
From mknappe@juniper.net  Wed Jan 12 01:31:57 2011
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From: Michael Knappe <mknappe@juniper.net>
To: Christian Hoene <hoene@uni-tuebingen.de>, "codec@ietf.org" <codec@ietf.org>
Date: Wed, 12 Jan 2011 01:32:35 -0800
Thread-Topic: [codec] Testing Codec Quality
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Christian - WOW, what a fantastic document!

I encourage everyone to read this through and provide comments back to the
mailing list. In particular, check out section 10.2 which lists detailed
considerations for the proposed semi-formal testing with the 6+ volunteer
'test houses'. One of the issues we've been dealing with in our discussions
is how to divide up testing of even a simplified list of codec configuratio=
n
and operational permutations (Christian easily came up with 48 different
test cases) between the test houses. The plan is to have a very short list
of basic codec test cases that every test house repeats, along with a longe=
r
list of permutations that is divided among the test houses.


We do need to add a section outlining the proposed widescale informal
listening tests via a web based mushra app that will be done alongside the
semi-formal testing. Look for that app soon at www.open-codec.com, we're
working on getting that app up and running as we speak.

Thanks again to Christian for all the effort put into this doc.

Cheers,=20

Mike




On 12/3/10 2:12 AM, "Christian Hoene" <hoene@uni-tuebingen.de> wrote:

> Hi,
>=20
> one important question back to the days where iLBC was submitted to the I=
ETF
> was: how do we know whether iLBC is a technical sound contribution to the
> IETF?
>=20
> The draft http://www.ietf.org/id/draft-hoene-codec-quality-00.pdf address=
es
> this question in the context of this WG. The first chapters describe well
> established procedures on how to test a codec (and goes a bit beyond that
> where needed). Finally, Chapter 10 suggests a procedure on how to test II=
AC
> (or Opus).
>=20
> Timely feedback is welcomed as the testing shall start soon.
>=20
> With best regards,
>=20
>  Christian
>=20
>=20
> ---------------------------------------------------------------
> Dr.-Ing. Christian Hoene
> Interactive Communication Systems (ICS), University of T=FCbingen
> Sand 13, 72076 T=FCbingen, Germany, Phone +49 7071 2970532
> http://www.net.uni-tuebingen.de/
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From fluffy@cisco.com  Thu Jan 13 14:25:21 2011
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From: Cullen Jennings <fluffy@cisco.com>
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Subject: [codec] Next Steps for WG
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The OPUS draft authors believe the bitstream for OPUS will be ready to =
be "frozen" by the end of the month. At that point we plan to spend the =
following month testing and, baring any surprises, will likely go to =
WGLC after that.=20

There has been some questions about what "frozen" means in IETF context. =
The bitstream could change any time up to when the IESG approves it =
which is after Working Group Last call and after IETF Last Call so there =
is no guarantee that nothing will change but ...  by "frozen" we mean =
that we believe no changes are needed and don't plan to make changes =
unless some significant problem is found.

Thanks,
Cullen, Jonathan, and Mike



From roman@telurix.com  Thu Jan 13 16:32:40 2011
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From: Roman Shpount <roman@telurix.com>
To: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [codec] Next Steps for WG
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--90e6ba47682b9fd2b10499c39aef
Content-Type: text/plain; charset=ISO-8859-1

Just out of curiosity, have the OPUS codec been review against the list of
requirements that we put together?

In particular, I raised the requirement that multiple encoded streams form
different sources can be efficiently combined. Is it possible with OPUS? If
it is, then how?
_____________________________
Roman Shpount - www.telurix.com


On Thu, Jan 13, 2011 at 5:29 PM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> The OPUS draft authors believe the bitstream for OPUS will be ready to be
> "frozen" by the end of the month. At that point we plan to spend the
> following month testing and, baring any surprises, will likely go to WGLC
> after that.
>
> There has been some questions about what "frozen" means in IETF context.
> The bitstream could change any time up to when the IESG approves it which is
> after Working Group Last call and after IETF Last Call so there is no
> guarantee that nothing will change but ...  by "frozen" we mean that we
> believe no changes are needed and don't plan to make changes unless some
> significant problem is found.
>
> Thanks,
> Cullen, Jonathan, and Mike
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--90e6ba47682b9fd2b10499c39aef
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Just out of curiosity, have the OPUS codec been review against the list of =
requirements that we put together?<br><br>In particular, I raised the requi=
rement that multiple encoded streams form different sources can be efficien=
tly combined. Is it possible with OPUS? If it is, then how?<br clear=3D"all=
">
_____________________________<br>Roman Shpount - <a href=3D"http://www.telu=
rix.com">www.telurix.com</a><br>
<br><br><div class=3D"gmail_quote">On Thu, Jan 13, 2011 at 5:29 PM, Cullen =
Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@c=
isco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); p=
adding-left: 1ex;">
<br>
The OPUS draft authors believe the bitstream for OPUS will be ready to be &=
quot;frozen&quot; by the end of the month. At that point we plan to spend t=
he following month testing and, baring any surprises, will likely go to WGL=
C after that.<br>

<br>
There has been some questions about what &quot;frozen&quot; means in IETF c=
ontext. The bitstream could change any time up to when the IESG approves it=
 which is after Working Group Last call and after IETF Last Call so there i=
s no guarantee that nothing will change but ... =A0by &quot;frozen&quot; we=
 mean that we believe no changes are needed and don&#39;t plan to make chan=
ges unless some significant problem is found.<br>

<br>
Thanks,<br>
Cullen, Jonathan, and Mike<br>
<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</blockquote></div><br>

--90e6ba47682b9fd2b10499c39aef--

From jmvalin@jmvalin.ca  Thu Jan 13 18:48:34 2011
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Hi Roman,

We believe that we meet all requirements that were identified. As for 
mixing multiple encoded streams, that was marked as desirable in the 
draft because it's very hard to achieve without sacrificing efficiency. 
To some extent, if you constrain the Opus encoder to a subset of the 
available options (CELT only without short blocks or post-filter), then 
you can create streams that can be mixed at roughly half the normal 
complexity. I think that's about as much as we can do here.

Cheers,

	Jean-Marc

On 11-01-13 07:34 PM, Roman Shpount wrote:
> Just out of curiosity, have the OPUS codec been review against the list
> of requirements that we put together?
>
> In particular, I raised the requirement that multiple encoded streams
> form different sources can be efficiently combined. Is it possible with
> OPUS? If it is, then how?
> _____________________________
> Roman Shpount - www.telurix.com <http://www.telurix.com>
>
>
> On Thu, Jan 13, 2011 at 5:29 PM, Cullen Jennings <fluffy@cisco.com
> <mailto:fluffy@cisco.com>> wrote:
>
>
>     The OPUS draft authors believe the bitstream for OPUS will be ready
>     to be "frozen" by the end of the month. At that point we plan to
>     spend the following month testing and, baring any surprises, will
>     likely go to WGLC after that.
>
>     There has been some questions about what "frozen" means in IETF
>     context. The bitstream could change any time up to when the IESG
>     approves it which is after Working Group Last call and after IETF
>     Last Call so there is no guarantee that nothing will change but ...
>       by "frozen" we mean that we believe no changes are needed and
>     don't plan to make changes unless some significant problem is found.
>
>     Thanks,
>     Cullen, Jonathan, and Mike
>
>
>     _______________________________________________
>     codec mailing list
>     codec@ietf.org <mailto:codec@ietf.org>
>     https://www.ietf.org/mailman/listinfo/codec
>
>
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec

From hoene@uni-tuebingen.de  Fri Jan 14 00:24:41 2011
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: "'Jean-Marc Valin'" <jmvalin@jmvalin.ca>, "'Roman Shpount'" <roman@telurix.com>
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The easiest way to mix the multiple streams is to take advantage of the DTX
feature, which is available in the Silk and hybrid mode. However, I am not
aware whether DTX or VAD is supported in the CELT mode.

