音量调节

复制import numpy as np
from scipy.io import wavfile

# 读取音频文件
rate, data = wavfile.read('input.wav')

# 标准化：将峰值归一化到 0dBFS（最大振幅 1.0）
dtype = data.dtype
max_val = np.iinfo(dtype).max if np.issubdtype(dtype, np.integer) else 1.0
peak = np.max(np.abs(data))
if peak > 0:
    gain = max_val / peak
    data = (data * gain).astype(dtype)

# 增益：增加 6dB（线性倍数 ≈ 2.0）
gain_db = 6.0
linear_gain = 10 ** (gain_db / 20)
data = np.clip(data * linear_gain, -max_val, max_val).astype(dtype)

# 限幅：硬限制在 -3dBFS 以内
limit_db = -3.0
limit_linear = max_val * (10 ** (limit_db / 20))
data = np.clip(data, -limit_linear, limit_linear).astype(dtype)

wavfile.write('output.wav', rate, data)

复制package main

import (
	"encoding/binary"
	"math"
	"os"
)

func main() {
	// 读取 16-bit PCM WAV 文件（简化：假设已知采样率和数据）
	data, _ := os.ReadFile("input.raw")
	samples := make([]int16, len(data)/2)
	for i := range samples {
		samples[i] = int16(binary.LittleEndian.Uint16(data[i*2:]))
	}

	// 标准化：找到峰值并归一化
	var peak int16
	for _, s := range samples {
		if abs := int16(math.Abs(float64(s))); abs > peak {
			peak = abs
		}
	}
	if peak > 0 {
		gain := float64(math.MaxInt16) / float64(peak)
		for i := range samples {
			samples[i] = int16(float64(samples[i]) * gain)
		}
	}

	// 增益 +3dB
	linearGain := math.Pow(10, 3.0/20)
	for i := range samples {
		val := float64(samples[i]) * linearGain
		if val > math.MaxInt16 {
			val = math.MaxInt16
		} else if val < math.MinInt16 {
			val = math.MinInt16
		}
		samples[i] = int16(val)
	}

	// 限幅：硬限制在 -6dBFS
	limit := int16(float64(math.MaxInt16) * math.Pow(10, -6.0/20))
	for i := range samples {
		if samples[i] > limit {
			samples[i] = limit
		} else if samples[i] < -limit {
			samples[i] = -limit
		}
	}

	// 写回文件
	out := make([]byte, len(samples)*2)
	for i, s := range samples {
		binary.LittleEndian.PutUint16(out[i*2:], uint16(s))
	}
	os.WriteFile("output.raw", out, 0644)
}

复制// 浏览器环境：使用 Web Audio API 处理音频缓冲区
async function normalizeGainLimit(audioBuffer) {
  const ctx = new OfflineAudioContext(
    audioBuffer.numberOfChannels,
    audioBuffer.length,
    audioBuffer.sampleRate
  );

  // 复制原始数据到离线上下文
  const source = ctx.createBufferSource();
  source.buffer = audioBuffer;
  source.connect(ctx.destination);
  source.start();

  // 渲染并获取处理后的 buffer
  const rendered = await ctx.startRendering();
  const channelData = rendered.getChannelData(0); // 仅处理单声道

  // 标准化：峰值归一化到 0dBFS
  let peak = 0;
  for (let i = 0; i < channelData.length; i++) {
    const abs = Math.abs(channelData[i]);
    if (abs > peak) peak = abs;
  }
  if (peak > 0) {
    const gain = 1.0 / peak;
    for (let i = 0; i < channelData.length; i++) {
      channelData[i] *= gain;
    }
  }

  // 增益 +6dB
  const linearGain = Math.pow(10, 6 / 20);
  for (let i = 0; i < channelData.length; i++) {
    channelData[i] = Math.min(1, Math.max(-1, channelData[i] * linearGain));
  }

  // 限幅：硬限制在 -3dBFS
  const limit = Math.pow(10, -3 / 20);
  for (let i = 0; i < channelData.length; i++) {
    channelData[i] = Math.min(limit, Math.max(-limit, channelData[i]));
  }

  return rendered;
}