SHAS: Approaching optimal Segmentation for End-to-End Speech Translation

Overview

SHAS: Approaching optimal Segmentation for End-to-End Speech Translation

In this repo you can find the code of the Supervised Hybrid Audio Segmentation (SHAS) method for End-to-End Speech Translation, proposed in Tsiamas et al. (2022). You can use our method with pre-trained models to segment a collection of audio files or train and fine-tune our method on your own segmented data. We provide instructions to replicate our results from the paper on MuST-C en-de and mTEDx es-en, fr-en, it-en, pt-en. You can also find easy-to-use implementations of other segmentation methods, like fixed-length, VAD, and the hybrid methods of Potapczyk and Przybysz (2020), Gállego et al. (2021), and Gaido et al. (2021).

Follow the instructions here to segment a collection of audio files, or the instruction here to replicate the results of the paper.

Abstract

Speech translation models are unable to directly process long audios, like TED talks, which have to be split into shorter segments. Speech translation datasets provide manual segmentations of the audios, which are not available in real-world scenarios, and existing segmentation methods usually significantly reduce translation quality at inference time. To bridge the gap between the manual segmentation of training and the automatic one at inference, we propose Supervised Hybrid Audio Segmentation (SHAS), a method that can effectively learn the optimal segmentation from any manually segmented speech corpus. First, we train a classifier to identify the included frames in a segmentation, using speech representations from a pre-trained wav2vec 2.0. The optimal splitting points are then found by a probabilistic Divide-and-Conquer algorithm that progressively splits at the frame of lowest probability until all segments are below a pre-specified length. Experiments on MuST-C and mTEDx show that the translation of the segments produced by our method approaches the quality of the manual segmentation on 5 languages pairs. Namely, SHAS retains 95-98% of the manual segmentation's BLEU score, compared to the 87-93% of the best existing methods. Our method is additionally generalizable to different domains and achieves high zero-shot performance in unseen languages.

Results

Citation

If you find SHAS or the contents of this repo useful for your research, please consider citing:

@misc{tsiamas2022shas,
      title={SHAS: Approaching optimal Segmentation for End-to-End Speech Translation}, 
      author={Ioannis Tsiamas and Gerard I. Gállego and José A. R. Fonollosa and Marta R. Costa-jussà},
      year={2022},
      eprint={2202.04774},
      archivePrefix={arXiv},
      primaryClass={cs.SD}
}

Usage

Clone this repository to $SHAS_ROOT:

git clone https://github.com/mt-upc/SHAS.git ${SHAS_ROOT}    

Create a conda environment using the environment.yml file and activate it:

conda env create -f ${SHAS_ROOT}/environment.yml && \
conda activate shas

Segmentation with SHAS

Download one of the available pre-trained segmentation frame classifiers required for the SHAS method:

English Spanish French Italian Portuguese Multilingual

Make sure that the audio files you want to segment are in .wav format, mono, and sampled at 16kHz. You can convert them with:

path_to_wavs=...                       # path to the audio files that will be segmented
ls ${path_to_wavs}/*.* | parallel -j 4 ffmpeg -i {} -ac 1 -ar 16000 -hide_banner -loglevel error {.}.wav

Segment a collection of audio files with the SHAS method. This includes inference with the classifier and application of a probabilistic Divide-and-Conquer (pDAC) algorithm:

python ${SHAS_ROOT}/src/supervised_hybrid/segment.py \
  -wavs $path_to_wavs \                       # path to the audio files that will be segmented
  -ckpt $path_to_checkpoint \                 # path to the checkpoint of a trained segmentation frame classifier
  -yaml $path_to_custom_segmentation_yaml \   # where to save the custom segmentation yaml file
  -max $max_segment_length                    # the core parameter of pDAC (in seconds, empirically values between 14-18 work well)

Segmentation with other methods

Length-based (fixed-length) segmentation:

python ${SHAS_ROOT}/src/segmentation_methods/length_based.py \
  -wavs $path_to_wavs \
  -yaml $path_to_custom_segmentation_yaml \
  -n $segment_length    # (in seconds)

