WAV音频文件压缩不起作用

问题描述

是否可以在不降低采样率的情况下压缩wav音频文件？

Is it possible to compress a wav audio file without reducing the sampling rate?

我有一个具有256位速率和8000Hz采样率的音频文件。我只想将比特率降低到128/64 kbs

I have an audio file with 256 bit rate and sampling rate - 8000Hz. I would just like to reduce the bit rate to 128/64 kbs

我尝试转换为mp3并返回到wav，
ffmpeg -i input.wav 1 .mp3
ffmpeg -i 1.mp3 -acodec pcm_s16le -ar 4000 out.wav
，但这也会降低采样率。
ffmpeg -i 1.mp3 -acodec pcm_s16le -ab 128 out.wav具有默认的256位速率

I tried converting to mp3 and back to wav, ffmpeg -i input.wav 1.mp3 ffmpeg -i "1.mp3" -acodec pcm_s16le -ar 4000 out.wavbut this reduced sampling rate as well. ffmpeg -i "1.mp3" -acodec pcm_s16le -ab 128 out.wav has default 256 bit rate

推荐答案

PCM（ WAV）未压缩，因此 -b：a / -ab 被忽略。

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计算PCM比特率

假设立体声输入，每秒8000个采样，每个采样16位：

PCM ("WAV") is uncompressed, so -b:a/-ab is ignored.

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Calculating PCM bitrate

Assuming a stereo input, 8000 samples per second, 16 bits per sample:

sample rate × number of channels × bits per sample = bitrate
8000 × 2 × 16 = 256000 bits/s, or 256 kb/s

确定通道，采样率，位深度

您可以只查看 ffmpeg -i input.wav 的输出，也可以使用 ffprobe 以获得更简洁的输出：

Determine channels, sample rate, bit depth

You can just view the output of ffmpeg -i input.wav or use ffprobe for a more concise output:

$ ffprobe -loglevel error -select_streams a -show_entries stream=sample_rate,channels,bits_per_sample -of default=nw=1 input.wav
sample_rate=8000
channels=2
bits_per_sample=16

降低比特率

您可以减少通道数量，更改采样率和/或更改位深度，但是另一种选择是使用无损压缩格式，例如FLAC：

Reducing bitrate

You can reduce the number of channels, change the sample rate, and/or change the bit depth, but another option is to use a lossless compressed format such as FLAC:

$ ffmpeg -i audio.wav audio.flac
$ ls -alh audio.wav audio.flac
  6.1M audio.flac
   11M audio.wac

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