neverlib 0.2.8__py3-none-any.whl → 0.3.0__py3-none-any.whl

neverlib/.history/audio_aug/audio_aug_20250827172559.py

@@ -1,470 +0,0 @@
-# -*- 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 limiter(wav, threshold=0.999):
-    """
-    简单限幅器, threshold=0.999 ~ -0.1dBFS
-    超过阈值的样本被压缩到阈值限制
-    """
-    peak = np.max(np.abs(wav))
-    if peak > threshold:
-        scalar = threshold / (peak + EPS)
-        wav = wav * scalar
-    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, hpf=False, sr=16000, order=4, cutoff=100):
-    """ 
-    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"
-    clean_tmp, noise_tmp = clean.copy(), noise.copy()
-    if hpf:
-        clean_tmp = HPFilter(clean_tmp, sr=sr, order=order, cutoff=cutoff)
-        noise_tmp = HPFilter(noise_tmp, sr=sr, order=order, cutoff=cutoff)
-    # 纯净语音功率, 噪声功率
-    clean_power, noise_power = np.mean(clean_tmp ** 2), np.mean(noise_tmp ** 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_changeClean(clean, noise, target_snr, clip_check=True, hpf=False, sr=16000, order=4, cutoff=100):
-    """ 
-    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"
-    clean_tmp, noise_tmp = clean.copy(), noise.copy()
-    if hpf:
-        clean_tmp = HPFilter(clean_tmp, sr=sr, order=order, cutoff=cutoff)
-        noise_tmp = HPFilter(noise_tmp, sr=sr, order=order, cutoff=cutoff)
-    # 纯净语音功率, 噪声功率
-    clean_power, noise_power = np.mean(clean_tmp ** 2), np.mean(noise_tmp ** 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_diff_changeNoise(clean, noise, target_snr, hpf=False, sr=16000, order=4, cutoff=100):
-    """ 
-    保持语音不变, 改变噪声的幅度, 和snr_aug_changeNoise作用等效
-    snr = 10 * log10(signal_power / k*noise_power) 
-    """
-    assert clean.shape == noise.shape, "clean and noise must have the same shape"
-    clean_tmp, noise_tmp = clean.copy(), noise.copy()
-    if hpf:
-        clean_tmp = HPFilter(clean_tmp, sr=sr, order=order, cutoff=cutoff)
-        noise_tmp = HPFilter(noise_tmp, sr=sr, order=order, cutoff=cutoff)
-    # 纯净语音功率, 噪声功率
-    clean_power, noise_power = np.mean(clean_tmp ** 2), np.mean(noise_tmp ** 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_diff_changeClean(clean, noise, target_snr, clip_check=True, hpf=False, sr=16000, order=4, cutoff=100):
-    """ 
-    保持噪声不变, 改变纯净语音的幅度, 和snr_aug_changeClean作用等效
-    snr = 10 * log10(signal_power / k*noise_power) 
-    """
-    assert clean.shape == noise.shape, "clean and noise must have the same shape"
-    clean_tmp, noise_tmp = clean.copy(), noise.copy()
-    if hpf:
-        clean_tmp = HPFilter(clean_tmp, sr=sr, order=order, cutoff=cutoff)
-        noise_tmp = HPFilter(noise_tmp, sr=sr, order=order, cutoff=cutoff)
-    # 纯净语音功率, 噪声功率
-    clean_power, noise_power = np.mean(clean_tmp ** 2), np.mean(noise_tmp ** 2)
-    source_snr = 10 * np.log10(clean_power / (noise_power + EPS) + EPS)
-    clean_dB = target_snr - source_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, hpf=False, sr=16000, order=4, cutoff=100):
-    """
-    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."
-    clean_tmp, noise_tmp = clean.copy(), noise.copy()
-    if hpf:
-        clean_tmp = HPFilter(clean_tmp, sr=sr, order=order, cutoff=cutoff)
-        noise_tmp = HPFilter(noise_tmp, sr=sr, order=order, cutoff=cutoff)
-    noisy = noise.copy()
-    index = random.randint(0, noise_len - clean_len)
-    noise_tmp = noise_tmp[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_tmp, noise_tmp, target_snr, clip_check=False, hpf=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 *= scale_factor
-    return noisy, np.pad(clean, ((index, noise_len - index - clean_len), (0, 0)))
-
-
-def get_snr_use_vad(wav, vad, sr=16000):
-    # 通过vad获得语音原始的snr
-    wav = HPFilter(wav, sr=sr, order=6, cutoff=100)
-    vadstart, vadend = vad["start"], vad["end"]
-    noise = np.concatenate([wav[:vadstart], wav[vadend:]], axis=0)
-    speech_segment = wav[vadstart:vadend]
-
-    # 计算信噪比
-    # 统计语音段的均方功率谱
-    P_speech_noise = np.mean(speech_segment ** 2)  # 语音+噪声功率
-    # P_speech_noise = np.mean(wav ** 2)  # 如果用全局的, 会存在噪声功率过大的问题, 导致snr过低
-    P_noise = np.mean(noise ** 2)  # 纯噪声功率
-
-    # 计算净语音功率(确保非负)
-    P_speech = max(P_speech_noise - P_noise, 1e-8)
-    if P_noise < 1e-8:
-        P_noise = 1e-8
-
-    snr = 10 * np.log10(P_speech / P_noise)  # 计算 SNR
-    # snr保留小数点后一位
-    return round(snr, 1)
-
-
-def snr_aug_vad_Interpolation(clean, noise, target_snr, vad, hpf=False, sr=16000, order=4, cutoff=100):
-    """
-    在已知clean_len<=noise_len的情况下, 将clean插入到noise中的snr aug方法,
-    使用VAD信息, 精确地找到语音位置
-    Args:
-        clean: 语音
-        noise: 噪声
-        vad: {"start": xxx, "end": xxx}
-    """
-    clean_len, noise_len = clean.shape[0], noise.shape[0]
-    assert clean_len <= noise_len, f"clean_len must be less than noise_len."
