I'm trying to save a series of images (16 bit grayscale pgm) as video. The video has to be compressed. My program has to be independent of the codecs installed in the system.
My initial idea was to use OpenCV for this, unfortunately it depends on codecs installed in the system (unless I'm missing something).
I feel like there should be a way to compile an encoder (H264 or similar would be perfect) into the program or redistribute it as a dll with my program. I just can't find any good up to date guidance/examples.
I've been swimming in the deep vast ocean of AV encoding for a couple of days and would really appreciate it if someone could point me to a right direction.
Thanks.
As Ben suggests, it would be a good idea to use an established library in your code.
FFMPEG is probably the most used at the moment - it can be used on the command line, with a 'wrapper' program or the libraries it is built with can be used directly.
I think the last case sounds like the one you want - you can find documentation here:
https://trac.ffmpeg.org/wiki/Using%20libav*
Note the comment about disambiguation at the start - this is important to understand as the project lib and the library (which is what you want) are different things.
and there is some notes in this answer on how to build it into a program:
FFMpeg sample program
Related
I am currently in the process of learn c++ and I have decided to make a simple program that takes a .png file and exports the link inside it. I have looked it up and discovered ZXing but I still can't figure out how to implement it easily.
The ones I've found consists of scanning using webcam etc instead of a very simple version.
First, you need a library to read the images. CImg is good, it is a header only library build on top of the base libraries for the different images format: https://cimg.eu/
Assuming the image is a clean QR code you can skip the complex computer vision recognition part.
For the decoding part I am not sure, I don't know ZXing but it looks like the c++ port is no longer maintained anyway. Still, once you have the clean image it should not be a problem to use it.
You might want to have a look at the most recent c++ port of ZXing: https://github.com/nu-book/zxing-cpp
The project includes a trivial example application that does exactly what you want.
So I'm trying to get a basic tool to output video/audio(s) to Twitch. I'm new to this side (AV) of programming so I'm not even sure what to look for. I'm trying to use mainly Windows infrastructure and third party where not available.
What are the steps of getting raw bitmap and wave data into a codec and then into a rtsp client and finally showing up on Twitch? I'm not looking for code. I'm looking for concepts so I can search for as I'm not absolutely sure what to search for. I'd rather not go through OBS source code to figure it out and use that as last resort.
So I capture the monitor via Output Duplication and also the Sound on the system as a wave and the microphone as another wave. I'm trying to push this to Twitch. I know that there's Media Foundation on Windows but I don't know how far to streaming it can get as I assume there no netcode integrated in it? And also the libav* collection in FFMPEG.
What are the basic steps of sending bitmap/wave to Twitch via any of thee above libraries or even others as long as they work on Windows. Please don't add code, I just need a not very long conceptual explanation and I'll take it from there. Try to cover also how bitrate and framerate gets regulated (do I have do it or the codec does it)?
Assume absolute noob level in this area (concept-wise not code-wise).
I've been wanting to play around with audio parsing for a while now but I haven't really been able to find the correct library for what I want to do.
I basically just want to parse through a sound file and get amplitudes/frequencies and other relevant information at certain times during the song (like every 10 ms or so) so I can graph the data for example where the song speeds up a lot and where it gets really loud.
I've looked at OpenAL quite a bit but it doesn't look like it provides this ability, other than that I have not had much luck with finding out where to start. If anyone has done this or used a library which can do this a point in the right direction would be greatly appreciated. Thanks!
For parsing and decoding audio files I had good results with libsndfile, which runs on Windows/OSX/Linux and is open source (LGPL license). This library does not support mp3 (the author wants to avoid licensing issues), but it does support FLAC and Ogg/Vorbis.
If working with closed source libraries is not a problem for you, then an interesting option could be the Quicktime SDK from Apple. This SDK is available for OSX and Windows and is free for registered developers (you can register as an Apple developer for free as well). With the QT SDK you can parse all the file formats that the Quicktime Player supports, and that includes .mp3. The SDK gives you access to all the codecs installed by QuickTime, so you can read .mp3 files and have them decoded to PCM on the fly. Note that to use this SDK you have to have the free QuickTime Player installed.
As far as signal processing libraries I honestly can't recommend any, as I have written my own functions (for speech recognition, in case you are curious). There are a few open source projects that seem interesting listed in this page.