Christian

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Friday, January 14, 2011 3:51 AM
> To: Roman Shpount
> Cc: codec@ietf.org
> Subject: Re: [codec] Next Steps for WG
> 
> Hi Roman,
> 
> We believe that we meet all requirements that were identified. As for
mixing
> multiple encoded streams, that was marked as desirable in the draft
because
> it's very hard to achieve without sacrificing efficiency.
> To some extent, if you constrain the Opus encoder to a subset of the
> available options (CELT only without short blocks or post-filter), then
you can
> create streams that can be mixed at roughly half the normal complexity. I
> think that's about as much as we can do here.
> 
> Cheers,
> 
> 	Jean-Marc
> 
> On 11-01-13 07:34 PM, Roman Shpount wrote:
> > Just out of curiosity, have the OPUS codec been review against the
> > list of requirements that we put together?
> >
> > In particular, I raised the requirement that multiple encoded streams
> > form different sources can be efficiently combined. Is it possible
> > with OPUS? If it is, then how?
> > _____________________________
> > Roman Shpount - www.telurix.com <http://www.telurix.com>
> >
> >
> > On Thu, Jan 13, 2011 at 5:29 PM, Cullen Jennings <fluffy@cisco.com
> > <mailto:fluffy@cisco.com>> wrote:
> >
> >
> >     The OPUS draft authors believe the bitstream for OPUS will be ready
> >     to be "frozen" by the end of the month. At that point we plan to
> >     spend the following month testing and, baring any surprises, will
> >     likely go to WGLC after that.
> >
> >     There has been some questions about what "frozen" means in IETF
> >     context. The bitstream could change any time up to when the IESG
> >     approves it which is after Working Group Last call and after IETF
> >     Last Call so there is no guarantee that nothing will change but ...
> >       by "frozen" we mean that we believe no changes are needed and
> >     don't plan to make changes unless some significant problem is found.
> >
> >     Thanks,
> >     Cullen, Jonathan, and Mike
> >
> >
> >     _______________________________________________
> >     codec mailing list
> >     codec@ietf.org <mailto:codec@ietf.org>
> >     https://www.ietf.org/mailman/listinfo/codec
> >
> >
> >
> >
> > _______________________________________________
> > codec mailing list
> > codec@ietf.org
> > https://www.ietf.org/mailman/listinfo/codec
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From koen.vos@skype.net  Fri Jan 14 13:38:49 2011
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Date: Fri, 14 Jan 2011 22:41:10 +0100 (CET)
From: Koen Vos <koen.vos@skype.net>
To: Christian Hoene <hoene@uni-tuebingen.de>
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Subject: Re: [codec] Next Steps for WG
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Even without DTX, the Voice modes will expose a Voice Activity flag in the decoder API. This will reveal without any decoding whether a payload contains active speech.

And all decoder modes have low CPU demands, so decoding all streams to determine which ones are most active is relatively cheap.

koen.


----- Original Message -----
From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: "Jean-Marc Valin" <jmvalin@jmvalin.ca>, "Roman Shpount" <roman@telurix.com>
Cc: codec@ietf.org
Sent: Friday, January 14, 2011 12:27:02 AM
Subject: Re: [codec] Next Steps for WG

The easiest way to mix the multiple streams is to take advantage of the DTX
feature, which is available in the Silk and hybrid mode. However, I am not
aware whether DTX or VAD is supported in the CELT mode.

Christian

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Jean-Marc Valin
> Sent: Friday, January 14, 2011 3:51 AM
> To: Roman Shpount
> Cc: codec@ietf.org
> Subject: Re: [codec] Next Steps for WG
> 
> Hi Roman,
> 
> We believe that we meet all requirements that were identified. As for
mixing
> multiple encoded streams, that was marked as desirable in the draft
because
> it's very hard to achieve without sacrificing efficiency.
> To some extent, if you constrain the Opus encoder to a subset of the
> available options (CELT only without short blocks or post-filter), then
you can
> create streams that can be mixed at roughly half the normal complexity. I
> think that's about as much as we can do here.
> 
> Cheers,
> 
> 	Jean-Marc
> 
> On 11-01-13 07:34 PM, Roman Shpount wrote:
> > Just out of curiosity, have the OPUS codec been review against the
> > list of requirements that we put together?
> >
> > In particular, I raised the requirement that multiple encoded streams
> > form different sources can be efficiently combined. Is it possible
> > with OPUS? If it is, then how?
> > _____________________________
> > Roman Shpount - www.telurix.com <http://www.telurix.com>
> >
> >
> > On Thu, Jan 13, 2011 at 5:29 PM, Cullen Jennings <fluffy@cisco.com
> > <mailto:fluffy@cisco.com>> wrote:
> >
> >
> >     The OPUS draft authors believe the bitstream for OPUS will be ready
> >     to be "frozen" by the end of the month. At that point we plan to
> >     spend the following month testing and, baring any surprises, will
> >     likely go to WGLC after that.
> >
> >     There has been some questions about what "frozen" means in IETF
> >     context. The bitstream could change any time up to when the IESG
> >     approves it which is after Working Group Last call and after IETF
> >     Last Call so there is no guarantee that nothing will change but ...
> >       by "frozen" we mean that we believe no changes are needed and
> >     don't plan to make changes unless some significant problem is found.
> >
> >     Thanks,
> >     Cullen, Jonathan, and Mike
> >
> >
> >     _______________________________________________
> >     codec mailing list
> >     codec@ietf.org <mailto:codec@ietf.org>
> >     https://www.ietf.org/mailman/listinfo/codec
> >
> >
> >
> >
> > _______________________________________________
> > codec mailing list
> > codec@ietf.org
> > https://www.ietf.org/mailman/listinfo/codec
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec

_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec

From roman@telurix.com  Fri Jan 14 21:24:47 2011
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Date: Sat, 15 Jan 2011 00:27:07 -0500
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From: Roman Shpount <roman@telurix.com>
To: Koen Vos <koen.vos@skype.net>
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Subject: Re: [codec] Next Steps for WG
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--00221532cb384582af0499dbcda1
Content-Type: text/plain; charset=ISO-8859-1

My question was not as much about the VAD, as about multiple stream
combining. What I want is to implement is switching between multiple talker
based on VAD (computation of which is done by some external methods). In
this case we need to indicate to the remote party that this is a new stream
and ideally get the remote decoder in such a state that it can immediately
start correctly decoding the new audio without a glitch. So, the minimum
requirement would be to have a flag or a packet that indicates the decoder
that it needs to reset its state. Ideally, we need an algorithm to get the
proper decoder state based on some audio history stored on the conferencing
server, and to send this decoder state to the remote party so that it can be
synchronized. I don't think this should affect the codec performance. This
is just something that needs to be accommodated in the bitstream by adding a
packet type which either resets the decoder state or sets it to some
specified value.

