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Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Hello everyone,

I'd like to know how to convert from 24-Bit / 96KHz FLAC to a lossy format in the best (i.e. recommended) way possible. From what i've read I should use a dedicated resampling programm first and then convert to mp3 / aac. Has any of the two any advantages in dealing with hd audio input and which parameters should i use? Do I need to do something about the bit-depth or anything else?

I would appreciate any tip or suggestion.
Chaos

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #1
1. Install foobar2000
2. Install foobar2000 "Free Encoder Pack"
3. Drag files to be converted onto foobar2000 main window
4. Select all, right click, and select "Convert" > "Quick convert" and the target format "MP3 (LAME)" > "Best quality"; post back if you have any other questions about converter settings

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #2
The hydrogenaudio wiki mentions that using lame for resampling is not optimal. Would you still recommend just using lame through foobar for converting hd audio or using mp3 for it in general?

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #3
Some resamplers "measure" better than others but I've never heard any difference/defects between the resampled file and the original  (always above 44.1kHz) no matter what software I was using.

I'm sure you know MP3 is lossy so you're not going to get perfection anyway.    It can sound perfect but the data is altered and if you do hear a compression artifact it's not from resampling.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #4
Some resamplers "measure" better than others but I've never heard any difference/defects between the resampled file and the original  (always above 44.1kHz) no matter what software I was using.

I'm sure you know MP3 is lossy so you're not going to get perfection anyway.    It can sound perfect but the data is altered and if you do hear a compression artifact it's not from resampling.

Yes, I do know that, I just want transparent results ^^ I personally use -V2 for anything that is red book audio (16-Bit / 44,1KHz) and I wasn't sure if inputting higher grade audio into LAME would require different settings for transparency.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #5
If you want transparent results then stick with lossless files. However, lossy encoded files can be transparent to you depending on your own hearing abilities. You mention you use LAME V2, and so if that is transparent for you then use it. Personally, I'd downsample (using sox) the original 24-bit / 96kHz FLACs to 16-bit / 48kHz and keep as archival versions, and then transcode to lossy from there. If your media player handles aac (m4a) then go with that over mp3.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #6
If your lossy encoder can handle 24 bit, then it might be better to resample to 24-bit to avoid adding dithering noise.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #7
If you want transparent results then stick with lossless files. However, lossy encoded files can be transparent to you depending on your own hearing abilities. You mention you use LAME V2, and so if that is transparent for you then use it. Personally, I'd downsample (using sox) the original 24-bit / 96kHz FLACs to 16-bit / 48kHz and keep as archival versions, and then transcode to lossy from there. If your media player handles aac (m4a) then go with that over mp3.

I keep the FLACs anyway on my NAS for future transcoding and stuff. How do I use sox then and which settings (command line arguments) should i use for aac? Anything special or the same recommended settings that are used for red book audio. I mostly need the files for my portable media player which has limited space. I just try to avoid an audible difference and most guides seem to be written for red book audio and I'm really not sure if I can apply the same settings to get the "same" quality so to speak when used on HD audio.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #8
As suggested, maybe the simplest way is to use foobar, add the encoder pack, and then, on the converting settings, ensure that you add any of the resamplers provided in foobar to convert the input to 44Khz ( with LAME, it's better to use 44Khz instead of 48Khz, because that's what it has been most tuned for, even though they should be very similar).
No intermediate files, no command line batch scripts...

And with a bit more work, (to get qaac and the Quicktime libs) you could also have m4a instead of mp3 in that same setup.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #9
As suggested, maybe the simplest way is to use foobar, add the encoder pack, and then, on the converting settings, ensure that you add any of the resamplers provided in foobar to convert the input to 44Khz ( with LAME, it's better to use 44Khz instead of 48Khz, because that's what it has been most tuned for, even though they should be very similar).
No intermediate files, no command line batch scripts...

And with a bit more work, (to get qaac and the Quicktime libs) you could also have m4a instead of mp3 in that same setup.

Okay, thanks for the answer. I will look into that.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #10
Try QAAC. I use it to translate all my high-resolution music into aac without worring resampling.

