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Multi Task Learning Speaker Embeddings

Code for the paper: "Leveraging speaker attribute information using multi task learning for speaker verification and diarization" submitted to ICASSP 2021.

The overall concept of this paper is that training speaker embedding extractors on auxiliary attributes (such as age or nationality) alongside speaker classification can lead to increased performance for verification and diarization. Training the embeddings in this multi-task fashion can improve the descriptiveness of the embedding space.

This is implemented by having multiple task-specific "heads" acting on the embeddings, such as x-vectors. Alongside speaker classification, one might employ age classification/regression, or nationality classification. A general diagram for this can be seen below:

Along with experimental code, this repository covers other data preparation tasks used in the paper, such as webscraping age information for lawyers in SCOTUS, and nationality for Wikipedia celebrities.


  • Requirements
  • SCOTUS Data Preparation
  • VoxCeleb Data Preparation
  • Experiments


This was tested on python 3.6.8

  • General requirements:
    • numpy
    • tqdm
    • sklearn
    • scipy
  • Web-scraping:
    • beautifulsoup4
    • wikipedia
    • wptools
    • Levenshtein
    • python-dateutil
  • Experiments:
    • Kaldi
    • torch
    • kaldi_io

You will also need access to the API, along with a custom google search API key, with instructions of how to obtain/set these up in the SCOTUS Data Preparation section.

SCOTUS Data Preparation

Obtaining case information

This work utilizes the information available via the Oyez project, which provides audio files and transcriptions for US Supreme Court oral arguments.

The Oyez project has an API, an example of which can be seen here:

Thankfully, another GitHub repo exists which has already scraped the information provided by the API for each case and auto-updates every week: This ~3.5GB repo will be used to extract the cases that we want audio and transcripts for.

The following command will clone this repo (specifically my fork, which doesn't update with new cases, to ensure consistency with the number of cases that I have tested my code with):

git clone

The location of this repository which contains the case information will be referred to $SCOTUS_CASE_REPO throughout the rest of the SCOTUS section. Of course, if you want to obtain the most up to date recordings, then you should clone the original repo by walkerdb.

Downloading Audio and Filtering Cases

Now that $SCOTUS_CASE_REPO exists and has been cloned from GitHub, we can now make us of the MTL-Speaker-Embeddings repo:

git clone
cd MTL-Speaker-Embeddings/scotus_data_prep

which puts us inside the scotus_data_prep folder, which is where the data prep scripts for SCOTUS are all present.

Next, we want to run the first step, which downloads mp3 files and filters the cases which we can use. This takes in as input the location of the case JSONs in $SCOTUS_CASE_REPO and then produces a data folder (of which the location is up to you) which we will call $BASE_OUTFOLDER, which will store all the SCOTUS data to be used later on.

python --case-folder $SCOTUS_CASE_REPO/oyez/cases --base-outfolder $BASE_OUTFOLDER

This parses through the cases in $SCOTUS_CASE_REPO and eliminates cases argued before October 2005, as this is when the Supreme Court started recording digitally, instead of using a reel-to-reel taping system. The taped recordings were excluded as there were a number of problems and defects with these recordings as detailed here: This script also eliminates cases where audio can't be found, and also ones where the transcription has a lot of invalid speaker turns (0 or less duration speaker turns).

The outcome should be a file structure in $BASE_OUTFOLDER something like so:

├── speaker_ids.json
├── audio
|   ├── 2007.07-330.mp3
|   └── ...
└── transcripts
    ├── 2007.07-330.json
    └── ...

Sometimes, downloading audio files may fail, but succeed on subsequent attempts. This script is safe to run multiple times, and will try and re-download files that are missing (while skipping ones that already have been processed).

If you have used my fork for $SCOTUS_CASE_REPO then you should end up with 2035 mp3s/jsons in the audio and transcripts folders respectively, although this may not be consistent, depending on the availability of every mp3.

