I'm writing a DirectShow source filter which is registered as a CLSID_VideoInputDeviceCategory, so it can be seen as a Video Capture Device (from Skype, for example, it is viewed as another WebCam).
My source filter is based on the VCam example from here, and, for now, the filter produces the exact output as this example (random colored pixels with one Video output pin, no audio yet), all implemented in the FillBuffer() method of the one and only output pin.
Now the real scenario will be a bit more tricky - The filter uses a file handle to a hardware device, opened using the CreateFile() API call (opening the device is out of my control, and is done by a 3Party library). It should then read chunks of data from this handle (usually 256-512 bytes chunk sizes).
The device is a WinUSB device and the 3Party framework just "gives" me an opened file handle to read chunks from.
The data read by the filter is a *.mp4 file, which is streamed from the device to the "handle".
This scenario is equivalent to a source filter reading from a *.mp4 file on the disk (in "chunks") and pushing its data to the DirectShow graph, but without the ability to read the file entirely from start to end, so the file size is unknown (Correct?).
I'm pretty new to DirectShow and I feel as though I'm missing some basic concepts. I'll be happy if anyone can direct me to solutions\resources\explanations for the following questions:
1) From various sources on the web and Microsoft SDK (v7.1) samples, I understood that for an application (such as Skype) to build a correct & valid DirectShow graph (so it will render the Video & Audio successfully), the source filter pin (inherits from CSourceStream) should implement the method "GetMediaType". Depending on the returned value from this implemented function, an application will be able to build the correct graph to render the data, thus, build the correct order of filters. If this is correct - How would I implement it in my case so that the graph will be built to render *.mp4 input in chunks (we can assume constant chunk sizes)?
2) I've noticed the the FillBuffer() method is supposed to call SetTime() for the IMediaSample object it gets (and fills). I'm reading raw *.mp4 data from the device. Will I have to parse the data and extract the frames & time values from the stream? If yes - an example would b great.
3) Will I have to split the data received from the file handle (the "chunks") to Video & Audio, or can the data be pushed to the graph without the need to manipulate it in the source filter? If split is needed - How can it be done (the data is not continuous, and is spitted to chunks) and will this affect the desired implementation of "GetMediaType"?
Please feel free to correct me if I'm using incorrect terminology.
Thanks :-)
This is a good question. On the one hand this is doable, but there is some specific involved.
First of all, your filter registered under CLSID_VideoInputDeviceCategory category is expected to behave as a live video source. By doing so you make it discoverable by applications (such as Skype as you mentioned), and those applications will be attempting to configure video resolution, they expect video to go at real time rate, some applications (such as Skype) are not expecting compressed video such H.264 there or would just reject such device. You can neither attach audio right to this filter as applications would not even look for audio there (not sure if you have audio on your filter, but you mentioned .MP4 file so audio might be there).
On your questions:
1 - You would have a better picture of application requirement by checking what interface methods applications call on your filter. Most of the methods are implemented by BaseClasses and convert the calls into internal methods such as GetMediaType. Yes you need to implement it, and by doing so you will - among other - enable your filter to connect with downstream filter pins by trying specific media types you support.
Again, those cannot me MP4 chunks, even if such approach can work in other DirectShow graphs. Implementing a video capture device you should be delivering exactly video frames, preferably decompressed (well those could be compressed too, but you are going to immediately have compatibility issies with applications).
A solution you might be thinking of is to embed a fully featured graph internally to which you inject your MP4 chunks, then the pipelines parse those, decodes and delivers to your custom renderer, taking frames on which you re-expose them off your virtual device. This might be a good design, though assumes certain understanding of how filters work internally.
2 - Your device is typically treated as/expected to be a live source, which means that you deliver video in realtime and frames are not necessarily time stamped. So you can put times there and yes you definitely need to extract time stamps from your original media (or have it done by internal graph as mentioned in item 1 above), however be prepared that applications strip time stamps especially for preview purposes, since the source is "live".
3 - Getting back to audio, you cannot implement audio on the same virtual device. Well you can, and this filter might be even working in a custom built graph, but this is not going to work with applications. They will be looking for separate audio device, and if you implement such, they will instantiate it separately. So you are expected to implement both virtual video and virtual audio source, and implement internal synchronization behind the scenes. This is where timestamps will be important, by providing them correctly you will keep lip sync in live session to what it was originally on the media file you are streaming from.
Related
I would like to make an app (Target pc windows) that let you modify the micro input in real time, like introducing sound effects or even modulating your voice.
I searched over the internet and only found people telling that it would not be possible without using a virtual audio cable.
However I know some apps with similar behavior (voicemod, resonance) not using a virtual audio cable so I would like some help about how can be done (just the name of a library capable would be enough) or where to start.
Firstly, you can use professional ready-made software for that - Digital audio workstation (DAW) in combination with a huge number of plugins for that.
See 5 steps to real-time process your instrument in the DAW.
And What is (audio) direct monitoring?
If you are sure you have to write your own, you can use libraries for real-time audio processing (as far as I know, C++ is better for this than C#).
These libraries really works. They are specially designed for realtime.
https://github.com/thestk/rtaudio
http://www.portaudio.com/
See also https://en.wikipedia.org/wiki/Csound
If you don't have a professional sound interface yet, but want to minimize a latency, read about Asio4All
The linked tutorial worked for me. In it, a sound is recorded and saved to a .wav.
The key to having this stream to a speaker would be opening a SourceDataLine and outputting to that instead of writing to a wav file. So, instead of outputting on line 59 to AudioSystem.write, output to a SourceDataLine write method.
IDK if there will be a feedback issue. Probably good to output to headphones and not your speakers!
