music quality

[Kuukunen]

It's a very bad idea to convert from one lossy format to another. You will usually lose a lot of quality in the conversion.

If you do that, you will have to rip from an original disc and encode.

Well I kinda meant re-encoding from the CDs. You're right, should've used "encode again". And don't tell me you don't have the CDs, you bad persons. :fear:

[Mulkero]

I find quality mp3's with dc++ very easily. You can see the quality from the filesize and avoid the small files. If I like an album, I buy it. And then encode it to my hd with even better quality because I don't listen to cd's very much.

[Sufficient]

Soulseek is awesome for music too. You can look at what encoding it is before u dl.

[NaiLed]

is it possible to have a bitrate of 0? i just noticed that a couple of my mp3s do, but they sound fine (they aren't actually 0kbps?)

[Kuukunen]

No, it most definitely is not. It's probably that whatever you're using for viewing the bitrate doesn't understand it. It might be variable bitrate or something. 0kbps would mean that the file's size is 0 bytes.

[Jeansi]

Stuff ripped from LP's, original master records and some live recordings I have as FLAC files, lossless compression. Files are about 20-60 megs depending on the lenght of the song.

Rest of my songs are mp3 with 192 kbit usually. I had some with even higher, up to 320, but after being forced to sell my expensive gear I couldn't hear any difference between 192 and 320 anymore so I downgraded them all to 192.

On my mp3-player I have 128 kbit because the sound quality is so crappy anyway that I can't notice any difference.

edit: Looks nice Jerome, I bet you're gonna be really frustrated when you move out and realize that you can't afford buying things like that for another 10 years

[Kuukunen]

Playing low quality music on high-quality speakers is a bad idea. My laptop connects to my hifi system via some monster cables. As you can see, it's fairly nice equipment, so I gotta be careful.

I'd like to see some more info on that. If you were talking about low quality records, I might believe you. I mean if they have lots of noise (hiss, whatever, dunno) and you played it with high volume the speakers' cones (whatever) would have to move pretty fast. But in the case of low-quality mp3, having low bitrate actually removes the high frequency noise. (It removes high frequency anything.)

[Jeansi]

Kuukunen, the sound will "mess up". Like with a highly compressed picture file compared to the original. You'll hear these things called artifacts in the sound.

Ah well, let wiki explain to save us all some time

One technique is to use a lower bitrate by resampling the audio. By reducing the sample rate, higher frequencies must be removed to conform to the Nyquist-Shannon sampling theorem. If the anti-aliasing filter works imperfectly, digital distortion or aliasing will be heard in the form of inharmonic frequencies reflected around the Nyquist frequency. (e.g. a 22.85 kHz tone processed with a Nyquist frequency of 22.05 kHz will result in a tone of 22.05 - (22.85 - 22.05) = 21.25 kHz. This can be generalised to outputF = NF x 2 - inputF) This may be subtle, but more severe levels of distortion can sound similar to ring modulation. (see Aliasing#An_audio_example) Lowering the amount of data (bits) captured per sample can result in loss of detail and dynamic range in the audio. The loss of quality in both methods will be uniform across the recording.

Another technique is to attempt to remove sounds that typical human hearing cannot perceive. As a human being cannot perceive the difference, the resulting data will be simpler (and thus compress better using lossless techniques). For example, in general human beings are unable to perceive a quiet tone simultaneously with a similar, but louder tone. A lossy compression technique might identify this quiet tone and attempt to remove it. As no algorithm is perfect and tradeoffs can be made to throw away additional data to reduce data rate, this will occasionally lead to perceivable sounds being discarded. As these sounds are, ideally, hard to perceive anyway, the result will generally be of flattening complexity, or muddying the sound.

Many systems attempt to replace the series of samples of audio with other representations. Typically these representations make it easier to attempt to eliminate non-perceivable sounds and make it easier to compress the data using traditional lossless techniques. One common technique is to represent the audio as the sum of a series of sine waves. The representation may not be perfect; in exchange for a more easily compressed description, accuracy may be sacrificed.

Many audio compression systems endeavor to match a target data rate, typically expressed in bits of data per second of audio. When using a constant data rate, simple portions of the recording (a simple tone or silence) will be easily compressed to the target rate; the resulting playback will be highly similar to the source audio. As more complex sections are recorded, the system will be forced to seriously reduce quality to meet the target rate; the resulting playback will display more artifacts. Many audio compression systems support Variable Bit Rate encoding, varying the target rate in an attempt to maintain a constant quality of reproduction.

you can try for yourself with these obvious example files:

http://ff123.net/training/training.html

Here's a visual example of this if it makes it easier for you to understand:


See the difference with the origianl pic (upper left) and the compressed one (lower left)??

[Mulkero]

With variable bitrate the bitrate can be zero when there's nothing to hear.

[Kuukunen]

I know Jeansi. What I said, it also removes the high-frequency sounds:

One technique is to use a lower bitrate by resampling the audio. By reducing the sample rate, higher frequencies must be removed to conform to the Nyquist-Shannon sampling theorem.

[Kuukunen]

With variable bitrate the bitrate can be zero when there's nothing to hear.

Well actually in theory you can make a CBR file with 0kbps too, there's just nothing in it. :P

But the mp3 format doesn't allow CBR to be 0, there's bunch of bitrates, lowest is 8 and highest 320. I thought VBR only allowed those same bitrates, but they might've added 0 to it.

[geekbot]

Hmm, not really. Let's see... This CD I have in my hand says it has 700MB or 80 mins... That means about 1166kbps.

This is not right. 700MB is the amount of data that the CD can store. 80 minutes is the length of a music CD you can burn in CD audio format- which is not related to audio compression.

Short answer to the original question- a higher bit rate means more information is retained in encoding.

In general, the higher the bit rate, the more bits are used in encoding and more true in should be. So with the same encoder, 192 kbps will lose less of the original sound than 128 kbps.

However there are many types of encoders as well as quality of encoders. MP2->MP3->MP4 are different generations of standards with increasing quality. A 128 kbps MP3 may sound as good as a 192 kbps MP2 because it does a better job of encoding. In general (according to geekbot), ogg and aac are better than mp3. I rip MP3s at 192 and aac at 160 because I can generally not tell that it's ripped at those levels.

If you can't tell the difference, then use the lower bit rate and save disk space. However, if you get better equipment (headphones, sound card, portable) you might start to hear a difference.

0 kbps is meaningless, just some hiccup with the file or the decoder.

Lossy and lossless in compression refers to if decoding back to the original is possible. With lossy audio, higher frequencies are thrown away to decrease file size. Lossless means you can get everything back.

[Kuukunen]

This is not right. 700MB is the amount of data that the CD can store. 80 minutes is the length of a music CD you can burn in CD audio format- which is not related to audio compression.

He was wondering about what is the bitrate of CD. Bitrate isn't actually related to just audio compression, it just means how many bits per second there is. Music is stored in the CD as bits, so that's the bitrate of CD. (Not that accurate, the numbers are very rounded, but something in that direction.)

0 kbps is meaningless, just some hiccup with the file or the decoder.

More likely it's a hiccup in the program showing the bitrate. Many programs have troubles with variable bitrates.

Lossy and lossless in compression refers to if decoding back to the original is possible. With lossy audio, higher frequencies are thrown away to decrease file size. Lossless means you can get everything back.

With lossy audio, they throw a lot away, but not necessarily the high frequencies. You can save a lot of space by removing high frequencies, because they demand the most bitrate, but it's also possible to save evenly all around the spectrum. At least in theory, you can check the compression methods of different formats if you want.