python PaddleSpeech实现婴儿啼哭识别

目录

• 一、基于PaddleSpeech的婴儿啼哭识别
• 1.项目背景
• 2.数据说明：
• 二、PaddleSpeech环境准备
• 三、数据预处理
• 1.数据解压缩
• 2.查看声音文件
• 3.音频文件长度处理
• 3.自定义数据集
• 四、模型训练
• 1.选取预训练模型
• 2.构建分类模型
• 3.finetune
• 五、模型训练
• 六、注意事项

一、基于PaddleSpeech的婴儿啼哭识别

1.项目背景

对婴儿来说，啼哭声是一种通讯的方式，一个非常有限的，但类似成年人进行交流的方式。它也是一种生物报警器，向外界传达着婴儿生理和心理的需求。基于啼哭声声波携带的信息，婴儿的身体状况才能被确定，疾病才能被检测出来。因此，有效辨识啼哭声，成功地将婴儿啼哭声“翻译”成“成人语言”，让我们能够读懂啼哭声的含义，有重大的实际意义。

2.数据说明：

• 1.训练数据集包含六类哭声，已人工添加噪声。

A：awake（苏醒）

B：diaper（换尿布）

C：hug（要抱抱）

D：hungry（饥饿）

E：sleepy（困乏）

F：uncomfortable（不舒服）

• 2.噪声数据来源Noisex-92标准数据库。

二、PaddleSpeech环境准备

# 环境准备：安装paddlespeech和paddleaudio
!python -m pip install -q -U pip --user
!pip install paddlespeech paddleaudio -U -q

!pip list|grep paddle

import warnings
warnings.filterwarnings("ignore")
import IPython
import numpy as np
import matplotlib.pyplot as plt
import paddle
%matplotlib inline

三、数据预处理

1.数据解压缩

# !unzip -qoa data/data41960/dddd.zip

2.查看声音文件

from paddleaudio import load
data, sr = load(file='train/awake/awake_0.wav', mono=True, dtype='float32')  # 单通道，float32音频样本点
print('wav shape: {}'.format(data.shape))
print('sample rate: {}'.format(sr))
# 展示音频波形
plt.figure()
plt.plot(data)
plt.show()

from paddleaudio import load
data, sr = load(file='train/diaper/diaper_0.wav', mono=True, dtype='float32')  # 单通道，float32音频样本点
print('wav shape: {}'.format(data.shape))
print('sample rate: {}'.format(sr))
# 展示音频波形
plt.figure()
plt.plot(data)
plt.show()

!paddlespeech cls --input train/awake/awake_0.wav

!paddlespeech help

3.音频文件长度处理

# 查音频长度
import contextlib
import wave
def get_sound_len(file_path):
    with contextlib.closing(wave.open(file_path, 'r')) as f:
        frames = f.getnframes()
        rate = f.getframerate()
        wav_length = frames / float(rate)
    return wav_length

# 编译wav文件
import glob
sound_files=glob.glob('train/*/*.wav')
print(sound_files[0])
print(len(sound_files))

# 统计最长、最短音频
sounds_len=[]
for sound in sound_files:
    sounds_len.append(get_sound_len(sound))
print("音频最大长度：",max(sounds_len),"秒")
print("音频最小长度：",min(sounds_len),"秒")

!cp train/hungry/hungry_0.wav ~/

!pip install pydub -q

# 音频信息查看
import math
import soundfile as sf
import numpy as np
import librosa
data, samplerate = sf.read('hungry_0.wav')
channels = len(data.shape)
length_s = len(data)/float(samplerate)
format_rate=16000
print(f"channels: {channels}")
print(f"length_s: {length_s}")
print(f"samplerate: {samplerate}")

