Not able to get the frame width and height from the rtsp url from h265 camera. Can I get any guidelines for fetching the resolution.
You can process the sps packets to find the height and width for your bit-stream Follow the Sequence parameter set RBSP semantics in the following link and move the bytes to process the sps data LINK: ITU H265 DOC
Related
I am trying to make a .mp4 file from the audio and video rtp packets of an IP camera.
When the audio format in the camera is configured as MPEG-2 ADTS i receive rtp packets with a payload size of 144 bytes, but these are made of two audio frames 72 bytes each.
However, when i configure the format as MPEG-1 the payload is made of only one audio frame.
What is the reason for this difference? I could get this info from some bits of the payload? as i do for the bitrate, samplerate, etc. I have read that the theoric packet size is 144 bytes, so how could i retrieve the frame size and number of frames in the package?
Besides, in order to calculate the theoric frame duration i am using the next formula:
time = 1/bitrate * framesize (in bytes) * 8
This is working well in the case of MPEG-2 with different combinations of bitrate and samplerate. However, it does not seem to work for the MPEG-1. Am i doing something wrong here?
I've a 16bit greyscale video stream from a LWIR (thermal camera) and I want to forward the stream over RTP without any compression.
gstreamer format is: video/x-raw,format=GRAY16_LE,width=640,height=520,framerate=9/1
But I can't find any plugin to transmit the data over RTP.
https://gstreamer.freedesktop.org/documentation/rtp/index.html?gi-language=c
Do you have an idea?
Thanks, Martin
Check for the specs of uncompressed video data over RTP:
https://www.rfc-editor.org/rfc/rfc4175
As you will notice your specific format is not covered by the specification.
Question: What does the Libav/FFmpeg decoding pipeline need in order to produce valid presentation timestamps (PTS) in the decoded AVFrames?
I'm decoding an H264 stream received via RTSP. I use Live555 to parse H264 and feed the stream to my LibAV decoder. Decoding and displaying is working fine, except I'm not using timestamp info and get some stuttering.
After getting a frame with avcodec_decode_video2, the presentation timestamp (PTS) is not set.
I need the PTS in order to find out for how long each frame needs to be displayed, and avoid any stuttering.
Notes on my pipeline
I get the SPS/PPS information via Live555, I copy these values to my AVCodecContext->extradata.
I also send the SPS and PPS to my decoder as NAL units, with the appended {0,0,0,1} startcode.
Live555 provides presentation timestamps for each packet, these are in most cases not monotonically increasing. The stream contains B-frames.
My AVCodecContext->time_base is not valid, value is 0/2.
Unclear:
Where exactly should I set the NAL PTS coming from my H264 sink (Live555)? As the AVPacket->dts, pts, none, or both?
Why is my time_basevalue not valid? Where is this information?
According to the RTP payload spec. It seems that
The RTP timestamp is set to the sampling timestamp of the content. A 90 kHz clock rate MUST be used.
Does this mean that I must always asume a 1/90000 timebase for the decoder? What if some other value is specified in the SPS?
Copy the live555 pts into the avpacket pts. Process the packet with avcodec_decode_video2, and then retrieve the pts from avframe->pkt_pts, these will be monotonically increasing.
There is no need to set anything in the codec context, apart from setting the SPS and PPS in the AVCodecContex extradata
You can find a good example in VLC's github:
Setting AVPacket pts: https://github.com/videolan/vlc/blob/master/modules/codec/avcodec/video.c#L983
Decoding AVPacket into AVFrame: https://github.com/videolan/vlc/blob/master/modules/codec/avcodec/video.c#L1014
Retrieving from AVFrame pts:
https://github.com/videolan/vlc/blob/master/modules/codec/avcodec/video.c#L1078
avcodec_decode_video2() reorders the frames so that decode order and presentation order is the same.
Even if you somehow convince ffmpeg to give you PTS on the decoded frame it should be the same as DTS.
//
// decode a video frame
//
avcodec_decode_video2
(
ctxt->video_st->codec,
frame,
&is_finished,
buffer
);
if (buffer->dts != AV_NOPTS_VALUE)
{
//
// you should end up here
//
pts = buffer->dts;
}
else
{
pts = 0;
}
//
// adjust time base
//
pts *= av_q2d(ctxt->video_st->time_base);
I use live555 to receive RTP video frame (frame encoded in H264). I use Live555 open my local .sdp file to receive frame data. I just saw DummySink::afterGettingFrame was called ceaselessly。 if fReceiveBuffer in DummySink is correct, Why FFMPEG cannot decode the frame? My code is wrong?
Here is my Code Snippet:
http://paste.ubuntu.com/12529740/
the function avcodec_decode_video2 is always return failed , its value less than zero
fReceiveBuffer is present one video frame？
Oh, Here is my FFMPEG init code need to open related video decoder:
http://paste.ubuntu.com/12529760/
I read the document related H264 again， I found out that I-frame(IDR) need SPS/PPS separated by 0x00000001 insert into the header and decoder have a capacity to decode the frame correctly. Here is a related solution
FFmpeg can't decode H264 stream/frame data
Decoding h264 frames from RTP stream
and now, My App works fine, it can decode the frame and convert it to OSD Image for displaying to screen .
I use Live555 h.264 stream client to query the frame packets from an IP camera, I use ffmpeg to decode the buffer and analysis the frame by OpenCV.(those pipeline are based on testRTSPClient sample, I decode the h.264 frame buffer in DummySink::afterGettingFrame() by ffmpeg)
And now I wanna stream the frame to another client(remote client) OnDemand mode in real-time, the frame may added the analysis result(boundingboxs, text, etc), how to use Live555 to achieve this?
Well, your best bet is to re-encode the resultant frame (with bounding boxes etc), and pass this to an RTSPServer process which will allow you to connect to it using an rtsp url, and stream the encoded data to any compatible rtsp client. There is a good reference on the FAQ for how to do this http://www.live555.com/liveMedia/faq.html#liveInput which walks you through the steps taken, and provides example source code which you can modify for your needs.