Pause-based segmentation with webrtc VAD:

python ${SHAS_ROOT}/src/segmentation_methods/pause_based.py \
  -wavs $path_to_wavs \
  -yaml $path_to_custom_segmentation_yaml \
  -l $frame_length \        # 10, 20 or 30
  -a $aggressiveness_mode   # 1, 2 or 3

Hybrid segmentation with either wav2vec 2.0 or VAD as pause predictor, and either the DAC or Streaming algorithms:

python ${SHAS_ROOT}/src/segmentation_methods/hybrid.py \
  -wavs $path_to_wavs \
  -yaml $path_to_custom_segmentation_yaml \
  -pause $pause_predictor \         # wav2vec or vad
  -alg $algorithm \                 # dac or strm
  -max $max_segment_length \        # (in seconds)
  -min $min_segment_length          # (in seconds) only active for the strm alg

More extensive usage

Follow these steps to replicate the results of the paper. Download the MuST-C and mTEDx data, prepare them for the Segmentation Frame Classifier training, train the classifier, generate a segmentation of a test set, translate the segments with Joint Speech-to-Text models from fairseq, do hypothesis-reference alignment, and compute BLEU scores.

Setting up the environment

Set the environment variables:

export SHAS_ROOT=...                # the path to this repo
export MUSTC_ROOT=...               # the path to save MuST-C v2.0
export MTEDX_ROOT=...               # the path to save mTEDx
export SEGM_DATASETS_ROOT=...       # the path to save the outputs of data_prep/prepare_dataset_for_segmentation
export ST_MODELS_PATH=...           # the path to the pre-trained joint-s2t models from fairseq
export RESULTS_ROOT=...             # the path to the results
export FAIRSEQ_ROOT=...             # the path to our fairseq fork
export MWERSEGMENTER_ROOT=...       # the path to the mwerSegmenter tool

Clone this repository to $SHAS_ROOT:

git clone https://github.com/mt-upc/SHAS.git ${SHAS_ROOT}    

If you want to evaluate a custom segmentation, the translated segments have to be aligned with the reference translations of the manual segmentation. We are using the mwerSegmenter for the alignment. Create a secondary python2 environment for using mwerSegmenter:

conda create -n p2-shas python=2.7

Download mwerSegmenter at ${MWERSEGMENTER_ROOT} and follow the instructions in ${MWERSEGMENTER_ROOT}/README to install it:

mkdir -p $MWERSEGMENTER_ROOT
wget https://www-i6.informatik.rwth-aachen.de/web/Software/mwerSegmenter.tar.gz
tar -zxvf mwerSegmenter.tar.gz -C ${MWERSEGMENTER_ROOT}
rm -r mwerSegmenter.tar.gz

Create a conda environment using the environment.yml file and activate it:

conda env create -f ${SHAS_ROOT}/environment.yml && \
conda activate shas

We are using fairseq for Speech Translation. Install our fork of fairseq:

git clone -b audio-segment-2022 https://github.com/mt-upc/fairseq-internal.git ${FAIRSEQ_ROOT}
pip install --editable ${FAIRSEQ_ROOT}

Note: You can also use the latest public fairseq version, but BLEU scores will have minor differences with the ones reported in the paper.

Data

Download MuST-C v2 en-de to $MUSTC_ROOT:
The dataset is available here. Press the bottom ”click here to download the corpus”, and select version V2.

Download the mTEDx x-en and ASR data to $MTEDX_ROOT:

mkdir -p ${MTEDX_ROOT}
mkdir -p ${MTEDX_ROOT}/log_dir
for lang_pair in {es-en,fr-en,pt-en,it-en,es,fr,pt,it}; do
  wget https://www.openslr.org/resources/100/mtedx_${lang_pair}.tgz -o ${MTEDX_ROOT}/log_dir/${lang_pair} -c -b -O - | tar -xz -C ${MTEDX_ROOT}
done

Convert to mono and downsample at 16kHz:

ls ${MTEDX_ROOT}/*/data/{train,valid,test}/wav/*.flac | parallel -j 12 ffmpeg -i {} -ac 1 -ar 16000 -hide_banner -loglevel error {.}.wav

Prepare the datasets for segmentation

We create two tsv files (talks, segments) for each triplet of dataset-lang_pair-split. These will be used during training to create training examples by random segmentation and during evaluation to create fixed segmentation for inference.

