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Join Date: Nov 2003
Location: West of Seattle
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Audiophiles -- Ogg Vorbis? Flac? MP3?
I've just upgraded my stereo system (NAD C320BEE, Infinity Primus 360s), and have suddenly discovered that compression quality really makes substantial difference in output quality. 128kbps mp3 is painful to listen to on the new system. For the most part, Ogg sounds pretty good at the higher quality levels, and FLAC seems like overkill, but definitely sounds fantastic. I'm hooked on the sheer convenience of having my entire music collection on the computer rather than having to swap CDs around all the time, but I'm struggling with formats.
Assuming that space is no issue, and that I have sufficient time to transfer all of my CDs onto the hard drive(s) in a new format, what's the recommendation from those who have been here before? Any technical details that are non-obvious in the ripping process? Any "lessons learned?" As they say -- only a fool learns from his own mistakes -- a truly wise man learns from the mistakes of others. ![]() Thanks in advance, Dan
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B58/732
Join Date: Feb 2000
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I usually go with 192 or 256 VBR MP3, depending on what I'm listening to. I'd use Ogg Vorbis if my iPod would only speak it.
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256kb/s MP3:s are impossible to differentiate from raw PCM, they have done numerous A/B tests using best hi-end equipments and most prominent audio geeks.
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The big advantage to OGG is that the format is open - and the ogg files can contain either just audio, just video, or both (ogg vorbis, ogg theora, etc)
FLAC is lossless, MP3 is lossy. Thats why the FLAC sounds sooo good and the 128kb/s mp3s don't. Either go with high rate MP3 (256 or 384 k/sec), FLAC, or uncompressed WAV (you said space wasn't an issue)
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Ian
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Alter Ego Racing
Join Date: May 2002
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If you really care about the sound (IMHO) just lsiten to the source, do not convert it to any format unless you were to upconvert it.
I use my Ipod for everyday listening but come back to my CD, SACD, DVD A... and my trusty transport and DAC if I'm serious about listening one day.
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Banned
Join Date: Apr 2005
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I do 256k mp3 and it sounds fine on my Polks....
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Quote:
Speaking of "upconverting" .... what exactly do you mean? Are you saying that I could take CD data and resample it to get better sound? ![]() ![]() Ogg sounds pretty good to my ear. I think I can tell a difference between the highest quality Ogg and the FLAC, but I can't convince myself that I'm not just hearing stuff. The midrange is fantastic, even in 128kbps mp3s -- I can hear the tackety-noises of the clarinet keys, for example. I'm still experimenting with mp3s. 256 should be "good enough," you say? Ok, I'll play in that range... Should it make any difference what player software I'm using, or is everyone using fundamentally public sources for the engines these days? I assume that the sound card makes a difference? Any recommendations? ... I love putting the PPOT crowd to the test. You people know all kinds of random stuff. It's like a pool of knowledge, and I'm just here with my fishing pole...
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Software doesn't matter - the mp3 compression algorythm doesn't change. As for sound cards, yes they make a difference. Not sure what the State of the Art is these days, I still use my old soundblaster with the extra RAM on it
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M-Audio cards are very good. Audiophile 192 or 2496.
I have a 2496 & use it spdif out to a processor for a PC driven front projection home theater. Sounds fine with CDs too but a good dedicated CD player will beat it. I stress the good part. M-Audio Ian
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There's now a 2496 on the way (thank you, eBay). Thanks for the hint on gear.
id10t, is that "algorythm," as in "algorythm and blues?" ![]()
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Alter Ego Racing
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Here's an extract of an article I got with my Simaudio Moon cd player which goes into details.
