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Help with improving audio quality for iMac

Tangband

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But what has Sample Rate Conversion to do with audio quality between platforms? Why convert the sample anyway to listen to it? You should listen at the original setting anyway.
When doing recordings, one always uses SRC at downmixing to 44,1 kHz ( from 96 kHz recording ) .
Audacity is very good software for SRC both for Windows and Mac.
Windows also have a really nasty limiter in the software at playback : CAudioLimiter
Read about all the correct settings you must have in a Windows machine to not destroy the music, here:
As I said - This has been discussed in tens of threads at audiosciencereview.:)

A Mac has only ”core” - its bitcorrect and need no drivers
 

Katji

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Its a software problem . Tons of info on the net. Microsofts SRC and limiter is really bad . But one can use Foobar instead , much better .
*That is not proof.*
Tons of crap too, not least with audiophile stuff.

"Destroy the music" - that is hype / gross exaggeration.

^^That thread: It "ended the debate" as in "no issue."

And - in general at least, and [ime] - Windows audio "tweaks" in audiophile forums/blogs cause problems, because they're from Windows 7 days. A mistake I made - reading that shit and changing related Windows settings and then i had to fix it - after I'd had no problems before.

There is also a thread where Amir gave some information about related Windows development, him/his team.

Foobar -- no thanks.
 

blueone

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If you really want to improve the sound quality of your set-up, I recommend acquiring REW and an appropriate mike, and some EQ software for the Mac, like Boom2, or one of the other Mac apps. I think worrying about sample rates and conversions is like chasing your tail. I use a Mac desktop system with Audioengine 5s and an SVS SB12 sub with Boom2 for EQ, and after an hour or so of fiddling around a couple of years ago, I've been completely satisfied, sometimes amazed and impressed, and haven't touched the adjustments since. EQ and measurement are the keys.
 

sarumbear

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When doing recordings, one always uses SRC at downmixing to 44,1 kHz ( from 96 kHz recording ) .
“One” should never do that. Everything should be recorded at the maximum available resolution and never changed. There’s no reason to do that. The only time it may be changed is at mastering as per the medium the track is distributed as.
 
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Tangband

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“One” should never do that. Everything should be recorded at the maximum available resolution and never changed. There’s no reason to do that. The only time it may be changed is at mastering as per the medium the track is distributed as.
In a purist perspective - you are absolutely right. But if you record something different than only classical instruments with two channels without effects , you have to mix it in a software program like Logic , using reverb plug ins ( often 48 KHz ) or compression ( 48 KHz ) and so on….
What you hear in a ” normal ” production is something thats in the beginning maybe was recorded at 96 KHz but has been thru SRC many times before if finally arrives at TIDAL or Spotify at 44,1 KHz .

One must remember that putting some reverb and compression on a drumkit ( always done ) in a mix demands resampling 2 times for the whole track ( or often 8 tracks for only the drumkit ) .

There is a lot of confusion about this .

Me , I always try to record acoustical instruments at native sampling rate at 96 KHz , using only two microphones, and scipping the whole mixing process.

But - I still have to do one digital manipulation before the recording is finished and thats ” normalisation” of the recording .

In an acoustical recording , you always have -10 dB as a margin for digital clipping . The recorded tracks will be about -10 as loud as a normal CD . At normalisation, you lift up the level to -1 dB . This is done in digital domain with 32 or 64 bit resolution.

My experience with Logic Pro and Audacity is that this simple digital manipulation ( in Logic Pro X its done at 64 bit internaly, 32 bit in audacity ) can be heard as a little less natural sound , unfortunately.

The -10 dB 96 KHz 24 bit recording will sound a bit better than the finnished -1 dB recording . The -10 dB recording will sound more natural and with less digital ” glare” .

This is a sad thing - because when I started recording , I always thought this was entirely inaudible wich is not true.
 
