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Bitrate conversion question.
Alright here's the problem.
I might be making a Xenosaga III rip for all the songs that are missing, but I encountered a small problem. I have access to all the songs on the DVD, but they have a bitrate of either 24000 or 48000. I can convert the wav files to those bitrates, but Exact Audio Copy only takes on 44100 bitrates. If I convert them to that bitrate there's a noticable quality drop, so my question is, is there any way to change it to that bitrate, without changing the actual quality? |
You're talking about sample rates, not bitrates, just FYI. It's 44100 Hz, not 44100 kb/s.
The best way to do it would probably be to use an encoder that doesn't force you to use 44100 Hz wav files as input. Try foobar2k or the BlueRazorLame frontend, as I know for a fact that they will take a 48000 Hz wav as input. If you're dead set on resampling to 44100 (which really isn't necessary), what you need is a resampler that does nothing but resample. If you're losing quality, your converter is probably doing more than just resampling, or the resampler could just suck. There may be a simpler way to do this, but my personal suggestion would be using the BeSweetGUI and running it through nothing but the SSRC resampler. Go here for BeSweet, which includes SSRC, and here for the BeSweetGUI, which makes using BeSweet easier. Extract them both and run BeSweetGUI, then use the browse button beside BeSweet.exe to point the GUI at BeSweet. After that, select your input and output files the same way. Next, make sure that only SSRC is ticked on that main page. The tooltip should say SSRC - Resampling frequency : 44100. If it doesn't, click the SSRC button over on the right and change the value in the 'set sampling rate of outputfile' to 44100. After that, you're ready to go. BeSweet should do nothing but run it through SSRC and you end up with a 44100 Hz wav file that should sound exactly like the 48000 or whatever one did. |
hmm, tenseiken was faster but here is my answer (i would use foobar for resampling AND mp3 conversion):
I guess you mean samplerate in stead of bitrate. Changing the samplerate (resample) will lower the quality. But if you use a good convertor the quality drop can be very small and unnoticable. Personally i would convert them all to 48khz (or maybe 44.1khz) and then convert them to mp3 with lame -V2. Foobar2000 has a neat convert option that you can use for resampling and mp3 encoding. |
Yeah my bad I guess it's samplerate.
I don't really mind if they stay 24000 or 48000 Khz. I just don't want the quality to drop, and it really noticable when I tried to convert it. I'll give Foobar a shot. Thanks. |
btw i've read that the SRC (Secret Rabit Code) resampler should give a better quality than the standard foobar resampler. www.mega-nerd.com/SRC/fb2k.html
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If if use "-V2 --vbr-new" LAME reports "Encoding as 24 khz"... so I´d say there is no conversion going on when you use LAME.
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Quote:
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Actually, I've read on the HA forums that SSRC (what Foobar uses by default, I believe) is more of a quick open-source sample-rate converter than a high-quality one. I have no idea about this SRC (Secret Rabbit Code), though. You could always try it out, and compare results.
Alternately, you could just use LAME to encode it without converting. It will handle 24KHz sample rates, although I'm not certain how well it works with them. |
You could also use any audio editing software such as Adobe Audition and upsample/downsample the .wavs respectively. I usually do this when the .wavs don't pass through LAME, however I doubt there's any quality loss at all so it doesn't really matter whether you do or don't.
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The thing is that foobar comes with the PPHS resampler nowadays. I've never heard of that one...
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Yeah. It comes with PPHS now. SSRC and PPHS are both high-quality. SSRC used to be highly recommended on HA. Now they recommend PPHS, and as far as I know, it's mostly for producing less load on the CPU load, not because the quality is noticeably better.
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