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Spanish translation of asr.mdx and add_new_pipeline.mdx (#20569)

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* Fix minor typo in question_answering.mdx

* Fixes minor typo in the english version of tasks/asr.mdx

* Update _toctree.yml

* Translate add_new_pipeline.mdx into Spanish

* Fixes some typos in the English version of add_new_pipeline.mdx

* Translate asr.mdx into Spanish

* Fixes small typos in add_new_pipeline.mdx

* Update docs/source/es/add_new_pipeline.mdx

Suggestion by @osanseviero

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/add_new_pipeline.mdx

Suggestion by @osanseviero: use "biblioteca" instead of "librería."

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/tasks/asr.mdx

Suggestion by @osanseviero.

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/add_new_pipeline.mdx

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/add_new_pipeline.mdx

Suggestion by @osanseviero.

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/add_new_pipeline.mdx

Suggestion by @osanseviero.

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/add_new_pipeline.mdx

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/tasks/asr.mdx

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/tasks/asr.mdx

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update docs/source/es/tasks/asr.mdx

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>

* Update asr.mdx

Co-authored-by: Omar Sanseviero <osanseviero@gmail.com>
This commit is contained in:
Alberto Mario Ceballos-Arroyo
2022-12-12 09:23:23 -05:00
committed by GitHub
parent 8d2fca07e8
commit 8286af6f54
6 changed files with 644 additions and 15 deletions
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docs/source/en/tasks/asr.mdx
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@@ -93,8 +93,8 @@ Take a look at the example again:
There are two fields:
- `audio`: a 1-dimensional `array` of the speech signal that must be called to load and resample the audio file.
- `transcription`: the target text.
- `audio`: a 1-dimensional `array` of the speech signal that must be called to load and resample the audio file.
- `transcription`: the target text.
## Preprocess
@@ -106,7 +106,7 @@ The next step is to load a Wav2Vec2 processor to process the audio signal:
>>> processor = AutoProcessor.from_pretrained("facebook/wav2vec2-base")
```
The MInDS-14 dataset has a sampling rate of 8000khz (you can find this information in its [dataset card](https://huggingface.co/datasets/PolyAI/minds14)), which means you'll need to resample the dataset to 16000kHz to use the pretrained Wav2Vec2 model:
The MInDS-14 dataset has a sampling rate of 8000kHz (you can find this information in its [dataset card](https://huggingface.co/datasets/PolyAI/minds14)), which means you'll need to resample the dataset to 16000kHz to use the pretrained Wav2Vec2 model:
```py
>>> minds = minds.cast_column("audio", Audio(sampling_rate=16_000))