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.
Related
I do have two computers which are used to control an industrial plant. One of them controls the plant, the other is used as a failsafe. They are directly connected over ethernet, and the inactive" one just mirrors the display of the main controller.
I did capture the network traffic between the two and when i open it up in wireshark i see its all X11 traffic. It does include the initial connection request and also includes all the "draw calls" in plain text.
I now want to "replay" this captured stream and recreate the screen content from it. Is there any program available which can do so? Ideally directly from the wireshark capture file
My thoughts so far:
I can easily replay the network data itself and send it to some socket, but the communication is specific to the session, e.g. some commands refer to specific handle values set up earlier. Its unlikely a new session would work with the same values so i cant just pipe it into some program
What you see from your connection is only your connection requests + events relevant to the windows created by you ( or other's client windows where your connection sets an event mask ), and because of that quite a lot is lost. I'm not aware of the programs that can reconstruct best possible version of the screen from one client traffic but it's certainly not possible to have 100% accurate copy of the screen and best possible model will be far away from real screen (unless your connection periodically polls for backing store content of each mapped window).
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.
We have a system where there are typically two processes running on the same system. One process handles the GUI and the other runs like a service (although for historical reasons, it's not a service, just an exe with no visible window).
The two processes undertake IPC mainly via registered messages asynchronously - i.e. we use RegisterWindowMessage() in both processes to define a large'ish set of messages that effectively form the API to the server process.
I have written a "hands-free" monitoring application that uses SetWindowsHookEx() to monitor and display the message queues of both processes and provide some level of decoding of the way the API is being utilised and how notifications are being propagated to the GUI process (each individual window can subscribe to notifications from the server directly).
So, there are a large number of messages in both directions so I have filtering and summary counts etc. so I can focus on particular activity. All this can be done without affecting the live code, which is good.
This all works well, but it now would be very useful to be able to "tag" a message originating in the GUI so I can trace the same message when it's processed by the server. This would be enormously useful for debugging and diagnosing system issues, but I can't find a clean way (actually I can't find any way!) of doing this without adding such support to our registered message API, which would be a lot of work and involves more risk than I'm comfortable with at the moment. It gets further complicated by the fact that the server pre-processes some messages and then does a PostMessage() back to itself to perform the action, so the originating message can get "lost".
Has anyone here tackled this type of problem? If so, can you give me some pointers? If not, then are there any documented or undocumented ways of adding a small block of data to a Windows message and retrieving it later? I've looked at SetMessageExtraInfo() but that seems to be per-queue rather than per-message.
FindWindow or FindWindowEx will give you the details of the GUI Window. Compare the details with message intercepted
The team I am working with has bought a CloudPort license (from CrossCheck Networks) and we are currently facing the problem of not being able to implement any sort of logic in the service Mocks (to control response selection). It would be something as simple as:
if (requestCounter++ == 1)
then
response = $fn:Global(MyFirstXmlString)$ // <-- this is CloudP syntax for vrbls
else
response = $fn:Global(MySecondXmlString)$.
We did not find any sample for using the Dll Plugin and neither of the two JScript and VBScript Tasks are working (i.e., our client machine gets back not the desired MySecondXmlString response but instead a Fault with
<faultstring>
ActiveX control '0e59f1d5-1fbe-11d0-8ff2-00a0d10038bc' cannot be instantiated
because the current thread is not in a single-threaded apartment.
</faultstring>.
Believe it or not, the fault above is being obtained even if the J- or VB-Script task is left empty! It's hard for us to believe that all the logic functionality advertised in the CloudPort UI is fake and that nothing is able to help one implement the kind of logic described above.
Any help would be appreciated!
Thanks,
Pompi
PS: A bit more details here on why the kind of logic described above is needed: We use SoapSonar in our testing framework to fire requests to a BizTalk orchestration application. The CP mocks are needed to simulate the environment of that BT orchestration. We cannot control individual mocked responses via SSonar requests: the (for cloudport: incoming) client requests are made by Production code and cannot be altered or controled by our SSonar client). The only Tasks functionality that worked for us is a DB-table working as an offline channel b/w SSonar and CP (SSonar writes in it and CP reads from it). CloudPort's reading of, say, responseXml's, from DB works fine but we cannot find a way to implement further behavior controlling logic on the CP side. Therefore, this stackoverflow posting. And thx for having the patience to read this whole shananigan :).
Don't think you can control this from the script.
The threading model should be controlled by the host, which I suppose uses windows's "vbscript.dll" for the actual execution.
So if you cannot find any settings under the tool's options or in the help :), you should look in the registry keys for the threading options of that ActiveX or "vbscript.dll"
That is the "ThreadingModel" value and try to change the values (you will also have to search the net for those, don't know them by heart).
There are chances that some other application (antivirus?) has changes the path to the dll that the COM interface should actually point to (see http://social.technet.microsoft.com/Forums/en-US/ieitpropriorver/thread/ac10bd5f-6d91-4aac-857c-0ed5758088ec)
Hope it helps.
I am building an music player using Background audio player agent on WP7. I want to enable communication between the UI part and the agent part. Many guides suggest using isolate storage, but I think that is not a good way
Is there any way to enable inter-process communication in Windows Phone 7
In Windows Phone 8 SDK, we can now use system-wide Mutex object.
It seems the foreground App and Background Agent run as separate processes on the phone. So even when you instantiate the same class, each process has a different instance.
The best solution I know about so far is to have each process map the "shared" data structure to an Isolated Storage file, then use a system wide Mutex (named Mutex) to prevent one process from reading the file when the other is writing it. It'll be simpler if one process is always the writer of the data structure, so it never has to worry about merging in changes made by the other process asynchronously. If each process must be the writer of some portion of the data structure, the usual case, consider separating those portions into separate data structures and separate Isolated Storage files, with one process reading one file and writing the other and the other process writing the first and reading the second. (all reads and writes within mutex. Use same mutex for both files and both processes to avoid deadlocks.)
try this:
phoneApplicationPage.State