HI I am trying to record rtsp stream coming from camera(H264 format).
I am using following gst command to do recording in MPEG4 Format
gst-launch -e rtspsrc location=rtsp://10.17.8.136/mediainput/h264 latency=100 ! decodebin ! ffenc_mpeg4 ! avimux ! filesink location=test.mp4
and H264 format
gst-launch-0.10 -e rtspsrc location="rtsp://10.17.8.136/mediainput/h264" latency=100 ! rtph264depay byte-stream=false ! capsfilter caps="video/x-h264,width=1920,height=1080,framerate=(fraction)25/1" ! mp4mux ! filesink location=testh264.mp4
Both are doing recording but i have observed that There is RAM mermory is gradually increasing.
Does gStreamer has memory leak. or there is problem in my pipeline command?
That is not a leak, the mp4 muxer is building the index table in memory, before writing it out to disk on eos.
Related
I am currently experiencing a small problem with GStreamer, here are more details:
Configuration:
Intel i7-6700
Intel HD Graphics 530
Ubuntu 18.04 LTS
GStreamer1.0
VAAPI plugin
I receive a UDP stream from a video source, this stream is sent in RAW UYVY format. Here is my command line to decode it:
gst-launch-1.0 -v udpsrc port="1234" caps = "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:2, depth=(string)8, width=(string)1920, height=(string)1080, colorimetry=(string)BT709-2, payload=(int)96, ssrc=(uint)1188110121, timestamp-offset=(uint)4137478200, seqnum-offset=(uint)7257, a-framerate=(string)25" ! rtpvrawdepay ! decodebin ! queue ! videoconvert ! xvimagesink
Problem as we can see on the screenshot below, the CPU load (right) is far too high for this kind of task and we can see the GPU load (left) which is almost zero.
To overcome this problem, I want to use the VAAPI graphics acceleration as I did in a previous project with H264 of which here is the command line below:
gst-launch-1.0 -v udpsrc port=1234 caps= "application/x-rtp, media\=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-level-id=(string)640028, payload=(int)96, ssrc=(uint)2665415388, timestamp-offset=(uint)3571350145, seqnum-offset=(uint)18095, a-framerate=(string)25" ! rtph264depay ! queue ! vaapih264dec low-latency=1 ! autovideosink
The line above works perfectly and the CPU has almost no more loads. So I adapt this command line to use it with a RAW stream, here is the command:
gst-launch-1.0 -v udpsrc port="1234" caps = "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:2, depth=(string)8, width=(string)1920, height=(string)1080, colorimetry=(string)BT709-2, payload=(int)96, ssrc=(uint)1188110121, timestamp-offset=(uint)4137478200, seqnum-offset=(uint)7257, a-framerate=(string)25" ! rtpvrawdepay ! vaapidecodebin ! videoconvert ! xvimagesink
It is the same line as the one at the beginning but I changed the element decodebin by vaapidecodebin as I had replaced avdec_h264 by vaapih264dec for my H264 stream. Unfortunately it doesn't work and I end up with this error:
WARNING: wrong pipeline: unable to connect rtpvrawdepay0 to vaapidecodebin0
How I can solve this problem? Do you have any leads to solve this problem?
What exactly are you trying to accelerate here? The CPU load is probably either due to the videoconvert as this is run in software to convert UYVY into a format your renderer supports (Hopefully that's another YUV format and not RGB) or it is the data transfer of the uncompressed data from CPU memory to GPU memory.
Note that transferring uncompressed image data is a much higher data rate than compressed H.264 video.
If you think the videoconvert is the expensive part you may want to try to use OpenGL for convert and displaying: .. ! glupload ! glcolorconvert ! glimagesink.
Maybe vaapipostproc can help you with color conversion if you don't want to go the OpenGL route.
I have streamed video via vlc player over rtsp and then I have displayed this video via gstreamer-0.10. However, While vlc was streaming video over rtsp, I suddenly lost stream in the first minute of stream before end of stream.
I have used following pipeline:
GST_DEBUG=2 gst-launch-0.10 rtspsrc location=rtsp://127.0.0.1:8554/test !
gstrtpjitterbuffer ! rtph264depay ! ffdec_h264 ! videorate ! xvimagesink
sync=false
I have got following output:
rtpjitterbuffer.c:428:calculate_skew: delta - skew: 0:00:01.103711536 too big, reset skew
rtpjitterbuffer.c:387:calculate_skew: backward timestamps at server, taking new base time
Got EOS from element "pipeline0".
Execution ended after 59982680309 ns.
Setting pipeline to PAUSED ...
gst_rtspsrc_send: got NOT IMPLEMENTED, disable method PAUSE
How to fix this problem ?
I have found solution. I have used rtspt://... instead of rtsp://... to enforce TCP instead of UDP.
gst-launch-0.10 rtspsrc location= rtspt://127.0.0.1:8554/test ! gstrtpjitterbuffer ! rtph264depay ! ffdec_h264 ! xvimagesink sync=false
I work on embedded linux. I want play video with minimum CPU. So after I completed compile, I tried play video with mplayer and gstreamer. Mplayer use CPU avarage %10-20. I want to obtain this perform on gstreamer. So I tried these command:
1- gst-launch filesrc location=video_path.mpeg ! mpegdemux ! mpeg2dec ! autovideosink
2-gst-launch-0.10 filesrc location=video_path.mpeg ! dvddemux ! mpegvideoparse ! mpeg2dec ! xvimagesink
These commands use avarage %10-20 CPU. This number that I want number. But audio did not work with these command. I tried added audio element but I could not achieve.