# MuST-C en-de
mkdir -p ${SEGM_DATASETS_ROOT}/MUSTC/en-de
for split in {train,dev,tst-COMMON}; do
  python ${SHAS_ROOT}/src/data_prep/prepare_dataset_for_segmentation.py \
    -y ${MUSTC_ROOT}/en-de/data/${split}/txt/${split}.yaml \
    -w ${MUSTC_ROOT}/en-de/data/${split}/wav \
    -o ${SEGM_DATASETS_ROOT}/MUSTC/en-de
done
# mTEDx
for lang_pair in {es-en,fr-en,pt-en,it-en,es-es,fr-fr,pt-pt,it-it}; do
  mkdir -p ${SEGM_DATASETS_ROOT}/mTEDx/${lang_pair}
  for split in {train,valid,test}; do
    python ${SHAS_ROOT}/src/data_prep/prepare_dataset_for_segmentation.py \
      -y ${MTEDX_ROOT}/${lang_pair}/data/${split}/txt/${split}.yaml \
      -w ${MTEDX_ROOT}/${lang_pair}/data/${split}/wav \
      -o ${SEGM_DATASETS_ROOT}/mTEDx/${lang_pair}
  done
done

Download pre-trained Speech Translation models

For translating the custom segmentations we are using the Joint Speech-to-Text models from fairseq. Download the bilingual model trained on MuST-C en-de and the multlingual model trained on mTEDx:

# joint-s2t-mustc-en-de
en_de_model_path=${ST_MODELS_PATH}/joint-s2t-mustc-en-de
mkdir -p $en_de_model_path
for file in {checkpoint_ave_10.pt,config.yaml,src_dict.txt,dict.txt,spm.model}; docheck
  wget https://dl.fbaipublicfiles.com/joint_speech_text_4_s2t/must_c/en_de/${file} -O $en_de_model_path/${file}
done
# joint-s2t-multilingual
mult_model_path=${ST_MODELS_PATH}/joint-s2t-multilingual
mkdir -p $mult_model_path
for file in {checkpoint17.pt,config.yaml,tgt_dict.txt,dict.txt,spm.model}; do
  wget https://dl.fbaipublicfiles.com/joint_speech_text_4_s2t/iwslt/iwslt_data/${file} -O $mult_model_path/${file}
done

To generate translation with the ST models, we have to modify the path of the spm.model in the task configs and remove some hardcoded paths from the cfg arguments of the checkpoints.

sed -i "s+/path/spm.model+${en_de_model_path}/spm.model+" ${en_de_model_path}/config.yaml
python ${SHAS_ROOT}/src/data_prep/fix_joint_s2t_cfg.py -c ${en_de_model_path}/checkpoint_ave_10.pt
sed -i "s+/path/spm.model+${mult_model_path}/spm.model+" ${mult_model_path}/config.yaml
python ${SHAS_ROOT}/src/data_prep/fix_joint_s2t_cfg.py -c ${mult_model_path}/checkpoint17.pt

Train a Segmentation Frame Classifier (SFC) model

For a monolingual model (for example on English speech):

experiment_name=en_sfc_model
python ${SHAS_ROOT}/src/supervised_hybrid/train.py \
    --datasets ${SEGM_DATASETS_ROOT}/MUSTC/en-de \
    --results_path ${RESULTS_ROOT}/supervised_hybrid \
    --model_name facebook/wav2vec2-xls-r-300m \
    --experiment_name $experiment_name \
    --train_sets tst-COMMON \
    --eval_sets dev \
    --batch_size 14 \
    --learning_rate 2.5e-4 \
    --update_freq 20 \
    --max_epochs 8 \
    --classifier_n_transformer_layers 1 \
    --wav2vec_keep_layers 15

