Livewire / AES67
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Fabien
Jul 11, 2018, 5:27:44 PM7/11/18
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Hello,
Has anyone been able to use odr with Livewire or AES67 audio streams?
Thanks
Have a nice day,
Fabien
Nick Piggott
Jul 12, 2018, 3:05:30 AM7/12/18
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Hi Fabien,
I've done this in a number of ODR setups.
If you look in the documentation for how to set up supervisor (http://wiki.opendigitalradio.org/RaspDAB/Automate_operation_with_Supervisor), under Special Note for Commands that start Child Processes, you'll see a sample command to extract linear audio, convert and encode it.
It works very well, although you'll need to use a high quality network card, and adjust IP routing on the machine to bridge between multicast and unicast networks.
Nick
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Fabien
Jul 12, 2018, 5:41:35 PM7/12/18
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Hi Nick,
Thanks for your answer.
If I understand correctly, I have to use rtpdump. How to download it? It is in this git https://github.com/columbia-irt/rtptools ?
Can you send me an Supervisor .conf file for example ? (my mail f...@bien.be)
Thanks
Best Regards,
Fabien
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Fabien
Jul 12, 2018, 6:05:26 PM7/12/18
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I just clone & install rtptools, and configured my supervisor encoder, but I have no sound
Can you tell me if my config is correct ?
[program:enc-test]
# DAB encoding using odr-audioenc
command=bash -c "rtpdump -F payload 239.192.3.248/5004 | sox -t raw -e signed-integer -r 48000 -c 2 -b 24 -B /dev/stdin -t raw --no-dither -r 48000 -c 2 -b 16 -L /dev/stdout gain 4 | odr-audioenc --format=raw --dab --bitrate=128 --dabmode=j --dabpsy=2 --pad=24 --pad-fifo=/home/dab/DAB/dls/test/test.pad --input=/dev/stdin --output=tcp://127.0.0.1:79001"
stopasgroup=true
autostart=true
autorestart=true
startretries=100
priority=10
stderr_logfile=/var/log/supervisor/enc-test.log
stdout_logfile=/var/log/supervisor/enc-test.log
and the log in supervisor return :
Welcome to ODR-AudioEnc v2.2.0-15-g60140af, compiled at Feb 9 2018, 20:10:39
Using 8 subchannels. AAC type: HE-AAC. channels=2, sample_rate=48000
AAC bitrate set to: 64000
Bandwidth is 10125
DAB+ Encoding: framelen=1920 (7680B)
Initialising VLC...
You are using VLC with size_t size callbacks
[0000000000f92dc8] pulse audio output error: PulseAudio server connection failure: Connection refused
Starting encoding
UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
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Fabien
Jul 12, 2018, 7:18:35 PM7/12/18
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I forgot to install SOX.
Now Supervisor no longer gives me an error for the encoder, but I still have no sound
Welcome to ODR-AudioEnc v2.2.0-15-g60140af, compiled at Feb 9 2018, 20:10:39
Using 8 subchannels. AAC type: HE-AAC. channels=2, sample_rate=48000
AAC bitrate set to: 64000
Bandwidth is 10125
DAB+ Encoding: framelen=1920 (7680B)
Nick Piggott
Jul 13, 2018, 6:13:07 AM7/13/18
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Hi,
You need to make sure that multicast IP is configured to route at the network layer.
Try running
rtpdump -F payload 239.192.3.248/5004 -o audio.wav
which will create an audio file called audio.wav. If the file size is 0 bytes, then no multicast is being received.
On most machines, there are two network cards:
* eth0 - connected to the normal LAN
* eth1 - connected to the Livewire dedicated LAN
You have to explicitly allow multicast to flow from eth1, otherwise it'll be ignored. I do this in the /etc/network/interfaces file
# Interface into LiveWire network
allow-hotplug eth1
iface eth1 inet static
address (a.b.c.d - the static address for this machine on the livewire network)
netmask 255.255.255.0
post-up route add -net 224.0.0.0 netmask 240.0.0.0 eth
The last line is important, as that's the multicast subnet routing.
Nick
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Fabien
Jul 13, 2018, 3:54:29 PM7/13/18
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Hi Nick,
Yes, it seems that it's a multicast problem.
