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  • SamY
    replied
    Originally posted by SAL9K

    This is great news, and I can verify that I'm now getting 24/[44.1, 48] Qobuz streams to Radio devices (not that I can tell the difference, but nice to know that it's streaming at the higher fmt).

    However, maybe I'm misreading this couple of statements, you mention "With this change, however, one can now specify the higher quality preferred format in the plugin without worrying about downsampling overhead", and follow with the next paragraph "anything over 24/48 being downsampled by LMS to 24/48". So, the LMS will indeed incur a transcoding penalty, given that it has to downsample for a particular player?
    Yes. Those statements were made in different contexts. The first paragraph was about players capable of decoding 24/96 maximum. These will no longer be sent content above that threshold, eliminating the downsampling that formerly occured in that situation.

    In the case of players capable of 24/48 maximum (e.g. SB3), the only way to play 24-bit content from Qobuz is to select the Hires (24-bit, <=96k) stream and accept the fact that content with a sampling rate above 48k (i.e. 24/88.2, 24/96) will be downsampled to 24/48. However, 24/44.1 and 24/48 content will be passed through without downsampling.

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  • SAL9K
    replied
    Originally posted by SamY
    Hello, fellow Qobuzians! There is a change in the latest plugin update that deserves some mention here. The logic that determines the quality of the stream to be sent to a particular player is now changed such that, if the quality of the "Preferred format" chosen in the plugin options is greater than what the player is capable of handling natively, the plugin will choose the highest quality stream it IS capable of handling instead. In my case, I have a Squeezelite player running on a RPi, feeding a Denon AVR which is capable of decoding up to 24/192 over its HDMI inputs, which I would like to take advantage of by specifying a "Preferred format" of "24 bits, >96kHz". The downside to doing this in the past, though, was that my other players are only capable of decoding up to 24/96 and, previous to this update, they would still be sent 24/192 content, resulting in LMS having to downsample / transcode the signal to 24/96 and incurring unnecessary (and not insignificant) resource strain on the LMS machine. In many cases, I would experience stuttering on these other players due to this transcoding overhead. With this change, however, one can now specify the higher quality preferred format in the plugin without worrying about downsampling overhead.

    One more "plus". SB Classics and any other players that handle only up to 24/48 decoding will no longer be limited by the plugin to the 16/44.1 flac stream. They will instead be given any hi-res content up to and including 24/96, with anything over 24/48 being downsampled by LMS to 24/48. This should be great news to SB Classic users, who can now experience Qobuz 24-bit content from those players for the first time.

    Thanks to darrell for coming up with this change, which I consider to be a big improvement in the plugin but has little visibility otherwise.
    This is great news, and I can verify that I'm now getting 24/[44.1, 48] Qobuz streams to Radio devices (not that I can tell the difference, but nice to know that it's streaming at the higher fmt).

    However, maybe I'm misreading this couple of statements, you mention "With this change, however, one can now specify the higher quality preferred format in the plugin without worrying about downsampling overhead", and follow with the next paragraph "anything over 24/48 being downsampled by LMS to 24/48". So, the LMS will indeed incur a transcoding penalty, given that it has to downsample for a particular player?

    Leave a comment:


  • darrell
    replied
    Originally posted by SamY
    darrell We should probably take this discussion private but I'm going to put this out here for anyone who wants to do a little listening experiment. I'm not an audio engineer but I do have dual degrees in Physics and Mathematics and I know the science behind the numbers. I just trust my ears more than science's limited understanding of psychoacoustics. Here is the assignment:
    1. Pick a song on Qobuz that is available in 24/44.1 and place it in the player queue. I suggest that you use the following song to begin with, which is one I randomly picked but which I believe is a good candidate for this exercise. It will also, of course, be helpful to use the same song for purposes of discussion. https://open.qobuz.com/track/57619322
    2. Make sure that all LMS player settings that would alter the digital signal are disabled, i.e. no replay gain, output level at 100%.
    3. Set your AVR (or equivalent player) for pure direct two-channel processing (the goal here is "bit perfect" playback from Qobuz to the DAC).
    4. Set the Qobuz plugin preferred quality option to 16/44.1.
    5. Set the player queue (one song) for repeat mode.
    6. Hit "Play" and verify from the technical info that it is the 16/44.1 stream.
    7. Listen carefully.
    8. While you are listening, go into the plugin settings and change the preferred quality to 24/96.
    9. Continue listening and verify that, when the song repeats, it is now the 24/44.1 stream
    10. Again listen carefully.
    11. Repeat as often as desired, switching between 16 and 24-bit versions and listening for differences of any kind.
    I won't try to describe what I hear right now except to say that for me the difference is not subtle. I look forward to hearing other opinions, although it should probably be moved to another thread.

    EDIT: In case the link doesn't take you to the exact track, it's the first one: "Dronning Fjellrose".
    Interesting. That's a very good album, by the way. I'm about to create a thread in the Audiophile sub-forum to continue this discussion.