I also tried gst-launch-1.0 playbin uri=file:///video_path.mpeg. Audio work with this command but CPU usage is so high and I don2t prefer this.
How can I work audio with 1 or 2 commands?
1- gst-launch filesrc location=video_path.mpeg ! mpegdemux ! mpeg2dec
! autovideosink
2-gst-launch-0.10 filesrc location=video_path.mpeg ! dvddemux !
mpegvideoparse ! mpeg2dec ! xvimagesink
With the above two pipelines you are asking gtreamer to just play video, as a result you aren’t getting any audio.
gst-launch filesrc location=video_path.mpeg ! mpegdemux name=demuxer
demuxer. ! queue ! mpeg2dec ! autovideosink demuxer. ! queue ! mad !
audioconvert ! audioresample ! autoaudiosink
The above pipeline should play both audio and video.
Note: If you have support for hardware decoding that would reduce further CPU usage.
I am trying to stream audio and video via Gstreamer via UDP but playback on VLC only returns video without audio. Currently I am using a sample of Big Buck Bunny and have confirmed that it does have audio. I am planning to use Snowmix to feed media to Gstreamer output in the future.
Streaming from file source via UDP to playback on VLC I currently perform by:
gst-launch-1.0 -v uridecodebin uri=file:///home/me/files/Snowmix-0.5.1/test/big_buck_bunny_720p_H264_AAC_25fps_3400K.MP4 ! queue ! videoconvert ! x264enc ! mpegtsmux ! queue ! udpsink host=230.0.0.1 port=4012 sync=true
which allows me to open a network stream in VLC on my Windows machine to receive packets and only plays video.
What am I missing from my command?
As RSATom stated previously, the audio is missing from the pipeline.
The correct pipeline for video and audio is the next (tested with the same input file):
gst-launch-1.0 -v uridecodebin name=uridec uri=file:///home/usuario/Desktop/map/big_buck_bunny_720p_H264_AAC_25fps_3400K.MP4 ! queue ! videoconvert ! x264enc ! video/x-h264 ! mpegtsmux name=mux ! queue ! udpsink host=127.0.0.1 port=5014 sync=true uridec. ! audioconvert ! voaacenc ! audio/mpeg ! queue ! mux.
Remember that in this case you're re-encoding all the content from the source video file, which means high CPU consumption. Other option would be to demux the content from the input file and mux again without encoding (using h264parse and aacparse).
i am using g streamer-0.10 on Ubuntu os for streaming an web cam video on to an rtmp server i am getting an video output but their is a problem in audio . Below command used for streaming
gst-launch-0.10 v4l2src ! videoscale method=0 ! video/x-raw-yuv,width=852,height=480,framerate=(fraction)24/1 ! ffmpegcolorspace ! x264enc pass=pass1 threads=0 bitrate=900 tune=zerolatency ! flvmux name=mux ! rtmpsink location='rtmp://..../live/testing' demux. alsasrc device="hw:0,0" ! audioresample ! audio/x-raw-int,rate=48000,channels=2,depth=16 ! pulseaudiosink
Blockquote
by running the above command i got an error
gstbaseaudiosrc.c(840): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 13920 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.
Blockquote
so the audio is not audible.
Help me out to solve this problem.
Thanks in advance
Ameeth
I don't understand your pipeline. What is "demux." in the middle?
The problem you are facing is because you have not seperated your elements with queues. Keep a queue before your sinks and after your sources to give the rest all seperate threads to run. It should allow get rid of the issue.
Since I don't have pulse audio or rtmp reciever in my system i have tested out the following and it works.
gst-launch-0.10 v4l2src ! ffmpegcolorspace ! queue ! x264enc pass=pass1 threads=0 bitrate=900000 tune=zerolatency ! queue ! flvmux name=mux ! fakesink alsasrc ! queue ! audioresample ! audioconvert ! queue ! autoaudiosink
You can change it accordingly and use it. The only thing I had to do to make it work and remove the error your are facing is to add the queues.
For me (Logitech c920 on Raspberry Pi3 w/ GStreamer 1.4.4) I was able to get rid of the "Dropped samples" warning by using audioresample to set the sampling rate of the alsasrc to something that flvmux liked. From gst-inspect-1.0 flvmux, it looks like flvmux only supports 5512, 11025, 22050, 44100 sample rates for x-raw and 5512, 8000, 11025, 16000, 22050, 44100 for mp4. Here's my working pipeline
gst-launch-1.0 -v -e \
uvch264src initial-bitrate=800000 average-bitrate=800000 iframe-period=2000 device=/dev/video0 name=src auto-start=true \
src.vidsrc ! video/x-h264,width=864,height=480,framerate=30/1 ! h264parse ! mux. \
alsasrc device=hw:1 ! 'audio/x-raw, rate=32000, format=S16LE, channels=2' ! queue ! audioresample ! "audio/x-raw,rate=44100" ! queue ! voaacenc bitrate=96000 ! mux. \
flvmux name=mux ! rtmpsink location="rtmp://live-sea.twitch.tv/app/MYSTREAMKEY"
I was surprised that flvmux didn't complain about getting an audio source that was at an unsupported sampling rate. Not sure if that's expected behavior.