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sarumbear

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In a purist perspective - you are absolutely right. But if you record something different than only classical instruments with two channels without effects , you have to mix it in a software program like Logic , using reverb plug ins ( often 48 KHz ) or compression ( 48 KHz ) and so on….
What you hear in a ” normal ” production is something thats in the beginning maybe was recorded at 96 KHz but has been thru SRC many times before if finally arrives at TIDAL or Spotify at 44,1 KHz .

One must remember that putting some reverb and compression on a drumkit ( always done ) in a mix demands resampling 2 times for the whole track ( or often 8 tracks for only the drumkit ) .

There is a lot of confusion about this .
I don't think there is. Keeping the sample rate unified is a standard practice.

Me , I always try to record acoustical instruments at native sampling rate at 96 KHz , using only two microphones, and scipping the whole mixing process.

But - I still have to do one digital manipulation before the recording is finished and thats ” normalisation” of the recording .
Thank you for agreeing with what I said. Within a relatively long reply all I see is that when people use plugins that are not set to the working sample rate you may have problems, if the host is Windows and if the conversion is done using Windows code. I have not met any professional DAW that does that.

I know that ProTools, Cubase & Abbleton Live handle every aspect of the signal within their code.

My experience with Logic Pro and Audacity is that this simple digital manipulation ( in Logic Pro X its done at 64 bit internaly, 32 bit in audacity )

You give example of Logic but that is a Mac only DAW. Your arguments is about Windows issues...
 
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fabius

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I believe I have the high resolution in Apple music enabled. Does that only affect audio coming from the Apple music app? I've wondered about that. I thought the equalizer within the Apple music app only affected the music when playing through Apple music.

Just to be clear, there’s the Apple Music streaming service - which is (or can be? I haven’t used it) in high resolution. So if you’re using that, then make sure it’s using high resolution (sounds like you have).

And then there’s the Apple Music app, through which you can listen to the Apple Music streaming service and/or your own MP3, AACs, FLACs, etc. The latter files are what they are and there’s nothing you can do to improve their resolution other than replacing them (they might be fine anyway; mine are a vast and terrible range!).

It’s baffling that Apple gives the two things the same name, and people are rarely explicit about which they’re referring to.

I haven’t used Boom2 but do use SoundSource to add EQ effects when listening through headphones - it also has an equaliser that works system-wide, unlike the one in the Apple Music app which, as you say, only affects what you play through that.
 

voodooless

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Why convert the sample anyway to listen to it?
Because on an operating system, you don't know what the sample rate is that you need at any moment. In fact, many sounds could be playing with various sample rates and bit depths at the same time. To facilitate this, you set one sample rate in the OS, and any sound source get resampled to that rate so that all sources can be properly mixed together. Windows still does this poorly. MacOS much better.
You should listen at the original setting anyway.
Yes, ideally you should. That's why any self-respecting audio player has exclusive access to the audio device to put it into the most ideal mode to play the audio.

To be fair, MacOS does not have a real exclusive mode, but comes close and can output bitperfect audio just fine. For Apple Music you'll need some help though: https://github.com/vincentneo/LosslessSwitcher
 
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sarumbear

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Because on an operating system, you don't know what the sample rate is that you need at any moment. In fact, many sounds could be playing with various sample rates and bit depths at the same time. To facilitate this, you set one sample rate in the OS, and any sound source get resampled to that rate so that all sources can be properly mixed together. Windows still does this poorly. MacOS much better.

Yes, ideally you should. That's why any self-respecting audio player has exclusive access to the audio device to put it into the most ideal mode to play the audio.

To be fair, MacOS does not have a real exclusive mode, but comes close and can output bitperfect audio just fine. For Apple Music you'll need some help though: https://github.com/vincentneo/LosslessSwitcher
But what recording software doesn’t control the sound module directly? Besides, ASIO is the standard way for a DAW to interface with audio hardware and it has its own control panel to make sure what you described doesn’t happen.

