Re "Issues with Large Files" and "Recent Radio Downloads Not Playing".

[RS]

>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.