I configured my Livewire network interface like you advised me :
auto lo
iface lo inet loopback
auto enp0s3 #Internet
iface enp0s3 inet static
address 10.30.10.61
netmask 255.255.255.0
gateway 10.30.10.1
dns-nameservers 10.30.10.1
auto enp0s8 #Livewire
iface enp0s8 inet static
address 10.30.40.105
netmask 255.255.255.0
post-up route add -net 224.0.0.0 netmask 240.0.0.0 ethifconfig
enp0s3 Link encap:Ethernet HWaddr 08:00:27:58:e4:83
inet adr:10.30.10.61 Bcast:10.30.10.255 Masque:255.255.255.0
adr inet6: fe80::a00:27ff:fe58:e483/64 Scope:Lien
adr inet6: 2a0a:800:50:10:a00:27ff:fe58:e483/64 Scope:Global
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Packets reçus:109809 erreurs:0 :3 overruns:0 frame:0
TX packets:68663 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 lg file transmission:1000
Octets reçus:91939319 (91.9 MB) Octets transmis:45196585 (45.1 MB)
enp0s8 Link encap:Ethernet HWaddr 08:00:27:d6:ec:f9
inet adr:10.30.40.105 Bcast:10.30.40.255 Masque:255.255.255.0
adr inet6: fe80::a00:27ff:fed6:ecf9/64 Scope:Lien
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Packets reçus:118 erreurs:0 :0 overruns:0 frame:0
TX packets:111 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 lg file transmission:1000
Octets reçus:11778 (11.7 KB) Octets transmis:12475 (12.4 KB)PS : interface's names are enp0x because Ubuntu is installed on a virtual machine via VirtualBox with two network cards in bridge mode
And, we can ping other Livewire devices on L2 network. But multicast doesn't seem to work because when I try "rtpdump -F payload 239.192.3.248/5004 -o audio.wav", my audio file stays at 0Bytes :-(
Do you have any idea ? :--)
Thanks a lot for your help,
Have a nice weekend,
Fabien
Fabien
Jul 13, 2018, 4:30:46 PM7/13/18
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Oh, apologize for my mistake.. : post-up route add -net 224.0.0.0 netmask 240.0.0.0 enp0s8
It works!!
thanks a lot ! :-)
Fabien
Jul 13, 2018, 5:05:47 PM7/13/18
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One last question: if I want to use multiple Livewire sources for multiple services, I can duplicate theses commands by just changing the multicast IP address ?
command=bash -c "rtpdump -F payload 239.192.1.1/5004 | sox -t raw -e signed-integer -r 48000 -c 2 -b 24 -B /dev/stdin -t raw --no-dither -r 48000 -c 2 -b 16 -L /dev/stdout gain 4 | odr-audioenc --format=raw --dab --bitrate=128 --dabmode=j --dabpsy=2 --pad=24 --pad-fifo=/home/dabsys/site/mot/sam.mot --input=/dev/stdin --output=tcp://127.0.0.1:9010" Because I just try with a second Livewire channel, and I have lot of audio drops in this second channel
Nick Piggott
Jul 15, 2018, 5:45:15 AM7/15/18
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Yes, but you really need a good network card for it to work.
I had problems pulling multiple channels on a cheap USB network adaptor. Running the same code on a machine with a better network card was more reliable. Also consider if you're overloading the USB bus, if the same machine is also streaming data from odr-dabmod to a modulator.
Fabien
Jul 15, 2018, 7:41:38 AM7/15/18
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Exact! With another network card it's ok :-) Thanks a lot !
Have a nice sunday,
Fabien
Nick Piggott
Jul 15, 2018, 2:52:02 PM7/15/18
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De rien!
Matthias Brändli
Jul 16, 2018, 1:24:35 AM7/16/18
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Hi Nick,
I'll add a scenario to the ODR-AudioEnc README that describes this
usage. Have you tried using the VLC input to receive the multicast stream?
Or did that fail for some reason?
Cheers
mpb
> <mailto:crc-mmbtools...@googlegroups.com>.
I'll add a scenario to the ODR-AudioEnc README that describes this
usage. Have you tried using the VLC input to receive the multicast stream?
Or did that fail for some reason?
Cheers
mpb
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Nick Piggott
Jul 16, 2018, 5:48:35 AM7/16/18
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Hi,
It's quite complicated to get VLC to pull Livewire/AE67 streams. The format can't be deduced by inspection of the packets, so if you try and use the rtp://@a.b.c.d format, it will fail...
