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app.py ADDED
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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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+
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+
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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-v3-7lang-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 = "homebrewltd/Ichigo-llama3.1-s-instruct-v0.3-phase-3"
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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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+
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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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+
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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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+
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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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+
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+ result = ''.join(f'<|sound_{num:04d}|>' for num in codes)
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+ return f'Transcribe the speech in this audio sample:<|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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+
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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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+
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+
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+ @spaces.GPU
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+ def process_input(audio_file=None):
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+
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+ for partial_message in process_audio(audio_file):
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+ yield partial_message
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+
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+
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+ @spaces.GPU
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+ def process_transcribe_input(audio_file=None):
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+
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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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+
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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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+
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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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+
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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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+
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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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+
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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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+
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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])
132
+ )
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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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+
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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:
141
+ 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"):
150
+ if file.endswith(".wav"):
151
+ good_examples.append([f"./examples/{file}"])
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+ bad_examples = []
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+ for file in os.listdir("./bad_examples"):
154
+ 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("# Ichigo-llama3-s: Llama3.1 with listening capabilities")
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+ gr.Markdown("Record your voice or upload audio and send it to the model.")
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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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+
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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="Send", 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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+
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+ convert_button = gr.Button("Convert to Audio", visible=False)
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+ submit_button = gr.Button("Send")
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+ transcrip_button = gr.Button("Make Model Transcribe the audio")
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+
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+ text_output = gr.Textbox(label="Generated Text")
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+
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+ def update_visibility(input_type):
177
+ 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):
183
+ 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]
188
+ )
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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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+
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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]
200
+ )
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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]
205
+ )
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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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+ # launch locally
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+ # iface.launch(server_name="0.0.0.0")
bad_examples/bad-What-is-Love.wav ADDED
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bad_examples/bad-who-bears-Obama.wav ADDED
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examples/Can-you-write-a-registration-letter.wav ADDED
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examples/Hello.wav ADDED
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examples/Who-is-Harry-Potter.wav ADDED
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examples/Write-an-email.wav ADDED
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examples/codeapythonscript.wav ADDED
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examples/generate_3_questions_you_can_ask_an_interviewer.wav ADDED
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examples/story.wav ADDED
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examples/what-is-the-color-of-the-elephant.wav ADDED
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examples/what-is-the-color-of-the-ocean.wav ADDED
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generate_audio.py ADDED
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1
+ import torchaudio
2
+
3
+ from whisperspeech.pipeline import Pipeline
4
+ import argparse
5
+
6
+ def parse_args():
7
+ parser = argparse.ArgumentParser(description="Convert text to audio.")
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+ parser.add_argument(
9
+ "--text",
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+ type=str,
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+ required=True,
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+ help="The text to convert to audio.",
13
+ )
14
+ return parser.parse_args()
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+
16
+ def convert_text_to_audio(pipe: Pipeline, text: str):
17
+ """Convert text to audio.
18
+
19
+ Args:
20
+ pipe (Pipeline): The pipeline to use for text-to-speech.
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+ text (str): The text to convert to audio.
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+
23
+ Returns:
24
+ torch.Tensor: The generated audio.
25
+ """
26
+ return pipe.generate(text)
27
+
28
+
29
+ def convert_text_to_audio_file(pipe: Pipeline, text: str, output_path: str):
30
+ """Convert text to audio and save it to a file.
31
+
32
+ Args:
33
+ pipe (Pipeline): The pipeline to use for text-to-speech.
34
+ text (str): The text to convert to audio.
35
+ output_path (str): The path to save the audio file.
36
+ """
37
+ pipe.generate_to_file(output_path, text)
38
+
39
+
40
+ class TTSProcessor:
41
+ def __init__(self, device: str):
42
+ """Initialize the TTS Processor with a specified device."""
43
+ self.pipe = Pipeline(
44
+ s2a_ref="collabora/whisperspeech:s2a-q4-tiny-en+pl.model", device=device
45
+ )
46
+
47
+ def get_reference_voice_embedding(self, path: str):
48
+ """Get the reference voice embedding from the given audio file.
49
+
50
+ Args:
51
+ path (str): The path to the audio file.
52
+ Returns:
53
+ torch.Tensor: The reference voice embedding."""
54
+ return self.pipe.extract_spk_emb(path).cpu()
55
+
56
+ def convert_text_to_audio(self, text: str, speaker=None):
57
+ """Convert text to audio.
58
+
59
+ Args:
60
+ text (str): The text to convert to audio.
61
+
62
+ Returns:
63
+ torch.Tensor: The generated audio.
64
+ """
65
+ return self.pipe.generate(text, speaker=speaker)
66
+
67
+ def convert_text_to_audio_file(self, text: str, output_path: str, speaker=None):
68
+ """Convert text to audio and save it to a file.
69
+
70
+ Args:
71
+ text (str): The text to convert to audio.
72
+ output_path (str): The path to save the audio file.
73
+ """
74
+ self.pipe.generate_to_file(output_path, text, speaker=speaker)
75
+ if __name__ == "__main__":
76
+ args = parse_args()
77
+ processor = TTSProcessor("cuda")
78
+ text = args.text
79
+ text = text.lower()
80
+ text_split = "_".join(text.lower().split(" "))
81
+ # remove the last character if it is a period
82
+ if text_split[-1] == ".":
83
+ text_split = text_split[:-1]
84
+ print(text_split)
85
+ path = f"./examples/{text_split}.wav"
86
+ processor.convert_text_to_audio_file(text, path)
87
+
requirements.txt ADDED
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+ openai-whisper==20231117
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+ IPython
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+ peft
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+ huggingface_hub
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+ matplotlib
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+ pyarrow
7
+ datasets
8
+ encodec
9
+ soundfile
10
+ gradio==4.39.0
11
+ transformers
12
+ bitsandbytes
13
+ torchvision
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+ vector_quantize_pytorch
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+ webdataset
16
+ whisperspeech
17
+ --extra-index-url https://download.pytorch.org/whl/cu121
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+ torch==2.2.0
19
+ torchaudio==2.2.0
20
+ fsspec==2024.6.1
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+ anyio==4.4.0
user_audio/0bf62a35-94bb-43f0-9a5f-9691c1691859_temp_audio.wav ADDED
Binary file (147 kB). View file