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Browse files- 1000_unigram.model +3 -0
- README.md +5 -5
- app.py +287 -0
- requirements.txt +5 -0
1000_unigram.model
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version https://git-lfs.github.com/spec/v1
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oid sha256:35a3a5130d52af7c3eb92cbf0c05bfed2f43c3204f3d17941a71cf8b46c84894
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size 257888
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README.md
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---
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title: Macedonian ASR Demo
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emoji:
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colorFrom:
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sdk: gradio
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sdk_version: 4.
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app_file: app.py
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pinned: false
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license: cc-by-4.0
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---
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title: Macedonian ASR Demo
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emoji: 👁
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colorFrom: purple
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colorTo: yellow
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sdk: gradio
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sdk_version: 4.41.0
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app_file: app.py
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pinned: false
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license: cc-by-4.0
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app.py
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import os
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# os.environ["PYTORCH_CUDA_ALLOC_CONF"] = "expandable_segments:True"
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import gc
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from functools import partial
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import gradio as gr
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import torch
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from speechbrain.inference.interfaces import Pretrained, foreign_class
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from transformers import T5Tokenizer, T5ForConditionalGeneration
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import librosa
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import whisper_timestamped as whisper
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline, Wav2Vec2ForCTC, AutoProcessor
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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torch.backends.cuda.matmul.allow_tf32 = True
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def clean_up_memory():
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gc.collect()
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torch.cuda.empty_cache()
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def recap_sentence(string):
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# Restore capitalization and punctuation using the model
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inputs = recap_tokenizer(["restore capitalization and punctuation: " + string], return_tensors="pt", padding=True).to(device)
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outputs = recap_model.generate(**inputs, max_length=768, num_beams=5, early_stopping=True).squeeze(0)
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recap_result = recap_tokenizer.decode(outputs, skip_special_tokens=True)
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return recap_result
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def return_prediction_w2v2(mic=None, file=None, device=device):
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if mic is not None:
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waveform, sr = librosa.load(mic, sr=16000)
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waveform = waveform[:30*sr]
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w2v2_result = w2v2_classifier.classify_file_w2v2(waveform, device)
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elif file is not None:
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waveform, sr = librosa.load(file, sr=16000)
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waveform = waveform[:30*sr]
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w2v2_result = w2v2_classifier.classify_file_w2v2(waveform, device)
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else:
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return "You must either provide a mic recording or a file"
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recap_result = recap_sentence(w2v2_result[0])
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# If the letter after punct is small, recap it
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for i, letter in enumerate(recap_result):
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if i > 1 and recap_result[i-2] in [".", "!", "?"] and letter.islower():
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recap_result = recap_result[:i] + letter.upper() + recap_result[i+1:]
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clean_up_memory()
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return recap_result
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def return_prediction_whisper(mic=None, file=None, device=device):
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if mic is not None:
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waveform, sr = librosa.load(mic, sr=16000)
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waveform = waveform[:30*sr]
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whisper_result = whisper_classifier.classify_file_whisper_mkd(waveform, device)
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elif file is not None:
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waveform, sr = librosa.load(file, sr=16000)
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waveform = waveform[:30*sr]
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whisper_result = whisper_classifier.classify_file_whisper_mkd(waveform, device)
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else:
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return "You must either provide a mic recording or a file"
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recap_result = recap_sentence(whisper_result[0])
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# If the letter after punct is small, recap it
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for i, letter in enumerate(recap_result):
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if i > 1 and recap_result[i-2] in [".", "!", "?"] and letter.islower():
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recap_result = recap_result[:i] + letter.upper() + recap_result[i+1:]
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clean_up_memory()
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return recap_result
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def return_prediction_compare(mic=None, file=None, device=device):
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# pipe_whisper.model.to(device)
