[BigZ12]

I still prefer 48khz over 44.1 due to compatability. I think the 44.1 multiples should just be dicontinued.

Also, is there a point of releasing digital recordings recorded at 24bits to vinyl? Imo only analog recordings should be on vinyl. It's pointless.

So..please tell a "noob" like me:
What do I set my computer to, when using a Audioquest Dragonfly Cobalt, Philips Fidelio X3 and play music with Apple Music? 24bit/44.1, 48, 88.2 or 96khz??
In the manual, Audioquest suggest 44.1khz mostly.

There's also an internal setting in Apple Music: Windows Audio Session or Direct sound. 16/24bit and 44.1 to 192khz. What should I use?

I have a trial with Tidal HiFi, where the Cobalt autodetect the bitrate and sampling freq. itself it seems. (green dragonfly (44.1khz) with "hifi" tracks, and purple for MQA/master)

 

[KeithPhantom]

So..please tell a "noob" like me:
What do I set my computer to, when using a Audioquest Dragonfly Cobalt, Philips Fidelio X3 and play music with Apple Music? 24bit/44.1, 48, 88.2 or 96khz??
In the manual, Audioquest suggest 44.1khz mostly.

There's also an internal setting in Apple Music: Windows Audio Session or Direct sound. 16/24bit and 44.1 to 192khz. What should I use?

I have a trial with Tidal HiFi, where the Cobalt autodetect the bitrate and sampling freq. itself it seems. (green dragonfly (44.1khz) with "hifi" tracks, and purple for MQA/master)

Can you use something like ASIO or WASAPI? I don't trust the internal audio processing Windows has implemented. But if you don't have any option, use the sample rate/bit rate most of your music has so you limit the amount of processing to be done and any possible nonlinearity introduced by bad software.

 

[BigZ12]

Can you use something like ASIO or WASAPI? I don't trust the internal audio processing Windows has implemented. But if you don't have any option, use the sample rate/bit rate most of your music has so you limit the amount of processing to be done and any possible nonlinearity introduced by bad software.

Asio and WASAPI are not possible with Apple Music. As I said, just settings under preferences with the following options:
Windows Audio session and Direct Sound - 16/24bit - 44.1-192khz

When connecting the Dragonfly Cobalt to Windows 10, I get 24bit/44.1 to 96khz as options.

 

[theaudiologist1]

So..please tell a "noob" like me:
What do I set my computer to, when using a Audioquest Dragonfly Cobalt, Philips Fidelio X3 and play music with Apple Music? 24bit/44.1, 48, 88.2 or 96khz??
In the manual, Audioquest suggest 44.1khz mostly.

There's also an internal setting in Apple Music: Windows Audio Session or Direct sound. 16/24bit and 44.1 to 192khz. What should I use?

I have a trial with Tidal HiFi, where the Cobalt autodetect the bitrate and sampling freq. itself it seems. (green dragonfly (44.1khz) with "hifi" tracks, and purple for MQA/master)

always stay bitperfect. If you have 44.1khz files (most CD rips are), go with 44.1. If you want to upsample, upsample in multiples of 2 (44.1 ---> 88.2/176.4/352.8, 48 ---> 96/192/386) to not cause quantization errors.

 

[Brahmsian]

Audirvana ups everything to 32 bits, so that’s what I listen to. Having said that, if you have a good recording to start with and decent headphones or speakers that are driven properly, even MP3 and AAC files sound quite good. Was listening to my old MP3 of Dudamel conducting Mahler’s 9th with the Los Angeles Philharmonic. Really enjoyed it. Far from bad sound despite the compression. And we’re talking a large-scale orchestral piece.

 

[theaudiologist1]

Audirvana ups everything to 32 bits, so that’s what I listen to. Having said that, if you have a good recording to start with and decent headphones or speakers that are driven properly, even MP3 and AAC files sound quite good. Was listening to my old MP3 of Dudamel conducting Mahler’s 9th with the Los Angeles Philharmonic. Really enjoyed it. Far from bad sound despite the compression. And we’re talking a large-scale orchestral piece.

what's the best lossy format? AAC, OGG Vorbis, or Opus?

 

[bigshot]

For high data rate, the best is AAC. And always use VBR. AAC can even go past 320 in VBR if necessary. (I can't imagine it being necessary, but it can.)

 

[theaudiologist1]

For high data rate, the best is AAC. And always use VBR. AAC can even go past 320 in VBR if necessary. (I can't imagine it being necessary, but it can.)

but I heard OGG/Opus can have higher quality at lower bitrate. Also, AAC is proprietary.