I would also prefer if this was working across all the codec modes and did
not require the conferencing server from restricting what kinds of encoded
audio it can accept.
_____________
Roman Shpount


On Fri, Jan 14, 2011 at 4:41 PM, Koen Vos <koen.vos@skype.net> wrote:

> Even without DTX, the Voice modes will expose a Voice Activity flag in the
> decoder API. This will reveal without any decoding whether a payload
> contains active speech.
>
> And all decoder modes have low CPU demands, so decoding all streams to
> determine which ones are most active is relatively cheap.
>
> koen.
>
>
> ----- Original Message -----
> From: "Christian Hoene" <hoene@uni-tuebingen.de>
> To: "Jean-Marc Valin" <jmvalin@jmvalin.ca>, "Roman Shpount" <
> roman@telurix.com>
> Cc: codec@ietf.org
> Sent: Friday, January 14, 2011 12:27:02 AM
> Subject: Re: [codec] Next Steps for WG
>
> The easiest way to mix the multiple streams is to take advantage of the DTX
> feature, which is available in the Silk and hybrid mode. However, I am not
> aware whether DTX or VAD is supported in the CELT mode.
>
> Christian
>
> > -----Original Message-----
> > From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> > Of Jean-Marc Valin
> > Sent: Friday, January 14, 2011 3:51 AM
> > To: Roman Shpount
> > Cc: codec@ietf.org
> > Subject: Re: [codec] Next Steps for WG
> >
> > Hi Roman,
> >
> > We believe that we meet all requirements that were identified. As for
> mixing
> > multiple encoded streams, that was marked as desirable in the draft
> because
> > it's very hard to achieve without sacrificing efficiency.
> > To some extent, if you constrain the Opus encoder to a subset of the
> > available options (CELT only without short blocks or post-filter), then
> you can
> > create streams that can be mixed at roughly half the normal complexity. I
> > think that's about as much as we can do here.
> >
> > Cheers,
> >
> >       Jean-Marc
> >
> > On 11-01-13 07:34 PM, Roman Shpount wrote:
> > > Just out of curiosity, have the OPUS codec been review against the
> > > list of requirements that we put together?
> > >
> > > In particular, I raised the requirement that multiple encoded streams
> > > form different sources can be efficiently combined. Is it possible
> > > with OPUS? If it is, then how?
> > > _____________________________
> > > Roman Shpount - www.telurix.com <http://www.telurix.com>
> > >
> > >
> > > On Thu, Jan 13, 2011 at 5:29 PM, Cullen Jennings <fluffy@cisco.com
> > > <mailto:fluffy@cisco.com>> wrote:
> > >
> > >
> > >     The OPUS draft authors believe the bitstream for OPUS will be ready
> > >     to be "frozen" by the end of the month. At that point we plan to
> > >     spend the following month testing and, baring any surprises, will
> > >     likely go to WGLC after that.
> > >
> > >     There has been some questions about what "frozen" means in IETF
> > >     context. The bitstream could change any time up to when the IESG
> > >     approves it which is after Working Group Last call and after IETF
> > >     Last Call so there is no guarantee that nothing will change but ...
> > >       by "frozen" we mean that we believe no changes are needed and
> > >     don't plan to make changes unless some significant problem is
> found.
> > >
> > >     Thanks,
> > >     Cullen, Jonathan, and Mike
> > >
> > >
> > >     _______________________________________________
> > >     codec mailing list
> > >     codec@ietf.org <mailto:codec@ietf.org>
> > >     https://www.ietf.org/mailman/listinfo/codec
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > codec mailing list
> > > codec@ietf.org
> > > https://www.ietf.org/mailman/listinfo/codec
> > _______________________________________________
> > codec mailing list
> > codec@ietf.org
> > https://www.ietf.org/mailman/listinfo/codec
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--00221532cb384582af0499dbcda1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

My question was not as much about the VAD, as about multiple stream combini=
ng. What I want is to implement is switching between multiple talker based =
on VAD (computation of which is done by some external methods). In this cas=
e we need to indicate to the remote party that this is a new stream and ide=
ally get the remote decoder in such a state that it can immediately start c=
orrectly decoding the new audio without a glitch. So, the minimum requireme=
nt would be to have a flag or a packet that indicates the decoder that it n=
eeds to reset its state. Ideally, we need an algorithm to get the proper de=
coder state based on some audio history stored on the conferencing server, =
and to send this decoder state to the remote party so that it can be synchr=
onized. I don&#39;t think this should affect the codec performance. This is=
 just something that needs to be accommodated in the bitstream by adding a =
packet type which either resets the decoder state or sets it to some specif=
ied value.<br>
<br>I would also prefer if this was working across all the codec modes and =
did not require the conferencing server from restricting what kinds of enco=
ded audio it can accept.<br clear=3D"all">_____________<br>Roman Shpount<br=
>

<br><br><div class=3D"gmail_quote">On Fri, Jan 14, 2011 at 4:41 PM, Koen Vo=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:koen.vos@skype.net">koen.vos@skyp=
e.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); paddin=
g-left: 1ex;">
Even without DTX, the Voice modes will expose a Voice Activity flag in the =
decoder API. This will reveal without any decoding whether a payload contai=
ns active speech.<br>
<br>
And all decoder modes have low CPU demands, so decoding all streams to dete=
rmine which ones are most active is relatively cheap.<br>
<font color=3D"#888888"><br>
koen.<br>
</font><div><div></div><div class=3D"h5"><br>
<br>
----- Original Message -----<br>
From: &quot;Christian Hoene&quot; &lt;<a href=3D"mailto:hoene@uni-tuebingen=
.de">hoene@uni-tuebingen.de</a>&gt;<br>
To: &quot;Jean-Marc Valin&quot; &lt;<a href=3D"mailto:jmvalin@jmvalin.ca">j=
mvalin@jmvalin.ca</a>&gt;, &quot;Roman Shpount&quot; &lt;<a href=3D"mailto:=
roman@telurix.com">roman@telurix.com</a>&gt;<br>
Cc: <a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
Sent: Friday, January 14, 2011 12:27:02 AM<br>
Subject: Re: [codec] Next Steps for WG<br>
<br>
The easiest way to mix the multiple streams is to take advantage of the DTX=
<br>
feature, which is available in the Silk and hybrid mode. However, I am not<=
br>
aware whether DTX or VAD is supported in the CELT mode.<br>
<br>
Christian<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:codec-bounces@ietf.org">codec-bounces@ietf.org=
</a> [mailto:<a href=3D"mailto:codec-bounces@ietf.org">codec-bounces@ietf.o=
rg</a>] On Behalf<br>
&gt; Of Jean-Marc Valin<br>
&gt; Sent: Friday, January 14, 2011 3:51 AM<br>
&gt; To: Roman Shpount<br>
&gt; Cc: <a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
&gt; Subject: Re: [codec] Next Steps for WG<br>
&gt;<br>
&gt; Hi Roman,<br>
&gt;<br>
&gt; We believe that we meet all requirements that were identified. As for<=
br>
mixing<br>
&gt; multiple encoded streams, that was marked as desirable in the draft<br=
>
because<br>
&gt; it&#39;s very hard to achieve without sacrificing efficiency.<br>
&gt; To some extent, if you constrain the Opus encoder to a subset of the<b=
r>
&gt; available options (CELT only without short blocks or post-filter), the=
n<br>
you can<br>
&gt; create streams that can be mixed at roughly half the normal complexity=
. I<br>
&gt; think that&#39;s about as much as we can do here.<br>
&gt;<br>
&gt; Cheers,<br>
&gt;<br>
&gt; =A0 =A0 =A0 Jean-Marc<br>
&gt;<br>
&gt; On 11-01-13 07:34 PM, Roman Shpount wrote:<br>
&gt; &gt; Just out of curiosity, have the OPUS codec been review against th=
e<br>
&gt; &gt; list of requirements that we put together?<br>
&gt; &gt;<br>
&gt; &gt; In particular, I raised the requirement that multiple encoded str=
eams<br>
&gt; &gt; form different sources can be efficiently combined. Is it possibl=
e<br>
&gt; &gt; with OPUS? If it is, then how?<br>
&gt; &gt; _____________________________<br>
&gt; &gt; Roman Shpount - <a href=3D"http://www.telurix.com" target=3D"_bla=
nk">www.telurix.com</a> &lt;<a href=3D"http://www.telurix.com" target=3D"_b=
lank">http://www.telurix.com</a>&gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; On Thu, Jan 13, 2011 at 5:29 PM, Cullen Jennings &lt;<a href=3D"m=
ailto:fluffy@cisco.com">fluffy@cisco.com</a><br>
&gt; &gt; &lt;mailto:<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</=
a>&gt;&gt; wrote:<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; =A0 =A0 The OPUS draft authors believe the bitstream for OPUS wil=
l be ready<br>
&gt; &gt; =A0 =A0 to be &quot;frozen&quot; by the end of the month. At that=
 point we plan to<br>
&gt; &gt; =A0 =A0 spend the following month testing and, baring any surpris=
es, will<br>
&gt; &gt; =A0 =A0 likely go to WGLC after that.<br>
&gt; &gt;<br>
&gt; &gt; =A0 =A0 There has been some questions about what &quot;frozen&quo=
t; means in IETF<br>
&gt; &gt; =A0 =A0 context. The bitstream could change any time up to when t=
he IESG<br>
&gt; &gt; =A0 =A0 approves it which is after Working Group Last call and af=
ter IETF<br>
&gt; &gt; =A0 =A0 Last Call so there is no guarantee that nothing will chan=
ge but ...<br>
&gt; &gt; =A0 =A0 =A0 by &quot;frozen&quot; we mean that we believe no chan=
ges are needed and<br>
&gt; &gt; =A0 =A0 don&#39;t plan to make changes unless some significant pr=
oblem is found.<br>
&gt; &gt;<br>
&gt; &gt; =A0 =A0 Thanks,<br>
&gt; &gt; =A0 =A0 Cullen, Jonathan, and Mike<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; =A0 =A0 _______________________________________________<br>
&gt; &gt; =A0 =A0 codec mailing list<br>
&gt; &gt; =A0 =A0 <a href=3D"mailto:codec@ietf.org">codec@ietf.org</a> &lt;=
mailto:<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a>&gt;<br>
&gt; &gt; =A0 =A0 <a href=3D"https://www.ietf.org/mailman/listinfo/codec" t=
arget=3D"_blank">https://www.ietf.org/mailman/listinfo/codec</a><br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; codec mailing list<br>
&gt; &gt; <a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/codec</a><br>
&gt; _______________________________________________<br>
&gt; codec mailing list<br>
&gt; <a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_bla=
nk">https://www.ietf.org/mailman/listinfo/codec</a><br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>

