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Subject: [codec] I-D Action: draft-ietf-codec-results-03.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           : Summary of Opus listening test results
	Author(s)       : Christian Hoene
                          Jean-Marc Valin
                          Koen Vos
                          Jan Skoglund
	Filename        : draft-ietf-codec-results-03.txt
	Pages           : 23
	Date            : 2013-05-17

Abstract:
   This document describes and examines listening test results obtained
   for the Opus codec and how they relate to the requirements.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-codec-results-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-codec-results-03


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-oggopus-01.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           : Ogg Encapsulation for the Opus Audio Codec
	Author(s)       : Timothy B. Terriberry
                          Ron Lee
                          Ralph Giles
	Filename        : draft-ietf-codec-oggopus-01.txt
	Pages           : 24
	Date            : 2013-05-24

Abstract:
   This document defines the Ogg encapsulation for the Opus interactive
   speech and audio codec.  This allows data encoded in the Opus format
   to be stored in an Ogg logical bitstream.  Ogg encapsulation provides
   Opus with a long-term storage format supporting all of the essential
   features, including metadata, fast and accurate seeking, corruption
   detection, recapture after errors, low overhead, and the ability to
   multiplex Opus with other codecs (including video) with minimal
   buffering.  It also provides a live streamable format, capable of
   delivery over a reliable stream-oriented transport, without requiring
   all the data, or even the total length of the data, up-front, in a
   form that is identical to the on-disk storage format.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-codec-oggopus-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-codec-oggopus-01


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


From giles@thaumas.net  Fri May 24 08:45:35 2013
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Subject: [codec] draft-ietf-codec-oggopus-01
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I've updated the Ogg Opus encapsulation draft.

https://tools.ietf.org/html/draft-ietf-codec-oggopus-01

Changes since the -00 version:

- Surround speaker labels for channel mapping family 1 are described
  directly in the draft. They still match those used by the Vorbis
  codec, but it's better to have them explicit.

- The surround section now documents the stereo downmix matrices we're
  using in opus-tools and Firefox, as an aid to consistent
  implementation.

- An Implementation Status section has been added, with a link to
  https://wiki.xiph.org/OggOpusImplementation. Assistance updating
  and expanding this wiki page is welcome.

- The Content Type section now makes clear that the '.opus' filename
  extention maps to 'audio/ogg', and that any use of this spec
  for video or other file types using Ogg encapsulation is separate
  from that.

- Several small formatting changes and wording improvements.

This version does not include Jean-Marc's suggested encoder guidelines
for stream splicing, but we expect to incorporate them after further
discussion.

 -r



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Subject: Re: [codec] draft-ietf-codec-oggopus-01
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Ralph Giles wrote:
> This version does not include Jean-Marc's suggested encoder guidelines
> for stream splicing, but we expect to incorporate them after further
> discussion.

Yes, apologies for taking so long to get to reviewing them. I'll reply 
to the other thread.


From tterribe@xiph.org  Fri May 24 10:11:17 2013
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Subject: Re: [codec] I-D Action: draft-ietf-codec-oggopus-00.txt
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Jean-Marc Valin wrote:
> Here's a revised patch that describes guidelines for encoders. These
> guidelines are related to gapless support, but I'm sure there's other
> recommendations we should give encoder implementers. The general idea is
> to minimize the number of broken or suboptimal encoders.

(as an individual contributor)
I like this much better than your earlier proposal that required special 
handling in the decoder, since it doesn't require any changes to 
already-deployed decoders, and keeps the decoding process (relatively) 
simple.

A few minor points:
- The start of the Encoder Guidelines section should forward-reference 
the LPC Extrapolation section.
- Need a reference on how to actually compute the linear prediction 
coefficients. We have references to both the Burg method and the Schur 
algorithm in RFC 6716, and source code in libvorbis. Some implementation 
guidance wouldn't be amiss.
- The language in general needs a little word-smithing. We can 
coordinate on that in git.

