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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] | |