ffmpeg:音频样本的字节序 [英] ffmpeg: Endianness of audio samples

问题描述

我使用ffmpeg的avcodec从c ++应用程序中的音乐文件中检索原始音频样本.对于文件，我对其进行了测试，看来这些文件样本的字节序为little-endian，但我想知道对于所有我尝试解码的文件来说，这是否总是正确的(即，来自ffmpeg的实现或至少是其架构，具体因为我的计算机的体系结构使用小端).如果没有，我认为这将取决于特定文件的编码格式.在那种情况下，如何检查要解码的每个文件采用哪种字节顺序?我在文档中找不到任何相关信息.

I use ffmpeg's avcodec to retrieve raw audio samples from music files in my c++ application. For files I test it on it appears that these files samples' endianness is little-endian, but I wonder if that will be always true for all files I'd try to decode (i.e. that comes from ffmpeg's implementation or at least it's architecture-specific since mine computer's architecture uses little endian). If not, I assume it would depend on particular file's encoding format. In that case how can I check which endianess applies for each file I'm decoding? I can't find any relevant information in the docs.

推荐答案

内部ffmpeg始终对音频样本使用本地字节序，因为这样可以更轻松地对数据执行各种操作(有关此问题的一些文档，请参见libavutil/samplefmt.h文件);根据文件格式的规定，编解码器的任务是转换为适当的字节序.作为一个简单的例子:有一个用于读/写原始样本的普通音频编解码器，称为pcm_*；例如有pcm_s16le和pcm_s16be.在小端架构上，pcm_s16le将不进行任何转换，而pcm_s16be在解码/编码数据时将交换字节.

Internally ffmpeg always uses native endianness for audio samples since it makes it easier to perform various manipulations on the data (see libavutil/samplefmt.h file for some documentation on the matter); it is codec's task to convert to/from an appropriate endianness as dictated by file format. As a simple example of this: there is a family of trivial audiocodecs for reading/writing raw samples called pcm_*; e.g. there are pcm_s16le and pcm_s16be. On little-endian architecture pcm_s16le will do no conversion while pcm_s16be will swap bytes when decoding/encoding data.

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