 

[bigshot]

AAC is not proprietary. It’s an open standard, and it’s widely supported. If you want super small files and don’t care so much about sound quality, opus is better. If you are looking for something that sounds as good as lossless with a smaller file size AAC is the best. I use AAC 256 VBR and it is as good as lossless at a much more convenient size.

 

[SoundAndMotion]

AAC is patented (link). End users don’t need to pay, but manufacturers or developers of codecs do. That’s why supporters of free and open source software (link) often won’t use it.

 

[bigshot]

I'm pretty sure the Fraunhofer FDK AAC codec is open source and it encodes and decodes AAC files. (I think it's backward engineered or something...) In effect though, it doesn't really matter to users though because AAC is the second most supported codec for compressing high fidelity music, second only to its predecessor MP3. It's not going anywhere. I think he was most concerned with compatibility, We don't want to confuse this guy. We all agree that AAC is definitely the best choice for a lossy codec with both the highest fidelity sound quality and compatibility.

 

[Steve999]

AAC is excellent. I am lousy at hearing lossy artifacts (and hope to stay that way) so anything above 192 kbps in any moderately effective codec is fine with me. I honestly don’t know what’s better in stereo in high bitrates, AAC or Opus. I’ve seen some streaming services turning to Opus if they sense one way or another (the tech is beyond me!) that the downstream gear can handle it. I’d bet the house on either one being transparent to me for the great majority of music under real world listening conditions at 192 kbps or above.

One thing I’ve worried about for a few years is how lossy codecs are handled by simulated surround sound setups, a.ka. upmixing to 2.1, 3.1, 5.1, etc. Industry seems to be quietly addressing this. I‘ve read that newer Bluetooth distributions can be better for that, since they distribute the lossy bits more evenly over the spectrum, so the upmixing algorithm steers the signal to the different channels more or less as hoped. I’ve also read that at higher bitrates Opus specifically tunes its bitrates so that it can be used as a stream for surround sound. Here, from the horse’s mouth: https://wiki.xiph.org/index.php?title=Opus_Recommended_Settings&mobileaction=toggle_view_desktop

A great deal of my listening is lossless streaming now (Amazon HD, Qobuz) (16-bit will do, the rest is marketing voodoo!), so a lot of this falls away as any kind of concern for me. Although right now I am listening to Apple Music (AAC) because, well, I feel like it. It was right there on my Apple TV menu and I’m so sick of the political junk on TV so I said what the hey. And in the meantime I did a head-fi Sound Science drive-by.

 

[Brahmsian]

Maybe someone can react to this:

”when a musician is being mixed in, say, 96kHz and and 24-bit, the output of that mixing session, the master tape or file that’s released for mastering to CD, tape, vinyl, etc. is very likely going to be 96kHz, 24-bit. So anything that’s done in mastering, including dithering down from 24-bit to 16-bit, downsampling from 96kHz to 44.1kHz, that’s all going to affect sound. These days, the changes are minor, possibly not even audible unless you’re really trained in critical listening. But there’s also the presumption that’s being done right. That there’s not record company mandate for crazy compression levels to make the music sound louder on the radio, etc.”
https://www.quora.com/Can-you-hear-...ty-between-24bit-and-32bit-192khz-music-files

 

[KeithPhantom]

So anything that’s done in mastering, including dithering down from 24-bit to 16-bit, downsampling from 96kHz to 44.1kHz, that’s all going to affect sound.

Again, changes -120 dBFS from the fundamental are NOT noticeable by humans, including outliers, even if you play the fundamental at 120 dB SPL (which I do not recommend), the differences are going to be all the way down to 0 dB SPL (assuming that 1 dBFS == 1 dB SPL).

 

[Brahmsian]

Again, changes -120 dBFS from the fundamental are NOT noticeable by humans, including outliers, even if you play the fundamental at 120 dB SPL (which I do not recommend), the differences are going to be all the way down to 0 dB SPL (assuming that 1 dBFS == 1 dB SPL).

He grants that they are ”possibly not even audible,” but then he says that it presumes that the dithering and downsampling to 16/44.1 are being done right--for instance, that there is no ”record company mandate for crazy compression levels to make the music sound louder on radio, etc.”

My own thinking on this is that if given the choice between a dithered and downsampled file and one without dithering and downsampling, I would choose the latter, all else being equal, even if I can’t hear a difference. I have the space for it, my CPU can handle the extra processing, and when they’re on sale on Qobuz the price difference is often negligible. But this is based on the assumption that Qobuz is selling me a copy of the ”master” and isn’t simply adding bits and upsampling redbook files. That is, I assume that when I buy a 24/88.2 file it was recorded that way. Of course, my DAC converts all my music to 32 bit anyway, and I’m still not sure what the point of that is.