如何将WAV转换为MP3和MP3，同时保持相同的尺寸
我找不到如何将WAV转换为MP3和MP3为WAV。谁知道如何将 .wav 文件转换为 .mp3 或 .ogg ，然后转换回.wav 匹配100％相同的大小（如未接触的）（如果可以在命令行中完成，则更好）。我尝试使用la脚，然后使用一些工具返回.wav，但是该文件将与每个字节100％字节匹配，就像从未被触摸过的话一样。有人知道可以帮助我的Sox或FFMPEG中的任何命令行吗？谢谢！
I cannot find out how I can convert a wav to mp3 and mp3 to wav. Does anyone know how to convert a .wav file into a .mp3 or .ogg and later convert back into .wav while matching 100% same size like untouched (if it can be done in the command line its much better). I tried to use LAME and later back to .wav with some tools but the file wouldn’t match 100% byte per byte like if it was never was touched. Does anyone know any command line in SoX or FFMPEG that can help me? Thanks!
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大多数WAV文件是原始PCM。 mp3是mp3。而且，大多数OGG文件都将包含Vorbis或Opus。
mp3，vorbis和opus都是有损的编解码器。他们通过利用我们听到的内容和所听到的内容，心理声学等，并节省带宽。带宽和音频质量之间的权衡。
您不能使用有损耗的编解码器的输出来返回原始源。因此，您绝对不能指望二进制比较输出并使它们变得相同。
您甚至在不了解源的情况下也无法真正获得相同的文件大小。例如，MP3编解码器的输入可能是24位音频，但是接收编解码器的输出几乎总是为16位配置。同样，这些有损的编解码器通常是不准确的。 MP3特别有问题。如果您有疑问，请阅读“无间隙播放”。
Most WAV files are raw PCM. MP3 is MP3. And, most Ogg files are going to contain Vorbis or Opus.
MP3, Vorbis, and Opus, are all lossy codecs. They work by taking advantage of what we hear and what we don't hear, psychoacoustics and all that, and saving bandwidth. It's tradeoff between bandwidth and audio quality.
You cannot use the output of a lossy codec to get back to the original source. Therefore, you definitely can't expect to binary compare the outputs and get them to be the same.
You also can't even get the same file size really without knowing more about the source. For instance, the input of your MP3 codec might have been 24-bit audio, but the output of the receiving codec is almost always going to be configured for 16-bit. Also, it's common for these lossy codecs to not be sample-accurate. MP3 in particular has a problem with this. Read up on "gapless playback" if you're in doubt.