All in all what I’m hearing is people who don’t know what they are doing can have more chance to mess things up using Windows than Mac. Then again those people will mess so many other things that will affect the sound than a rate conversion error.
 

voodooless

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But what recording software doesn’t control the sound module directly?
But we're talking playback here. Recording is a totally different ballgame.
Besides, ASIO is the standard way for a DAW to interface with audio hardware and it has its own control panel to make sure what you described doesn’t happen.
You'll probably don't want to use a DAW for playback of your music collection, do you? But yes, also for any decent music player, ASIO or other direct hardware access must be configured for the best sound quality.
All in all what I’m hearing is people who don’t know what they are doing can have more chance to mess things up using Windows than Mac. Then again those people will mess so many other things that will affect the sound than a rate conversion error.
Probably :)
 

sarumbear

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When doing recordings, one always uses SRC at downmixing to 44,1 kHz ( from 96 kHz recording ) .
But if you record something different than only classical instruments with two channels without effects , you have to mix it in a software program like Logic , using reverb plug ins ( often 48 KHz ) or compression ( 48 KHz ) and so on….
As far as I can see the conversation is about recording.
Because on an operating system, you don't know what the sample rate is that you need at any moment. In fact, many sounds could be playing with various sample rates and bit depths at the same time. To facilitate this, you set one sample rate in the OS, and any sound source get resampled to that rate so that all sources can be properly mixed together. Windows still does this poorly. MacOS much better.
What sounds will be playing at the same time when you are listening to music? You play a track and send that to your DAC, which will select the correct sampling frequency. What am I missing?
 

voodooless

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What sounds will be playing at the same time when you are listening to music? You play a track and send that to your DAC, which will select the correct sampling frequency. What am I missing?
The DAC is set to the correct frequency by the driver. This is either controlled by the OS mixer, which will resample all incoming audio to that frequency, or by some player application that has exclusive access to the driver and fully controls what the sound device is doing. In the last case, no other sounds can be played, otherwise, things like email sounds, YouTube or incoming phone calls all share the same output device via the OS mixer.
 

sarumbear

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The DAC is set to the correct frequency by the driver. This is either controlled by the OS mixer, which will resample all incoming audio to that frequency, or by some player application that has exclusive access to the driver and fully controls what the sound device is doing. In the last case, no other sounds can be played, otherwise, things like email sounds, YouTube or incoming phone calls all share the same output device via the OS mixer.
I understand but why would any audiophile will use an internal sound card as a DAC to feed their Hi-Fi? Why not use an external USB DAC? Compare to the rest of the equipment the cost is negligible.
 

voodooless

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As far as I can see the conversation is about recording.
Well, it went all over the place though ;) You asked:
Why convert the sample anyway to listen to it?
So I delivered a reason why that happens on a multifunction device like a computer :) The Windows mixer sample rate converter was brought up after all.
 

voodooless

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I understand but why would any audiophile will use an internal sound card as a DAC to feed their Hi-Fi? Why not use an external USB DAC? Compare to the rest of the equipment the cost is negligible.
Internal or external is of no consequence. It works exactly the same.
 

sarumbear

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Internal or external is of no consequence. It works exactly the same.
Not really. You don’t have to use the Windows mixer for listening music. You can (and must) bypass it.
 

voodooless

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Not really. You don’t have to use the Windows mixer for listening music.
That has nothing to do with internal or external. I can feed the USB DAC with the windows mixer just fine, as I can use it with an ASIO-enabled audio player in exclusive mode. An internal soundcard can do exactly the same.
 

sarumbear

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That has nothing to do with internal or external. I can feed the USB DAC with the windows mixer just fine, as I can use it with an ASIO-enabled audio player in exclusive mode. An internal soundcard can do exactly the same.
If you feed the external DAC via the windows mixer wouldn’t you suffer the same sampling conversion issues? When I bypass the Windows Mixer for Chrome browser and play Apple Music tracks, using their Web Player, the external DAC matches the differing sample rate and depth of the tracks. However, system sounds will be heard from the PC sound output. No mixing is involved as it should be. No one wants system beeps from their Hi-Fi would they?
 
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