--00221532cb384582af0499dbcda1--

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Subject: Re: [codec] Next Steps for WG
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On 01/14/2011 11:27 PM, Roman Shpount wrote:
> My question was not as much about the VAD, as about multiple stream
> combining. What I want is to implement is switching between multiple
> talker based on VAD (computation of which is done by some external
> methods). In this case we need to indicate to the remote party that this
> is a new stream and ideally get the remote decoder in such a state that
> it can immediately start correctly decoding the new audio without a
> glitch. So, the minimum requirement would be to have a flag or a packet
> that indicates the decoder that it needs to reset its state. Ideally, we
> need an algorithm to get the proper decoder state based on some audio
> history stored on the conferencing server, and to send this decoder
> state to the remote party so that it can be synchronized. I don't think
> this should affect the codec performance. This is just something that
> needs to be accommodated in the bitstream by adding a packet type which
> either resets the decoder state or sets it to some specified value.

This sort of mechanism already exists when using RTP as the transport 
mechanism, by setting the marker bit and changing the SSRC to indicate 
that the payload in the packet is from a different source than the 
previous packet. In my opinion there's no need for the codec bitstream 
to have any provisions for such an indication.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

From roman@telurix.com  Sat Jan 15 14:05:10 2011
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From: Roman Shpount <roman@telurix.com>
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--002215048dabcf65020499e9c684
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First of all, CODEC definition should be independent from the transport
protocol, and we might need this functionality when RTP SSRC are not
available.

Furthermore, in case of RTP, there are two problems with your suggestion:

1. Quite a few clients do not allow remote party to change SSRC without the
re-Invite. SIP clients note SSRC of the first received RTP packet and ignore
RTP packets with different SSRC

2. Even when the client allows switching of SSRC on the fly, a few initial
RTP packets are typically discarded due to RTP probation. After this,
clients typically need to pre-fill the jitter buffer, and only after this
start audio playback. This produces from 40ms to 100ms gap in audio. This is
very audible and highly undesirable.

Finally, what I was looking for was a bit more then just decoder reset.
Ideally I wanted to set decoder state to a known value to start decoding
audio packets that I will send to it.

P.S. As a side note, Intel Performance Primitives CODECs implement a packet
type in its wave file format that resets the decoder. This packet is used to
simplify implementation of performance and regression tests. I think they
are using standard sized packet with all bits set to zero for this purpose.
We can at least do something similar.

_____________
Roman Shpount


On Sat, Jan 15, 2011 at 8:12 AM, Kevin P. Fleming <kpfleming@digium.com>wrote:

> On 01/14/2011 11:27 PM, Roman Shpount wrote:
>
>> My question was not as much about the VAD, as about multiple stream
>> combining. What I want is to implement is switching between multiple
>> talker based on VAD (computation of which is done by some external
>> methods). In this case we need to indicate to the remote party that this
>> is a new stream and ideally get the remote decoder in such a state that
>> it can immediately start correctly decoding the new audio without a
>> glitch. So, the minimum requirement would be to have a flag or a packet
>> that indicates the decoder that it needs to reset its state. Ideally, we
>> need an algorithm to get the proper decoder state based on some audio
>> history stored on the conferencing server, and to send this decoder
>> state to the remote party so that it can be synchronized. I don't think
>> this should affect the codec performance. This is just something that
>> needs to be accommodated in the bitstream by adding a packet type which
>> either resets the decoder state or sets it to some specified value.
>>
>
> This sort of mechanism already exists when using RTP as the transport
> mechanism, by setting the marker bit and changing the SSRC to indicate that
> the payload in the packet is from a different source than the previous
> packet. In my opinion there's no need for the codec bitstream to have any
> provisions for such an indication.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--002215048dabcf65020499e9c684
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First of all, CODEC definition should be independent from the transport pro=
tocol, and we might need this functionality when RTP SSRC are not available=
.<br><br>Furthermore, in case of RTP, there are two problems with your sugg=
estion:<br>
<br>1. Quite a few clients do not allow remote party to change SSRC without=
 the re-Invite. SIP clients note SSRC of the first received RTP packet and =
ignore RTP packets with different SSRC<br><br>2. Even when the client allow=
s switching of SSRC on the fly, a few initial RTP packets are typically dis=
carded due to RTP probation. After this, clients typically need to pre-fill=
 the jitter buffer, and only after this start audio playback. This produces=
 from 40ms to 100ms gap in audio. This is very audible and highly undesirab=
le.<br>
<br>Finally, what I was looking for was a bit more then just decoder reset.=
 Ideally I wanted to set decoder state to a known value to start decoding a=
udio packets that I will send to it.<br><br>P.S. As a side note, Intel Perf=
ormance Primitives CODECs implement a packet type in its wave file format t=
hat resets the decoder. This packet is used to simplify implementation of p=
erformance and regression tests. I think they are using standard sized pack=
et with all bits set to zero for this purpose. We can at least do something=
 similar.<br>
<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Sat, Jan 15, 2011 at 8:12 AM, Kevin P=
. Fleming <span dir=3D"ltr">&lt;<a href=3D"mailto:kpfleming@digium.com">kpf=
leming@digium.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, =
204); padding-left: 1ex;">
<div class=3D"im">On 01/14/2011 11:27 PM, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
My question was not as much about the VAD, as about multiple stream<br>
combining. What I want is to implement is switching between multiple<br>
talker based on VAD (computation of which is done by some external<br>
methods). In this case we need to indicate to the remote party that this<br=
>
is a new stream and ideally get the remote decoder in such a state that<br>
it can immediately start correctly decoding the new audio without a<br>
glitch. So, the minimum requirement would be to have a flag or a packet<br>
that indicates the decoder that it needs to reset its state. Ideally, we<br=
>
need an algorithm to get the proper decoder state based on some audio<br>
history stored on the conferencing server, and to send this decoder<br>
state to the remote party so that it can be synchronized. I don&#39;t think=
<br>
this should affect the codec performance. This is just something that<br>
needs to be accommodated in the bitstream by adding a packet type which<br>
either resets the decoder state or sets it to some specified value.<br>
</blockquote>
<br></div>
This sort of mechanism already exists when using RTP as the transport mecha=
nism, by setting the marker bit and changing the SSRC to indicate that the =
payload in the packet is from a different source than the previous packet. =
In my opinion there&#39;s no need for the codec bitstream to have any provi=
sions for such an indication.<br>
<font color=3D"#888888">
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href=3D"mailto:kfleming@digium.com" target=3D=
"_blank">kfleming@digium.com</a><br>
Check us out at <a href=3D"http://www.digium.com" target=3D"_blank">www.dig=
ium.com</a> &amp; <a href=3D"http://www.asterisk.org" target=3D"_blank">www=
.asterisk.org</a></font><div><div></div><div class=3D"h5"><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>