To add an effect, the AudioInputLine has to be accessed and processed in segments. In each segment the following needs to happen:
obtain the byte array from the AudioInputLine
convert the audio bytes to PCM
apply your audio effect to the PCM (if the effect is a volume change over time, this could be done by progressively altering a volume factor between 0 to 1, multiplying the factor against the PCM)
convert back to audio bytes
write to the SourceDataLine
All these steps have been covered in StackOverflow posts.
The link tutorial does some simplification in how file locations, threads, and the stopping and starting are handled. But most importantly, it shows a working, live audio line from the microphone.
I'm trying to do composition with two separate video sources in Media Foundation. I am attempting to encode a video with a video overlay. To do so I am attempting to use the Video Resizer on the smaller input.
I've seen several threads on this, but I thought I'd ask around in any case.
Basically the idea is to create two source readers and a sink writer. The source files are h264, so I use the reader to decode into YUY2. While processing samples, I send the appropriate sample to the Resize MFT, then down the line (I haven't made it this far) I combine the two images to create the overlay effect with MFCopyImage.
My question is: I am getting an E_INVALIDARG when I call ProcessInput on the Resize MFT.
To initialize the mft, I am giving it the appropriate type from the reader via SetInput Type. After that I am setting all the appropriate properties via the PropertyStore, and then updating the framesize for the output type of the MFT. I have read the documentation and modeled my implementation according to the MFT Processing Model.
None of these steps raise any red flags until I actually attempt to use ProcessInput.
Although I have limited experience in Windows Media Foundation, I have been able to use the Framerate DSP with success. I would appreciate any advice.
Thank you!
For anyone else stuck in a similar situation, I ended up not using the Resizer MFT but the Video Processor MFT which worked with much less effort.
I'm searching for a way to analyse the content of internet radios. I want to write a ruby client that can get the current track, next track, band, bpm and other meta information from a stream (e.g. a radio on shoutcast).
Does anybody know how to do this? And how do I record that stream into a mp3 or aac file?
Maybe there is a library that can already do this, I haven't one so far.
regards
I'll answer both of your questions.
Metadata
What you are seeking isn't entirely possible. Information on the next track is not available (keep in mind not all stations are just playing songs from a playlist... many offer live content). Advanced metadata such as BPM is not available. All you get is something like this:
Some Band - Some Song
The format of {artist} - {song title} isn't always followed either.
With those caveats, you can get that metadata from a stream by connecting to the stream URL and requesting the metadata with the following request header:
Icy-MetaData: 1
That tells the server to send the metadata, which is interleaved into the stream. Every 8KB or so (specified by the server in a response header), you'll find a chunk of metadata to parse. I have written up a detailed answer on how to parse that here: Pulling Track Info From an Audio Stream Using PHP The prior question was language-specific, but you will find that my answer can be easily implemented in any language.
Saving Streams to Disk
Audio playing software is generally very resilient to errors. SHOUTcast servers are built on this principal, and are not knowledgeable about the data going through them. They just receive data from an encoder, and when the client requests the stream, they start sending that data at an arbitrary point.
You can use this to your advantage when saving stream data. It is possible to simply write the stream data as it comes in to a file. Most audio players will play them without problem. I have tested this with MP3 and AAC.
If you want a more conformant file, you will have to use a library or parse the stream yourself to split on the appropriate frames, and then handle bit reservoir issues in your code. This is a lot of work, and generally isn't worth doing unless you find your files have real compatibility problems.
I'm working on a media player using Media foundation. I want to support VOB files playback. However, media foundation currently does not support the VOB container. Therefore I wish to use DirectShow for the same.
My idea here is not to take an alternate path using a DirectsShow graph, but just grab a video frame and pass it to the same pipeline in media foundation. In media foundation, I have an 'IMFSourceReader' which simply reads frames from the video file. Is there a direct show equivalent, which just gives me the frames without needing to create a graph, start playback cycle, and then trying to extract frames from the renders pin? (To be more clear, does DirectsShow support an architecture wherein it could give me raw frames without actually having to play the video?)
I've read about ISampleGrabber but its deprecated and I think it won't fit my architecture. I've not worked on DirectShow before.
Thanks,
Mots
You have to build a graph and accept frames from the respective parser/demultiplexer filter which will read container and deliver individual frames on its output.
The playback does not have to be realtime, nor you need to fake painting those video frames somewhere. Once you get the data you need in Sample Grabber filter, or a customer filter, you can terminate pipeline with a Null Renderer. That is, you can arrange getting frames you need in a more or less convenient way.
You can use Monogram frame grabber filter to connect the VOB DS filter's output - it works great. See the comments there for how to connect the output to external application.
I ran a test today with a DirectShow graph I assembled that had a Capture Filter assigned to my VOIP phone at the top of the graph. The app takes the audio from the capture filter and writes a WAV file, as part of the filter graph's operations. Out of curiosity I ran two copies of the program, fully expecting one of them to throw an error complaining that the capture device was "in use". Much to my surprise both copies of the program worked fine and each created its own WAV file of the recorded audio. The audio in both files was smooth and without problem and twins of each other in regards to the contained audio data.
Can I count on all DirectShow capture filters to exhibit the ability to be shared between multiple filter graphs? Or is every device/driver different?
If the filter instances don't share internally any exclusive access resources (such as hardware, specific TCP ports etc), you are free to duplicate them within a process, or in multiple processes. There are no implications as for specific filter to be only active in a single instance throughout the system.
Important example include:
USB video capture: a video capture device is normally intended to be used by one application only, so as soon as it is active it is locked no other application or filter instance can capture from it
Audio playback: popular user mode API for audio is a layer on top of actual playback implementation, internally a driver mixes audio from mutliple audio-enabled applications; so when you play audio, there is no exclusive lock involved because actualy device is shared between applications and there is code running around which combines audio from the applications transparently.