> odr-audioenc --vlc-uri=rtp://@239.192.58.253 --dab --bitrate=128 --channels=2 --rate=48000 ==dabmode=J --output=tcp://127.0.0.1:9099
You are using VLC with size_t size callbacks
[00007fd7080046c8] rtp demux error: unspecified payload format (type 96)
[00007fd7080046c8] rtp demux: A valid SDP is needed to parse this RTP stream.
[00007fd7080046c8] core demux error: SDP required
[00007fd7080046c8] core demux error: A description in SDP format is required to receive the RTP stream. Note that rtp:// URIs cannot work with dynamic RTP payload format (96).
There's a utility on the Axia website to help create an SDP file
If you plug in the details of the multicast to this, it creates an SDP file
v=0
o=Node 1 1 IN IP4 10.100.1.1
s=DAB_Audio_Enc_Service
t=0 0
a=type:multicast
c=IN IP4 239.192.58.253
m=audio 5004 RTP/AVP 97
a=rtpmap:97 L24/48000/2
But then something doesn't work.
Initialising VLC...
You are using VLC with size_t size callbacks
Setting outbuf size to 4092
Starting encoding
Detected fault in input! No data in time.
The other reason I wanted to use rtpdump | sox | odr-audioenc is that I tend to believe that the re-sampler in sox is better than that in vlc, and I needed to bump an 8dB boost into the audio, as most livewire networks are configured for -23dBFS, which is too quiet for broadcast. (Resampling is a painful experience in vlc). You've got to shift the audio from 24 bit to 16 bit, although thankfully in most cases the sample rate (in this case 48kHz) doesn't need changing. (You may come across multicast setups using 44.1kHz as a master sample (nngggghhhhhh) and of course you may also have to re-sample to other sample rates in DAB/DAB+).
Nick
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Matthias Brändli
Jul 16, 2018, 6:13:13 AM7/16/18
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Hi,
thanks for the clarifictions. In the meantime, I found a discussion on
this list about generating the SDP file, and also added that to the README.
Maybe there's a way to get the SDP input to work too. Does it at least
work with "plain VLC" or with cvlc?
mpb
On 16/07/18 11:48, Nick Piggott wrote:
> Hi,
>
> It's quite complicated to get VLC to pull Livewire/AE67 streams. The
> format can't be deduced by inspection of the packets, so if you try and
> use the rtp://@a.b.c.d format, it will fail...
>
>> odr-audioenc --vlc-uri=rtp://@239.192.58.253 <http://239.192.58.253>
> <crc-mm...@googlegroups.com <mailto:crc-mm...@googlegroups.com>>
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> <mailto:crc-mmbtools%2Bunsu...@googlegroups.com>>.
thanks for the clarifictions. In the meantime, I found a discussion on
this list about generating the SDP file, and also added that to the README.
Maybe there's a way to get the SDP input to work too. Does it at least
work with "plain VLC" or with cvlc?
mpb
On 16/07/18 11:48, Nick Piggott wrote:
> Hi,
>
> It's quite complicated to get VLC to pull Livewire/AE67 streams. The
> format can't be deduced by inspection of the packets, so if you try and
> use the rtp://@a.b.c.d format, it will fail...
>
> --dab --bitrate=128 --channels=2 --rate=48000 ==dabmode=J
> --output=tcp://127.0.0.1:9099 <http://127.0.0.1:9099>
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Nick Piggott
Jul 16, 2018, 6:53:30 AM7/16/18
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Hi,
I noticed that the SDP Generator on the website is giving the wrong payload type. It should be 96, not 97.
v=0
o=Node 1 1 IN IP4 10.100.1.1
s=DAB_Audio_Enc_Service
t=0 0
a=type:multicast
c=IN IP4 239.192.58.253
m=audio 5004 RTP/AVP 96
a=rtpmap:96 L24/48000/2
cvlc will pull this audio fine, when routing to a raw demuxer:
cvlc file:///home/dabsys/file.sdp --no-video --sout file/raw:stream.wav
You can load this stream.wave file into an audio editor (specifying 24 bit audio, big endian bit order, 48kHz sample rate).
However, if you try and get this to happen in odr-audioenc, vlc seems to choke on the conversion...
