This is a tutorial of training and evaluating a transformer wait-k simultaneous model on MUST-C English-Germen Dataset, from SimulMT to SimulST: Adapting Simultaneous Text Translation to End-to-End Simultaneous Speech Translation.
MuST-C is multilingual speech-to-text translation corpus with 8-language translations on English TED talks.
This section introduces the data preparation for training and evaluation. If you only want to evaluate the model, please jump to Inference & Evaluation
Download and unpack MuST-C data to a path
${MUSTC_ROOT}/en-${TARGET_LANG_ID}
, then preprocess it with
# Additional Python packages for S2T data processing/model training
pip install pandas torchaudio sentencepiece
# Generate TSV manifests, features, vocabulary,
# global cepstral and mean estimation,
# and configuration for each language
cd fairseq
python examples/speech_to_text/prep_mustc_data.py \
--data-root ${MUSTC_ROOT} --task asr \
--vocab-type unigram --vocab-size 10000 \
--cmvn-type global
python examples/speech_to_text/prep_mustc_data.py \
--data-root ${MUSTC_ROOT} --task st \
--vocab-type unigram --vocab-size 10000 \
--cmvn-type global
We need a pretrained offline ASR model. Assuming the save directory of the ASR model is ${ASR_SAVE_DIR}
.
The following command (and the subsequent training commands in this tutorial) assume training on 1 GPU (you can also train on 8 GPUs and remove the --update-freq 8
option).
fairseq-train ${MUSTC_ROOT}/en-de \
--config-yaml config_asr.yaml --train-subset train_asr --valid-subset dev_asr \
--save-dir ${ASR_SAVE_DIR} --num-workers 4 --max-tokens 40000 --max-update 100000 \
--task speech_to_text --criterion label_smoothed_cross_entropy --report-accuracy \
--arch convtransformer_espnet --optimizer adam --lr 0.0005 --lr-scheduler inverse_sqrt \
--warmup-updates 10000 --clip-norm 10.0 --seed 1 --update-freq 8
A pretrained ASR checkpoint can be downloaded here
Fixed pre-decision indicates that the model operate simultaneous policy on the boundaries of fixed chunks.
Here is a example of fixed pre-decision ratio 7 (the simultaneous decision is made every 7 encoder states) and
a wait-3 policy model. Assuming the save directory is ${ST_SAVE_DIR}
fairseq-train ${MUSTC_ROOT}/en-de \
--config-yaml config_st.yaml --train-subset train_st --valid-subset dev_st \
--save-dir ${ST_SAVE_DIR} --num-workers 8 \
--optimizer adam --lr 0.0001 --lr-scheduler inverse_sqrt --clip-norm 10.0 \
--criterion label_smoothed_cross_entropy \
--warmup-updates 4000 --max-update 100000 --max-tokens 40000 --seed 2 \
--load-pretrained-encoder-from ${ASR_SAVE_DIR}/checkpoint_best.pt \
--task speech_to_text \
--arch convtransformer_simul_trans_espnet \
--simul-type waitk_fixed_pre_decision \
--waitk-lagging 3 \
--fixed-pre-decision-ratio 7 \
--update-freq 8
fairseq-train ${MUSTC_ROOT}/en-de \
--config-yaml config_st.yaml --train-subset train_st --valid-subset dev_st \
--save-dir ${ST_SAVE_DIR} --num-workers 8 \
--optimizer adam --lr 0.0001 --lr-scheduler inverse_sqrt --clip-norm 10.0 \
--warmup-updates 4000 --max-update 100000 --max-tokens 40000 --seed 2 \
--load-pretrained-encoder-from ${ASR_SAVE_DIR}/${CHECKPOINT_FILENAME} \
--task speech_to_text \
--criterion latency_augmented_label_smoothed_cross_entropy \
--latency-weight-avg 0.1 \
--arch convtransformer_simul_trans_espnet \
--simul-type infinite_lookback_fixed_pre_decision \
--fixed-pre-decision-ratio 7 \
--update-freq 8
SimulEval is used for evaluation. The following command is for evaluation.
git clone https://github.com/facebookresearch/SimulEval.git
cd SimulEval
pip install -e .
simuleval \
--agent ${FAIRSEQ}/examples/speech_to_text/simultaneous_translation/agents/fairseq_simul_st_agent.py
--source ${SRC_LIST_OF_AUDIO}
--target ${TGT_FILE}
--data-bin ${MUSTC_ROOT}/en-de \
--config config_st.yaml \
--model-path ${ST_SAVE_DIR}/${CHECKPOINT_FILENAME} \
--output ${OUTPUT} \
--scores
The source file ${SRC_LIST_OF_AUDIO}
is a list of paths of audio files. Assuming your audio files stored at /home/user/data
,
it should look like this
/home/user/data/audio-1.wav
/home/user/data/audio-2.wav
Each line of target file ${TGT_FILE}
is the translation for each audio file input.
Translation_1
Translation_2
The evaluation runs on the original MUSTC segmentation.
The following command will generate the wav list and text file for a evaluation set ${SPLIT}
(chose from dev
, tst-COMMON
and tst-HE
) in MUSTC to ${EVAL_DATA}
.
python ${FAIRSEQ}/examples/speech_to_text/seg_mustc_data.py \
--data-root ${MUSTC_ROOT} --lang de \
--split ${SPLIT} --task st \
--output ${EVAL_DATA}
The --data-bin
and --config
should be the same in previous section if you prepare the data from the scratch.
If only for evaluation, a prepared data directory can be found here. It contains
spm_unigram10000_st.model
: a sentencepiece model binary.spm_unigram10000_st.txt
: the dictionary file generated by the sentencepiece model.gcmvn.npz
: the binary for global cepstral mean and variance.config_st.yaml
: the config yaml file. It looks like this. You will need to set the absolute paths forsentencepiece_model
andstats_npz_path
if the data directory is downloaded.
bpe_tokenizer:
bpe: sentencepiece
sentencepiece_model: ABS_PATH_TO_SENTENCEPIECE_MODEL
global_cmvn:
stats_npz_path: ABS_PATH_TO_GCMVN_FILE
input_channels: 1
input_feat_per_channel: 80
sampling_alpha: 1.0
specaugment:
freq_mask_F: 27
freq_mask_N: 1
time_mask_N: 1
time_mask_T: 100
time_mask_p: 1.0
time_wrap_W: 0
transforms:
'*':
- global_cmvn
_train:
- global_cmvn
- specaugment
vocab_filename: spm_unigram10000_st.txt
Notice that once a --data-bin
is set, the --config
is the base name of the config yaml, not the full path.
Set --model-path
to the model checkpoint.
A pretrained checkpoint can be downloaded from here, which is a wait-5 model with a pre-decision of 280 ms.
The result of this model on tst-COMMON
is:
{
"Quality": {
"BLEU": 13.94974229366959
},
"Latency": {
"AL": 1751.8031870037803,
"AL_CA": 2338.5911762796536,
"AP": 0.7931395378788959,
"AP_CA": 0.9405103863210942,
"DAL": 1987.7811616943081,
"DAL_CA": 2425.2751560926167
}
}
If --output ${OUTPUT}
option is used, the detailed log and scores will be stored under the ${OUTPUT}
directory.
The quality is measured by detokenized BLEU. So make sure that the predicted words sent to the server are detokenized.
The latency metrics are
- Average Proportion
- Average Lagging
- Differentiable Average Lagging
Again they will also be evaluated on detokenized text.