lien de téléchargement : [http://download.sound4.biz download.sound4.biz]

=2.0=
Compatible with new GUI (Need Server/Remote 4.0.0)

==2.0.29==
(2021-04-09)

*Fix PTP timestamp race condition, which can lead to reboot when traffic is too heavy

==2.0.28==
(2021-03-24)

*Can specify PTP Announce/Request Interval and Timeout
*Support IGMP v3 (Experimental)

==2.0.27==
(2021-02-23)

'''Livewire GPIO: reversed hardware level logic from previous version'''

*Revert the GPIO logic level for Livewire. For Axia, a Low level is active (and not high)

==2.0.26==
(2020-06-19)

*Corrected: AES67 Slave clock does not switch from a Master to another

==2.0.25==
(2019-12-02)

*Corrected: Default broadcast is not applied
*Correct scale so upgrading from old version does not remove AES67 input settings.

==2.0.24==
(2019-11-26)

*Corrected handling of 1-channel stream (copy left channel to right)
*Better PTP timestamping (rx timestamp pool empty)

==2.0.23==
(2019-10-10)

*Corrections for Dante compatibility (constant o= in SDP for reconnexion after reboot ; accept numbers in Session name)

==2.0.21==
(2019-04-30)

*Avoid to save volatile data in flash
*Other minor changes

==2.0.20==
(2019-03-28)

*Change DSP for Dante compatibility.
*Corrected SAP session name check
*Corrected Livewire GPO snake reconnect

==2.0.19==
(2018-11-29)

*Corrected mDNS and SAP advertising
*Corrected PTP internal sync
*Corrections for GPIO display in data description
*(cards) NEEDS Board 1.4.11

==2.0.18==
(2018-10-30)

*Manage Livewire GPIO.

==2.0.17==
(2018-10-22)

*SAP advertising
*Unicode corrections

==2.0.16==
(2018-10-01)

*Reduce log

==2.0.15==
(2018-09-27)

*Correct internal sync when Livewire master
*Show error when LANAUDIO Hostname is in conflict
*Force IGMP v2 for Cisco Catalyst
*Hostname 63 characters
*DNS-SD reply correction
*Don't allow Master when IP is not set
*Status bar should not show errors on channels out of license

==2.0.14==
(2018-06-22)

*License management correction for sharing stream (On Both L/R, Separated L/R)
*BUG: problems with URL, Hostname, ...

==2.0.13==
(2018-05-03)

*Realtime optimization
*Faster recovery after Ethernet replug
*Can now use ISO chars for Channel names

==2.0.12==

*Can deactivate MDNS advertising

==2.0.10==

*Improved Multicast management and packet drops (Debug version without mdnsd, rtsp and ptpmng)
*Increased watchdog timeout to 5s

==2.0.8==

*Corrected sync regul to be stable when sync packet is late or sync is lost
*Corrected memory leak

==2.0.7==

*Memory use optimization
*Potential race condition corrected
*Better Multicast packet management

==2.0.6==

*Livewire Live Audio: buffer size works
*Correct synchro loss on packet reordered
*NOTE: previous setting was doing nothing in real

==2.0.5==

*AES67: when adress is IP, can select format (so works with raw streams)
*NOTE: new internal browser

==2.0.4==

*Correct Livewire protocol extra double-quote after LABL

==2.0.3==

*Correct Livewire Payload (value can be incompatible with some Livewire parts).
*Was using the AES67 Custom Format to choose Payload number.
*BUG: Livewire protocol has LABL with double quote (LABL:"Ch2"" SHAB:0 FASM:1 BASM:1)

==2.0.2==

*Correct internal memory leaks.
*Advertising (SpiSkt) more robust.
*BUG: Livewire protocol has LABL with double quote (LABL:"Ch2"" SHAB:0 FASM:1 BASM:1)

==2.0.1==

*Compatible with new GUI (Need Server/Remote 4.0.0)

 
[[Category:Versions]]

2021-03-04T17:09:27Z

Marcel: /* IPv4 Multicast addresses */

[[Category:AES67]]
==Generalities==
These docs are useful :

AES67 Practical Guide (from [https://www.ravenna-network.com RAVENNA]) 
[[File:AES67 Practical Guide.pdf|thumb]] 

== Transport ==

=== IPv4 Multicast addresses ===
AES67 does not automatically attribute multicast address, it's under user's responsibility.

