import streamlit as st import streamlit_ext as ste import openai from pydub import AudioSegment from pytube import YouTube import pytube import io from pyannote.audio import Pipeline from pyannote.audio.pipelines.utils.hook import ProgressHook from pyannote.database.util import load_rttm from pyannote.core import Annotation, Segment, notebook import time import json import torch import urllib.parse as urlparse from urllib.parse import urlencode import os import matplotlib matplotlib.use('Agg') from matplotlib import pyplot as plt st.set_page_config( page_title="Speech-to-chat", page_icon = '🌊' ) def create_audio_stream(audio): return io.BytesIO(audio.export(format="wav").read()) def add_query_parameter(link, params): url_parts = list(urlparse.urlparse(link)) query = dict(urlparse.parse_qsl(url_parts[4])) query.update(params) url_parts[4] = urlencode(query) return urlparse.urlunparse(url_parts) def youtube_video_id(value): """ Examples: - http://youtu.be/SA2iWivDJiE - http://www.youtube.com/watch?v=_oPAwA_Udwc&feature=feedu - http://www.youtube.com/embed/SA2iWivDJiE - http://www.youtube.com/v/SA2iWivDJiE?version=3&hl=en_US """ query = urlparse.urlparse(value) if query.hostname == 'youtu.be': return query.path[1:] if query.hostname in ('www.youtube.com', 'youtube.com'): if query.path == '/watch': p = urlparse.parse_qs(query.query) return p['v'][0] if query.path[:7] == '/embed/': return query.path.split('/')[2] if query.path[:3] == '/v/': return query.path.split('/')[2] # fail? return None def load_rttm_file(rttm_path): return load_rttm(rttm_path)['stream'] def load_audio(uploaded_audio): return AudioSegment.from_file(uploaded_audio) # Set your OpenAI, Hugging Face API keys openai.api_key = os.getenv('openai') hf_api_key = os.getenv('hf') st.title("Speech Diarization and Speech-to-Text with PyAnnote and Whisper") reddit_thread = 'https://www.reddit.com/r/dataisbeautiful/comments/17413bq/oc_speech_diarization_app_that_transcribes_audio' with st.expander('About', expanded=True): st.markdown(f''' Given an audio file this app will - [x] 1. Identify and diarize the speakers using `pyannote` [HuggingFace Speaker Diarization api](https://huggingface.co/pyannote/speaker-diarization-3.0) - [x] 2. Transcribe the audio and attribute to speakers using [OpenAi Whisper API](https://platform.openai.com/docs/guides/speech-to-text/quickstart) - [ ] 3. Set up an LLM chat with the transcript loaded into its knowledge database, so that a user can "talk" to the transcript of the audio file (WIP) This version will only process up to first 6 minutes of an audio file due to limited resources of Streamlit.io apps. A local version with access to a GPU can process 1 hour of audio in 1 to 5 minutes. If you would like to use this app at scale reach out directly by creating an issue on github [🤖](https://github.com/KobaKhit/speech-to-text-app/issues)! Rule of thumb, for this Streamlit.io hosted app it takes half the duration of the audio to complete processing, ex. g. 6 minute youtube video will take 3 minutes to diarize. [github repo](https://github.com/KobaKhit/speech-to-text-app) ''') option = st.radio("Select source:", ["Upload an audio file", "Use YouTube link","See Example"], index=2) # Upload audio file if option == "Upload an audio file": uploaded_audio = st.file_uploader("Upload an audio file (MP3 or WAV)", type=["mp3", "wav","mp4"]) with st.expander('Optional Parameters'): rttm = st.file_uploader("Upload .rttm if you already have one", type=["rttm"]) transcript_file = st.file_uploader("Upload transcipt json", type=["json"]) youtube_link = st.text_input('Youtube link of the audio sample') if uploaded_audio is not None: st.audio(uploaded_audio, format="audio/wav", start_time=0) audio_name = uploaded_audio.name audio = load_audio(uploaded_audio) # sample_rate = st.number_input("Enter the sample rate of the