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Improving a HTPC audio chain.

Aerith Gainsborough

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So far you've tried to endlessly debate every topic we've touched on...
We went through the whole clipping thing about 304,000 times when it should have been obvious the first time
I'm not going to do this again.
As expected. You make ridiculous blanket statements and do not have the data to back them up. This is ASR. When you make such a statement, expect other members to challenge them and ask you to back it up.

From the site you linked, it becomes obvious that it is a setting issue on your system:
As I said: windows CAN do these things but they are not always on. It would be ridiculous if they were.

Bass ManagementThere are two bass management modes: forward bass management and reverse bass management. Forward bass management filters out the low frequency content of the audio data stream. The forward bass management algorithm redirects the filtered output to the subwoofer or to the front-left and front-right loudspeaker channels, depending on the channels that can handle deep bass frequencies. This decision is based on the setting of the LRBig flag. To set the LRBig flag, the user uses the Sound applet in Control Panel to access the Bass Management Settings dialog box. The user selects a check box to indicate, for example, that the front-right and front-left speakers are full range and this action sets the LRBig flag. To clear this flag, select the check box. Reverse bass management distributes the signal from the subwoofer channel to the other output channels. The signal is directed either to all channels or to the front-left and front-right channels, depending on the setting of the LRBig flag. This process uses a substantial gain reduction when mixing the subwoofer signal into the other channels. The bass management mode that is used depends on the availability of a subwoofer and the bass-handling capability of the main speakers. In Windows, the user provides this information via the Sound applet in Control Panel.
 
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tuskenraider

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As others have stated: the weak link is the lack pf proper Room Correction in your AVR.

What will make an audible difference:
- Get a subwoofer. Even with tower speakers, these often provide better bass extension. Get a model with built in DSP in order to be more independent of the AVR's Room Correction implementation.
- Get an AVR with Audyssey/Dirac Live. Especially when watching movies, this is FAR more convenient that messing around with multichannel USB DACs.

What will not make an audible difference:
- getting a "better" DAC (SINAD is hopelessly overrated)
- getting a "better" amp (see above unless you actually need more power or lower noise floor)
- messing around with windows playback protocols. People obsess over WASAPI, resampling, exclusive mode ect, yadda yadda. Ultimately, it doesn't make a lick of difference. You can measure a quality improvement but you cannot hear it in actual program material (unless some software seriously fucks up the audio stream that is).

Since you already use a PC s a source device, you could try out the 14 day demo for Dirac Live's Studio version to see if you like it, provided you have a measurement microphone. With the new standalone mode, it's easier to use than ever.
So I have the Yamaha RX-A3080 AVR, Parasound 2-channel amp for the Revel F208 mains, SVS PB2+ subwoofer and let the AVR power the center and side surrounds. I cross over the mains at 40hz. I've played with 80hz as well. I have a 5-channel ATI amp that is just too much at 106 lbs., so I'm selling that. The Parasound has been repaired 3 times in the last 2 years and is 20yo, so I'm replacing it with a Audiophonics NCore amp. I just installed the Dirac trial on my HTPC and tested it a little yesterday. I'm doing some quick studying to figure how to implement it through the AVR to prevent any overlap of room correction systems. I have REW and a UMIK-1 for measurements that I understand the basics of using. As noted previously, I also have a Denon 4700H in my second system I could swap with the Yamaha since it clearly seems to be the better AVR for room correction.

Winamp is a knife, Jriver is a swiss-army knife. You'll get out of JRiver what you put into it. It's that simple.
Sure looks that way. I just want to open a playlist or a folder with music files, have the songs load and play, then throw on Milkdrop visualization and chill. Then be able to clear out the playlist at will and load something else. Linking to libraries, grouping, tagging, auto syncing, etc. is a bunch of crap I don't need. I'm new to the interface, but god forbid I could just be able to right click in a playlist window and be able to just clear it out all out once. The knife is absolutely perfect for my needs.
 