Web scraping

Next, we want to scrape the approximate DoB of each speaker found in the recordings downloaded, which is stored in $BASE_OUTFOLDER/speaker_ids.json.

First of all, we need to set up and obtain a few things in order to fill out inside of scotus_data_prep.

Avvo API

Instructions on how to obtain access to the Avvo API are here:

Once this is set up, you can fill out with your own AVVO_API_ACCESS_TOKEN. (Note: AVVO_CSE_ID is covered in the next section)

Custom Google Search

A custom google search will be needed, which can be set up here:

You will want to click Add to create a new search engine, and under Sites to search, you will want to enter, with the Language specified as English.

After clicking Create, this should succesfully create this custom search. The control panel will take you to where you can find the Search Engine ID, which you can fill into AVVO_CSE_ID in

You will also need a google API key, which can be set up here:

Once this is obtained, fill out GOOGLE_API_KEY in

Running the webscraper

python --base_outfolder $BASE_OUTFOLDER

This will go through the names in speaker_ids.json and try and scrape their dates of birth (DoB) from the following sources:

For the final three sources, the year discovered for graduation/admission to practice law will be subtracted by 25 to obtain an approximate date of birth. The dates of birth, and additional information about the source of each DoB are stored in pickled dictionaries in $BASE_OUTFOLDER:

├── speaker_ids.json
├── dobs.p
├── dobs_info.p
├── audio
|   └── ...
└── transcripts
    └── ...

Like, this script is also safe to re-run, and will try to re-scrape names for which no DoB has been found. If you wish to skip the names which have been attemped, and are present in the dobs.p dictionary, the step2 script can be run with a --skip-attempted flag.

Prepping data for feature extraction

python --base_outfolder $BASE_OUTFOLDER

This prepares verification and diarization data folders that are ready for Kaldi to extract features from. As a result, the recordings are changed into the consistent naming format YEAR-XYZ, and this mapping from orignal names to new names is stored in a JSON file.

Verification/training data is made by splitting up long utterances into non-overlapping 10s segments, with minimum length 4s. Utterances are named in a consistent fashion, and the age (in days) of each speaker at the time of each utterance is calculated, and placed into an utt2age file.

Diarization data is made by splitting into 1.5s segments with 0.75s shift. A ref.rttm file is also created.

This should result in the following file structure (in addition to what was previously shown):

├── ...
├── orec_recid_mapping.json
├── recid_orec_mapping.json
├── veri_data
|   ├── utt2spk
|   ├── spk2utt
|   ├── utt2age
|   ├── wav.scp
|   ├── segments
|   ├── real_utt2spk
|   └── real_spk2utt
└── diar_data
    ├── utt2spk
    ├── spk2utt
    ├── ref.rttm
    ├── wav.scp
    ├── segments
    ├── real_utt2spk
    └── real_spk2utt

The real_{utt2spk|spk2utt} are there as Kaldi feature extraction insists on speaker ids being the prefix to an utterance name - these will be sorted out later on.

Feature extraction

We will use Kaldi to extract the features. This kaldi feature extraction script is found in

You will need to edit at the top of this script and fill in your own $BASE_OUTFOLDER


base_outfolder=/PATH/TO/BASE_OUTFOLDER <--- Edit here


We used the egs/voxceleb/v2 recipe folder to carry out this script (although the only thing specific to this recipe is the conf/mfcc.conf).

To carry this out, we will assume you have Kaldi installed at $KALDI_ROOT

cp $KALDI_ROOT/egs/voxceleb/v2/
cd $KALDI_ROOT/egs/voxceleb/v2

Making train/test splits

After changing back your working directory to scotus_data_prep, we can run the final stage script.

python --base_outfolder $BASE_OUTFOLDER --train-proportion 0.8 --pos-per-spk 12

This trims down the data folders to match what features have been extracted, and splits the recordings into train and test according to --train-proportion. It also makes a verification task for test set utterances, excluding speakers seen in the training set, generating trials of pairs of utterances to compare.

