app.py

import spaces
import torch
import gradio as gr
import yt_dlp as youtube_dl
import tempfile
import os
import locale
import whisper
import datetime
import subprocess
import pyannote.audio
from pyannote.audio.pipelines.speaker_verification import PretrainedSpeakerEmbedding
from pyannote.audio import Audio
from pyannote.core import Segment
import wave
import contextlib
from sklearn.cluster import AgglomerativeClustering
import numpy as np

device = "cuda" if torch.cuda.is_available() else "cpu"
BATCH_SIZE = 4
FILE_LIMIT_MB = 100
COMPUTE_TYPE = "float32" 
YT_LENGTH_LIMIT_S = 600  # limit to 1 hour YouTube files

num_speakers = 2
language = "French"
model_size = 'tiny'
model_name = model_size

def getpreferredencoding(do_setlocale = True):
    return "UTF-8"

locale.getpreferredencoding = getpreferredencoding
embedding_model = PretrainedSpeakerEmbedding(
    "speechbrain/spkrec-ecapa-voxceleb",
    device=torch.device("cpu"))
model = whisper.load_model(model_size).to(device)
audio = Audio()

def segment_embedding(segment,duration,path):
    start = segment["start"]
    # Whisper overshoots the end timestamp in the last segment
    end = min(duration, segment["end"])
    clip = Segment(start, end)
    waveform, sample_rate = audio.crop(path, clip)

    # Convert waveform to single channel
    waveform = waveform.mean(dim=0, keepdim=True)

    return embedding_model(waveform.unsqueeze(0))

def time(secs):
  return datetime.timedelta(seconds=round(secs))

@spaces.GPU
def transcribe(path, task):
    if path is None:
        raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting your request.")
    
    if path[-3:] != 'wav':
        subprocess.call(['ffmpeg', '-i', path, "audio.wav", '-y'])
        path = "audio.wav"
    result = model.transcribe(path,fp16=False)
    segments = result["segments"]
    print(segments)
    with contextlib.closing(wave.open(path,'r')) as f:
        frames = f.getnframes()
        rate = f.getframerate()
        duration = frames / float(rate)
    
    embeddings = np.zeros(shape=(len(segments), 192))
    for i, segment in enumerate(segments):
        embeddings[i] = segment_embedding(segment,duration=duration,path=path)
    embeddings = np.nan_to_num(embeddings)
    clustering = AgglomerativeClustering(num_speakers).fit(embeddings)
    labels = clustering.labels_
    output_text=""
    for i in range(len(segments)):
        segments[i]["speaker"] = '**SPEAKER ' + str(labels[i] + 1) + "**"
    for (i, segment) in enumerate(segments):
        if i == 0 or segments[i - 1]["speaker"] != segment["speaker"]:
            output_text += "  "+segment["speaker"] + ' : '
        output_text += segment["text"][1:] + ' <br>  '
    return output_text



def _return_yt_html_embed(yt_url):
    video_id = yt_url.split("?v=")[-1]
    return f'<center><iframe width="500" height="320" src="https://www.youtube.com/embed/{video_id}"></iframe></center>'

def download_yt_audio(yt_url, filename):
    ydl_opts = {
        "format": "bestaudio/best",
        "outtmpl": filename,
        "postprocessors": [{
            "key": "FFmpegExtractAudio",
            "preferredcodec": "wav",
            "preferredquality": "192",
        }],
    }
    
    with youtube_dl.YoutubeDL(ydl_opts) as ydl:
        ydl.download([yt_url])

@spaces.GPU
def yt_transcribe(yt_url, task):
    html_embed_str = _return_yt_html_embed(yt_url)
    
    with tempfile.TemporaryDirectory() as tmpdirname:
        filepath = os.path.join(tmpdirname, "audio.wav")
        download_yt_audio(yt_url, filepath)
        
        result = model.transcribe(audio, batch_size=BATCH_SIZE)
    
    return html_embed_str, result["text"]

demo = gr.Blocks(theme=gr.themes.Ocean())

mf_transcribe = gr.Interface(
    fn=transcribe,
    inputs=[
        gr.Audio(sources="microphone", type="filepath"),
        gr.Radio(["transcribe", "translate"], label="Task", value="transcribe"),
    ],
    outputs="text",
    title="VerbaLens Demo 1 : Prototype",
    description="Transcribe long-form microphone or audio inputs using WhisperX.",
    allow_flagging="never",
)

file_transcribe = gr.Interface(
    fn=transcribe,
    inputs=[
        gr.Audio(sources="upload", type="filepath", label="Audio file"),
        gr.Radio(["transcribe", "translate"], label="Task", value="transcribe"),
    ],
    outputs=gr.Markdown(label="Sortie Markdown",height=500),
    title="VerbaLens Demo 1 : Prototype",
    description="Transcribe uploaded audio files using WhisperX.",
    allow_flagging="never",
)

yt_transcribe = gr.Interface(
    fn=yt_transcribe,
    inputs=[
        gr.Textbox(lines=1, placeholder="Paste the URL to a YouTube video here", label="YouTube URL"),
        gr.Radio(["transcribe", "translate"], label="Task", value="transcribe"),
    ],
    outputs=["html", "text"],
    title="VerbaLens Demo 1 : Prototyping",
    description="Transcribe YouTube videos using WhisperX.",
    allow_flagging="never",
)

with demo:
    gr.TabbedInterface([mf_transcribe, file_transcribe, yt_transcribe], ["Microphone", "Audio file", "YouTube"])

demo.queue().launch(ssr_mode=False)