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Frequency response vs HD 24bit track sample rate

post #1 of 8
Thread Starter 

Hey guys,

 

So I got my hands on some 24bit flac that has 96,000Hz sample rate. While I understand that the higher bit rate and sample rate the better it sounds, is sample related to "frequency response" in any way? I'm using the audio technica ATH-M50 and it has a response from 15-28,000 Hz, what does that mean? Even the normal mp3 songs recorded at 44,100 Hz is above the 28,000 Hz upper range.

 

Thanks!

post #2 of 8

You're basically asking about the Nyquist rate, the idea as it pertains to audio being that you can reconstruct any waveform if your sampling rate is more than twice the highest frequency in the waveform.  Generally speaking there isn't much in the way of evidence in favor of frequencies higher than 20,000 Hz being perceived or altering perception of it.  I haven't seen any test to see if the higher sampling rate allows for a better reconstruction of the original waveform, though it probably wouldn't do anything.  

 

The idea behind 24 bit audio is that there are more possible levels between silence and the loudest volume.  16 bit audio has 65536 levels, and 24 bit audio has 16777216.  Additionally some DACs running in 24 bit mode (even with 16 bit files) will have a lower noise floor.  


Edited by Phos - 2/9/12 at 2:27pm
post #3 of 8

without any techno talk-those frequencies are NOT related. One is the quality of the audio, the other is the dynamic response of the headphone that plays them. A nice headphone response will sound like **** with **** audio and vice versa.

post #4 of 8

Thats really cool to know Phos!

post #5 of 8
Quote:
Originally Posted by mingamo View Post

Thats really cool to know Phos!


To put the levels in perspective before you think 16 bit is a weakness, each bit allows for approximately 6 dB of dynamic range above the digital noise floor. So 16 bit allows for peak volume of 96 dB before the noise floor is audible. That's enough for almost any situation, and even in situations where louder is needed, the noise floor is likely to be buried in the ambient noise of the room anyway. You would have to play at ~110 dB peaks in a room with ~20-30 dB ambient volume for the weaknesses of 16 bit audio to be noticeable. In comparison, I listen at ~80 dB peaks in a room with ambient volume around 40-50 dB.

 

24 bits allows for 144 dB of dynamic range, something no playback system can actually achieve anyway.

post #6 of 8

Remastering in 24 bit vs. 16 bit is a no brainer. You get better results with the 24 bit file than with the 16 bit. Imagine a sine wave (~), the "bits" are used to represent the steps used to map out the distance going up and down on the waveform itself. The more bits you have the finer the steps become and the more resolution you will have. It also dependes on the sampling frequency of the 24 bit file as well. If you have a 96 KHz sampling frequency this will be better than twice as fine as a CD which has 44.1 KHz sample rate. The sample rate is the criteria for how wide the steps on the sine wave will be. The DAC (digital to analog converter) will then take the numbers and turn it into a sound that is pretty close to the original. The finer the steps, the closer to the original it will be.  

 

Again the bits are the (rise) of the "stairs" and the sample rate is the (run) (if you are into linear algebra)! I hope that I explained it well. Actually it is very common to see 24/196 in all recording studios now, since they do alot of mastering there, it makes sense to use those rates, even if your endgame is the "lowly" CD, or a dreaded MP3 file. 5,644,800 Hz or 5.6448 MHz sampling rates are possible and used today.

 

If you try to sample a real world sound from a band playing hundreds of frequencies at a time and only use 44.1 KHz sampling freq, then you can draw a pretty good 22 KHz sine wave. But since there are a bunch of other frequencies on top of the others it makes more sense to use a faster sampling rate to get the most resolution possible, and try to map out the complexities of all of the combined sounds. That's why you hear so much more detail in the higher bit/ higher sample music.   

 

With the Frequency response of the headphones it is a rule of thumb (sort of) that if the freq response is wide like 15Hz-28KHz you have a better quality driver in the phones. But that is not what makes them sound good, for the most part. It just means that the phones are capable of producing dB (sound pressure, decibel) at 15Hz as well as 28KHz and everything in between. The dB will change dramatically between the frequencies and a good phone will be as flat (not change dB) along the frequencies. But all headphones are different and none are perfect. It is actually theoretically impossible to make a totally flat responding speaker, because of the change in impedence with the changes in frequency. The more impedence the coil has the less dB you get. The higher the frequency the more impedence you get in an inductor, which is what makes the speakers move less, giving you less dB. There are other speaker designs that don't use coils for motors, and they are more expensive, and have different issues with physics.


Edited by Megaohmz - 2/10/12 at 1:38am
post #7 of 8

What a mess!
 

