asr-inference / whisper2.py
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from transformers import WhisperForConditionalGeneration, WhisperProcessor
import torchaudio
import torch
import librosa
import ffmpeg
MODEL_NAME = "openai/whisper-large-v3"
device = "cuda:0" if torch.cuda.is_available() else "cpu"
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
print("[ INFO ] Device: ", device)
#torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
torch_dtype = torch.float32
model = WhisperForConditionalGeneration.from_pretrained(MODEL_NAME, torch_dtype=torch_dtype).to(device)
processor = WhisperProcessor.from_pretrained(MODEL_NAME)
def convert_forced_to_tokens(forced_decoder_ids):
forced_decoder_tokens = []
for i, (idx, token) in enumerate(forced_decoder_ids):
if token is not None:
forced_decoder_tokens.append([idx, processor.tokenizer.decode(token)])
else:
forced_decoder_tokens.append([idx, token])
return forced_decoder_tokens
def change_formate(input_file):
ffmpeg.input(input_file).output("output.wav", **{'ar': '16000'}).run(overwrite_output=True) #loglevel='quiet'
return "output.wav"
def generate(audio):
# audio = change_formate(audio)
input_audio, sample_rate = torchaudio.load(audio)
input_audio = torchaudio.transforms.Resample(sample_rate, 16000)(input_audio)
#metadata = torchaudio.info(audio)
#length1 = math.ceil(metadata.num_frames / metadata.sample_rate)
length = librosa.get_duration(path=audio)
input_speech = input_audio[0]
if length <= 30:
input_features = processor(input_speech,
sampling_rate=16_000,
return_tensors="pt", torch_dtype=torch_dtype).input_features.to(device)
else:
input_features = processor(input_speech,
return_tensors="pt",
truncation=False,
padding="longest",
return_attention_mask=True,
sampling_rate=16_000).input_features.to(device)
forced_decoder_ids = []
forced_decoder_ids.append([1,50270]) #[1, '<|ca|>']
forced_decoder_ids.append([2,50262]) #[2, '<|es|>']
forced_decoder_ids.append([3,50360]) #[3, '<|transcribe|>']
forced_decoder_ids_modified = forced_decoder_ids
idx = processor.tokenizer.all_special_tokens.index("<|startofprev|>")
forced_bos_token_id = processor.tokenizer.all_special_ids[idx]
prompt = " transcribe an audio containing code-switching between es and ca"
prompt_tokens = processor.tokenizer(prompt, add_special_tokens=False).input_ids
# we need to force these tokens
forced_decoder_ids = []
for idx, token in enumerate(prompt_tokens):
# indexing starts from 1 for forced tokens (token at position 0 is the SOS token)
forced_decoder_ids.append([idx + 1, token])
# now we add the SOS token at the end
offset = len(forced_decoder_ids)
forced_decoder_ids.append([offset + 1, model.generation_config.decoder_start_token_id])
# now we need to append the rest of the prefix tokens (lang, task, timestamps)
offset = len(forced_decoder_ids)
for idx, token in forced_decoder_ids_modified:
forced_decoder_ids.append([idx + offset , token])
model.config.forced_decoder_ids = forced_decoder_ids
model.generation_config.forced_decoder_ids = forced_decoder_ids
if length <= 30:
pred_ids = model.generate(input_features,
return_timestamps=True,
decoder_start_token_id=forced_bos_token_id,
max_new_tokens=128)
#exclude prompt from output
forced_decoder_tokens = convert_forced_to_tokens(forced_decoder_ids)
output = processor.decode(pred_ids[0][len(forced_decoder_tokens) + 1:], skip_special_tokens=True)
else:
pred_ids = model.generate(input_features,
return_timestamps=True,
decoder_start_token_id=forced_bos_token_id,
logprob_threshold=-1.0,
compression_ratio_threshold=1.35,
temperature=(0.0, 0.2, 0.4),
no_speech_threshold=0.1,
)
output = processor.batch_decode(pred_ids, skip_special_tokens=True)
if length <= 30:
return output[1:]
else:
return output[0]