
From lars@ibp.de  Tue Jun 12 14:42:31 2012
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Subject: [codec] draft-spittka-payload-rtp-opus-00: a=fmtp:101 application=audio
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Hi,

I have integrated the Opus codec into a SIP UA (based on pjsip). The =
intended application is for a broadcast (podcast) studio: a guest calls =
into the studio, talks (and maybe even sings) with the host, and the =
conversation is broadcast.

For this application, the highest available quality is desirable.=20

I have experimented with Opus in the appliation=3Daudio setting and am =
pretty happy with it. But reading draft-spittka-payload-rtp-opus-00, I =
find no provision to encourage the other party to send an Opus stream =
encoded with the "audio" application setting.

I'd like to propose an additional "application" parameter that can take =
the value "audio", "lowdelay" (and maybe "voip"):

In my example, the studio would offer (or answer with):

  m=3Daudio 54312 RTP/AVP 101
  a=3Drtpmap:101 opus/48000
  a=3Dfmtp:101 application=3Daudio

This would prompt the encoder on the other side to use the audio =
application mode of Opus.

I hope this makes sense. If it does, I'd be very pleased if it could be =
included in the next draft.=20

- Lars Immisch=

From jmvalin@jmvalin.ca  Tue Jun 12 15:03:48 2012
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Subject: Re: [codec] draft-spittka-payload-rtp-opus-00: a=fmtp:101 application=audio
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Hi,

The idea here is that the sender should know better than the receiver
the optimal way to encode the audio. This is why there's no application=
parameter.

Cheers,

	Jean-Marc

On 12-06-12 05:42 PM, Lars Immisch wrote:
> Hi,
> 
> I have integrated the Opus codec into a SIP UA (based on pjsip). The
> intended application is for a broadcast (podcast) studio: a guest
> calls into the studio, talks (and maybe even sings) with the host,
> and the conversation is broadcast.
> 
> For this application, the highest available quality is desirable.
> 
> I have experimented with Opus in the appliation=audio setting and am
> pretty happy with it. But reading draft-spittka-payload-rtp-opus-00,
> I find no provision to encourage the other party to send an Opus
> stream encoded with the "audio" application setting.
> 
> I'd like to propose an additional "application" parameter that can
> take the value "audio", "lowdelay" (and maybe "voip"):
> 
> In my example, the studio would offer (or answer with):
> 
> m=audio 54312 RTP/AVP 101 a=rtpmap:101 opus/48000 a=fmtp:101
> application=audio
> 
> This would prompt the encoder on the other side to use the audio
> application mode of Opus.
> 
> I hope this makes sense. If it does, I'd be very pleased if it could
> be included in the next draft.
> 
> - Lars Immisch _______________________________________________ codec
> mailing list codec@ietf.org 
> https://www.ietf.org/mailman/listinfo/codec
> 
> 
> 

From lars@ibp.de  Tue Jun 12 15:34:00 2012
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Hi,

> The idea here is that the sender should know better than the receiver
> the optimal way to encode the audio. This is why there's no =
application=3D
> parameter.

I understand that reasoning. A hi-fi music on hold implementation would =
just use application=3Daudio and send it unless the receiver asks for =
bandwidth restrictions.

But in our usecase (guest calls into broadcast studio), the receiver =
does know better. The recording will be broadcast and the receiver wants =
a way of saying: "Hey, send your best quality (because this will be =
broadcast)".

This addition would solve a real problem and create no interoperability =
concerns.

If you can think of other ways to solve this problem, I'd be pleased to =
hear them (I toyed with the idea of having an opus-audio codec, but =
dismissed it because application=3Daudio seemed a better fit, in =
particular because the hypothetical opus-audio and the opus codec can =
interoperate).

- Lars Immisch=

From jmvalin@mozilla.com  Tue Jun 12 17:38:13 2012
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On 06/12/2012 06:33 PM, Lars Immisch wrote:
> But in our usecase (guest calls into broadcast studio), the
> receiver does know better. The recording will be broadcast and the
> receiver wants a way of saying: "Hey, send your best quality
> (because this will be broadcast)".

Just saying "encode at high bit-rate" is enough to get the best
quality. Also, note that the "application" setting has nothing to do
with the Opus format. It's just an option in the current libopus API.
Other Opus implementation may not have that option at all, or may
expose the equivalent through a totally different API. This is why it
cannot be included in the SDP.

Cheers,

	Jean-Marc

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Hi,

>> But in our usecase (guest calls into broadcast studio), the
>> receiver does know better. The recording will be broadcast and the
>> receiver wants a way of saying: "Hey, send your best quality
>> (because this will be broadcast)".
>=20
> Just saying "encode at high bit-rate" is enough to get the best
> quality. Also, note that the "application" setting has nothing to do
> with the Opus format. It's just an option in the current libopus API.
> Other Opus implementation may not have that option at all, or may
> expose the equivalent through a totally different API. This is why it
> cannot be included in the SDP.

Well, the draft talks about "audio mode" and "voice mode" [1]. I should =
have suggested the parameter name "mode" and not "application" to stay =
within the draft's language.

But how can the receiver say: "encode at high bit-rate"? I don't see how =
that can be done.

a=3Dfmtp: 101 maxaveragebitrate=3D128000 (or any x > 32000) is a bit =
weak, because that's always true per Table 1.=20

What about maxaveragbitrate=3Dunlimited?

