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Rick, Are you able to capture the file onto your Mac or PC from the iPod via the USB interface ? Then it would be possible to determine if the iPod is doing the ALAC decoding correctly. (It seems you are confident that the stored file is OK). Chris.

Gordon, Rick, What is the form of connection between the iPod and the Alpine unit ? If it is just USB (ie. digital connection) the problem can then only be incorrect decoding of the ALAC file (by the iPod) or sample timing (ie. jitter in the Alpine's DAC and/or USB interface). There isn't anything else to get wrong ! If however, the Alpine unit is actually taking the analogue output from the iPod, then you are at the mercy of the DAC in the iPod, which won't have very good jitter performance. Again, it could be that the iPod isn't correctly decoding the ALAC file, but I doubt that. Apple will almost certainly be using the same algorithm as they use in iTunes. Chris.

Hi rom661, Assuming that nothing went wrong during the ripping process, the playback bitstreams from AIFF or ALAC should be identical. Lossless encoding is by definition, lossless, so will be bit-for-bit identical to the original. So this only really leaves jitter, and I can see that this could be worse using ALAC on the iPod. In decoding an ALAC file, the processor in the iPod is having to work a lot harder than for AIFF, and it will therefore draw more current. This in turn will cause supply voltage fluctuation, and in turn clock jitter. So if the Alpine unit doesn't do any jitter reduction (highly unlikely), its DAC will do a worse job with ALAC. With AIFF, the iPod doesn't have to do much, other than write the L and R samples out to the USB port, so it is only lightly loaded. I can't think of any other cause. Chris.

Matt, One more option for compressed files (for iPod or suchlike) would be to use the Variable Bitrate (VBR) option with AAC. This dynamically varies the bitrate according to the music content - increasing the rate if there is a lot of high frequency content. Typically, AAC-VBR increases the file size by ~ 10%, but gives a significant increase in quality. AAC-VBR at 128k would sound better than MP3 at 192k. Chris.

Hi Matt, Unless you have a particular reason for wanting MP3, you would be better off using AAC, as it is a more advanced codec (part of the MPEG4 standard). AAC at 128k is going to be pretty similar to MP3 at 192k. Are you pushed for storage space, as if you want the best quality it would be better to use a lossless codec such as FLAC or Apple lossless ? That said, in tests I made a while ago, I couldn't tell the difference between AAC at 256k and lossless ! Chris.

If the CD sounds good when played using your PC, that suggests the OS and replay software are OK, so the finger points to the ripping software. I'm not familiar with dbpoweramp, as I only ever use iTunes for CD importing, and that gives a bit-perfect copy (on a Mac). Have you tried iTunes for the PC ? Chris.

Earlier in this thread, I noted that not many people can hear much above 16kHz, and I certainly can't. It would therefore be very interesting to make a recording at 44.1kHz sample rate, but with the audio bandlimited to 16kHz to begin with. This would allow a very benign filter characteristic to be used, both in the recording ADC and in the replay DAC. I suspect that 16 bits would be quite sufficient too, although it would make sense to use 24 bits for the initial signal capture and editing. As a related aside, it is also worth noting that VHF FM stereo transmissions (in the UK at least) use digital links between the studios and the transmitters, which operate at 32kHz sample rate and 13 bit resolution. Provided you have a good tuner and a fairly strong signal, it can sound really quite good. Chris.

Hi Chris, Yes, you are right that the majority of DAC chips these days are based on the Sigma-Delta principle, which always contain internal oversampling (brick wall) FIR filters. Typically, these increase the sample rate to 8x the input rate, which is normally enough to allow the Sigma-Delta modulator to work properly. But many DAC box and CD player manufacturers often also use sample rate converter (ASRC) chips in front of the DAC chips to increase the sample rate by a non-integer factor to the typically trendy 192kHz. ASRC chips effectively also contain a brick wall interpolation filter function, so you now have two such filters in series. There is some evidence that the resultant excessive time dispersion (due to the two filters) is audible and undesirable. However, the plus side of the ASRC chips is that they contain narrowband digital PLLs, which greatly reduce the clock jitter fed to the DAC. My opinion on this is that a better solution would be to dispense with the ASRC, but use reclocking to remove the jitter, then you only have the one brick wall filter (in the DAC itself). If you are using SPDIF, a good way to do this is to use one of Wolfson's SPDIF receiver chips. Unlike every other SPDIF chip out there, the Wolfson chip really does suppress jitter. Chris.

Re: "My DAC is only 44/16, but I'm thrilled with the sound I hear from it. I may be a purist, but I don't like the thought of upsampling standard resolution." I tend to agree. The current trend to use upsampling is a bit of a red herring, IMO. Some of the (allegedly) best sounding DACs are non-oversampling. And from a purely engineering perspective, DACs get less linear the faster they have to operate. Chris.

Hi Crion, I think the problem here is that you need the 'near brick wall' filter to do the upsampling anyway, so the damage is already done. It could be that they are using an apodizing filter which does encroach on the 20kHz passband somewhat, thus achieving the desired raised cosine (or whatever) shape. It would be interesting to get hold of Bob Stuart's AES paper on the subject to find out the details. Chris.

Re :- "The improved filtering (which works to mitigate/eliminate pre-echos and ringing from 44.1kHz/48kHz sources) is performed within the upsampling stage of the new 808.2 CD-Player and the DSP7200 Speakers." I think that the Meridian marketing machine has got itself confused there ! An apodizing (or windowing) filter must, by definition, start to roll-off within the passband. So in the context of audio, they can only really be used at the higher sample rates. At 44.1 or 48kHz, there isn't enough guard-band between the wanted band and the closest alias to fit a useful windowing function in. True, it is a good argument for higher sample rates such as 96, 176.4 or 192kHz, but isn't a viable technique at 44.1 or 48kHz. Chris.

Actually, I think that even 21 bits and 70kHz is overkill. Unless you are a very small child, or a cat, most people can't hear much above ~16kHz, so on that basis 44.1kHz is just fine. But what you really need is a DAC with an output filter that cuts off at 16kHz. None of them do of course, because of the industry's obsession with specmanship. A dynamic range of 21 bits (128dB) is probably also over the top. Here's a little known fact (in audiophile circles, anyway) :- The instantaneous dynamic range of the human ear is typically ~30dB. Yes, really ! The other 90dB or so is achieved with signal processing in the brain, and is mostly an AGC effect. It is this property which is exploited in lossy codecs in the form of their psycho-acoustic masking. So 24 bits is somewhat excessive too. I think a perfectly adequate spec would be 18 bits and 50kHz. Chris.

Re :- http://6moons.com/audioreviews/whitelightning/moonshine.html They would be OK for speaker cables, but not good for interconnects as they are completely unscreened, so wide open to RFI.

Hi Peter, Those cables look ideal, and very sensibly priced too. Chris.

Hi Chris, If you look at the conductivity 'league table' for metals, silver is best, copper 2nd and gold 3rd, but there isn't a huge difference between them. The gold plating is necessary to avoid oxidation. It is widely used on RF connectors where the frequencies involved can be many GHz (ie. ~ 1,000,000 times higher than audio), and there are absolutely no adverse effects. Chris.