This codebase puts together tools and experiments to analyze self-supervised speech representations. These analysis techniques can be used to replicate the findings presented in Layer-Wise analysis of a self-supervised speech representation model and Comparative layer-wise analysis of self-supervised speech models.
The codebase currently supports data loading and feature extraction for the following publicly available pre-trained models:
The following canonical correlation analysis (CCA) and mutual information (MI) experiments are currently supported by the codebase (please refer our papers for more details on individual experiments):
- cca-intra
- cca-mel
- cca-phone
- cca-word
- cca-agwe
- cca-glove
- mi-phone
- mi-word
Follow the next steps in order to generate property content trends for pretrained models from raw wavforms and alignments. Each step is accompanied by a short explanation.
You can also find an abridged and collated version of these steps at examples/recipe.sh.
. examples/recipe.sh $path_to_librispeech_data $ckpt_dir $pckg_dir
Perform steps 0, 1, and 2a below, and read the accompanying README.md before running the script.
Install the relevant model packages to $pckg_dir and download the pre-trained models in $ckpt_dir. The default inference in most model packages does not directly return layerwise outputs. This can that can be easily fixed with minor edits to their model files. The edited files are added to the modellib_addons/ directory.
Replace the original files in the model packages with these edited versions before proceeding.
git clone https://github.com/ankitapasad/layerwise-analysis.git
pip install -r requirements.txt
# for wordsim
git submodule update
git submodule init
Note that since this repo is intended to be used for one or more of the models listed above, please make sure to install these libraries in the same environment that has all the necessary installations and dependencies for the corresponding model(s).
Currently, all the experiments use Librispeech, so before proceeding further make sure you have the dataset downloaded. Download and extract all the files into the $path_to_librispeech_data directory, such that this directory has a folder for each dataset split.
Follow the next two steps to prepare data files.
. scripts/prepare_alignment_files.sh librispeech $path_to_librispeech_data $alignment_data_dir
This will download and reformat the phone and word alignment files for Librispeech and save the alignments as dictionary files to $alignment_data_dir. These .json files map each phone/word type to a list of tuples (utt_id, path_to_wav, start_time, end_time). This might take 30 minutes. Note: Uncomment step #3 commands if you intend to apply MI tools, this will add a few more minutes of processing time.
data_sample=1
dataset_split=dev-clean
span=frame
. scripts/create_librispeech_data_samples.sh $data_sample $path_to_librispeech_data $alignment_data_dir $dataset_split $span
This will randomly sample audio utterances and phone and word segment instances from Librispeech. The list of sampled utterance ids will be saved to the data_samples/librispeech directory.
The data_sample variable is an identifier for the data sample set. You can have more than one data_sample sets, generated by passing a different identifier, and repeat all the analysis experiments on each data_sample set to check for robustness (using mean and standard deviation for instance).
Understanding this is not necessary for CCA experiments since those are performed on the smaller dev splits. The following is relevant when sampling and processing utterances from train splits.
For phone and word segments, the sampled set can be split further into subsets identified with numbers 0, 1, 2, .... These ids are referred to as subset_id in the next steps. This is done so that each subset can be processed in parallel for feature extraction. The extracted representations for each subset are concatenated into one single representation matrix before evaluating analysis scores.
Currently, each subset is under 10000 seconds. This threshold can be changed by passing the dur_threshold argument to the create_data_samples.py token-level line.
Example: Extract representations from the pre-trained wav2vec2.0 model for dev-clean split
model_name=wav2vec_base
dataset_split=dev-clean
data_sample=1
subset_id=0
Set save_dir_pth as the directory where the extracted representations will be saved. The script will generate sub-folders within $save_dir_pth for each different model and each different type of representation. Pass the same $save_dir_pth variable to all the subsequent scripts.
Frame-level representations from the 7 convolutional layers
rep_type=local
span=frame
. scripts/extract_rep.sh $model_name $ckpt_dir $data_sample $rep_type $span $subset_id $dataset_split $save_dir_pth $pckg_dir
Similarly for extracting representations from transformer layers, run the above script with following changes to the arguments:
- Frame-level representations from the 12 transformer layers:
rep_type=contextualized; span=frame - Mean-pooled phone-level representations from the 12 transformer layers:
rep_type=contextualized; span=phone - Mean-pooled word-level representations from the 12 transformer layers:
rep_type=contextualized; span=word
The extracted features will be saved to the $save_dir_pth/$model_name/librispeech_$dataset_split_sample1 directory
Note: This part is only necessary for wordsim experiments.
