neverlib 0.2.4__py3-none-any.whl → 0.2.6__py3-none-any.whl

neverlib/.history/audio_aug/audio_aug_20250826222008.py

@@ -0,0 +1,197 @@
+# -*- coding:utf-8 -*-
+# Author:凌逆战 | Never
+# Date: 2024/9/27
+"""
+
+"""
+import random
+import numpy as np
+import soundfile as sf
+from scipy import signal
+from neverlib.utils import EPS
+from neverlib.filter import HPFilter
+
+def volume_norm(wav):
+    """
+    音量归一化
+    :param wav: (T,)
+    :return: (T,)
+    """
+    wav = wav / (np.max(np.abs(wav)) + 1e-8)
+    return wav
+
+
+def add_reverb(wav, rir, ratio=1, mode="same"):
+    """添加混响,
+    Args:
+        wav: [T, channel]
+        rir: [T, channel]
+        ratio:  0-1
+        mode: "same" for SE or "full" for kws
+    """
+    if random.random() < ratio:
+        wav = signal.fftconvolve(wav, rir, mode=mode)  # (28671, 3)
+        # note: 建议过完添加混响后再进行归一化, 否则可能会出现溢出
+        # 防止削波
+        if np.max(np.abs(wav)) > 1:
+            scale_factor = 1 / np.max(np.abs(wav))
+            wav *= scale_factor
+    return wav
+
+
+def snr_aug_changeNoise(clean, noise, target_snr, hp=False):
+    """ 
+    保持语音不变, 改变噪声的幅度
+    HP: 高通滤波, 如果你的数据工频干扰较高, 建议设置为True, 否则snr会不准
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    noise_scale = np.sqrt(clean_power / (noise_power * 10 ** (target_snr / 10) + EPS))
+    noisy = clean + noise_scale * noise
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1:
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean *= scale_factor
+    return noisy, clean
+
+
+def snr_aug_changeNoise_dB(clean, noise, target_snr, hp=False):
+    """ 
+    保持语音不变, 改变噪声的幅度
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    source_snr = 10 * np.log10(clean_power / (noise_power + EPS) + EPS)
+    noise_dB = source_snr - target_snr     # 噪声还差多少dB
+    noise_gain = 10 ** (noise_dB / 20)
+    noisy = clean + noise_gain * noise
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1:
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean *= scale_factor
+    return noisy, clean
+
+
+
+def snr_aug_changeClean(clean, noise, target_snr, clip_check=True, hp=False):
+    """ 
+    保持噪声不变，改变纯净语音的幅度以达到目标信噪比
+    snr = 10 * log10(k*signal_power/ noise_power)
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    # 计算纯净信号需要的幅度因子
+    clean_scale = np.sqrt(noise_power * 10 ** (target_snr / 10) / (clean_power + EPS))
+    noisy = clean * clean_scale + noise
+    # 防止削波
+    if clip_check:
+        if np.max(np.abs(noisy)) > 1:
+            scale_factor = 1 / np.max(np.abs(noisy))
+            noisy *= scale_factor
+            clean *= (scale_factor * clean_scale) 
+    return noisy, clean
+
+
+def snr_aug_changeClean_dB(clean, noise, target_snr, clip_check=True):
+    """ 
+    保持噪声不变, 改变纯净语音的幅度
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    source_snr = 10 * np.log10(clean_power / (noise_power + EPS) + EPS)
+    clean_dB = source_snr - target_snr     # 纯净语音还差多少dB
+    clean_gain = 10 ** (clean_dB / 20)
+    noisy = clean_gain * clean + noise
+    # 防止削波
+    if clip_check:
+        if np.max(np.abs(noisy)) > 1:
+            scale_factor = 1 / np.max(np.abs(noisy))
+            noisy *= scale_factor
+            clean *= (scale_factor * clean_gain)
+    return noisy, clean
+
+
+def snr_aug_Interpolation(clean, noise, target_snr):
+    """
+    在已知clean_len<=noise_len的情况下
+    将clean插入到noise中的snr aug方法
+    Args:
+        clean: 语音
+        noise: 噪声
+        snr: snr=random.uniform(*snr_range)
+    """
+    clean_len, noise_len = clean.shape[0], noise.shape[0]
+    assert clean_len <= noise_len, f"clean_len must be less than noise_len."
