How does kbps in MP3 relate to WAV bit/sampling rate?

[korben_dallas]

So if I use dBpowerAMP to rip a 44.1kHz/16bit wav file from a CD to MP3.. is that sampling rate and bit resolution lost?

When people talk about using Foobar to upconvert/upsample, does that also apply to MP3 files?

 

[OracleGuy]

Once you convert from lossless (WAV) to lossy (MP3), well, they call it lossy for a reason! MP3 compresses by throwing away some of the information. If you rip at a decent bitrate (I like LAME VBR at the best quality setting), you'll find that the resulting file may sound "just as good" as the WAV, depending on your ears and equipment.

What you don't want to do is rip to lossy and then convert to some other format, because you'll degrade the sound with each conversion.

Ideally you want to rip from WAV to a lossless format (I use FLAC), because the file compresses some, but you haven't thrown anything away, so you could later create an MP3 with the same quality as if converting from the original WAV.

You need lots of disk space, but I always say "disk is cheap". I'm re-ripping all my MP3s as FLAC.

 

[Rempert]

An mp3 will have the same sample rate as the file it was ripped from. They should also be capable of the same dynamic range. There is no direct relationship between the kbps of an mp3 and it's sample rate. They can be resampled in the same way as the source file.

Obviously, no amount of upsampling will get rid of the distortions that can be caused by the compression.

 

[Albatross05]

Quote:

Originally Posted by OracleGuy Once you convert from lossless (WAV) to lossy (MP3), well, they call it lossy for a reason! MP3 compresses by throwing away some of the information. If you rip at a decent bitrate (I like LAME VBR at the best quality setting), you'll find that the resulting file may sound "just as good" as the WAV, depending on your ears and equipment. What you don't want to do is rip to lossy and then convert to some other format, because you'll degrade the sound with each conversion. Ideally you want to rip from WAV to a lossless format (I use FLAC), because the file compresses some, but you haven't thrown anything away, so you could later create an MP3 with the same quality as if converting from the original WAV. You need lots of disk space, but I always say "disk is cheap". I'm re-ripping all my MP3s as FLAC.

I've been using the .APE format, with pretty good (to my ears) results. How does APE compare to FLAC? I'm assuming there must be a reason as to why APE isn't (or doesn't seem to be) as widely supported as FLAC?

 

[lordgibbness]

Quote:

Originally Posted by Albatross05 I've been using the .APE format, with pretty good (to my ears) results. How does APE compare to FLAC? I'm assuming there must be a reason as to why APE isn't (or doesn't seem to be) as widely supported as FLAC?

FLAC is opensource, APE is not. It should be pretty good to your ears as it is lossless and hence will sound identical to the original cd.

 

[necropimp]

ape is also slower

it usually compresses better than FLAC but like i said FLAC is faster

personally i decided to use wavpack when i go lossless because it compresses a little better than flac and it's even faster...

 

[breez]

Quote:

Originally Posted by lordgibbness FLAC is opensource, APE is not.

From the website:

Quote:

Originally Posted by http://www.monkeysaudio.com/ Freely available source code, simple SDK and non-restrictive licensing - other developers can easily use Monkey's Audio in their own programs -- and there are no evil restrictive licensing agreements

 

[shimage]

Quote:

Originally Posted by Albotross05 I've been using the .APE format, with pretty good (to my ears) results. How does APE compare to FLAC? I'm assuming there must be a reason as to why APE isn't (or doesn't seem to be) as widely supported as FLAC?

All lossless codecs should decode exactly the same, since they're ... well, lossless... The only issues are those of convenience (time, space, compatibility, stability, etc.). Monkey's Audio was closed source for a very long time, and it's still windows only. FLAC was more stable (i.e. it didn't make fatal changes to the format, which ape was succeptible to for a long time) and was open, so it was popular within certain crowds. IIRC, file traders back in the day preferred shorten, though, for reasons unknown to me.
Quote:

Originally Posted by necropimp personally i decided to use wavpack when i go lossless because it compresses a little better than flac and it's even faster...

How much better? I'm just curious since I thought that the difference in compression between most codecs was pretty small. I don't really care much about speed since the limiting factor on my computer is the rate at which it pulls the data off of the CD... Oh, I almost forgot, does it have a CLI frontend for calculating replaygain tags?