    Leave a comment:


  • SamY
    replied
    darrell We should probably take this discussion private but I'm going to put this out here for anyone who wants to do a little listening experiment. I'm not an audio engineer but I do have dual degrees in Physics and Mathematics and I know the science behind the numbers. I just trust my ears more than science's limited understanding of psychoacoustics. Here is the assignment:
    1. Pick a song on Qobuz that is available in 24/44.1 and place it in the player queue. I suggest that you use the following song to begin with, which is one I randomly picked but which I believe is a good candidate for this exercise. It will also, of course, be helpful to use the same song for purposes of discussion. https://open.qobuz.com/track/57619322
    2. Make sure that all LMS player settings that would alter the digital signal are disabled, i.e. no replay gain, output level at 100%.
    3. Set your AVR (or equivalent player) for pure direct two-channel processing (the goal here is "bit perfect" playback from Qobuz to the DAC).
    4. Set the Qobuz plugin preferred quality option to 16/44.1.
    5. Set the player queue (one song) for repeat mode.
    6. Hit "Play" and verify from the technical info that it is the 16/44.1 stream.
    7. Listen carefully.
    8. While you are listening, go into the plugin settings and change the preferred quality to 24/96.
    9. Continue listening and verify that, when the song repeats, it is now the 24/44.1 stream
    10. Again listen carefully.
    11. Repeat as often as desired, switching between 16 and 24-bit versions and listening for differences of any kind.
    I won't try to describe what I hear right now except to say that for me the difference is not subtle. I look forward to hearing other opinions, although it should probably be moved to another thread.

    EDIT: In case the link doesn't take you to the exact track, it's the first one: "Dronning Fjellrose".
    Last edited by SamY; 2023-09-22, 05:42.

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  • darrell
    replied
    I wonder if any differences between 16 and 24 bit could be down to different mastering? Or are you taking a 24/48 recording and converting it to 16/48 yourself? My gut feeling is that with an identical master you'd be hard pressed to notice a difference between a noise floor of -95dB vs -120dB in any kind of normal listening room, unless the recording was made at an extremely low reference level. I have an old test CD with a recording of very low level Japanese water chimes, which is supposed to demonstrate the difference between "normal" 16 bit mastering and Sony's big thing back in the early '90s, Super Bit Mapping, which they claimed gave CD the benefits of 20 bit.

    Leave a comment:


  • SamY
    replied
    Originally posted by darrell
    At our age, we'd be fine with 32khz sampling rate!
    Good point. Then again, maybe my inability to hear frequencies above 12kHz results in a heightened sensitivity to the content below that threshold, similar to what they say about a blind person having superior overall hearing ability compared to the sighted. ​​​​​​

    Leave a comment:


  • darrell
    replied
    Originally posted by SamY

    I don't think there was actually any resampling occurring in that case. The plugin was simply choosing the 16/44.1 (CD quality) Qobuz stream instead of the 24/48 one.
    Doh! I blame my age. But it's still avoiding the resampling of 44.1 to 48 in the receiver's DSP.

    Originally posted by SamY

    In my opinion, the SB3 gaining access to the 24-bit dynamic range of the Hi-res stream is the prime benefit of that part of the change. After much critical listening (no, not double-blind ), I'm more convinced that I can hear the difference between 16/44.1 and 24/44.1 than between 24/44.1 and 24/192, all scientific arguments aside.
    At our age, we'd be fine with 32khz sampling rate!

    Leave a comment:


  • SamY
    replied
    Originally posted by darrell
    when using my SB3 (aka SB Classic) 48K content was being resampled to 44.1K by LMS,
    I don't think there was actually any resampling occurring in that case. The plugin was simply choosing the 16/44.1 (CD quality) Qobuz stream instead of the 24/48 one. In my opinion, the SB3 gaining access to the 24-bit dynamic range of the Hi-res stream is the prime benefit of that part of the change. After much critical listening (no, not double-blind ), I'm more convinced that I can hear the difference between 16/44.1 and 24/44.1 than between 24/44.1 and 24/192, all scientific arguments aside.

    Sam also mentioned in passing that we might be able to introduce player-specific overrides to the default Quobuz streaming quality preference: if we could do that, you could, for example, limit the Qobuz stream to MP3 for a kitchen radio.
    ​​​​​​This change pretty much removes the need for a per-player option, in my opinion, as the player's firmware / Squeezelite preference options can be used to specify the desired / supported streaming quality for all sources, including the Qobuz plugin.

    Leave a comment:


  • darrell
    replied
    Originally posted by Jeff07971
    Thanks for the update
    This seems a good idea, not that I ever had a problem with downsampling.
    What version of the plugin has this update ?​
    I don't get too worked up about resampling/downsampling either, but I authored this change because noticed that when using my SB3 (aka SB Classic) 48K content was being resampled to 44.1K by LMS, and then (I have it on good authority) resampled back to 48K by the Audyssey DSP room correction in my Denon. I think that unless you've got a really expensive receiver anything but pure/direct/straight mode is going to resample/downsample to 48K. I just wanted to get rid of the unnecessary resampling load on SBS and the receiver.

