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---
license: apache-2.0
pipeline_tag: automatic-speech-recognition
tags:
  - pytorch
  - audio
  - speech
  - automatic-speech-recognition 
  - whisper
  - wav2vec2

model-index:
  - name: whisper_medium_fp16_transformers
    results:
      - task: 
        type: automatic-speech-recognition
        name: Automatic Speech Recognition
        dataset:
          type: common_voice
          name: Common Voice (14.0) (Hindi) (test.tsv -> 2557 samples used)
          metrics:
            - type: wer
              value: 1.7
              name: Test WER
              description: Word Error Rate
            - type: mer
              value: 1.1
              name: Test MER
              description: Match Error Rate
            - type: wil
              value: 3,584
              name: Test WIL
              description: Word Information Lost
            - type: wip
              value: 112
              name: Test WIP
              description: Word Information Preserved
            - type: cer
              value: 1.7
              name: Test CER
              description: Character Error Rate
        
      - task: 
        type: automatic-speech-recognition
        name: Automatic Speech Recognition
        dataset:
          type: common_voice
          name: Common Voice (14.0) (English) (test.tsv -> 2557 samples used)
          metrics:
            - type: wer
              value: -
              name: Test WER
              description: Word Error Rate
            - type: mer
              value: -
              name: Test MER
              description: Match Error Rate
            - type: wil
              value: -
              name: Test WIL
              description: Word Information Lost
            - type: wip
              value: -
              name: Test WIP
              description: Word Information Preserved
            - type: cer
              value: -
              name: Test CER
              description: Character Error Rate

widget:
  - example_title: Librispeech sample 1
    src: https://cdn-media.huggingface.co/speech_samples/sample1.flac
  - example_title: Librispeech sample 2
    src: https://cdn-media.huggingface.co/speech_samples/sample2.flac

language:
  - en
  - zh
  - de
  - es
  - ru
  - ko
  - fr
  - ja
  - pt
  - tr
  - pl
  - ca
  - nl
  - ar
  - sv
  - it
  - id
  - hi
  - fi
  - vi
  - he
  - uk
  - el
  - ms
  - cs
  - ro
  - da
  - hu
  - ta
  - 'no'
  - th
  - ur
  - hr
  - bg
  - lt
  - la
  - mi
  - ml
  - cy
  - sk
  - te
  - fa
  - lv
  - bn
  - sr
  - az
  - sl
  - kn
  - et
  - mk
  - br
  - eu
  - is
  - hy
  - ne
  - mn
  - bs
  - kk
  - sq
  - sw
  - gl
  - mr
  - pa
  - si
  - km
  - sn
  - yo
  - so
  - af
  - oc
  - ka
  - be
  - tg
  - sd
  - gu
  - am
  - yi
  - lo
  - uz
  - fo
  - ht
  - ps
  - tk
  - nn
  - mt
  - sa
  - lb
  - my
  - bo
  - tl
  - mg
  - as
  - tt
  - haw
  - ln
  - ha
  - ba
  - jw
  - su
  
---
## Versions:

- CUDA: 12.1
- cuDNN Version: 8.9.2.26_1.0-1_amd64

* tensorflow Version: 2.12.0
* torch Version: 2.1.0.dev20230606+cu12135
* transformers Version: 4.30.2
* accelerate Version: 0.20.3

## Model Benchmarks:

- RAM: 2.8 GB (Original_Model: 5.5GB)
- VRAM: 1812 MB (Original_Model: 6GB)
- test.wav: 23 s (Multilingual Speech i.e. English+Hindi)
  - **Time in seconds for Processing by each device**

  | Device Name       | float32 (Original)   | float16 | CudaCores | TensorCores |
  | ----------------- | -------------------- | ------- | --------- | ----------- |
  | 3060              | 1.7                  | 1.1     | 3,584     | 112         |
  | 1660 Super        | OOM                  | 3.3     | 1,408     | -           |
  | Collab (Tesla T4) | 2.8                  | 2.2     | 2,560     | 320         |
  | Collab (CPU)      | 35                   | -       | -         | -           |
  | M1 (CPU)          | -                    | -       | -         | -           |
  | M1 (GPU -> 'mps') | -                    | -       | -         | -           |
  

  - **NOTE: TensorCores are efficient in mixed-precision calculations**
  - **CPU -> torch.float16 not supported on CPU (AMD Ryzen 5 3600 or Collab GPU)**
- Punchuation: True

## Model Error Benchmarks:

- **WER: Word Error Rate**
- **MER: Match Error Rate**
- **WIL: Word Information Lost**
- **WIP: Word Information Preserved**
- **CER: Character Error Rate**

### Hindi (test.tsv -> 2557 samples used) [Common Voice 14.0](https://commonvoice.mozilla.org/en/datasets)
  |                   | WER                  | MER     | WIL       | WIP         | CER |
  | ----------------- | -------------------- | ------- | --------- | ----------- | --- |
  | Original_Model    | -                    | -       | -         | -           | -   |
  | This_Model        | -                    | -       | -         | -           | -   |

### English
  |                   | WER                  | MER     | WIL       | WIP         | CER |
  | ----------------- | -------------------- | ------- | --------- | ----------- | --- |
  | Original_Model    | -                    | -       | -         | -           | -   |
  | This_Model        | -                    | -       | -         | -           | -   |

- **'jiwer' library is used for calculations**

## Code:
  - ### [$\textbf{Will be soon Uploaded on Github}$ ](https://github.com/devasheeshG)

## Usage
A file ``__init__.py`` is contained inside this repo which contains all the code to use this model.

Firstly, clone this repo and place all the files inside a folder.
### Make sure you have git-lfs installed (https://git-lfs.com)
```bash
git lfs install
git clone https://huggingface.co/devasheeshG/whisper_medium_fp16_transformers
```
**Please try in jupyter notebook**

```python
# Import the Model
from whisper_medium_fp16_transformers import Model
```

```python
# Initilise the model
model = Model(
            model_name_or_path='whisper_medium_fp16_transformers',
            cuda_visible_device="0", 
            device='cuda',
      )
```

```python
# Load Audio
audio = model.load_audio('whisper_medium_fp16_transformers/test.wav')
```

```python
# Transcribe (First transcription takes time)
model.transcribe(audio)
```