So I'd like to stream video from an RTSP stream using OpenCV, and immediately save the raw video to a file without decoding. We need high performance and might be processing as many as 500 cameras on one machine. Which means that even though typically small and 2% CPU, the extra 10x CPU needed to decode every frame adds up. When I run with ffmpeg command line it's getting so little usage it shows 0.0% CPU.
The recommendation I've seen before is just use ffmpeg, reason I'd like OpenCV is:
A) We might need to do some image analysis in the future, wouldn't be on all frames though, just a small sample of them, so all still need to be saved frame by frame to file without decoding, but some will separately be decoded.
B) Simpler to implement than ffmpeg (that in the future would need to pass select frames to OpenCV)
Edit: I've tried using VideoWriter::fourcc('X', '2', '6', '4'), and -1 in order to try to skip encoding. Seems like it goes ahead and does it anyway even though it's already in that format?
Thoughts? Thanks!
I'm using ffmpeg to read an h264 RTSP stream from a Cisco 3050 IP camera and reencode it to disk as h264 (there are reasons why I'm not just using -codec:copy).
The ffmpeg version is as follows:
ffmpeg version 3.2.6 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 6.3.0 (Alpine 6.3.0)
I've also tried with ffmpeg 2.8.14-0ubuntu0.16.04.1 and the latest ffmpeg built from source (I used this commit) and see the same behaviour as below.
The command I'm running is:
ffmpeg -rtsp_transport udp -i 'rtsp://<user>:<pw>#<ip>:554/StreamingSetting?version=1.0&action=getRTSPStream&ChannelID=1&ChannelName=Channel1' -r 10 -c:v h264 -crf 23 -x264-params keyint=60:min-keyint=60 -an -f ssegment -segment_time 60 -strftime 1 /output/%Y%m%d_%H%M%S.ts -abort_on empty_output
I get a variety of errors at a fairly steady rate of at least one per second. Here's a sample:
[rtsp # 0x7f268c5e9220] max delay reached. need to consume packet
[rtsp # 0x7f268c5e9220] RTP: missed 40 packets
[h264 # 0x55b1e115d400] left block unavailable for requested intra mode
[h264 # 0x55b1e115d400] error while decoding MB 0 12, bytestream 114567
[h264 # 0x55b1e115d400] concealing 3889 DC, 3889 AC, 3889 MV errors in I frame
The most common one is 'error while decoding MB x x, bytestream x'. This corresponds to severe corruption in the video file when played back.
I see many references to that error message on stackoverflow and elsewhere, but I've yet to find a satisfying explanation or workaround. It comes from this line which appears to correspond to missing data at the end of the stream. 'left block unavailable' comes from here and also looks like missing data.
Others have suggested using -rtsp_transport tcp instead (1, 2, 3) which in my case just gives a slightly different mix of errors, and still video corruption:
[h264 # 0x557923191b00] left block unavailable for requested intra4x4 mode -1
[h264 # 0x557923191b00] error while decoding MB 0 28, bytestream 31068
[h264 # 0x557923191b00] concealing 2609 DC, 2609 AC, 2609 MV errors in I frame
[rtsp # 0x7f88e817b220] CSeq 5 expected, 0 received.
Using Wireshark I confirmed that in both UDP and TCP mode, all of the packets are making it from the camera to the PC (sequential RTP sequence numbers without any missing) which makes me think the data is being lost after it arrives at ffmpeg.
I also see similar behaviour when running the same command against a Panasonic WV-SFV110 camera, but with less frequent errors overall. Switching from UDP to TCP on the Panasonic camera reduces but does not completely eliminate the errors/corruption.
I also tried a similar command with VLC and got similar errors (cvlc rtsp://<user>:<pw>#<ip>/MediaInput/h264 :sout='#transcode{vcodec=h264}:std{access=file, mux=ts, dst="output.ts"}) -- presumably the code hasn't diverged much since libav forked from ffmpeg.
The camera is plugged directly into a PoE port on the PC so network congestion can't be a problem. Given that the PC has enough CPU to keep up encoding the live stream, it seems to me a problem with ffmpeg that it still drops data from the TCP stream.
Qualitatively, there are several factors which seem to make the problem worse:
Higher video resolution
Higher system load on the machine running ffmpeg (e.g. transcoding to a low res .avi file produces fewer errors than transcoding to h264 VBR; using -codec:copy eliminates all errors except a couple while ffmpeg is starting up)
Greater motion within the camera view
What the does the error mean? And what can I do about it?
Looking at the initial error message:
[rtsp # 0x7f268c5e9220] max delay reached. need to consume packet
[rtsp # 0x7f268c5e9220] RTP: missed 40 packets
I guess that you are loosing UDP packets. The rest of the H.264 error messages are caused by receiving an incomplete bitstream.
Now key is to isolate the issue. Is your network dropping packets? Or is your sever too slow or overloaded receiving the UDP (RTP).
First I'd check the UDP buffer size of your OS. https://access.redhat.com/documentation/en-US/JBoss_Enterprise_Web_Platform/5/html/Administration_And_Configuration_Guide/jgroups-perf-udpbuffer.html
If increasing the UDP buffer size doesn't help - use ffmpeg with -codec:copy to lower the CPU load. Do you still get errors?
