Rockbox Technical Forums
Support and General Use => Audio Playback, Database and Playlists => Topic started by: BZFrank on April 27, 2016, 08:15:21 PM
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Howdy,
this is something is stumbled upon while trying to get a bunch of AAC files (extracted as stream from videos) to play in Rockbox on a Sansa Clip+. The plain AAC files are not shown and renaming them to m4a or mp4 does not work either as Rockbox refuses to play them.
I didn't want to transcode the AAC data to something else (due to loss of quality) so I used MP4Box to mux the AAC into an MP4 container.
mp4box -add in.acc -new out.mp4
Now Rockbox plays them without any issues and I am a happy camper.
https://gpac.wp.mines-telecom.fr/mp4box
Of course this is not exactly rocket science but I have seen quite a number of questions about this topic while looking for a solution, so I hope this is useful for anyone seeking wisdom. ;)
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Yeah, raw AAC is not supported. It has to be MP4/M4A.
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This is the exact issue I was having today.
for those wanting to use ffmpeg you can do the same thing like this:
ffmpeg -i input.aac -acodec copy output.mp4
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This didn't work with ffmpeg. Although Rockbox tries to play the m4a file, it freezes up every few seconds and needs a hard reset.
Also didn't work with mp4box, same thing happens... not sure what is different here compared to OP. The AAC files are output by streamripper from local radio station. ffmpeg identifies the input files as HE-AAC and so does Rockbox with the output. Using a Sansa e280... maybe it isn't fast enough to play the mp4?
Any other ideas on how to get these files playing with as little processing/transcoding as possible (using linux tools)?
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The e280v1 is not fast enough to play aac-he. The v2 is fast enough, but it sounds like you have the v1.
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OK thanks, probably right... how could I tell which version it is?
Also what would be the best way to make these files playable with least possible quality loss? I opened up the aac file in Audacity and took a look at the spectrogram, looks like it's about 34 kHz. I tried transcoding it to mp3 q9 (worst) and it cut the audio way down to 22 kHz and doubled the file size :'(
Ogg q0 (worst) was better, cutting it to 30 kHz and increasing the file size by 50%.... any other ideas? thanks
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Normal hearing only goes up to 16 KHz, so there shouldn't be higher frequencies in normal audio.
I would convert the files to Ogg or regular AAC.
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Well yeah, I was looking at the total frequency of the file, so a 30 kHz Ogg file with 2 channels would be 15 kHz for each channel (ear), so it should be pretty OK.
but your statement reminds me of something quite interesting. Research has been done that instruments can produce frequencies over 100 kHz and, even though we don't hear these frequencies, they still have an effect on the tiny bones in our inner ear and our bodies, contributing to the overall "listening experience"... but short of a live concert, would need top-notch recordings and audio-reproduction equipment to re-experience the frequencies. But cutting out those frequencies does have effects... something to think about.
https://www.cco.caltech.edu/~boyk/spectra/spectra.htm
But for listening in the car, the Ogg looks like the best bet. AAC produces the cleanest copy but the file size more than quadruples!