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Update app.py
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app.py
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@@ -1,50 +1,209 @@
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import gradio as gr
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import torch
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from transformers import AutoTokenizer, LlamaForCausalLM
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import spaces
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import gradio as gr
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import torch
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import spaces
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import torchaudio
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from whisperspeech.vq_stoks import RQBottleneckTransformer
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from encodec.utils import convert_audio
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from transformers import AutoModelForCausalLM, AutoTokenizer, BitsAndBytesConfig, pipeline
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from transformers import StoppingCriteria, StoppingCriteriaList, TextIteratorStreamer
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from threading import Thread
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import logging
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import os
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from generate_audio import (
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TTSProcessor,
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)
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import uuid
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device = "cuda" if torch.cuda.is_available() else "cpu"
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vq_model = RQBottleneckTransformer.load_model(
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"whisper-vq-stoks-medium-en+pl-fixed.model"
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).to(device)
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# tts = TTSProcessor('cpu')
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use_8bit = False
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llm_path = "akjindal53244/Llama-3.1-Storm-8B"
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tokenizer = AutoTokenizer.from_pretrained(llm_path)
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model_kwargs = {}
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if use_8bit:
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model_kwargs["quantization_config"] = BitsAndBytesConfig(
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load_in_8bit=True,
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llm_int8_enable_fp32_cpu_offload=False,
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llm_int8_has_fp16_weight=False,
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)
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else:
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model_kwargs["torch_dtype"] = torch.bfloat16
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model = AutoModelForCausalLM.from_pretrained(llm_path, **model_kwargs).to(device)
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@spaces.GPU
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def audio_to_sound_tokens_whisperspeech(audio_path):
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vq_model.ensure_whisper('cuda')
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wav, sr = torchaudio.load(audio_path)
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if sr != 16000:
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wav = torchaudio.functional.resample(wav, sr, 16000)
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with torch.no_grad():
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codes = vq_model.encode_audio(wav.to(device))
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codes = codes[0].cpu().tolist()
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result = ''.join(f'<|sound_{num:04d}|>' for num in codes)
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return f'<|sound_start|>{result}<|sound_end|>'
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@spaces.GPU
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def audio_to_sound_tokens_whisperspeech_transcribe(audio_path):
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vq_model.ensure_whisper('cuda')
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wav, sr = torchaudio.load(audio_path)
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if sr != 16000:
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wav = torchaudio.functional.resample(wav, sr, 16000)
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with torch.no_grad():
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codes = vq_model.encode_audio(wav.to(device))
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codes = codes[0].cpu().tolist()
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result = ''.join(f'<|sound_{num:04d}|>' for num in codes)
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return f'<|reserved_special_token_69|><|sound_start|>{result}<|sound_end|>'
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# print(tokenizer.encode("<|sound_0001|>", add_special_tokens=False))# return the audio tensor
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# print(tokenizer.eos_token)
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@spaces.GPU
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def text_to_audio_file(text):
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# gen a random id for the audio file
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id = str(uuid.uuid4())
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temp_file = f"./user_audio/{id}_temp_audio.wav"
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text = text
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text_split = "_".join(text.lower().split(" "))
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# remove the last character if it is a period
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if text_split[-1] == ".":
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text_split = text_split[:-1]
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tts = TTSProcessor("cuda")
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tts.convert_text_to_audio_file(text, temp_file)
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# logging.info(f"Saving audio to {temp_file}")
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# torchaudio.save(temp_file, audio.cpu(), sample_rate=24000)
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print(f"Saved audio to {temp_file}")
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return temp_file
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@spaces.GPU
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def process_input(audio_file=None):
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for partial_message in process_audio(audio_file):
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yield partial_message
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@spaces.GPU
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def process_transcribe_input(audio_file=None):
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for partial_message in process_audio(audio_file, transcript=True):
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yield partial_message
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class StopOnTokens(StoppingCriteria):
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def __call__(self, input_ids: torch.LongTensor, scores: torch.FloatTensor, **kwargs) -> bool:
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# encode </s> token
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stop_ids = [tokenizer.eos_token_id, 128009] # Adjust this based on your model's tokenizer
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for stop_id in stop_ids:
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if input_ids[0][-1] == stop_id:
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return True