+    noisy = noise.copy()
+    index = random.randint(0, noise_len - clean_len)
+    noise = noise[index:index + clean_len, :]
+    # 这里必须把clip_check设置为False, 否则外面的noise和里面的不一致
+    noisy_tmp, clean_tmp = snr_aug_changeClean(clean, noise, target_snr, clip_check=False)
+    noisy[index:index + clean_len, :] = noisy_tmp
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1: 
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean_tmp *= scale_factor
+    return noisy, clean_tmp
+
+
+def get_audio_segments(wav_len, audio_path_list, sr=16000):
+    """
+    从音频列表中随机拼接指定长度音频
+    Args:
+        wav_len: 需要返回的音频长度
+        audio_path_list: 音频路径列表
+        sr: 采样率
+    Returns:返回指定长度的音频
+    """
+    audio_len = 0
+    wav_list = []
+    while audio_len < wav_len:
+        audio_path = random.choice(audio_path_list)
+        wav, wav_sr = sf.read(audio_path, always_2d=True, dtype='float32')
+        assert wav_sr == sr, f"音频采样率是{wav_sr}, 期望{sr}"
+        audio_len += len(wav)
+        wav_list.append(wav)
+    wav = np.concatenate(wav_list, axis=0)
+    if len(wav) > wav_len:
+        # 随机截取clean_len
+        start = random.randint(0, len(wav) - wav_len)
+        wav = wav[start:start + wav_len, :]
+    return wav
+
+
+def volume_aug(wav, range, rate, method="linmax"):
+    """音量增强 """
+    if random.random() < rate:
+        target_level = random.uniform(range[0], range[1])
+        if method == "dbrms":
+            wav_rms = (wav ** 2).mean() ** 0.5
+            scalar = 10 ** (target_level / 20) / (np.max(wav_rms) + EPS)
+        elif method == "linmax":
+            ipt_max = np.max(np.abs(wav))
+            # wav/wav_max*target_level=target_level_wav
+            # 处理后音频的 最大值就是target_level
+            scalar = target_level / (ipt_max + EPS)
+        else:
+            raise ValueError("method must be 'dbrms' or 'linmax'")
+        wav *= scalar
+    return wav

neverlib/.history/audio_aug/audio_aug_20250826222017.py

@@ -0,0 +1,200 @@
+# -*- coding:utf-8 -*-
+# Author:凌逆战 | Never
+# Date: 2024/9/27
+"""
+
+"""
+import random
+import numpy as np
+import soundfile as sf
+from scipy import signal
+from neverlib.utils import EPS
+from neverlib.filter import HPFilter
+
+def volume_norm(wav):
+    """
+    音量归一化
+    :param wav: (T,)
+    :return: (T,)
+    """
+    wav = wav / (np.max(np.abs(wav)) + 1e-8)
+    return wav
+
+
+def add_reverb(wav, rir, ratio=1, mode="same"):
+    """添加混响,
+    Args:
+        wav: [T, channel]
+        rir: [T, channel]
+        ratio:  0-1
+        mode: "same" for SE or "full" for kws
+    """
+    if random.random() < ratio:
+        wav = signal.fftconvolve(wav, rir, mode=mode)  # (28671, 3)
+        # note: 建议过完添加混响后再进行归一化, 否则可能会出现溢出
+        # 防止削波
+        if np.max(np.abs(wav)) > 1:
+            scale_factor = 1 / np.max(np.abs(wav))
+            wav *= scale_factor
+    return wav
+
+
+def snr_aug_changeNoise(clean, noise, target_snr, hp=False):
+    """ 
+    保持语音不变, 改变噪声的幅度
+    HP: 高通滤波, 如果你的数据工频干扰较高, 建议设置为True, 否则snr会不准
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    noise_scale = np.sqrt(clean_power / (noise_power * 10 ** (target_snr / 10) + EPS))
+    noisy = clean + noise_scale * noise
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1:
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean *= scale_factor
