I’m in search for a solution that makes it possible to read the Timing Reference Signal (TRS) and Ancillary data (HANC and VANC) of a serial digital video. The TRS gives information about the start and the end of active video (SAV/EAV), the Ancillary data gives, for example, information about embedded audio. With this I want to code an application that analyzes the data that is transported in the non-picture area of serial video.
I read much about GStreamer and found with GstVideo Ancillary a collection that makes it possible to handle the Ancillary Data of a video.
Unfortunately, it’s not clear to me, how this collection works. For me, it looks like that this collection can only construct Ancillary data for a video and it’s not possible to read ancillary data from a detected videostream.
Another idea is to read the whole video stream and display, in a first step, the data words of the stream. TRS and ANC packets have to start with a special sequence of identifiers that makes it possible to localize them. Is GStreamer for this the right choice? Are there better recommended libraries for this task?
Related
I have an application that gets video samples from a frame grabber card via DirectShow. The application then does some processing and sends the video signal over a network. I now want to duplicate this video signal such that another DirectShow-enabled software (like Skype) can use the original input signal, too.
I know that you can create Tee filters in DirectShow like the one used to split a video signal for recording and preview. However, as I understand, this filter is only useful within a single graph, ie I cannot use it to forward the video from my process to eg Skype.
I also know that I could write my own video source, but this would run in the process of the consuming application. The problem is that I cannot put the logic of my original application in such a video source filter.
The only solution I could think of is my application writing the frames to a shared memory block and a video source filter reading it from there. Synchronisation would be done using a shared mutex or so. Could that work? I specifically do not like the synchronisation part?
And more importantly, is there a better solution to solve this problem?
The APIs work as you identified: a video capture application, such as Skype, is requesting video stream without interprocess communication in mind, there is no IPC involved to consume output generated in another process. Your challenge here is to provide this IPC yourself so that one application is generating the data, and then another extends existing API (virtual video source device) and picks existing data, then delivers as generated.
With video, you have a relatively big stream of data and you are interested in avoiding its excessive copying. File mappings (AKA shared memory) are the right thing to do: you put bytes in one process and they are immediately visible in another. You can synchronize access to the data using names events and mutexes which both processes use collaboratively - to signal availability of new buffer of data, as indication that used buffer is no longer in use etc.
I want to be able to break a song into packets and have access to these individual packets.
The reason for that is that I want to send each individual packet over the network using an experimental network protocol called Named Data Network.
As the packets arrive at the destination I want to play them. So I want to implement a streaming functionality. The only difference is the network layer that I will use. This network layer is not based on IP.
Does anyone know any C/C++ implementation of breaking a song file into pieces and then playing these packets individually? I looked over Gstreamer, but it seems complicated to get individual packets from its pipeline structure.
I found this reference which was the closest to what I wanted, however it was not so clear for me: how can I parse audio raw data recorder with gstreamer?
Summarizing the points I need:
Break a song into packets
Play the audio content of a single packet (or a small set of packets).
Thank you very much for the help!
An MP3 file is just a succession of MP3 frames. Each frame is made of a header and a data block.
Splitting the MP3 file as MP3 frames will involve parsing the MP3 file. You can refer to this documentation for a good description of the format.
Note that in the case of mpeg layer 3 codec, frames are not independant. In the worst case, 9 input frames may be needed before beeing able to decode one single frame.
What I would do instead of this
I guess you could probably ignore most of these details and focus on the streaming problem itself. Here is what I would try to build first:
on the sender side, split a file into packets, and send them one by one using your system. Command example: send_stream test.mp3
on the receiver side, receive the packets and rebuild the original file. Command example: receive_stream test.mp3
Once you have this working fine, modify the receiver program so that it writes the packets in-order on the standard output. This will allow you to redirect stdout to a file
# sender side did not change
send_stream test.mp3
# receiver side
receive_stream > test.mp3
Then, you can use madplay to play the mp3 while it is received simply by redirecting receive_stream output to madplay:
# madplay - tells madplay to read its input from standard input.
receive_stream | madplay -
For a good mp3 decoder, take a look at MAD.
I have written an application which triggers an IP Camera to stream it's data (MPEG4) over RTP. This works fine so far - I start to setup and start the stream with the corresponding RTSP commands ( DESCRIBE, SETUP and PLAY ).
While streaming I receive the usual Sender Reports and send my own Receiver Reports - Everything is working fine here.
Now with the application mentioned above, I do NOT read the stream. I have a seperate hardware , which just logs all the stuff going over the Ethernet ( a little bit like Wireshark ). Now when the whole streaming is finished I can download those logs from my hardware and extract data from them.
So what I have then is a logfile with all the data from the RTP stream as raw data.
My question would now is: How do I write this appropriately into a MPEG4 file? I know this is a very broad question and I don't expect to get a step-by-step tutorial. But actually I am a bit overwhelmed and don't know where to start.If I just memcpy all the Payload from the RTP messages sequentially into a MPEG4 file it doesn't work. Now I am also a bit confused by SDP and stuff.
Well maybe someone has a link or some help for me..?
You should first read RFC3016, which describes the RTP format of MPEG-4 stream, then you'll know how to extract MPEG-4 frames from the RTP stream.
I actually changed from MPEG4 to H.264 - it actually was a little bit easier to write a video file like this. For H.264 this answer covers it pretty much:
How to process raw UDP packets so that they can be decoded by a decoder filter in a directshow source filter
Okay I got the following problem:
I have an IP Camera which is able to stream MPEG4 data over RTP
I am able to connect to this camera via RTSP
I can receive the raw RTP data.