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# mms_model.to(device)
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if mic is not None:
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waveform, sr = librosa.load(mic, sr=16000)
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waveform = waveform[:30*sr]
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whisper_mkd_result = whisper_classifier.classify_file_whisper_mkd(waveform, device)
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# result_generator_w2v2 = w2v2_classifier.classify_file_w2v2(mic, device)
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whisper_result = whisper_classifier.classify_file_whisper(waveform, pipe_whisper, device)
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mms_result_generator = whisper_classifier.classify_file_mms(waveform, processor_mms, mms_model, device)
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elif file is not None:
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waveform, sr = librosa.load(file, sr=16000)
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waveform = waveform[:30*sr]
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whisper_mkd_result = whisper_classifier.classify_file_whisper_mkd(waveform, device)
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# result_generator_w2v2 = w2v2_classifier.classify_file_w2v2(file, device)
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whisper_result = whisper_classifier.classify_file_whisper(waveform, pipe_whisper, device)
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mms_result_generator = whisper_classifier.classify_file_mms(waveform, processor_mms, mms_model, device)
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else:
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return "You must either provide a mic recording or a file"
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# pipe_whisper.model.to("cpu")
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# mms_model.to("cpu")
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segment_results_whisper = ""
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prev_segment_whisper = ""
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# segment_results_w2v2 = ""
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# prev_segment_w2v2 = ""
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segment_results_mms = ""
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prev_segment_mms = ""
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recap_result_whisper_mkd = recap_sentence(whisper_mkd_result[0])
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recap_result_whisper = recap_sentence(whisper_result[0])
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recap_result_mms = recap_sentence(mms_result_generator[0])
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# If the letter after punct is small, recap it
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for i, letter in enumerate(recap_result_whisper_mkd):
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if i > 1 and recap_result_whisper_mkd[i-2] in [".", "!", "?"] and letter.islower():
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recap_result_whisper_mkd = recap_result_whisper_mkd[:i] + letter.upper() + recap_result_whisper_mkd[i+1:]
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for i, letter in enumerate(recap_result_whisper):
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if i > 1 and recap_result_whisper[i-2] in [".", "!", "?"] and letter.islower():
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recap_result_whisper = recap_result_whisper[:i] + letter.upper() + recap_result_whisper[i+1:]
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for i, letter in enumerate(recap_result_mms):
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if i > 1 and recap_result_mms[i-2] in [".", "!", "?"] and letter.islower():
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recap_result_mms = recap_result_mms[:i] + letter.upper() + recap_result_mms[i+1:]
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clean_up_memory()
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return "Буки-Whisper:\n" + recap_result_whisper_mkd + "\n\n" + "MMS:\n" + recap_result_mms + "\n\n" + "OpenAI Whisper:\n" + recap_result_whisper
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# yield "Our W2v2: \n" + segment_results_w2v2 + "\n\n" + "MMS transcript:\n" + segment_results_mms
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# Load Whisper model
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model_id = "openai/whisper-large-v3"
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whisper_model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id, torch_dtype=torch.float16, low_cpu_mem_usage=True, use_safetensors=True, attn_implementation="sdpa")
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whisper_model.to(device)
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processor = AutoProcessor.from_pretrained(model_id)
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pipe_whisper = pipeline(
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"automatic-speech-recognition",
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model=whisper_model,
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tokenizer=processor.tokenizer,
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feature_extractor=processor.feature_extractor,
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torch_dtype=torch.float16,
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return_timestamps=True,
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device=device,
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)
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# Load MMS model
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model_id = "facebook/mms-1b-all"
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processor_mms = AutoProcessor.from_pretrained(model_id)
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mms_model = Wav2Vec2ForCTC.from_pretrained(model_id)
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mms_model = mms_model.to(device)
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mms_model.eval()
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processor_mms.tokenizer.set_target_lang("mkd")
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mms_model.load_adapter("mkd")
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# Create a partial function with the device pre-applied
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return_prediction_whisper_with_device = partial(return_prediction_whisper, device=device)
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return_prediction_w2v2_with_device = partial(return_prediction_w2v2, device=device)
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return_prediction_with_device_compare = partial(return_prediction_compare, device=device)
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# Load the ASR models
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w2v2_classifier = foreign_class(source="Macedonian-ASR/wav2vec2-aed-macedonian-asr", pymodule_file="custom_interface_app.py", classname="ASR")
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w2v2_classifier = w2v2_classifier.to(device)
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w2v2_classifier.eval()
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whisper_classifier = foreign_class(source="Macedonian-ASR/whisper-large-v3-macedonian-asr", pymodule_file="custom_interface_app.py", classname="ASR")
