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Baresip using gst_video #36
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Hello Alfredh, |
Could you share some detail on your previous resolution and/or changes to gstvideo for others who might be interested before just extending this Issue with a new question? |
the cairo module is using libcairo to draw letters, please have a look .. aside from that, I think we should implement a very simple text-drawing function close this ticket for now. |
Hello Alfredh,
First of all, I want to say that you have done an amazing job with baresip. With the right pitch baresip could become very popular among new developers like myself.
So far I have managed to connect to another SIP client locally as well as through the stunserver. I have managed to establish both audio and video connection between the two remote UA. The config file for the above purpose is pasted below for others who need any reference. Now, what I am trying to do is create an appication wherein I receive video feed from the remote UA who have more than two webcam and display it on one display panel as merged.I have tried to use the gstreamer module (gst_video)to do the merge of the multiple video but I am not really able to get round it. I have the gst_video.so build successfully.(after adding lot of library dependencies )
I have a sample gstreamer code which does merging of video feed(as an example).
[code]
gst-launch -e videomixer name=mix ! ffmpegcolorspace ! xvimagesink
v4l2src device=/dev/video0 ! 'video/x-raw-yuv, framerate=24/1, width=320, height=240' !
videobox border-alpha=0 top=0 left=0 ! mix.
v4l2src device=/dev/video1 ! 'video/x-raw-yuv,width=320,height=240,framerate=30/1' !
videobox border-alpha=0 top=0 left=-320 ! mix.
[/code]
I want to use the above code in gst_video module to receive more than one cam feed from remote UA and display it on one output panel as merged(similar to ones used by security camera viewers). What I was trying to do so far was to edit the encode file (gst_video) and fit the gst-launch code(from above) into the gst_encoder_init function by replacing the appsrc part.On the config file I am using gst_video.so instead of avcodec.so Anyways, without much success with this approach I am looking for alternate solutions. I was wondering to change the rem-0.4.6/src/vidmix/vidmix.c file in some way to achieve my objective.I hope I was able to narrate my issue.
The config for establishing audio and video link
[code]
baresip configuration
------------------------------------------------------------------------------
Core
poll_method poll # poll, select
SIP
sip_trans_bsize 128
sip_listen 192.168..**:5060#Put your IP address;with the default port(5060)
sip_certificate cert.pem
Audio
audio_player alsa,default
audio_source alsa,default
audio_alert alsa,default
audio_srate 8000-48000
audio_channels 1-2
ausrc_srate 48000
auplay_srate 48000
ausrc_channels 0
auplay_channels 0
Video
video_source v4l2,/dev/video0
video_source fakevideo
video_display x11,nil
video_size 352x288
video_bitrate 500000
video_fps 25
AVT - Audio/Video Transport
rtp_tos 184
rtp_ports 10000-20000
rtp_bandwidth 512-1024 # [kbit/s]
rtcp_enable yes
rtcp_mux no
jitter_buffer_delay 5-10 # frames
rtp_stats no
Network
dns_server 10.0.0.1:53
net_interface eth0
BFCP
bfcp_proto udp
------------------------------------------------------------------------------
Modules
module_path .
UI Modules
module stdio.so
module cons.so
module evdev.so
module httpd.so
Audio codec Modules (in order)
module opus.so
module silk.so
module amr.so
module g7221.so
module g722.so
module g726.so
module g711.so
module gsm.so
module l16.so
module speex.so
module bv32.so
Audio filter Modules (in encoding order)
module vumeter.so
module sndfile.so
module speex_aec.so
module speex_pp.so
module plc.so
Audio driver Modules
module alsa.so
module portaudio.so
Video codec Modules (in order)
module gst_video.so
module avcodec.so
module vpx.so
Video filter Modules (in encoding order)
module selfview.so
Video source modules
module v4l.so
module v4l2.so
module avformat.so
module fakevideo.so
module x11grab.so
module cairo.so
Video display modules
module x11.so
module sdl2.so
Audio/Video source modules
module rst.so
module gst.so
Media NAT modules
module stun.so
module turn.so
module ice.so
module natpmp.so
Media encryption modules
module srtp.so
module dtls_srtp.so
------------------------------------------------------------------------------
Temporary Modules (loaded then unloaded)
module_tmp uuid.so
module_tmp account.so
------------------------------------------------------------------------------
Application Modules
module_app auloop.so
module_app contact.so
module_app menu.so
module_app mwi.so
module_app natbd.so
module_app presence.so
module_app syslog.so
module_app vidloop.so
------------------------------------------------------------------------------
Module parameters
cons_listen 0.0.0.0:5555
http_listen 0.0.0.0:8000
evdev_device /dev/input/event0
Speex codec parameters
speex_quality 7 # 0-10
speex_complexity 7 # 0-10
speex_enhancement 0 # 0-1
speex_vbr 0 # Variable Bit Rate 0-1
speex_vad 0 # Voice Activity Detection 0-1
speex_agc_level 8000
NAT Behavior Discovery
natbd_server creytiv.com
natbd_interval 600 # in seconds
Selfview
video_selfview window # {window,pip}
selfview_size 64x64
ICE
ice_turn no
ice_debug no
ice_nomination regular # {regular,aggressive}
ice_mode full # {full,lite}
[/code]
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