Multicast address range is : ''224.0.0.0/4''. The group includes the addresses from ''224.0.0.0'' to ''239.255.255.255 (cf https://en.wikipedia.org/wiki/Multicast_address<nowiki/>)''

For Audio over IP usage it's recommended to use 239.x.y.z range ("Administratively scoped, private use within an organization").

Actually some usual sub-ranges are :

*239.192.y.z => Livewire (Livestream and Standard Stereo Streams)
**239.192.255.1 to 4 are Livewire clock/advertising/GPIO
*239.193.y.z => Livewire (Back Standard Streams)
*239.195.y.z => Livewire (Back Live Streams)
*239.255.255.255 => SAP (advertisement) Dante

Never use 224.x.y.z range which is reserved for Network administration and control.

 

=====Windows=====
Under Windows, there may be difficulties with multicast routing when more than one Network Interfaces is used. 

===UDP Port===
Usually port 5004 is used for RTP (audio) and 5005 for RTCP (control) (cf http://www.networksorcery.com/enp/protocol/ip/ports05000.htm) but you may choose another number:

*RTP port must be even
*RTCP port = RTP port + 1
*don't use port under 1024 which are Unix "system ports"

 
===QOS===
 

== Clocking ==
AES67 uses PTP (''Precise Time Protocol'', IEEE1588v2-2008) as clock reference for sub-sample synchronization.

You have to configure some parameters, which are normally chosen according to a "PTP Profile" (1588, AES67, ST2110) and must be the same for all the audio devices of your network.

These parameters are critical for good and fast synchronization, especially when a Master Clock is disappearing and another one takes precedence.

Parameters :

* domain [0..255] : choose a common number, 0 is AES67 default, 127 is SMPTE ST2110 default
* priority1 and priority2 [0..255] : (for Master only) used to elect the clock Master. Lower values take precedence. priority1 is used first, then Clock Quality, then priority2. Default is 128,128.
* sync delay (in seconds) : (for Master only) mean delay between SYNC messages. Default is 0.125s
* announce interval (in seconds) : (for Master only) mean delay between ANNOUNCE messages. Default is 2s
* announce timeout (in packets) : number of announce interval that has to pass before timeout (which launch a Best Master election). Default is 3 packets.
* request interval (in seconds) : (for Slave only) mean delay between DELAY_REQ messages. Default is 1s

 
==Troubleshooting==
See [[LANAUDIO]]
[[Category:LANAUDIO]]

LANAUDIO

2021-03-01T15:08:11Z

Marcel:

Dante

2021-03-01T14:59:25Z

Marcel: /* Tools */

See [[Yamaha Tio1608-D|Yamaha Tio1608-D]] page for an example

==Configuration==
- AES67 clocking on both Dante and SOUND4

- select "SAP Advertising" on SOUND4

- Some DANTE devices accept only 1ms packets, L24 (L16 is accepted but does not work : noise). This is the "AES67 Low Latency (Standard)" audio Profile. You may try the "AES67 Very Low Latency" profile with 250µs packets.

- VLAN : if no VLAN, use default AES67 profiles : No 802.1 tagging. Else must change AES67 Profiles configuration

- Dante AES67 default Multicast Address Range = 239.69.x.y (69 prefix may be changed in Dante Controller config)

===For Dante to SOUND4===
Normally just select session from list.

Manual setting :

- on Dante source device you have to retrieve the Multicast IP address of the stream (something like 239.x.y.z). Also try to get the port, it is usually 5004

- on SOUND4 you have to select AES67, and write down the IP address in the "Session" field, with port if not 5004 (eg "239.193.4.68:5004")

- two new fields appear that has to be configured according to stream config : "Sample Format" (usually "L24") and "Nb Channels" (usually 2 ; if the stream has more than 2 only the first two will be used)

You should now receive it

===For SOUND4 to Dante :===
- on SOUND4, select "AES67 Low Latency (Standard)" and chose a Multicast IP address that is free, in the Dante AES67 Multicast Address Range (see above). Default port is 5004 if not given

- Check Stream format in Advanced/AES67 Profiles for "Sample Format" (normally L24) and "Nb Channels" (normally 2)

- stream should appear on Dante Controller, with a "+" drop-down to select channels. If no "+", SDP is not recognised.

