I'm getting a sony walkman nwz-s618 mp3 player. The response rate is 20-20000Hz. Does this means it filters frequency outside of that range (even if I use headphones that goes outside that range, like up to 23000Hz)? If so then I want to encode wav or 320kb/s Nero digital files such that the frequency is cut off at exactly 20000Hz. This is because I do not want any wasted bits in a higher frequency that is going to get filtered out. I would rather those wasted bits get used to improve the part of the file that is within that range. This should make the encoded spectrum wave (shown in adobe audition) resemble, within the range, as closely to the source spectrum wave as possible. Now when I encode with lame cbr at 320kb/s the file is cut off at about 20000Hz. While this is somewhat fine, I would rather use a more accurate codec such as wav or 320kb/s ACC Nero digital instead. But with these I have no options on how to set the frequency to a max of 20000Hz. Are there any programs or encoders that allow me the option to encode wav and or 320kb/s Nero digital to be cut off at exactly 20000Hz? Thanks!
Do you really think you know what you're talking about? Read up a bit on that which you talk of (such as: http://en.wikipedia.org/wiki/Analog_sound_vs._digital_sound or http://en.wikipedia.org/wiki/Hearing_(sense)) and come back with a question that makes sense. The upper frequency limit is set by sample rate, so if you sample at 44.1 KHz it will never pass over 22.05 KHz anyway. I mean you could low-pass filter your music before encoding but there's really nothing to speak of there.
I know this stuff already. You misunderstand me. You shouldn't always assume that it is something someone else don't understand instead of you. That way you don't make your and me look like a donkey, if you know what I mean. The Nero digital encoders I've seen didn't have an option for the sample rate/bandwidth at all. The WAV encoders I've seen didn't offer a sampling rate option of 40000Hz. So in short, I'm asking which programs or hacks gives these options. Also, even though most WAV encoders I've seen offered nothing between the sampling rates of 32kHz and 44.1kHz, I've seen a program which offered the sampling rate of 36kHz, which gives me a max of 18kHz per channel. But I want 20kHz max per channel as both I can hear up to 20kHz (my hearing is good) and my player will filter anything higher than that anyway. Also, if you encode a lame file with cbr at 320kb/s it will filter the frequency exactly at 20kHz per channel. But if you encode at 256kb/s then you get a max of 18kHz per channel, etc. You can see this in the graph under adobe audition. Now the codec FAAC has the option of choosing the sampling rate per channel. So I can easily choose a bandwidth of 20kHz and any bitrate I want to make the perfect file for me. But I don't like FAAC but only WAV and Nero digital as they are more accurate than FAAC (lame is close though). In clarifying my question, are there any programs or hacks I can use to limit the bandwidth to 20kHz per channel or sampling rate to 40kHz for WAV and Nero digital? I will try Audacity though. Thanks!
Why would you want to encode at a non standard rate? Will your player even handle it? The specs on SONY's site were pretty vague. They didn't specify the flatness of the '20Hz-20KHz' response or the acceptable sampling rates of the playable formats. If you are really concerned that you are wasting bits by encoding stuff above 20 KHz, then just low pass filter the source. Odds are it was passed through an LP filter before it was recorded originally, anyway.
