Official Opus support will be in the next version of Airtime. AAC+ support should be there too but it won't work out of the box since we can't legally ship the codec - however we will have detailed wiki for enabling it :)

Hello Martin,

On Wed, Mar 13, 2013 at 11:21 AM, Martin Konecny <<br />airtime-dev@lists.sourcefabric.org> wrote:

> Official Opus support will be in the next version of Airtime. AAC+ support
> should be there too but it won't work out of the box since we can't legally
> ship the codec.


Will that be 2.3.1 or 2.4?

If it's going to be a while, I may just build liquidsoap 1.1.0 and try to
hack ls_script.liq as the low-latency functionality provided by opus will
be really helpful for my use case.

Thanks,
-Bill

2.4.0 will be the first with Opus support. If you try compiling yourself,
there are instructions here:

https://wiki.sourcefabric.org/display/CC/Building+Liquidsoap+from+Source


On Wed, Mar 13, 2013 at 4:28 PM, Bill Burton <<br />airtime-dev@lists.sourcefabric.org> wrote:

> Hello Martin,
>
> On Wed, Mar 13, 2013 at 11:21 AM, Martin Konecny <<br />> airtime-dev@lists.sourcefabric.org> wrote:
>
> > Official Opus support will be in the next version of Airtime. AAC+
> support
> > should be there too but it won't work out of the box since we can't
> legally
> > ship the codec.
>
>
> Will that be 2.3.1 or 2.4?
>
> If it's going to be a while, I may just build liquidsoap 1.1.0 and try to
> hack ls_script.liq as the low-latency functionality provided by opus will
> be really helpful for my use case.
>
> Thanks,
> -Bill
>
>

It should be the same process as when you added the AAC+ encoder.

Where are all the awesome Opus mobile streaming apps hiding? The good folks at tunein.com managed to ignore Opus yet again in the latest and greatest TuneIn 3.6 for iOS. Major bummer. Then again, it doesn't even parse metadata on Vorbis streams. Weak sauce. Same true for Android??

Perhaps you can take the reference encoder and see if that performs as expected? If so, we can file a ticket for Liquidsoap.

It may be helpful to contact Liquidsoap on their mailing list and ask what
the defaults are. Or create a ticket on their github page asking them to
put this into their documention.

https://github.com/savonet/liquidsoap


On Thu, Jun 13, 2013 at 12:28 PM, Roger Wilco <<br />airtime-dev@lists.sourcefabric.org> wrote:

> There's an awful lot to tweak... what (or where) are the airtime defaults?
>
>
> - vbr: one of "none", "constrained" or "unconstrained"
> - application: One of "audio", "voip" or "restricted_lowdelay"
> - complexity: Integer value between 0 and 10.
> - max_bandwidth: One of "narrow_band", "medium_band", "wide_band",
> "super_wide_band" or "full_band"
> - samplerate: input samplerate. Must be one of: 8000, 12000, 16000,
> 24000 or 48000
> - frame_size: encoding frame size, in milliseconds. Must be one of: 2.5,
> 5., 10., 20., 40. or 60..
> - bitrate: encoding bitrate, in kbps. Must be a value between 5 and 512.
> You can also set it to "auto".
> - channels: currently, only 1 or 2 channels are allowed.
> - mono, stereo: equivalent to channels=1 and channels=2.
> - signal: one of "voice" or "music"
>
> http://savonet.sourceforge.net/doc-svn/encoding_formats.html
>
>

Yes exactly.


On Thu, Jun 13, 2013 at 3:00 PM, Roger Wilco <<br />airtime-dev@lists.sourcefabric.org> wrote:

> The only ones specifically set by Airtime would be Channels, Bitrate,
> right? Everything else would be "default" ?
>
>

Hello Albert,

On Mon, Jun 17, 2013 at 7:42 AM, Albert FR <<br />airtime-dev@lists.sourcefabric.org> wrote:

> Hi,
>
> Someone have found an app for streaming an opus stream ?


Depending on your use case, you may be able to use liquidsoap. This is my
plan to replace edcast for live programming but I need to test my script
more. In my case, Skype is connected to a phone conference service and
then liquidsoap would relay the audio from the loopback device into the
Airtime Master Source.

I'm also testing a relay from Airtime to sound card to send audio to a
Paltalk.com chat room and phone conference via Skype (listen only).

Right now I'm testing with Ogg Vorbis but as soon as I can upgrade to
Airtime 2.4, plan to use Opus for both inbound and outbound relays.

Because there's no GUI with liquidsoap, I'm using the Pira CZ Silence
Detector (www.pira.cz) to show volume levels.

In the liquidsoap source, there's at least one example that uses a Python
GUI to invoke liquidsoap (
https://github.com/savonet/liquidsoap/tree/master/gui).

Hope this helps,
-Bill

Opus is a good choice for low Latency Voice/Audio ...
but OGG Vorbis @ 48kbps (q=-0.1 )sounds slightly better for music.

Edcast is still the line-in upstream tool of choice... someone should add the user name field in stream conf.
http://code.google.com/p/edcast-reborn/

Just compare ... try Foobar with Kernel-Sound or ASIO output
http://stream.lmx.fm:8000/club.48.opus.m3u
http://stream.lmx.fm:8000/club.48.ogg.m3u
http://stream.lmx.fm:8000/club.48.aac.m3u

Encoder:
 %opus(samplerate=48000,bitrate=48,max_bandwidth="full_band",application="audio",signal="music")
 %vorbis(samplerate=48000,quality=-0.1)
 %aacplus(samplerate=48000,bitrate=48)

We use is mostly mp3 (@192kbps+) for our radio program ... rarely CD or Vinyl.
Currently we also compute the signal with lossy transcoding. (edcast,airtime,liquidsoap)
I intend to use netJack or similar for internal audio transport to reduce latency and loss.

For this aacplus stream i use Liquidsoap 1.0.1 the libaacplus-ocaml-dynlink .deb package ...

We are currently looking for a good codec for our Smartphone App.
If needed I can create even more streams with different codecs.

@Roger FDK added

Just compare ...
http://stream.lmx.fm:8000/club.48.opus.m3u
http://stream.lmx.fm:8000/club.48.ogg.m3u
http://stream.lmx.fm:8000/club.48.aac.m3u
http://stream.lmx.fm:8000/club.48.fdk.aac.m3u

Encoder:
 %opus(samplerate=48000,bitrate=48,max_bandwidth="full_band",application="audio",signal="music")
 %vorbis(samplerate=48000,quality=-0.1)
 %aacplus(samplerate=48000,bitrate=48)
 %fdkaac(bitrate=48, aot="mpeg4_he_aac_v2")