Since MS Teams for linux only enumerates audio devices on startup, I have to restart the application when switching e.g. to my headphones. As a workaround, I'd like to create a persistent "alias" sink in pulseaudio, which I can re-route to any real sink on the fly (e.g. like pactl move-sink-input or the pavucontrol sink selection), without the application noticing.
Pulseaudio has some modules for "backing" sinks with others (for different purposes) but they all appear to fix the backing sink on startup: Tried module-combine-sink, module-remap-sink.
With module-null-sink I can create a virtual sink and have its output as "monitor" source, but I cannot get this source back to a (real) sink like my headphones.
I know I'm not answering your question, but this might help you work around your original problem.
I use ms teams in Google Chrome (I'd prefer Firefox, but teams calls don't work in FF). Then:
Some of the changes in devices will be enumerated and reflected, if not, a simple refresh is enough and is much easier than closing and re-opening the standalone application
The "Skype is using microphone" annoying problem is no longer observed
Related
I am testing .NET version of ZeroMQ to understand how to handle network failures. I put the server (pub socket) to one external machine and debugging the client (sub socket). If I stop my local Wi-Fi connection for seconds, then ZeroMQ automatically recovers and I even get remaining values. However, if I disable Wi-Fi for longer time like a minute, then it just gets stuck on a frame waiting. How can I configure this period when ZeroMQ is still able to recover? And how can I reconnect manually after, say, several minutes? How can I understand that the socket is locked and I need to kill/open again?
Q :" How can I configure this ... ?"
A :Use the .NET versions of zmq_setsockopt() detailed parameter settings - family of link-management parameters alike ZMQ_RECONNECT_IVL, ZMQ_RCVTIMEO and the likes.
All other questions depend on your code.
If using blocking-forms of the .recv()-methods, you can easily throw yourself into unsalvageable deadlocks, best never block your own code ( why one would ever deliberately lose one's own code domain-of-control ).
If in a need to indeed understand low-level internal link-management details, do not hesitate to use zmq_socket_monitor() instrumentation ( if not available in .NET binding, still may use another language to see details the monitor-instance reports about link-state and related events ).
I was able to find an answer on their GitHub https://github.com/zeromq/netmq/issues/845. Seems that the behavior is by design as I got the same with native zmq lib via .NET binding.
I have an app which is built with libVLC, accessing the standard installed libvlc-dev libraries, plug ins, headers etc on Ubuntu Linux.
My app generally works perfectly, and its mostly there to receive UDP streams and convert to something else.
However, I am having a really weird issue in a specific mode, and so far I have invested around 30 development hours in trial and error trying to solve it - and I am hoping some VLC genius here can unlock the puzzle.
It revolves around http:// based URL sources. Typically for HLS, but the same issue happens with any http based source.
IMPORTANT: If I launch my app in terminal, everything works perfectly (including http streams). However, if I 'sublaunch' the same app with the same launch parameters from my parent process using fork() and then execv(), it fails to play any http based streams (although things like UDP do still work perfectly).
I have checked the obvious things like ensuring the VLC_PLUGIN_PATH is set correctly, and I have exhaustively compared all other environment variables in the 2 launch states, without finding anything obviously related.
After enabling full logging, I can see there is a glaring difference during the url opening process - and it seems something is amiss when evaluating plug in suitability.
In Terminal Launch:
looking for access_demux module matching "http": 20 candidates
no access_demux modules matched
creating access: http://...snip....myfeed.m3u8
looking for access module matching "http": 28 candidates
resolving ..snip....myfeed
outgoing request:
and the stream plays fine
However, when forked, and execv, I see the following:
looking for access_demux module matching "http": 40 candidates
no access_demux modules matched
creating access: http://...snip....myfeed.m3u8
looking for access module matching "http": 56 candidates
and it sticks right there and does not ever even make an http call out.
Of course the odd thing which I hope may be a clue is that the forked environment finds twice as many candidates when matching. However, it fails to complete the http access stage, and goes no further.
This is driving me crazy, and I have given up 5 times so far, only to come back for another try. However, I have exhausted what I can discover via logging, and I am really hopeful a VLC developer here might be able to point me in the right direction.
