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Dear @sanchit-gandhi,
I was following your tutorial, [Fine-Tune Whisper For Multilingual ASR with 🤗 Transformers](https://huggingface.co/blog/fine-tune-whisper), to fine-tune Whisper with a dataset i…
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Transformers 4.35 only supports speculative decoding for batch size == 1. In order to use speculative decoding for batch size > 1, please make sure to use this branch: https://github.com/huggingface/t…
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## ❓ Questions and Help
### Before asking:
1. search the iss…
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Could you add a native speech to speech / audio-to-audio support with encoder (tokenizer) and decoder (back to audio waves)
I was able to implement a decoder only model, I first used audio codec to…
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使用https://github.com/modelscope/FunASR/blob/main/runtime/http/readme_zh.md
这里的文档自行build了一个http服务端，可以正常启动
但curl -F \"file=@example.wav\" 127.0.0.1:80调用的时候出现错误
basic_string::_M_construct null not v…
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I need to make certain modifications to the code, such as converting the frequency of the WAV file before reading it and then transcribing the speech. However, if I run transcribe_wav.py directly, it …
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In UNIT4 : Pretrained models for audio classification
We’ll load an official [Audio Spectrogram Transformer](https://huggingface.co/docs/transformers/model_doc/audio-spectrogram-transformer) checkpo…
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I try to use the feature extractor on my audiofiles.
My audio files are all 16000Hz and 5 seconds long.
The `waveform.shape[1]` is 80000
```python
input_values = feature_extractor(waveform, sampli…
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### Describe the bug
```import torch
from transformers import pipeline
import bentoml
pipe = pipeline(
"automatic-speech-recognition",
)
bentoml.transformers.save_model(
"automatic-s…
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Hello! I am trying to reproduce the results that were achieved by pretrained models described in [librispeech_example.md](https://github.com/pytorch/fairseq/blob/main/examples/speech_to_text/docs/libr…