(yt_url): video_id = yt_url.split("?v=")[-1] HTML_str = ( f'
' "
" ) return HTML_str def download_yt_audio(yt_url, filename): info_loader = youtube_dl.YoutubeDL() try: info = info_loader.extract_info(yt_url, download=False) except youtube_dl.utils.DownloadError as err: raise gr.Error(str(err)) file_length = info["duration_string"] file_h_m_s = file_length.split(":") file_h_m_s = [int(sub_length) for sub_length in file_h_m_s] if len(file_h_m_s) == 1: file_h_m_s.insert(0, 0) if len(file_h_m_s) == 2: file_h_m_s.insert(0, 0) file_length_s = file_h_m_s[0] * 3600 + file_h_m_s[1] * 60 + file_h_m_s[2] if file_length_s > YT_LENGTH_LIMIT_S: yt_length_limit_hms = time.strftime("%HH:%MM:%SS", time.gmtime(YT_LENGTH_LIMIT_S)) file_length_hms = time.strftime("%HH:%MM:%SS", time.gmtime(file_length_s)) raise gr.Error(f"Maximum YouTube length is {yt_length_limit_hms}, got {file_length_hms} YouTube video.") ydl_opts = {"outtmpl": filename, "format": "worstvideo[ext=mp4]+bestaudio[ext=m4a]/best[ext=mp4]/best"} with youtube_dl.YoutubeDL(ydl_opts) as ydl: try: ydl.download([yt_url]) except youtube_dl.utils.ExtractorError as err: raise gr.Error(str(err)) def yt_transcribe(model, yt_url, task): html_embed_str = _return_yt_html_embed(yt_url) with tempfile.TemporaryDirectory() as tmpdirname: filepath = os.path.join(tmpdirname, "video.mp4") download_yt_audio(yt_url, filepath) with open(filepath, "rb") as f: inputs = f.read() pipe = pipeline( task="automatic-speech-recognition", model=model, chunk_length_s=30, device=device, ) inputs = ffmpeg_read(inputs, pipe.feature_extractor.sampling_rate) inputs = {"array": inputs, "sampling_rate": pipe.feature_extractor.sampling_rate} text = pipe(inputs, batch_size=BATCH_SIZE, generate_kwargs={"language": "latvian", "task": task}, return_timestamps=True)["text"] return html_embed_str, text demo = gr.Blocks() transcribe = gr.Interface( fn=transcribe, inputs=[ gr.Dropdown([ ("tiny", "RaivisDejus/whisper-tiny-lv"), ("small", "RaivisDejus/whisper-small-lv"), ("large", "AiLab-IMCS-UL/whisper-large-v3-lv-late-cv17") ], label="Model", value="RaivisDejus/whisper-small-lv"), gr.Audio(sources=["upload", "microphone"],type="filepath", label="Audio"), gr.Radio([("Transcribe", "transcribe"), ("Translate to English", "translate",)], label="Task", value="transcribe"), ], outputs=gr.Textbox(label="Transcription", lines=15), title="Latvian speech recognition: Transcribe Audio", description=(""" Test Latvian speech recognition (STT) models. Three models are available: * [tiny](https://huggingface.co/RaivisDejus/whisper-tiny-lv) - Fastest, requiring least RAM, but also least accurate * [small](https://huggingface.co/RaivisDejus/whisper-small-lv) - Reasonably fast, reasonably accurate, requiring reasonable amounts of RAM * [large](https://huggingface.co/AiLab-IMCS-UL/whisper-large-v3-lv-late-cv17) - Most accurate, developed by scientists from [ailab.lv](https://ailab.lv/). Requires most RAM and for best performance should be run on a GPU. To improve speech recognition quality, more data is needed, donate your voice on [Balsu talka](https://balsutalka.lv/) """ ), allow_flagging="never", ) yt_transcribe = gr.Interface( fn=yt_transcribe, inputs=[ gr.Dropdown([ ("tiny", "RaivisDejus/whisper-tiny-lv"), ("small", "RaivisDejus/whisper-small-lv"), ("large", "AiLab-IMCS-UL/whisper-large-v3-lv-late-cv17") ], label="Model", value="RaivisDejus/whisper-small-lv"), gr.Textbox(lines=1, placeholder="Paste the URL to a YouTube video here", label="YouTube URL"), gr.Radio([("Transcribe", "transcribe"), ("Translate to English", "translate",)], label="Task", value="transcribe") ], outputs=["html", "text"], title="Latvian speech recognition: Transcribe YouTube", description=(""" Test Latvian speech recognition (STT) models. Three models are available: * [tiny](https://huggingface.co/RaivisDejus/whisper-tiny-lv) - Fastest, requiring least RAM, but also least accurate * [small](https://huggingface.co/RaivisDejus/whisper-small-lv) - Reasonably fast, reasonably accurate, requiring reasonable amounts of RAM * [large](https://huggingface.co/AiLab-IMCS-UL/whisper-large-v3-lv-late-cv17) - Most accurate, developed by scientists from [ailab.lv](https://ailab.lv/). Requires most RAM and for best performance should be run on a GPU. To improve speech recognition quality, more data is needed, donate your voice on [Balsu talka](https://balsutalka.lv/) """ ), allow_flagging="never", ) with demo: gr.TabbedInterface([transcribe, yt_transcribe], ["Microphone / Audio file", "YouTube"]) demo.queue(max_size=10) demo.launch()