Quote:
Originally Posted by Megaohmz View Post

Remastering in 24 bit vs. 16 bit is a no brainer. You get better results with the 24 bit file than with the 16 bit. Imagine a sine wave (~), the "bits" are used to represent the steps used to map out the distance going up and down on the waveform itself. The more bits you have the finer the steps become and the more resolution you will have. It also dependes on the sampling frequency of the 24 bit file as well. If you have a 96 KHz sampling frequency this will be better than twice as fine as a CD which has 44.1 KHz sample rate. The sample rate is the criteria for how wide the steps on the sine wave will be. The DAC (digital to analog converter) will then take the numbers and turn it into a sound that is pretty close to the original. The finer the steps, the closer to the original it will be.  

 

The reconstruction filters get rid of the "steps", it's a wholly inaccurate image of what's really happening. Whatever sample rate you are using the filtered analog signal has no "steps". Also, resolution is a matter of sampling rate, dynamic rage is a matter of bit depth, both notions are different. In general, it's true that the greater the sampling rate/but depth is, the more accurately you can digitize a signal, but in audio, you first run into the limits of human hearing (20000 Hz and 0 dB SPL at its most sensitive range) and as important, the lack of musical content above 20000 hertz and the ambient noise of your listening room.

 

Again the bits are the (rise) of the "stairs" and the sample rate is the (run) (if you are into linear algebra)! I hope that I explained it well. Actually it is very common to see 24/196 in all recording studios now, since they do alot of mastering there, it makes sense to use those rates, even if your endgame is the "lowly" CD, or a dreaded MP3 file. 5,644,800 Hz or 5.6448 MHz sampling rates are possible and used today.

 

It makes sense to use them in studios because they are doing a lot of editing and thus limiting noise and calculations artifacts is a huge bonus, it makes much much less sense for playback. Besides, comparing DSD sample rates to PCM sample rates is confusing to the reader since it's a totally different technology, most of the DSD coding is noise, it relies on pulse density modulation and the frequency used has a totally different purpose unrelated to to the Nyquist frequency of PCM (pulse code modulation).

 

If you try to sample a real world sound from a band playing hundreds of frequencies at a time and only use 44.1 KHz sampling freq, then you can draw a pretty good 22 KHz sine wave. But since there are a bunch of other frequencies on top of the others it makes more sense to use a faster sampling rate to get the most resolution possible, and try to map out the complexities of all of the combined sounds. That's why you hear so much more detail in the higher bit/ higher sample music.   

 

Again you fail to understand the basis of a linear system and the frequency domain, adding soundwaves of different frequencies don't make you suddenly need higher frequencies to represent them. Unless you somehow postulate that higher than 22000 Hz frequencies are necessary for reproducing music even when they are inaudible, there's no reason to encode higher than 22000 Hz information.

 

With the Frequency response of the headphones it is a rule of thumb (sort of) that if the freq response is wide like 15Hz-28KHz you have a better quality driver in the phones. But that is not what makes them sound good, for the most part. It just means that the phones are capable of producing dB (sound pressure, decibel) at 15Hz as well as 28KHz and everything in between. The dB will change dramatically between the frequencies and a good phone will be as flat (not change dB) along the frequencies. But all headphones are different and none are perfect.

 

Agreed on that part, the frequency range given is in no way indicative of the final performance of headphones.

 

It is actually theoretically impossible to make a totally flat responding speaker, because of the change in impedence with the changes in frequency. The more impedence the coil has the less dB you get. The higher the frequency the more impedence you get in an inductor, which is what makes the speakers move less, giving you less dB. There are other speaker designs that don't use coils for motors, and they are more expensive, and have different issues with physics.

 

It's quite possible to make a totally flat (with 2 dB in anechoic conditions) speaker, a number of manufacturers certainly succeeded.


 

 

 

post #8 of 8

As far as steps are concened I was trying to explain how the software takes samples of various points of time and the amplitude of the signal. I was just using a staircase analogy (Graph) so that it can be visualized. Sorry if it came out confusing. It is hard to explain without rewriting an entire book on it.

 

But I do understand that some people believe that you don't have to sample any higher than what a CD uses since humans can't hear above 20 KHz, and that the CD ids the holy grail of recording. I happen to not believe that a CD is "Perfect for human hearing". I actually believe that the combination of multiple frequencies of during a recording is WAY more complex than just trying to acurately reproduce an arbitrary 20 KHz sine wave. And using a much higher Bit and time sample rate is more desireable for reproducing real life sound.

 

Take for example if you had a person who could only hear 60 Hz to 10 KHz and is severely bass deaf like a lot of people are, would an MP3 at 128Kbps sound as good to them as a CD would? I don't know for sure, but I'll put my money on that they will like the CD better.    

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