- Generate samples of words from the train set. Note that there is a
num_instancesvariable inside the script, that is the value for number of instances' representations averaged for each word embedding.
. scripts/create_word_samples.sh $model_name $path_to_librispeech_data $alignment_data_dir $save_dir_pth $num_instances
This will sample the word instances and divide all words into subsets, such that each subset is smaller than 10000 seconds (for processing speech and parallelization).
- Extract representation, you'll run the following for each
subset_id. The argumentsubfnamedenotes the sample directory name that is set here and for which you wish to extract the embeddings.
. scripts/extract_static_word_embed.sh extract $model_name $ckpt_dir $subfname $save_dir_pth $subset_id
- Once all the subsets are processed, combine the representations to form an embedding map.
. scripts/extract_static_word_embed.sh combine $model_name $ckpt_dir $subfname $save_dir_pth
- Download and store GloVe and AGWE embeddings maps as dictionary files.
. scripts/save_embeddings.sh $save_dir_pth $alignment_data_dir agwe
. scripts/save_embeddings.sh $save_dir_pth $alignment_data_dir glove
. scripts/save_embeddings.sh $save_dir_pth $alignment_data_dir one-hot
Example: Canonical correlation analysis between the extracted representations and mel filterbank features for representations extracted at a frame-level
exp_name=cca_mel
span=frame
. scripts/get_cca_scores.sh $model_name $data_sample $exp_name $span $save_dir_pth
In order to evaluate a single layer at a time, pass $layer_num to the same script
exp_name=cca_mel
span=frame
layer_num=T4 # process transformer layer 4
. scripts/get_cca_scores.sh $model_name $data_sample $exp_name $span $save_dir_pth $layer_num
The following CCA experiments are possible:
| $exp_name | $span |
|---|---|
| cca_mel | frame |
| cca_intra | frame |
| cca_phone | phone |
| cca_word | word |
| cca_agwe | word |
| cca_glove | word |
Example: Mutual information between $span labels and the extracted phone segments for representations extracted from layer 1. The iter_num denotes the iteration number. To account for the randomness introduced by k-means clustering, we run this experiment for each sample set multiple times.
iter_num=0
layer_num=1
span=phone # or word
. scripts/get_mi_scores.sh $span $layer_num $iter_num $model_name $data_sample $save_dir_pth
. scripts/get_wordsim_scores.sh $model_name $subfname $save_dir_pth
The results from all the above measures will be saved at logs/librispeech_${model_name}/
-
Thanks to Ju-Chieh Chou (@jjery2243542) for help with testing the codebase.
-
Thanks to Lugosch et al. for making Librispeech alignments publicly available.
Loren Lugosch, Mirco Ravanelli, Patrick Ignoto, Vikrant Singh Tomar, and Yoshua Bengio, "Speech Model Pre-training for End-to-End Spoken Language Understanding", Interspeech 2019
Michael McAuliffe, Michaela Socolof, Sarah Mihuc, Michael Wagner, and Morgan Sonderegger, "Montreal Forced Aligner: trainable text-speech alignment using Kaldi", Interspeech 2017
- Thanks to Shane Settle (@shane-settle) for providing acoustically grounded word embeddings trained on Librispeech data
Shane Settle, Kartik Audhkhasi, Karen Livescu, and Michael Picheny, “Acoustically grounded word embeddings for improved acoustics-to-word speech recognition”, in ICASSP, 2019
- Thanks to Raghu et al. for making their CCA implementation publicly available. We use their library with some modifications and corrections.
Maithra Raghu, Justin Gilmer, Jason Yosinski, and Jascha Sohl-Dickstein, "SVCCA: Singular Vector Canonical Correlation Analysis for Deep Learning Dynamics and Interpretability", NeurIPS 2017
Ari S. Morcos, Maithra Raghu, and Samy Bengio, "Insights on Representational Similarity in Deep Neural Networks with Canonical Correlation", NeurIPS 2018