Pure Music

[Lars]

Channel D has released a player only version of Pure Vinyl called Pure Music. $99 but on sale for $79 through March 14th.

 

http://www.pure-music-player.com/

 

[The Computer Audiophile]

I just downloaded the demo version. It's changing the sample rate but not the bit depth. I'll wait until it outputs bit perfect audio before trying it again.

 

[davidada]

Hi,

Do you know if amarra does both?

 

[Lars]

Pure Vinyl and Amarra do both. I haven't checked this other program yet.

 

[Nyal Mellor]

Poo 10.5 only

 

I really need to get off 10.4!!!

 

Anyone have a spare upgrade disc (PPC mac)

 

[bottlerocket]

On the bit depth (word length), I find that neither Amarra Mini or Pure Music changes the bit depth. For example if I play a 24/96 track and then change to a 16/44.1 track, I see 24/44.1 in Audio Mini.

 

Chris, can you explain more about Pure Music not being bit perfect? How did you confirm this?

 

 

 

 

 

[The Computer Audiophile]

Hi bottlerocket - This is interesting. I'm not sure if I've found a bug or limitation in both Amarra and Pure Music.

 

When using the built-in optical output on a MacBook Pro or Mac Pro neither player can play a 16-bit song first followed by a 24-bit song and output 24-bits on the second song. I believe neither player can change the bit depth using the optical output. When using a Lynx card Amarra will report bit depth changes and it appears to be changing the bits. I'm testing this by sending HDCD encoded material to the Alpha DAC and checking the HDCD indicator. If the Least Significant Bit is not correct the indicator will not illuminate. The Lynx and Pure Music also appear to work bit perfect.

 

The Lynx bit depth is always set at 32-bit and there are no other options.

 

This raises questions about sending equal or greater than the max number of bits each track requires. If I set Audio Midi to 24-bits and use the optical output I can play 16-bit and 24-bit tracks and illuminate the HDCD indicator. Tacking on the extra zeros for 16-bit tracks is better than removing 8-bits from the 24-bit tracks.

 

I have run a ton of tests tonight. I'm still trying to process the results. I'm tempted to say neither app can adjust the bit depth and that's that. But, I think some of my tests indicated otherwise. Or, when I had the dCS Paganini DAC in here it displayed the bit depth of the tracks being played. I know for a fact it showed the correct rate for the tracks, but I think it just looks at the track and not how many bits are being sent by the audio card. If this is the case it would lead me to the lack of bit depth changing capability in both apps.

 

I hope other readers can run some tests.

 

 

[PeterSt]

Hi there Chris,

 

Or, when I had the dCS Paganini DAC in here it displayed the bit depth of the tracks being played. I know for a fact it showed the correct rate for the tracks,

 

I recall very well there was a great discussion about what the dCS showed, and it was the conclusion it could recognize the LSByte being all zeros, and thus shows 16 ("as siginifant" ?).

 

This may help too, at working it out :

 

I myself (XXHE) spit out 24 bits (or 32 actually when the device allows it), no matter the source material is 16. This is because you *may* want to use the volume, and then it starts using those other 8 bits. But if not, the least significant byte remains zero, and all HDCD receivers I have heard of, show "bit perfect", also those who say to receive 24 bits.

 

Now, we may even wonder what it means - or what we want when two subsequent tracks change of bit depth; In the case of XXHE the audio engine is restarted and physically the device is addressed differently. BUT, what does not change (and cannot change I think) is that you, the user, in between state that your device wants to receive 16 bits only, once the before track was 24. Because (this may be hard to get) a 16 bit track always has the potential to "need" 24, and this is again when the volume is needed (or other DSP stuff has to jump in on user command). So, in advance you state what the DAC should receive (16 bits or 24 bits), and the playback is going to behave like that. This is unrelated to the source material ...

 

The above is how I do it, and I don't see it go otherwise while staying praticle.

The most important might be that either case is "bit perfect", as long as there's two bytes only, or when more, the LSBytes are zero (for 16 bit material).

 

Of course, when a first track is 16bits and a second is 24, while the receiver "stays" on 16, things are very wrong.

 

I hope I made clear what I wanted to make clear !