And a bit of explanatory text like this (modified from Section 7.2.1):

   o  The parameter "maxaveragebitrate" is a unidirectional receive-only
      parameter. The value "unlimited" indicates that the local receiver=20=

      wishes to receive payload data that was encoded with the audio =
mode=20
      of the codec. Other numerical values reflects limitations of the =
local=20
      receiver and the sender of the other side MUST NOT send with an =
average=20
      bit rate higher than "maxaveragebitrate" as it might overload the =
network
      and/or receiver.  The parameter "maxaveragebitrate" typically will
      not compromise interoperability; however, dependent on the set
      value of the parameter the performance of the application may
      suffer and should be set with care.

That would be good enough for the use case at hand, but it isn't easy on =
the reader of the draft. "mode=3Daudio" seems clearer and needs less =
text. But I don't care how it is done as long as the receiver can =
indicate to the sender that it wants the highest quality that the sender =
can produce.

- - Lars

[1] There is a typo in the draft: the section title in 3.1.1 should =
probably be "Voice mode"
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Cc: codec@ietf.org, Tim Pritlove <tim@pritlove.org>
Subject: Re: [codec] draft-spittka-payload-rtp-opus-00: a=fmtp:101 application=audio
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On 06/13/2012 06:52 PM, Lars Immisch wrote:
> Well, the draft talks about "audio mode" and "voice mode" [1]. I 
> should have suggested the parameter name "mode" and not
> "application" to stay within the draft's language.

Actually, the audio mode/voice mode will go away in the next revision
- -- it's just not needed considering that the sender knows better.

> But how can the receiver say: "encode at high bit-rate"? I don't
> see how that can be done.

You can do that with the b=AS: parameter

> a=fmtp: 101 maxaveragebitrate=128000 (or any x > 32000) is a bit 
> weak, because that's always true per Table 1.

First, table is just a rule of thumb for the recommended bandwidth
bandwidth in each more. It's not about defining maxaveragebitrate.
Also, we've realised that maxaveragebitrate needs to be a hard cap and
that it's not a good idea to use it for controlling the rate.

> What about maxaveragbitrate=unlimited?

Indeed, we'll make maxaveragbitrate *default* to unlimited in the next
draft. Controlling the bitrate should be done with the b= attribute.

	Jean-Marc
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Subject: Re: [codec] draft-spittka-payload-rtp-opus-00: a=fmtp:101 application=audio
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> On 06/13/2012 06:52 PM, Lars Immisch wrote:
>> Well, the draft talks about "audio mode" and "voice mode" [1]. I=20
>> should have suggested the parameter name "mode" and not
>> "application" to stay within the draft's language.
>=20
> Actually, the audio mode/voice mode will go away in the next revision
> - -- it's just not needed considering that the sender knows better.
>=20
>> But how can the receiver say: "encode at high bit-rate"? I don't
>> see how that can be done.
>=20
> You can do that with the b=3DAS: parameter

Ah, ok. RFC 3556

>> a=3Dfmtp: 101 maxaveragebitrate=3D128000 (or any x > 32000) is a bit=20=

>> weak, because that's always true per Table 1.
>=20
> First, table is just a rule of thumb for the recommended bandwidth
> bandwidth in each more. It's not about defining maxaveragebitrate.
> Also, we've realised that maxaveragebitrate needs to be a hard cap and
> that it's not a good idea to use it for controlling the rate.
>=20
>> What about maxaveragbitrate=3Dunlimited?
>=20
> Indeed, we'll make maxaveragbitrate *default* to unlimited in the next
> draft. Controlling the bitrate should be done with the b=3D attribute.

That sounds like it would address my concerns.

Thank you,

- - Lars Immisch
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Subject: [codec] I-D Action: draft-ietf-codec-opus-15.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Internet Wideband Audio Codec Working Gro=
up of the IETF.

	Title           : Definition of the Opus Audio Codec
	Author(s)       : Jean-Marc Valin
                          Koen Vos
                          Timothy B. Terriberry
	Filename        : draft-ietf-codec-opus-15.txt
	Pages           : 330
	Date            : 2012-06-18

Abstract:
   This document defines the Opus interactive speech and audio codec.
   Opus is designed to handle a wide range of interactive audio
   applications, including Voice over IP, videoconferencing, in-game
   chat, and even live, distributed music performances.  It scales from
   low bitrate narrowband speech at 6 kb/s to very high quality stereo
   music at 510 kb/s.  Opus uses both linear prediction (LP) and the
   Modified Discrete Cosine Transform (MDCT) to achieve good compression
   of both speech and music.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-codec-opus

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-codec-opus-15

A diff from previous version is available at:
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-codec-opus-15


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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Subject: [codec] I-D Action: draft-ietf-codec-opus-16.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Internet Wideband Audio Codec Working Gro=
up of the IETF.

	Title           : Definition of the Opus Audio Codec
	Author(s)       : Jean-Marc Valin
                          Koen Vos
                          Timothy B. Terriberry
	Filename        : draft-ietf-codec-opus-16.txt
	Pages           : 330
	Date            : 2012-06-28

Abstract:
   This document defines the Opus interactive speech and audio codec.
   Opus is designed to handle a wide range of interactive audio
   applications, including Voice over IP, videoconferencing, in-game
   chat, and even live, distributed music performances.  It scales from
   low bitrate narrowband speech at 6 kb/s to very high quality stereo
   music at 510 kb/s.  Opus uses both linear prediction (LP) and the
   Modified Discrete Cosine Transform (MDCT) to achieve good compression
   of both speech and music.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-codec-opus

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-codec-opus-16

A diff from previous version is available at:
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-codec-opus-16


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

