syganymede

I hope everyone has read this from Apple: images.apple.com/itunes/mastered-for-itunes/docs/mastered_for_itunes.pdf It is a first attempt to improve the quality of iTunes music, and not before time. However I think, just as many before them, they have fallen into a trap. It is just fine to develop these sophisticated tools that do improve the signal processing. But the fundamental error is that it is not correct to limit the sound chain to what the human ear can hear, what is needed is a sound chain that can reproduce the sounds made by the original instruments, and these have very high harmonics, up to 50-100KHz. Let's hope that Apple's announcement of iTunes+ is just a first step in persuading the music industry to up its game and release 24/96 music in non-compressed file formats.

I have been looking for a pair of "desktop"-size speakers with built in Apple Airplay connection. I can find ones at £300-400, with far too many features that I don't want - including horrible DSP processing and remote controls etc. What I want is a basic, simple pair of speakers, round £100, with the focus on audio quality not on fancy features. Any ideas?

I was faced with the same problem of how to interface my Apple TV TOSLINK digital output to my Spendor loudspeakers. After a bit of research I made up a homebrew amplifier with these components 1 MUSE audio Mini TDA1543 x 4 DAC - Hong Kong for £35 2 Two Hypex UcD180 amplifier modules (Class D, 100-150W) €170 3 A power supply module - rectifier, smoothing 4 x 4700uF from Hong Kong £15 4 A toroid transformer 2 x 22V out, 160VA all in a small box. Works very well. If only our Chinese friends would get on the job and copy this, I bet we could have it retailing at £99 or so...

My understanding is that OSX 10.6 does audio processing at up to 32bit, OSX Lion does it at up to 64bit in Core Audio. But that iOS used on the iPod, iPad, iPhone is only a 16 bit OS, and so audio processing is limited to 16bit... If confirmed this is the case then it might explain why the quality from iPods is poor, in addition to the use of AAC lossy audio. Even if you use Apple Lossless the processing and DAC are still 16bit... It also might account for the volume control being so poor, as it has only 16bit processing to handle a 16bit audio word. Thus turning down the volume is essentially just shifting the high end bits lower and truncating? Anyway the audio quality of iPods etc is poor, and doesn't bear comparison to a good lossless 24/96 recording played back through a wide band system.

Just seen the article in Macworld (http://www.macworld.com/article/161078/2011/07/how_to_rip_audio_from_your_bluray_discs.html) describing a tortuous but workable way ti rip audio from Blu Ray disk. About time this problem was solved. Blu Ray is one way to get HD audio, even if studios don't seem to want us to have it.

Can I add a couple of things? 1 As far as 64bit is concerned you can see what runs in this mode by Apple Icon > About this Mac > More Info... 2 What I found is that Quicktime, AudioMIDI and Core audio all run in 64bit. But iTunes, sound studio and soundflower are 32bit. I don't have Ayrewave or Amarra but those who do can check to see what it says. 3 As far as I can work out OSX is 32 or 64bit, as above. But iOS is only 16bit. I believe, if this is true, then this is the reason why the iPad and iPhone and Apple TV are poorer than any MacBook at audio reproduction and cannot handle 24/96 or so. In particular the iPod and iPhone volume controls do a poor job of reducing the volume as they truncate the 16 bits, whereas a 32bit floating point solution like Core audio performs much better for the digital volume control. So don't use iPad, iPod or iPhone for playback. Use Remote to control your Mac. 4 Streaming to Airpport Express and Apple TV is at 16/48 I believe and they do not have volume controls - much as people would like the Apple TV to have one! It would result in very poor audio.

When you turn down the volume digitally it is like changing from 16 bit to 15 or less bits. For example you might just shift the 16 bits to the right putting bit 16 into 15 etc. Thus you throw away the LSB, i.e. bit 1. To overcome this some software does two things, uses 24 or 32 bit to increase resolution, andh aving done that randomises, or dithers the LSB. This is why the volume control in iTunes on a Mac, sounds better than the volume control on an iPad. If you don't want to lose resolution (i.e. number of bits representing your sound) then use an analog volume control. You do not need an active pre-amp with any gain, provided the output of your DAC is big enough to drive your power amp, thus the volume control only has to attenuate the signal. So what you need is just a potentiometer in a box! Say 10$ or so...

I am fairly familiar with digital volume controls. On the one hand they just shift the bits right, cutting off the lower ones. Then again some shift right then dither the LSB. Am I right? My question is how does Mac/iTunes handle this versus iPad/iPod app? For me the Mac iTunes volume control sounds much better than that on the iPad when not at max. Is it that on the Mac we have more processing power and 32it Core Audio, but on the iPad we don't have suh powerful CPU's?