On your problem with having to change cd's, I used to copy music into SVHS tapes for "longer playing sets. Whilts the sound was not as good as the CD it was superior than I now get with my Ipod (IMHO). Here's some good reading (which might be copyrighted and should be credited to Simaudio): Basic Digital Audio Theory Let’s start with the basic Pulse Coded Modulation (PCM) theory. PCM is a method of converting an audio signal from its native analog format into the digital domain where it is comprised of only zeros and ones. This is done on a Compact Disc (CD) for example, and then the original analog signal is recovered by playing the CD in a CD player. PCM works as follows: The analog signal is defined by a value on two axis, amplitude and time. At the sampling frequency, 44.1kHz for CD audio, the amplitude of the signal is encoded into 16-bits words. This means that every 1/44,100th of a second, the closest numerical value to the amplitude of the original analog signal is stored. Since 16-bit words are used, there are 65,536 possible values (216). When the analog signal is recreated using a digital filter followed by a digital-to-analog converter (DAC), found inside every CD Player, all of these encoded numerical values are restored to their original amplitudes with respect to the time axis, recreating a signal with a staircase shape; each step’s limit corresponds to the finite value in time and amplitude of the encoded digital signal. This signal is then sent to a reconstruction filter that “smoothes” out the staircase shaped signal, creating an output signal without all the jagged steps. This result is an analog signal, which should closely resemble that of the original analog signal. What are the limitation of this system? One limitation results from the approximations required to encode the real amplitude of the signal into a discrete value. In reality, the signal always lies between two consecutive levels. This causes noise in the digital domain and the limits dynamic range; with more bits, you have better amplitude precision and an equivalent increase in dynamic range. Another limitation is the sampling frequency. With a higher sampling frequency, you have more samples of the signal in one timeframe. This extends the available bandwidth of the system. Imagine a decrease in the size of the steps as the amplitude resolution (number of bits) increases and/or sampling frequency increases. A finer staircase shape (less jagged appearance) allows the use of a simpler reconstruction filter, which is directly related to sound quality. The less effect the filter needs to have, the less it affects sound quality. In more scientific terms, the sampling frequency dictates the highest possible frequency the system can reproduce. Nyquist has demonstrated the mathematical law which states that a sampling system can reproduce frequencies of up to half the sampling frequency. The system has to filter out all frequencies above half the sampling frequency, which is where unwanted artifacts of the digitized signal (the staircase) reside. The audio CD has a specified bandwidth ceiling of 20kHz, leaving a small gap between that upper frequency and half the sampling rate which is 22.05kHz. The filter must be very steep (high order) to remove information above 22.05kHz but still leaving information under 20kHz. Such a filter was developed at the inception of CD playback and was named the “brickwall filter”. This filter had a terrible impact on sound quality. Oversampling The oversampling technique was developed to get away from a “brickwall filter”. A digital system interpolates new points between the different original samples to obtain an artificially higher sampling rate. This allows the use of a less aggressive filter because it doesn’t have to eliminate frequencies as close to the frequencies it must not affect. It first began as four times (4x) oversampling (i.e. 176.4kHz), then later eight times (8x) oversampling (i.e. 352.8kHz). The digital filter must perform many mathematical calculations to determine the value of the point it must add to the original digital signal. Often, this calculated value may fall between two discrete values, so the oversampling system must round off the value to the closest discrete value. To increase the precision of the resulting calculated value, DACs and digital filters with more than 16-bits of resolution were therefore introduced. We have seen 18-bit, 20-bit and 24- bit digital filters and DACs. It is important to note that oversampling creates an artificially higher sampling frequency, which does not extend the real frequency response of the original media or the system, but simply extends the frequencies that need to be filtered out, allowing for a simpler and better sounding analog filter. Upsampling & Upconversion One of the latest storage mediums is the popular the Digital Versatile Disc (DVD). When developing this new standard, a higher-than-CD resolution PCM format was adopted with a maximum resolution of 24-bits/96kHz. For the professional market, this new format had to be compatible with the CD's 16-bit/44.1kHz resolution. This would allow the conversion of original recordings to the new standard. So a sample-rate converter chip, which is nothing more than an oversampling digital filter, was created to actually convert any digital signal from one standard format to another format. For example, a 16-bit/32kHz signal can then be converted to 24-bit/96kHz and 24-bit/96kHz can also be converted to 16-bit/48kHz. This gave rise to the marketing hype with the concepts of upsampling and upconversion, which claims could upsample or upconvert your 16-bit/44.1kHz CD to a 24-bit/96kHz resolution digital signal prior to the digital to analog conversion, resulting in DVD-audio like quality from CD. While this statement is a great idea for marketing purposes and is surely impressive to most consumers, it is technically only half true, and is not the best way to improve the audio quality that can be derived from CDs. Why? Digital filtering is digital filtering regardless of name assigned to it, and how the interpolation is made still relies solely on the arithmetic calculations implanted in dedicated hardware or software. The main difference is how well the “mechanics” of the mathematics will assist in the signal’s reconstruction. When changing the sampling rate, it is better to maintain an integer multiple of the original signal’s sample rate, so the processing is kept simple. More importantly, the end result is more accurate, thus enabling a higher fidelity of sound reproduction. A two times (2x) oversampling system will double the sampling rate, by adding one easy to find numerical value in between each actual sample. For example, when a 44.1kHz digital signal is processed, a 88.2kHz digital signal is obtained. It is simple, effective and precise because it is a direct multiple of the original digital signal. For an upsampler to make a 96kHz digital signal from a 44.1kHz signal, it will have to perform awkward mathematical operations to obtain a 96kHz signal. (96kHz / 44.1KHz equals 2.1768707…). This results in a less accurate output from the digital filter, with everything else following (i.e. digital-to-analog conversion and analog filtering) also being less accurate. As well, exactly like oversampling, the artificially higher sampling frequency created by an upsampler doesn’t increase the actual frequency response of the system, but simply increases the lower limit of the frequencies that need to be eliminated. What does all this really mean? You can hear the differences between the various types of digital filters, regardless of the marketing names used. Over the past two decades, we have witnessed vast improvements in both digital filters and DACs. It is demonstrably true and there does exist real progress. However, the “latest and greatest” upsampling method is not necessarily better than the classic oversampling method. In fact, and most probably, these latest methods actually deteriorate sound quality if the conversion takes the sampling rate to a frequency that is not a direct integer multiple of the original sampling rate, being 44.1 kHz for audio CD.