:-)

Peter

 

[bottlerocket]

Thanks for that reply Chris. I resolved to 1 time set my MAC to 24 bit using Audio Midi and then Amarra Mini just switches the sample rate back and forth between 44.1khz and 88 and 96 etc. I think this matches what Peter suggests too. Makes sense to me that its better to zero fill a 24 bit word when going from 24 to 16 bit, than to truncate at 16 bits when going from 16 to 24.

 

BTW, I am using a MAC digital optical out.

 

I asked this question to Amarra if they switched both the word length and the sample rate or just the sample rate and they never answered me.

 

 

 

[cfmsp]

thanks Lars,

 

Do you know how to compare Pure Music to iTunes (a la Amarra)?

 

once Pure Music is opened, I can't figure out how to play music via iTunes only - e.g. to switch back and forth from one to the other.

 

More importantly, when Pure Music is closed, I cannot play music via iTunes without restarting it first. Pure Music does not seem to reconnect the output stream to iTunes upon it's own shutdown.

 

The combination of these two issues makes it impossible to instantly compare one app to the other. The closest I can get to that feature of Amarra is to start a song in iTunes, then open Pure Music, which re-starts whatever song was playing at the beginning.

 

 

There's also a bug whereby Pure Music does not recognize the new song selected in iTunes, and continues playing the original selection, even though iTunes indicates that it's playing the different selection. This continues even after stopping and re-starting the song via iTunes.

 

 

Any experience with either of these issues?

 

it sounds different in my system than the earlier version of PV I tried.

 

thanks

clay

 

 

 

 

[Mr.C]

One thing I noticed with PureMusic is that with the Hiface it is crackle city, whereas PV is maybe an occasional pop, but generally good.

 

[Lars]

Hi Clay,

 

I have not listened to Pure Music and probably won't try it. Pure Vinyl is working very well for me, but I haven't tried the latest version just posted. Anyway, I'm getting a bit worn out testing all of this stuff. It's time to enjoy some tunes.

 

Steve

 

[cfmsp]

 

 

enjoy yourself then, Lars

 

I'm rather tired of trying to 'hear' differences with all the changes I've incorporated as well.

 

clay

 

 

 

[cbarber]

Well I for one have been really happy w/ pure music today...

 

Was comparing pure music side by side w/ amarra (just load some songs into amarra's playlist mode and play them, then play the same ones with pure music). Feeling like amarra definitely has its own certain sound - smooth, but condensed. Sometimes this is better. And the equalizer is important. Pure music takes away the harshness of itunes but allows the spacing of the instruments and what feels like more interesting and vinyl-like timbre. Could it just be that this new sound has caught my ear b/c it's different from what I've been listening to and old songs sound new again? Will have to see how this holds up over the next few weeks...

 

[bsn]

So, it appears that the Pure Music Version does not have the Upsampling feature of Pure Vinyl. I know that many audiophiles don't want that feature, but some people may want it and it does not appear to be there. Just thought I would point that out.

 

[bsn]

My Bad... It appears it will be in the final product.

 

"Note: this option will be disabled until the final (non-Beta) release of Pure Music, which will

include the same high quality 64 bit real-time upsampling feature currently included in the iTunes

Music Server feature of Pure Vinyl."

 

[JR_Audio]

Just an small input from my side. I am back home, so I had a chance to test the new Pure Music 1.0b1, on Mac OS X 10.6.2.

 

It us 100 % Bit True for 16 and for 24 Bit, for AIFF and WAV (I haven't tested ALAC), it does automatic Sample Rate change (tested 44.1, 48, 88.2 and 96K), forward and backward, with and without memory play, and is Bit True in all tested sample rates.

 

Juergen

 

[The Computer Audiophile]

Hi Juergen - Did Pure Music switch he bit depth for you? If you use an audio device that can be set to 16-bit I don't think PM can change it to 24-bit when needed (Amarra is the same way). Maybe there should be something in the manual that say set Audio Midi to 24-bit.

 

[JR_Audio]

Chris: I am not sure, whether I understand your question correctly, but I set the Bit Depth in the Audio MIDI Setup to the resolution of my Audio Interface or DAC.

 

In case I have a 16 Bit Interface or DAC, then I will test only 16 Bit Signals, because 24 Bit signals will be truncated to 16 Bit, so if then 16 Bit is OK, everything is fine.