For a multilingual model trained on (English, Spanish, French, Italian, Portuguese) speech:

experiment_name=mult_sfc_model
python ${SHAS_ROOT}/src/supervised_hybrid/train.py \
    --datasets ${SEGM_DATASETS_ROOT}/MUSTC/en-de,${SEGM_DATASETS_ROOT}/mTEDx/es-es,${SEGM_DATASETS_ROOT}/mTEDx/fr-fr,${SEGM_DATASETS_ROOT}/mTEDx/it-it,${SEGM_DATASETS_ROOT}/mTEDx/pt-pt \
    --results_path ${RESULTS_ROOT}/supervised_hybrid \
    --model_name facebook/wav2vec2-xls-r-300m \
    --experiment_name $experiment_name \
    --train_sets train,train,train,train,train \
    --eval_sets dev,valid,valid,valid,valid \
    --batch_size 14 \
    --learning_rate 2.5e-4 \
    --update_freq 20 \
    --max_epochs 8 \
    --classifier_n_transformer_layers 2 \
    --wav2vec_keep_layers 15

(The above commands assume 1 active GPU, adjust accordingly the update_freq if you are using more)

Create a segmentation the SHAS method

Segment a collection of audio files, by doing inference with a trained Segmentation Frame Classifier and applying a probabilistic Divide-and-Conquer (pDAC) algorithm:

python ${SHAS_ROOT}/src/supervised_hybrid/segment.py \
  -wavs $path_to_wavs \                       # path to the audio files that will be segmented
  -ckpt $path_to_checkpoint \                 # path to the checkpoint of a trained segmentation frame classifier
  -yaml $path_to_custom_segmentation_yaml \   # where to save the custom segmentation yaml file
  -max $max_segment_length                    # the core parameter of pDAC (in seconds, empirically values between 14-18 work well)

Translate the segments and evaluate the translations

The eval_custom_segmentation.sh performs the following tasks:

  • (1): translates the segments using an ST model;
  • (2): does hypothesis-reference alignment with mwerSegmenter;
  • (3): computes scores with sacreBLEU;
bash ${SHAS_ROOT}/src/eval_scripts/eval_custom_segmentation.sh \
  $path_to_wavs \                               # path to the audio files that will be segmented
  $path_to_custom_segmentation_yaml \           # path to the custom segmentation yaml from segment.py
  $path_to_original_segmentation_yaml \         # path to the original segmentation yaml
  $path_to_original_segment_transcriptions \    # path to the text file of the original segment transcriptions
  $path_to_original_segment_translations \      # path to the text file of the original segment translations
  $src_lang \                                   # the source language id (for example: en)
  $tgt_lang \                                   # the target language id (for example: de)
  $path_to_st_model_ckpt                        # path to the checkpoint of the joint-s2t model (use the joint-s2t-mustc-en-de for en source and joint-s2t-multilingual for the rest)
Owner
Machine Translation @ UPC
Hi, we are the UPC Machine Translation Group! 👋
Machine Translation @ UPC
Sapiens is a human antibody language model based on BERT.

Sapiens: Human antibody language model ____ _ / ___| __ _ _ __ (_) ___ _ __ ___ \___ \ / _` | '_ \| |/ _ \ '

Merck Sharp & Dohme Corp. a subsidiary of Merck & Co., Inc. 13 Nov 20, 2022
Simple program that translates the name of files into English

Simple program that translates the name of files into English. Useful for when editing/inspecting programs that were developed in a foreign language.

0 Dec 22, 2021
Natural Language Processing for Adverse Drug Reaction (ADR) Detection

Natural Language Processing for Adverse Drug Reaction (ADR) Detection This repo contains code from a project to identify ADRs in discharge summaries a

Medicines Optimisation Service - Austin Health 21 Aug 05, 2022
Source code for the paper "TearingNet: Point Cloud Autoencoder to Learn Topology-Friendly Representations"

TearingNet: Point Cloud Autoencoder to Learn Topology-Friendly Representations Created by Jiahao Pang, Duanshun Li, and Dong Tian from InterDigital In

InterDigital 21 Dec 29, 2022
✨Rubrix is a production-ready Python framework for exploring, annotating, and managing data in NLP projects.