# 统一到34s
from pydub import AudioSegment
audio = AudioSegment.from_wav('hungry_0.wav')
print(str(audio.duration_seconds))
i = 1
padded = audio
while padded.duration_seconds * 1000 < 34000:
    padded = audio * i
    i = i + 1
padded[0:34000].set_frame_rate(16000).export('padded-file.wav', format='wav')

import math
import soundfile as sf
import numpy as np
import librosa
data, samplerate = sf.read('padded-file.wav')
channels = len(data.shape)
length_s = len(data)/float(samplerate)
format_rate=16000
print(f"channels: {channels}")
print(f"length_s: {length_s}")
print(f"samplerate: {samplerate}")

# 定义函数，如未达到最大长度，则重复填充，最终从超过34s的音频中截取
from pydub import AudioSegment
def convert_sound_len(filename):
    audio = AudioSegment.from_wav(filename)
    i = 1
    padded = audio*i
    while padded.duration_seconds * 1000 < 34000:
        i = i + 1
        padded = audio * i
    padded[0:34000].set_frame_rate(16000).export(filename, format='wav')

# 统一所有音频到定长
for sound in sound_files:
    convert_sound_len(sound)

3.自定义数据集

import os
from paddlespeech.audio.datasets.dataset import AudioClassificationDataset
class CustomDataset(AudioClassificationDataset):
    # List all the class labels
    label_list = [
        'awake',
        'diaper',
        'hug',
        'hungry',
        'sleepy',
        'uncomfortable'
    ]
    train_data_dir='./train/'
    def __init__(self, **kwargs):
        files, labels = self._get_data()
        super(CustomDataset, self).__init__(
            files=files, labels=labels, feat_type='raw', **kwargs)
    # 返回音频文件、label值
    def _get_data(self):
        '''
        This method offer information of wave files and labels.
        '''
        files = []
        labels = []
        for i in range(len(self.label_list)):
            single_class_path=os.path.join(self.train_data_dir, self.label_list[i])            
            for sound in os.listdir(single_class_path):
                # print(sound)
                if 'wav' in sound:
                    sound=os.path.join(single_class_path, sound)
                    files.append(sound)
                    labels.append(i)
        return files, labels

# 定义dataloader
import paddle
from paddlespeech.audio.features import LogMelSpectrogram
# Feature config should be align with pretrained model
sample_rate = 16000
feat_conf = {
  'sr': sample_rate,
  'n_fft': 1024,
  'hop_length': 320,
  'window': 'hann',
  'win_length': 1024,
  'f_min': 50.0,
  'f_max': 14000.0,
  'n_mels': 64,
}
train_ds = CustomDataset(sample_rate=sample_rate)
feature_extractor = LogMelSpectrogram(**feat_conf)
train_sampler = paddle.io.DistributedBatchSampler(
    train_ds, batch_size=64, shuffle=True, drop_last=False)
train_loader = paddle.io.DataLoader(
    train_ds,
    batch_sampler=train_sampler,
    return_list=True,
    use_buffer_reader=True)

四、模型训练

1.选取预训练模型

选取cnn14作为 backbone，用于提取音频的特征：

from paddlespeech.cls.models import cnn14
backbone = cnn14(pretrained=True, extract_embedding=True)

2.构建分类模型

SoundClassifer接收cnn14作为backbone模型，并创建下游的分类网络：

import paddle.nn as nn
class SoundClassifier(nn.Layer):
    def __init__(self, backbone, num_class, dropout=0.1):
        super().__init__()
        self.backbone = backbone
        self.dropout = nn.Dropout(dropout)
        self.fc = nn.Linear(self.backbone.emb_size, num_class)
    def forward(self, x):
        x = x.unsqueeze(1)
        x = self.backbone(x)
        x = self.dropout(x)
        logits = self.fc(x)
        return logits
model = SoundClassifier(backbone, num_class=len(train_ds.label_list))

