In our application, we are processing audio files using ffmpeg. Specifically, we use the NodeJS library fluent-ffmpeg, (npm link).
Our audio files are generated from various text to speech providers. We recently noticed that when we converted audio using ssml to add pauses to the generated audio, the duration on the file is no longer correct. Upon further investigation, we noticed that the standard audios were also incorrect, just more accurate overall due to the more consistent data. When we put a pause at the beginning of the audio, the estimate was the worst, overshooting it by a very large margin (e.g., a 25s audio clip would read as 3 minutes long, but skip to the end when playing past the 25s mark.
I did some searching and research into the structure of MP3 files, and to me it seems like the issue is because the duration gets estimated by various audio players. Windows media player is an example, but Firefox's web player seems to also do this. I tried changing the ffmpeg command from using .audioQuality(0), which sets ffmpeg to use VBR, to .audioBitrate(320), which tells ffmpeg to use a constant bitrate.
For reference, the we are using libmp3lame, and the full command that gets run is the following, for the VBR and CBR cases respectively:
For VBR (broken durations): ffmpeg -i <URL> -acodec libmp3lame -aq 0 -f mp3 pipe:1
For CBR (correct duration): ffmpeg -i <URL> -acodec libmp3lame -b:a 320k -f mp3 pipe:1
Note: we then pipe the output to the requesting client application after sending the appropriate file headers, hence the pipe:1 output. The input is a cloud storage url where the source file is located
This fixes our problem of having a correct duration, and it makes sense to me why this would fix it if the problem was because the duration is being estimated by some of these players / audio consumers. But, this came at the cost that the file size was significantly larger, which also makes sense to me. While testing we found that compared to the same file in WAV, the VBR mp3 was about 10% of the WAV file size, while the CBR mp3 was still 50% of the WAV file size. This practically defeats the purpose of supporting the mp3 format for our use-case, which is a smaller but slightly lossy alternative to the large WAV file.
While researching, I found that there can be ID3 tags in a chunk at the beginning of the mp3 file, specifying information for the consumer of the audio to know the duration before potentially having processed the whole file. But, I also found that there doesn't seem to be a standard, at least for duration. More things like song title, album, artist, etc.
My question is, is there a way to get the proper duration onto an mp3 file, preferably via some ffmpeg mechanism, while still using VBR? Thanks!
