gstreamer

cannot change output format in gstreamer

这一生的挚爱 提交于 2019-12-25 05:19:19
问题 I am using a camera which supports 1080 and 720p videos at both 30 and 25 frames per second. I can get the webcam to stream with the following gstreamer command on my ARM board gst-launch-1.0 -v v4l2src device=/dev/video0 ! "video/x-raw, width=1920, height=1080, format=(string)YUY2, framerate=30/1" ! xvimagesink sync=false -e However, I find that I cannot change any of these parameters. For example, if I change the framerate to 25/1 , I get the error: error: streaming task paused, reason not

Looping playback of a list of Gst.Sample with GstApp.AppSrc

浪子不回头ぞ 提交于 2019-12-25 04:52:05
问题 I'm trying to write a simple music player using GStreamer. I want to play any arbitrary music file ABS_FILE_PATH , store the samples for other purposes and later loop over these indefinitely, once the original end of stream is reached. Now playing the music works fine until short after the last sample of the track was played. Most of the time there's just silence, but sometimes there are one or two audible samples indicating, that the track just started playing again. The same holds for the

TS Stream Transport Over UDP from .TS files

∥☆過路亽.° 提交于 2019-12-25 03:38:45
问题 I would like to transport ts stream (.ts files) over UDP. I have used the below pipeline, gst-launch-1.0 filesrc location=a.ts ! tsparse ! rtpmp2tpay ! udpsink host="IP" port=port But, i am getting end of the stream within fraction of seconds. At, Receiver end i am receiving some data only. Warning arises when i set do-timestamp=true for filesrc. 0:00:00.051245178 4751 0xe494a0 WARN mpegtspacketizer mpegtspacketizer.c:1388:calculate_skew: delta - skew: 5124091:34:25.044918089 too big, reset

Create live video of jpeg snapshots using Gstreamer and JavaScript setTimeout()

别等时光非礼了梦想. 提交于 2019-12-25 02:53:40
问题 I am trying to create a live "video" stream using an tag on a web page. A Gstreamer pipeline continually overwrites a file "snapshot.jpeg" with a new frame grabbed from a webcam using video4linux2 with a framerate of 15 fps. A web page renders the image without caching every 100 ms. The problem is that I get ERR_CONTENT_LENGTH_MISMATCH (in browser console) for the image source on many frames. This is shown as a broken link in the browser. GStreamer 0.10 syntax gst-launch v4l2src ! video/x-raw

how do you build gstreamer's gst-launch pipelines?

不问归期 提交于 2019-12-25 01:44:27
问题 Let's say you have a video file. As far as I searched, you first need to know what container it uses by mediainfo command. $ mediainfo your_path_to_a_video.file you then need to find a demuxer for the container, so you do $ gst-inspect-1.0 | grep your_container_name_such_as_ogg now that you have a proper demuxer, such as oggdemux , you can split video and audio. If you want to display the video, you first need to know the codec name, and you will need to decode it to output to the screen.

Push images into Gstreamer pipeline

孤街浪徒 提交于 2019-12-24 23:02:37
问题 I have been following many examples about pushing an image into a Gstreamer pipeline but still I can't make my code work. Any suggestion (beside telling me to try with Gstreamer1.0 instead of 0.10) will be very appreciated. I want to understand what is wrong in the following script that feeds an appsrc element with a jpeg image. Later I will use the same code to feed openCv images that I obtain from my camera but first I want to understand the basics by making this simple example work.

Add sound to the GStreamer mosaic

喜夏-厌秋 提交于 2019-12-24 21:28:56
问题 This is the mosaic code GStreamer 2x2 gst-launch-1.0 -e \ videomixer name=mix \ sink_0::xpos=0 sink_0::ypos=0 sink_0::alpha=0\ sink_1::xpos=0 sink_1::ypos=0 \ sink_2::xpos=200 sink_2::ypos=0 \ sink_3::xpos=0 sink_3::ypos=100 \ sink_4::xpos=200 sink_4::ypos=100 \ rtmpsrc location='rtmp://streaming.example.com:1935/209147924'\ ! decodebin name=dec1 ! videoconvert ! videoscale \ ! video/x-raw,width=200,height=100 \ ! mix.sink_1 \ decodebin1. ! queue ! audioconvert ! audiomixer name=amix \

How can I use gstreamer & smpte to concatenate 2 video files with gst-launch?

江枫思渺然 提交于 2019-12-24 20:26:27
问题 I have 2 video files (vid1.mov and vid2.mov), both have the same frame size and frame rate. I want to have 1 final video with shows vid1.mov and then vid2.mov, one after the other. I also want there to be a transition from one video to another (rather than an abrupt change of video), and have discovered the smpte plugin for gstreamer, which goes what I want. Using gst-launch on the ubuntu linux command line, how can I merge the 2 videos together with a transition? (Assume I want to use the

Gstreamer on iphone, blackberry, android, nokia? How would it perform?

…衆ロ難τιáo~ 提交于 2019-12-24 19:12:35
问题 Gstreamer on IPhone, Android, Blackberry, Nokia. How can we make it using C or D or Vala languages? Or should we just use Java or Lua? 1) Glib porting to IPhone, Android, Blackberry, Nokia is available? For Android, there is a tips http://gstreamer.freedesktop.org/wiki/GstreamerAndroid_InstallInstructions For IPhone, no idea For Nokia, no idea For Blackberry, no idea Wish someone put some answers and feedback on this topic. 回答1: I don't know much about mobile development, but your best bet

OpenCv + Gstreamer from an app, getting initial 30s delay

|▌冷眼眸甩不掉的悲伤 提交于 2019-12-24 18:56:42
问题 So my application is exposing an RTP stream using new VideoWriter(pipeline-definition); The pipeline definition is: appsrc is-live=1 do-timestamp=1 format=3 stream-type=0 min-latency=0 max-latency=500000000 ! queue leaky=2 max-size-time=500000000 ! videoconvert ! video/x-raw ! x264enc ! h264parse ! rtph264pay config-interval=10 pt=96 ! udpsink host=127.0.0.1 port=9000 The problem I'm faced with is 30s delay in the stream when viewing it in VLC. No matter what I do, VLC is always 29-30s behind