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# Copyright (c) 2024 NVIDIA CORPORATION.
# Licensed under the MIT license.
import os
import yaml
import gradio as gr
import librosa
from pydub import AudioSegment
import soundfile as sf
import numpy as np
import torch
import laion_clap
from inference_utils import prepare_tokenizer, prepare_model, inference
from data import AudioTextDataProcessor
def load_laionclap():
model = laion_clap.CLAP_Module(enable_fusion=True, amodel='HTSAT-tiny').cuda()
model.load_ckpt(ckpt='630k-audioset-fusion-best.pt')
model.eval()
return model
def int16_to_float32(x):
return (x / 32767.0).astype(np.float32)
def float32_to_int16(x):
x = np.clip(x, a_min=-1., a_max=1.)
return (x * 32767.).astype(np.int16)
def load_audio(file_path, target_sr=44100, duration=33.25, start=0.0):
if file_path.endswith('.mp3'):
audio = AudioSegment.from_file(file_path)
if len(audio) > (start + duration) * 1000:
audio = audio[start * 1000:(start + duration) * 1000]
if audio.frame_rate != target_sr:
audio = audio.set_frame_rate(target_sr)
if audio.channels > 1:
audio = audio.set_channels(1)
data = np.array(audio.get_array_of_samples())
if audio.sample_width == 2:
data = data.astype(np.float32) / np.iinfo(np.int16).max
elif audio.sample_width == 4:
data = data.astype(np.float32) / np.iinfo(np.int32).max
else:
raise ValueError("Unsupported bit depth: {}".format(audio.sample_width))
else:
with sf.SoundFile(file_path) as audio:
original_sr = audio.samplerate
channels = audio.channels
max_frames = int((start + duration) * original_sr)
audio.seek(int(start * original_sr))
frames_to_read = min(max_frames, len(audio))
data = audio.read(frames_to_read)
if data.max() > 1 or data.min() < -1:
data = data / max(abs(data.max()), abs(data.min()))
if original_sr != target_sr:
if channels == 1:
data = librosa.resample(data.flatten(), orig_sr=original_sr, target_sr=target_sr)
else:
data = librosa.resample(data.T, orig_sr=original_sr, target_sr=target_sr)[0]
else:
if channels != 1:
data = data.T[0]
if data.min() >= 0:
data = 2 * data / abs(data.max()) - 1.0
else:
data = data / max(abs(data.max()), abs(data.min()))
return data
@torch.no_grad()
def compute_laionclap_text_audio_sim(audio_file, laionclap_model, outputs):
try:
data = load_audio(audio_file, target_sr=48000)
except Exception as e:
print(audio_file, 'unsuccessful due to', e)
return [0.0] * len(outputs)
audio_data = data.reshape(1, -1)
audio_data_tensor = torch.from_numpy(int16_to_float32(float32_to_int16(audio_data))).float().cuda()
audio_embed = laionclap_model.get_audio_embedding_from_data(x=audio_data_tensor, use_tensor=True)
text_embed = laionclap_model.get_text_embedding(outputs, use_tensor=True)
cos = torch.nn.CosineSimilarity(dim=1, eps=1e-6)
cos_similarity = cos(audio_embed.repeat(text_embed.shape[0], 1), text_embed)
return cos_similarity.squeeze().cpu().numpy()
inference_kwargs = {
"do_sample": True,
"top_k": 50,
"top_p": 0.95,
"num_return_sequences": 10
}
config = yaml.load(open('chat.yaml'), Loader=yaml.FullLoader)
clap_config = config['clap_config']
model_config = config['model_config']
text_tokenizer = prepare_tokenizer(model_config)
DataProcessor = AudioTextDataProcessor(
data_root='./',
clap_config=clap_config,
tokenizer=text_tokenizer,
max_tokens=512,
)
laionclap_model = load_laionclap()
model = prepare_model(
model_config=model_config,
clap_config=clap_config,
checkpoint_path='chat.pt'
)
def inference_item(name, prompt):
item = {
'name': str(name),
'prefix': 'The task is dialog.',
'prompt': str(prompt)
}
processed_item = DataProcessor.process(item)
outputs = inference(
model, text_tokenizer, item, processed_item,
inference_kwargs,
)
laionclap_scores = compute_laionclap_text_audio_sim(
item["name"],
laionclap_model,
outputs
)
outputs_joint = [(output, score) for (output, score) in zip(outputs, laionclap_scores)]
outputs_joint.sort(key=lambda x: -x[1])
return outputs_joint[0][0]
with gr.Blocks(title="Audio Flamingo - Demo") as ui:
gr.HTML(
"""
<div style="text-align: center; max-width: 900px; margin: 0 auto;">
<div
style="
display: inline-flex;
align-items: center;
gap: 0.8rem;
font-size: 1.5rem;
"
>
<h1 style="font-weight: 700; margin-bottom: 7px; line-height: normal;">
Audio Flamingo: A Novel Audio Language Model with Few-Shot Learning and Dialogue Abilities
</h1>
</div>
<p style="margin-bottom: 10px; font-size: 125%">
<a href="https://arxiv.org/abs/2402.01831">[Paper]</a> <a href="https://github.com/NVIDIA/audio-flamingo">[Code]</a> <a href="https://audioflamingo.github.io/">[Demo]</a>
</p>
</div>
"""
)
gr.HTML(
"""
<div>
<h3>Model Overview</h3>
Audio Flamingo is an audio language model that can understand sounds beyond speech.
It can also answer questions about the sound in natural language.
Examples of questions include:
"Can you briefly describe what you hear in this audio?",
"What is the emotion conveyed in this music?",
"Where is this audio usually heard?",
or "What place is this music usually played at?".
</div>
"""
)
name = gr.Textbox(
label="Audio file path (choose one from: audio/wav{1--6}.wav)",
value="audio/wav5.wav"
)
prompt = gr.Textbox(
label="Instruction",
value='Can you briefly describe what you hear in this audio?'
)
with gr.Row():
play_audio_button = gr.Button("Play Audio")
audio_output = gr.Audio(label="Playback")
play_audio_button.click(fn=lambda x: x, inputs=name, outputs=audio_output)
inference_button = gr.Button("Inference")
output_text = gr.Textbox(label="Audio Flamingo output")
inference_button.click(
fn=inference_item,
inputs=[name, prompt],
outputs=output_text
)
ui.queue()
ui.launch()
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