-    clean_tmp, noise_tmp = clean.copy(), noise.copy()
-    noisy = noise.copy()
-    index = random.randint(0, noise_len - clean_len)
-    noise = noise[index:index + clean_len, :]  # 现在语音和噪声长度一致
-    if hpf:
-        clean_tmp = HPFilter(clean_tmp, sr=sr, order=order, cutoff=cutoff)
-        noise_tmp = HPFilter(noise_tmp, sr=sr, order=order, cutoff=cutoff)
-    # 只根据语音段求SNR
-    clean_vad = clean_tmp[vad["start"]:vad["end"]]
-    noise_tmp = noise_tmp[vad["start"]:vad["end"]]
-    power_clean, power_noise = np.mean(clean_vad ** 2), np.mean(noise_tmp ** 2)
-    snr_in = 10 * np.log10(power_clean / (power_noise + EPS) + EPS)
-    clean_dB = target_snr - snr_in  # 语音还差多少dB
-    # noise_dB = snr_in - target_snr    # 噪声还差多少dB
-    clean_gain = 10 ** (clean_dB / 20)
-    noisy_tmp = clean_gain * clean + noise
-
-    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 *= (scale_factor * clean_gain)
-    return noisy, np.pad(clean, ((index, noise_len - index - clean_len), (0, 0)))
-
-
-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
-
-
-# ----------------------------------------------------------------
-# 音量增强
-# ----------------------------------------------------------------
-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 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))
-            scalar = target_level / (ipt_max + EPS)
-        else:
-            raise ValueError("method must be 'dbrms' or 'linmax'")
-        wav *= scalar
-    return wav
-
-
-def volume_aug_dbrms(wav, target_level, hpf=False, sr=16000, order=4, cutoff=100):
-    """
-    音量增强, 使用dbrms方法
-    Args:
-        wav: 音频
-        target_level: 目标音量, 单位dB
-        hpf: 是否高通滤波
-        sr: 采样率
-        order: 滤波器阶数
-        cutoff: 截止频率
-    """
-    wav_tmp = wav.copy()
-    if hpf:
-        wav_tmp = HPFilter(wav_tmp, sr=sr, order=order, cutoff=cutoff)
-    wav_rms = (wav_tmp ** 2).mean() ** 0.5
-    scalar = 10 ** (target_level / 20) / (np.max(wav_rms) + EPS)
-    wav_opt = wav_tmp * scalar
-    wav_opt = limiter(wav_opt)
-    return wav_opt
-
-
-def volume_aug_linmax(wav, target_level, hpf=False, sr=16000, order=4, cutoff=100):
-    """
-    音量增强, 使用linmax方法
-    Args:
-        wav: 音频
-        target_level: 目标音量, 单位dB
-        hpf: 是否高通滤波
-        sr: 采样率
-        order: 滤波器阶数
-        cutoff: 截止频率
-    """
-    assert target_level > 0 and target_level < 1, "target_level must be between 0 and 1"
-    wav_tmp = wav.copy()
-    if hpf:
-        wav_tmp = HPFilter(wav_tmp, sr=sr, order=order, cutoff=cutoff)
-    wav_max = np.max(np.abs(wav_tmp))
-    scalar = target_level / (wav_max + EPS)
-    wav_opt = wav_tmp * scalar
-    wav_opt = limiter(wav_opt)
-    return wav_opt
-
-
-import pyloudnorm as pyln
-
-def volume_aug_lufs(wav, target_lufs, hpf=False, sr=16000, order=4, cutoff=100):
-    """
-    音量增强, 使用lufs方法, 
-    LUFS是“感知响度” → 跟人耳听感对齐，而且符合国际响度标准。
-    
-    LUFS 使用 感知加权（K-weighting）
-    - 高频增强（模拟人耳在 3~6kHz 的敏感）
-    - 低频衰减（降低 <100Hz 对响度的影响）。
-    使用 短时块（400ms）能量 + 响度门限（-70 LUFS） 过滤极静音段。
-    
-    Args:
-        wav: 音频
-        target_lufs: 目标音量, 单位lufs
-        hpf: 是否高通滤波
-        sr: 采样率
-        order: 滤波器阶数
-        cutoff: 截止频率
-        
-    补充信息: 
-    ## 推荐的 target_lufs 值（行业参考）
-    平台	推荐目标 LUFS
-    YouTube / Spotify	-14
-    Apple Music	-16
-    广播 / TV	-23
-    游戏音频	-16 ~ -18
-    有声书	-18 ~ -20
-    """
