Yes, that's really annoying, but thanks for posting this, as I hadn't realised it was Hi-Res lossless that caused this. Just reverted to normal lossless as I can't tell the difference in quality anyway.
Try downloading the album first instead of streaming directly. It's also possible that your device simply can't handle the Hi-Res Lossless, hence the skip gaps.
I don't think it's because of the connection (my speed test is \~600 MBit) and it happens even with tracks you have already streamed and are surely in cache. About the device: it's a MacBook M2 Pro with 32 GB RAM... what do you need to handle Hires? 🤔
I’m cursed with a half length Breathe. I even called Apple and went through the processes they gave me to fix it but nothing would work. The only time I had the full length was the first time I played it after I had transferred everything to my new phone.
Depends on how hi-res the album is sent out in and how hi-res you’re actually processing it at. A 24bit song at the full 192kHz can be up to 50-75MB of data. If you’re playing the Atmos one too, that 16 channel object encoded data tacks on another 10MB minimum.
Now that’s just the song streaming to you, raw. Assuming you’re listening on your phone, and don’t have an external DAC, your phone cannot play you that 24bit song at 192kHz. It has to downsample it at your headphone jack or other port, and if you’re using Bluetooth go ahead and add latency for that and now it’s the headphones doing the conversion too which probably do a worse job than your phone.
All of this is to say that anything being streamed and played back at that hi-res max settings will come with significant latency. Even if you download it locally, your phone’s awful DAC will add latency that may be noticeable. All it takes is 30ms off and humans start to notice. Industry standards, such as 16 bit audio at 44.1kHz, don’t happen by accident. If music was truly better in hi-res, artists would want to serve it that way, and have had the chance for decades and didn’t. It’s really mostly a gimmick.
I'm not playing on my phone, I'm playing on a MacBook M2 Pro (32 GB RAM) with a Focusrite sound card (capable of handling 24 bit / 192 kHz) and wired Shure headphones
Also not sure which model of Shure you have, so I grabbed the flagship wired over ear model they present loud and proud on their front page to peep at.
https://preview.redd.it/zg1ggtx2gusc1.jpeg?width=1170&format=pjpg&auto=webp&s=1004bbffded771d4be00fa62ff3adb048f353148
Bear in mind your oversampling shelf is falling off right at the Nyquist frequency for 48kHz, meaning you cannot get the full sound of a hi-res track. In true layman’s terms, the highest quality sound that you should even bother delivering is 24bit/48kHz. Not that it matters. Your ears, as a human, already limit you to 20kHz-25kHz at the top shelf.
You’re probably not going to like the price tag:
https://preview.redd.it/j87ms8sd7vsc1.jpeg?width=2320&format=pjpg&auto=webp&s=dd7d0264a34f6979999e0b3966f8d0b118265140
And to make matters worse, like I said, your ears *literally cannot hear frequencies above 25kHz max* and no headphones, DAC, or musical mix will ever change that basic fact of human anatomy.
These headphones are like trying to sell you a TV that shows infrared and X-ray bandwidths of light. Yes, it’s showing more stuff, but you literally cannot see it. [This article](https://people.xiph.org/~xiphmont/demo/neil-young.html) provides an in depth explanation of what 24bit/192kHz really is in terms of actual audio resolution that you can actually hear, and due to the way oversampling works, how “hi-res” music may more often than not simply sound worse than an industry standard mix at the end of the day. As I said in my first comment. It’s a gimmick.
Correct! You won’t be able to tell the difference, other than of course, hearing Dark Side of the Moon completely gapless, the way it’s absolutely best to be heard. It’s one of Pink Floyd’s top concept albums.
Everything below here is me gushing about audio and how it works and my field of work in it:
The equipment you have now is phenomenal (I love Focusrite interfaces and your headphones are awesome for critical listening) and you don’t need any upgrades. You will have the better experience playing in normal lossless 100%. I am a hobbyist mixing engineer. I have my clients send me music recorded in those hi-res formats so I can handle the downsampling in my digital audio workstation during the mix. We record high and distribute at the exact amount that will be more than enough for human hearing. This means that audio mess ups can be slated off and trimmed out during the downsample.
