Audio quality degradation with dual CIS connections on nRF5340

[Valentin Sagstetter]

Hello,

My application uses three nRF5340dk for LE Audio: two Unicast Servers and a Unicast Client. The Client scans for servers, establishes connections, performs discovery, and initiates audio streaming. The audio data from the CIS streams is written into ring buffers and is then combined into stereo packages and sent to the PC using UAC2.

The problem is that audio quality degrades with two simultaneous connections there is continuous buzzing in the recorded audio. I tried to increase the ACL connection interval to 1 second, but there was no improvement. Adding event spacing in the network core config did not help either. When the Unicast Client is only connected to a single Unicast Server, the audio quality is consistently good.

I am using the LE Audio API provided by Zephyr with the Basic Audio Profile. For the Network Core image, I am using the ipc_radio image, and the application core speed is set to 128MHz. I am using nRF Connect SDK v3.1.1.

My question is if there is a way to make this configuration work. I think I'm missing something, but I don't have an idea of what it could be. I'm still very new to Zephyr, and this is the first Nordic chip I'm programming. Any help would be much appreciated.

Kind regards,

Valentin Sagstetter

[Maria Gilje]

Hi Valentin Sagstetter, 

Could you please check if the characteristics for the audio streams changes when you add a second connection? I can help to find out how check for this in your firmware, but a link to the source code you are using is very helpful (i.e. a GitHub-link).

Best regards,

Maria

[Valentin Sagstetter]

Hello Maria,

Thank you for your help. I've uploaded the source code to GitHub: github.com/.../le_audio_to_uac2.
Let me know if you need any additional information.

Kind regards,

Valentin

[Maria Gilje]

Hi Valentin, 

The characteristics I'm thinking of are the ones in Lc3AudioDecoder::extractLc3Config(). When using that function in Lc3AudioDecoder::initialize, could you verify that the sample rate and frame duration stays the same when adding a second connection? It can also be worth to verify the configurations in the equivalent functions on the encoder side as well. 

If the configuration stays the same, please let me know what type of audio you are streaming: is it voice or audio? 

Just so you are aware: I will take some time off starting this evening due to christmas holidays. I will continue supporting you when I return the week after christmas. 

Best regards,

Maria

[Valentin Sagstetter]

Hello Maria,

I looked into the CodecConfig and realized I had the same Location set for both Encoders when creating the unicast group. I changed one Encoder to use Location FRONT_LEFT and the other to Location FRONT_RIGHT. I tested this configuration with the Galaxy Buds 3 Pro and it worked well.

Thank you for your help and Happy Holidays!

Kind regards,

Valentin

[Maria Gilje]

Hi Valentin, 

Good find! It's always nice to return to work to find solved cases.

Happy holidays to you as well! 

Best regards,

Maria