Audio compression rate

Hi,
is there a way to change the audio compression rate on my Ipod? Bass and treble sound very distorted when connected to my home stereo amplifier.
Many thanks
acer aspire 3500 Windows XP

Distortion can be caused by several things -
1) Simply overloading the input to your hi-fi or by a high data compression rate or a combination of the two. Check the input sensitivity rating of the socket you are using for the iPod on your hi-fi is around the 1 volt level (RMS). If is much less than 1V (say 300mV or less), even a lowish volume setting on the iPod could cause distortion, at least in the bass. Input overload from high bass levels can cause clipping distortion that can blow speaker tweeters (not the woofers) and sound terrible. Also, file compression at less than 128kbps causes progressively more disagreeable distortion right across the audio bandwidth.
2) Downloaded files are sometimes highly compressed (less than 68kbps) and are not worth playing from any half-decent hi-fi. Also, avoid ripping CD's at less than 128kbps if you want to play back through your hi-fi. Preferably, use a bigger bit rate than 128kbps. iTunes material should be fine, at least for the non-audiophile, as it's AAC files are not too highly compressed.
3) If you use the Sound Enhancer (iTunes preferences) or an iPod Equalizer Preset, you could push the overall output toward overload for the hi-fi input.
Happy listening.
iPod 5G video 30GB

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Audio compress rate

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I apologize if this has already been asked, but I have been searching everywhere and I simply cannot find the answer to this.
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Hey Hacienda,
I might not have the experience in audio work you have since I've only been doing this for the past 6 years or so. But I've been a musician for far longer than that, and I've learned A LOT mostly from really smart people in the industry. So, I'm not gonna lie to you and say that I've done extensive testing in this area because I simply do not have the equipment, nor the money to buy it (WAY too expensive). But we do share the neophyte status when it comes to video editing :-P
Anyways, the Nyquist Theorem is not a theory, which is what people are led to believe. It is a theorem, meaning it's already mathematically proven. It is proven that, as long as you follow the premise of capturing twice the highest frequency of the sound source, you'll get a perfect reproduction of it. To capture more than that is a waste of bandwith specially because most people won't even hear above 18KHz, nor do they have the equipment to reproduce such frequencies. Most consumer systems and audio gear, including those found in professional studios, go up to about 22KHz. You need to spend BIG dolars for anything that goes beyond that. So, who are we really making music for here? The super rich? Dolphins?
Now, I know you're not just talking about higher frequencies, but the amount of samples needed to recronstruct a perfect copy of the original waveform. OK, well, this is the kind of snake oil marketing BS I was talking about. The biggest one being that 1bit DSD crap that Sony/Phillips is pushing. Adding more samples to the recording will not make any difference on how faithfully you can reproduce a sound. It'll just make the files bigger for no reason. Again, the Nyquist Theorem already proves this. This is FACT! Here's a link I found interesting regarding these audio industry lies, maybe you will too: http://theaudiocritic.com/back_issues/The_Audio_Critic_26_r.pdf It starts on page 5, but the one pertaining this discussion is lie #3 on page 6. :-D
Don't forget that modern converters already sample at much higher frequencies than the target sampling rate. I believe my RME Fireface 400 samples at 5.6MHz, which is twice the amount of samples compared to DSD technology, before going back down to the target rate. But, like I said, it does so for other reasons and NOT because it needs that many samples in order to faithfully reproduce a waveform. Of more importance are the quality of the FIR (Finite Impulse Response) filter and the clock inside the converters. These components are what make a converter high grade, among others. The converter chips themselves are very inexpensive (in the tens of dolars) which why you hear some companies advertizing having the same converter chip as a ProTools HD rig (not the best example I know).
By the way, I didn't say humans can only hear up to 20KHz. I'm sure there are people who can hear above that. My point was that the 20Hz - 20KHz range is what's generally accepted as an average for humans (which implies that there are people who can hear avobe/below that). Also, the reason why modern-day pop records causes headaches and sound horrible is because of a totally different issue known as "The Loudness War" (I'm sure you know about it so I won't go into details). However, I do agree with you as far as compressed audio goes. Unfortunately, there's a reason for that and there's nothing we can do about it until the day Internet bandwith becomes more accessible and cheaper. Eventually it'll get to the point where uncompressed audio can be streamed reliably through the net. But, until then, we're stuck with MP3, AAC, DTS and other audio compression formats. As far as digital media distribution goes, it's the future and companies are seeing that. More and more people download music rather than buying CDs, so I do believe those numbers are accurate. Just look at sales from iTunes and even games like Guitar Hero and Rock Band. It's just a matter of time.
Take care!

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Inconsistent audio sample rates....?

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Also check this [Link|http://help.sap.com/saphelp_45b/helpdata/EN/0d/d30b734a0c11d182b80000e829fbfe/content.htm]
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Edited by: Bhavik G. Shroff on Sep 21, 2009 2:39 PM
Edited by: Bhavik G. Shroff on Sep 21, 2009 2:44 PM

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