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 Post subject: AES67 Sample Rate
PostPosted: Thu Feb 20, 2025 3:25 am 

Joined: Mon Oct 07, 2024 4:13 am
Posts: 5
Hi. Using ST 10.50 (64 bit ALSA) on Ubuntu Studio. ST sample rate is set to 192khz for an MPX output via a Behringer UMD202HD. My issue is that my Dante plant is set to 48 kHz, and if I try to utilize the AES67 input on ST, the resulting audio sounds like chipmunks on speed. For the moment, I'm using a Avio Dante dongle to get audio into ST, but would like to use AES67 directly at some point. Any ideas on how to easily do a sample rate conversion for the AES67 ST input? Thanks in advance!


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 Post subject: Re: AES67 Sample Rate
PostPosted: Sun Feb 23, 2025 8:12 pm 

Joined: Mon Oct 07, 2024 4:13 am
Posts: 5
I can see this is not a popular topic. BTW, just changed my FM MPX sound card to a SMSL SU-1, instead of the Behringer UMC202HD...sounds great with much less overshoot. Now, if I can only figure out how to use the AES67 input with ST set to 192 KHz, and a Dante plant at 48 KHz.


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 Post subject: Re: AES67 Sample Rate
PostPosted: Fri Feb 28, 2025 12:06 pm 

Joined: Thu Apr 21, 2011 1:52 pm
Posts: 124
HI,

I have never tried, with dante and I don't know the behavior with linux....

I share my experience in livewire, I hope it helps...

Input aes67/livewire, input 48khz automatically blocked by ST based on the master clock.

Out HD, if you use aes67, it will use the network clock (I use axia asio, I also tested aes67 in the latest beta and it works), if you use a physical output (sound card) use wdm, KS, wasapi or mme drivers to test based on the hardware, in my case it worked by mixing various in/out drivers... mme and KS (without having buffer problems)

Out Fm always WDM (mme and KS work for me) (192KHZ from windows control panel) to test based on the hardware.

P.S. I don't know why I can't insert screenshots


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 Post subject: Re: AES67 Sample Rate
PostPosted: Thu Mar 13, 2025 8:58 pm 

Joined: Mon Oct 07, 2024 4:13 am
Posts: 5
Quote:
HI,

I have never tried, with dante and I don't know the behavior with linux....

I share my experience in livewire, I hope it helps...

Input aes67/livewire, input 48khz automatically blocked by ST based on the master clock.

Out HD, if you use aes67, it will use the network clock (I use axia asio, I also tested aes67 in the latest beta and it works), if you use a physical output (sound card) use wdm, KS, wasapi or mme drivers to test based on the hardware, in my case it worked by mixing various in/out drivers... mme and KS (without having buffer problems)

Out Fm always WDM (mme and KS work for me) (192KHZ from windows control panel) to test based on the hardware.

P.S. I don't know why I can't insert screenshots
Thanks for the guidance.

I've been experimenting with Beta 10.51-011 (64 bit ALSA Stand Alone) utilizing it for a FM MPX out and the new AES67 output. I realize that the AES67 HD output is experimental. I keep running into sample rate issues. For instance, because Stereo Tool is globally set for 192kHz to accomodate the MPX out, the AES67 out for HD is also at 192kHz, as a result. I get an HD out audio glitch/dropout every so often (which could be caused by several factors in my setup.) The AES67 standard calls for recommended support of 48 kHz even though 192kHz is possible. My Dante plant here is 48kHz to maintain compatability with all of my Dante and other AES67 endpoints.

I'm sure users of Stereo Tool utilizing 48kHz all the way around may not have any issue.

Hans, is it possible to have selectable sample rates on on each Stereo Tool audio input and output?

