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  1. #41
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    Quote Originally Posted by pippin View Post
    yep. And one effect of server-side upsampling is a dramatic increase of bandwidth requirements. Going from 44.1/16 to 192/24 means you increase the bandwidth required by almost a factor of 7!
    OK fair enough.

    For uncompressed PCM:
    16/44.1 = 1,411 Kbps
    24/192 = 9,216 Kbps

    Your earlier claim, though, about being more than HD Video doesn't jive with me. My HD programming on my TV is around 15Mbps (for 1080i 60) and Blu-Rays (1080p 24) are around 30Mbps.

    But that's neither here nor there. If we talk about server-side SoX, then bandwidth is an issue. But John, please chime in here, aren't you talking about SoX in the streamer? Or did I misread that. What's the CPU load effect at that point (compared to doing it server-side)?

  2. #42
    Senior Member pippin's Avatar
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    Don't want to distract this even more, but you are correct, I caught an additional factor of two for PCM but your video streams are MPEG rates, I was talking about H.264 which runs at around half the rate.

    So it's about the same as a 1080p stream, the ones I looked at here were all in the 10 MBit range.
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  3. #43
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    I started this because I was using the upsampling built into Squeezelite(libsoxr), not using SoX in LMS so I can't offer any clues as to how to get that to work properly. The Squeezelite resampling option is not the same as in SoX the program arguments, although the underlying code is the same.

    The one I'm using right now is mI:::28, which is medium quality, intermediate phase and 28 bit depth. I'm not quite sure exactly how this translates to SoX arguments.

    The upsampling is done in Squeezelite on the Wanboard and then sent to my own USB DAC, which uses a standard XMOS UAC2 interface, I2S is run through isolators, on isolated side are low jitter clocks, reclocker and the DAC chip is PCM5142 with very low noise regulators on all of this. The low jitter clock is sent back through an isolator to the XMOS interface.

    The PCM5142 has several builtin filters, some of which are better than others. If you feed it 352.8/384 it turns off the internal filters. So by using resampling in Squeezelite to 352/384 I can bypass the implementation in the chip and just use the software filter.

    Even if upsampling to a lower rate it can still be advantageous, upsampling to 88.2 with the above paramters is a significant improvement. Going to 176.4 is even better but the best can be achieved by going high enough that the built in filters are completely bypassed.

    In all these cases the network traffic stays the same, but the USB data rate goes up since the upsampling is being done in squeezelite.

    I have not had time to go into depth trying all kinds of different parameters, I spent a few nights trying different things and came up with the above. I've been listening to it for some time now and am still enthralled with what it is doing.

    BTW the load on the Wandboard processor is about 8% when using this setting. When using the default 20 bit setting it is about 4% and when not doing any upsampling its about 2%.

    Klaus, to your statement that upsampling should not be necessary, the answer is of course YES. The issue is that as far as I can tell all DAC chips with builtin filters are compromised sonically, the upsampling is an attempt to bypass these filters with an external filter that is more sonically "transparent". So yep it IS a band-aid, but one that is currently necessary for most DACs.

    If the internal filters are not disabled, exactly how the internal filters interact with the external filter are going to be very DAC specific. In the case of the chip I'm using the filters get simpler as the sample rate goes up so even though they are still there if you upsample to an intermediate rate, the total result sounds better than the builtin filter going from 44.1.

    The best sounding parameters for going all the wayto 352.8 and 88.2 will almost certainly be different, but I have not explored it yet.

    John S.
    Last edited by JohnSwenson; 2013-07-12 at 13:43.

  4. #44
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    Quote Originally Posted by JohnSwenson View Post
    BTW the load on the Wandboard processor is about 8% when using this setting. When using the default 20 bit setting it is about 4% and when not doing any upsampling its about 2%.
    Cool. Thanks for the info. Do those percentages include FLAC decoding, or are you feeding it PCM?

  5. #45
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    Quote Originally Posted by edwardian View Post
    Cool. Thanks for the info. Do those percentages include FLAC decoding, or are you feeding it PCM?
    I'm usually sending flac over the network these days since I don't have a new enough server to handle 176 and 192 pcm. The above numbers were for sending flac at 44.1 over ethernet cable.

    John S.

