How to use OpenAIs Whisper to transcribe and diarize audio files
Whisper is an State-of-the-Art speech recognition system from OpenAI that has been trained on 680,000 hours of multilingual and multitask supervised data collected from the web. This large and diverse dataset leads to improved robustness to accents, background noise and technical language. In addition, it enables transcription in multiple languages, as well as translation from those languages into English. OpenAI released the models and code to serve as a foundation for building useful applications that leverage speech recognition.
One big downside of Whisper is though, that it can not tell you who is speaking in a conversation. That's a problem when analyzing conversations. This is where diarization comes in. Diarization is the process of identifying who is speaking in a conversation.
In this tutorial you will learn how to identify the speakers, and then match them with the transcriptions of Whisper.
We will use pyannote-audio
to accomplish this. Let's get started!
First, we need to prepare the audio file. We will use the first 20 minutes of Lex Fridmans podcast with Yann download.
To download the video and extract the audio, we will use yt-dlp
package.
!pip install -U yt-dlp
We will also need ffmpeg installed
!wget -O - -q https://github.com/yt-dlp/FFmpeg-Builds/releases/download/latest/ffmpeg-master-latest-linux64-gpl.tar.xz | xz -qdc| tar -x
Now we can do the actual download and audio extraction via the command line.
!yt-dlp -xv --ffmpeg-location ffmpeg-master-latest-linux64-gpl/bin --audio-format wav -o download.wav -- https://youtu.be/SGzMElJ11Cc
Now we have the download.wav
file in our working directory. Let's cut the first 20 minutes of the audio. We can use the pydub package for this with just a few lines of code.
!pip install pydub
from pydub import AudioSegment
t1 = 0 * 1000 # works in milliseconds
t2 = 20 * 60 * 1000
newAudio = AudioSegment.from_wav("download.wav")
a = newAudio[t1:t2]
a.export("audio.wav", format="wav")
audio.wav
is now the first 20 minutes of the audio file.
pyannote.audio
is an open-source toolkit written in Python for speaker diarization. Based on PyTorch
machine learning framework, it provides a set of trainable end-to-end neural building blocks that
can be combined and jointly optimized to build speaker diarization pipelines. pyannote.audio
also
comes with pretrained models and pipelines covering a wide range of domains for voice activity
detection, speaker segmentation, overlapped speech detection, speaker embedding reaching
state-of-the-art performance for most of them.
Installing Pyannote and running it on the video audio to generate the diarizations.
!pip install pyannote.audio
from pyannote.audio import Pipeline
pipeline = Pipeline.from_pretrained('pyannote/speaker-diarization')
DEMO_FILE = {'uri': 'blabal', 'audio': 'audio.wav'}
dz = pipeline(DEMO_FILE)
with open("diarization.txt", "w") as text_file:
text_file.write(str(dz))
Lets print this out to see what it looks like.
print(*list(dz.itertracks(yield_label = True))[:10], sep="\n")
The output:
(<Segment(2.03344, 36.8128)>, 0, 'SPEAKER_00')
(<Segment(38.1122, 51.3759)>, 0, 'SPEAKER_00')
(<Segment(51.8653, 90.2053)>, 1, 'SPEAKER_01')
(<Segment(91.2853, 92.9391)>, 1, 'SPEAKER_01')
(<Segment(94.8628, 116.497)>, 0, 'SPEAKER_00')
(<Segment(116.497, 124.124)>, 1, 'SPEAKER_01')
(<Segment(124.192, 151.597)>, 1, 'SPEAKER_01')
(<Segment(152.018, 179.12)>, 1, 'SPEAKER_01')
(<Segment(180.318, 194.037)>, 1, 'SPEAKER_01')
(<Segment(195.016, 207.385)>, 0, 'SPEAKER_00')
This looks pretty good already, but let's clean the data a little bit:
def millisec(timeStr):
spl = timeStr.split(":")
s = (int)((int(spl[0]) * 60 * 60 + int(spl[1]) * 60 + float(spl[2]) )* 1000)
return s
import re
dz = open('diarization.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) - spacermilli
end = millisec(end) - spacermilli
lex = not re.findall('SPEAKER_01', string=l)
dzList.append([start, end, lex])
print(*dzList[:10], sep='\n')
[33, 34812, True]
[36112, 49375, True]
[49865, 88205, False]
[89285, 90939, False]
[92862, 114496, True]
[114496, 122124, False]
[122191, 149596, False]
[150018, 177119, False]
[178317, 192037, False]
[193015, 205385, True]
Now we have the diarization data in a list. The first two numbers are the start and end time of the speaker segment in milliseconds. The third number is a boolean that tells us if the speaker is Lex or not.