I recommend that you start simple, for example working on analyzing amplitude data, which is readily available from the PCM samples without having to do any processing. Being able to visualize the data is very useful, I have found Audacity to be an excellent visualization tool, and since it is open source you can build your own tests inside it.
Good luck!
I am working on a C++ project with openCV. It is a simple web cam application with basic features like capturing pictures and videos. I have already been able to save video (w/o audio). Since openCV doesnot support audio processing, I was wondering if there is any way I can record audio separately in a different file and later combine those together to get one video file.
While searching on the internet, I did hear something about using ffmpeg with openCV. But I just cant figure out how to do it exactly.....
Can you guys help me? I would be very grateful... Thankyou!
P.S. I have used openCV and QT (for GUI)
As you said, opencv doesn't by itself deal with audio. However once you get a separate audio and video file, you can combine them using a technique called muxing. There are many many ways to do this. I use VirtualDub for most of my muxing needs, although it is windows only (not sure if that's a problem). I know ffmpeg is also capable of muxing (via the command line interface), I can't recall what the command is. There's also mplayer and a multitude of other programs out there to do this.
as far as i know openCV is good for video/image processing. To support audio processing, you can use other libraries e.g. PortAudio or C-sound.
I've always been curious about audio conversion software, but I have never seen a proper explanation from a beginners point of view as to how to write a simple program that converts for example, a mp3 file to a wav. I'm not asking about any of the complex algorithms involved, just a small example using a simple library. Searching on SO, I came up with several names including:
Lame
The Synthesis Toolkit
OpenAL
DirectSound
But I'm unable to find a straightforward example of any of these libraries. Usually I don't mind wading through tons of code, but here I have absolutely no knowledge about the subject and so I always feel like I'm shooting in the dark.
Anyone here have a simple example / tutorial on converting a sound file using any of these libraries? My question is specifically directed towards C/C++ because those are the two languages I'm currently learning and so I'd like to continue to focus on them.
Edit: One thing I forgot to mention: I'm on a *NIX system.
Thanks everyone for the responses! I sort of cobbled them together to successfully make a small utility that converts a AIFF/WAV/etc file to an mp3 file. There seems to be some interest in this question, so here it what I did, step by step:
Step 1:
Download and install the libsndfile library as suggested by James Morris. This library is very easy to use – its only shortcoming is it won't work with mp3 files.
Step 2:
Look inside the 'examples' folder that comes with libsndfile and find generate.c. This gives a nice working example of converting any non-mp3 file to various file formats. It also gives a glimpse of the power behind libsndfile.
Step 3:
Borrowing code from generate.c, I created a c file that just converts an audio file to a .wav file. Here is my code: http://pastie.org/719546
Step 4:
Download and install the LAME encoder. This will install both the libmp3lame library and the lame command-line utility.
Step 5:
Now you can peruse LAME's API or just fork & exec a process to lame to convert your wav file to an mp3 file.
Step 6: Bring out the champagne and caviar!
If there is a better way (I'm sure there is) to do this, please let me know. I personally have never seen a step-by-step roadmap like this so I thought I'd put it out there.
For converting between various formats (except MP3) check libsndfile http://mega-nerd.com/libsndfile/
Libsndfile is a library designed to
allow the reading and writing of many
different sampled sound file formats
(such as MS Windows WAV and the
Apple/SGI AIFF format) through one
standard library interface.
During read and write operations,
formats are seamlessly converted
between the format the application
program has requested or supplied and
the file's data format. The
application programmer can remain
blissfully unaware of issues such as
file endian-ness and data format
It is also simple to use, with the API following the style of the Standard C library function names:
http://mega-nerd.com/libsndfile/api.html
And examples are included in the source distribution.
For actual audio output, another library will be needed, SDL as already mentioned might be a good place to start. While SDL can also read/write audio files, libsndfile is far superior.
If your curious about DSP and computers, take a look at the Synthesis Toolkit. It's sweet. It's designed for learning. The examples and tutorials they have on their website are straightforward and thorough. Keep in mind, the guys who wrote it, wrote it so they could create acoustic models of real instruments. As a result, they've included some instruments that are just plain wacky, but fun. It will give you a core understanding of processing PCM sound. And you'll probably be able to hack together some fun little noisemakers while your at it.
https://ccrma.stanford.edu/software/stk/
Check libmad http://mad.sourceforge.net " "M"peg "A"udio "D"ecoder library", should provide a good example.
Also for an easy cross-platform audio handling, check SDL http://www.libsdl.org/.
Hope that helps.