Dante/AES67-RTP-Stream to SBC: Latency

[Nicolas Dufresne]

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
> jack):
>
> 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://239.69.212.18:5004 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
property.


>    1. Do I miss something in my Gstreamer Pipeline to be able to reduce
>    latency?
>    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:
> v=0
> o=- 1484410 1484415 IN IP4 192.168.88.32
> s=avio : 2
> c=IN IP4 239.69.212.18/32
> t=0 0
> a=keywds:Dante
> m=audio 5004 RTP/AVP 97
> i=2 channels: Left, Right
> a=recvonly
> a=rtpmap:97 L24/48000/2
> a=ptime:1
> a=ts-refclk:ptp=IEEE1588-2008:00-1D-C1-FF-FE-50-BD-E3:0
> a=mediaclk:direct=161726307
>
> Thanks in advance for your feedback & help.
>
>
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