To make things work for my use case, I'd need a 48kHz input sample rate to make the AES67 Dante Input work for me. Right now, I cannot utilize the AES67 input in Stereo Tool because the input is expecting a 192kHz sample rate my Dante gear cannot provide. I'm stuck with using a Dante AVIO dongle to feed Stereo Tool. A 192kHz sample rate for the FM output is fine as I have a good DAC to deal with providing the MPX out, and a seperate 48 kHz output for the HD AES67 out would be useful. Ideally, a 44.1 kHz sample rate would be perfect for most all HD Radio importers. I'm pulling in the AES67 HD audio output from Stereo Tool with GStreamer on a Linux machine and doing the sample rate conversion there. Gstreamer works great with most AES67 sources! I've gotten around the need for a HD Radio 44.1kHz sample rate by using ALSA to do the conversion for my HD transmitter.

This is really a great product, and the ability to utilize different sample rates for all I/O would really maximize flexibility!


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 Post subject: Re: AES67 Sample Rate
PostPosted: Fri Mar 14, 2025 10:04 am 

Joined: Thu Apr 21, 2011 1:52 pm
Posts: 124
Quote:
Quote:
HI,

I have never tried, with dante and I don't know the behavior with linux....

I share my experience in livewire, I hope it helps...

Input aes67/livewire, input 48khz automatically blocked by ST based on the master clock.

Out HD, if you use aes67, it will use the network clock (I use axia asio, I also tested aes67 in the latest beta and it works), if you use a physical output (sound card) use wdm, KS, wasapi or mme drivers to test based on the hardware, in my case it worked by mixing various in/out drivers... mme and KS (without having buffer problems)

Out Fm always WDM (mme and KS work for me) (192KHZ from windows control panel) to test based on the hardware.

P.S. I don't know why I can't insert screenshots
Thanks for the guidance.

I've been experimenting with Beta 10.51-011 (64 bit ALSA Stand Alone) utilizing it for a FM MPX out and the new AES67 output. I realize that the AES67 HD output is experimental. I keep running into sample rate issues. For instance, because Stereo Tool is globally set for 192kHz to accomodate the MPX out, the AES67 out for HD is also at 192kHz, as a result. I get an HD out audio glitch/dropout every so often (which could be caused by several factors in my setup.) The AES67 standard calls for recommended support of 48 kHz even though 192kHz is possible. My Dante plant here is 48kHz to maintain compatability with all of my Dante and other AES67 endpoints.

I'm sure users of Stereo Tool utilizing 48kHz all the way around may not have any issue.

Hans, is it possible to have selectable sample rates on on each Stereo Tool audio input and output?

To make things work for my use case, I'd need a 48kHz input sample rate to make the AES67 Dante Input work for me. Right now, I cannot utilize the AES67 input in Stereo Tool because the input is expecting a 192kHz sample rate my Dante gear cannot provide. I'm stuck with using a Dante AVIO dongle to feed Stereo Tool. A 192kHz sample rate for the FM output is fine as I have a good DAC to deal with providing the MPX out, and a seperate 48 kHz output for the HD AES67 out would be useful. Ideally, a 44.1 kHz sample rate would be perfect for most all HD Radio importers. I'm pulling in the AES67 HD audio output from Stereo Tool with GStreamer on a Linux machine and doing the sample rate conversion there. Gstreamer works great with most AES67 sources! I've gotten around the need for a HD Radio 44.1kHz sample rate by using ALSA to do the conversion for my HD transmitter.

This is really a great product, and the ability to utilize different sample rates for all I/O would really maximize flexibility!
Hi,

StereoTool, if you use aes67 input, expects a sampling rate of 48khz (standard aes67), which will also be fine for HD output (48khz), the only thing you will be forced to change is the sampling rate of the sound card for FM, which must necessarily be 192khz, just set the alsa driver and change the sampling rate of the sound card, if it is possible, I repeat, I do not use linux.

Best
Luca


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 Post subject: Re: AES67 Sample Rate
PostPosted: Sat Mar 15, 2025 3:22 am 

Joined: Mon Oct 07, 2024 4:13 am
Posts: 5
Luca,

Thanks again for the tips. Linux is a different beast for sure!