--002215048dabcf65020499e9c684--

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Subject: Re: [codec] Next Steps for WG
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On 15 Jan 2011, at 22:07, Roman Shpount wrote:
> First of all, CODEC definition should be independent from the =
transport protocol, and we might need this functionality when RTP SSRC =
are not available.
>=20
> Furthermore, in case of RTP, there are two problems with your =
suggestion:
>=20
> 1. Quite a few clients do not allow remote party to change SSRC =
without the re-Invite. SIP clients note SSRC of the first received RTP =
packet and ignore RTP packets with different SSRC

If they do, then they're broken. RFC 3550 is very clear that the RTP =
SSRC can change at any time, due to SSRC collisions, changes in the =
transport address of the source, or source restarts.

> 2. Even when the client allows switching of SSRC on the fly, a few =
initial RTP packets are typically discarded due to RTP probation. After =
this, clients typically need to pre-fill the jitter buffer, and only =
after this start audio playback. This produces from 40ms to 100ms gap in =
audio. This is very audible and highly undesirable.

The two initial packets needed to establish an RTP stream as valid don't =
have to be discarded. They can be used to pre-fill the jitter buffer, =
and decoded once the stream is established as being valid. Only if the =
jitter buffer is smaller than two packets (the MIN_SEQUENTIAL parameter =
used to establish a source as valid) should a gap in the playout occur =
due to an SSRC change, and even then that gap should be possible to =
conceal.=20

Colin




> Finally, what I was looking for was a bit more then just decoder =
reset. Ideally I wanted to set decoder state to a known value to start =
decoding audio packets that I will send to it.
>=20
> P.S. As a side note, Intel Performance Primitives CODECs implement a =
packet type in its wave file format that resets the decoder. This packet =
is used to simplify implementation of performance and regression tests. =
I think they are using standard sized packet with all bits set to zero =
for this purpose. We can at least do something similar.
>=20
> _____________
> Roman Shpount
>=20
>=20
> On Sat, Jan 15, 2011 at 8:12 AM, Kevin P. Fleming =
<kpfleming@digium.com> wrote:
> On 01/14/2011 11:27 PM, Roman Shpount wrote:
> My question was not as much about the VAD, as about multiple stream
> combining. What I want is to implement is switching between multiple
> talker based on VAD (computation of which is done by some external
> methods). In this case we need to indicate to the remote party that =
this
> is a new stream and ideally get the remote decoder in such a state =
that
> it can immediately start correctly decoding the new audio without a
> glitch. So, the minimum requirement would be to have a flag or a =
packet
> that indicates the decoder that it needs to reset its state. Ideally, =
we
> need an algorithm to get the proper decoder state based on some audio
> history stored on the conferencing server, and to send this decoder
> state to the remote party so that it can be synchronized. I don't =
think
> this should affect the codec performance. This is just something that
> needs to be accommodated in the bitstream by adding a packet type =
which
> either resets the decoder state or sets it to some specified value.
>=20
> This sort of mechanism already exists when using RTP as the transport =
mechanism, by setting the marker bit and changing the SSRC to indicate =
that the payload in the packet is from a different source than the =
previous packet. In my opinion there's no need for the codec bitstream =
to have any provisions for such an indication.
>=20
> --=20
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming@digium.com
> Check us out at www.digium.com & www.asterisk.org
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec



--=20
Colin Perkins
http://csperkins.org/




--Apple-Mail-379--1003818200
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><div>On 15 Jan 2011, at 22:07, Roman Shpount =
wrote:</div><blockquote type=3D"cite">First of all, CODEC definition =
should be independent from the transport protocol, and we might need =
this functionality when RTP SSRC are not available.<br><br>Furthermore, =
in case of RTP, there are two problems with your suggestion:<br>
<br>1. Quite a few clients do not allow remote party to change SSRC =
without the re-Invite. SIP clients note SSRC of the first received RTP =
packet and ignore RTP packets with different =
SSRC<br></blockquote><div><br></div><div>If they do, then they're =
broken. RFC 3550 is very clear that the RTP SSRC can change at any time, =
due to SSRC collisions, changes in the transport address of the source, =
or source restarts.</div><br><blockquote type=3D"cite">2. Even when the =
client allows switching of SSRC on the fly, a few initial RTP packets =
are typically discarded due to RTP probation. After this, clients =
typically need to pre-fill the jitter buffer, and only after this start =
audio playback. This produces from 40ms to 100ms gap in audio. This is =
very audible and highly =
undesirable.<br></blockquote><div><br></div><div>The two initial packets =
needed to establish an RTP stream as valid don't have to be discarded. =
They can be used to pre-fill the jitter buffer, and decoded once the =
stream is established as being valid. Only if the jitter buffer is =
smaller than two packets (the MIN_SEQUENTIAL parameter used to establish =
a source as valid) should a gap in the playout occur due to an SSRC =
change, and even then that gap should be possible to =
conceal.&nbsp;</div><div><br></div><div>Colin</div><div><br></div><div><br=
></div><div><br></div><br><blockquote type=3D"cite">Finally, what I was =
looking for was a bit more then just decoder reset. Ideally I wanted to =
set decoder state to a known value to start decoding audio packets that =
I will send to it.<br><br>P.S. As a side note, Intel Performance =
Primitives CODECs implement a packet type in its wave file format that =
resets the decoder. This packet is used to simplify implementation of =
performance and regression tests. I think they are using standard sized =
packet with all bits set to zero for this purpose. We can at least do =
something similar.<br>
<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Sat, Jan 15, 2011 at 8:12 AM, =
Kevin P. Fleming <span dir=3D"ltr">&lt;<a =
href=3D"mailto:kpfleming@digium.com">kpfleming@digium.com</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt =
0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class=3D"im">On 01/14/2011 11:27 PM, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; =
border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
My question was not as much about the VAD, as about multiple stream<br>
combining. What I want is to implement is switching between multiple<br>
talker based on VAD (computation of which is done by some external<br>
methods). In this case we need to indicate to the remote party that =
this<br>
is a new stream and ideally get the remote decoder in such a state =
that<br>
it can immediately start correctly decoding the new audio without a<br>
glitch. So, the minimum requirement would be to have a flag or a =
packet<br>
that indicates the decoder that it needs to reset its state. Ideally, =
we<br>
need an algorithm to get the proper decoder state based on some =
audio<br>
history stored on the conferencing server, and to send this decoder<br>
state to the remote party so that it can be synchronized. I don't =
think<br>
this should affect the codec performance. This is just something =
that<br>
needs to be accommodated in the bitstream by adding a packet type =
which<br>
either resets the decoder state or sets it to some specified value.<br>
</blockquote>
<br></div>
This sort of mechanism already exists when using RTP as the transport =
mechanism, by setting the marker bit and changing the SSRC to indicate =
that the payload in the packet is from a different source than the =
previous packet. In my opinion there's no need for the codec bitstream =
to have any provisions for such an indication.<br>
<font color=3D"#888888">
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href=3D"mailto:kfleming@digium.com" =
target=3D"_blank">kfleming@digium.com</a><br>
Check us out at <a href=3D"http://www.digium.com/" =
target=3D"_blank">www.digium.com</a> &amp; <a =
href=3D"http://www.asterisk.org/" =
target=3D"_blank">www.asterisk.org</a></font><div><div></div><div =
class=3D"h5"><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>=