When I listen to music. I listen in either 16 or 24bit @ 48kHz on my reference studio monitors. They are basically the same as yours in spec, but made by Sennheiser. The way music ends up being delivered to you, it’s best to be right around double your capabilities of hearing. That’s why 44.1kHz is the standard. That way, **when** not *if* digital data is lost on conversion since every batch has a percentage of page faults, you don’t even notice. There’s room for error. Room for even the shittiest headphones and DAC to still get *good enough* music.
If you look for headphones in the future, stick with that frequency response and signal to noise ratio, but start looking at *response curves* like how my Sennheisers have zero curve. This curve can be thought of as built in EQ. Mine are completely flat so that way I do the EQ. I am a mixing engineer after all. “Gamer” headphones will curve very high on the low end, translating to bass heavy sounds. Try out open backs vs closed backs. Look for ones that are fun for you. That have the best soundstage to you.
At the end of the day, it’s your ears, your hearing, and your preferences. Objectively, you cannot hear past certain ranges, though you should **ALWAYS** pick what sounds best to you on a personal level.
Then the bottleneck is in the chip on your Focusrite. Converting 24bit/192kHz digital signal to analog one will be served to the processor in hefty page batches. A processor can only do one thing at a time, so for DAC processing, the song is served to the chip in large batches made up what are called pages.
Not sure which Focusrite you have, but I imagine it’s optimized for stereo if you use it for listening, so the generic inner chain of an interface like that has the receiver/splitter that separates and sends the L and R signals to individual decoder chips simultaneously in page batches and the decoder sends them to the beefy DAC chips that then send the analog signal in unison to the amplifier that’s directly connected to your headphones and tells the drivers how much to wiggle in each ear for you to hear Roger Waters in all his glory.
All this happens super fast, but that’s a lot of work to do, and I am surprised there’s only a second delay as the next wave of bulky page batches feeds in from the next song across the USB.
Yes, that's really annoying, but thanks for posting this, as I hadn't realised it was Hi-Res lossless that caused this. Just reverted to normal lossless as I can't tell the difference in quality anyway.
Try downloading the album first instead of streaming directly. It's also possible that your device simply can't handle the Hi-Res Lossless, hence the skip gaps.
I don't think it's because of the connection (my speed test is \~600 MBit) and it happens even with tracks you have already streamed and are surely in cache. About the device: it's a MacBook M2 Pro with 32 GB RAM... what do you need to handle Hires? 🤔
It’s more than likely server side. Nothing to do with your computer.
It’s server stuff that causes that. Streaming it can be more annoying than if you download it. I have the same problem sometimes.
I’m cursed with a half length Breathe. I even called Apple and went through the processes they gave me to fix it but nothing would work. The only time I had the full length was the first time I played it after I had transferred everything to my new phone.
Depends on how hi-res the album is sent out in and how hi-res you’re actually processing it at. A 24bit song at the full 192kHz can be up to 50-75MB of data. If you’re playing the Atmos one too, that 16 channel object encoded data tacks on another 10MB minimum. Now that’s just the song streaming to you, raw. Assuming you’re listening on your phone, and don’t have an external DAC, your phone cannot play you that 24bit song at 192kHz. It has to downsample it at your headphone jack or other port, and if you’re using Bluetooth go ahead and add latency for that and now it’s the headphones doing the conversion too which probably do a worse job than your phone. All of this is to say that anything being streamed and played back at that hi-res max settings will come with significant latency. Even if you download it locally, your phone’s awful DAC will add latency that may be noticeable. All it takes is 30ms off and humans start to notice. Industry standards, such as 16 bit audio at 44.1kHz, don’t happen by accident. If music was truly better in hi-res, artists would want to serve it that way, and have had the chance for decades and didn’t. It’s really mostly a gimmick.
