We have an "MC1362 Camera" and an "Inspecta-5" frame grabber in our lab. There is program in LABVIEW11 which gets the data from a frame grabber, however as the Labview is slow my supervisor has told me to write a program in c++ to get the data from the frame grabber. I have no idea how to write a c++ program to connect to a frame grabber and do the data acquisition. I know how to write software in c++, but have never tried programming to connect to hardware and read data from it. Is there any specific library or framework which can help me, or any tutorial?
Please, if anybody knows, help me in this matter.
Update:just to add, we are doing medical image analysis, and a laser illuminate a subject, so camera will take pictures and pass it to the computer. I need to grab the pictures and analysis them.
You basically have a couple of options,
1 see if there is an SDK for the grabber card, if there is this is usually easier then option 2 but is of course restricted to work with that grabber or familly of grabber cards, we do it this way with the eurysys grabber cards.
2 assuming you are running on a windows platform, implement a DirectShow filtergraph and write your own ouput filter to get the data, the SDK for DirectShow is quiet good and has many examples. This approach is far more flexible and you should be able to use a number of grabber but its also alot more complex, we do it this way for USB / some other inbuilt grabbers.
Our software is done in Delphi 7 but its just importing DLLs, for C++ should be no problem and most SDK's are written round C++ anyway.
I know its not much but its a place to start.
Update
Just done a quick Google search and there is an SDK for that Grabber and on first looks its seams fairly straight forward.
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I've been stuck on this problem for weeks now and Google is no help, so hopefully some here can help me.
I am programming a software sound mixer in C++, getting audio packets from the network and Windows microphones, mixing them together as PCM, and then sending them back out over the network and to speakers/USB headsets. This works. I have a working setup using the PortAudio library to handle the interface with Windows. However, my supervisors think the latency could be reduced between this software and our system, so in an attempt to lower latency (and better handle USB headset disconnects) I'm now rewriting the Windows interface layer to directly use WASAPI. I can eliminate some buffers and callbacks doing this, and theoretically use the super low latency interface if that's still not fast enough for the higher ups.
I have it only partially working now, and the partially part is what is killing me here. Our system has the speaker and headphones as three separate mono audio streams. The speaker is mono, and the headset is combined from two streams to be stereo. I'm outputting this to windows as two streams, one for a device of the user's choice for speaker, and one of another device of the user's choice for headset. For testing, they're both outputting to the default general stereo mix on my system.
I can hear the speaker perfectly fine, but I cannot hear the headset, no matter what I try. They both use the same code path, they both go through a WMF resampler to convert to 2 channel audio at the sample rate Windows wants. But I can hear the speaker, but never the headset stream.
It's not an exclusive mode problem: I'm using shared mode on all streams, and I've even specifically tried cutting down the streams to only the headset, in case one was stomping the other or something, and still the headset has no audio output.
It's not a mixer problem upstream, as I haven't modified any code from when it worked with PortAudio streams. I can see the audio passing through the mixer and to the output via my debug visualizers.
I can see the data going into the buffer I get from the system, when the system calls back to ask for audio. I should be hearing something, static even, but I'm getting nothing. (At one point, I bypassed the ring buffer entirely and put random numbers directly into the buffer in the callback and I still got no sound.)
What am I doing wrong here? It seems like Windows itself is the problem or something, but I don't have the expertise on Windows APIs to know what, and I'm apparently the most expert for this stuff in my company. I haven't even looked yet as to why the microphone input isn't working, and I've been stuck on this for weeks now. If anyone has any suggestions, it'd be much appreciated.
Check the re-sampled streams: output the stereo stream to the speaker, and output the mono stream to the handset.
Use IAudioClient::IsFormatSupported to check supported formats for the handset.
Verify your code using an mp3 file. Use two media players to play different files with different devices simultaneously.
I'm trying to develop a little application in which you can load a mp3 file and play it in variable speeds! (I know it already exists :-) )
I'm using Qt and C++. I already have the basic player but I'm stuck with the rate thing, because I want to change the rate smoothly (like in Mixxx) without stopping the playback! The QMediaPlayer always stops if I change the value and creates a gap in the sound. Also I don't want the pitch to change!
I already found something called "SoundTouch" but now I'm completely clueless what to do with it, how to process my mp3 data and how to get it to the player! The "SoundTouch" Library is capable of doing what I want, i got that from the samples on the homepage.
How do I have to import the mp3 file, so I can process it with the SoundTouch functions
How can I play the output from the SoundTouch function? (Perhaps QMediaPlayer can do the job?)
How is that stuff done live? I have to do some kind of stream I guess? So I can change the speed during play and keep on playing without gaps. Graphicaly in my head it has to be something that sits between the data and the player, where all data has to go through live, with a small buffer (20-50 ms or so) behind to avoid gaps during processing future data.
Any help appreciated! I'm also open to any another solution then "SoundTouch" as long as I can stay with Qt/C++!
(Second thing: I want to view a waveform overview aswell as moving part of it (around actual position of the song), so I could also use hints on how to get the waveform data)
Thanks in advance!
As of now (Qt 5.5) this is impossible to do with QMediaPlayer only. You need to do the following:
Decode the audio using GStreamer, FFMpeg or (new) QAudioDecoder: http://doc.qt.io/qt-5/qaudiodecoder.html - this will give you raw PCM stream;
Apply SoundTouch or some other library to this raw data to change the pitch. If GPL is ok, take a look at http://nsound.sourceforge.net/examples/index.html, if you develop proprietary stuff, STK might be a better choice: https://ccrma.stanford.edu/software/stk/
Output the modified data into audio device by using QAudioOutput.
