Transcoding and down sampling

[sviatapolk]

I've never messed with the settings for transcoding and down sampling and I never installed any separate transcodes myself. And I have a few questions:
-the wiki seems to mp,y that you need to install transcodes. But all my FLAC files have been ending up as smaller than FLAC files on my iSub client (and I turned transcoding off on that, over wifi) so is subsonic in fact performing transcoding?
-my music is either in lac or in mp3. I would like to modify subsonic so that the mp3 files are NOT downsampled. I want subsonic to serve them to my client as they are, be it V0, 320kbps, 192kbps. It's an old PC and I'm assume eliminating this step will speed things up (correct me if I'm wrong). What do I need to change to effect this?
-I DO want FLAC files to be transcoded to MP3 for reasons of file size and bandwidth.

So I would like to change the settings on the transcoding screen to eliminate Mp3 to MP3 transcoding/down sampling, but to ensure that other formats are transcoded to MP3.

Again, my current settings are the default settings and I've installed, to my knowledge, no additional components.

[ezraghast]

To change transcoding settings, log in to the Subsonic interface, go to Settings and Transcoding. Make sure 'mp3' is not included in the 'Convert from' field and save settings if you had to remove it. Depending on your available bandwidth/streaming method, you might want to remove ogg, m4a and wma from the 'Convert from' field too. A lot (including FLAC) can be streamed without conversion, bandwidth permitting. What kind of device are you running Subsonic from?

[sviatapolk]

I'm using iSub on my iPhone. I turned off transcoding over Wifi on the client, but there is still the downsampling on the server that takes place. I can tell within my home network that a 192kbps MP3 album will download 5-10x faster than a 256kbps album. So clearly subsonic is downsampling (to v0???) server side.

Please look at the attached image. I'm not sure what I need to do precisely.

Again, I want my Flac transcoded to MP3, but I want mty MP3s (the majority of my library) to not be downsampled. I have no M4A OGG, or other formats so they're non-issues.

[GJ51]

Isn't the downsampling determined by the setting in the app?

On the Android app in settings there is a section that allows you to limit the bitrate and also set it to unlimited.

[audioscavenger]

i'm trying to transcode with the lates top on the line technics : mp3 (256-320) / flac --> HE-AAC v2
i hope it's possible easily
i'll let u know ^^

[audioscavenger]

ok here some results for a transcode of a 6MB/160kbps MP3 file to 4 different AAC profiles at 32kbps :

Code: Select all

ffmpeg -i yves0.mp3 -v 0 -b:a 32k -codec:a libfdk_aac                      yves.libfdk_aac.32.m4a
ffmpeg -i yves0.mp3 -v 0 -b:a 32k -codec:a libfdk_aac -profile:a aac_he    yves.libfdk_aac.aac_he.32.m4a
ffmpeg -i yves0.mp3 -v 0 -b:a 32k -codec:a libfdk_aac -profile:a aac_he_v2 yves.libfdk_aac.aac_he_v2.32.m4a
ffmpeg -i yves0.mp3 -v 0 -b:a 32k -codec:a aac -strict experimental        yves.aac.32.m4a
ffmpeg -i yves0.mp3 -v 0 -b:a 32k -codec:a libfaac                         yves.libifaac.32.m4a

time   size name
0:0:30 1.3M yves.libfdk_aac.32.m4a
0:0:37 1.2M yves.libfdk_aac.aac_he.32.m4a
0:0:31 1.2M yves.libfdk_aac.aac_he_v2.32.m4a
0:0:41 1.4M yves.aac.32.m4a
0:0:37 2.3M yves.libifaac.32.m4a

when hearing the sound (yves de ruiter, trance), which mixes many different frequencies, with big bass and high tones, the heard quality sounds like this:

libfdk_aac (AAC_HE_V2 >= AAC_HE > AAC_LC) >> libfaac >>> native ffmpeg AAC

i think the native ffmpeg AAC to be the worst since it introduces strong distortion. on the contrary, libfaac is quite stable sounds like blowed.
AAC_HE_V2 is definitely the best for bitrates under 48kbps.

here is the line to use for streaming players from subsonic:
Name: m4a audio
Convert From: mp3 ogg oga aac m4a flac wav wma aif aiff ape mpc shn
Convert To: mp4
Step 1: ffmpeg -i %s -v 50 -b:a 32k -codec:a libfdk_aac -profile:a aac_he_v2 -f adts -

both adts and mp4 file format will work, however mp4 seems not to work for every files
well, it's transcoding, however i hear nothing, jw-player won't start.

though, jw-player SHOULD be able to play aac: http://www.longtailvideo.com/support/jw ... at-support
Extension(s) aac, m4a, f4a
Mimetype audio/mp4

i tried libfaac, aac, and many other methods such as -frag_duration 2 and the flv method too, none is working

what am i doing wrong ?

[jonsaddles]

Anyone had any success trying to sort this down sampling out?
No matter what settings I use I get nothing at all unless everything is set to a higher bitrate than the track is recorded in.
I tried adding mp3 into the list of formats to transcode from but to no effect.