3.finetune

# 定义优化器和 Loss
optimizer = paddle.optimizer.Adam(learning_rate=1e-4, parameters=model.parameters())
criterion = paddle.nn.loss.CrossEntropyLoss()

from paddleaudio.utils import logger
epochs = 20
steps_per_epoch = len(train_loader)
log_freq = 10
eval_freq = 10
for epoch in range(1, epochs + 1):
    model.train()
    avg_loss = 0
    num_corrects = 0
    num_samples = 0
    for batch_idx, batch in enumerate(train_loader):
        waveforms, labels = batch
        feats = feature_extractor(waveforms)
        feats = paddle.transpose(feats, [0, 2, 1])  # [B, N, T] -> [B, T, N]
        logits = model(feats)
        loss = criterion(logits, labels)
        loss.backward()
        optimizer.step()
        if isinstance(optimizer._learning_rate,
                      paddle.optimizer.lr.LRScheduler):
            optimizer._learning_rate.step()
        optimizer.clear_grad()
        # Calculate loss
        avg_loss += loss.numpy()[0]
        # Calculate metrics
        preds = paddle.argmax(logits, axis=1)
        num_corrects += (preds == labels).numpy().sum()
        num_samples += feats.shape[0]
        if (batch_idx + 1) % log_freq == 0:
            lr = optimizer.get_lr()
            avg_loss /= log_freq
            avg_acc = num_corrects / num_samples
            print_msg = 'Epoch={}/{}, Step={}/{}'.format(
                epoch, epochs, batch_idx + 1, steps_per_epoch)
            print_msg += ' loss={:.4f}'.format(avg_loss)
            print_msg += ' acc={:.4f}'.format(avg_acc)
            print_msg += ' lr={:.6f}'.format(lr)
            logger.train(print_msg)
            avg_loss = 0
            num_corrects = 0
            num_samples = 0

[2022-08-24 02:20:49,381] [   TRAIN] - Epoch=17/20, Step=10/15 loss=1.3319 acc=0.4875 lr=0.000100
[2022-08-24 02:21:08,107] [   TRAIN] - Epoch=18/20, Step=10/15 loss=1.3222 acc=0.4719 lr=0.000100
[2022-08-24 02:21:08,107] [   TRAIN] - Epoch=18/20, Step=10/15 loss=1.3222 acc=0.4719 lr=0.000100
[2022-08-24 02:21:26,884] [   TRAIN] - Epoch=19/20, Step=10/15 loss=1.2539 acc=0.5125 lr=0.000100
[2022-08-24 02:21:26,884] [   TRAIN] - Epoch=19/20, Step=10/15 loss=1.2539 acc=0.5125 lr=0.000100
[2022-08-24 02:21:45,579] [   TRAIN] - Epoch=20/20, Step=10/15 loss=1.2021 acc=0.5281 lr=0.000100
[2022-08-24 02:21:45,579] [   TRAIN] - Epoch=20/20, Step=10/15 loss=1.2021 acc=0.5281 lr=0.000100 

五、模型训练

top_k = 3
wav_file = 'test/test_0.wav'
n_fft = 1024
win_length = 1024
hop_length = 320
f_min=50.0
f_max=16000.0
waveform, sr = load(wav_file, sr=sr)
feature_extractor = LogMelSpectrogram(
    sr=sr, 
    n_fft=n_fft, 
    hop_length=hop_length, 
    win_length=win_length, 
    window='hann', 
    f_min=f_min, 
    f_max=f_max, 
    n_mels=64)
feats = feature_extractor(paddle.to_tensor(paddle.to_tensor(waveform).unsqueeze(0)))
feats = paddle.transpose(feats, [0, 2, 1])  # [B, N, T] -> [B, T, N]
logits = model(feats)
probs = nn.functional.softmax(logits, axis=1).numpy()
sorted_indices = probs[0].argsort()
msg = f'[{wav_file}]\n'
for idx in sorted_indices[-1:-top_k-1:-1]:
    msg += f'{train_ds.label_list[idx]}: {probs[0][idx]:.5f}\n'
print(msg)    

 [test/test_0.wav]
diaper: 0.50155
sleepy: 0.41397
hug: 0.05912

六、注意事项

• 1.自定义数据集，格式可参考文档；
• 2.统一音频尺寸（例如音频长度、采样频率）

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