Many thanks for any ideas, tips, gut instincts or whatever.
Thanks !
SOLVED: In the parent app, we happen to be calling: signal(SIGCHLD,SIG_IGN); at some point before the fork(). (so this is inherited by the child presumably) In this condition when forked and execved libVLC cannot work with http sources. There must be some behaviour which relies on SIGCHLD in VLC's http handling. We could solve the problem either by removing signal(SIGCHLD,SIG_IGN); from the parent, or by adding signal(SIGCHLD, SIG_DFL); to the child libVLC app. Once we do this, libVLC behaves as expected.
Recently, I wanted to get my hands dirty with Core Audio, so I started working on a simple desktop app that will apply effects (eg. echo) on the microphone data in real-time and then the processed data can be used on communication apps (eg. Skype, Zoom, etc).
To do that, I figured that I have to create a virtual microphone, to be able to send processed (with the applied effects) data over communication apps. For example, the user will need to select this new microphone (virtual) device as Input Device in a Zoom call so that the other users in the call can hear her with her voiced being processed.
My main concern is that I need to find a way to "route" the voice data captured from the physical microphone (eg. the built-in mic) to the virtual microphone. I've spent some time reading the book "Learning Core Audio" by Adamson and Avila, and in Chapter 8 the author explains how to write an app that a) uses an AUHAL in order to capture data from the system's default input device and b) then sends the data to the system's default output using an AUGraph. So, following this example, I figured that I also need to do create an app that captures the microphone data only when it's running.
So, what I've done so far:
I've created the virtual microphone, for which I followed the NullAudio driver example from Apple.
I've created the app that captures the microphone data.
For both of the above "modules" I'm certain that they work as expected independently, since I've tested them with various ways. The only missing piece now is how to "connect" the physical mic with the virtual mic. I need to connect the output of the physical microphone with the input of the virtual microphone.
So, my questions are:
Is this something trivial that can be achieved using the AUGraph approach, as described in the book? Should I just find the correct way to configure the graph in order to achieve this connection between the two devices?
The only related thread I found is this, where the author states that the routing is done by
sending this audio data to driver via socket connection So other apps that request audio from out virtual mic in fact get this audio from user-space application that listen for mic at the same time (so it should be active)
but I'm not quite sure how to even start implementing something like that.
The whole process I did for capturing data from the microphone seems quite long and I was thinking if there's a more optimal way to do this. The book seems to be from 2012 with some corrections done in 2014. Has Core Audio changed dramatically since then and this process can be achieved more easily with just a few lines of code?
I think you'll get more results by searching for the term "play through" instead of "routing".
The Adamson / Avila book has an ideal play through example that unfortunately for you only works for when both input and output are handled by the same device (e.g. the built in hardware on most mac laptops and iphone/ipad devices).
Note that there is another audio device concept called "playthru" (see kAudioDevicePropertyPlayThru and related properties) which seems to be a form of routing internal to a single device. I wish it were a property that let you set a forwarding device, but alas, no.
Some informal doco on this: https://lists.apple.com/archives/coreaudio-api/2005/Aug/msg00250.html
I've never tried it but you should be able to connect input to output on an AUGraph like this. AUGraph is however deprecated in favour of AVAudioEngine which last time I checked did not handle non default input/output devices well.
I instead manually copy buffers from the input device to the output device via a ring buffer (TPCircularBuffer works well). The devil is in the detail, and much of the work is deciding on what properties you want and their consequences. Some common and conflicting example properties:
minimal lag
minimal dropouts
no time distortion
In my case, if output is lagging too much behind input, I brutally dump everything bar 1 or 2 buffers. There is some dated Apple sample code called CAPlayThrough which elegantly speeds up the output stream. You should definitely check this out.
And if you find a simpler way, please tell me!
Update
I found a simpler way:
create an AVCaptureSession that captures from your mic
add an AVCaptureAudioPreviewOutput that references your virtual device
When routing from microphone to headphones, it sounded like it had a few hundred milliseconds' lag, but if AVCaptureAudioPreviewOutput and your virtual device handle timestamps properly, that lag may not matter.