From tterribe@xiph.org  Fri May 24 10:19:43 2013
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Subject: Re: [codec] I-D Action: draft-ietf-codec-oggopus-00.txt
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Jean-Marc Valin wrote:
>>> +<t>On the last frame of the first stream, encoding an independent frame by
>>> +turning off all forms of inter-frame prediction (de-emphasis is allowed).</t>
>>
>> How do people turn that off?
>
> That's actually something that needs to be cleanly exposed in the
> libopus API. The functionality is already there and you can probably get
> it to work by specifying 100% packet loss, but it needs to be a
> dedicated setting.

Well, I still think you need to explicitly list the SILK and CELT 
features that should be disabled (since, as you say, you don't want to 
refer to the libopus API), not just one feature that you shouldn't disable.

>>> +<t>setting the granulepos of the past page to a point near the end of the last
>>> +frame.</t>
>>
>> How do they determine where that point should be?
>
> I mean that any point near the end would work. Any thoughts on how to
> make this clearer?

I think you need to say more than this. In some contexts you may not be 
able to choose how many samples are in the file (encoding a CD with no 
inter-track gaps, but where you wish to preserve sample-accurate track 
boundaries, for example). That means you may need to do some frame size 
juggling to ensure the end of the track lies at "a point near the end of 
the last frame". How near is enough? The last half? The last quarter? In 
what size frame? Someone without intimate DSP knowledge and knowledge of 
the Opus format is not going to be able to make informed decisions here, 
so we should provide some guidance.

From oej@edvina.net  Thu May 30 00:14:20 2013
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From: "Olle E. Johansson" <oej@edvina.net>
Date: Thu, 30 May 2013 09:14:10 +0200
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Subject: [codec] Fwd: [asterisk-dev] Opus and VP8
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Please comment on this statement about Opus from Matt Jordan, the=20
project leader for the Asterisk project.

Regards,
/Olle

Vidarebefordrat brev:

