Originally Posted by gettinI9300
So this this what normally happens when you encode an mp3?
In other words, you can never take a lower bitrate mp3 and encode it, raising the bitrate?
1st - What is ENCODING ? Especially Analog to Digital Encoding.
An audio signal is the best in its purest ANALOG form. Nothing is better than a HIGH QUALITY audio master.
When an audio signal goes from the analog to digital domain the BOX that does this conversion takes SAMPLES of the sound at discrete points on the audio signal curve. The change in value of audio signal from an INSTANT X to an INSTANT Y is measured. The gap between the instant X and instant Y defines the sampling 'frequency' i.e. HOW FAST did you sample ? or How many samples did you make in ONE second. i.e. 48 Khz, 44.1, etc. Why are these taken at 48, 44.1... this goes to a thing called Nyquist rate which is better left to another chapter (You can google for Nyquist Rate / theorem)
Now when the audio signal changes from INSTANT X to INSTANT Y.. it is measured by the BOX. The ability of the box to DISTINGUISH small changes between the ValueAt(X) and ValueAt(Y) is defined by its ABILITY to map information.
an e.g. if I have the 2 numbers... 2.45 and 2.449... The 1st number had 2 decimal places for PRECISION and the 2nd has 3 decimal places. So when measuring any value on the 1st scale we will have either the ROUNDING of 2.449 to 2.45 or TRUNCATING of it to 2.44. This PRECISION defines how PRECISELY you are able to store and measure VALUES.
So at any given point when the ACTUAL value cannot be accurately stored as is it is rounded or truncated to the 'best' match on the measurement scale.
This introduces what is called as the QUANTIZATION ERROR. i.e. The inaccuracy.
The BITRATE defines how many BITS of information you can use to store this measured value. Hence 192kbps = 192 x 1000 bits per second. That is a very accurate measurement. Ofcourse there is no end to it.
A combination of better Bitrate and sampling rate bring in LOT of INFORMATION. This DIGITAL information as a part of your FILE is used by a DECODER to "REPRODUCE" the information ... as close as possible to the original.
It shall NEVER be the exact same again. But, the ability of OUR SENSES to distinguish between the ORIGINAL and the RECORDED is well...
This was just one step... i.e. Analog Master to Digital Master... Then it goes through several stages of SOUND enchancements, mixing and mastering to reach your CD. Then you rip a CD to your HDD as a RAW / WAV. Encode that RAW/ WAV into MP3 which uses its own little algorithm to further COMPRESS the information. .............
Which is why a lot of purists dont believe in CDs. And a lot of CD purists hate mp3s.... unless they are super high quality encoded.
Getting back to your question. If you SAMPLE your audio once and have introduced Quantization Errors... the next time you sample it... again.. you have introduced MORE in addition to the existing ones. And any increase in bitrate wont mean anything because your MEASUREMENT points were already LIMITED the first time around.