 

[Albatross05]

Quote:

Originally Posted by shimage All lossless codecs should decode exactly the same, since they're ... well, lossless... The only issues are those of convenience (time, space, compatibility, stability, etc.). Monkey's Audio was closed source for a very long time, and it's still windows only. FLAC was more stable (i.e. it didn't make fatal changes to the format, which ape was succeptible to for a long time) and was open, so it was popular within certain crowds. IIRC, file traders back in the day preferred shorten, though, for reasons unknown to me.

Thanks for the answers. I think I'll just keep using APE, under the circumstances as my hard drive isn't large enough to rip all my music unless I use the lowest quality anyway. I just rip my faves. I agree though....it is a bit slow and would be pretty painful if you had a lot of stuff to rip.

 

[korben_dallas]

Quote:

Originally Posted by Rempert An mp3 will have the same sample rate as the file it was ripped from. They should also be capable of the same dynamic range.

Quote:

Originally Posted by Rempert There is no direct relationship between the kbps of an mp3 and it's sample rate.

Ummm, kinda sounds like you are contradicting yourself.

Perhaps I need to know what is being received by the card. Does a sound card understand what a Wave file is? Does it understand what an MP3 file is? Or does the media player tell the OS, and the OS tells the device driver, and the device driver tells the sound card hardware.. with translations and lots of 1's and 0's at every point?

Oh.. is there a format between the most compressed lossless and least compressed MP3? Something that would be a "perfect balance"?

 

[Guild]

Interesting comments. Regarding bits: the flac file I'm currently playing in foobar reads as: 16 bits_per_sample and 919 bitrate. What's the difference? It seems backwards to me.

Quote:

Originally Posted by korben_dallas Oh.. is there a format between the most compressed lossless and least compressed MP3? Something that would be a "perfect balance"?

IMHO, "--alt-preset insane" is that balance; although, perfect it can never be.

 

[necropimp]

Quote:

Originally Posted by shimage How much better? I'm just curious since I thought that the difference in compression between most codecs was pretty small. I don't really care much about speed since the limiting factor on my computer is the rate at which it pulls the data off of the CD... Oh, I almost forgot, does it have a CLI frontend for calculating replaygain tags?

well when i tested i used [WITH_TEETH] and FLAC compressed to about 360MB wavpack was 349MB

and i did a silence test out of boredom 5 minutes of silence compressed to 19KB in wavpack FLAC was 75KB but i doubt people are gonna bother too much with silent tracks

and i don't know about the replaygain question... i just do that in foobar

 

[EdipisReks]

Quote:

Originally Posted by korben_dallas Ummm, kinda sounds like you are contradicting yourself.

looks like you are getting sample rate and bit rate confused.

 

[EdipisReks]

Quote:

Originally Posted by Guild Interesting comments. Regarding bits: the flac file I'm currently playing in foobar reads as: 16 bits_per_sample and 919 bitrate. What's the difference? It seems backwards to me.

the sample rate refers to the frequency range of the source that is sampled, and how many "slices" are taken. in this case 16bit probably means 41.1 khz, as i'm assuming it was ripped from a cd and no upsampling was used. the bit rate is the actual amount of data being played per second. this site may help you guys out, as it is spells out the basics.

 

[TURBO]

Quote:

Originally Posted by korben_dallas So if I use dBpowerAMP to rip a 44.1kHz/16bit wav file from a CD to MP3.. is that sampling rate and bit resolution lost? When people talk about using Foobar to upconvert/upsample, does that also apply to MP3 files?

Up sampling
Is the process where the data stream is stretched out by interpolation (guessing the points in between), this is done mathematically.


Over sampling
Is a technique used when transferring frequency waves from analog to digital and vice versa; the signal is sampled many times over and above that actually required by the sampling frequency. Sometimes, there is a perceived noise reduction when over sampling from 44.1 kHz to 192 KHz and over due to, the noise will be spread over a larger band after the over sampling (this is called noise-shaping).

In other words, these terms will be used like this:
Over sampling will be a more dramatic up sampling of the frequencies and usually done by a DAC. Example: 4.1 kHz rose to 352.8 kHz
Up sampling will be used in a lower scale. Example: 44.1 kHz to 192 kHz

Bit deph: It means how many bits are flowing thru the algorithms of a data stream. In other words, how much information is running thou the great high way of rock and roll.