+    return noisy, clean
+
+
+def snr_aug_changeNoise_dB(clean, noise, target_snr, hp=False):
+    """ 
+    保持语音不变, 改变噪声的幅度
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    source_snr = 10 * np.log10(clean_power / (noise_power + EPS) + EPS)
+    noise_dB = source_snr - target_snr     # 噪声还差多少dB
+    noise_gain = 10 ** (noise_dB / 20)
+    noisy = clean + noise_gain * noise
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1:
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean *= scale_factor
+    return noisy, clean
+
+
+
+def snr_aug_changeClean(clean, noise, target_snr, clip_check=True, hp=False):
+    """ 
+    保持噪声不变，改变纯净语音的幅度以达到目标信噪比
+    snr = 10 * log10(k*signal_power/ noise_power)
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    # 计算纯净信号需要的幅度因子
+    clean_scale = np.sqrt(noise_power * 10 ** (target_snr / 10) / (clean_power + EPS))
+    noisy = clean * clean_scale + noise
+    # 防止削波
+    if clip_check:
+        if np.max(np.abs(noisy)) > 1:
+            scale_factor = 1 / np.max(np.abs(noisy))
+            noisy *= scale_factor
+            clean *= (scale_factor * clean_scale) 
+    return noisy, clean
+
+
+def snr_aug_changeClean_dB(clean, noise, target_snr, clip_check=True, hp=False):
+    """ 
+    保持噪声不变, 改变纯净语音的幅度
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    source_snr = 10 * np.log10(clean_power / (noise_power + EPS) + EPS)
+    clean_dB = source_snr - target_snr     # 纯净语音还差多少dB
+    clean_gain = 10 ** (clean_dB / 20)
+    noisy = clean_gain * clean + noise
+    # 防止削波
+    if clip_check:
+        if np.max(np.abs(noisy)) > 1:
+            scale_factor = 1 / np.max(np.abs(noisy))
+            noisy *= scale_factor
+            clean *= (scale_factor * clean_gain)
+    return noisy, clean
+
+
+def snr_aug_Interpolation(clean, noise, target_snr):
+    """
+    在已知clean_len<=noise_len的情况下
+    将clean插入到noise中的snr aug方法
+    Args:
+        clean: 语音
+        noise: 噪声
+        snr: snr=random.uniform(*snr_range)
+    """
+    clean_len, noise_len = clean.shape[0], noise.shape[0]
+    assert clean_len <= noise_len, f"clean_len must be less than noise_len."
+    noisy = noise.copy()
+    index = random.randint(0, noise_len - clean_len)
+    noise = noise[index:index + clean_len, :]
+    # 这里必须把clip_check设置为False, 否则外面的noise和里面的不一致
+    noisy_tmp, clean_tmp = snr_aug_changeClean(clean, noise, target_snr, clip_check=False)
+    noisy[index:index + clean_len, :] = noisy_tmp
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1: 
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean_tmp *= scale_factor
+    return noisy, clean_tmp
+
+
+def get_audio_segments(wav_len, audio_path_list, sr=16000):
+    """
+    从音频列表中随机拼接指定长度音频
+    Args:
+        wav_len: 需要返回的音频长度
+        audio_path_list: 音频路径列表
+        sr: 采样率
+    Returns:返回指定长度的音频
+    """
+    audio_len = 0
+    wav_list = []
+    while audio_len < wav_len:
+        audio_path = random.choice(audio_path_list)
+        wav, wav_sr = sf.read(audio_path, always_2d=True, dtype='float32')
+        assert wav_sr == sr, f"音频采样率是{wav_sr}, 期望{sr}"
+        audio_len += len(wav)
+        wav_list.append(wav)
+    wav = np.concatenate(wav_list, axis=0)
+    if len(wav) > wav_len:
+        # 随机截取clean_len
+        start = random.randint(0, len(wav) - wav_len)
