Dante/AES67-RTP-Stream to SBC: Latency
nicolas at ndufresne.ca
Tue Feb 18 14:28:46 UTC 2020
Le mar. 18 févr. 2020 09 h 15, CVX DEV <cvxdev at gmail.com> a écrit :
> Hi all,
> I am currently playing around with an AudioOverIP Project and wondered if
> you could help me out. I have a LAN, with an Audio Source
> (Dante/AES67-RTP-Stream) which I would like to distribute to multiple
> receivers (SBC (e.g. RaspberryPi) with an Audio Output (e.g. Headphone
> Source (e.g. PC)-->Dante-Audio-USB-Dongle-->AES67/RTP-Multicast-Stream-->LAN-Network-Switch-->RPI (Gstreamer --> AudioJack)
> I currently use the following Gstreamer Pipeline command on the RPi:
> gst-launch-1.0 -v udpsrc uri=udp://18.104.22.168:5004 caps="application/x-rtp,channels=(int)2,format=(string)S16LE,media=(string)audio,payload=(int)96,clock-rate=(int)48000,encoding-name=(string)L24" ! rtpL24depay ! audioconvert ! alsasink device=hw:0,0
> It all works fine, but if I watch a video on the PC and listen to the
> Audio on the RPI, I have some latency (~200-300ms), therefore my questions
Alsasink has 200ms latency as default configuration. See buffer-time
> 1. Do I miss something in my Gstreamer Pipeline to be able to reduce
> 2. What is the minimal Latency to be expected with RTP-Streams, is
> *<50ms* achievable?
> 3. Would the latency occur due to the network or due to the speed of
> the RPi?
> 4. Since my audio-input is not a Gstreamer input, I assume
> rtpjitterbuffer or similar would not help to decrease latency /
> improve sync?
> Here the SDP-extract from the Audio-Source:
> o=- 1484410 1484415 IN IP4 192.168.88.32
> s=avio : 2
> c=IN IP4 22.214.171.124/32
> t=0 0
> m=audio 5004 RTP/AVP 97
> i=2 channels: Left, Right
> a=rtpmap:97 L24/48000/2
> Thanks in advance for your feedback & help.
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
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