- on Dante sink device you normally just have to configure this address and port if requested.

Note :

- Dante Controller only used for connection. Dante device is then "subscribed" to the stream (to the Mcast IP address) forever (no need Controller afterwards).
 

==Tools==
See [[LANAUDIO]]

====RAV2SAP====
Translate Ravenna mDNS <=> Dante SAP

May be used to diagnose advertisement problems (SDP SAP)

https://www.ravenna-network.com/aes67/rav2sap/
==Troubleshooting==
See [[LANAUDIO]]

====S4->Dante====
Advertised (checkmark under Dante Controller) but no sound (HP greyed) : check clock is AES67. If clock not synched, no sound on DANTE side.
[[Category:AES67]]
[[Category:LANAUDIO]]

AoIP

2021-03-01T14:54:30Z

Marcel: Redirected page to LANAUDIO

#REDIRECT [[LANAUDIO]]

AoIP

2021-03-01T14:54:17Z

Marcel: Created blank page

Audio Over IP

2021-03-01T14:53:41Z

Marcel: Redirected page to LANAUDIO

#REDIRECT [[LANAUDIO]]

Audio Over IP

2021-03-01T14:53:17Z

Marcel: Created blank page

Dante

2021-03-01T13:17:51Z

Marcel: /* For SOUND4 to Dante : */

See [[Yamaha Tio1608-D|Yamaha Tio1608-D]] page for an example

==Configuration==
- AES67 clocking on both Dante and SOUND4

- select "SAP Advertising" on SOUND4

- Some DANTE devices accept only 1ms packets, L24 (L16 is accepted but does not work : noise). This is the "AES67 Low Latency (Standard)" audio Profile. You may try the "AES67 Very Low Latency" profile with 250µs packets.

- VLAN : if no VLAN, use default AES67 profiles : No 802.1 tagging. Else must change AES67 Profiles configuration

- Dante AES67 default Multicast Address Range = 239.69.x.y (69 prefix may be changed in Dante Controller config)

===For Dante to SOUND4===
Normally just select session from list.

Manual setting :

- on Dante source device you have to retrieve the Multicast IP address of the stream (something like 239.x.y.z). Also try to get the port, it is usually 5004

- on SOUND4 you have to select AES67, and write down the IP address in the "Session" field, with port if not 5004 (eg "239.193.4.68:5004")

- two new fields appear that has to be configured according to stream config : "Sample Format" (usually "L24") and "Nb Channels" (usually 2 ; if the stream has more than 2 only the first two will be used)

You should now receive it

===For SOUND4 to Dante :===
- on SOUND4, select "AES67 Low Latency (Standard)" and chose a Multicast IP address that is free, in the Dante AES67 Multicast Address Range (see above). Default port is 5004 if not given

- Check Stream format in Advanced/AES67 Profiles for "Sample Format" (normally L24) and "Nb Channels" (normally 2)

- stream should appear on Dante Controller, with a "+" drop-down to select channels. If no "+", SDP is not recognised.

- on Dante sink device you normally just have to configure this address and port if requested.

Note :

- Dante Controller only used for connection. Dante device is then "subscribed" to the stream (to the Mcast IP address) forever (no need Controller afterwards).
 

==Tools==

===RAV2SAP===
Translate Ravenna mDNS <=> Dante SAP

May be used to diagnose advertisement problems (SDP SAP)

https://www.ravenna-network.com/aes67/rav2sap/
==Troubleshooting==
See [[LANAUDIO]]

====S4->Dante====
Advertised (checkmark under Dante Controller) but no sound (HP greyed) : check clock is AES67. If clock not synched, no sound on DANTE side.
[[Category:AES67]]
[[Category:LANAUDIO]]

RS232 and GPIO Connector

2021-02-23T17:15:33Z

Marcel:

The male DB25 connector is used for both RS232 and GPIO, providing 8 General Purpose Inputs and 8 General Purpose Outputs.

