readme: ---
  license: apache-2.0
  tags:
  - automatic-speech-recognition
  - NbAiLab/NPSC
  - xxx-robust-speech-event
  - no
  - nb-NO
  datasets:
  - NbAiLab/NPSC
  language:
  - nb-NO
  model-index:
  - name: wav2vec2-xls-r-1b-npsc-bokmaal-low-27k
    results:
    - task:
        name: Automatic Speech Recognition 
        type: automatic-speech-recognition
      dataset:
        name: NPSC
        type: NbAiLab/NPSC
        args: 16K_mp3_bokmaal
      metrics:
         - name: Test (Bokmål) WER
           type: wer
           value: 0.06686424124625939
         - name: Test (Bokmål) CER
           type: cer
           value: 0.025697763468940576
  ---
  
  # Norwegian Wav2Vec2 Model - 1B - Bokmål
  This achieves the following results on the test set with a 5-gram KenLM:
  - WER: 0.0668
  - CER: 0.0256
  
  Without using a language model, we are getting these results:
  - WER: ???
  - CER: ???
  
  ## Model description
  This is one of several Wav2Vec-models created during the 🤗 hosted [Robust Speech Event](https://discuss.huggingface.co/t/open-to-the-community-robust-speech-recognition-challenge/13614?s=09). In parallell with the event, the team also converted the [Norwegian Parliamentary Speech Corpus (NPSC)](https://huggingface.co/datasets/NbAiLab/NPSC) to the 🤗 Dataset format and used that as the main source for training.
  
  We do release all code developed during the event so that the Norwegian NLP community can build upon this to develop even better Norwegian ASR models. The finetuning of these models are not very compute demanding. You should after following the instructions here, be able to train your own automatic speech recognition system in less than a day with an average GPU.
  
  ## Team
  The following people contributed to building this model: Rolv-Arild Braaten, Per Egil Kummervold, Andre Kåsen, Javier de la Rosa, Per Erik Solberg, and Freddy Wetjen. 
  
  ## Training procedure
  To reproduce these results, we strongly recommend that you follow the [instructions from HuggingFace](https://github.com/huggingface/transformers/tree/master/examples/research_projects/robust-speech-event#talks) to train a simple Swedish model.
  
  When you have verified that you are able to do this, create a fresh new repo. You can then start by copying the files ```run.sh``` and ```run_speech_recognition_ctc.py``` from our repo. Running this will create all the other necessary files, and should let you reproduce our results. With some tweaks to the hyperparameters, you might even be able to build an even better ASR. Good luck!
  
  ### Language Model
  As you see from the results above, adding even a simple 5-gram language will significantly improve the results.  🤗 has provided another [very nice blog](https://huggingface.co/blog/wav2vec2-with-ngram) about how to add a 5-gram language model to improve the ASR model. You can build this from your own corpus, for instance by extracting some suitable text from the [Norwegian Colossal Corpus](https://huggingface.co/datasets/NbAiLab/NCC). You can also skip some of the steps in the guide, and copy the [5-gram model from this repo](https://huggingface.co/NbAiLab/XLSR-300M-bokmaal/tree/main/language_model).
  
  
  ### Parameters
  The following parameters were used during training:
  ```
  --dataset_name="NbAiLab/NPSC"
  --model_name_or_path="facebook/wav2vec2-xls-r-1b"
  --dataset_config_name="16K_mp3_bokmaal"
  --output_dir="./"
  --overwrite_output_dir
  --num_train_epochs="40"
  --per_device_train_batch_size="12"
  --per_device_eval_batch_size="12" 
  --gradient_accumulation_steps="2" 
  --learning_rate="2e-5" 
  --warmup_steps="2000" 
  --length_column_name="input_length" 
  --evaluation_strategy="steps" 
  --text_column_name="text" 
  --save_steps="500" 
  --eval_steps="500" 
  --logging_steps="100" 
  --layerdrop="0.041" 
  --attention_dropout="0.094" 
  --activation_dropout="0.055" 
  --hidden_dropout="0.047" 
  --save_total_limit="3"
  --freeze_feature_encoder 
  --feat_proj_dropout="0.04" 
  --mask_time_prob="0.082" 
  --mask_time_length="10" 
  --mask_feature_prob="0.25" 
  --mask_feature_length="64" 
  --gradient_checkpointing
  --min_duration_in_seconds="0.5" 
  --max_duration_in_seconds="30.0" 
  	--ctc_zero_infinity=True 
  --use_auth_token 
  --seed="42" 
  --fp16 
  --group_by_length 
  --do_train --do_eval 
  --push_to_hub 
  --preprocessing_num_workers="16"
  ```
  
  Following this settings, the training might take 3-4 days on an average GPU. You should however get a decent model and faster results by tweaking these parameters
  
  | Parameter| Comment | 
  |:-------------|:-----|
  | per_device_train_batch_size       | Adjust this to the maximum of available memory. 16 or 24 might be good settings depending on your system  |
  |gradient_accumulation_steps |Can be adjusted even further up to increase batch size and speed up training without running into memory issues |
  | learning_rate|Can be increased, maybe as high as 1e-4. Speeds up training but might add instability |
  | epochs| Can be decreased significantly. This is a huge dataset and you might get a decent result already after a couple of epochs|
  
  
  