Parallelize encoding of audio-only segments in ffmpeg

Question

We are looking to decrease the execution time of segmentation/encoding from wav to aac segmented for HTTP live streaming using ffmpeg to segment and generate a m3u8 playlist by utilizing all the cores of our machine.

In one experiment, I had ffmpeg directly segment a wav file into aac with libfdk_aac, however it took quite a long time to finish.

In the second experiment, I had ffmpeg segment a wav file as is (wav) which was quite fast (< 1 second on our machines), then use GNU parallel to execute ffmpeg again to encode the wav segments to aac and manually changed the .m3u8 file without changing their durations. This was performed much faster however "silence" gaps could be heard when streaming the output audio.

I have initially tried the second scenario using mp3 and result was still quite the same. Though I've read that lame adds padding during encoding (http://scruss.com/blog/2012/02/21/generational-loss-in-mp3-re-encoding/), does this this mean that libfdk_aac also adds padding during encoding?

Maybe this one is related to this question: How can I encode and segment audio files without having gaps (or audio pops) between segments when I reconstruct it?

Answer 1

According to section 4 of HLS Specification, we have this:

A Transport Stream or audio elementary stream segment MUST be the continuation of the encoded media at the end of the segment with the previous sequence number, where values in a continuous series, such as timestamps and Continuity Counters, continue uninterrupted

"Silence" gaps are 99,99% of times related to wrong counters/discontinuity. Because you wrote that you manually changed the .m3u8 file without changing their durations I deduce you tried to cut the audio by yourself. It can't be done.

An HLS stream can't have a parallelizable creation because of these counters. They must follow a sequence [ MPEG2-TS :-( ]. You better get a faster processor.