Speech Transcription with Whisper v1.1

Date completed	March 19, 2024
Release where first appeared	OpenWillis v2.1
Researcher / Developer	Vijay Yadav, Anzar Abbas, Georgios Efstathiadis

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1 – Use

import openwillis as ow

transcript_json, transcript_text = ow.speech_transcription_whisper(filepath = '', model = '', compute_type = '', batch_size = '', hf_token = '', language = '', min_speakers = '', max_speakers = '', context = '')

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2 – Methods

This function transcribes speech into text using WhisperX, an open source library built atop Whisper, a speech transcription model developed by OpenAI. The function allows for offline speech transcription where computational resources are available. It is best suited for researchers transcribing speech on institutional or cloud-based machines, ideally with GPU access.

The function allows for multiple models, tiny, small, medium, large and large-v2.

• The tiny model, with 39M parameters, is suitable for personal machines and can handle relatively short audio files.
• The small model, with 244M parameters, offers a balance of performance and file length.
• The medium model, with 769M parameters, is a step up for processing longer audio files.
• On the other hand, the large and large-v2 models, with 1550M parameters, demand GPU resources for efficient processing of large recordings.

Naturally, transcription accuracy will be higher with the larger models. There is also the option to change the default compute type and batch size for the transcription. Refer to the WhisperX instruction set for further information on the computational resources required.

The user will also need a Hugging Face token to access the underlying models. To acquire a token, they will need to create an account on Hugging Face, accept user conditions for the segmentation, voice activity detection, and diarization models, and go to account settings to access their token.

Whisper can support several languages/dialects, namely Afrikaans, Arabic, Armenian, Azerbaijani, Belarusian, Bosnian, Bulgarian, Catalan, Chinese, Croatian, Czech, Danish, Dutch, English, Estonian, Finnish, French, Galician, German, Greek, Hebrew, Hindi, Hungarian, Icelandic, Indonesian, Italian, Japanese, Kannada, Kazakh, Korean, Latvian, Lithuanian, Macedonian, Malay, Marathi, Maori, Nepali, Norwegian, Persian, Polish, Portuguese, Romanian, Russian, Serbian, Slovak, Slovenian, Spanish, Swahili, Swedish, Tagalog, Tamil, Thai, Turkish, Ukrainian, Urdu, Vietnamese, and Welsh. The language codes for each can be found below.

If no language is specified, Whisper will automatically detect the language and process it as such. However, language classification accuracy may vary depending on the language and its similarity to other languages. We recommend always specifying the language, if known.

2.2.1 – Transcription only

In case of a straightforward transcription, the function simply passes the source audio through Whisper.

In transcript_json, a word-by-word and phrase-by-phrase transcript is saved as a JSON.

In transcript_text, a string of the entire transcript, including punctuation, is saved.

2.2.2 – Transcription + Speaker labeling

When the source audio contains multiple speakers, the function will automatically label each speaker in the output JSON as speaker0, speaker1, speaker2, and so on at both the word and the phrase level.

If the number of speakers in the source audio is known to the user, they may specify max_speakers = n to limit the number of speakers that will be labeled. Please note that this is only an upper limit. If the model detects fewer than the expected number of speakers, it will output the JSON as such.

As is to be expected, speaker labeling is not 100% accurate.

For phrases that are labeled as one speaker but contain words labeled for more than one speaker, each word’s speaker label will be reassigned to the phrase-level identification if it does not match the phrase’s speaker label.

2.2.3 – Transcription + Speaker labeling + Speaker identification

If the source audio is a recording of a structured clinical interview, the user may specify the scale being administered using the context parameter and the function will automatically identify which of the speakers is the clinician and which of the speakers is the participant.

Then, at the word and phrase level, the speaker label is changed to either clinician or participant.

When context is specified, the user must set speaker_labels = 'True' and max_speakers = 2.

Currently supported contexts are the following clinical scales:

• MADRS, conducted in accordance with SIGMA, for which context = 'madrs'.
• PANSS, conducted in accordance with the SCI-PANSS, for which context = 'panss'.
• HAM-A, conducted in accordance with Hamilton Anxiety Rating Scale (SIGH-A), for which context = 'ham-a'.
• CAPS past week conducted in accordance with DSM-5 (CAPS-5) Past Week Version, for which context = 'caps-past-week'.
• CAPS past month conducted in accordance with DSM-5 (CAPS-5) Past Month Version, for which context = 'caps-past-month'.
• CAPS DSM IV conducted in accordance with Clinician-Administered PTSD Scale For DSM-IV, for which context = 'caps-iv'.
• MINI conducted in accordance with Version 7.0.2 for DSM-5, for which context = 'mini'.
• CAINS conducted in accordance with CAINS (v1.0), for which context = 'cains'.

