Hello, all --
I am attempting to set up an audio stream on my home network for music listening. I have a desktop PC and a little headless server, both running Arch Linux. The server is attached to speakers and contains all my music files on disk (and I've been using MPD to play those files for some time - no problem with that), but I need to use the desktop machine to reliably access some of the streaming music services that I use ... so, long story short, after a lot of web browsing and experimentation, I have determined that I need to run an audio stream from the desktop machine to the server.
[Oh, BTW, my answer to anything involving PulseAudio is NO. Don't have it, don't want it, don't want to hear about it. Thank you. However, I would definitely consider using Jack if my current effort fails. I use, like, and trust Jack. But for this purpose it seems like the fewer components involved, the better.]
Anyway, I found a document entitled "Streaming desktop audio on Linux without PulseAudio or JACK" (http://www.pogo.org.uk/~mark/trx/streaming-desktop-audio.html), which seems to address my need, and I have followed the recipe up to the point where you manually send the audio stream. Except that I am using mtrx (https://github.com/VittGam/mtrx) instead of trx. That's because I found mtrx before I installed trx, and based on its name and the fact that the code is more recent, presumably its author believes it is an improvement on trx - and since I don't have the expertise to be able to independently judge that claim, I just have to assume he knows what he's doing ;-) . Anyway, I also tried trx when I ran into the issue I'm about to describe, but the 'rx' program segfaulted when I sent the stream, so that's probably not going to work.
So, when I send an audio stream according to the instructions, I can hear the music over the speakers as expected, but the quality is not really acceptable. There are constant momentary dropouts. I suspect that this has to do with network performance and/or buffering, because the 'mrx' program running on the server is emitting error messages about buffer underruns. So I have a rough idea of what might fix this:
1) Adjust the buffer size? However, unlike trx, which allows you to specify a specific buffer size, mtrx has an 'alsa buffer multiplier' parameter, with a default value of 3. However, I have no idea what exactly that represents, or what the range of reasonable values is. Also, if I am setting this parameter, do I need to set it on both the server and the client - and do I need to set it to the *same value* on both sides?
2) Fix something in the network. I guess that's slightly off-topic, but I've never done anything with UDP over a network, so any tips would be appreciated. Note that while the desktop is connected via ethernet, the server is on WiFi.
a) Tweak some settings on the router. But I have no idea what settings to tweak.
b) Run an ethernet cable to the server. Not hard, but as a renter I am trying not to
attach too many things to the walls.
TIA for any assistance!
I am attempting to set up an audio stream on my home network for music listening. I have a desktop PC and a little headless server, both running Arch Linux. The server is attached to speakers and contains all my music files on disk (and I've been using MPD to play those files for some time - no problem with that), but I need to use the desktop machine to reliably access some of the streaming music services that I use ... so, long story short, after a lot of web browsing and experimentation, I have determined that I need to run an audio stream from the desktop machine to the server.
[Oh, BTW, my answer to anything involving PulseAudio is NO. Don't have it, don't want it, don't want to hear about it. Thank you. However, I would definitely consider using Jack if my current effort fails. I use, like, and trust Jack. But for this purpose it seems like the fewer components involved, the better.]
Anyway, I found a document entitled "Streaming desktop audio on Linux without PulseAudio or JACK" (http://www.pogo.org.uk/~mark/trx/streaming-desktop-audio.html), which seems to address my need, and I have followed the recipe up to the point where you manually send the audio stream. Except that I am using mtrx (https://github.com/VittGam/mtrx) instead of trx. That's because I found mtrx before I installed trx, and based on its name and the fact that the code is more recent, presumably its author believes it is an improvement on trx - and since I don't have the expertise to be able to independently judge that claim, I just have to assume he knows what he's doing ;-) . Anyway, I also tried trx when I ran into the issue I'm about to describe, but the 'rx' program segfaulted when I sent the stream, so that's probably not going to work.
So, when I send an audio stream according to the instructions, I can hear the music over the speakers as expected, but the quality is not really acceptable. There are constant momentary dropouts. I suspect that this has to do with network performance and/or buffering, because the 'mrx' program running on the server is emitting error messages about buffer underruns. So I have a rough idea of what might fix this:
1) Adjust the buffer size? However, unlike trx, which allows you to specify a specific buffer size, mtrx has an 'alsa buffer multiplier' parameter, with a default value of 3. However, I have no idea what exactly that represents, or what the range of reasonable values is. Also, if I am setting this parameter, do I need to set it on both the server and the client - and do I need to set it to the *same value* on both sides?
2) Fix something in the network. I guess that's slightly off-topic, but I've never done anything with UDP over a network, so any tips would be appreciated. Note that while the desktop is connected via ethernet, the server is on WiFi.
a) Tweak some settings on the router. But I have no idea what settings to tweak.
b) Run an ethernet cable to the server. Not hard, but as a renter I am trying not to
attach too many things to the walls.
TIA for any assistance!
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