Next, we will attach the audio segements according to the diarization, with a spacer as the delimiter.
from pydub import AudioSegment
import re
sounds = spacer
segments = []
dz = open('diarization.txt').read().splitlines()
for l in dz:
start, end = tuple(re.findall('[0-9]+:[0-9]+:[0-9]+\.[0-9]+', string=l))
start = int(millisec(start)) #milliseconds
end = int(millisec(end)) #milliseconds
segments.append(len(sounds))
sounds = sounds.append(audio[start:end], crossfade=0)
sounds = sounds.append(spacer, crossfade=0)
sounds.export("dz.wav", format="wav") #Exports to a wav file in the current path.
print(segments[:8])
[2000, 38779, 54042, 94382, 98036, 121670, 131297, 160702]
Next, we will use Whisper to transcribe the different segments of the audio file. Important: There is a version conflict with pyannote.audio resulting in an error. Our workaround is to first run Pyannote and then whisper. You can safely ignore the error.
Installing Open AI Whisper.
!pip install git+https://github.com/openai/whisper.git
Running Open AI whisper on the prepared audio file. It writes the transcription into a file. You can adjust the model size to your needs. You can find all models on the model card on Github.
!whisper dz.wav --language en --model base
[00:00.000 --> 00:04.720] The following is a conversation with Yann LeCun,
[00:04.720 --> 00:06.560] his second time on the podcast.
[00:06.560 --> 00:11.160] He is the chief AI scientist at Meta, formerly Facebook,
[00:11.160 --> 00:15.040] professor at NYU, touring award winner,
[00:15.040 --> 00:17.600] one of the seminal figures in the history
[00:17.600 --> 00:20.460] of machine learning and artificial intelligence,
...
In order to work with .vtt files, we need to install the webvtt-py library.
!pip install -U webvtt-py
Lets take a look at the data:
import webvtt
captions = [[(int)(millisec(caption.start)), (int)(millisec(caption.end)), caption.text] for caption in webvtt.read('dz.wav.vtt')]
print(*captions[:8], sep='\n')
[0, 4720, 'The following is a conversation with Yann LeCun,']
[4720, 6560, 'his second time on the podcast.']
[6560, 11160, 'He is the chief AI scientist at Meta, formerly Facebook,']
[11160, 15040, 'professor at NYU, touring award winner,']
[15040, 17600, 'one of the seminal figures in the history']
[17600, 20460, 'of machine learning and artificial intelligence,']
[20460, 23940, 'and someone who is brilliant and opinionated']
[23940, 25400, 'in the best kind of way,']
...
Next, we will match each transcribtion line to some diarizations, and display everything by generating a HTML file. To get the correct timing, we should take care of the parts in original audio that were in no diarization segment. We append a new div for each segment in our audio.
# we need this fore our HTML file (basicly just some styling)
preS = '<!DOCTYPE html>\n<html lang="en">\n <head>\n <meta charset="UTF-8">\n <meta name="viewport" content="width=device-width, initial-scale=1.0">\n <meta http-equiv="X-UA-Compatible" content="ie=edge">\n <title>Lexicap</title>\n <style>\n body {\n font-family: sans-serif;\n font-size: 18px;\n color: #111;\n padding: 0 0 1em 0;\n }\n .l {\n color: #050;\n }\n .s {\n display: inline-block;\n }\n .e {\n display: inline-block;\n }\n .t {\n display: inline-block;\n }\n #player {\n\t\tposition: sticky;\n\t\ttop: 20px;\n\t\tfloat: right;\n\t}\n </style>\n </head>\n <body>\n <h2>Yann LeCun: Dark Matter of Intelligence and Self-Supervised Learning | Lex Fridman Podcast #258</h2>\n <div id="player"></div>\n <script>\n var tag = document.createElement(\'script\');\n tag.src = "https://www.youtube.com/iframe_api";\n var firstScriptTag = document.getElementsByTagName(\'script\')[0];\n firstScriptTag.parentNode.insertBefore(tag, firstScriptTag);\n var player;\n function onYouTubeIframeAPIReady() {\n player = new YT.Player(\'player\', {\n height: \'210\',\n width: \'340\',\n videoId: \'SGzMElJ11Cc\',\n });\n }\n function setCurrentTime(timepoint) {\n player.seekTo(timepoint);\n player.playVideo();\n }\n </script><br>\n'
postS = '\t</body>\n</html>'
from datetime import timedelta
html = list(preS)
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
start = start / 1000.0
startStr = '{0:02d}:{1:02d}:{2:02.2f}'.format((int)(start // 3600),
(int)(start % 3600 // 60),
start % 60)
html.append('\t\t\t<div class="c">\n')
html.append(f'\t\t\t\t<a class="l" href="#{startStr}" id="{startStr}">link</a> |\n')
html.append(f'\t\t\t\t<div class="s"><a href="javascript:void(0);" onclick=setCurrentTime({int(start)})>{startStr}</a></div>\n')
html.append(f'\t\t\t\t<div class="t">{"[Lex]" if dzList[i][2] else "[Yann]"} {c[2]}</div>\n')
html.append('\t\t\t</div>\n\n')
html.append(postS)
s = "".join(html)
with open("lexicap.html", "w") as text_file:
text_file.write(s)
print(s)
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