A Question About Bitrate…

Today I ripped a CD of classical music with the program "Power2Go". It came with my Lenovo machine. I checked the bitrate after the rip and it said that it was at 1,411 kbps! I thought that 320 kbps was the max.

I’m learning new things each day about this stuff. I also converted an audio file (Online Audio Converter – ?onvert Sound to MP3, WAV, MP4, M4A, OGG or iPhone Ringtones (http://online-audio-converter.com/)) to wav and it is at 3,072 kbps.

So I’m assuming that wav is able to hold a higher bitrate than mp3 (I think that FLAC is probably the highest quality). Could someone explain this?

FLAC is not the highest quality. It is one type of losless format. It is equal to any other lossless format. But, FLAC is compressed and so takes up about half the space of an uncompressed wav file. It takes a file and compresses it. In an mp3 this compression causes data loss, which is why it is called a lossy source. The data equates to sound quality. The more data lost, the more detail lost in the compression, the poorer the sound quality will be. The reason FLAC is popular is that it still compresses, but there is NO data lost and so you get more space with no loss of audio quality. You can convert from wav to FLAC (or any other lossless source) with no loss of sound quality. However once the file is converted to mp3, and the data is lost it cannot be restored. You cannot take a file that was converted to mp3 128, and upconvert to a 320 without the sound degrading even further. The result will be more space taken up, but it all being full of empty data. THe sound will be worse than the 128 file because any conversion within the mp3 range, whether up or down causes data to be lossed. The more you process it the worse it will get. Thus it being called lossy, as I mentioned, and why a lot of the folks doing album editing here prefer flac or lossless sources. Some like Sonicadventure will not edit lossy and so the "Deluxe" albums he produces are far rarer, because true lossless sources for a lot of material here just is not easily obtained.

From what I gather from casually reading comments posted here anyway

———- Post added at 06:28 AM ———- Previous post was at 06:21 AM ———-

Btw, the reason many folks here prefer a variable bit rate for their mp3’s rather than a continual set rate is that only certain maounts of data are needed at any given time per track. Coninual bit rate of 320 will make the file that large, whether it is useless empty space or not. But a vbr file only uses exactly as much space as is needed for the audio being played. Same sound quality, less space, hence considered superior, although thinking that way takes some getting used to. Depending on the original audio sources and what was done to them, an mp3 128 file may have superior audio to an mp3 320 file.

also, CD bitrate having 1411 kbs, so any lossless copying will also have 1411 kbs (though sometimes having smaller filesize, depending on container, like flac, ape, tak, wavpack, or other lossless coec)

The bitrate can be expressed as the amount of information per second from two sources: the decoded audio stream or the encoded data present on the file (but not yet decoded). The two bitrates you mention are of the first type:

– Those 1411 kbps from the FLAC file are, as pointed out by Zeratul13, the typical bitrate of CDs and correspond to a decoded stream of 16 bits and 44.1 kHz samples per second in 2 channels:

<pre> 2 * 16 * 44100 = 1411200 bits/second</pre>
Every second of the decoded audio needs 1411.2 kbits, no matter if it is total silence or a loud complex sound.

– Conversely you see 3072 kbps in a FLAC or WAV file if the audio has a faster sampling frequency of 96 kHz and the same bit depth and number of channels:

<pre> 2 * 16 * 96000 = 3072000 bits/second</pre>

For the second type of bitrate in compressed-lossy formats like MP3, the numbers 320, 192 or 128 tell the amount of encoded kilobits to be read from the file (or a mean estimation for non-constant bitrates) in order to deliver 1 second of real audio.

But you may also see this kind of information for lossless files storing sound of legit CD quality at 1411 kbps when some program shows a bitrate like, for example, "758 kbps". That number is computed from the size of the file (which is lossless but also compressed) and the duration of the audio, and it gives an indication about the complexity of the sound for a given encoding algorithm. So, you could have the same original audio in two lossless files showing two bitrates of this kind, like "814 kbps" and "769 kbps"; the difference would depend on the compression mode or the lossless formats used with those identical streams.

:bearclap: Very good everyone. I enjoyed reading the longer posts.

With kbps you’re looking at "data transfer". That is all. However, it does have implications for lossy audio compression, because the lower the rate, the lower the "quality". This shouldn’t be confused with the sample rate of the audio – this figure is 44100Hz for Audio-CDs and 48000Hz for DVD-Video soundtracks.

While uncompressed formats such as WAV will have a fixed (and relatively high) kbps rate, this value can be reduced by using compression algorithms. There are various options available.

The two main branches are:

Lossless
Lossy

With lossless, the idea is to pack your data as much as possible, but then be able to unpack it without a single bit of data being altered – this is somewhat like a zip file. But audio, like video and images, can be compressed to a high degree by being a bit cheeky and cutting corners – and guess what, those corners are the kbps rates. By lowering the amount of data a compression session packs in, less will be needed to be unpacked. The trick is to use a rate which will not betray any quality loss to the listener.

Awesome! Thank you guys so much! I read every bit of it. I think that this is a good little thread for people just like me who want to know more about this subject. Thank you all for your extensive explanations, you get smarter every day!