from __future__ import unicode_literals import youtube_dl import yt_dlp from pydub import AudioSegment from pyannote.audio import Pipeline import re import whisper import os import ffmpeg import subprocess import gradio as gr import traceback import json pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization", use_auth_token="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") from pydub.effects import speedup import moviepy.editor as mp import datetime import torch import pyannote.audio from pyannote.audio.pipelines.speaker_verification import SpeechBrainPretrainedSpeakerEmbedding #PyannoteAudioPretrainedSpeakerEmbedding from pyannote.audio import Audio from pyannote.core import Segment import wave import contextlib from sklearn.cluster import AgglomerativeClustering import numpy as np import json from datetime import timedelta from transformers import T5ForConditionalGeneration, T5Tokenizer __FILES = set() wispher_models = list(whisper._MODELS.keys()) def correct_grammar(input_text,num_return_sequences=1): torch_device = 'cuda' if torch.cuda.is_available() else 'cpu' tokenizer = T5Tokenizer.from_pretrained('deep-learning-analytics/GrammarCorrector') model = T5ForConditionalGeneration.from_pretrained('deep-learning-analytics/GrammarCorrector').to(torch_device) batch = tokenizer([input_text],truncation=True,padding='max_length',max_length=len(input_text), return_tensors="pt").to(torch_device) results = model.generate(**batch,max_length=len(input_text),num_beams=2, num_return_sequences=num_return_sequences, temperature=1.5) generated_sequences = [] for generated_sequence_idx, generated_sequence in enumerate(results): text = tokenizer.decode(generated_sequence, clean_up_tokenization_spaces=True, skip_special_tokens=True) generated_sequences.append(text) generated_text = "".join(generated_sequences) _generated_text = "" for idx, _sentence in enumerate(generated_text.split('.'), 0): if not idx: _generated_text+=_sentence+'.' elif _sentence[:1]!=' ': _generated_text+=' '+_sentence+'.' elif _sentence[:1]=='': pass else: _generated_text+=_sentence+'.' return _generated_text def CreateFile(filename): __FILES.add(filename) return filename def RemoveFile(filename): if (os.path.isfile(filename)): os.remove(filename) def RemoveAllFiles(): for file in __FILES: if (os.path.isfile(file)): os.remove(file) def Transcribe_V1(NumberOfSpeakers, SpeakerNames="", audio="temp_audio.wav"): SPEAKER_DICT = {} SPEAKERS = [speaker.strip() for speaker in SpeakerNames.split(',') if len(speaker)] def GetSpeaker(sp): speaker = sp if sp not in list(SPEAKER_DICT.keys()): if len(SPEAKERS): t = SPEAKERS.pop(0) SPEAKER_DICT[sp] = t speaker = SPEAKER_DICT[sp] else: speaker = SPEAKER_DICT[sp] return speaker def millisec(timeStr): spl = timeStr.split(":") s = (int)((int(spl[0]) * 60 * 60 + int(spl[1]) * 60 + float(spl[2]) )* 1000) return s def preprocess(audio): t1 = 0 * 1000 t2 = 20 * 60 * 1000 newAudio = AudioSegment.from_wav(audio) a = newAudio[t1:t2] spacermilli = 2000 spacer = AudioSegment.silent(duration=spacermilli) newAudio = spacer.append(a, crossfade=0) newAudio.export(audio, format="wav") return spacermilli, spacer def diarization(audio): as_audio = AudioSegment.from_wav(audio) DEMO_FILE = {'uri': 'blabal', 'audio': audio} if NumberOfSpeakers: dz = pipeline(DEMO_FILE, num_speakers=NumberOfSpeakers) else: dz = pipeline(DEMO_FILE) with open(CreateFile(f"diarization_{audio}.txt"), "w") as text_file: text_file.write(str(dz)) dz = open(CreateFile(f"diarization_{audio}.txt")).read().splitlines() dzList = [] for l in dz: start, end = tuple(re.findall('[0-9]+:[0-9]+:[0-9]+\.