I'm using OPUS with avcodec to encode sounds and stream it using my own protocol.
It works with the MP2 codec so far but when I'm switching to OPUS, I have this issue :
[opus # 1b06d040] Error parsing the packet header.
I suppose that unlike MP2, I need to generate a header for my OPUS encoded data stream but I don't know how.
Can someone explain me how to do that? Thanks.
This error comes from ff_opus_parse_packet() failing, which handles the raw opus packet header, what the specification calls the 'TOC' (for table-of-contents) byte and optional subframe lengths. It means libavcodec couldn't find the packet duration where it expected.
So probably your custom protocol is corrupting the data, returning the wrong data length, or you're otherwise not splitting the opus packet out of your framing layer correctly.
You don't need to invent your own protocol if you don't want to. There are two established designs: Opus over RTP for interactive use (like live chat where latency matters) is documented in RFC 7587. For HTTP streaming, file storage for recording, playback and other applications like that use the Ogg container, documented here. There are implementations of both of these in libavformat. See rtpenc.c, oggenc.c and oggparseopus.c if you're curious about the details.
Related
I am trying to transmit TS packets using Ethernet. I am using C++ and ffmpeg libraries. At the moment I can send a hevc encoded ts stream via Ethernet successfully. But the output data rate varies. I want to maintain a constant(approximately) data rate..
I am using "av_interleaved_write_frame()" to transmit the TS packets.
I know this can be achieved using NULL packet transmission. Can anyone tell me how to do this using ffmpeg?
Thank you.
What you are trying to achieve is called Constant BitRate: you should set minrate, maxrate and bitrate to the same value to get it.
cf similar questions for more detailed examples:
https://superuser.com/a/314355/329216
How to force Constant Bit Rate using FFMPEG
And interesting external links:
https://support.octoshape.com/entries/25126002-Encoding-best-practices-using-ffmpeg
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.
I deal with rtp packets of gtalk video. I want to make video using gtalk rtp payload data. According to my search gtalk use h264 codec for video.
I combined all of rtp payload which is send with gtalk video and wanted to play ffplay using this comment "ffplay -f h264 "filename" but I can't see
anything and I take this error "Could not find codec parameters (Video: h264, yuv420p)". I think my wrong is combining rtp payload. how can I play this payload?
Thanks for your helps.
Cheers.
It could be that you need the sequence and picture parameters sets (SPS and PPS) which are often transferred during session setup. Standard protocols used for this are RTSP and SIP though I have no clue whether gtalk uses either of these.