+        wav = wav[start:start + wav_len, :]
+    return wav
+
+
+def volume_aug(wav, range, rate, method="linmax"):
+    """音量增强 """
+    if random.random() < rate:
+        target_level = random.uniform(range[0], range[1])
+        if method == "dbrms":
+            wav_rms = (wav ** 2).mean() ** 0.5
+            scalar = 10 ** (target_level / 20) / (np.max(wav_rms) + EPS)
+        elif method == "linmax":
+            ipt_max = np.max(np.abs(wav))
+            # wav/wav_max*target_level=target_level_wav
+            # 处理后音频的 最大值就是target_level
+            scalar = target_level / (ipt_max + EPS)
+        else:
+            raise ValueError("method must be 'dbrms' or 'linmax'")
+        wav *= scalar
+    return wav

neverlib/.history/audio_aug/audio_aug_20250826222046.py

@@ -0,0 +1,203 @@
+# -*- coding:utf-8 -*-
+# Author:凌逆战 | Never
+# Date: 2024/9/27
+"""
+
+"""
+import random
+import numpy as np
+import soundfile as sf
+from scipy import signal
+from neverlib.utils import EPS
+from neverlib.filter import HPFilter
+
+def volume_norm(wav):
+    """
+    音量归一化
+    :param wav: (T,)
+    :return: (T,)
+    """
+    wav = wav / (np.max(np.abs(wav)) + 1e-8)
+    return wav
+
+
+def add_reverb(wav, rir, ratio=1, mode="same"):
+    """添加混响,
+    Args:
+        wav: [T, channel]
+        rir: [T, channel]
+        ratio:  0-1
+        mode: "same" for SE or "full" for kws
+    """
+    if random.random() < ratio:
+        wav = signal.fftconvolve(wav, rir, mode=mode)  # (28671, 3)
+        # note: 建议过完添加混响后再进行归一化, 否则可能会出现溢出
+        # 防止削波
+        if np.max(np.abs(wav)) > 1:
+            scale_factor = 1 / np.max(np.abs(wav))
+            wav *= scale_factor
+    return wav
+
+
+def snr_aug_changeNoise(clean, noise, target_snr, hp=False):
+    """ 
+    保持语音不变, 改变噪声的幅度
+    HP: 高通滤波, 如果你的数据工频干扰较高, 建议设置为True, 否则snr会不准
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    noise_scale = np.sqrt(clean_power / (noise_power * 10 ** (target_snr / 10) + EPS))
+    noisy = clean + noise_scale * noise
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1:
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean *= scale_factor
+    return noisy, clean
+
+
+def snr_aug_changeNoise_dB(clean, noise, target_snr, hp=False):
+    """ 
+    保持语音不变, 改变噪声的幅度
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    source_snr = 10 * np.log10(clean_power / (noise_power + EPS) + EPS)
+    noise_dB = source_snr - target_snr     # 噪声还差多少dB
+    noise_gain = 10 ** (noise_dB / 20)
+    noisy = clean + noise_gain * noise
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1:
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean *= scale_factor
+    return noisy, clean
+
+
+
+def snr_aug_changeClean(clean, noise, target_snr, clip_check=True, hp=False):
+    """ 
+    保持噪声不变，改变纯净语音的幅度以达到目标信噪比
+    snr = 10 * log10(k*signal_power/ noise_power)
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    # 计算纯净信号需要的幅度因子
+    clean_scale = np.sqrt(noise_power * 10 ** (target_snr / 10) / (clean_power + EPS))
+    noisy = clean * clean_scale + noise
+    # 防止削波
+    if clip_check:
+        if np.max(np.abs(noisy)) > 1:
+            scale_factor = 1 / np.max(np.abs(noisy))
+            noisy *= scale_factor
+            clean *= (scale_factor * clean_scale) 
+    return noisy, clean
