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I am doing multi-room audio in my house as described at posts like this one:

Multi-room audio with multicast RTP
http://www.danplanet.com/blog/2014/11/26/multi-room-audio-with-multicast-rtp/

My problem is that sound quality is poor. In particular, there are frequent pops. I have a Gigabit wired LAN and all computers are Intel Core i5 or i7 (no Raspberry Pi's or other low power devices). (I believe Intel CPU's are little endian.)

My PA configuration is described in more detail here:
https://unix.stackexchange.com/a/471787/15010

Since then I have added latency_msec=1000 to module-rtp-recv on each receiver.

On the sender, I am thinking about adding rate=44100 channels=2 format=s16le. However, those are already the defaults on all devices:

  • PulseAudio Version: 12.2
  • Default Sample Specification: s16le 2ch 44100Hz

Also, all are synchronized with an NTP server:

sudo timedatectl status
System clock synchronized: yes
NTP service: active

First question: how do I add rate=44100 channels=2 format=s16le when loading the module using pactl instead of changing /etc/pulse/default.pa?

Second, am I on the right track with those proposed changes? What else could be responsible for my poor sound quality? My hardware is pretty high end and the GigE network has good performance (although I would have to learn how to quantify that performance if that becomes necessary).

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As stated in the question, I added latency_msec=1000 to module-rtp-recv on each receiver and that did not resolve the issue. Since then, I changed it to latency_msec=4000 and that did resolve the issue. I did not try intermediate values.

I consider this only half an answer. I would still like to fine tune other parameters such as rate and format, but so far I have come across the correct instructions for doing so. If anyone comes up with a better answer, I'll accept yours. For now, this is the best answer I came up with through trial and error; it was enough to make the music experience significantly better.

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