Pulseaudio monitor

By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service. The dark mode beta is finally here. Change your preferences any time. Stack Overflow for Teams is a private, secure spot for you and your coworkers to find and share information. Faced strange PulseAudio monitor device i. It works for example 30 seconds and prints read bytes count. Problem is that bytes count vastly differs if something is played during program run.

I've executed this program and in parallel executed bash for loop consisting aplay with short tada. But if instead of several aplay with short wav I run mplayer with long mp3 file - there is no such difference, bytes count is similar to case when no sound is played.

Such behaviour causes failure of sound processing code in my real project, so if somebody could suggest how to solve it - I'll be very grateful. Similar code on Windows, Qt based and using Stereo Mixer device as analog of PulseAudio monitor works without such problems. Test 2. Bash for loop "for i in seq 1 22; do aplay tada. Test 3. Test 4. Not a full answer, but a little explanation and workaround:. You're probably hitting a known bug we have with rewind handling in monitor sources "known" means in this case that the symptoms are known, but not the exact cause.

When a stream starts to play to the monitored sink, the sink will rewrite its playback buffer contents this is called "rewinding"and this causes a glitch in the monitor source some extra audio apparently gets pushed to the recording stream, judging from your experiments. I hope to fix this some day, but it doesn't look like I'll have time for it in the near future.

If you have the time and motivation to investigate and fix the bug, that would be awesome. As a workaround, you can probably reduce the magnitude of the error by reducing the playback buffer size.

Learn more. Strange PulseAudio monitor device behaviour Ask Question. Asked 3 years, 2 months ago. Active 3 years, 2 months ago. Viewed times. Dmitriy Vinokurov.

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Dmitriy Vinokurov Dmitriy Vinokurov 2 2 silver badges 15 15 bronze badges.I'm working on driving an analog VU meter from my Raspberry Pi using whatever audio is going out the Pi's sound outputs. The de facto Linux sound system, PulseAudioallows any sound output or "sink" in PulseAudio's nonclementure to be monitored. In PulseAudio land, each sink has a corresponding "source" called the monitor source which can be read just like any other other PulseAudio input such as a microphone.

In fact, to help with volume meter style applications, PulseAudio even allows you to ask for peak level measurements, which means you can sample the monitor sink at a low frequency, with low CPU utilisation, but still produce a useful volume display.

When this feature is used, each sample read indicates the peak level since the last sample.

How to install Pulseaudio equalizer on Linux

This makes it a little difficult to figure out how to get everthing to hang together. Using the code from various open source projects primarily veromix-plasmoid and pavumeteralong with the API reference, I was able to develop a fairly minimal code example that will hopefully be useful to others trying to do something similar. Although this example is written in Python, it is using the PulseAudio C API directly via ctypes so it should hopefully still be relevant if your application is written in C or another language.

Here's the demo code. The latest version is also available on Bitbucket. If you're not sure of the sink, just run it - the program prints the details of all available sinks to stdout. If all goes to plan you should see a basic volume display in the console when sound is actually playing!

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It needs the name of a sink to monitor and the sampling rate. Iterating over a PeakMonitor instance will give 8-bit peak level samples actually they're from to - see comment in code and comment from Tanu on this article.

The main function implements a simple volume display. Some of the constants at the top of the code control how the volume level is displayed.

This goes for API calls that return information about the sound configuration of the system as well as API calls that return or accept actual sound data. The reason for this is that the available sound devices may change at any point as a program is running e. It doesn't seem like a good idea to constantly have the CPU doing that much work. I plan on trying out Cython to convert the PulseAudio interation parts to C or rewriting the entire program in pure C since it's not really taking much advantage of Python's features.

More on this to come. Please note that I'm by no means a PulseAudio expert. Please let me know if there's anything I could be doing better and I'll update the article.

This article is actually quite far behind where I'm actually at with the VU meter project. I'm hoping to publish some more articles this week. Update : I've updated the code and article to explain about why peak samples only range from to See Tanu's comment below too.

Update : There's now a new article that describes how this code is used to actually drives a VU meter from the Raspberry Pi. Welcome to the blog of Menno Finlay-Smits. Here you will find infrequent postings, mainly - but not always - about software development, technology and general geekery. I build stuff at The Cacophony Projectbut the opinions expressed here are my own.

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Elsewhere menn0 mastodon.The latter two share a set of speakers, but I hate having to climb under the desk to switch the cables around, and wanted a better way to switch them back and forth.

