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[Mikkel]

Hello,

I'm using WASAPI-eventstyle and I always had the idea that it completely skipped the ASUS control panel settings of my ASUS HDAV1.3 Deluxe. It appears that isn't the case since speaker selection and sample rate selection in the ASUS control panel does indeed affect how many channels to output to and at what sample rate.

So now I have a question about how to setup JRiver and my soundcard:
1. Should I set the asus control panel to playback at the same sample rate as the source material?
2. Or should I just set the asus control panel to any sample rate I like, e.g. 48khz, 96 khz or 192khz depending on my preference?
3. A third quesiton, I guess, is if it will harm the sound quality and my use of convolution (see below) if I do upsampling on the soundcard irrespective of source sample rate.

My DSP settings in jriver are:
1. OUTPUT FORMAT
    No upsampling
    Force 24 bit depth
    Upmix to 7.1
2. CONVOLUTION
    Using convolution for 7.1 playback, autoselect filter based on sample rate is selected (I have filters for 44.1khz and 48khz), normalize selected

I hope you will help me sort out this mess :-)


Best regards,
Mikkel

[mojave]

I would recommend you using the ASUS Xonar Unified Drivers. This will allow you to use ASIO output with the HDAV1.3. With ASIO output, the ASUS control panel is completely bypassed and the sample rate will be automatically matched to the media. ASIO also automatically matches the highest bit depth supported by the card so it will use 24 bits regardless of what is set in Output Format.

Once you switch to ASIO, you can leave your DSP and convolution settings as you currently have them and everything will playback at its native sample rate and the convolution engine will select the correct convolution file.

[Mikkel]

I would recommend you using the ASUS Xonar Unified Drivers. This will allow you to use ASIO output with the HDAV1.3. With ASIO output, the ASUS control panel is completely bypassed and the sample rate will be automatically matched to the media. ASIO also automatically matches the highest bit depth supported by the card so it will use 24 bits regardless of what is set in Output Format.

Once you switch to ASIO, you can leave your DSP and convolution settings as you currently have them and everything will playback at its native sample rate and the convolution engine will select the correct convolution file.

I do in fact use the unified drivers. But I still experience glitches in the sound so that is why I use Wasapi (unfortunately)

[mojave]

I would set the ASUS control panel to use the source sample rate if possible. If nt, then set the ASUS control panel's sample rate to your preference. I would probably use 48 kHz. To keep the soundcard driver from resampling, I would then set Output Format to resample everything to the rate set in the control panel. In Convolution no auto matching will be used since you are now using just one sample rate.

[Mikkel]

I would set the ASUS control panel to use the source sample rate if possible. If nt, then set the ASUS control panel's sample rate to your preference. I would probably use 48 kHz. To keep the soundcard driver from resampling, I would then set Output Format to resample everything to the rate set in the control panel. In Convolution no auto matching will be used since you are now using just one sample rate.

Hi mojave,

I tried out asio once again and it seems to work. If I do other things in the background it can cause glitches in the audio but mostly not.
Thanks for recommending it :-).


/Mikkel

[mojave]

Hi mojave,

I tried out asio once again and it seems to work. If I do other things in the background it can cause glitches in the audio but mostly not.
Thanks for recommending it :-).


/Mikkel

In the ASIO Settings in JRiver you can increase the buffering to see if that will help. I also uncheck "use large hardware buffers" and then click "Open driver control panel" in the tools section. This opens a little ASUS ASIO control panel that allows you to change the bit-depth and latency. Make sure the bit-depth is set to 24 bit and increase the latency and try playback to see if it fixes the issue.