<a href=3D"https://www.ietf.org/mailman/listinfo/codec" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>
_______________________________________________<br>codec mailing =
list<br><a =
href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>https://www.ietf.org/=
mailman/listinfo/codec<br></blockquote></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div><br =
class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline"></span>
</div>
<br></body></html>=

--Apple-Mail-379--1003818200--

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Date: Tue, 18 Jan 2011 23:35:20 +0100 (CET)
From: Koen Vos <koen.vos@skype.net>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [codec] Next Steps for WG
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Roman: 

Jean-Marc and I discussed this a bit, and this is how we see it: 

Resetting the decoder state would lead to a discontinuity in the output signal and may create an objectionable "click" sound. Moreover, the incoming stream that's enabled in the switching node at the moment of the decoder reset may depend on the last frame before the switch. So the first few frames would not be decoded correctly. 

Coding and transmitting the decoder state doesn't overcome the discontinuity problem. And it would add a lot of complications and code to Opus. 

A better method seems to be: 
1. Determine if the first frame after switching has significant dependencies on the previous frame (i.e., long-term prediction in the case of SILK, or energy prediction in the case of CELT) 
2a: If yes: decode the first frame and re-encode it independently. After that, switch to the new stream. 
2b. If no: simply switch to the new stream immediately. 
This ensures a smooth transition and fast convergence to the correct output signal for the new stream. 

best, 
koen. 


----- Original Message -----
From: "Roman Shpount" <roman@telurix.com> 
To: "Kevin P. Fleming" <kpfleming@digium.com> 
Cc: codec@ietf.org 
Sent: Saturday, January 15, 2011 2:07:28 PM 
Subject: Re: [codec] Next Steps for WG 

First of all, CODEC definition should be independent from the transport protocol, and we might need this functionality when RTP SSRC are not available. 

Furthermore, in case of RTP, there are two problems with your suggestion: 

1. Quite a few clients do not allow remote party to change SSRC without the re-Invite. SIP clients note SSRC of the first received RTP packet and ignore RTP packets with different SSRC 

2. Even when the client allows switching of SSRC on the fly, a few initial RTP packets are typically discarded due to RTP probation. After this, clients typically need to pre-fill the jitter buffer, and only after this start audio playback. This produces from 40ms to 100ms gap in audio. This is very audible and highly undesirable. 

Finally, what I was looking for was a bit more then just decoder reset. Ideally I wanted to set decoder state to a known value to start decoding audio packets that I will send to it. 

P.S. As a side note, Intel Performance Primitives CODECs implement a packet type in its wave file format that resets the decoder. This packet is used to simplify implementation of performance and regression tests. I think they are using standard sized packet with all bits set to zero for this purpose. We can at least do something similar. 

_____________ 
Roman Shpount 



On Sat, Jan 15, 2011 at 8:12 AM, Kevin P. Fleming < kpfleming@digium.com > wrote: 



On 01/14/2011 11:27 PM, Roman Shpount wrote: 


My question was not as much about the VAD, as about multiple stream 
combining. What I want is to implement is switching between multiple 
talker based on VAD (computation of which is done by some external 
methods). In this case we need to indicate to the remote party that this 
is a new stream and ideally get the remote decoder in such a state that 
it can immediately start correctly decoding the new audio without a 
glitch. So, the minimum requirement would be to have a flag or a packet 
that indicates the decoder that it needs to reset its state. Ideally, we 
need an algorithm to get the proper decoder state based on some audio 
history stored on the conferencing server, and to send this decoder 
state to the remote party so that it can be synchronized. I don't think 
this should affect the codec performance. This is just something that 
needs to be accommodated in the bitstream by adding a packet type which 
either resets the decoder state or sets it to some specified value. 

This sort of mechanism already exists when using RTP as the transport mechanism, by setting the marker bit and changing the SSRC to indicate that the payload in the packet is from a different source than the previous packet. In my opinion there's no need for the codec bitstream to have any provisions for such an indication. 

-- 
Kevin P. Fleming 
Digium, Inc. | Director of Software Technologies 
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 
skype: kpfleming | jabber: kfleming@digium.com 
Check us out at www.digium.com & www.asterisk.org 



_______________________________________________ 
codec mailing list 
codec@ietf.org 
https://www.ietf.org/mailman/listinfo/codec 


_______________________________________________ 
codec mailing list 
codec@ietf.org 
https://www.ietf.org/mailman/listinfo/codec 