The --pos-per-speaker option determines how many positive trials there are per test set speaker. If too high a value is selected, this may error out as it cannot select enough utterances, so if that occurs, try lowering this value.

This should yield the following (once again in addition to what was produced before):

├── ...
├── veri_data_nosil
|   ├── train
|   |   ├── utt2spk
|   |   ├── spk2utt
|   |   ├── utt2age
|   |   ├── feats.scp
|   |   └── utts
|   ├── test
|   |   ├── veri_pairs
|   |   └── ...
|   └── ...
└── diar_data_nosil
    ├── train
    |   ├── ref.rttm
    |   └── ...
    ├── test
    |   ├── ref.rttm
    |   └── ...
    └── ...

VoxCeleb Data Preparation



Training an embedding extractor is done via

python --cfg configs/example.cfg

This runs according to the config file configs/example.cfg which we will detail and explain below. The following config file trains an xvector architecture embedding extractor with an age classification head with 10 classes and 0.1 weighting of the age loss function.

Config File

# Path to the datasets
train = /PATH_TO_BASE_OUTFOLDER/veri_data_nosil/train
test = /PATH_TO_BASE_OUTFOLDER/veri_data_nosil/test

# Allowed model_type : ['XTDNN', 'ETDNN']
model_type = XTDNN
# Allowed classifier_heads types:
# ['speaker', 'nationality', 'gender', 'age', 'age_regression', 'rec']
classifier_heads = speaker,age

# Allowed classifier_types: 
# ['xvec', 'adm', 'adacos', 'l2softmax', 'xvec_regression', 'arcface', 'sphereface']
classifier_types = xvec,xvec
classifier_lr_mults = [1.0, 1.0]
classifier_loss_weights = [1.0, 0.1]
# Allowed smooth_types:
# ['twoneighbour', 'uniform']
classifier_smooth_types = none,none

input_dim = 30
lr = 0.2
batch_size = 500
max_seq_len = 350
no_cuda = False
seed = 1234
num_iterations = 50000
momentum = 0.5
scheduler_steps = [40000]
scheduler_lambda = 0.5
multi_gpu = False
classifier_lr_mult = 1.
embedding_dim = 256

model_dir = exp/example_exp
# At each checkpoint, the model will be evaluated on the test data, and the model will be saved
checkpoint_interval = 500

num_age_bins = 10

Most of the parameters in this configuration file are fairly self explanatory.

The most important setup is the classifier_heads field, which determines what tasks are being applied to the embedding. This also determines the number of parameters in each field in [Optim], which have to match the number of classifier heads.

Supported Auxiliary Tasks

The currently supported tasks are 'speaker', 'nationality', 'gender', 'age', 'age_regression', 'rec', and each has a requirement for the data that must be present in both train and test folders order for this to be evaluated:

  • age and age_regression
    • Age classification or Age regression
    • Requires: utt2age file
      • 2 column file of format:
      • <utt> <age(int)>
  • gender
    • Gender classification
    • Requires: spk2gender file
      • 2 column file of format:
      • <spk> <gender(str/int)>
  • nationality
    • Nationality classification
    • Requires spk2nat file
      • 2 column file of format:
      • <spk> <nationality(str/int)>
  • rec
    • Recording ID classification
    • This is typically accompanied by a negative loss weight. When a negative loss weight is given, this automatically applies a Gradient Reversal Layer (GRL) to that classification head.
    • Requires utt2rec file (but not in test folder):
      • 2 column file of format:
      • <utt> <rec(str)>

Resuming training

The following command will resume an experiment from the checkpoint at 25000 iterations:

python --cfg configs/example.cfg --resume-checkpoint 25000


Code for the paper: "Leveraging speaker attribute information using multi task learning for speaker verification and diarization" submitted to ICASSP 2021




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