My good friend Tyler B suggested that I run the line out from the XBOX into the line-in on my sound card, and just let my computer handle the audio in the same way that it handles music and movies. In theory, this works great. After quite a bit of googling, I found the answer on the Ubuntu forums. Open up a terminal and use the following commands to permanently enable this behaviour. The first line instructs PulseAudio one of the many ways that your system talks with the underlying sound hardware to load a module called loopback, which unsurprisingly, loops incoming audio back through your outputs.

This means that you can hear everything that comes into your line-in port in real time.

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Note that this behaviour does not extend to your microphone input by design. The second line simply tells PulseAudio to load this module whenever the system starts.

Hi, I also tried this and it worked and I was really excited. Only problem is : the lags behind by about 1 sec, which kinda beats the point of playing video games through line-in :S. Thank you! This was one of the things that has kept my last Windows machine from being formatted.

Now to find a way to play those Popcap games under Linux. Great article, my only question is how do you unload the module? I found a way to fix it. Basically make a script that starts on startup no need to add the above loopback module. In the script but the line:. Best of luck. Please help. Many thanks! Thanks to a Uberuser I know I was able to roll back the changes doing this:. The first line only affects your current session.

It should have no effect after you restart your system. To remove that line from that file use this command in terminal:. That just opens the file with gedit with root permissions. Restart your system and things should hopefully be back the way they were. How do you do that with pulse? Yes, yes, yes! Thank you so much for this! This fixed everything, and now I can finally listen while I record!

All you have access to in your GUI mixers nowadays is global pulseaudio level when you should have BOTH pulseaudio and hardware levels and switches.PulseAudio is a general purpose sound server intended to run as a middleware between your applications and your hardware devices, either using ALSA or OSS.

It also offers easy network streaming across local devices using Avahi if enabled. While its main purpose is to ease audio configuration, its modular design allows more advanced users to configure the daemon precisely to best suit their needs.

Install the pulseaudio package. Some PulseAudio modules are not included in the main package and must be installed separately if needed:.

pulseaudio monitor

By default, PulseAudio is configured to automatically detect all sound cards and manage them. It takes control of all detected ALSA devices and redirects all audio streams to itself, making the PulseAudio daemon the central configuration point. The daemon should work mostly out of the box, only requiring a few minor tweaks. While PulseAudio usually runs fine out of the box and requires only minimal configuration, advanced users can change almost every aspect of the daemon by either altering the default configuration file to disable modules or writing your own from scratch.

The daemon by itself does nothing without its modules except to provide an API and host dynamically loaded modules. The audio routing and processing tasks are all handled by various modules, including PulseAudio's native protocol itself provided by module-native-protocol-unix. Clients reach the server through one of many protocol modules that will accept audio from external sources, route it through PulseAudio and eventually have it go out through a final other module.

The output module does not have to be an actual sound output: it can dump the stream into a file, stream it to a broadcasting server such as Icecastor even just discard it.

You can find a detailed list of all available modules at Pulseaudio Loadable Modules. This is the main configuration file to configure the daemon itself.

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It defines base settings like the default sample rates used by modules, resampling methods, realtime scheduling and various other settings related to the server process. These can not be changed at runtime without restarting the PulseAudio daemon.

The defaults are sensible for most users, see the pulse-daemon. Boolean options accepts any of these: trueyeson and 1 as well as falsenooff and 0. This file is a startup script and is used to configure modules. This will make the daemon load the CLI module and will accept the configuration directly from the command line, and output resulting information or error messages on the same terminal.This lets you attach to a given Sink and chain more stuff to it, for example, recording the audio that is playing through PulseAudio at any given moment.

This is very handy for creating, for example, PubQuiz-style clips of songs, movies, etc. Would you provide a link for an updated sox? The latest version in the Ubuntu repositories still causes this problem. The sox from Ubuntu Jaunty 9.

Comment by xuzo — April 20, am. This avoids the SOX problem pointed out by vadim, kees and 3pipe and, at the same time, gets you Comment by hacklschorsch — October 26, am. Comment by kees — November 18, am. Thanks so much. I just wasted a bunch of time trying to do this the wrong way and it is such a relief to have found your solution. Comment by Ben Smith — January 14, am. Thank you for this script! I added the ability to start when a sound occurs and to split in multiple files after 2s of silence.