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Sure looks that way. I just want to open a playlist or a folder with music files, have the songs load and play, then throw on Milkdrop visualization and chill. Then be able to clear out the playlist at will and load something else. Linking to libraries, grouping, tagging, auto syncing, etc. is a bunch of crap I don't need. I'm new to the interface, but god forbid I could just be able to right click in a playlist window and be able to just clear it out all out once. The knife is absolutely perfect for my needs.

Take a look at Media Player Classic Black Edition. Install it, run through the settings real quick. Enable MPC Renderer in the audio section and set it for Exclusive Mode. Exit the program.

Now all you have to do is click on a movie or music file in windows explorer and it will pop up and play it for you ... Easy Peasy.
You can also save playlists and when you click them, it pops up to play them from explorer too.

After you use it a bit, you will find you can customize it's behaviour very easily, too...
 
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As expected. You make ridiculous blanket statements and do not have the data to back them up. This is ASR. When you make such a statement, expect other members to challenge them and ask you to back it up.

From the site you linked, it becomes obvious that it is a setting issue on your system:
As I said: windows CAN do these things but they are not always on. It would be ridiculous if they were.

Okay so I take it you listen in shared mode?
What depth and rate do you have set in the advanced pane of your output device?
Because that is the depth and rate that you are listening to everything at no matter what.
In shared mode you won't hear a difference... because there is no difference to hear!

Usually this panel defaults to "CD Quality" (16/44.1) which is not bad... but until you've heard a movie at it's native 16/48 or a Flack file at 24/96 you won't have a clue how much better it can be. The improvement is being erased by resampling, limiting, and lord knows what else your sound drivers are doing to the files.

As I pointed out on my website ... by the time you are flipping 96k down to 44.1 more than half of the samples in the original file are being thrown out. And when you go the other way more than half of what is being played is extrapolated from the original and inserted as extra samples... more than half ... how can that possibly be a good thing?

When I play a 24/96 file in Exclusive mode, it is played at 24/96, bit exact, to what is in the file. Movies play at 16/48 and compressed music plays at 16/44.1 ... as intended when the files were created.

I don't think there is a settings issue in my system ... I think you don't have the first clue how much better yours could be.

Now please... go ruin someone else's day.
 

Aerith Gainsborough

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I'm doing some quick studying to figure how to implement it through the AVR to prevent any overlap of room correction systems
Sound like a cool system already!
I think the easiest way would be to set the Yammy to "direct" and disable all prost processing on the AVR side while testing Dirac. That's how I do it with mine (it too ahs YPAO but I never use it).

I think, if you like the Dirac PC implementation, you could be done with the system itself after getting the NCore amp. Next step would be to look at room treatments, if applicable.

Don't feel bad regarding JRiver. I never got to vibe with it as well and stuck to the more simple foobar2000, that handles all my audio needs faithfully. :'D
 

Aerith Gainsborough

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As I pointed out on my website ... by the time you are flipping 96k down to 44.1 more than half of the samples in the original file are being thrown out. And when you go the other way more than half of what is being played is extrapolated from the original and inserted as extra samples... more than half ... how can that possibly be a good thing?
Most of the data in a 96K file is just meaningless noise to begin with, it can safely be deleted because it refers to frequencies that are outside of human hearing to begin with (let alone being out of band for most speaker systems). 44.1K is already an overkill format as far as human hearing is concerned. Outside of music production, there is really no need for anything beyond it. It's a ploy of the music industry because they wanted a High-resolution pendant to the HD/UHD of the video industry.

Yes, real time resampling of Windows is a bit messy, since it has to have a low footprint but just because you can see alterations in an analyzer, does not mean that you get audible changes. What happens some 80-90dB down the signal is completely irrelevant to our ears in any practical scenario.

OBVIOUSLY I had it set to exclusive when I did listening comparisons. Anything else would be idiotic on my part. I was even paranoid enough to download RME's test samples to verify that the signal reached the DAC bit-perfect. :'D
There was no audible difference, not that I would expect there to be any in the first place.
In general, I set my system to 48KHz/24Bit because I want to avoid any resampling of my piano VSTi due to latency concerns most of my Audio is 44.1K/16 though.
 