So what problems do I have now?
1. Extract Data
What is the data I actually want? I know that I have to trunkate the RTP Header - but is there anything else I need to cut from the RTP packets?
2. Packetization Mode
I read that I should expect a field Packetization Mode in my SDP- well it's not there. Does that mean I have to assume some kind of standard packetization mode?
3. Depacketization
If I got it right I need to buffer all incoming frames with the Marker Bit = false until I get a frame with Marker Bit = true to get a complete MPEG4 Frame. What exactly do I have to understand by MPEG4 Frame? Keyframe + data until next keyframe?
4. Decode
Do I have the decode the data any further then? In other threads I saw that people used another decoder - but what is there left to decode? I mean the camera should send the data already MPEG4 coded?
5. Libraries
If I really need to decode the data, are there any open libraries I could use for that? Or maybe there is even a library which has some functions where I can just dump my RTP data and then magic happens and I get my mp4. ( But I assume there will be nothing like that .. )
Note: Everything I want to do should be part of my own application, meaning for example, I can't use an external software to parse the data.
Well long story short - I'd really need some kind of step by step explanation for this to do. I know this is a broad question but I don't know any further. I also looked into the RFCs, but I couldnt extract much information out of them.
Also I already looked up these two Questions:
How to process raw UDP packets so that they can be decoded by a decoder filter in a directshow source filter
MPEG4 extract from RTP payload
But also the long answer from the first question could not make everything clear to me.
UPDATE: Well I informed a bit further and now I don't know where to look anymore. It seems that all the packetization stuff etc. is actually not needed for my purpose. I also recorded a stream with openRTSP. When I open those files in a Hex-Editor I see that there are 16 Bytes which I can't identify, followed by the config part of the SDP. Then the frame starts with the usual 00 00 01 B6. Also oprenRTSP adds some kind of tail to the MP4 - well I actually don't know what I need and whats just some "extra" stuff which isn't mandatory.
I know that I have to trunkate the RTP Header - but is there anything
else I need to cut from the RTP packets?
RTP packet might have stuff data from a file format (such as MP4) or it could have directly based on RFC 3640 or something similar. You need to find that out.
What exactly do I have to understand by MPEG4 Frame? Keyframe + data
until next keyframe? Do I have the decode the data any further then?
In other threads I saw that people used another decoder - but what is
there left to decode? I mean the camera should send the data already
MPEG4 coded?
You should explore basics of MPEG compression to appreciate this fully. The depacketization only give you a string of bits. This is compressed data. You need to uncompress it (decode it) to see it on the screen.
are there any open libraries I could use for that?
try ffmpeg or MPEG4IP
I would like to convert a TS file to mpeg file. Is there any documents describing such process?
I know the TS architecture, but I don't know mpeg's file architecture. Any info on this subject will highly appreciated.
Thank you.
What you are probably wanting to do is convert from MPEG-TS (Transport Stream) to MPEG-PS (Program Stream). MPEG-PS is the format of a standard .mpg file as well as the format DVD video uses.
You probably should get a hold on the standard which is ISO/IEC 13818-1. This standard contains all of the MPEG-TS and MPEG-PS container details (it does not cover the coded video which is covered in ISO/IEC 13818-2).
Luckily, this conversion is rather simple since most of the entire MPEG-PS structure is contained within the MPEG-TS format. The transport stream contains a series of 188 byte packets that each have a header. PES (Program Elementary Stream) packets are contained within the packet payloads. PES packets contain the actual video or audio payload. A PES packet can be any length and most of the time they span several TS packets. Demuxing the PES packets from the transport stream really just involves removing the TS headers and concatenating the payload data correctly to form the PES packets.
Once you have a stream of PES packets, you will multiplex them into the Program Stream format as laid out in the standard. So basically, you don't need to parse the PES packets or their content, you can just lift them from one format and insert them into the other.
Even though the conversion is fairly simple, it still requires quite a bit of work since you will need to become pretty familiar with the container standard and be meticulous with your parsing of the bitstream to get things right. So even though I say the conversion is simple, that is only in the sense that it is simple compared to other format conversions where you might have to dig down further into the video data.
I am trying to add some good resources that might help.
Here are some documents that explains the details of Transport and Program streams and associated packetization structures.
This explains the differences between Transport stream and Program stream. http://www.vbrick.com/docs/VB_WhitePaper_TransportStreamVSProgramStream_rd2.pdf
This explains the over view of MPEG and includes packetization as well.
http://www.img.lx.it.pt/~fp/cav/Additional_material/MPEG2_overview.pdf
THis explains the other aspects of transport streams on how programs are selected using tables etc. http://www.bitrouter.com/pdf/tutorial-psip.pdf
Basically, you need to depacketize the transport stream and decompose into PES packets (along with the time stamps) and then apply the program stream packetization process.
The crucial thing is how do you maintain the relative gap and timing of the packets in PS streams when you mux it back. Hence, you must preserve the PTS/DTS timestamps in the PES packets.
I am listing some tools here - that are good example for part of your work - and they are better known to be with compliance to MPEG2 systems standard.
tstools ( http://tstools.berlios.de/)
mplex (from mjpegtools)
dvb-mplex (part of libdvb, http://www.metzlerbros.org/dvb/)
DVB-replax (also part of libdvb, http://freshmeat.net/projects/dvb-replex/ or http://www.metzlerbros.org/dvb/)
avidemux. http://avidemux.sourceforge.net/
Another good way to begun learning is to use Gstreamer plug-in framework if you want to understand the broader flow quickly.
FFMPEG can be used to convert from a TS to MPEG. More info here.