I am using visual studio.
I need to write a c++ dll, that plays video (h265 codec) in the background. The video shouldn't been seen, only the sound should be heard. I need to catch every frame and gives it to the host application.
I couldn't find a simple API that let me access the current frame of a player.
Any help would be appreciated
This question is very broad, but, there are three choices:
libVLC - easy to use and implement, but you don't have full access (but for your requirements, it should be enough)
ffmpeg - you can spawn the ffmpeg command line tool and just get play audio output. This should be easy. If you want more control, you can use ffmpeg as library, what is kinda hard to use and implement, but you have full control over process of decoding and playback
gstreamer - same as ffmpeg
Related
In my fun project, I'm downloading video file from youtube, and writing to a file on local disk. Simultaneously I want to play it. The objective is to cache the file on local disk, so that when I want to see the video again, the app can play it locally, thereby saving bandwidth.
I'm using Python 3.3.1, PyQt4/Phonon and LibVLC. So far, I'm able to do the following things:
Given a youtube watch url, I can download the video file and then play it using both PyQt4/Phonon and LibVLC, independently. It is not streaming.
Since LibVLC supports streaming, I'm able to play the given url through streaming.
The second is very close to what I want to do, but since it doesn't save the file on disk, next time I cannot play the same video locally.
I'm looking for some guidelines as to how to proceed from here. In particular, how to play a video from an incomplete file which is still being written into.
I'm completely fine with any API (that does the job) as long as it is:
Python 3.3.1 (preferably)
C
C++.
And I'm looking for alternative approaches also, if my current approach is not correct or makes the problem more difficult than it actually is.
VLC supports playback of incomplete files, so if you're up for a bit of non-blocking I/O and/or parallel code, you should be able to start the download and after a sufficient amount has been written, use LibVLC to start playback. Depending on what compression algorithm is used, you may need to buffer enough so that there's always several seconds of data left in the buffer -- if I recall correctly, some of the more modern algorithms record deltas and index information going forward and backward.
You may get a few warnings / error messages / Exceptions, but I would not assume that they're fatal -- let the playback quality be your guide!
This is somewhat similar to some of the suggestions from the comments above, and is also related to a lot of what #abarnert said, to a lesser extent some of the exchange with #StackedCrooked.
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 need to add webcam video capture to a legacy MFC C++ application. The video needs to be saved as MP4. Did a bit of googling but didn't come across anything that looked promising. Any suggestions on the best approach?
EDIT:
Windows platform.
EDIT:
Must be compatible with XP
There are a few popular options to choose from:
DirectShow API - it does not have stock MPEG-4 compressors for video and audio, neither it has a stock multiplexor for .MP4 format, though there is an excellent free multiplexor from GDCL: http://www.gdcl.co.uk/mpeg4/. Also there is decent documentation, a lot of samples
Media Foundation API - it has everything you need (codecs, multiplexor) but only in Windows 7 (maybe even not all edtions)
FFmpeg and libavcodec/libavformat are definitely relevant, however H.264 encoder is only available under GPL license, not sure about video capture part right there, and you might have hard time looking for documentation and samples.
I'd say look at OpenCV as a library, hook into their video capture for that aspect, it can write out to mp4 but you'll need a couple of other libs for handling the output stream (on Linux I'd say ffmpeg and x264), that should get the buffer into the container with a reasonable amount of hassle.
I'm trying to get mpg123 audio decoder to work with QT on windows. How do i play the decoded audio data at the right speed with Qmultimedia module in push mode. Currently i'm using simple timer to get it to play audio but it's not very efficient way to do it, if I do anything else at the same time audio get all distorted. Is there any better way to send the decoded data to audio output? It would be nice if anyone could point me to any nice examples using Qmultimedia module and Qaudiooutput class. I've tried to figure out QT example project "audiooutput" but it seems that it's also using timer to send audio to output in push mode.. Hope that I'm not too confusing.
I also had to figure that out and I would also suggest using the Phonon framework to do this.
It uses Windows Media Player as host on Windows, QuickTime on Mac and some KDE stuff on Linux.
So it's pretty platform independent.
If you need more low-level functionality, you should take a look into an open-source project called portaudio. It's very easy to use and you can manipulate or even fill buffers from code.
I used it to build an oscillator.
Hope that helps!
Best,
guitarflow
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.