[webrtc] Add initial mmfw_webrtc.ini files

[Version] 0.3.2
[Issue Type] Add

Change-Id: I9f1738e413a71304163f9fc4cec89052a7755a05
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/media-config-simulator/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-simulator/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..917ab55
--- /dev/null
+++ b/media-config-simulator/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = v4l2src
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-artik10/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-artik10/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..c7eb8f9
--- /dev/null
+++ b/media-config-target-artik10/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = v4l2src
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-hawkp/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-hawkp/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..c7eb8f9
--- /dev/null
+++ b/media-config-target-hawkp/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = v4l2src
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-n4/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-n4/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..c7eb8f9
--- /dev/null
+++ b/media-config-target-n4/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = v4l2src
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-rpi3-spk/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-rpi3-spk/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..c7eb8f9
--- /dev/null
+++ b/media-config-target-rpi3-spk/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = v4l2src
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-rpi3/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-rpi3/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..c7eb8f9
--- /dev/null
+++ b/media-config-target-rpi3/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = v4l2src
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-tm1/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-tm1/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..e8608d8
--- /dev/null
+++ b/media-config-target-tm1/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = camerasrc
+; values below will override the default one of [media source] above
+video raw format = SN12
+video width = 640
+video height = 480
+video framerate = 30
+video codec = h264
+video hw encoder element = sprdenc_h264
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-tw1/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-tw1/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..819ac8d
--- /dev/null
+++ b/media-config-target-tw1/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-tw2/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-tw2/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..819ac8d
--- /dev/null
+++ b/media-config-target-tw2/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-tw3/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-tw3/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..819ac8d
--- /dev/null
+++ b/media-config-target-tw3/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/media-config-target-u3/hal/etc/multimedia/mmfw_webrtc.ini b/media-config-target-u3/hal/etc/multimedia/mmfw_webrtc.ini
new file mode 100644 (file)
index 0000000..819ac8d
--- /dev/null
+++ b/media-config-target-u3/hal/etc/multimedia/mmfw_webrtc.ini
@@ -0,0 +1,109 @@
+[general]
+; generating dot file representing pipeline state
+generate dot = no
+dot path = /tmp
+
+; | separated list of arguments that will pass to gst_init
+gstreamer arguments = --gst-debug=webrtcbin:7,3
+
+; comma separated list of elements that will not use in the gstreamer pipeline
+gstreamer excluded elements =
+
+; latency of RTP jitterbuffer
+rtp jitterbuffer latency = 100
+
+; FEC setting of RTP packets
+use ulpfec red = yes
+
+; default STUN server URL
+stun server = stun://stun.l.google.com:19302
+
+
+[media source]
+; default values for video source pipeline (e.g, videotest, camera, screen)
+video raw format = I420
+video width = 320
+video height = 240
+video framerate = 30
+video codec = vp8
+video hw encoder element =
+video drc support = no
+; default values for audio source pipeline (e.g, audiotest, mic)
+audio raw format = S16LE
+audio samplerate = 8000
+audio channels = 1
+audio codec = opus
+audio hw encoder element =
+
+
+[source videotest]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+video drc support = yes
+
+
+[source camera]
+source element = videotestsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source screen]
+source element = waylandsrc
+; values below will override the default one of [media source] above
+;video raw format =
+;video width =
+;video height =
+;video framerate =
+;video codec =
+;video hw encoder element =
+;video drc support =
+
+
+[source audiotest]
+source element = audiotestsrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[source mic]
+source element = pulsesrc
+; values below will override the default one of [media source] above
+;audio raw format =
+;audio samplerate =
+;audio channels =
+;audio codec =
+;audio hw encoder element =
+
+
+[rendering sink]
+; comma separated list of elements, it should be one by one per codec type
+audio hw decoder elements =
+video hw decoder elements =
+
+
+[vpxenc params]
+;threads =
+;end usage =
+;cpu used =
+;target bitrate =
+;keyframe max dist =
+;min quantizer =
+;max quantizer =
+;undershoot =

diff --git a/packaging/media-config.spec b/packaging/media-config.spec
index 1851ecec0272de0eeaaf69c31b34ba2441d9d35f..6534da54186b05a7f9f3fdfd59206d3074405601 100644 (file)
--- a/packaging/media-config.spec
+++ b/packaging/media-config.spec
@@ -1,6 +1,6 @@
 Name:       media-config
 Summary:    Multimedia Framework system configuration package
-Version:    0.3.1
+Version:    0.3.2
 Release:    0
 Group:      Multimedia/Configuration
 License:    LGPL-2.1 and Apache-2.0