Spaces:
Running
on
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Running
on
Zero
add whisper with language tags and prompt
Browse files- __pycache__/app.cpython-310.pyc +0 -0
- __pycache__/whisper.cpython-310.pyc +0 -0
- __pycache__/whisper2.cpython-310.pyc +0 -0
- app.py +5 -22
- requirements.txt +2 -0
- whisper2.py +103 -0
__pycache__/app.cpython-310.pyc
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Binary file (1.23 kB). View file
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__pycache__/whisper.cpython-310.pyc
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Binary file (2.45 kB). View file
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__pycache__/whisper2.cpython-310.pyc
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Binary file (2.45 kB). View file
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app.py
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@@ -1,34 +1,18 @@
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import torch
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import gradio as gr
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from transformers import pipeline
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from transformers.pipelines.audio_utils import ffmpeg_read
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import tempfile
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import os
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MODEL_NAME = "
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BATCH_SIZE = 8
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FILE_LIMIT_MB = 1000
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device = 0 if torch.cuda.is_available() else "cpu"
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pipe = pipeline(
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task="automatic-speech-recognition",
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model=MODEL_NAME,
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chunk_length_s=30,
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device=device,
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)
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def transcribe(inputs
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if inputs is None:
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raise gr.Error("Cap fitxer d'àudio introduit! Si us plau pengeu un fitxer "\
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"o enregistreu un àudio abans d'enviar la vostra sol·licitud")
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text = pipe(inputs, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=True)["text"]
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return text
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description_string = "Transcripció automàtica de micròfon o de fitxers d'àudio.\n Aquest demostrador s'ha desenvolupat per"\
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file_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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gr.Audio(sources=["upload", "microphone"], type="filepath", label="Audio")
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gr.Radio(["transcribe", "translate"], label="Task", value="transcribe"),
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],
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outputs="text",
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title="Transcripció automàtica d'àudio",
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import gradio as gr
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from whisper2 import generate
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MODEL_NAME = "/whisper-large-v3"
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def transcribe(inputs):
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if inputs is None:
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raise gr.Error("Cap fitxer d'àudio introduit! Si us plau pengeu un fitxer "\
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"o enregistreu un àudio abans d'enviar la vostra sol·licitud")
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return generate(audio=inputs)
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description_string = "Transcripció automàtica de micròfon o de fitxers d'àudio.\n Aquest demostrador s'ha desenvolupat per"\
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file_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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gr.Audio(sources=["upload", "microphone"], type="filepath", label="Audio")
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],
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outputs="text",
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title="Transcripció automàtica d'àudio",
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requirements.txt
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@@ -2,3 +2,5 @@ git+https://github.com/huggingface/transformers
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torch
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yt-dlp
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gradio==4.20.0
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torch
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yt-dlp
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gradio==4.20.0
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torchaudio==2.2.1
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librosa==0.10.1
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whisper2.py
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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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MODEL_NAME = "openai/whisper-large-v3"
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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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device = "cpu"
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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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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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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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def generate(audio):
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input_audio, sample_rate = torchaudio.load(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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input_speech = input_audio[0]
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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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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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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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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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# 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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# 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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# 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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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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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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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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return output[0]
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