如何缩放和混合音频? [英] How to scale and mux audio?

问题描述

第一个问题是音频重新缩放.我正在尝试重做 doc/examples/transcode_aac.c ，以便它也可以从41100重新采样到48000，其中包含一个警告，提示它无法执行操作.

First problem is with audio rescaling. I'm trying to redo doc/examples/transcode_aac.c so that it also resamples from 41100 to 48000, it contained a warning that it can't do it.

使用 doc/examples/resampling_audio.c 作为参考，我发现在进行 swr_convert 之前，我需要在输出端找到带有像这样的代码:

Using doc/examples/resampling_audio.c as a reference, I saw that before doing swr_convert, I need to find the number of audio samples at the output with the code like this:

    int dst_nb_samples = av_rescale_rnd( input_frame->nb_samples + swr_get_delay(resampler_context, 41100),
                                         48000, 41100, AV_ROUND_UP);

问题是，当我只设置 int dst_nb_samples = input_frame-> nb_samples (它是1024)时，它会正常编码和播放，但是当我这样做时 av_rescale_rnd 事情(导致1196年发生)，音频会变慢并失真，就像音频中有跳音一样.

Problem is, when I just set int dst_nb_samples = input_frame->nb_samples (which is 1024), it encodes and plays normally, but when I do that av_rescale_rnd thing (which results in 1196), audio is slowed down and distorted, like there are skips in the audio.

第二个问题是尝试将 webm 与 opus 音频混合.

Second problem is with trying to mux webm with opus audio.

当我将 AVStream-> time_base 设置为 1/48000 ，并将 AVFrame-> pts 增加960时，结果文件在播放器中以更大的文件播放.17秒的音频显示为16毫秒的音频，但可以正常播放.

When I set AVStream->time_base to 1/48000, and increase AVFrame->pts by 960, the resulted file is played in the player as a file that is much bigger. 17 seconds audio shows as 16m11s audio, but it plays normally.

当我将 pts 增大20时，它会正常显示，但是在编码过程中有很多 [libopus @ 00ffa660]队列输入在时间上向后消息.与 pts 30相同，仍然有这些消息.

When I increase pts by 20, it displays normally, but has a lot of [libopus @ 00ffa660] Queue input is backward in time messages during the encoding. Same for pts 30, still has those messages.

我应该尝试 time_scale 1/1000吗? webm 始终具有毫秒级的时间码，而 opus 则具有20ms的数据包大小(在48000 Hz时有960个样本).

Should I try time_scale 1/1000? webm always have timecodes in milliseconds, and opus have packet size of 20ms (960 samples at 48000 Hz).

搜索 pts + = 20;

这是整个文件，我所做的所有修改都用//MINE 标记: http://www.mediafire.com/file/jlgo7x4hiz7bw64/transcode_aac.c

Here is the whole file, all modification I did are marked with //MINE: http://www.mediafire.com/file/jlgo7x4hiz7bw64/transcode_aac.c

这是我在 http://www.mediafire.com上测试过的文件/file/zdy0zarlqw3qn6s/480P_600K_71149981_soundonly.mkv

推荐答案

最简单的方法是使用swr_convert_frame，它获取一帧并将其重新采样为完全不同的帧.您可以在此处了解更多信息: https://ffmpeg.org/doxygen/3.2/swresample_8h_source.html

The easiest way to achieve that is by using swr_convert_frame which take a frame and resample it to a completely different one. You can read more about it here: https://ffmpeg.org/doxygen/3.2/swresample_8h_source.html

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