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Delete whisper2.py

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  1. whisper2.py +0 -114
whisper2.py DELETED
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- from transformers import WhisperForConditionalGeneration, WhisperProcessor
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- import torchaudio
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- import torch
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- import librosa
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- import ffmpeg
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-
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- MODEL_NAME = "openai/whisper-large-v3"
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-
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- device = "cuda:0" if torch.cuda.is_available() else "cpu"
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- device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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-
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- print("[ INFO ] Device: ", device)
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- #torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
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- torch_dtype = torch.float32
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-
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- model = WhisperForConditionalGeneration.from_pretrained(MODEL_NAME, torch_dtype=torch_dtype).to(device)
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- processor = WhisperProcessor.from_pretrained(MODEL_NAME)
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-
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-
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- def convert_forced_to_tokens(forced_decoder_ids):
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- forced_decoder_tokens = []
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- for i, (idx, token) in enumerate(forced_decoder_ids):
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- if token is not None:
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- forced_decoder_tokens.append([idx, processor.tokenizer.decode(token)])
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- else:
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- forced_decoder_tokens.append([idx, token])
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- return forced_decoder_tokens
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-
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-
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- def change_formate(input_file):
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- ffmpeg.input(input_file).output("output.wav", **{'ar': '16000'}).run(overwrite_output=True) #loglevel='quiet'
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- return "output.wav"
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-
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-
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- def generate(audio):
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- # audio = change_formate(audio)
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- input_audio, sample_rate = torchaudio.load(audio)
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- input_audio = torchaudio.transforms.Resample(sample_rate, 16000)(input_audio)
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- #metadata = torchaudio.info(audio)
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- #length1 = math.ceil(metadata.num_frames / metadata.sample_rate)
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- length = librosa.get_duration(path=audio)
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-
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- input_speech = input_audio[0]
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-
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-
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- if length <= 30:
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- input_features = processor(input_speech,
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- sampling_rate=16_000,
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- return_tensors="pt", torch_dtype=torch_dtype).input_features.to(device)
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-
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- else:
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- input_features = processor(input_speech,
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- return_tensors="pt",
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- truncation=False,
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- padding="longest",
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- return_attention_mask=True,
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- sampling_rate=16_000).input_features.to(device)
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- forced_decoder_ids = []
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- forced_decoder_ids.append([1,50270]) #[1, '<|ca|>']
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- forced_decoder_ids.append([2,50262]) #[2, '<|es|>']
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- forced_decoder_ids.append([3,50360]) #[3, '<|transcribe|>']
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-
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- forced_decoder_ids_modified = forced_decoder_ids
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- idx = processor.tokenizer.all_special_tokens.index("<|startofprev|>")
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- forced_bos_token_id = processor.tokenizer.all_special_ids[idx]
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-
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- prompt = " transcribe an audio containing code-switching between es and ca"
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- prompt_tokens = processor.tokenizer(prompt, add_special_tokens=False).input_ids
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-
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- # we need to force these tokens
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- forced_decoder_ids = []
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- for idx, token in enumerate(prompt_tokens):
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- # indexing starts from 1 for forced tokens (token at position 0 is the SOS token)
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- forced_decoder_ids.append([idx + 1, token])
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-
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- # now we add the SOS token at the end
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- offset = len(forced_decoder_ids)
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- forced_decoder_ids.append([offset + 1, model.generation_config.decoder_start_token_id])
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-
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- # now we need to append the rest of the prefix tokens (lang, task, timestamps)
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- offset = len(forced_decoder_ids)
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- for idx, token in forced_decoder_ids_modified:
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- forced_decoder_ids.append([idx + offset , token])
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-
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- model.config.forced_decoder_ids = forced_decoder_ids
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- model.generation_config.forced_decoder_ids = forced_decoder_ids
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-
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-
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- if length <= 30:
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- pred_ids = model.generate(input_features,
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- return_timestamps=True,
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- decoder_start_token_id=forced_bos_token_id,
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- max_new_tokens=128)
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- #exclude prompt from output
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- forced_decoder_tokens = convert_forced_to_tokens(forced_decoder_ids)
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- output = processor.decode(pred_ids[0][len(forced_decoder_tokens) + 1:], skip_special_tokens=True)
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-
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- else:
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- pred_ids = model.generate(input_features,
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- return_timestamps=True,
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- decoder_start_token_id=forced_bos_token_id,
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- logprob_threshold=-1.0,
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- compression_ratio_threshold=1.35,
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- temperature=(0.0, 0.2, 0.4),
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- no_speech_threshold=0.1,
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- )
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- output = processor.batch_decode(pred_ids, skip_special_tokens=True)
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-
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-
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- if length <= 30:
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- return output[1:]
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- else:
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- return output[0]
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-