> Fr=E5n: Matthew Jordan <mjordan@digium.com>
> =C4mne: Re: [asterisk-dev] Opus and VP8
> Datum: 29 maj 2013 20:55:58 CEST
> Till: asterisk-dev@lists.digium.com
> Svara till: Asterisk Developers Mailing List =
<asterisk-dev@lists.digium.com>
>=20
> On 05/25/2013 05:19 AM, Hans Witvliet wrote:
>> -----Original Message-----
>> From: Olle E. Johansson <oej@edvina.net>
>> Reply-to: Asterisk Developers Mailing List
>> <asterisk-dev@lists.digium.com>
>> To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com>
>> Cc: Olle E. Johansson <oej@edvina.net>
>> Subject: Re: [asterisk-dev] Opus and VP8
>> Date: Fri, 24 May 2013 13:26:29 +0200
>>=20
>>=20
>> 24 maj 2013 kl. 12:51 skrev Lorenzo Miniero <lminiero@gmail.com>:
>>=20
>>> PS: a few months ago I also talked, on the #asterisk-dev IRC, about
>>> the support I added for both Opus (transcoding) and VP8 =
(passthrough)
>>> in Asterisk, codecs that are currently the default ones used in
>>> WebRTC. I checked whether there was an interest in a patch for them,
>>> but at the time there were some concerns about the copyright status =
of
>>> Opus that prevented it to be considered for integration in Asterisk.
>>> Has this situation changed in the meanwhile? I can open a separate
>>> thread for this if needed.
>>>=20
>> Lorenzo,
>>=20
>>=20
>> Good seeing you here!
>>=20
>>=20
>> Due to legal issues I don't think Digium can accept a contribution of
>> Opus and VP8 in the svn repositories today.
>>=20
>>=20
>> I would encourage you, if you have these patches, to publish them on =
a
>> web site like github or sourceforge so w all can help you test it. I
>> really would like for these to be available for the community in an =
easy
>> form.
>>=20
>=20
> <snip>
>=20
> Hello! I'm going to comment here specifically to clarify Digium's
> position on Opus and VP8 as codecs and their inclusion in Asterisk.
>=20
> To start, pass through support in the form of a format module is fine
> for both Opus and VP8. It involves no transcoding and hence cannot
> violate any claims against their technology. We'd be happy to see =
format
> modules in Asterisk.
>=20
> VP8 is the easier of the two to clarify. A codec for VP8 is probably =
not
> appropriate, regardless of any patent or IPR issues. Asterisk doesn't
> perform video transcoding. Video transcoding is an intensive operation
> that performs poorly without hardware augmentation. We've always taken
> the stance that software video transcoding in Asterisk would cause =
more
> problems then it would solve; as such, VP8 as a codec is best left
> outside of Asterisk.
>=20
> The real question is: what about Opus?
>=20
> Before that, a word about the American patent system.
>=20
> The American patent system has devolved into what can only be =
charitably
> described as mafia-inspired extortion. Non-practicing entities (NPEs)
> are groups of lawyers who have not and never will produce, market, or
> sell a product. The only actions they perform are filing infringement
> claims against businesses and individuals, regardless of whether or =
not
> that business or individual actually violates a patent, with the sole
> purpose of extracting as much money out of said business or individual
> as they can. The cost of fighting these claims is enormous. The cost =
of
> losing a fight against even one of these claims is crippling. The NPEs
> know this. Technical merit, logic, rationale, or any kind of morality
> has no applicability here: these folks exist solely to find new and =
more
> creative ways to make claims against you and take your money.
>=20
> They'd be happy to put you out of business in the process.
>=20
> Back to Opus.
>=20
> There are several IPRs filed against Opus with the unfortunate =
licensing
> declaration of "Reasonable and Non-Discriminatory License to All
> Implementers with Possible Royalty/Fee." These IPRs have not been
> clarified, and the entities making these claims have not moved one way
> or the other regarding their claims. If any one of these entities
> decides to play the NPE game (see: Alcatel-Lucent), they could crush
> Digium like a bug. They could go after every user, integrator, and
> developer of Asterisk as well. It has the potential of spelling the =
end
> of the Asterisk project. The risk of this unfortunately does not =
justify
> the inclusion of Opus as a codec in Asterisk.
>=20
> Question: I am a user, integrator, and developer of Asterisk that does
> not work for Digium. Since Digium holds the copyright of Asterisk, how
> am I at risk?
>=20
> Answer: I have no idea. I do know that logic and reasoning does not
> apply where patents are concerned. Caveat emptor.
>=20
> Question: Asterisk is an open source project. Doesn't that protect me
> somehow?
>=20
> Answer: No. The GPLv2 specifically states "that any patent must be
> licensed for everyone's free use or not licensed at all". There are
> additional sections that further explain how patents affect software
> licensed under GPLv2; suffice to say that the sections exist to =
protect
> the freedom of the software; not to protect you from patent trolls.
>=20
> Question: If all of this is true, why does Google, Mozilla, Xiph.org,
> and others implement Opus?
>=20
> Answer: They either have an army of lawyers, are willing to roll the
> dice on their future, or are ignorant of how the patent system works.
>=20
> Question: This is messed up. If all of this is true, how can we ever
> innovate in areas where patents have ever been filed?
>=20
> Answer: You can't. The system is broken.
>=20
> Question: What can I do about it?
>=20
> Answer: Contact your government officials. Complain. The only way this
> situation will get fixed is if the laws are changed. Note that there =
is
> at least one bill being brought up in the U.S. Senate to address these
> exact deficiencies in the American patent system (and possibly more in
> the House); if you are a U.S. citizen I highly recommend you contact
> your elected Senators/Representatives and express your opinion(s).
>=20
> I hope this helps everyone understand why we've made our decision. We
> all hope that this situation changes in the near future, but until =
then,
> we'll have to limit our support of these codecs in Asterisk to
> pass-through only.
>=20
> Thanks
>=20
> Matt
>=20
> --=20
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>=20