ZolaIII

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I'm using a WASAPI driver for Winamp and JRiver Media. I can see the frequency change on the Yamaha's information display as I change through different resolutions of audio files. JRiver also has an informational tab that shows if anything in the audio signal is being modified, which it doesn't as I'm using a WASAPI output on that player as well. If I switch to a default DirectSound driver, I can see the output to the AVR shows the static bit-rate that is selected in the advanced sound properties tab regardless file resolution.
You can do all the signal processing you need on PC - JRiver and leave AVR in direct mode (or whatever). That AVR has balanced inputs so if you choose to go with external DAC be sure to use them, there won't be much (if any) benefit of using better performing DAC. My advice is use either Toslink input or network wired/wireless DLNA server to fead it (with comprehensive DSP chain like JRiver of course) instead of Nv HDMI which is very limited (both in terms of supported PCM formats and quality) if you wish to improve things without investing big.
Initial setup for JRiver DLNA server is a bit fussy and not exactly problem free but once done you will forget about it. It needs a couple minutes to come on when you start the JRiver and that's about it regarding use, of course it supports full DSP chain (EQ, PEQ, room correction, VST and Winamp plugin's... transcoding and such) as DLNA server.
 

Aerith Gainsborough

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My advice is use either Toslink input or network wired/wireless DLNA server to fead it (with comprehensive DSP chain like JRiver of course) instead of Nv HDMI which is very limited (both in terms of supported PCM formats and quality) if you wish to improve things without investing big.
He also mentioned watching movies from the PC, so SPDIF is automatically worse quality for surround signals due to mandatory lossy compression. HDMI is the way to go in that case.
 

ZolaIII

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He also mentioned watching movies from the PC, so SPDIF is automatically worse quality for surround signals due to mandatory lossy compression. HDMI is the way to go in that case.
Nv HDMI is very limited regarding PCM support 36000/44100/48000 Hz and 16 bit only. I think he can live with lossy Dolby's with SPDIF/Toslink and PCM up to 192000/ 24 bit and of course I ment DLNA as for music only. Suggestion to use optical/network or wireless input is to minimise input IMD as much as possible not for convince purpose. I don't find a performance improvement of about 10 dB SINAD as small or insignificant and he will get that much by not using DSP block on amplifier (6~7 dB difrence) and avoiding HDMI (3~4 dB improvement). Even more so when it's for free.

Best regards.
 
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Aerith Gainsborough

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Nv HDMI is very limited regarding PCM support 36000/44100/48000 Hz and 16 bit only.
Your information is out of date.

SINAD in this context is completely irrelevant to audio applications. We're building stereo system not setting up a measurement rig. <_<
The HDMI SINAD will be way WAY better than anything his AVR DAC + AMP can do anyway.

The ability to watch multichannel movies using Dolby THD or DTS-MA with lossless audio is far more important in this case.

It boggles my mind how you can suggest lossy compression for that sake of some arbitrary SINAD value that is inaudible in either case. O_O
 
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But isn’t the question: how can that, in isolation, be a noticeable thing?
If you've never heard the "bit exact" reproduction, it would not be.
It would simply be your music as you're used to it.

But once you do...
 
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Nv HDMI is very limited regarding PCM support 36000/44100/48000 Hz and 16 bit only.

It should also be noted that USB Audio 1 is also limited it maxes out at 24b/96k which is good enough for most files, but to get beyond that you need to install an updated USB Audio 2 driver for your DAC... In Win7 through 10 you get that from your DAC's manufacturer.

(I believe Windows 11 now includes the USB Audio 2 driver...)
 

ZolaIII

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@Aerith Gainsborough am I (outdated)? I use Yamaha amp and it used to be HDMI to TV and Toslink from there (it still is for TV) and you won't get any other options than stated one's there. Difrence between 70 dB SINAD and 80 dB SINAD is much more significant than 110 to 120 dB as that's the real audible range (at least for most folks).
Lossy can be inaudible when done right (WavPack hybrid lossy for example) and that's for two chenel high dynamic range materials, in multichannel surround it doesn't even have to be done great.
All what I told you can get mesurent and A-B testing verification here and elsewhere and don't ask me why it is so (ask OEM's, industrial insider's or who ever you wish). You can argue about it as much as you like but do it with someone else as I don't really care.
Have a nice time and goodbye.
 