audio", min_value=8000, max_value=48000) # audio = audio.set_frame_rate(sample_rate) # use youtube link elif option == "Use YouTube link": youtube_link_raw = st.text_input("Enter the YouTube video URL:") youtube_link = f'https://youtu.be/{youtube_video_id(youtube_link_raw)}' with st.expander('Optional Parameters'): rttm = st.file_uploader("Upload .rttm if you already have one", type=["rttm"]) transcript_file = st.file_uploader("Upload transcipt json", type=["json"]) if youtube_link_raw: st.write(f"Fetching audio from YouTube: {youtube_link}") try: yt = YouTube(youtube_link) audio_stream = yt.streams.filter(only_audio=True).first() audio_name = audio_stream.default_filename st.write(f"Downloaded {audio_name}") except pytube.exceptions.AgeRestrictedError: st.stop('Age restricted videos cannot be processed.') try: os.remove('sample.mp4') except OSError: pass audio_file = audio_stream.download(filename='sample.mp4') time.sleep(2) audio = load_audio('sample.mp4') st.audio(create_audio_stream(audio), format="audio/mp4", start_time=0) # sample_rate = st.number_input("Enter the sample rate of the audio", min_value=8000, max_value=48000) # audio = audio.set_frame_rate(sample_rate) # except Exception as e: # st.write(f"Error: {str(e)}") elif option == 'See Example': youtube_link = 'https://www.youtube.com/watch?v=TamrOZX9bu8' audio_name = 'Stephen A. Smith has JOKES with Shannon Sharpe' st.write(f'Loaded audio file from {youtube_link} - Stephen A. Smith has JOKES with Shannon Sharpe 👏😂') if os.path.isfile('example/steve a smith jokes.mp4'): audio = load_audio('example/steve a smith jokes.mp4') else: yt = YouTube(youtube_link) audio_stream = yt.streams.filter(only_audio=True).first() audio_file = audio_stream.download(filename='sample.mp4') time.sleep(2) audio = load_audio('sample.mp4') if os.path.isfile("example/steve a smith jokes.rttm"): rttm = "example/steve a smith jokes.rttm" if os.path.isfile('example/steve a smith jokes.json'): transcript_file = 'example/steve a smith jokes.json' st.audio(create_audio_stream(audio), format="audio/mp4", start_time=0) # Diarize if "audio" in locals(): st.write('Performing Diarization...') # create stream duration = audio.duration_seconds if duration > 360: st.info('Only processing the first 6 minutes of the audio due to Streamlit.io resource limits.') audio = audio[:360*1000] duration = audio.duration_seconds # Perform diarization with PyAnnote # "pyannote/speaker-diarization-3.0", # use_auth_token=hf_api_key pipeline = Pipeline.from_pretrained( "pyannote/speaker-diarization-3.0", use_auth_token=hf_api_key) if torch.cuda.device_count() > 0: # use gpu if available pipeline.to(torch.device('cuda')) # run the pipeline on an audio file if 'rttm' in locals() and rttm != None: st.write(f'Loading {rttm}') diarization = load_rttm_file(rttm) else: # with ProgressHook() as hook: audio_ = create_audio_stream(audio) # diarization = pipeline(audio_, hook=hook) diarization = pipeline(audio_) # dump the diarization output to disk using RTTM format with open(f'{audio_name.split(".")[0]}.rttm', "w") as f: diarization.write_rttm(f) # Display the diarization results st.write("Diarization Results:") annotation = Annotation() sp_chunks = [] progress_text = f"Processing 1/{len(sp_chunks)}..." my_bar = st.progress(0, text=progress_text) counter = 0 n_tracks = len([a for a in diarization.itertracks(yield_label=True)]) for turn, _, speaker in diarization.itertracks(yield_label=True): annotation[turn] = speaker progress_text = f"Processing {counter}/{len(sp_chunks)}..." my_bar.progress((counter+1)/n_tracks, text=progress_text) counter +=1 temp = {'speaker': speaker, 'start': turn.start, 'end': turn.end, 'duration': turn.end-turn.start, 'audio': audio[turn.start*1000:turn.end*1000]} if 'transcript_file' in locals() and transcript_file == None: temp['audio_stream'] = create_audio_stream(audio[turn.start*1000:turn.end*1000]) sp_chunks.append(temp) # plot notebook.crop = Segment(-1, duration + 1) figure, ax = plt.subplots(figsize=(10,3)) notebook.plot_annotation(annotation, ax=ax, time=True, legend=True) figure.tight_layout() # save to file st.pyplot(figure) st.write('Speakers and Audio Samples') with st.expander('Samples', expanded=True): for speaker in set(s['speaker'] for s in sp_chunks): temp = max(filter(lambda d: d['speaker'] == speaker, sp_chunks), key=lambda x: x['duration']) speak_time = sum(c['duration'] for c in filter(lambda d: d['speaker'] == speaker, sp_chunks)) rate = 100*min((speak_time, duration))/duration speaker_summary = f"{temp['speaker']} ({round(rate)}% of video duration): start={temp['start']:.1f}s stop={temp['end']:.1f}s" if youtube_link != None: speaker_summary += f" {add_query_parameter(youtube_link, {'t':str(int(temp['start']))})}" st.write(speaker_summary) st.audio(create_audio_stream(temp['audio'])) # st.write("Transcription with Whisper ASR:") st.divider() # # Perform transcription with Whisper ASR st.write('Transcribing using Whisper API (150 requests limit)...') container = st.container() limit = 150 progress_text = f"Processing 1/{len(sp_chunks[:limit])}..." my_bar = st.progress(0, text=progress_text) with st.expander('Transcript', expanded=True): if 'transcript_file' in locals() and transcript_file != None: with open(transcript_file,'r') as f: sp_chunks_loaded = json.load(f) for i,s in enumerate(sp_chunks_loaded): if s['transcript'] != None: transcript_summary = f"{s['speaker']} start={float(s['start']):.1f}s end={float(s['end']):.1f}s: {s['transcript']}" if youtube_link != None: transcript_summary += f" {add_query_parameter(youtube_link, {'t':str(int(s['start']))})}" st.write(transcript_summary) progress_text = f"Processing {i+1}/{len(sp_chunks_loaded)}..." my_bar.progress((i+1)/len(sp_chunks_loaded), text=progress_text) transcript_json = sp_chunks_loaded transcript_path = f'example-transcript.json' else: sp_chunks_updated = [] for i,s in enumerate(sp_chunks[:limit]): if s['duration'] > 0.1: audio_path = s['audio'].export('temp.wav',format='wav') try: transcript = openai.Audio.transcribe("whisper-1", audio_path)['text'] except Exception: transcript = '' pass if transcript !='' and transcript != None: s['transcript'] = transcript transcript_summary = f"{s['speaker']} start={s['start']:.1f}s end={s['end']:.1f}s : {s['transcript']}" if youtube_link != None: transcript_summary += f" {add_query_parameter(youtube_link, {'t':str(int(s['start']))})}" sp_chunks_updated.append({'speaker':s['speaker'], 'start':s['start'], 'end':s['end'], 'duration': s['duration'],'transcript': transcript}) progress_text = f"Processing {i+1}/{len(sp_chunks[:limit])}..." my_bar.progress((i+1)/len(sp_chunks[:limit]), text=progress_text) st.write(transcript_summary) transcript_json = [dict((k, d[k]) for k in ['speaker','start','end','duration','transcript'] if k in d) for d in sp_chunks_updated] transcript_path = f'{audio_name.split(".")[0]}-transcript.json' with open(transcript_path,'w') as f: json.dump(transcript_json, f) with container: st.info(f'Completed transcribing') @st.cache_data def convert_df(string): # IMPORTANT: Cache the conversion to prevent computation on every rerun return string.encode('utf-8') transcript_json_download = convert_df(json.dumps(transcript_json)) c1_b,c2_b = st.columns((1,2)) with c1_b: ste.download_button( "Download transcript as json", transcript_json_download, transcript_path, ) header = ','.join(transcript_json[0].keys()) + '\n' for s in transcript_json: header += ','.join([str(e) if ',' not in str(e) else '"' + str(e) + '"' for e in s.values()]) + '\n' transcript_csv_download = convert_df(header) with c2_b: ste.download_button( "Download transcript as csv", transcript_csv_download, f'{audio_name.split(".")[0]}-transcript.csv' )