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whisper_classifier = whisper_classifier.to(device)
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whisper_classifier.eval()
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# Load the T5 tokenizer and model for restoring capitalization
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recap_model_name = "Macedonian-ASR/mt5-restore-capitalization-macedonian"
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recap_tokenizer = T5Tokenizer.from_pretrained(recap_model_name)
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recap_model = T5ForConditionalGeneration.from_pretrained(recap_model_name, torch_dtype=torch.float16)
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recap_model.to(device)
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recap_model.eval()
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mic_transcribe_whisper = gr.Interface(
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fn=return_prediction_whisper_with_device,
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inputs=gr.Audio(sources="microphone", type="filepath"),
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outputs=gr.Textbox(),
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allow_flagging="never",
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live=False,
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)
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# file_transcribe_whisper = gr.Interface(
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# fn=return_prediction_whisper_with_device,
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# inputs=gr.Audio(sources="upload", type="filepath"),
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# outputs=gr.Textbox(),
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# allow_flagging="never",
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# live=False
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# )
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mic_transcribe_w2v2 = gr.Interface(
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fn=return_prediction_w2v2_with_device,
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inputs=gr.Audio(sources="microphone", type="filepath"),
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outputs=gr.Textbox(),
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allow_flagging="never",
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live=False,
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)
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# file_transcribe_w2v2 = gr.Interface(
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# fn=return_prediction_w2v2_with_device,
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# inputs=gr.Audio(sources="upload", type="filepath"),
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# outputs=gr.Textbox(),
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# allow_flagging="never",
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# live=False
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# )
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mic_transcribe_compare = gr.Interface(
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fn=return_prediction_with_device_compare,
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inputs=gr.Audio(sources="microphone", type="filepath"),
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outputs=gr.Textbox(),
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allow_flagging="never",
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live=False,
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)
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# file_transcribe_compare = gr.Interface(
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# fn=return_prediction_with_device_compare,
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# inputs=gr.Audio(sources="upload", type="filepath"),
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# outputs=gr.Textbox(),
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# allow_flagging="never",
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# live=False
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# )
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project_description = '''
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## Автори:
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1. **Дејан Порјазовски**
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2. **Илина Јакимовска**
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3. **Ордан Чукалиев**
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4. **Никола Стиков**
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Оваа колаборација е дел од активностите на **Центарот за напредни интердисциплинарни истражувања ([ЦеНИИс](https://ukim.edu.mk/en/centri/centar-za-napredni-interdisciplinarni-istrazhuvanja-ceniis))** при УКИМ.
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## Во тренирањето на овој модел се употребени податоци од:
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240 |
+
1. Дигитален архив за етнолошки и антрополошки ресурси (ДАЕАР) при Институтот за етнологија и антропологија, Природно-математички факултет при УКИМ.
|
241 |
+
2. Аудио верзија на меѓународното списание „ЕтноАнтропоЗум“ на Институтот за етнологија и антропологија, Природ��о-математички факултет при УКИМ.
|
242 |
+
3. Аудио подкастот „Обични луѓе“ на Илина Јакимовска
|
243 |
+
4. Научните видеа од серијалот „Наука за деца“, фондација КАНТАРОТ
|
244 |
+
5. Македонска верзија на Mozilla Common Voice (верзија 18.0)
|
245 |
+
'''
|
246 |
+
|
247 |
+
# Custom CSS
|
248 |
+
css = """
|
249 |
+
.gradio-container {
|
250 |
+
background-color: #f0f0f0; /* Set your desired background color */
|
251 |
+
}
|
252 |
+
.custom-markdown p, .custom-markdown li, .custom-markdown h2, .custom-markdown a {
|
253 |
+
font-size: 15px !important;
|
254 |
+
font-family: Arial, sans-serif !important;
|
255 |
+
}
|
256 |
+
.gradio-container {
|
257 |
+
background-color: #f3f3f3 !important;
|
258 |
+
}
|
259 |
+
"""
|
260 |
+
|
261 |
+
transcriber_app = gr.Blocks(css=css, delete_cache=(60, 120))
|
262 |
+
|
263 |
+
with transcriber_app:
|
264 |
+
state = gr.State()
|
265 |
+
gr.Markdown(project_description, elem_classes="custom-markdown")
|
266 |
+
|
267 |
+
# gr.TabbedInterface(
|
268 |
+
# [mic_transcribe_whisper, mic_transcribe_compare],
|
269 |
+
# ["Буки-Whisper транскрипција", "Споредба на модели"],
|
270 |
+
# )
|
271 |
+
# state = gr.State(value=[], delete_callback=lambda v: print("STATE DELETED"))
|
272 |
+
|
273 |
+
gr.TabbedInterface(
|
274 |
+
[mic_transcribe_whisper, mic_transcribe_w2v2, mic_transcribe_compare],
|
275 |
+
["Буки-Whisper транскрипција", "Буки-W2v2 транскрипција", "Споредба на модели"],
|
276 |
+
)
|
277 |
+
state = gr.State(value=[], delete_callback=lambda v: print("STATE DELETED"))
|
278 |
+
|
279 |
+
transcriber_app.unload(return_prediction_whisper)
|
280 |
+
transcriber_app.unload(return_prediction_compare)
|
281 |
+
transcriber_app.unload(return_prediction_w2v2)
|
282 |
+
|
283 |
+
|
284 |
+
# transcriber_app.launch(debug=True, share=True, ssl_verify=False)
|
285 |
+
if __name__ == "__main__":
|
286 |
+
transcriber_app.queue()
|
287 |
+
transcriber_app.launch(share=True)
|
requirements.txt
ADDED
@@ -0,0 +1,5 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
speechbrain
|
2 |
+
transformers
|
3 |
+
librosa
|
4 |
+
whisper_timestamped
|
5 |
+
accelerate
|