Hi. I have an encoder which writes to a buffer, and I want to
push that buffer to the rtpmp4vpay. So does there exist an element (or an
element which I could edit) which could take in a custom buffer and output a
GstBuffer, so that I can push this buffer to the rtpmp4vpay??? It would be good
if such a framework exists. Thanks in advance. Regards, Neel. ------------------------------------------------------------------------------ Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi all, I had a command line command that copies video streams from a network into a file. When I try to change it to a code I face with problem. There is no error and the file is created but the content of the file is empty. I have also used sync=FALSE for the filesink element. I used two bins one for the audio and another for the video and then put the two bins and the avimux and filesink in the pipeline and started the pipepline. I even checked the state of the pipeline and it is in running state, but the file is empty. Pease help me. The following is the case:
I wanted to change the following into a code: gst-launch-0.10 -v avimux name=mux ! filesink location=videoandaudio.avi sync=true {udpsrc port=5000 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" num-buffers=5000 ! queue ! rtph263pdepay ! 'video/x-h263,width=320,height=240,framerate=(fraction)25/1' ! queue} ! mux.video_0 {udpsrc port=5002 caps="application/x-rtp,media=(string)audio,payload=(int)96, rate=(int)8000, encoding-name=(string)PCMA" ! queue ! rtppcmadepay ! 'audio/x-alaw, rate=(int)8000, channels=(int)1' ! queue} ! mux.audio_0; And the following is what I did: gst_bin_add_many (GST_BIN (audbin), udpsource_audio, queue_audio, pcmadepay, NULL); gst_bin_add_many (GST_BIN (vidbin), udpsource_video, queue_video_1, h263depay, NULL); gst_bin_add_many (GST_BIN (pipeline), vidbin, audbin, avimuxer, filedes, NULL); gst_element_link_many (udpsource_video, queue_video_1, h263depay, NULL); gboolean link_ok_video = gst_element_link_filtered (h263depay, avimuxer, caps_video_2); gst_element_link_many (udpsource_audio, queue_audio, pcmadepay, NULL); gboolean link_ok_audio = gst_element_link_filtered (pcmadepay, avimuxer, caps_audio_2); gst_element_link_many (avimuxer, filedes, NULL); gboolean pipstate=gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING); Thank you. - Zelalem S. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! ------------------------------------------------------------------------------ Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
request Sink pad of your avimux, link it to your audiobin.
use gst-inspect "ELEMENT_TO_VIEW" to view your element pads, some of them aren't always available. some are dynamic. Hope that helps. On Tue, Jun 23, 2009 at 1:00 PM, Zelalem Sintayehu <[hidden email]> wrote:
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In reply to this post by Mehta, Neel
On Tue, Jun 23, 2009 at 6:35 AM, Mehta, Neel <[hidden email]> wrote:
You are looking for "appsrc", you feed him GstBuffers and it pushes them to the next element. More info here.
-- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) ------------------------------------------------------------------------------ Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
In reply to this post by Thabelo Mmbengeni-2
Hi Thabelo, thank you for your response. I requested a sink for both audio and video and linked it to avimux as follows:
gst_pad_link (gst_element_get_pad (h263depay, "src"), gst_element_get_pad (avimuxer, "video_0")); gst_pad_link (gst_element_get_pad (pcmadepay, "src"), gst_element_get_pad (avimuxer, "audio_0")); But the result is the same. It creates the file but the file is empty. BTW, i had used gst_element_link_filtered instead of the above statements previously. - Zelalem S. Date: Tue, 23 Jun 2009 13:42:09 +0200 From: [hidden email] To: [hidden email] Subject: Re: [gst-devel] Problem with changing command line command to code request Sink pad of your avimux, link it to your audiobin. use gst-inspect "ELEMENT_TO_VIEW" to view your element pads, some of them aren't always available. some are dynamic. Hope that helps. On Tue, Jun 23, 2009 at 1:00 PM, Zelalem Sintayehu <[hidden email]> wrote:
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