From: batguano999
Sent: Wednesday, January 25, 2017 15:50

The next question is how to convert AVC3 to AVC1 or AAC. I have already
tried using ffmpeg and its aac codec. It was very slow and I still couldn't
play the file. I have also tried the Nero encoder


This is pointless.
If you're converting from lossy to lossy aac you might as well convert from lossy to lossy mp3.

What I had in mind was something like
mp4box -bs-switching no
to force AVC1. Also whether it is possible to force uniform sample rate to minimise the size of the staz table.

By the way, what about this?
Create a 30 minute "raw aac" test file with FFmpeg...
ffmpeg -f lavfi -i sine=d=1800 -y 1800_testfile.aac
Mux it into m4a with mp4creator...
mp4creator -create=1800_testfile.aac 1800_testfile.m4a
See if it plays OK... with your "£19 AGPtEK player"

From: iz
Sent: Wednesday, January 25, 2017 14:30

One other long shot to try is to remove the MP4 container by remuxing to plain old ADTS AAC. However, I'm guessing that even if that file played you >may not be able to seek/ff/rw, so of limited use. You could also feed the raw fragmented MP4 DASH file to your player, but I suspect that is even less >likely to be >supported.

I had already tried playing the raw file in VLC. It played, but I noticed seeking was slow. The file with the ADTS container did play in VLC, and seeking seemed normal.

With -f lavfi I got
Requested output format 'lavfi' is not a suitable output format
tfr1800.aac: Invalid argument

(I had renamed the raw file from .m4a to .aac but I got the same message when I renamed it back to .m4a) I used -f adts instead. The result was not playable in the AGPtEK player.

I did not understand the -i parameter so I used -t for the duration.

These were the commands
ffmpeg -i tfr.aac -f adts -t 1800 -acodec copy tfr1800.aac
mp4creator -create tfr1800.aac tfr1800.m4a

Without -acodec copy it converted from aac to aac using ffmpeg's built in codec, which was very slow. The resultant file with a duration of 1800s (30min) did play on the AGPtEK player, but durations of 1900s and more did not play. That is an improvement on 14min, but I noticed some drop outs not in the FLAC conversion. I think I need to stick to FLAC or 320kbit/s MP3 for music and 128kbit/s MP3 for speech.

I still have a lot to learn about the relationship between containers and content, and why a container can cause a problem for a player, or why AAC is so much more difficult to decode than MP3 or FLAC. (The reason I commented on price was that AAC players generally seem to be a lot more expensive than MP3 players.) I have been struck by how astonishingly little non-BBC AAC material is available for testing. There is a lot of free MP3 material. I do not use iTunes.

Thank you both for your suggestions and comments.






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