It seems to me that the iCloud choice of AAC lossy files is going to have a longer term effect on how the mass market buys its music. Major deliveries today are by MP3, AAC, CD and FLAC. With most at 16bit/44.1kHz, except FLAC for 24bit/96-192kHz. But Apple's dominant choice of AAC could have an infkuence on the sales of CD's which are higher quality through being uncompressed, and thus lossless. If this happens the whole market will move to a lower, not higher, reproduction quality. So what would be nice to see 1 A move to using 16/48 for AAC files = better dybamic range 2 An offer of 24/48 FLAC tracks for better dynamics, timbre, etc, this being the logical upgrade from CDs 3 A HD offering at 24/96-192 for "aiming high" users. What do you think? Is the CD dead?

We need some answers about bit rate and word lengths in the Apple eco-sphere. This is the source material we have today: CD 16/44.1 -> iTunes rip at 16/44.1 CD 16/44.1 -> iTunes rip at 16/48 HD audio 24/96 -> iTunes 24/96 downloads (maybe in FLAC or WAV converted to AIFF or ALAC?) On the Mac lots of things happen with audio: 1 Conversion programs are available for CD ripping and file format changes. I have a small contribution to this part of the confusion, I purchase FLAC files and convert to AIFF using the program XLD, which seems to me to be ideal. Others may have different solution. iTunes also has a good reputation for direct CD ripping. 2 The setting of AudioMIDI (number of bits and rate) can be changed for the output up to 24/96, what effect does this have? I know for certain you have to restart iTunes if you change the settings as it reads them only once at startup. 3 Playing a file with a different bits/rate may or may not up-/down-sample. For example with AudioMIDI set to 24/96, what comes out when you play a 16/44.1 file? And vice versa, play a 24/96 with AudioMIDI set to 16/44.1, what comes out? Is it the setting of AudioMIDI that defines the output itself or the setting iTunes reads from AudioMIDI when it starts up? For example if I start iTunes with AudioMIDI set at 16/44.1, then change it to 24/96, does iTunes continue to output 16/44.1 or does the system up-sample to 24/96? And vice versa. 4 Now the confusion gets worse if you stream the files to an Airport Express or New Apple TV. What conversions does iTunes make, does AudioMIDI have any effect, what is the bits/rate spec for the stream,is this different for Ethernet wired or WiFi connections, what conversion do the AE or ATV do, if any, and what does it output for each of the stored file types. We need a complete and clear description of all of these issues so we can resolve the many, many discussions that are on the forum. Any volunteers to prepare the manual?

I believe that the output of the new ATV is at 48kHz as this is the maximum allowed by the DVD spec for external audio output, via the HDMI it is allowed to transfer up to 24/192. This limitation is to stop file copying at high resolution. Any comments?

Audio streams are first encoded using codecs such as MP3 or iTunes AAC, then streamed using standard internet protocols. So the disadvantage of lossy MP3/AAC compression still exists and we don't get any better audio quality. Just open access, on a subscriber basis, to huge libraries. Think Spotify! It is perfectly possible to stream CD quality, or even 24bit/96khz files, but the bandwidth neede would be much increased. However we all seem happy to sign up for ISPs that let you happily watch BBC iPlayer, which consumes enough bandwidth to stream HD Audio, e.g. FLAC HD, that I do wonder why someone doesn't start a web site with HD Audio streaming...

Have a look at the Audio-GD Sparrow DAC. I uses the very best IC Chips and should give great sound. For just $240.

I have been looking into the need for a simple amplifier for this application for over a year. What I have come up with is a choice of how to design it, but I do not have the ability to make it. If anyone is interested this is the line up I would propose - style, vertical square tube matching the new Apple TV - SPDIF only input at up to 24/96, OK so the Apple TV today only supports 16/48 max, but one can hope... - chips to use, TORX147 - WM8805 - WM8742 - hi end Op Amp for filters - Hypex class D amp modules. Non switching power supply. - target is to have virtually DC to above 45kHz bandwidth. Price around $600. Sorry if this is a bit technical, but I do believe that we have to be aware of the best in semiconductors to start to chose DACs and amplifiers. If anyone is interested to persue this idea, contacf me.

There's a lot of discussion about the middle of the system. Recording standards (studios systems, A/D conversion), distribution systems (CD, SACD, downloads...), and reproduction electronics (music servers, DACs). But the digital part of these is susceptible to accurate analysis, what people often refer to as "bit perfect", where what you get out of your computer is what the A/D convertor in the studio put in. Apart from choosing the distribution system's bandwidth (16/44, 24/96 etc), that part is the easy one.<br /> <br /> What is more important is the beginning and end of the system<br /> <br /> - Microphones, preamplifiers<br /> - Power amplifiers and speakers.<br /> <br /> Especially important are the transducers (mics and LSs).<br /> <br /> How about some focus on these?