[0-9]+', string=l)) start = millisec(start) end = millisec(end) lex = GetSpeaker(re.findall('(SPEAKER_[0-9][0-9])', string=l)[0]) dzList.append([start, end, lex]) sounds = spacer segments = [] dz = open(CreateFile(f"diarization_{audio}.txt")).read().splitlines() for l in dz: start, end = tuple(re.findall('[0-9]+:[0-9]+:[0-9]+\.[0-9]+', string=l)) start = millisec(start) end = millisec(end) segments.append(len(sounds)) sounds = sounds.append(as_audio[start:end], crossfade=0) sounds = sounds.append(spacer, crossfade=0) sounds.export(CreateFile(f"dz_{audio}.wav"), format="wav") return f"dz_{audio}.wav", dzList, segments def transcribe(dz_audio): model = whisper.load_model("medium") result = model.transcribe(dz_audio) # for _ in result['segments']: # print(_['start'], _['end'], _['text']) captions = [[((caption["start"]*1000)), ((caption["end"]*1000)), caption["text"]] for caption in result['segments']] conversation = [] for i in range(len(segments)): idx = 0 for idx in range(len(captions)): if captions[idx][0] >= (segments[i] - spacermilli): break; while (idx < (len(captions))) and ((i == len(segments) - 1) or (captions[idx][1] < segments[i+1])): c = captions[idx] start = dzList[i][0] + (c[0] -segments[i]) if start < 0: start = 0 idx += 1 if not len(conversation): conversation.append([dzList[i][2], c[2]]) elif conversation[-1][0] == dzList[i][2]: conversation[-1][1] += c[2] else: conversation.append([dzList[i][2], c[2]]) #print(f"[{dzList[i][2]}] {c[2]}") return conversation, ("".join([f"{speaker} --> {text}\n" for speaker, text in conversation])) spacermilli, spacer = preprocess(audio) dz_audio, dzList, segments = diarization(audio) conversation, t_text = transcribe(dz_audio) RemoveAllFiles() return (t_text, ({ "data": [{"speaker": speaker, "text": text} for speaker, text in conversation]})) def Transcribe_V2(model, num_speakers, speaker_names, audio="temp_audio.wav"): model = whisper.load_model(model) # embedding_model = SpeechBrainPretrainedSpeakerEmbedding("speechbrain/spkrec-ecapa-voxceleb") embedding_model = SpeechBrainPretrainedSpeakerEmbedding( "speechbrain/spkrec-ecapa-voxceleb", device = torch.device('cuda' if torch.cuda.is_available() else 'cpu') ) SPEAKER_DICT = {} default_speaker_names = ['A','B','C','D','E','F','G','H','I','J','K','L','M','N','O','P','Q','R','S','T','U','V','W','X','Y','Z'] SPEAKERS = [speaker.strip() for speaker in speaker_names.split(',') if len(speaker)] def GetSpeaker(sp): speaker = sp if sp not in list(SPEAKER_DICT.keys()): if len(SPEAKERS): t = SPEAKERS.pop(0) SPEAKER_DICT[sp] = t speaker = SPEAKER_DICT[sp] elif len(default_speaker_names): t = default_speaker_names.pop(0) SPEAKER_DICT[sp] = t speaker = SPEAKER_DICT[sp] else: speaker = SPEAKER_DICT[sp] return speaker # audio = Audio() def diarization(audio): def millisec(timeStr): spl = timeStr.split(":") s = (int)((int(spl[0]) * 60 * 60 + int(spl[1]) * 60 + float(spl[2]) )* 1000) return s as_audio = AudioSegment.from_wav(audio) DEMO_FILE = {'uri': 'blabal', 'audio': audio} hparams = pipeline.parameters(instantiated=True) hparams["segmentation"]["min_duration_off"] -= 0.25 pipeline.instantiate(hparams) if num_speakers: dz = pipeline(DEMO_FILE, num_speakers=num_speakers) else: dz = pipeline(DEMO_FILE) with open(CreateFile(f"diarization_{audio}.txt"), "w") as text_file: text_file.write(str(dz)) dz = open(CreateFile(f"diarization_{audio}.txt")).read().splitlines() print(dz) dzList = [] for l in dz: start, end = tuple(re.findall('[0-9]+:[0-9]+:[0-9]+\.