ZolaIII

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It should also be noted that USB Audio 1 is also limited it maxes out at 24b/96k which is good enough for most files, but to get beyond that you need to install an updated USB Audio 2 driver for your DAC... In Win7 through 10 you get that from your DAC's manufacturer.

(I believe Windows 11 now includes the USB Audio 2 driver...)
Now days even remotely modern OS-es and audio player's have integrated USB Audio 2.0 support so that is not an issue, however EMI over USB can and is.
 
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Now days even remotely modern OS-es and audio player's have integrated USB Audio 2.0 support so that is not an issue, however EMI over USB can and is.
If you are hooking up a modem and it requires a special driver to work ... then your system is USB A1. On USB A2 it would "just work".
Windows from Win7 to Win10 needed the update and a modem driver. The best I can determine MS didn't start with the USB A2 standard until Windows 11 was released and then started putting it into Win10 as an update.

As for the EMI on USB thing... that is so overblown it's actually laughable. ALL digital noise abatement methods are used in USB hardware, including differential signalling (the digital version of balanced audio) and common mode rejection on all inputs. Since there is no audio on the usb cable itself, there's nothing to mess up your precious music.
 

ZolaIII

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@Douglas Blake you missed a point. You can use old OS without support or if implemented one isn't good (on Android for example) and support embedded in audio player (USB kernel streaming in JRiver for example) to use it.
EMI problem is largely underrated in audio industry as it is, however it's very much a real problem. Difrence between just different USB cables can be up to 10 dB. Real most offending source is switching power supply's or better say power grid hum than goes trough (which you really can't block). On great engineered equipment which is authended regarding this (both regarding power source and proper shielding) this is not an issue (in normal use conditions) like newer hier grade desktop standalone DAC's but with power amplifiers it still is (along with plethora of other equipment).
 
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@Douglas Blake you missed a point. You can use old OS without support or if implemented one isn't good (on Android for example) and support embedded in audio player (USB kernel streaming in JRiver for example) to use it.
Not exactly. WASAPI Exclusive mode requires the player itself to decode/render the file as a PCM stream.
Without the new USB drivers you will be limited to 24/96, no matter what player you're using.



EMI problem is largely underrated in audio industry as it is, however it's very much a real problem. Difrence between just different USB cables can be up to 10 dB. Real most offending source is switching power supply's or better say power grid hum than goes trough (which you really can't block). On great engineered equipment which is authended regarding this (both regarding power source and proper shielding) this is not an issue (in normal use conditions) like newer hier grade desktop standalone DAC's but with power amplifiers it still is (along with plethora of other equipment).

Utter horsepucky.

USB ports ALL use differential signalling with common mode rejection at every input. The bits transferred to the device (a DAC in this case) are not the bits on the wire... they are copies of the bits on the wire transferred after common mode rejection... By the time the PCM stream gets to the DAC's inputs it has been filtered, copied, buffered and latched... 4 steps removed from what's happening on the cable.

If it's good enough to transfer gigabytes of data, error free, to a hard disk, printer, etc. it is more than good enough to transfer PCM audio.

This whole "usb cables make a difference" thing is just predatory marketing by companies hawking over priced garbage.
 

ZolaIII

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Sure thing, pick an ADC, bunch of USB cables and RMA and happy measuring, been there done that.
I don't use expense gimmick cables in any form, just good ones and there are perfectly good USB ones that cost couple bucks.
 

Aerith Gainsborough

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If it's good enough to transfer gigabytes of data, error free, to a hard disk, printer, etc. it is more than good enough to transfer PCM audio.
I think they are less worried about digital signal integrity than port noise, EMI etc being transferred along with it and messing with the analog stage of the DAC.

I can see a crappy cable doing that.

This is also why many people worship at the Altar of SPDIF, although that can have it's own issues.
 
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