How to read video file using v4l2 - linux

I want to read a video file using v4l2, say an AVI file. And read it frame by frame.
As far as I can tell I need to use the read() function. But how isn't very clear to me. There are also hardly any examples available. So maybe a simple example on how to do this would help.

This is not what the Video4Linux2 (V4L2) API is for. It is not designed for reading multimedia files from disk, decoding them and playing them. Rather, it is designed to interface to assorted multimedia input devices (like webcams, microphones, TV tuners, and video capture devices), capture A/V data, and play it.
Take it from the V4L2 API introduction:
Video For Linux Two is [...] a kernel interface for analog radio and
video capture and output drivers.
For reading an AVI file and decoding/playing it (programmatically) on Linux, look into FFmpeg or GStreamer.

Related

Linux: how to dump audio output PCM bit stream like tcpdump

I am trying to do some audio debugging on my Linux system.
I learned how to record the sound of the current playing media but how can I get the PCM data without DAC/ADC?
I mean, just like wireshark or tcpdump tool, is there some sort of alsadump that I can make use of?
I want to do bit-exact comparison of the output PCM data to make sure the audio processing algorithm (which is an executable binary) worked correctly.
Thanks a lot.

Adding audio effects (reverb etc..) to a BackgroundAudioPlayer driven streaming audio app

I have a windows phone 8 app which plays audio streams from a remote location or local files using the BackgroundAudioPlayer. I now want to be able to add audio effects, for example, reverb or echo, etc...
Please could you advise me on how to do this? I haven't been able to find a way of hooking extra audio processing code into the pipeline of audio processing even through I've read much about WASAPI, XAudio2 and looked at many code examples.
Note that the app is written in C# but, from my previous experience with writing audio processing code, I know that I should be writing the audio code in native C++. Roughly speaking, I need to find a point at which there is an audio buffer containing raw PCM data which I can use as an input for my audio processing code which will then write either back to the same buffer or to another buffer which is read by the next stage of audio processing. There need to be ways of synchronizing what happens in my code with the rest of the phone's audio processing mechanisms and, of course, the process needs to be very fast so as not to cause audio glitches. Or something like that; I'm used to how VST works, not how such things might work in the Windows Phone world.
Looking forward to seeing what you suggest...
Kind regards,
Matt Daley
I need to find a point at which there is an audio buffer containing
raw PCM data
AFAIK there's no such point. This MSDN page hints that audio/video decoding is performed not by the OS, but by the Qualcomm chip itself.
You can use something like Mp3Sharp for decoding. This way the mp3 will be decoded on the CPU by your managed code, you can interfere / process however you like, then feed the PCM into the media stream source. Main downside - battery life: the hardware-provided codecs should be much more power-efficient.

Audio Playback control in C++

I'm working on a project that requires me to sync an audio playback(preferably an mp3 file) with my program.
My program reads a motion file from a txt file and output's it onto the serial port at a particular rate. At the same time an audio file has to be played back on the speaker. This audio file has to be in sync with the data..that is to say after say transmittin 100 bytes of data, the audio mustve played back to a predefined time.
What would be the tools used to play and control audio like this?
a tutorial would be great!
Thanks!!
In general, when working with audio, you want to synchronize other sources to audio. This is for several reasons, but most important is that audio runs on a clock running on its own hardware. You'll have to get timing information from that clock. There is a guide here written for using portaudio, but the principles apply to other situations:
http://www.portaudio.com/docs/portaudio_sync_acmc2003.pdf

making USB video in Linux

I am working on an embedded device using Linux that will read video, process and modify every frame and then return USB video stream. I don't know how to make USB video from a sequence of frames. Can someone direct me where to start?
Take a look at http://electron.mit.edu/~gsteele/ffmpeg/
It shows you how to make video from a sequence of images using ffmpeg and mencoder
Yes, take a look at OpenCV.
There are lots of code around here to show you how to use the library. For instance, take a look at: OpenCV: process every frame

Can v4l2 be used to read audio and video from the same device?

I have a capture card that captures SDI video with embedded audio. I have source code for a Linux driver, which I am trying to enhance to add video4linux2 support. My changes are based on the vivi example.
The problem I've come up against is that all the example I can find deal with only video or only audio. Even on the client side, everything seems to assume v4l is just video, like ffmpeg's libavdevice.
Do I need to have my driver create two separate devices, a v4l2 device and an alsa device? It seems like this makes the job of keeping audio and video in sync much more difficult.
I would prefer some way for each buffer passed between the driver and the app (through v4l2's mmap interface) contain a frame, plus some audio that matches up (with respect to time) with that frame.
Or perhaps have each buffer contain a flag indicating if it is a video frame, or a chunk of audio. Then the time stamps on the buffers could be used to sync things up.
But I don't see a way to do this with the V4L2 API spec, nor do I see any examples of v4l2-enabled apps (gstreamer, ffmpeg, transcode, etc) reading both audio and video from a single device.
Generally, the audio capture part of a device shows up as a separate device. It's usually a different physical device (posibly sharing a card), which makes sense. I'm not sure how much help that is, but it's how all of the software I'm familiar with works...
There are some spare or reserved fields in the v4l2 buffers that can be used to pass audio or other data from the driver to the calling application via pointers to mmaped buffers.
I modified the BT8x8 driver to use this approach to pass data from an A/D card synchronized to the video on Ubuntu 6.06.
It worked OK, but the effort of maintaining my modified driver caused me to abandon this approach.
If you are still interested I could dig out the details.
IF you want your driver to play with gstreamer etc. a separate audio device generally is what is expected.
Most of the cheap v4l2 capture card's audio is only an analog pass through with a volume control requiring a jumper to capture the audio via the sound card's line input.

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