/usr/share/farstream/0.1/fsrtpconference/default-element-properties is in libfarstream-0.1-0 0.1.2-3.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 | # Put the desired properties in the style of
#
# [element name]
# prop1=val1
#
# Video codecs default to 256 kbps
#
# 100ms for the jitterbuffer is a good tradeoff
[gstrtpbin]
latency=100
[x264enc]
byte-stream=1
bitrate=256
profile=baseline
# Zerolatency is needed because the default latency is pretty high
tune=zerolatency
# 3 is "veryfast", presets go from 1 (ultrafast) to 10 (veryslow)
speed-preset=3
# These are all included in profile=baseline but stay here for older versions
bframes=0
b-adapt=0
cabac=0
dct8x8=0
# With zerolatency, threads are per slice, but slices confuse some decoders
threads=1
# Access-Unit Delimiters are a waste of bandwidth
aud=0
# Try to make GOBs as small as possible
[ffenc_h263]
rtp-payload-size=1
[theoraenc]
bitrate=256
[vp8enc]
bitrate=256000
max-latency=1
speed=2
error-resilient=true
[rtppcmupay]
ptime-multiple=20000000
[rtppcmapay]
ptime-multiple=20000000
# Set appropriate buffer/latency parameters for voip. The key parameter is
# buffer-time, which determines the latency in the conventional sense (X us of
# buffering between client and playback/capture. We take a conservatively high
# value for these to lower CPU load on less powerful systems.
[pulsesink]
latency-time=20000
buffer-time=60000
[pulsesrc]
latency-time=20000
buffer-time=40000
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