I am taking my first dives in to the WASAPI system of windows and I do not know if what I want is even possible with the windows API.
I am attempting to write program that will record the sound from various programs and break each in to a separate recorded track/audio file. From the reseacrch I have done I know the unit I need to record is the various audio sessions being rendered to a endpoint, and the normal way of recording is by taking the render endpoint and performing a loopback. However from what I have read so far in the MSDN the only interaction with sessions I can do is through IAudioSessionControl and that does not provide me with a way to get a copy of the stream for the session.
Am I missing something that would allow me to do this with the WASAPI (or some other windows API) and get the individual sessions (or individual streams) before they are mixed together to form the endpoint or is this a imposable goal?
The mixing takes place inside the API (WASAPI) and you don't have access to buffers of other audio clients, esp. that they don't exist in the context of the current process in first place. Perhaps one's best (not so good, but there are no better alternatives) way would be to hook the API calls and intercept data on its way to WASAPI, if the task in question permits dirty tricks like this.
Why not use the Windows scheduler?
I have several applications that have to run at certain times according to business rules not the typical every weekday at 1pm.
I also need a way for the applications to provide feedback of their progress so that I can have rules that notify me when the applications are running slow or aren't even running anymore.
What Windows API should I be looking into? (like, a time version of the FileWatcher apis)
What's the best way to have the application notify the scheduler of its progress (files, sockets, windows messages, ???)?
For Vista/Win2k8, there's the nice Task Scheduler 2.0 API: http://msdn.microsoft.com/en-us/library/aa384138(VS.85).aspx. Previous version have the Task Scheduler 1.0 API, but I've never used it.
AppControls has a CronJob component that you can use to create scheduled events. This saves your program from having to wake up every minute and check the schedule itself. Instead, just schedule the job and indicate a callback method.
I have used this component for scheduling jobs myself and have been very happy with the way that it works.
I think what you really want is a common framework for your applications that report to something (you or the system messages or tracing or perfmon, event log, whatever) and also to receive via some inter process protocol a way to receive messages and respond.
based on the reporting you can change the scheduling or make changes, etc.
So, there is some monitor app, and then each of your other apps does common reporting.
events I can think of:
- started
- stopped
- error
- normal log messages
- and of course specific things your apps do.
I think there are probably existing classes/framework that do this - you'll have to check around.
If it were me, I would make a service that could talk to all the other apps and perhaps was even an http server. It would be able to route messages to particular apps and start stop those processes and query them.
There are lots of ways to do what you want though. those were just off the top of my head.
Alternatively you might just be able to get these to be services and they handle messages sent to them. Their normal processing does nothing until they are "woken up" with some task command.
You have more questions in one. Normally you should split them. But let's overlook this and try to answer.
To schedule certain events (including running an application): Use TJvScheduledEvents from JVCL. IMHO JVCL is the best Delphi open source library around with extensive number of components, developers & support. TJvScheduledEvents is quite neat, uses threads for event scheduling and also you have in JVCL a detailed editor for your events (it needs a small hack to use it though).
To provide 'feedback' from your applications to a (remote) central point: A very very very good solution (if your requirements permit) is to log the progress of your applications in a table (let's call it LOG) on a Firebird server. In LOG you can have the following fields: COMPUTER, USERNAME, APPNAME, MSG, LOGDATE (etc. etc.). In the After Insert trigger of the LOG table you can fire an event (let's call it NEW_LOG). In your console app you can register the interest for this event and so, your application will be automatically updated with everything which happens in any of your applications, so you can do log analysis, graphs etc. Of course you can do it with IB, but IB costs.
...going on Windows API route you need headers (which probably aren't translated), you'll encounter our dearest Pointers/PChars etc. etc. Of course, building from scratch everything isn't worthwhile but when this is already done in a Delphi way, why don't use it?
Use service with a timer that is fired regulary (for example each minute). It reads the schedule and looks if some are due before the next iteration. If so, you can execute them.
You can add an interface that shows all running apps. For the feedback and query that using a desktop application.