Speaker identification is done by comparing the transcribed text with expected rater prompts from the clinical scale. This comparison is done using a pre-trained multilingual sentence transformer model, which works by mapping different sentences to the pre-trained embedding space, based on their underlying meaning. The embeddings are compared using cosine similarity. The closer the embeddings are in similarity, the more similar they are in meaning. The speaker whose audio more closely matches the expected rater prompts is labeled as the clinician, while the other speaker is labeled as the participant. This comparison is done in a manner that is agnostic of the language the speech is in.

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3 – Inputs

3.1 – filepath

Type	String
Description	A local path to the file that the user wants to process. Most audio and video file types are supported.

3.2 – model

Type	String; default is tiny
Description	Name of model the user wants to utilize, with the options being tiny, small, medium, large and large-v2.

3.3 – compute_type

Type	String; default is int16
Description	The type of computation to be used during transcription. Only if running on CPU, if running on GPU compute_type will be float16.

3.4 – batch_size

Type	Integer; default is 16
Description	The number of audio samples to be processed together in one batch during transcription. Adjusting this can help manage memory usage and processing speed.

3.5 – hf_token

Type	String
Description	The user’s Hugging Face token as described in the function methods

3.6 – language

Type	String; optional – if left empty; resorts to auto-detection
Description	Language of the source audio file

english	en
chinese	zh
german	de
spanish	es
russian	ru
korean	ko
french	fr
japanese	ja
portuguese	pt
turkish	tr
polish	pl
catalan	ca
dutch	nl
arabic	ar
swedish	sv
italian	it
indonesian	id
hindi	hi
finnish	fi
vietnamese	vi
hebrew	he
ukrainian	uk
greek	el
malay	ms
czech	cs
romanian	ro
danish	da
hungarian	hu
tamil	ta
norwegian	no
thai	th
urdu	ur
croatian	hr
bulgarian	bg
lithuanian	lt
latin	la
maori	mi
malayalam	ml
welsh	cy
slovak	sk
telugu	te
persian	fa
latvian	lv
bengali	bn
serbian	sr
azerbaijani	az
slovenian	sl
kannada	kn
estonian	et
macedonian	mk
breton	br
basque	eu
icelandic	is
armenian	hy
nepali	ne
mongolian	mn
bosnian	bs
kazakh	kk
albanian	sq
swahili	sw
galician	gl
marathi	mr
punjabi	pa
sinhala	si
khmer	km
shona	sn
yoruba	yo
somali	so
afrikaans	af
occitan	oc
georgian	ka
belarusian	be
tajik	tg
sindhi	sd
gujarati	gu
amharic	am
yiddish	yi
lao	lo
uzbek	uz
faroese	fo
haitian creole	ht
pashto	ps
turkmen	tk
nynorsk	nn
maltese	mt
sanskrit	sa
luxembourgish	lb
myanmar	my
tibetan	bo
tagalog	tl
malagasy	mg
assamese	as
tatar	tt
hawaiian	haw
lingala	ln
hausa	ha
bashkir	ba
javanese	jw
sundanese	su

3.7 – max_speakers

Type	Integer; optional
Description	This parameter sets the maximum number of speakers to be identified and labeled in audio or video transcriptions.

3.8 – context

Type	String; optional
Description	In case the source audio is the recording of a known clinical scale, specification of the clinical scale. If scale is provided, num_speakers is assumed to be 2.

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4 – Outputs

4.1 – transcript_json

Type	JSON
Description	This is a word-wise and phrase-wise transcript saved as a JSON

4.2 – transcript_text

Type	String
Description	The transcription, compiled into a string

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5 – Dependencies

Below are dependencies specific to calculation of this measure.

Dependency	License	Justification
WhisperX	BSD-4	Library for transcribing and diarizing speakers in recordings.
Whisper	MIT	Library for transcribing speech
Pyannote	MIT	Library for speaker diarization
SentenceTransformer	Apache 2.0	A pre-trained BERT-based sentence transformer model to compute the similarity between the speech.