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Topic: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy (Read 1226 times) previous topic - next topic
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preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Sometimes lossless files that I buy are 48 or even 96khz samplerate. I use FB2k to encode to LAME V3 for my listening library, and use the dBpoweramp/SSRC resampler (from the "available DSPs" menu in FB2k's converter) to get 44.1khz mp3s whenever I start with lossless at a higher samplerate.

I've wondered if there is a more efficient way to get from higher-samplerate lossless to 44.1 khz mp3. Perhaps
[highsamplerate FLAC --> resampler --> LAME]
has some downside to the two-step approach, and there's a way to do this as a single step internal to the LAME encoder.

I also have no reason to expect I could ABX any of the differences here, and am not particularly worried about this. But I'm asking this question because folks here (a) understand the math and process steps, and (b) are aware and honest of what can actually matter for differences being noticeable.

It's also possible that there's no good reason to go to 44.1 khz mp3 files. Given that a given lossy preset will have lowpass at a some frequency level, changing the official samplerate of the lossy file doesn't change the bitrate, and 48khz is normal enough that most hardware handles it and 44.1 just fine. The only reason I've stuck with 44.1 khz mp3 files is because gapless algorithm on my phone app (Musicolet) seems to handle live-concert or other continuous-play stuff better if it's all at 44.1 khz.
I also still use EAC's wav editor (which can only handle normal CD files, 44.1 khz stereo), although that doesn't matter in these cases. I can probably find something better and more flexible, but haven't put in the effort to figure out what such a thing would be, and switch.

Thanks for any feedback here!
God kills a kitten every time you encode with CBR 320

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #1
I'd let LAME do it all in one shot.

I just did a quick experiment with Kabuu Audio Converter and it handled the compression & resampling fine.   I know some resamplers "measure better" than others but I've never heard a difference, no matter what I was using, and you're converting to lossy anyway.

...I have some "ripped" DVD concerts that I've converted to MP3 and I've left them at 48kHz.  But I'm not trying to play them gapless and I don't have an iPhone.  (MP3 doesn't support 96kHz.)

...My compromise is to make a concert-length file plus separate song files with the applause/crowd noise faded in-and-out so I can play them individually and intermix with my other music.    

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #2
I'd let LAME do it all in one shot.

Thanks Doug. By "let LAME do it all in one shot" do you mean
a) just let LAME default to reproducing the samplerate (at least with 48khz files)
b) set LAME itself to output 44.1khz files as part of the internal process.
If (b), do you know if there's a way to do this within the FB2k converter interface? I'm pretty sure there's a way to specify within command line, but I haven't used that in 15 years.

...I have some "ripped" DVD concerts that I've converted to MP3 and I've left them at 48kHz.  But I'm not trying to play them gapless

I discovered that my phone's music app didn't handle gapless on 48khz files through a similar situation. I had a live album that came with a concert DVD, and the DVD had a few songs that the live audio CD didn't. So I pulled those out and converted, initially to 48 khz for the files from the DVD. I re-did the songs from the DVD to 44.1 after noticing the handling-gapless problem with my phone player.
God kills a kitten every time you encode with CBR 320

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #3
LAME can, by commandline, handle resampling internally while encoding. For my case, starting with a higher-samplerate input file, I would add
--resample 44.1
to the commandline

It doesn't appear to FB2k can tell LAME to do this, nor can LameDropXPd. I've usually tagged my files with RG album gain tags, and with commandline I may have to translate those into --scale X commands. For example, an album gain of -6.0 dB would translate into --scale 0.5
God kills a kitten every time you encode with CBR 320

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #4
If you switch to 'Custom' in the LAME encoder window, you can add '--resample 44.1' to the commandline and it will work. I just gave it a try and it was successful. :)

Edit: I'm referring to foobar.

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #5
If you switch to 'Custom' in the LAME encoder window, you can add '--resample 44.1' to the commandline and it will work. I just gave it a try and it was successful. :)

Thanks John! It's probably been 2 decades since I've interacted with you here.

One other Q for you: Do you think that using --resample 44.1 internally while encoding using LAME is meaningfully different (at a theoretical/mathematical level, I wouldn't expect it to be ABX-able) than using one of the resampler DSPs in Foobar before kicking over to LAME? Or would you say, apart from gapless considerations, there's really no value in having 44.1 lossy files vs 48 anyway?
God kills a kitten every time you encode with CBR 320

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #6
If you switch to 'Custom' in the LAME encoder window, you can add '--resample 44.1' to the commandline and it will work. I just gave it a try and it was successful. :)

Thanks John! It's probably been 2 decades since I've interacted with you here.

One other Q for you: Do you think that using --resample 44.1 internally while encoding using LAME is meaningfully different (at a theoretical/mathematical level, I wouldn't expect it to be ABX-able) than using one of the resampler DSPs in Foobar before kicking over to LAME? Or would you say, apart from gapless considerations, there's really no value in having 44.1 lossy files vs 48 anyway?
I am sure, but not qualifed mathematically to comment, that the resampler DSPs in foobar are superior, but, personally, I encode 48kHz files as 48kHz and 44.1 as 44.1. I am certain that I couldn't ABX any difference either way, it's just a personal preference.

2 decades!! There's a frightening thought!! Not sure I'll be around in another 2 decades, but it's worth a try! ;)

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #7
I am sure, but not qualifed mathematically to comment, that the resampler DSPs in foobar are superior, but, personally, I encode 48kHz files as 48kHz and 44.1 as 44.1. I am certain that I couldn't ABX any difference either way, it's just a personal preference.

2 decades!! There's a frightening thought!! Not sure I'll be around in another 2 decades, but it's worth a try! ;)

Yeah, time keeps moving and all. Kids, living in California, etc.

Thanks for the feedback there. I may as well just push them to 48 khz, apart from live concerts where everything handling gapless properly really matters to me.

I'd figured that FLAC --> resampler DSP --> LAME mp3 would be worse, because of that extra step compared to doing everything in one step (which is me making significant assumptions about the math going on inside of LAME). But whatever the case, 48 FLAC --> 48 mp3 is simpler. Not that I expect to be able to ABX any of it anyway.
God kills a kitten every time you encode with CBR 320

 

Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy

Reply #8
That is how I've done it ;   Input > resampler dbpoweramp (44.1)  > convert mono to stereo > Replaygain album mode [optional-works in any player]

This way you always end up with normalized cd-audio format files. No problems so far.  If the input is already 2ch-44.1 then nothing
is changed.