 

In case this would be a 24 Bit Interface or DAC, then I will test 16 Bit and 24 Bit Signals for Bit True. I will not look back at this Audio MIDI Setup, I just set it to 24 Bit and if 16 Bit Signals came through with 100 % Bit True, this means that Bit 17 to 24 is stable Zero all the time, so it would make no difference if I would have set the Audio MIDI Setup to 16 or to 24 Bit.

 

Is this what you where asking? Because why should a software switch bit depth, because the software didn’t know the resolution of your interface of DAC, so you should set this correctly. This is the same with WASAPI under Windows 7 or Vista. So in WASAPI you have to set the same bit width, as your connected interface or DAC.

 

Juergen

 

 

[The Computer Audiophile]

Hi Jeurgen - This comes into play with an interface like the built-in optical output on a Mac. It supports 16, 20, and 24 bit output. I've always thought it best to output the same bit depth as the music being played. Thus thought Audio Midi should switch between 16 and 24 bit to be exact. My Lynx card only outputs 32 bit so there are no options.

 

Isn't this a combination of what the DAC and the digital I/O support, not just what the DAC supports? Maybe I'm making a bigger deal about this than needs to be made and everyone should just set the bit depth at 24-bit if possible.

 

[barrows]

I do not think that there is any downside to outputting 24 bits from the player to the DAC for 16 bit files, as I believe 16 bit files are just padded with zeros, and the native 16 bits are not changed in any way. In fact, some engineers have mentioned to me that many DAC chips will sound better getting the 24 bits even with 16 bit source material (not sure I understand why though, something to do with noise shaping maybe, perhaps a true expert like Gordon Rankin could answer this question...) I think one will find that when outputting 24 bits from a 16 bit file, the 16 bits stay exactly the same (bit perfect) with 8 zeros added below for properly designed player software.

 

[JR_Audio]

If the software is working correct, than this setting is absolutely signal independent (as what PeterSt was also saying), because then everything above the signal bit width is filled with static Zeros.

 

You have to set it to the minimum value of your signal chain that is behind your playback software, so the interface and the DAC. A BiPhase signal is limited to maximal 24 Bit, so this is the case with SPDIF, AESEBU or TOSLINK, so here 24 Bit would be correct.

 

Even if the Lynx Card supports 32 Bit between the software and the card, at the digital out of the Lynx Card to the Berkley DAC (for example) you can only transfer max 24 Bit, so also here 24 Bit would be correct, if you want to stay Bit True.

 

And also in the case you use a digital volume control between the software and the lynx card, you should and could use 32 Bit, but when you finally want to transfer the digital signal to the Berkley DAC, you need to set the dither of the digital volume to 24 Bit.

 

This has nothing to do with the internal bit depth of software for EQ or Volume Control, because this resolution is prior the point, where the software gives the signal to the outer world.

 

Does this answer the question?

 

Juergen

 

 

[JR_Audio]

Borrows: When connection the digital devices via BiPhase (SPDIF, AESEBU or Toslink), than it does make no difference, whether you are transferring for example only the 16 Bit signal or the 16 Bit signal with padded 8 Zeros.

 

But internally with I2S interfaces (or in some very rate external I2S interfaces), there is a difference, whether you only transfer the 16 Bit signal, or the 16 Bit signal with padded 8 zeros. But to go in this field deeper, it will be too complicated, because then you have to split up again, between very rare NOS DACs and the typical oversampling DACs, or digital oversampling filter. So I will stop here.

 

Juergen

 

 

[earflappin]

Slightly off topic, but I was wondering what people consider to be the optimal interface for the Berkeley Alpha DAC? I use a Lynx AES16, but know from trying an Antelope DA that the Lynx produces sufficient levels of jitter and noise to audibly degrade the Berkeley's performance. I see some use the Legato (async USB/SPDIF) and others use an external clock for their Lynx card. Berkeley is working on their own AES reclocker so they agree the Lynx on its own leaves room for improvement.

 

Would appreciate the benefit of others experimentations. Thanks.

 

David

 

[JR_Audio]

Earflappin: The lynx is a very good sound card with automatic sample rate switching, but you are right and I can confirm, concerning jitter, leaves room for some improvement. When using external clocks, this improves the jitter and sound, but then no longer automatic sample rate change.

 

Juergen