Since aac support to 96khz it would automatic keep on 96khz or resample to 48khz depands on the bitrate.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #11
Since aac support to 96khz it would automatic keep on 96khz or resample to 48khz depands on the bitrate.
Are you sure?  I just resampled a folder of 24/96 FLAC to m4a using QAAC -tvbr 100 and -c 320.  The resulting files had been resampled to 48000 and, like mp3, do not have a "sample size".

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #12
Hi!, Here is the best way in my setup at least.
First, you need to downsample to the most according to the samplerate and this is the half of the original, IE: 96000/88200 to 48000/44100. I personally use SOX plugin in Foobar2000.
Second, about the resolution in bits, lossy encoders works native in 32bit floating point, that is, you don't need to work here.
Third, i choose somewhat 256bps in opus enc, or qaac, notice that opus by default the sample rate is 48000 so, if you want 44100, the choose is qaac.
tl:dr
foobar2000>sox(downsample 2x)>opus 256 or better

thanks you for this forum, i learnned a lot.
11 years latter.... i finally created an account.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #13
Are you sure?  I just resampled a folder of 24/96 FLAC to m4a using QAAC -tvbr 100 and -c 320.  The resulting files had been resampled to 48000 and, like mp3, do not have a "sample size".
Sorry , Maybe in QAAC 96khz is only for HE-AAC.
You can check out all the output by
Code: [Select]
qaac.exe --format

Totally,
QAAC automatic half the sample rate if needed.
And Most of all it use the Sox resampler(libsoxrate.dll)
So it is the best solution for me.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #14
Are you sure?  I just resampled a folder of 24/96 FLAC to m4a using QAAC -tvbr 100 and -c 320.  The resulting files had been resampled to 48000 and, like mp3, do not have a "sample size".
Sorry , Maybe in QAAC 96khz is only for HE-AAC.
You can check out all the output by
Code: [Select]
qaac.exe --format

Totally,
QAAC automatic half the sample rate if needed.
And Most of all it use the Sox resampler(libsoxrate.dll)
So it is the best solution for me.

Just to clarify. Does QAAC automatically use SoX for resampling or do I need to do that manually before? And if I do that manually before what settings should I set in SoX?

To everbody: Thanks for all your answers, I already learned a lot. I think I will use QAAC with or without resampling beforehand (Depends on the answer to my question above).

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #15
Just to clarify. Does QAAC automatically use SoX for resampling or do I need to do that manually before? And if I do that manually before what settings should I set in SoX?

I only found the most detailed document in Japanese.
Here is a Google translated link
https://translate.google.com/translate?hl=en&sl=ja&tl=en&u=https%3A%2F%2Fkamedo2.hatenablog.jp%2Fentry%2F20130625%2F1372177052

After download QAAC, there is a libsoxr.dll in the same folder with qaac.
QAAC uses sox for the best quality resampling by default.

If you delete libsoxr.dll it would switch to use the Apple native resampler but never do that...

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #16
Just to clarify. Does QAAC automatically use SoX for resampling or do I need to do that manually before? And if I do that manually before what settings should I set in SoX?

I only found the most detailed document in Japanese.
Here is a Google translated link
https://translate.google.com/translate?hl=en&sl=ja&tl=en&u=https%3A%2F%2Fkamedo2.hatenablog.jp%2Fentry%2F20130625%2F1372177052

After download QAAC, there is a libsoxr.dll in the same folder with qaac.
QAAC uses sox for the best quality resampling by default.

If you delete libsoxr.dll it would switch to use the Apple native resampler but never do that...

Oh, thank you very much for that info. It seems that the foobar encoder pack doesn't include it for some reason. I just added the dll to the encoder folder. Hope that fixes that.

Re: Best way to convert HD audio (24-Bit / 96 KHz) into mp3 /aac

Reply #17
It seems that the foobar encoder pack doesn't include it for some reason. I just added the dll to the encoder folder. Hope that fixes that.
Encoder pack is compact and doesn't bundle anything unnecessary. If for some reason you think Apple doesn't handle some sample rate well enough you can use foobar's DSPs to resample. Doing the processing with foobar DSP should also give you performance benefit as the task will be done in another thread.

 
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