==Connector==
[[File:DB25.png|thumb]]
{| class="wikitable"
!Pin
!Function
!Direction
|-
|1
|GND
|Ground
|-
|2
|RS_TXD
|Out
|-
|3
|RS_RXD
|In
|-
|4
|RS_RTS
|Out
|-
|5
|RS_CTS
|In
|-
|6
|GPI8
|Opto isolated Input
|-
|7
|GND
|Ground
|-
|8
|GPI6
|Opto isolated Input
|-
|9
|GPI5
|Opto isolated Input
|-
|10
|GPI4
|Opto isolated Input
|-
|11
|GPI3
|Opto isolated Input
|-
|12
|GPI2
|Opto isolated Input
|-
|13
|GPI1
|Opto isolated Input
|-
|14
|GPOCOM
|Common GPO rail
|-
|15
|GPO1
|Solid State Relay
|-
|16
|GPO2
|Solid State Relay
|-
|17
|GPO3
|Solid State Relay
|-
|18
|GPO4
|Solid State Relay
|-
|19
|GPO5
|Solid State Relay
|-
|20
|GPO6
|Solid State Relay
|-
|21
|GPO7
|Solid State Relay
|-
|22
|GPI7
|Opto isolated Input
|-
|23
|GPO8
|Solid State Relay
|-
|24
|GPICOM
|Common GPI High-side rail
|-
|25
|GP5V
| +5V (Out), Fuse protected (100mA)
|}

==RS232==
The RS232 port is mainly used to feed the SOUND4 device with RDS UECP commands.

The DB25-Male GPIO+RS232 connector is compatible with the V.24 standard, "DTE" side, meaning that it's the same as a computer-side connector. Consequently when connecting a computer to the SOUND4 device, you have to use a Null Modem (cross cable).

Only the RXD/TXD signals are used for UECP, there is no handshaking : the CTS/RTS are not used

So if your PC COM is male SUBD-9, the connection must be at minimum :

*PC COM DB9M TXD (pin 3) must go to (pin 3) DB25M RS_RXD
*PC COM DB9M RXD (pin 2) must go to (pin 2) DB25M RS_TXD
*PC COM DB9M GND (pin 5) must go to (pin 1) DB25M GND

(see [https://en.wikipedia.org/wiki/Null_modem Null_modem wiring diagram])

(Optionally other V24 signals can be connected (CTS, RTS, CD, DSR, DTR) so a fully wired branded Null-Modem cable or adapter should work)

'''Notes :'''

*the RS232 output voltage is approx. ±6V. Full ±12V is accepted at input.
*the output RS_TXD pin is not active (so is 0V instead of +6V) until there is activity on RS_RXD pin.
*A validated cable reference is : https://fr.rs-online.com/web/p/products/1828825/

==Using GPI==
To activate one Input, GPI pin would be pulled to ground , with a voltage applied on the GPICOM pin (Common to all GPI).

Using external power supply is the recommended method in order to avoid possible ground loops between equipment, as shown in Figure 1-1. The maximum allowed external power supply for logic control is 48 volts DC.

'''NOTE''' the presence of Current Limiting Resistors per GPI pin. The intention is to limit the current to 20mA for each GPI pin. Use the table below to choose the suitable Resistor’s value.
[[File:GPI external power supply.png|center|570x570px|Sample usage of external power supply for GPI connection]]
If the equipment being controlled is electrically isolated, then the use of the GPIO port’s power supply is acceptable.
[[File:GPI processor power supply.png|center|437x437px]]
'''CAUTION''': The use of current limiting resistor per GPI pin is required for some voltages, see table (each input has an internal 330ohms protection).

'''NOT PROTECTING THE GPI COULD DAMAGE YOUR DEVICE.'''
{| class="wikitable"
|+Current limiting resistor value
!VDC
!External Resitor
|-
|5
|0
|-
|6
|0
|-
|12
|680 / 0.25 Watt
|-
|24
|1.8k / 0.5 Watt
|-
|48
|3.9k / 1 Watt
|}

==Using GPO==
The GPO portion of the GPIO port are Solid State Relays. Current should be limited to 100 mA per GPO pin of a port. Maximum allowed voltage is 48 volts. The following diagram shows the recommended connections for outputs with the use of an external power supply.
[[File:GPO external power supply.png|center|460x460px|Sample GPO connection using external power supply]]
If necessary, a Current Limiting Resistors must be used to limit the current to 100mA for each GPO pin.

