Model reference · open weights
wav2vec2-large-xlsr-cantonese is an open-weight audio or speech model from ctl. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.
Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.
What it is
| Released by | ctl |
|---|---|
| Type | Audio & music |
| Task | Speech→text |
| Runs with | transformers |
| Released | 2022-03-02 |
| Popularity | 2k downloads / month |
| Licence | Open weights |
About
Fine-tuned facebook/wav2vec2-large-xlsr-53 on Cantonese using the Common Voice. When using this model, make sure that your speech input is sampled at 16kHz.
The model can be used directly (without a language model) as follows:
import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
test_dataset = load_dataset("common_voice", "zh-HK", split="test[:2%]")
processor = Wav2Vec2Processor.from_pretrained("ctl/wav2vec2-large-xlsr-cantonese")
model = Wav2Vec2ForCTC.from_pretrained("ctl/wav2vec2-large-xlsr-cantonese")
resampler = torchaudio.transforms.Resample(48_000, 16_000)
# Preprocessing the datasets.
# We need to read the aduio files as arrays
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = torchaudio.load(batch["path"])
batch["speech"] = resampler(speech_array).squeeze().numpy()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
predicted_ids = torch.argmax(logits, dim=-1)
print("Prediction:", processor.batch_decode(predicted_ids))
print("Reference:", test_dataset["sentence"][:2])
The model can be evaluated as follows on the Chinese (Hong Kong) test data of Common Voice.
!pip install jiwer
import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re
import argparse
lang_id = "zh-HK"
model_id = "ctl/wav2vec2-large-xlsr-cantonese"
chars_to_ignore_regex = '[\,\?\.\!\-\;\:"\“\%\‘\”\�\.\⋯\!\-\:\–\。\》\,\)\,\?\;\~\~\…\︰\,\(\」\‧\《\﹔\、\—\/\,\「\﹖\·\']'
test_dataset = load_dataset("common_voice", f"{lang_id}", split="test")
cer = load_metric("cer")
processor = Wav2Vec2Processor.from_pretrained(f"{model_id}")
model = Wav2Vec2ForCTC.from_pretrained(f"{model_id}")
model.to("cuda")
resampler = torchaudio.transforms.Resample(48_000, 16_000)
# Preprocessing the datasets.
# We need to read the aduio files as arrays
def speech_file_to_array_fn(batch):
batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
speech_array, sampling_rate = torchaudio.load(batch["path"])
batch["speech"] = resampler(speech_array).squeeze().numpy()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
# Preprocessing the datasets.
# We need to read the aduio files as arrays
def evaluate(batch):
inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
pred_ids = torch.argmax(logits, dim=-1)
batch["pred_strings"] = processor.batch_decode(pred_ids)
return batch
result = test_dataset.map(evaluate, batched=True, batch_size=16)
print("CER: {:2f}".format(100 * cer.compute(predictions=result["pred_strings"], references=result["sentence"])))
Test Result: 15.51 %
The Common Voice train, validation were used for training.
The script used for training will be posted here
From the published model card. Full card on the HuggingFace links in the sidebar.
Benchmarks
As published on the model card — the maker's own numbers, not measured by AxForge.
| Task | Dataset | Metric | Score |
|---|---|---|---|
| Speech Recognition | Common Voice zh-HK | Test CER | 15.360 |
Using it via the API
Once AxForge deploys ctl-wav2vec2-large-xlsr-cantonese for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (ctl-wav2vec2-large-xlsr-cantonese below is illustrative; you get the exact model name on deployment.)
$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \ -H "Authorization: Bearer $AXFORGE_API_KEY" \ -F model="ctl-wav2vec2-large-xlsr-cantonese" -F file=@audio.mp3
Create an account — your API key is available in the console. 3M free tokens every 30 days with every new account.