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return False
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@spaces.GPU
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def process_audio(audio_file, transcript=False):
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if audio_file is None:
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raise ValueError("No audio file provided")
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logging.info(f"Audio file received: {audio_file}")
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logging.info(f"Audio file type: {type(audio_file)}")
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sound_tokens = audio_to_sound_tokens_whisperspeech_transcribe(audio_file) if transcript else audio_to_sound_tokens_whisperspeech(audio_file)
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logging.info("Sound tokens generated successfully")
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# logging.info(f"audio_file: {audio_file.name}")
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messages = [
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{"role": "user", "content": sound_tokens},
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]
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stop = StopOnTokens()
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input_str = tokenizer.apply_chat_template(messages, tokenize=False)
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input_ids = tokenizer.encode(input_str, return_tensors="pt")
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input_ids = input_ids.to(model.device)
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streamer = TextIteratorStreamer(tokenizer, timeout=10., skip_prompt=True, skip_special_tokens=True)
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generation_kwargs = dict(
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input_ids=input_ids,
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streamer=streamer,
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max_new_tokens=1024,
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do_sample=False,
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stopping_criteria=StoppingCriteriaList([stop])
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)
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thread = Thread(target=model.generate, kwargs=generation_kwargs)
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thread.start()
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partial_message = ""
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for new_token in streamer:
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partial_message += new_token
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if tokenizer.eos_token in partial_message:
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break
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partial_message = partial_message.replace("assistant\n\n", "")
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yield partial_message
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# def stop_generation():
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# # This is a placeholder. Implement actual stopping logic here if needed.
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# return "Generation stopped.", gr.Button.update(interactive=False)
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# take all the examples from the examples folder
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good_examples = []
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for file in os.listdir("./examples"):
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if file.endswith(".wav"):
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good_examples.append([f"./examples/{file}"])
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bad_examples = []
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for file in os.listdir("./bad_examples"):
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if file.endswith(".wav"):
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bad_examples.append([f"./bad_examples/{file}"])
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examples = []
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examples.extend(good_examples)
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examples.extend(bad_examples)
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with gr.Blocks() as iface:
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gr.Markdown("# Llama3.1-S: checkpoint Aug 19, 2024")
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gr.Markdown("Enter text to convert to audio, then submit the audio to generate text or Upload Audio")
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gr.Markdown("Powered by [Homebrew Ltd](https://homebrew.ltd/) | [Read our blog post](https://homebrew.ltd/blog/llama3-just-got-ears)")
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with gr.Row():
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input_type = gr.Radio(["text", "audio"], label="Input Type", value="audio")
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text_input = gr.Textbox(label="Text Input", visible=False)
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audio_input = gr.Audio(label="Audio", type="filepath", visible=True)
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# audio_output = gr.Audio(label="Converted Audio", type="filepath", visible=False)
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convert_button = gr.Button("Make synthetic audio", visible=False)
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submit_button = gr.Button("Chat with AI using audio")
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transcrip_button = gr.Button("Make Model transcribe the audio")
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text_output = gr.Textbox(label="Generated Text")
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def update_visibility(input_type):
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return (gr.update(visible=input_type == "text"),
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gr.update(visible=input_type == "text"))
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def convert_and_display(text):
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audio_file = text_to_audio_file(text)
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return audio_file
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def process_example(file_path):
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return update_visibility("audio")
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input_type.change(
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update_visibility,
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inputs=[input_type],
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outputs=[text_input, convert_button]
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)
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convert_button.click(
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convert_and_display,
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inputs=[text_input],
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outputs=[audio_input]
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)
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submit_button.click(
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process_input,
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inputs=[audio_input],
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outputs=[text_output]
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)
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transcrip_button.click(
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process_transcribe_input,
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inputs=[audio_input],
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outputs=[text_output]
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)
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gr.Examples(examples, inputs=[audio_input])
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iface.queue()
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iface.launch(
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