[0-9]+', string=l)) start = millisec(start) end = millisec(end) lex = GetSpeaker(re.findall('(SPEAKER_[0-9][0-9])', string=l)[0]) dzList.append([start, end, lex]) return dzList def get_output(segments): # print(segments) conversation=[] for (i, segment) in enumerate(segments): # print(f"{i}, {segment["speaker"]}, {segments[i - 1]["speaker"]}, {}") if not len(conversation): conversation.append([str(timedelta(seconds=float(segment['start']))),str(timedelta(seconds=float(segment['end']))),GetSpeaker(segment["speaker"]), segment["text"].lstrip()]) elif conversation[-1][2] == GetSpeaker(segment["speaker"]): conversation[-1][3] += segment["text"].lstrip() else: conversation.append([str(timedelta(seconds=float(segment['start']))),str(timedelta(seconds=float(segment['end']))),GetSpeaker(segment["speaker"]), segment["text"].lstrip()]) # if i == 0 or segments[i - 1]["speaker"] != segment["speaker"]: # if i != 0: # conversation.append([GetSpeaker(segment["speaker"]), segment["text"][1:]]) # segment["speaker"] + ' ' + str(time(segment["start"])) + '\n\n' # conversation[-1][1] += segment["text"][1:] # return output for idx in range(len(conversation)): conversation[idx][3] = correct_grammar(conversation[idx][3]) return ("".join([f"[{start}] - {speaker} \n{text}\n" for start, end, speaker, text in conversation])), ({ "data": [{"start": start, "end":end, "speaker": speaker, "text": text} for start, end, speaker, text in conversation]}) def get_duration(path): with contextlib.closing(wave.open(path,'r')) as f: frames = f.getnframes() rate = f.getframerate() return frames / float(rate) def make_embeddings(path, segments, duration): embeddings = np.zeros(shape=(len(segments), 192)) for i, segment in enumerate(segments): embeddings[i] = segment_embedding(path, segment, duration) return np.nan_to_num(embeddings) def segment_embedding(path, segment, duration): 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) return embedding_model(waveform[None]) def add_speaker_labels(segments, embeddings, num_speakers): clustering = AgglomerativeClustering(num_speakers).fit(embeddings) labels = clustering.labels_ for i in range(len(segments)): segments[i]["speaker"] = 'SPEAKER ' + str(labels[i] + 1) def time(secs): return datetime.timedelta(seconds=round(secs)) duration = get_duration(audio) if duration > 4 * 60 * 60: return "Audio duration too long" # print(json.dumps(diarization(audio))) result = model.transcribe(audio) # print(json.dumps(result)) segments = result["segments"] num_speakers = min(max(round(num_speakers), 1), len(segments)) if len(segments) == 1: segments[0]['speaker'] = 'SPEAKER 1' else: embeddings = make_embeddings(audio, segments, duration) add_speaker_labels(segments, embeddings, num_speakers) return get_output(segments) # return output def AudioTranscribe(NumberOfSpeakers=None, SpeakerNames="", audio="", retries=5, model='base'): print(f"{NumberOfSpeakers}, {SpeakerNames}, {retries}") if retries: # subprocess.call(['ffmpeg', '-i', audio,'temp_audio.wav']) try: subprocess.call(['ffmpeg', '-i', audio,'temp_audio.wav']) except Exception as ex: traceback.print_exc() return AudioTranscribe(NumberOfSpeakers, SpeakerNames, audio, retries-1) if not (os.path.isfile("temp_audio.wav")): return AudioTranscribe(NumberOfSpeakers, SpeakerNames, audio, retries-1) return Transcribe_V2(model, NumberOfSpeakers, SpeakerNames) else: raise gr.Error("There is some issue ith Audio Transcriber. Please try again later!") def VideoTranscribe(NumberOfSpeakers=None, SpeakerNames="", video="", retries=5, model='base'): if retries: try: clip = mp.VideoFileClip(video) clip.audio.write_audiofile("temp_audio.wav") # command = f"ffmpeg -i {video} -ab 160k -ac 2 -ar 44100 -vn temp_audio.wav" # subprocess.call(command, shell=True) except Exception as ex: traceback.print_exc() return VideoTranscribe(NumberOfSpeakers, SpeakerNames, video, retries-1) if not (os.path.isfile("temp_audio.wav")): return VideoTranscribe(NumberOfSpeakers, SpeakerNames, video, retries-1) return Transcribe_V2(model, NumberOfSpeakers, SpeakerNames) else: raise gr.Error("There is some issue ith Video Transcriber. Please try again later!") def YoutubeTranscribe(NumberOfSpeakers=None, SpeakerNames="", URL="", retries = 5, model='base'): if retries: if "youtu" not in URL.lower(): raise gr.Error(f"{URL} is not a valid youtube URL.") else: RemoveFile("temp_audio.wav") ydl_opts = { 'format': 'bestaudio/best', 'outtmpl': 'temp_audio.%(ext)s', 'postprocessors': [{ 'key': 'FFmpegExtractAudio', 'preferredcodec': 'wav', }], } try: with yt_dlp.YoutubeDL(ydl_opts) as ydl: ydl.download([URL]) except: return YoutubeTranscribe(NumberOfSpeakers, SpeakerNames, URL, retries-1) stream = ffmpeg.input('temp_audio.m4a') stream = ffmpeg.output(stream, 'temp_audio.wav') RemoveFile("temp_audio.m4a") return Transcribe_V2(model, NumberOfSpeakers, SpeakerNames) else: raise gr.Error(f"Unable to get video from {URL}") with gr.Blocks() as yav_ui: with gr.Row(): with gr.Column(): with gr.Tab("Youtube", id=1): ysz = gr.Dropdown(label="Model Size", choices=wispher_models , value='base') yinput_nos = gr.Number(label="Number of Speakers", placeholder="2") yinput_sn = gr.Textbox(label="Name of the Speakers (ordered by the time they speak and separated by comma)", placeholder="If Speaker 1 is first to speak followed by Speaker 2 then -> Speaker 1, Speaker 2") yinput = gr.Textbox(label="Youtube Link", placeholder="https://www.youtube.com/watch?v=GECcjrYHH8w") ybutton_transcribe = gr.Button("Transcribe", show_progress=True, scroll_to_output=True) with gr.Tab("Video", id=2): vsz = gr.Dropdown(label="Model Size", choices=wispher_models, value='base') vinput_nos = gr.Number(label="Number of Speakers", placeholder="2") vinput_sn = gr.Textbox(label="Name of the Speakers (ordered by the time they speak and separated by comma)", placeholder="If Speaker 1 is first to speak followed by Speaker 2 then -> Speaker 1, Speaker 2") vinput = gr.Video(label="Video") vbutton_transcribe = gr.Button("Transcribe", show_progress=True, scroll_to_output=True) with gr.Tab("Audio", id=3): asz = gr.Dropdown(label="Model Size", choices=wispher_models , value='base') ainput_nos = gr.Number(label="Number of Speakers", placeholder="2") ainput_sn = gr.Textbox(label="Name of the Speakers (ordered by the time they speak and separated by comma)", placeholder="If Speaker 1 is first to speak followed by Speaker 2 then -> Speaker 1, Speaker 2") ainput = gr.Audio(label="Audio", type="filepath") abutton_transcribe = gr.Button("Transcribe", show_progress=True, scroll_to_output=True) with gr.Column(): with gr.Tab("Text"): output_textbox = gr.Textbox(label="Transcribed Text", lines=15) with gr.Tab("JSON"): output_json = gr.JSON(label="Transcribed JSON") ybutton_transcribe.click( fn=YoutubeTranscribe, inputs=[yinput_nos,yinput_sn,yinput, ysz], outputs=[output_textbox,output_json] ) abutton_transcribe.click( fn=AudioTranscribe, inputs=[ainput_nos,ainput_sn,ainput, asz], outputs=[output_textbox,output_json] ) vbutton_transcribe.click( fn=VideoTranscribe, inputs=[vinput_nos,vinput_sn,vinput, vsz], outputs=[output_textbox,output_json] ) yav_ui.launch(debug=True)