I took your advice, and set my Stereo Tool sample rate to 48kHz. Got the AES67 input working fine. AES67 HD output was also 48 kHz sample rate, and magically the SMSL DAC I use for the MPX automatically set itself to 192kHz! I think the key was to set the synchronization between different sound cards to 'Off'. Still have glitches with the AES67 HD output, so I went pack to the Behinger UMC204HD device I was using to feed analog HD audio out to the sound card in HD Radio importer. The AES67 Dante input has no glitches and works fine. That will free up the AVIO dongle I was using for input...maybe I'll set things up so the AVIO dongle can send AES67 audio to the HD Radio Importer.

I'll wait to hear from Hans that the AES67 HD output is ready for prime time. It is neat to experiment with at the moment.

So, no issues with sample rates at the moment. All seems good, except the glitch on the AES67 HD out, which caused my audio diversity alignment to change over time. It could be an issue with clocking of GStreamer receiver, not sure...I know there is a way to provide PTP to the pipeline, but I haven't figured that out yet.


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 Post subject: Re: AES67 Sample Rate
PostPosted: Sat Mar 15, 2025 2:11 pm 

Joined: Thu Apr 21, 2011 1:52 pm
Posts: 124
Quote:
Luca,

Thanks again for the tips. Linux is a different beast for sure!

I took your advice, and set my Stereo Tool sample rate to 48kHz. Got the AES67 input working fine. AES67 HD output was also 48 kHz sample rate, and magically the SMSL DAC I use for the MPX automatically set itself to 192kHz! I think the key was to set the synchronization between different sound cards to 'Off'. Still have glitches with the AES67 HD output, so I went pack to the Behinger UMC204HD device I was using to feed analog HD audio out to the sound card in HD Radio importer. The AES67 Dante input has no glitches and works fine. That will free up the AVIO dongle I was using for input...maybe I'll set things up so the AVIO dongle can send AES67 audio to the HD Radio Importer.

I'll wait to hear from Hans that the AES67 HD output is ready for prime time. It is neat to experiment with at the moment.

So, no issues with sample rates at the moment. All seems good, except the glitch on the AES67 HD out, which caused my audio diversity alignment to change over time. It could be an issue with clocking of GStreamer receiver, not sure...I know there is a way to provide PTP to the pipeline, but I haven't figured that out yet.
hi,

well, I'm happy to help....
in theory the synchronization between various sound cards should be activated...if it doesn't give you buffer problems, better that way

best
luca


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 Post subject: Re: AES67 Sample Rate
PostPosted: Tue Mar 25, 2025 5:49 pm 
Site Admin
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Joined: Mon Mar 17, 2008 1:40 am
Posts: 11414
Sorry for not responding sooner, we're all traveling at the moment. Indeed, you can set the sample rate to 48, the FM will automatically upsample by a factor 2 or 4 if needed.

Now, those glitches. I was kinda afraid that that could happen. If you open the detailed settings, there should be a display that shows if AES67 is running on the internal clock (which would be bad) or PTP. If it uses PTP, it will also show offset values that are updated every second. My question is: If it uses PTP (if not, the question is why not), is there any correspondence between those dropout and large displayed offset values? The one thing on the todo list for AES67 support is to smoothe out large jumps in this - but, with a high quality PTP source there really shouldn't be any, I think at least.


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 Post subject: Re: AES67 Sample Rate
PostPosted: Mon Apr 14, 2025 9:07 pm 

Joined: Mon Oct 07, 2024 4:13 am
Posts: 5
Hans,

Many thanks for the reply. Hope the NAB treated you well!

I set up my HD Radio feed to use a Audinate AVIO dongle on both ends to handle the audio transport. I'll wait to hear the Linux Beta builds are ready to go before trying out the AES67 output again.

I've also been traveling for work, and havent had a chance to get back on this!

Have a great week!


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