This strategy uses Qt as much as possible, and brings you the best platform coverage (you still lose Android though as it does not support QAudioOutput)
I just want to know if its possible to write a program in c++ on Windows which can grab frames of video from a web cam by making use of a USB library like libusb.
I have looked at http://www.dreamincode.net/forums/topic/148707-introduction-to-using-libusb-10/ . They define a basic sequence of the steps one must follow while using libusb to work with usb devices. To summarize:
1) Find and connect to the USB device
2) Send and receive data
3) Release the USB device
libusb handles part 1) and 3). So, the question comes down to part 2). I wonder if it is possible/feasible to send commands to the camera and receive data via USB by adhering to some type of standard (UVC 1.5 for example). I'm thinking that not knowing some proprietary, device specific information might be a problem though.
If this is possible but super hard, I would appreciate any insights into what the biggest challenge would be.
The hardware I have in mind are low-end logitech USB cameras.
Also, I am fully aware that OpenCV, Video for Windows (VFW), DirectShow, and FFMPEG can do web-cam video capture. I am interested in this project as a learning experience.
Thanks in advance!
Since you're obviously a bit inexperienced, I'll give an answer in high-level terms.
To talk to someone, or something, you need to speak a common language, or have a translator. Now, there are many different USB devices, and they speak many different languages, but OpenCV can act as a translator for many of those. If you don't want to use that, you will need to find another translator, or possibly learn to speak the camera's native language.
The first option (another translator) is no option for you, because you don't want a translator.
The second option is something we can't help you with, because we don't know which camera you have.
I've stumbled through some code to enumerate my microphone devices (with some help), and am able to grab the "friendly name" and "clsid" information from each device.
I've done some tinkering with GraphEd.exe to try and figure out how I can take audio from directshow and write it to a file (I'm not currently concerned about the format, wav should be fine), and can't seem to find the right combination.
One of the articles I've read linked to this Windows SDK sample, but when I examined the code, I ended up getting pretty confused at how to use that code, ie. setting the output file, or specifying which audio capture device to use.
I also came across a codeguru article that has a nicely featured audio recorder, but it does not have an interface for selecting the audio device, and I can't seem to find where it statically picks which recording device to use.
I think I'd be most interested in figuring out how to use the Windows SDK sample, but any explanation on either of the two approaches would be fantastic.
Edit: I should mention my knowledge and ability as a win32 COM programmer is very low on the scale, so if this is easy, just explain it to me like I'm five, please.
Recording audio into file with DirectShow needs you to build the right filter graph, as you should have figured out already. The parts include:
The device itself, which you instantiate via moniker (not CLSID!), it is typically PCM format
Multiplexer component that converts streams into container format
File Writer Filter that takes file-compatible stream and writes into a file
The tricky moment is #2 since there is not standard component available. Windows SDK samples however contains the missing part - WavDest Filter Sample. Building it and making it ready for use, you can build a graph that records from device into .WAV file.
Your graph will look like this, and it's built easily programmatically as well:
I noticed that I have a variation of WavDest installed with Google Earth - for the case you have troubles building it yourself and you will be looking for prebuilt binary.
You can instruct ffmpeg to record from a directshow device, and output to a file.
I'm developing a screencasting utility in C++.
It basically captures desktop frames and creates an AVI file. The algorithm is as follows:
Create a thread: this->m_hThread=CreateThread(NULL,0,thScreenCapture,this,0,NULL);
Capture desktop in thScreenCapture n times per second (like 5 fps).
obj->Capture();
In Capture(), append the bitmap data to the avi file.
this->appendBitmapToAvi(this->avifile, bmp);
This utility also records sound. So, in method thScreenCapture, sound data is also being appended to the avi file.
The problem is that a lag occurs between frames and sound when more than 6 frames (this can change depending on hardware configuration) are captured per second.
I'm seeking for a solution to optimize the algorithm. A solution may be buffering frames in memory, not appending all of them to the avi file on-the-fly. But this makes the code more complex because I have to deal with the sound data that's being captured in a different thread.
What do you suggest to increase the fps value that this utility supports without losing synchronization?
Are you writing the AVI file yourself? A noble effort, but there are APIs to help with this task.
If you're working on the windows platform, I'd suggest considering using the DirectShow or Media Foundation APIs to mux the audio and video to disk. DirectShow is the API for A/V capture, streaming and muxing on the windows platform.
This article on CodeProject talks about audio & video sync issues and the mechanism that DirectShow uses to overcome this difficulty.
Essentially, a reference clock is employed and the frames are timestamped.
There's a very active DirectShow community that is an extremely useful resource for new folks. TMH's website is well worth checking out - he's an MS MVP and is an active member of the community.
I hope this helps!
You could take a look at the source for other screencasting software, such as CamStudio, to see how they are doing it.
If your program is disk bound (and I suspect it is), then things might improve by using compression (this is how the big name programs, such as Camtasia Studio, operate)
Use a circular double or triple buffer to store the bitmap and sound each frame and use a separate thread to add the bitmap and sound to the avi. So data-collection is in one thread, data is in a circular (thread-safe) buffer and the data-storage is in another thread.
What OS are you targeting? If tyou are working on Windows XP I would take a look at some of the DirectShow code at http://tmhare.mvps.org/downloads.htm, specifically Filter Graph Library.