    Then SamY suggested that we add the selection of the 96K max stream from Qobuz unless the player is capable of 192K, again to cut down on processing (and in that case, bandwidth).

    Another thought - If you're using Audyssey DSP (or similar) and the UPnPBridge plugin to have the stream sent directly to the receiver, you could have LMS downsample 96K and 192K content to 48K using UPnP plugin settings, which would reduce the load on the receiver as well as your local network bandwidth requirement.

    Sam also mentioned in passing that we might be able to introduce player-specific overrides to the default Quobuz streaming quality preference: if we could do that, you could, for example, limit the Qobuz stream to MP3 for a kitchen radio. Would there be any demand for such a thing?

    Leave a comment:


  • Jeff07971
    replied
    Hi Michael

    Yup I'm on 2.16.4 so all good

    Thanks
    Jeff

    Leave a comment:


  • mherger
    replied
    Originally posted by Jeff07971
    Thanks for the update
    This seems a good idea, not that I ever had a problem with downsampling.
    What version of the plugin has this update ?​
    The latest . I did f... up a first release of this change. So please double check what you have. Should be 2.16.4 at least.

    Leave a comment:


  • Jeff07971
    replied
    Thanks for the update
    This seems a good idea, not that I ever had a problem with downsampling.
    What version of the plugin has this update ?​

    Leave a comment:


  • d6jg
    replied
    Originally posted by SamY
    Hello, fellow Qobuzians! There is a change in the latest plugin update that deserves some mention here. The logic that determines the quality of the stream to be sent to a particular player is now changed such that, if the quality of the "Preferred format" chosen in the plugin options is greater than what the player is capable of handling natively, the plugin will choose the highest quality stream it IS capable of handling instead. In my case, I have a Squeezelite player running on a RPi, feeding a Denon AVR which is capable of decoding up to 24/192 over its HDMI inputs, which I would like to take advantage of by specifying a "Preferred format" of "24 bits, >96kHz". The downside to doing this in the past, though, was that my other players are only capable of decoding up to 24/96 and, previous to this update, they would still be sent 24/192 content, resulting in LMS having to downsample / transcode the signal to 24/96 and incurring unnecessary (and not insignificant) resource strain on the LMS machine. In many cases, I would experience stuttering on these other players due to this transcoding overhead. With this change, however, one can now specify the higher quality preferred format in the plugin without worrying about downsampling overhead.

    One more "plus". SB Classics and any other players that handle only up to 24/48 decoding will no longer be limited by the plugin to the 16/44.1 flac stream. They will instead be given any hi-res content up to and including 24/96, with anything over 24/48 being downsampled by LMS to 24/48. This should be great news to SB Classic users, who can now experience Qobuz 24-bit content from those players for the first time.

    Thanks to darrell for coming up with this change, which I consider to be a big improvement in the plugin but has little visibility otherwise.
    Good move. I can probably bring my DAC32/DIGI32 back into play. They are both capped at 24/96 and were taken out of service due to the stuttering referred to when trying to play 24/192

    Leave a comment:


  • SamY
    replied
    Hello, fellow Qobuzians! There is a change in the latest plugin update that deserves some mention here. The logic that determines the quality of the stream to be sent to a particular player is now changed such that, if the quality of the "Preferred format" chosen in the plugin options is greater than what the player is capable of handling natively, the plugin will choose the highest quality stream it IS capable of handling instead. In my case, I have a Squeezelite player running on a RPi, feeding a Denon AVR which is capable of decoding up to 24/192 over its HDMI inputs, which I would like to take advantage of by specifying a "Preferred format" of "24 bits, >96kHz". The downside to doing this in the past, though, was that my other players are only capable of decoding up to 24/96 and, previous to this update, they would still be sent 24/192 content, resulting in LMS having to downsample / transcode the signal to 24/96 and incurring unnecessary (and not insignificant) resource strain on the LMS machine. In many cases, I would experience stuttering on these other players due to this transcoding overhead. With this change, however, one can now specify the higher quality preferred format in the plugin without worrying about downsampling overhead.

    One more "plus". SB Classics and any other players that handle only up to 24/48 decoding will no longer be limited by the plugin to the 16/44.1 flac stream. They will instead be given any hi-res content up to and including 24/96, with anything over 24/48 being downsampled by LMS to 24/48. This should be great news to SB Classic users, who can now experience Qobuz 24-bit content from those players for the first time.

    Thanks to darrell for coming up with this change, which I consider to be a big improvement in the plugin but has little visibility otherwise.

    Leave a comment:


  • darrell
    replied
    The thing is that the LMS library database doesn't have anywhere to store the tag "Work" and until that changes (one day!) the enhanced views are only available, as Sam says, when browsing albums in the Qobuz plug-in. You can add albums from the various plug-in screens to your Qobuz favourites, and then play from there in order to make use of the classical enhancements. Your Qobuz favourites list is sort-able via a setting in the plug-in options. Thinking about it, a search within favourites would be nice, I'll look into the possibility of adding that feature.

    Leave a comment:

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