-    wav_tmp = wav.copy()
-    if hpf:
-        wav_tmp = HPFilter(wav_tmp, sr=sr, order=4, cutoff=1000)
-
-    # Step2: 创建 LUFS 测量器（ITU-R BS.1770）
-    meter = pyln.Meter(sr, block_size=0.400)  # block_size=400ms
-
-    # Step3: 测量当前 LUFS
-    loudness = meter.integrated_loudness(wav_tmp)   
-
-    # Step4: 计算增益并应用
-    loudness_diff = target_lufs - loudness
-    scalar = 10 ** (loudness_diff / 20.0)
-    wav = wav * scalar
-
-    wav = limiter(wav, threshold=0.999) # Step5: 限幅
-    return wav
-
-def measure_loudness(wav, sr):
-    """
-    测量音频的 Peak / RMS / LUFS，以及峰均比（Crest Factor）
-
-    参数:
-        wav: np.ndarray, 音频波形（范围 [-1, 1]）
-        sr: int, 采样率
-
-    返回:
-        dict:
-            - peak_dbfs: 峰值(dBFS)
-            - rms_dbfs: 均方根电平(dBFS)
-            - lufs: 感知响度（LUFS，ITU-R BS.1770-4标准）
-            - crest_factor_db: 峰均比(dB)，峰值与RMS的差值
-    """
-    EPS = 1e-9
-
-    # 1. Peak
-    peak_linear = np.max(np.abs(wav))
-    peak_dbfs = 20 * np.log10(peak_linear + EPS)
-
-    # 2. RMS
-    rms_val = np.sqrt(np.mean(wav ** 2))
-    rms_dbfs = 20 * np.log10(rms_val + EPS)
-
-    # 3. LUFS
-    meter = pyln.Meter(sr, block_size=0.400)  # 400ms 块
-    loudness_lufs = meter.integrated_loudness(wav)
-
-    # 4. Crest Factor (峰均比)
-    crest_factor_db = peak_dbfs - rms_dbfs
-
-    return {
-        "peak_dbfs": peak_dbfs,
-        "rms_dbfs": rms_dbfs,
-        "lufs": loudness_lufs,
-        "crest_factor_db": crest_factor_db
-    }
-    
-    
-def volume_convert(value, 
-                   from_unit="linear", to_unit="dBFS", 
-                   crest_factor_db=None, lufs_offset=None):
-    """
-    音量单位转换函数
-
-    参数:
-        value: float
-            输入值（可能是线性幅度、dBFS、LUFS）
-        from_unit: str
-            输入单位: "linear", "dBFS", "RMS_dBFS", "LUFS"
-        to_unit: str
-            输出单位: "linear", "dBFS", "RMS_dBFS", "LUFS"
-        crest_factor_db: float | None
-            峰均比（用于 Peak <-> RMS 的转换）
-        lufs_offset: float | None
-            LUFS 与 RMS 的差值（用于 RMS <-> LUFS 转换）
-            例如, 对人声，LUFS ≈ RMS_dBFS - 1.5
-
-    返回:
-        float
-    """
-    EPS = 1e-9
-
-    # Step 1: 统一转换成线性幅度（以满刻度 1.0 为基准）
-    if from_unit == "linear":
-        lin_val = value
-    elif from_unit in ("dBFS", "Peak_dBFS"):
-        lin_val = 10 ** (value / 20)
-    elif from_unit == "RMS_dBFS":
-        lin_val = 10 ** (value / 20)
-    elif from_unit == "LUFS":
-        if lufs_offset is None:
-            raise ValueError("LUFS ↔ RMS 转换需要提供 lufs_offset")
-        rms_dbfs = value + lufs_offset
-        lin_val = 10 ** (rms_dbfs / 20)
-    else:
-        raise ValueError(f"未知单位：{from_unit}")
-
-    # Step 2: 从线性幅度转换到目标单位
-    if to_unit == "linear":
-        return lin_val
-    elif to_unit in ("dBFS", "Peak_dBFS"):
-        return 20 * np.log10(lin_val + EPS)
-    elif to_unit == "RMS_dBFS":
-        if from_unit in ("dBFS", "Peak_dBFS") and crest_factor_db is not None:
-            # 用Peak -> RMS
-            peak_dbfs = 20 * np.log10(lin_val + EPS)
-            return peak_dbfs - crest_factor_db
-        return 20 * np.log10(lin_val + EPS)
-    elif to_unit == "LUFS":
-        if lufs_offset is None:
-            raise ValueError("RMS ↔ LUFS 转换需要提供 lufs_offset")
-        rms_dbfs = 20 * np.log10(lin_val + EPS)
-        return rms_dbfs - lufs_offset
-    else:
-        raise ValueError(f"未知单位：{to_unit}")