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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 = 'π'
)
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 [π€](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 ππ')
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'
)
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