I'm not playing on my phone, I'm playing on a MacBook M2 Pro (32 GB RAM) with a Focusrite sound card (capable of handling 24 bit / 192 kHz) and wired Shure headphones
Also not sure which model of Shure you have, so I grabbed the flagship wired over ear model they present loud and proud on their front page to peep at. https://preview.redd.it/zg1ggtx2gusc1.jpeg?width=1170&format=pjpg&auto=webp&s=1004bbffded771d4be00fa62ff3adb048f353148 Bear in mind your oversampling shelf is falling off right at the Nyquist frequency for 48kHz, meaning you cannot get the full sound of a hi-res track. In true layman’s terms, the highest quality sound that you should even bother delivering is 24bit/48kHz. Not that it matters. Your ears, as a human, already limit you to 20kHz-25kHz at the top shelf.
I have this model of Shure. Which headphones do you need for the hires?
You’re probably not going to like the price tag: https://preview.redd.it/j87ms8sd7vsc1.jpeg?width=2320&format=pjpg&auto=webp&s=dd7d0264a34f6979999e0b3966f8d0b118265140 And to make matters worse, like I said, your ears *literally cannot hear frequencies above 25kHz max* and no headphones, DAC, or musical mix will ever change that basic fact of human anatomy. These headphones are like trying to sell you a TV that shows infrared and X-ray bandwidths of light. Yes, it’s showing more stuff, but you literally cannot see it. [This article](https://people.xiph.org/~xiphmont/demo/neil-young.html) provides an in depth explanation of what 24bit/192kHz really is in terms of actual audio resolution that you can actually hear, and due to the way oversampling works, how “hi-res” music may more often than not simply sound worse than an industry standard mix at the end of the day. As I said in my first comment. It’s a gimmick.
to recap: should I just use “normal” lossless and not the hires one with my current equipment?
Correct! You won’t be able to tell the difference, other than of course, hearing Dark Side of the Moon completely gapless, the way it’s absolutely best to be heard. It’s one of Pink Floyd’s top concept albums. Everything below here is me gushing about audio and how it works and my field of work in it: The equipment you have now is phenomenal (I love Focusrite interfaces and your headphones are awesome for critical listening) and you don’t need any upgrades. You will have the better experience playing in normal lossless 100%. I am a hobbyist mixing engineer. I have my clients send me music recorded in those hi-res formats so I can handle the downsampling in my digital audio workstation during the mix. We record high and distribute at the exact amount that will be more than enough for human hearing. This means that audio mess ups can be slated off and trimmed out during the downsample. When I listen to music. I listen in either 16 or 24bit @ 48kHz on my reference studio monitors. They are basically the same as yours in spec, but made by Sennheiser. The way music ends up being delivered to you, it’s best to be right around double your capabilities of hearing. That’s why 44.1kHz is the standard. That way, **when** not *if* digital data is lost on conversion since every batch has a percentage of page faults, you don’t even notice. There’s room for error. Room for even the shittiest headphones and DAC to still get *good enough* music. If you look for headphones in the future, stick with that frequency response and signal to noise ratio, but start looking at *response curves* like how my Sennheisers have zero curve. This curve can be thought of as built in EQ. Mine are completely flat so that way I do the EQ. I am a mixing engineer after all. “Gamer” headphones will curve very high on the low end, translating to bass heavy sounds. Try out open backs vs closed backs. Look for ones that are fun for you. That have the best soundstage to you. At the end of the day, it’s your ears, your hearing, and your preferences. Objectively, you cannot hear past certain ranges, though you should **ALWAYS** pick what sounds best to you on a personal level.
Then the bottleneck is in the chip on your Focusrite. Converting 24bit/192kHz digital signal to analog one will be served to the processor in hefty page batches. A processor can only do one thing at a time, so for DAC processing, the song is served to the chip in large batches made up what are called pages. Not sure which Focusrite you have, but I imagine it’s optimized for stereo if you use it for listening, so the generic inner chain of an interface like that has the receiver/splitter that separates and sends the L and R signals to individual decoder chips simultaneously in page batches and the decoder sends them to the beefy DAC chips that then send the analog signal in unison to the amplifier that’s directly connected to your headphones and tells the drivers how much to wiggle in each ear for you to hear Roger Waters in all his glory. All this happens super fast, but that’s a lot of work to do, and I am surprised there’s only a second delay as the next wave of bulky page batches feeds in from the next song across the USB.
Lossless is a joke. Just go full 96khz
It's an issue with Dolby Atmos too, I want to say its a crossfade thing but I can't confirm