Comment by kaapstorm — January 16, pm. Sorry about that. Comment by eric — March 6, pm. Thanks for the updates! Comment by kees — April 6, pm. Comment by Al Williams — July 12, pm. Powered by WordPress. Comments What I would like to do: have my line-in audio coming from my monitor playing on game console to my motherboard and listen to the audio through my PC. The problem is I cannot use my microphone to talk to other people over VoiP programs like Discord. In pavucontrol-qt I have line in set as the port under input devices but if I want to talk over discord it pucks up the audio from the monitor, like-wise if I have microphone selected in pavucontrol-qt I can use my microphone to talk but the loopback makes me hear my voice and I can't hear audio from my monitor anymore.

I was able to do this perfectly on Windows 7 with 'Listen to this device' and Windows audio being able to select a 'Default communications device'. I tried something similar in Pulseaudio a few years ago and found it too complex and unreliable and lot of extra latency delay from looping through mutiple modules.

Only want to use my microphone with Discord, otherwise the people I'm talking to will only hear the audio from my game. I figured out a way to do it with ALSAMixer's loopback, but I need to leave pavucontrol-qt open in order for it to work. Under pavucontrol-qt change your input device to Rear Microphone since Discord depends on Pulseaudio.

Leave pavucontrol-qt open, change your monitor's input to your game consoles and try Discord's microphone check.

PulseAudio: Monitoring your Line-In Interface

If there's a way to do this while having pavucontrol-qt closed that would be great or to close it to the system tray or a jack-for-dummies guide that'd be great. No idea if alsa has as much latency as pulse but I haven't noticed any awful delay. If microphone is single audio source required for Discord, should be able to configure microphone in Discord settings or pavucontrol - t 2. Thank you, got it to work!

It's not a big deal since I use KDE's volume in the system tray. This topic was automatically closed 30 days after the last reply. New replies are no longer allowed. Pulseaudio: Listen to line in through speakers and use microphone for communication Newbie Corner. ALSA loopback in hardware is better option. Forum kindly sponsored by.Note all methods described here stream raw PCM audio over the network. This can use pretty much network bandwidth around 1. If you get choppy sound, try setting a lower sample rate for the network stream.

Furthermore, even while many WiFi connections can sustain such bitrates, often the jitter in packet latency makes transmitting low-latency audio over a wireless link infeasible in practice. See Server Strings for an explanation of the format. In this FAQ entry all locations you can specify the server to use are listed. All the methods that connect to the daemon over the network using the native protocol need module-native-protocol-tcp loaded.

This includes tunnels and Zeroconf setups. With this module loaded, the server listens on port for incoming client connections.

For authentication you need the same auth cookies on all sides. Alternatively the authorization cookies can be stored in the X11 server. The server must have module-native-protocol-tcp loaded.

These two modules are not loaded in the default configuration because they might open PulseAudio for remote attackers.

These properties are much like environment variables, but will be available remotely if you SSH to another machine with X11 forwarding. You can see a list of PulseAudio related properties by doing:.

pulseaudio monitor

Note that only the X11 properties are forwarded over the SSH tunnel, but the pulseaudio client still connects to the server using its own native protocol.

With a tunnel you can create a new sink that forwards all audio over the network to another server. For the sink at the remote server the tunnel looks like just another stream connecting over the network. The same holds for sources. See the documentation on module-tunnel for details on the module arguments. Setting up a tunnel requires a running PulseAudio daemon on the remote server with module-native-protocol-tcp loaded, just like with the direct connection.

After the tunnel is set up, client applications connect to the tunnel sink on the local PulseAudio daemon. This has the advantage that you can switch the stream seamlessly between a local hardware sink and the tunnel sink.

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With a direct connection the client generally has to be restarted in order to switch servers. A direct connection has the advantage that the client has more control over buffering parameters. In order to avoid having to setup tunnel manually between computers on a network, Zeroconf can be used.

pulseaudio monitor

Setup module-zeroconf-publish and module-zeroconf-discover manually or use the check box in paprefs. In addition make sure to load the module-native-protocol-tcp and that it allows connections from other hosts, see Authorization above.

PulseAudio/Examples

These modules are not loaded in the default configuration because they might open PulseAudio for remote attackers. It is a well-known protocol for transferring audio and video data over IP. They can be used in both multicast and unicast fashions. For more information about using these technologies with PulseAudio have a look on the modules documentation.

This will make rtp the default sink, i. On the client sides just load the reciever module:.

pulseaudio monitor

Now you can play your favourite music on the sender side and all clients will output it simultaneously. BTW: You can have more than one sender machine set up like this.


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