+
+
+def snr_aug_changeClean_dB(clean, noise, target_snr, clip_check=True, hp=False):
+    """ 
+    保持噪声不变, 改变纯净语音的幅度
+    snr = 10 * log10(signal_power / k*noise_power) 
+    """
+    assert clean.shape == noise.shape, "clean and noise must have the same shape"
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    # 纯净语音功率, 噪声功率
+    clean_power, noise_power = np.mean(clean ** 2), np.mean(noise ** 2)  
+    source_snr = 10 * np.log10(clean_power / (noise_power + EPS) + EPS)
+    clean_dB = source_snr - target_snr     # 纯净语音还差多少dB
+    clean_gain = 10 ** (clean_dB / 20)
+    noisy = clean_gain * clean + noise
+    # 防止削波
+    if clip_check:
+        if np.max(np.abs(noisy)) > 1:
+            scale_factor = 1 / np.max(np.abs(noisy))
+            noisy *= scale_factor
+            clean *= (scale_factor * clean_gain)
+    return noisy, clean
+
+
+def snr_aug_Interpolation(clean, noise, target_snr, hp=False):
+    """
+    在已知clean_len<=noise_len的情况下
+    将clean插入到noise中的snr aug方法
+    Args:
+        clean: 语音
+        noise: 噪声
+        snr: snr=random.uniform(*snr_range)
+    """
+    clean_len, noise_len = clean.shape[0], noise.shape[0]
+    if hp:
+        clean = HPFilter(clean, sr=16000, order=6, cutoff=100)
+        noise = HPFilter(noise, sr=16000, order=6, cutoff=100)
+    assert clean_len <= noise_len, f"clean_len must be less than noise_len."
+    noisy = noise.copy()
+    index = random.randint(0, noise_len - clean_len)
+    noise = noise[index:index + clean_len, :]
+    # 这里必须把clip_check设置为False, 否则外面的noise和里面的不一致
+    noisy_tmp, clean_tmp = snr_aug_changeClean(clean, noise, target_snr, clip_check=False, hp=False)
+    noisy[index:index + clean_len, :] = noisy_tmp
+    # 防止削波
+    if np.max(np.abs(noisy)) > 1: 
+        scale_factor = 1 / np.max(np.abs(noisy))
+        noisy *= scale_factor
+        clean_tmp *= scale_factor
+    return noisy, clean_tmp
+
+
+def get_audio_segments(wav_len, audio_path_list, sr=16000):
+    """
+    从音频列表中随机拼接指定长度音频
+    Args:
+        wav_len: 需要返回的音频长度
+        audio_path_list: 音频路径列表
+        sr: 采样率
+    Returns:返回指定长度的音频
+    """
+    audio_len = 0
+    wav_list = []
+    while audio_len < wav_len:
+        audio_path = random.choice(audio_path_list)
+        wav, wav_sr = sf.read(audio_path, always_2d=True, dtype='float32')
+        assert wav_sr == sr, f"音频采样率是{wav_sr}, 期望{sr}"
+        audio_len += len(wav)
+        wav_list.append(wav)
+    wav = np.concatenate(wav_list, axis=0)
+    if len(wav) > wav_len:
+        # 随机截取clean_len
+        start = random.randint(0, len(wav) - wav_len)
+        wav = wav[start:start + wav_len, :]
+    return wav
+
+
+def volume_aug(wav, range, rate, method="linmax"):
+    """音量增强 """
+    if random.random() < rate:
+        target_level = random.uniform(range[0], range[1])
+        if method == "dbrms":
+            wav_rms = (wav ** 2).mean() ** 0.5
+            scalar = 10 ** (target_level / 20) / (np.max(wav_rms) + EPS)
+        elif method == "linmax":
+            ipt_max = np.max(np.abs(wav))
+            # wav/wav_max*target_level=target_level_wav
+            # 处理后音频的 最大值就是target_level
+            scalar = target_level / (ipt_max + EPS)
+        else:
+            raise ValueError("method must be 'dbrms' or 'linmax'")
+        wav *= scalar
+    return wav