gstreamer

I changed the stream url in the included Tutorial 5 (a basic media player) to a h.264/mp3 media stream (from its original ogv stream) and it started complaining about some missing plugins. After doing some googling I found Prajnashi's gst-ffmpeg plugin for Android https://github.com/prajnashi/gst-ffmpeg but it is certainly not suitable for the SDK. If the SDK does not have the required plugin how do I go about resolving this? I am new to GStreamer and have no idea how to compile any of the existing plugins for Android. You just need to add $(GSTREAMER_PLUGINS_CODECS_RESTRICTED) to your Android

I am streaming an mp4(mpeg-4) file from one device to another using gstreamer over RTP stream. Basically I am splitting up the mp4 file into its audio and video file and then sending it all to the other device where it gets streamed. Now, I want to save the mp4 file to disk in the other device, but my problem is that I am able to save the audio and video files seperately and it cannot be played individually. I am confused on how to combine both the audio and video rtp streams to form my mp4 file back and save it to a file in the other device. Here are the command line codes : Sender(Server)

I want to stream some random bytes to Gstreamer and display it as follows: [Rand Bytes]--[Video source=appsrc]--[Video sink=ximagesink] The following Python code I found in this SO post works source = gst.element_factory_make("appsrc", "source") caps = gst.Caps("video/x-raw-gray,bpp=16,endianness=1234,width=320,height=240,framerate=(fraction)10/1") source.set_property('caps',caps) source.set_property('blocksize',320*240*2) source.connect('need-data', self.genRandBytes) colorspace = gst.element_factory_make('ffmpegcolorspace') #To remove videosink = gst.element_factory_make('ximagesink') caps =

I am using Python bindings for Gstreamer and am using the following pipeline to convert a wav file to mp3. I used one of the suggestions in this question , with some modifications (as I was getting some errors when original syntax was used) gst.parse_launch("filesrc location=C:\\music.wav ! decodebin ! audioconvert ! lame ! filesink location=C:\\music.mp3") When I run this code in Python, I get no errors. However, it doesn't generate music.mp3 file. What else do I need to do so that it creates a new file music.mp3 your pipeline is correct - or more specifically, your choice of elements and

EDIT: I forgot to mention this is working with an axis encoder. I've recently started using gStreamer-sharp and I can get a video playing with playbin but I need my video to be live, thus a need to set the latency. but I have come across this error: GST_ELEMENT_PADS gstelement.c:722:gst_element_add_pad:<Source> adding pad 'recv_rtp_src_0_219678342_96' 0:00:11.033000000 13088 05418DA0 INFO basesrc gstbasesrc.c:2519:gst_base_src_loop:<udpsrc1> pausing after gst_pad_push() = not-linked 0:00:11.033000000 13088 05418DA0 WARN basesrc gstbasesrc.c:2582:gst_base_src_loop:<udpsrc1> error: Internal data

I've been strugling for two weeks to create an environment for building a gstreamer plugin on windows (needed for a songbird addon). I've installed MSYS, MinGW and Cygwin, then installed GStreamer OSSBuild, and I also downloaded the sources for Songbird, which come with their own precompiled version of gstreamer. I was unable to run gst-inspect (or any other gstreamer applications) from the songbird sources and I figured I will settle for OSSBuild (as I was able to run gst-inspect from the compiled OSSBuild). When following the instructions for building a GST plugin ( found here ) through,

I'm trying to fetch the video file from my local directory,enable the stream from server and capture these frames from my Client side.I have used the following pipelines: Server side: gst-launch -v filesrc location=/home/gokul/Videos/Econ_TestVideo/radioactive.mp4 ! qtdemux ! rtpmp4vpay ! udpsink host=192.168.7.61 port=5000 sync=true Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP4V-ES, profile-level-id=(string)1, config=(string

I would like to use gstreamer to save an arbitrary clip from one audio file to a new file. For example, a segment from 1 minute to 2 minutes in the original. How do I do it? You need gnonlin . See http://www.jonobacon.org/2006/12/27/using-gnonlin-with-gstreamer-and-python/ You won't need a gnlcomposition because you only want one segment. Use a gnlfilesource with its start and duration set to 0, 1 minute, and media-start and media-duration set to 1 minute, 1 minute. All times and durations are in nanoseconds. Take 5 seconds from source.mp3 starting at 10 seconds, write to destination.ogg : gst

I am looking for an example/help for displaying a gstreamer-sharp feed in a WinForms application. I am using VS 2012 and have the "glue" project built for this version of VS. I also have glib-sharp, gstreamer-sharp installed and referenced by my project. I have the gstreamer bin directory set as my project's working directory. If have the following code in a button click handler I get the GStreamer D3D video sink test Window which pops-up on top of my Form. Gst.Application.Init(); var pipeline = Parse.Launch(@"videotestsrc ! videoconvert ! autovideosink"); pipeline.SetState(State.Playing); I