I believe I've identified the root cause of the "Lost Control of Audio Device" message with Yamaha ASIO

After a lot of testing, I no longer believe this is related to TIDAL, my music library, USB stability, or the Yamaha driver installation.

Here are my findings:

  • 44.1 kHz → 44.1 kHz :white_check_mark: No issue

  • 48 kHz → 48 kHz :white_check_mark: No issue

  • 96 kHz → 96 kHz :white_check_mark: No issue

  • 44.1 kHz → 48/96 kHz :cross_mark: “Lost Control of Audio Device”

  • 48 kHz → 44.1/96 kHz :cross_mark: “Lost Control of Audio Device”

  • 96 kHz → 44.1/48 kHz :cross_mark: “Lost Control of Audio Device”

To eliminate album changes as a factor, I created a queue with tracks of different sample rates. The result was exactly the same. The error only occurs when the sample rate changes between consecutive tracks. If the next track has the same sample rate, playback continues seamlessly.

I also tested increasing the Resync Delay to 3000 ms and 5000 ms. This did not resolve the issue—it only delayed the start of playback.

For comparison, I tested WASAPI. It never produced the error because Windows was fixed at 96 kHz, so Roon resampled every track to 96 kHz. The Yamaha DAC never had to switch its sample rate, which explains why the issue never occurred.

With ASIO, Roon correctly switches the Yamaha DAC to the native sample rate of each track (44.1, 48, 96 kHz, etc.). During that transition, Roon briefly reports “Lost Control of Audio Device.” As soon as the Yamaha finishes switching, the ASIO device is available again, and simply pressing Play resumes playback immediately.

My environment:

  • Yamaha R-N800A

  • Yamaha Steinberg USB Driver v2.1.9

  • Windows 11

  • Latest Roon version

  • ASIO output

  • Stable mode, 512-sample buffer

At this point, I don’t believe the issue is caused by my hardware. It appears to be related to how the ASIO stream is handled during sample-rate transitions.

Has anyone else with a Yamaha ASIO device observed the same behaviour?

Adding Resync delay will delay the start of playback. It also gives the DAC time to resync to the new sample rate. Consequently, this suggests that it is the DAC that requires the delay since control of the device is lost while resyncing. Keep this as low as possible.

However, given the way the DAC behaves, you may want to consider upsampling everything to 192 kHz or similar.

This is your related support ticket, which provides some explanation.

Thanks much for the suggestion, I understand that upsampling everything to 192 kHz would avoid the sample-rate transitions, but my goal is to achieve bit-perfect playback.

Why ? The dac chip will do oversampling anyway. And roons upsampling algorithm beats the one in your dac

Your Yamaha has a SABRE ES9080Q chip and supports PCM384khz but,

YPAO supports room correction up to 192khz. So I would feed the dac 192khz.

Thanks Marc, would definitely give a try :slight_smile:

I developed that problem when I installed an EtherRegen. SuperDad worked with me to solve the issue and finally narrowed it down to latency issues in my network. Roon has some settings that can help alleviate that problem and that along with changes to some of the switches the signal was traveling through solved it for me.

Thanks for sharing your experience. In my case, I’m using the Yamaha R-N800A via USB ASIO rather than a network endpoint. I was able to reproduce the issue consistently only when changing between tracks with different sample rates (44.1 → 48 → 96 kHz, etc.). The RAAT logs show a device_reset from the Yamaha ASIO driver during the sample-rate transition, so at the moment it seems unrelated to network latency.

Have you fried increasing the buffer size in the driver control panel?

Thanks for the suggestion. I did some more testing while monitoring the Yamaha Steinberg USB Driver. I noticed that the driver automatically adjusts its buffer size depending on the playback sample rate. For example, both 44.1 kHz and 48 kHz use a 2048-sample buffer, while 96 kHz uses a 4096-sample buffer. However, I still see the “Lost Control of Audio Device” message when switching from 44.1 kHz to 48 kHz, even though the buffer size remains the same. This makes me wonder if the issue is related to how the driver reinitializes when the sample rate changes, rather than the buffer size itself. Happy to hear your thoughts if I’m overlooking something. Attached the pictures for your reference.

Reports elsewhere suggest that a larger buffer size eliminates the issue. Therefore, experiment with a larger buffer size.