  6. #46
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    John,


    could I ask You witch USB DAC are you using? If is kind of a KIt or a commercial product, could you point me to the source?

    is a kind of the one you'll be using in CSP1?

    thx a lot.

    p.s.

    I could not understand why if you upsample at i.e. 352.8 you still need a software filter, is not assured this way that aliasing is far away from the audible spectrum (reason why the hardware filter is disabled, I suppose)?


    Marco.
    __________________________________________________ ______________________
    Author of C-3PO plugin, Squeezelite-R2, Falcon Web interface - See www.marcoc1712.it

  7. #47
    Can anyone provide me with covert.conf codes to do this within LMS?

  8. #48
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    Try this:

    The file needs to be called :

    custom-convert.conf

    The custom-convert.conf overrides convert.conf.

    It just has to have these 3 lines

    Code:
    flc flc * *
                 # FT:{START=--skip=%t}U:{END=--until=%v}
                 [flac] -dcs $START$ $END$ -- $FILE$  |  [sox] -D -q -t wav - -t flac -e signed  -C 0 -b 24 - rate -v -I -a -b 98 96000

    inside for flac only resampling. You can play with the resampling options. I think above is close to what John suggested earlier.

    You might realize that I output a 24bit (-b 24) stream. That's important to avoid additional dithering, when feeding 16 bit material.

    We need to recode to flac again since bit and sample rate changes can not be streamed as PCM, which is a weakness
    of the LMS server.

    You

    1. need to restart the server to activate above changes and
    2. you have to make sure you've got flac flac decoding activaded in advanced server settings / file formats.
    Under flc/flc you'll see the option showing flac/sox. All options above need to be disabled!


    You'll find more in depth infos here.


    Good luck.
    Last edited by soundcheck; 2013-10-30 at 01:27.

  9. #49
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    Sorry for using this forum to contact you, but i can t send you a message

    Dear Mr Swenson,
    My name is Alex and I am a Master student (Media Management and Entrepreneurship) at the University Fresenius, Cologne. I have seen you in this fantastic blog and was really impressed by your ideas and facts.

    My team of three is looking for hardware experts who could help us with the hardware design of our product. We are currently developing a Smart Home Solution, which offers an all in one solution for media content and transfers it all to your tv or sound system.

    About our Product:
    Our product is a HDMI-stick or -box and also an application. The stick is for a television or a sound stereo system. The application runs also on smartphones. The stick transfers audio (spotify, simfy and other audio online services) and video (netflix, videoload and other video services) content, own content (videos, fotos from dropbox for example) and free tv to your television screen. The smartphone would be the remote.

    Our project team consists of business modell experts (economic experts) and we need people, who can design our hardware and realize our product. For software developing, we are in contact with the ipeng developers. For the user interface, we have some experts from Germany. The business modell is running too.

    However we would need hardware experts to create a wonderful product.

    We hope you will be interested in our idea and would like to know more about it. For more information I could also send you a presentation, just let me know.

    Best wishes
    Alex

  10. #50
    Quote Originally Posted by soundcheck View Post
    Try this:

    The file needs to be called :

    custom-convert.conf

    The custom-convert.conf overrides convert.conf.

    It just has to have these 3 lines

    Code:
    flc flc * *
                 # FT:{START=--skip=%t}U:{END=--until=%v}
                 [flac] -dcs $START$ $END$ -- $FILE$  |  [sox] -D -q -t wav - -t flac -e signed  -C 0 -b 24 - rate -v -I -a -b 98 96000

    inside for flac only resampling. You can play with the resampling options. I think above is close to what John suggested earlier.

    You might realize that I output a 24bit (-b 24) stream. That's important to avoid additional dithering, when feeding 16 bit material.

    We need to recode to flac again since bit and sample rate changes can not be streamed as PCM, which is a weakness
    of the LMS server.

    You

    1. need to restart the server to activate above changes and
    2. you have to make sure you've got flac flac decoding activaded in advanced server settings / file formats.
    Under flc/flc you'll see the option showing flac/sox. All options above need to be disabled!


    You'll find more in depth infos here.


    Good luck.
    Cheers - it doesn't want to work at 352.8 or 384kHz, which is what I need to bypass the digital filters of my PCM5102 chip :-(

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