Since you want to reencode consider using Intel Quicksync -vcodec h264_qsv or some other hardware encoder lowering your CPU load.
The question is not so much about if the PC has enough CPU. But more about identifying the bottle neck in the processing pipeline. Your H.264 encoder (x264) may over subscribe your CPU so that you get momentary peak loads that result in packet drops. Try limiting the number of threads for x264 and/or lower the quality to 'fast' or 'faster'.
It does sound like packet loss is an issue. Higher video resolution and greater motion both increase the bitrate of the encoded video stream which will increase your packet loss. Depending on which packet is lost, you will see varying errors in the decoding process as you indicated in your post.
The higher system load running ffmpeg also indicates that your network card might be dropping packets, when e.g. ffmpeg takes too long to read them while it is busy transcoding the video.
First question is what is your network topology? Streaming over the public Internet is a lot harder than streaming over your LAN. What kind of switches/routers are in the network?
Next question, what bitrate is your camera streaming at? Try reducing this and check the results. Be systematic in your approach i.e.
don't transcode at first.
just receive the video.
write it to file.
Check for packet loss/video artifacts.
start at lower bitrates e.g. 100kbps and increase this if no loss is evident
The next thing I would try to do is to increase the size of the receiver buffers. While I am not that familiar with ffmpeg, it looks like you can set it via recv_buffer_size as indicated here. You then need to work out a reasonably big enough size based on your camera configuration to store e.g. a couple (5?) of seconds of video data. Check if there are less artifacts as you increase the receiver buffer size or longer periods without artifacts.
Of course if your processor is too slow to transcode the video in real-time, you will run out of space sooner or later, in which case, you might have to transcode to a lower resolution/bitrate or use less intensive encoder settings, etc or run the transcoding on a faster machine.
Also, note that adjusting receiver buffer size will not compensate for packet loss occurring on the public Internet so the above will help assuming you're streaming on a local network that supports the bitrate of the camera. If you exceed the bandwidth of the network you can expect packet loss. In that case streaming over TCP could help somewhat (at least until the receiver buffer overruns eventually).
More things you can try if the above does not help or solve the problem completely:
Sniff the incoming traffic with wireshark or tcpdump.
Have a look at the traces. Filter the trace using "RTSP".
You should be able to see the RTP traffic where consecutive RTP packets have increasing sequence numbers e.g. 20, 21, 22, 23, etc. If you see missing sequence numbers, then you've got packet loss and try streaming over TCP. Repeat the trace while streaming over TCP. Also, remember to increase the receiver buffer size also when streaming over TCP.
In summary you have a pipeline architecture and you need to determine where in the pipeline the loss is occurring:
camera -> network -> receiver buffer (OS) -> application (ffmpeg)
I have a video source which gives me a raw h264 stream. I need to re-stream this live input in a way it is cross-compatible and playable without any plugin. I've tried using ffmpeg+ffserver to produce a fragmented mp4, but unfortunately my iPhone isn't playing it.
Is there a way to make it (raw h264 in mp4 container) playable in iOS's Safari, or maybe another cross-platform container?
Ps: i'm using a raspberry pi 3 to host ffmpeg processes, so i'm avoiding re encoding tasks; instead i'm just trying to fit my raw h264 in a "ios-compatible" container and make it accessible through a media server.
For live streams you must use HTTP Live Streaming (HLS) with either the traditional MPEG-TS or fMP4 for newer iOS versions (see Apple HLS examples).
With FFmpeg you can use the hls muxer. The hls_segment_type option is used to choose between mpegts and fmp4.
I am developing a Live Streaming application where mobile application (iOS/Android) records the video and Audio and then encodes raw pixels to h.264 and AAC using VideoToolBox and AudioToolBox, these encoded pixels are converted to PES (Packetized Elementary Stream) separately (Video & Audio). Now we are stuck at what to transfer to the server either PES or MPEG-TS, which one gives the minimum latency and packet loss like Periscope, Meerkat, Qik, UStream and other live streaming applications.
For transmitting which networking protocol is best suitable TCP/UDP.
And what is required at server to receive these packets. I know FFMPEG will trans-code and generates the segmented files(.ts) and .m3u8 file for HLS streaming, but do we need any pipe before FFMPEG layer?
Please give me some ideas in terms of which is best and what are pros and cons of each.
Thanks
Shiva.
Please help me here.
I want to packetise and stream an HEVC encoded bitstream that resides on the server (e.g.Linux stream server) to a client machine( Linux) where the RTP headers will be removed and and the bitstream decoded to YUV using HEVC decoder.
The stages will includes:
Encoding the raw YUV and obtain an HEVC bitstream ---Server
packetise the bitstream/encapsulate using RTP----server
RTP packets streaming over UDP
Remove RTP and store --- client
decode bitstream using HEVC decoder ---- client
At the moment, I have only encoded the videos.
I want to start by packetizing the bitstream from the encoding process.
I will be grateful if someone could help me with information on how to extract HEVC NAL units and packetize for streaming over a network.
Many thanks.
James