✨A Python framework to explore, label, and monitor data for NLP projects

Recognai 1.5k Jan 02, 2023
CDLA: A Chinese document layout analysis (CDLA) dataset

CDLA: A Chinese document layout analysis (CDLA) dataset 介绍 CDLA是一个中文文档版面分析数据集,面向中文文献类(论文)场景。包含以下10个label: 正文 标题 图片 图片标题 表格 表格标题 页眉 页脚 注释 公式 Text Title

buptlihang 84 Dec 28, 2022
auto_code_complete is a auto word-completetion program which allows you to customize it on your need

auto_code_complete v1.3 purpose and usage auto_code_complete is a auto word-completetion program which allows you to customize it on your needs. the m

RUO 2 Feb 22, 2022
Code for EMNLP 2021 main conference paper "Text AutoAugment: Learning Compositional Augmentation Policy for Text Classification"

Code for EMNLP 2021 main conference paper "Text AutoAugment: Learning Compositional Augmentation Policy for Text Classification"

LancoPKU 105 Jan 03, 2023
AI Assistant for Building Reliable, High-performing and Fair Multilingual NLP Systems

AI Assistant for Building Reliable, High-performing and Fair Multilingual NLP Systems

Microsoft 37 Nov 29, 2022
The RWKV Language Model

RWKV-LM We propose the RWKV language model, with alternating time-mix and channel-mix layers: The R, K, V are generated by linear transforms of input,

PENG Bo 877 Jan 05, 2023
Mapping a variable-length sentence to a fixed-length vector using BERT model

Are you looking for X-as-service? Try the Cloud-Native Neural Search Framework for Any Kind of Data bert-as-service Using BERT model as a sentence enc

Han Xiao 11.1k Jan 01, 2023
Comprehensive-E2E-TTS - PyTorch Implementation

A Non-Autoregressive End-to-End Text-to-Speech (text-to-wav), supporting a family of SOTA unsupervised duration modelings. This project grows with the research community, aiming to achieve the ultima

Keon Lee 114 Nov 13, 2022
Named Entity Recognition API used by TEI Publisher

TEI Publisher Named Entity Recognition API This repository contains the API used by TEI Publisher's web-annotation editor to detect entities in the in

e-editiones.org 14 Nov 15, 2022
COVID-19 Chatbot with Rasa 2.0: open source conversational AI

COVID-19 chatbot implementation with Rasa open source 2.0, conversational AI framework.

Aazim Parwaz 1 Dec 23, 2022
Chatbot with Pytorch, Python & Nextjs

Installation Instructions Make sure that you have Python 3, gcc, venv, and pip installed. Clone the repository $ git clone https://github.com/sahr

Rohit Sah 0 Dec 11, 2022
Code associated with the "Data Augmentation using Pre-trained Transformer Models" paper

Data Augmentation using Pre-trained Transformer Models Code associated with the Data Augmentation using Pre-trained Transformer Models paper Code cont

44 Dec 31, 2022
Unofficial Implementation of Zero-Shot Text-to-Speech for Text-Based Insertion in Audio Narration

Zero-Shot Text-to-Speech for Text-Based Insertion in Audio Narration This repo contains only model Implementation of Zero-Shot Text-to-Speech for Text

Rishikesh (ऋषिकेश) 33 Sep 22, 2022
PyTorch implementation of convolutional neural networks-based text-to-speech synthesis models

Deepvoice3_pytorch PyTorch implementation of convolutional networks-based text-to-speech synthesis models: arXiv:1710.07654: Deep Voice 3: Scaling Tex

Ryuichi Yamamoto 1.8k Dec 30, 2022
News-Articles-and-Essays - NLP (Topic Modeling and Clustering)

NLP T5 Project proposal Topic Modeling and Clustering of News-Articles-and-Essays Students: Nasser Alshehri Abdullah Bushnag Abdulrhman Alqurashi OVER

2 Jan 18, 2022
Revisiting Pre-trained Models for Chinese Natural Language Processing (Findings of EMNLP 2020)

This repository contains the resources in our paper "Revisiting Pre-trained Models for Chinese Natural Language Processing", which will be published i

Yiming Cui 463 Dec 30, 2022