Currently we use alsasink in gstreamer clients to connect to pipewire. Alsasink has 10ms latency by default, which produces 480-sample buffers at 48 kHz. This eventually forces pipewire to use 256-sample buffers internally in order to keep up. This is not good enough for many use cases and causes underruns. We should instead use 21.3 ms latency, which at 48 kHz translates to 1024-sample buffers. This matches the latency we have in the qemu virtual sound card for instance, but also on others, and it produces much smoother results.