------=_Part_1112856_1656715763.1295390120404
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><head><style type=3D'text/css'>p { margin: 0; }</style></head><body><=
div style=3D'font-family: Times New Roman; font-size: 12pt; color: #000000'=
>Roman:<br><br>Jean-Marc and I discussed this a bit, and this is how we see=
 it:<br><br>Resetting the decoder state would lead to a discontinuity in th=
e output signal and may create an objectionable "click" sound.&nbsp; Moreov=
er, the incoming stream that's enabled in the switching node at the moment =
of the decoder reset may depend on the last frame before the switch. So the=
 first few frames would not be decoded correctly.<br><br>Coding and transmi=
tting the decoder state doesn't overcome the discontinuity problem.&nbsp; A=
nd it would add a lot of complications and code to Opus.&nbsp; <br><br>A be=
tter method seems to be:<br>1. Determine if the first frame after switching=
 has significant dependencies on the previous frame (i.e., long-term predic=
tion in the case of SILK, or energy prediction in the case of CELT)<br>2a: =
If yes: decode the first frame and re-encode it independently.&nbsp; After =
that, switch to the new stream.<br>
2b. If no: simply switch to the new stream immediately.<br>This ensures a s=
mooth transition and fast convergence to the correct output signal for the =
new stream.<br><br>best,<br>koen.<br><br><br><hr id=3D"zwchr"><b>From: </b>=
"Roman Shpount" &lt;roman@telurix.com&gt;<br><b>To: </b>"Kevin P. Fleming" =
&lt;kpfleming@digium.com&gt;<br><b>Cc: </b>codec@ietf.org<br><b>Sent: </b>S=
aturday, January 15, 2011 2:07:28 PM<br><b>Subject: </b>Re: [codec] Next St=
eps for WG<br><br>First of all, CODEC definition should be independent from=
 the transport protocol, and we might need this functionality when RTP SSRC=
 are not available.<br><br>Furthermore, in case of RTP, there are two probl=
ems with your suggestion:<br>
<br>1. Quite a few clients do not allow remote party to change SSRC without=
 the re-Invite. SIP clients note SSRC of the first received RTP packet and =
ignore RTP packets with different SSRC<br><br>2. Even when the client allow=
s switching of SSRC on the fly, a few initial RTP packets are typically dis=
carded due to RTP probation. After this, clients typically need to pre-fill=
 the jitter buffer, and only after this start audio playback. This produces=
 from 40ms to 100ms gap in audio. This is very audible and highly undesirab=
le.<br>
<br>Finally, what I was looking for was a bit more then just decoder reset.=
 Ideally I wanted to set decoder state to a known value to start decoding a=
udio packets that I will send to it.<br><br>P.S. As a side note, Intel Perf=
ormance Primitives CODECs implement a packet type in its wave file format t=
hat resets the decoder. This packet is used to simplify implementation of p=
erformance and regression tests. I think they are using standard sized pack=
et with all bits set to zero for this purpose. We can at least do something=
 similar.<br>
<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Sat, Jan 15, 2011 at 8:12 AM, Kevin P=
. Fleming <span dir=3D"ltr">&lt;<a href=3D"mailto:kpfleming@digium.com" tar=
get=3D"_blank">kpfleming@digium.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px so=
lid rgb(204, 204, 204); padding-left: 1ex;">
<div class=3D"im">On 01/14/2011 11:27 PM, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
My question was not as much about the VAD, as about multiple stream<br>
combining. What I want is to implement is switching between multiple<br>
talker based on VAD (computation of which is done by some external<br>
methods). In this case we need to indicate to the remote party that this<br=
>
is a new stream and ideally get the remote decoder in such a state that<br>
it can immediately start correctly decoding the new audio without a<br>
glitch. So, the minimum requirement would be to have a flag or a packet<br>
that indicates the decoder that it needs to reset its state. Ideally, we<br=
>
need an algorithm to get the proper decoder state based on some audio<br>
history stored on the conferencing server, and to send this decoder<br>
state to the remote party so that it can be synchronized. I don't think<br>
this should affect the codec performance. This is just something that<br>
needs to be accommodated in the bitstream by adding a packet type which<br>
either resets the decoder state or sets it to some specified value.<br>
</blockquote>
<br></div>
This sort of mechanism already exists when using RTP as the transport mecha=
nism, by setting the marker bit and changing the SSRC to indicate that the =
payload in the packet is from a different source than the previous packet. =
In my opinion there's no need for the codec bitstream to have any provision=
s for such an indication.<br>
<font color=3D"#888888">
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href=3D"mailto:kfleming@digium.com" target=3D=
"_blank">kfleming@digium.com</a><br>
Check us out at <a href=3D"http://www.digium.com" target=3D"_blank">www.dig=
ium.com</a> &amp; <a href=3D"http://www.asterisk.org" target=3D"_blank">www=
.asterisk.org</a></font><div><div></div><div class=3D"h5"><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>
<br>_______________________________________________<br>codec mailing list<b=
r>codec@ietf.org<br>https://www.ietf.org/mailman/listinfo/codec<br></div></=
body></html>
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From roman@telurix.com  Tue Jan 18 14:52:37 2011
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Date: Tue, 18 Jan 2011 17:55:09 -0500
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From: Roman Shpount <roman@telurix.com>
To: Koen Vos <koen.vos@skype.net>
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--00221532c58cdf480e049a26cabe
Content-Type: text/plain; charset=ISO-8859-1

Thank you, this is very helpful.

What I was considering was something similar: Typically you have one speaker
in the conference with short periods of double talk. I was thinking that
decoding, mixing and encoding using independent frames the double talk
period with the few initial frames of the new speaker, and then switching to
a new speaker stream would sound the most natural and will only require
re-encoding for short periods of time.

Thank you for your great work,
_____________________________
Roman Shpount - www.telurix.com


On Tue, Jan 18, 2011 at 5:35 PM, Koen Vos <koen.vos@skype.net> wrote:

