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lazycatken

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  1. My idea is to use Mac mini as a headless music server. I can set iTunes to automatically ripping CD, but need a way to eject CD! If manually editing metadata remotely is possible, that will be great! I can't realize why Apple Remote can't show/play the CD inside the controlled Mac...
  2. The hardware of EMU-0404 can support up to 24bits, thus, set 24 or 32 are identical. BTW, EMU-0404 USB is sensitive to power. You had better use an isolation transformer.
  3. I'm using EMU-0404 USB. It's a good product, good C/P ratio. But just found it's sensitive to AC power supply, just like all other audio equipments. You had better supply it via isolation transformer. BTW, you don't have to learn how to use those bundled recording softwares. Just use Audacity, if you Mac.
  4. That could be possible.... One of my experiences, I ever tried to check the output of iTunes on my MacBook Pro 15" OS X 10.6.3, default system output is SoundFlower, i.e., iTunes outputs SoundFlower then my program check the stream from SoundFlower. I found that it's strange, while tracks on hard disk was playing, the tail of every track was not silence as it should (I checked digital data), but this did not happened while tracks on CD was playing. The signal added onto silence looked like dither(under -80dB, I can't remember actual value)... The above problem happened only when iTunes is launched while OS default output device is a 16bits device. SoundFlower is always 32bits float format, the OS internal format. Perhaps, Gordon could give us more information...
  5. Perhaps, this graph can give you some idea about what might happen in your system. http://2.bp.blogspot.com/_QtIX994Ul0c/SwtYPwjxeZI/AAAAAAAAAxQ/g2ZyyLH3NvY/s640/AK4380+88.2K+IR+high.png The upper is impulse response of an ideal DAC, ie. digital signal, the lower is the one of my DAC driven by my Mac Book Pro optical output. Looks so bad... You can expect the sound of the lower one will be blur, not precise, too much bass.... Room correction can do impulse response correction, not just amplitude frequency response correction. This is the impulse response of a system, measured in listening position. http://2.bp.blogspot.com/_QtIX994Ul0c/SreRwfIj1FI/AAAAAAAAAlQ/4YsHJfF2OI4/s400/IR-original.png And this is the simulation result of correction done by DRC. http://1.bp.blogspot.com/_QtIX994Ul0c/SreRhpNtjMI/AAAAAAAAAlA/jjFFWezm7Yo/s400/IR-sim5.png
  6. "I would suspect that adding the m2tech would improve the sound of the DAC in the receiver since it is probably more sensitive to Jitter than some of the newer DAC designs." That's true. But how's the price of those newer low-jitter DACs? And, the question of SoutherMan is "Low-end recommendation"...
  7. I would recommend you take a low jitter S/PDIF, it will cost least, but gain most. This is the impulse response of Accuphase DP-500 driven by my Mac Book Pro http://4.bp.blogspot.com/_QtIX994Ul0c/SwtYZFmCBtI/AAAAAAAAAxY/soEcICje--Q/s1600/DP-500vsEC.png Looks terrible...What's the hell? That's the result of phase shift and the root cause is jitter. This is phase frequency response of Philips CD482 vs NAD502, both cheap and old . http://2.bp.blogspot.com/_QtIX994Ul0c/Sz9UwrpSWRI/AAAAAAAAA6s/EYs3lz449Dg/s1600-h/NAD502+FR+phase.png CD482 looks much better. That's why it's famous of good C/P. This is impulse response of ideal vs my MacBook Pro drives AK4380 DAC, which is based on AKM AK4380 evaluation board. http://2.bp.blogspot.com/_QtIX994Ul0c/SwtYPwjxeZI/AAAAAAAAAxQ/g2ZyyLH3NvY/s1600/AK4380+88.2K+IR+high.png Looks the same as expensive Accuphase DP-500. And this AK4380 driven by EMU-0404 USB, optical & coaxial. http://4.bp.blogspot.com/_QtIX994Ul0c/SwtY-86TH3I/AAAAAAAAAyA/siqzDGLwHjg/s1600/EMU-0404+Toslink+vs+Coaxial.png Very close to ideal! Thus, we can see, even the expensive Accuphase DP-500 does not handle any jitter from S/PDIF. You know they just take a S/PDIF receiver chip, which costs a few dollars, that's all... What's it different with your AV amp? Just better power supply, which cause some improvement, but not much on measurement... All of us agree that Human ears are much sensitive than measurement. In other words, any difference in measurement result will cause hugh difference hearing by ears. Here's some posts on my blog, sorry, it's in Chinese, but you can check graphs. http://records2ear.blogspot.com/2009/12/jitterdacimpulse-response.html http://records2ear.blogspot.com/2010/01/denon-dcd-1650aesony-xa3essony-scd-1-vs.html http://records2ear.blogspot.com/2010/01/philips-cd482-vs-nad-502.html
  8. Just had a glance at Apple sample code "PlayFile". Here's the section where AudioUnitGraph is setup //////////////////////////////////////////////////////////////////////// // output node cd.componentType = kAudioUnitType_Output; cd.componentSubType = kAudioUnitSubType_DefaultOutput; cd.componentManufacturer = kAudioUnitManufacturer_Apple; AUNode outputNode; XThrowIfError (AUGraphAddNode (theGraph, &cd, &outputNode), "AUGraphAddNode"); // file AU node AUNode fileNode; cd.componentType = kAudioUnitType_Generator; cd.componentSubType = kAudioUnitSubType_AudioFilePlayer; XThrowIfError (AUGraphAddNode (theGraph, &cd, &fileNode), "AUGraphAddNode"); /////////////////////////////////////////////////////////////////////////// Default Output audio unit is invoked, but without any setting. File Player audio unit is set as the format of the file. Looks like, all the conversions are done by Default Output audio unit. This a simple way to play a file. Does iTunes do this way? Another way is to do sample rate conversion by application. Audiotoolbox SDK provides a function to create an audio converter object. /////////////////////////////////////////////////////////////////////// AudioConverterNew Creates a new audio converter object based on specified audio formats. OSStatus AudioConverterNew ( const AudioStreamBasicDescription *inSourceFormat, const AudioStreamBasicDescription *inDestinationFormat, AudioConverterRef *outAudioConverter ); //////////////////////////////////////////////////////////////////////// The property of a converter can be set, including these two: /////////////////////////////////////////////////////////////////////// Sample Rate Conversion Quality Identifiers Specifiers for sample rate conversion quality, used for the kAudioConverterSampleRateConverterQuality property. enum { kAudioConverterQuality_Max = 0x7F, kAudioConverterQuality_High = 0x60, kAudioConverterQuality_Medium = 0x40, kAudioConverterQuality_Low = 0x20, kAudioConverterQuality_Min = 0 }; Sample Rate Conversion Complexity Identifiers Specifiers for the sample rate conversion algorithm, used for the kAudioConverterSampleRateConverterComplexity property. enum { kAudioConverterSampleRateConverterComplexity_Linear = 'line', kAudioConverterSampleRateConverterComplexity_Normal = 'norm', kAudioConverterSampleRateConverterComplexity_Mastering = 'bats', }; ///////////////////////////////////////////////////////////////////////// So, questions: If iTunes do sample rate conversion, what level of quality it uses? If iTunes doesn't do sample rate conversion, this means, it's done by Default Output audio unit, , what level of quality that Default Output audio unit uses?
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