[0000000001001f28] core input debug: Stream buffering done (1000 ms in 996 ms)
[0000000001001f28] core input debug: Decoder wait done in 0 ms
[00007f2c00000eb8] core encoder debug: looking for encoder module matching "any": 20 candidates
[00007f2c00000eb8] araw encoder debug: samplerate:48000Hz channels:2 bits/sample:32
[00007f2c00000eb8] core encoder debug: using encoder module "araw"
[00007f2c10001ed8] core stream out debug: input 's32l' 48000 Hz Stereo frame=1 samples/8 bytes
[00007f2c10001ed8] core stream out debug: conversion: 's32l'->'f32l' 48000 Hz->48000 Hz Stereo->Stereo
[00007f2c00011118] core audio converter debug: looking for audio converter module matching "any": 12 candidates
[00007f2c00011118] audio_format audio converter debug: s32l->f32l, bits per sample: 32->32
[00007f2c00011118] core audio converter debug: using audio converter module "audio_format"
[00007f2c10001ed8] core stream out debug: conversion pipeline complete
[00007f2c00011518] core audio resampler debug: looking for audio resampler module matching "any": 3 candidates
[00007f2c00011518] core audio resampler debug: using audio resampler module "samplerate"
Detected fault in input! No data in time.
It might be that there are additional parameters you can force into odr-audioenc, which in turn help libvlc understand what's going on?
Nick
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Matthias Brändli
Jul 25, 2018, 4:07:59 AM7/25/18
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You can use -L to pass more parameters to vlc, maybe there's a way to
get it working with that.
I'm afraid I can't be of much help, without a LiveWire signal available
here.
mpb
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get it working with that.
I'm afraid I can't be of much help, without a LiveWire signal available
here.
mpb
> wrote:
>
> Hi,
>
> thanks for the clarifictions. In the meantime, I found a discussion on
> this list about generating the SDP file, and also added that to the
> README.
>
> Maybe there's a way to get the SDP input to work too. Does it at least
> work with "plain VLC" or with cvlc?
>
> mpb
>
> On 16/07/18 11:48, Nick Piggott wrote:
> > Hi,
> >
> > It's quite complicated to get VLC to pull Livewire/AE67 streams. The
> > format can't be deduced by inspection of the packets, so if you
> try and
> > use the rtp://@a.b.c.d format, it will fail...
> >
> >> odr-audioenc --vlc-uri=rtp://@239.192.58.253
> <http://239.192.58.253> <http://239.192.58.253>
>
> Hi,
>
> thanks for the clarifictions. In the meantime, I found a discussion on
> this list about generating the SDP file, and also added that to the
> README.
>
> Maybe there's a way to get the SDP input to work too. Does it at least
> work with "plain VLC" or with cvlc?
>
> mpb
>
> On 16/07/18 11:48, Nick Piggott wrote:
> > Hi,
> >
> > It's quite complicated to get VLC to pull Livewire/AE67 streams. The
> > format can't be deduced by inspection of the packets, so if you
> try and
> > use the rtp://@a.b.c.d format, it will fail...
> >
> >> odr-audioenc --vlc-uri=rtp://@239.192.58.253
> <mailto:crc-mmbtools%252Buns...@googlegroups.com>>
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Fabien
Jul 25, 2018, 9:34:25 AM7/25/18
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Hi Matthias,
PS : IIf you want to test on my ODR virtual test machine in Teamviewer, with livewire sources. Do not hesitate to contact me
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Matthias Brändli
Jul 25, 2018, 10:20:33 AM7/25/18
to crc-mm...@googlegroups.com
Hi Fabien,
> PS : IIf you want to test on my ODR virtual test machine in Teamviewer,
> with livewire sources. Do not hesitate to contact me
Thanks for the proposal, but I don't have time to look into this.
However, if there's anything I can do to help you all from the community
to do tests and improve the tools, tell me.
As a general remark, I hope to one day be able to share the maintenance
responsibility of the mmbTools with someone else, because the low bus
factor is bad for the project.
https://en.wikipedia.org/wiki/Bus_factor
mpb
> PS : IIf you want to test on my ODR virtual test machine in Teamviewer,
> with livewire sources. Do not hesitate to contact me
However, if there's anything I can do to help you all from the community
to do tests and improve the tools, tell me.
As a general remark, I hope to one day be able to share the maintenance
responsibility of the mmbTools with someone else, because the low bus
factor is bad for the project.
https://en.wikipedia.org/wiki/Bus_factor
mpb
Fabien
Jul 25, 2018, 2:58:51 PM7/25/18
to mmbtools
Hi Matthias,
I understand.
That said, the technique with rtpdump|sox works great!
Not easy to establish a large community on such a precise domain
thanks for your work!
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