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module-rtp: use sess.latency.msec also for sender

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Use the sess.latency.msec also for the sender and use it to control the
NODE_LATENCY. Make it a float to be in line with the other time values.
Set is to a default of ptime, which was what it used to be.

This makes it possible to set the ptime to a smaller value than the
sess.latency.msec so that we send out multiple packets per quantum.
This will result in some bursty output for now but with a timer that can
be improved later.

Update the docs a little, mention the new rtp.ptime and rtp.frametime.
This commit is contained in:
Wim Taymans 2024-01-25 15:42:16 +01:00
parent ec825086f1
commit c37f9f9cf0
4 changed files with 24 additions and 12 deletions
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9
src/modules/module-rtp-sink.c
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@ -52,9 +52,14 @@
* - `net.mtu = <int>`: MTU to use, default 1280
* - `net.ttl = <int>`: TTL to use, default 1
* - `net.loop = <bool>`: loopback multicast, default false
* - `sess.min-ptime = <int>`: minimum packet time in milliseconds, default 2
* - `sess.max-ptime = <int>`: maximum packet time in milliseconds, default 20
* - `sess.min-ptime = <float>`: minimum packet time in milliseconds, default 2
* - `sess.max-ptime = <float>`: maximum packet time in milliseconds, default 20
* - `sess.name = <str>`: a session name
* - `rtp.ptime = <float>`: size of the packets in milliseconds, default up to MTU but
* between sess.min-ptime and sess.max-ptime
* - `rtp.framecount = <int>`: number of samples per packet, default up to MTU but
* between sess.min-ptime and sess.max-ptime
* - `sess.latency.msec = <float>`: target node latency in milliseconds, default as rtp.ptime
* - `sess.ts-offset = <int>`: an offset to apply to the timestamp, default -1 = random offset
* - `sess.ts-refclk = <string>`: the name of a reference clock
* - `sess.media = <string>`: the media type audio|midi|opus, default audio