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Any true audiophile will absolutely shun ANY compression technique which isn't lossless imo. If you can't hear the difference, then you need a better system
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One more thing Dan. If you intend to RIP all of your cds to your pc, and you want to ensure that you are getting a 'bit for bit' copy, then you simply MUST use something good like EAC (Exact Audio Copy), and use it properly, otherwise, you'll likely loose bits in the process. Audio CD formats are not like computer data files (i.e. .wav, FLAC, SHN, etc) and contain no check sum (ensuring data integrity) algorithms. That's a whole 'nuther topic of discussion....
Caveat...All of this probably doesn't matter to the average listener, but you did direct you question to 'audiophiles'...and we're pretty anal about this stuff ![]() |
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Nothing sounds better than a good vinyl record.
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I agree with kach22i. I have a difficult time listening to CD's on a really good system- vinyl wins hands down. I have done the test with several audiophile friends- then they believe. Here is a simple test to those of us old enough to remember vinyl records. Remember when you would put the same album on and play it loud over and over again. Try that with one of your CD's- my bet is by play #2 you are looking for something else to do. CD's are not musical. Now the new formats such as SACD are coming along and are very nice.
Yes I listen to my ipod, actually more than my audiophile system. But listen as close to the source as you can, no compression, no copies. Your source will make more of a difference, once you reach a certain level, than your other equipment. Listen more Gary |
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For all of my music I use 320k non-VBR MP3s.
I have a set of Polk Audio Signature Series speakers that are very clear (you can hear things on them that you would never hear on a cheap set of speakers, especially in the background, etc.) The 320k is basically impossible to tell from a CD, while 128k is very very obvious. I use all direct rips, no downloads, and have 2000+ in my collection. It just fits on a 250GB drive right now with not much free space. If you do decide on MP3, I really can't see any reason to not use 320k with the prices of storage. You can get a 250GB Drive and a USB2.0 External enclosure for it for around $150 (cost of about 10 CDs) and store 2000ish discs on it.
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I don't disagree with you guys at all regarding vinyl sounding the best 'IF' the source is pristine. The downside to vinyl is that each subsequent play is wearing the grooves a bit, and after hundreds or thousands of plays (even if you meticulously clean it each time), they don't stay pristine. That being said, everything in the digital realm is simply trying to emulate what the music is originally...i.e. analog, so it's all about compromises. I have several thousand cds (many of them are EXCELLENT recordings of live shows, so vinyl isn't even a option). Gary, I hear what you're saying about listening to a CD as being 'tiring to your ears', and that was certainly the case back in the mid-80s, but the DACs have come a LONG way since then. I can still do with cds what I used to do w/ vinyl and it still sounds SWEET over and over again... it's all about the system. Rob, my primary speakers are vintage Polk SDA 1.2s (170 lb towers w/ 13 drivers each) with each tower being driven by a 1000 watt amp (two amps, bi-amped left/right), so I know what ya mean. For convenience, I have 5 multi-disc players (total capacity: 1300 cds) all w/ digital connections to my preamp (again, its all about compromises). I built this system several years ago (before many of todays options were available), but everyone that's ever heard my system is amazed. As far as compression goes, there's absolutely nothing inferior as long as it's 'lossless compression' (i.e. FLAC), as what comes out is EXACTLY the same as the original uncompressed source. (MP3 is another thing entirely however). Also Rob, when you say you use 'direct rip', what do you mean? Unless you're using EAC (or something similar), there is no way to guarantee that the copy is 'bit for bit identical' to the original. Again, due to the inherent limitations of the red book audio format of CDs (they do NOT contain any type of check sum algorithms), occasional 'bit errors' are in fact 'the nature of the beast'. As you might can tell, this stuff is also a passion for me, and we could debate this forever, but I'd rather discuss Steve Wong chips with Loren
![]() Last edited by KFC911; 12-16-2005 at 09:13 AM.. |
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Yeah, mine are SDAs also, but the 4ft, 100lb "baby" ones.
![]() I didn't mean direct rip bit for bit, I was just meaning to say that everything in my collection comes from a CD that I personally cleaned and touched. No downloads, no files from friends, etc. And yeah, MP3 is lossy, but I just don't think my ears are good enough once it gets to the 256 point, 320 is just for good measure. I think people underestimate good speakers comparatively with other things as well. There are certain things about lossy compression that good speakers point out and other things that bad speakers point out. And some of the new formats are amazing. Last year at CES I was treated to a demo of the new Bang & Olufsen BeoLab5 speakers. They had 4 set up in the Hotel Room as a surround system. I have never heard anything so loud in my life that didn't create even the slightest ear fatigue, it was amazing. 2500w per speaker built in, and they have a mic that sets their levels automatically based on where in the room they sit with the touch of a button. And just to prove to us that they weren't like the older models that lacked on bass response, he switched from the Dave Matthews and Classical he was playing to Insane Clown Posse and it was just incredible. Unfortunately, at $37k for a set, I won't be getting any anytime soon. Needless to say, it was DVD-Audio, not MP3 at that point. ![]()
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