'''NOT PROTECTING THE GPO COULD DAMAGE YOUR DEVICE.'''

If the device being controlled is electrically isolated, than the internal GP5V supply can be used, maintaining a 100mA limit on current drawn.
[[File:GPO processor power supply.png|center|465x465px|Sample GPO connection without external power supply]]
'''NOTE''': GPO pins and GPOCOM are not polarized, current can flow both directions.

==Internal connections of the GPIO port==
GPIO port provides 8 GPI (opto isolated inputs) and 8 GPO (solid state relays). Port is capable of driving a combined current of 100mA. Each GPI pin should be limited to 20mA of current.

Figure below shows a simplified diagram of the internal wiring behind the connector. The EMI Filters’ parts are omitted for the sake of simplicity.

All of the inputs and all of the outputs on the GPIO port are grouped together. The 8 GPOutputs are on 8 separate output pins, but they share the same “Common Return” connection GPOCOM on pin 14. Similarly, the 8 GPInput pins share one high-side rail GPICOM, connected to pin 24.
[[File:GPIO internal connections.png|center|thumb|619x619px|GPIO internal connections]]
 
[[Category:RS232]]
[[Category:RDS]]
[[Category:Connectors]]
[[Category:GPIO]]

2021-02-23T17:08:52Z

Marcel:

RDS Metada works now as in Streaming Extension (Serveur v4).

<code>Metadata source kind</code> can be:

*<code>File/URL</code>: In this case, use the Source field to set the file path or the URL. You may use User/Password and poll time for ftp and http URLs.
*<code>TCP Server</code>: will listen on the port set in "Source Port". It can optionally be bind to an IP, if the PC has multiple IP. This uses the PC/Admin IP, not the board.

Then, you need to select a parser.

Parsers are scripts in the [https://www.lua.org/ LUA language]. This is well documented on the Web.

The easier is to use XML. The script support to use [[wikipedia:XPath|XPath]] for XML parsing. This is also well documented on the Web and very flexible.

You need to put scripts in(see [[Files directories]]):

*Windows: <code>C:\ProgramData\SOUND4\Server\Data\metaparsers</code>
*Linux: <code>/opt/sound4/Data/rdsmetaparsers</code>
*Standalone: <code>\\rdsmetadataparser\</code>

A script called "MyParser.lua" should appear after in the "Metadata parsers" list as "MyParser".

For instance, to parse this XML file:<syntaxhighlight lang="xml">
<live>
<radio_id>Oldies Network</radio_id>
<current_song>
<Startime>17:53:14</Startime>
<artist><![CDATA[THE CHRISTIANS]]></artist>
<title><![CDATA[WHATS IN A WORD]]></title>
<duration>00:04:49</duration>
<cover_url>http://www.allcdcovers.com/image_system/images/f/4/f4efd22a515abcace470f093c26633fd.jpg</cover_url>
<categorie>G</categorie>
</current_song>
</live>
</syntaxhighlight>The MyParser.lua can be written like this:<syntaxhighlight lang="lua">
UseXmlParser()

function OnXMLParsed()
local artist=XPath('//current_song/artist/text()')
local title=XPath('//current_song/title/text()')
meta:Add("title", title)
meta:Add("artist", artist)
end
</syntaxhighlight>This script means:

<code>UseXmlParser()</code> ''Set the script in XML mode parsing''
<code>function OnXMLParsed()</code> ''Function called when XML has been parsed''

<code>local artist=XPath('//current_song/artist/text()')</code> ''Create a variable artist, using XPath to extract the artist content that is in <whatever><current_song><artist>name</...''

<code>local title=XPath('//current_song/title/text()')</code> ''Create a variable title, using XPath to extract the title content that is in <whatever><current_song><title>name</...''
<code>meta:Add("title", title)</code> ''Set the metadata field title to the value we got''

<code>meta:Add("artist", artist)</code> ''Set the metadata field artist to the value we got''

<code>end</code> ''End of function''

You can then use the metadata values in all Dynamic fields with <code>{artist}</code> and <code>{title}</code>.

For instance, set Dynamic RT: "RT Use Dynamic" = Yes, Mode= User and set "Dynamic RT"="<code>Now playing {title} by {artist}</code>"
[[Category:RDS]]