> Roman:
>
> Jean-Marc and I discussed this a bit, and this is how we see it:
>
> Resetting the decoder state would lead to a discontinuity in the output
> signal and may create an objectionable "click" sound.  Moreover, the
> incoming stream that's enabled in the switching node at the moment of the
> decoder reset may depend on the last frame before the switch. So the first
> few frames would not be decoded correctly.
>
> Coding and transmitting the decoder state doesn't overcome the
> discontinuity problem.  And it would add a lot of complications and code to
> Opus.
>
> A better method seems to be:
> 1. Determine if the first frame after switching has significant
> dependencies on the previous frame (i.e., long-term prediction in the case
> of SILK, or energy prediction in the case of CELT)
> 2a: If yes: decode the first frame and re-encode it independently.  After
> that, switch to the new stream.
> 2b. If no: simply switch to the new stream immediately.
> This ensures a smooth transition and fast convergence to the correct output
> signal for the new stream.
>
> best,
> koen.
>
>
> ------------------------------
> *From: *"Roman Shpount" <roman@telurix.com>
> *To: *"Kevin P. Fleming" <kpfleming@digium.com>
> *Cc: *codec@ietf.org
> *Sent: *Saturday, January 15, 2011 2:07:28 PM
>
> *Subject: *Re: [codec] Next Steps for WG
>
> First of all, CODEC definition should be independent from the transport
> protocol, and we might need this functionality when RTP SSRC are not
> available.
>
> Furthermore, in case of RTP, there are two problems with your suggestion:
>
> 1. Quite a few clients do not allow remote party to change SSRC without the
> re-Invite. SIP clients note SSRC of the first received RTP packet and ignore
> RTP packets with different SSRC
>
> 2. Even when the client allows switching of SSRC on the fly, a few initial
> RTP packets are typically discarded due to RTP probation. After this,
> clients typically need to pre-fill the jitter buffer, and only after this
> start audio playback. This produces from 40ms to 100ms gap in audio. This is
> very audible and highly undesirable.
>
> Finally, what I was looking for was a bit more then just decoder reset.
> Ideally I wanted to set decoder state to a known value to start decoding
> audio packets that I will send to it.
>
> P.S. As a side note, Intel Performance Primitives CODECs implement a packet
> type in its wave file format that resets the decoder. This packet is used to
> simplify implementation of performance and regression tests. I think they
> are using standard sized packet with all bits set to zero for this purpose.
> We can at least do something similar.
>
> _____________
> Roman Shpount
>
>
> On Sat, Jan 15, 2011 at 8:12 AM, Kevin P. Fleming <kpfleming@digium.com>wrote:
>
>> On 01/14/2011 11:27 PM, Roman Shpount wrote:
>>
>>> My question was not as much about the VAD, as about multiple stream
>>> combining. What I want is to implement is switching between multiple
>>> talker based on VAD (computation of which is done by some external
>>> methods). In this case we need to indicate to the remote party that this
>>> is a new stream and ideally get the remote decoder in such a state that
>>> it can immediately start correctly decoding the new audio without a
>>> glitch. So, the minimum requirement would be to have a flag or a packet
>>> that indicates the decoder that it needs to reset its state. Ideally, we
>>> need an algorithm to get the proper decoder state based on some audio
>>> history stored on the conferencing server, and to send this decoder
>>> state to the remote party so that it can be synchronized. I don't think
>>> this should affect the codec performance. This is just something that
>>> needs to be accommodated in the bitstream by adding a packet type which
>>> either resets the decoder state or sets it to some specified value.
>>>
>>
>> This sort of mechanism already exists when using RTP as the transport
>> mechanism, by setting the marker bit and changing the SSRC to indicate that
>> the payload in the packet is from a different source than the previous
>> packet. In my opinion there's no need for the codec bitstream to have any
>> provisions for such an indication.
>>
>> --
>> Kevin P. Fleming
>> Digium, Inc. | Director of Software Technologies
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> skype: kpfleming | jabber: kfleming@digium.com
>> Check us out at www.digium.com & www.asterisk.org
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--00221532c58cdf480e049a26cabe
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Thank you, this is very helpful. <br><br>What I was considering was somethi=
ng similar: Typically you have one speaker in the conference with short per=
iods of double talk. I was thinking that decoding, mixing and encoding usin=
g independent frames the double talk period with the few initial frames of =
the new speaker, and then switching to a new speaker stream would sound the=
 most natural and will only require re-encoding for short periods of time.<=
br>
<br>Thank you for your great work,<br clear=3D"all">_______________________=
______<br>Roman Shpount - <a href=3D"http://www.telurix.com">www.telurix.co=
m</a><br>
<br><br><div class=3D"gmail_quote">On Tue, Jan 18, 2011 at 5:35 PM, Koen Vo=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:koen.vos@skype.net">koen.vos@skyp=
e.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); paddin=
g-left: 1ex;">
<div><div style=3D"font-family: Times New Roman; font-size: 12pt; color: rg=
b(0, 0, 0);">Roman:<br><br>Jean-Marc and I discussed this a bit, and this i=
s how we see it:<br><br>Resetting the decoder state would lead to a discont=
inuity in the output signal and may create an objectionable &quot;click&quo=
t; sound.=A0 Moreover, the incoming stream that&#39;s enabled in the switch=
ing node at the moment of the decoder reset may depend on the last frame be=
fore the switch. So the first few frames would not be decoded correctly.<br=
>
<br>Coding and transmitting the decoder state doesn&#39;t overcome the disc=
ontinuity problem.=A0 And it would add a lot of complications and code to O=
pus.=A0 <br><br>A better method seems to be:<br>1. Determine if the first f=
rame after switching has significant dependencies on the previous frame (i.=
e., long-term prediction in the case of SILK, or energy prediction in the c=
ase of CELT)<br>
2a: If yes: decode the first frame and re-encode it independently.=A0 After=
 that, switch to the new stream.<br>
2b. If no: simply switch to the new stream immediately.<br>This ensures a s=
mooth transition and fast convergence to the correct output signal for the =
new stream.<br><br>best,<br>koen.<br><br><br><hr><b>From: </b>&quot;Roman S=
hpount&quot; &lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">rom=
an@telurix.com</a>&gt;<br>
<b>To: </b>&quot;Kevin P. Fleming&quot; &lt;<a href=3D"mailto:kpfleming@dig=
ium.com" target=3D"_blank">kpfleming@digium.com</a>&gt;<br><b>Cc: </b><a hr=
ef=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br><b>Sen=
t: </b>Saturday, January 15, 2011 2:07:28 PM<div class=3D"im">
<br><b>Subject: </b>Re: [codec] Next Steps for WG<br><br></div><div><div></=
div><div class=3D"h5">First of all, CODEC definition should be independent =
from the transport protocol, and we might need this functionality when RTP =
SSRC are not available.<br>
<br>Furthermore, in case of RTP, there are two problems with your suggestio=
n:<br>
<br>1. Quite a few clients do not allow remote party to change SSRC without=
 the re-Invite. SIP clients note SSRC of the first received RTP packet and =
ignore RTP packets with different SSRC<br><br>2. Even when the client allow=
s switching of SSRC on the fly, a few initial RTP packets are typically dis=
carded due to RTP probation. After this, clients typically need to pre-fill=
 the jitter buffer, and only after this start audio playback. This produces=
 from 40ms to 100ms gap in audio. This is very audible and highly undesirab=
le.<br>

<br>Finally, what I was looking for was a bit more then just decoder reset.=
 Ideally I wanted to set decoder state to a known value to start decoding a=
udio packets that I will send to it.<br><br>P.S. As a side note, Intel Perf=
ormance Primitives CODECs implement a packet type in its wave file format t=
hat resets the decoder. This packet is used to simplify implementation of p=
erformance and regression tests. I think they are using standard sized pack=
et with all bits set to zero for this purpose. We can at least do something=
 similar.<br>

<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Sat, Jan 15, 2011 at 8:12 AM, Kevin P=
. Fleming <span dir=3D"ltr">&lt;<a href=3D"mailto:kpfleming@digium.com" tar=
get=3D"_blank">kpfleming@digium.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px so=
lid rgb(204, 204, 204); padding-left: 1ex;">

<div>On 01/14/2011 11:27 PM, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
My question was not as much about the VAD, as about multiple stream<br>
combining. What I want is to implement is switching between multiple<br>
talker based on VAD (computation of which is done by some external<br>
methods). In this case we need to indicate to the remote party that this<br=
>
is a new stream and ideally get the remote decoder in such a state that<br>
it can immediately start correctly decoding the new audio without a<br>
glitch. So, the minimum requirement would be to have a flag or a packet<br>
that indicates the decoder that it needs to reset its state. Ideally, we<br=
>
need an algorithm to get the proper decoder state based on some audio<br>
history stored on the conferencing server, and to send this decoder<br>
state to the remote party so that it can be synchronized. I don&#39;t think=
<br>
this should affect the codec performance. This is just something that<br>
needs to be accommodated in the bitstream by adding a packet type which<br>
either resets the decoder state or sets it to some specified value.<br>
</blockquote>
<br></div>
This sort of mechanism already exists when using RTP as the transport mecha=
nism, by setting the marker bit and changing the SSRC to indicate that the =
payload in the packet is from a different source than the previous packet. =
In my opinion there&#39;s no need for the codec bitstream to have any provi=
sions for such an indication.<br>

<font color=3D"#888888">
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href=3D"mailto:kfleming@digium.com" target=3D=
"_blank">kfleming@digium.com</a><br>
Check us out at <a href=3D"http://www.digium.com" target=3D"_blank">www.dig=
ium.com</a> &amp; <a href=3D"http://www.asterisk.org" target=3D"_blank">www=
.asterisk.org</a></font><div><div></div><div><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>
<br>_______________________________________________<br>codec mailing list<b=
r><a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br=
><a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></div></div></blockquote></div><br>

--00221532c58cdf480e049a26cabe--

From koen.vos@skype.net  Tue Jan 18 15:11:22 2011
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From: Koen Vos <koen.vos@skype.net>
To: Roman Shpount <roman@telurix.com>
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That should work. 
Make sure to get the time alignment right by adjusting for look-ahead etc. 
And only the first frame of the double talk period has to be coded independently; for the others you could let the encoder decide. In general, whenever you're adding or removing a stream from the mix you want an independent frame. 

best, 
koen. 


----- Original Message -----
From: "Roman Shpount" <roman@telurix.com> 
To: "Koen Vos" <koen.vos@skype.net> 
Cc: codec@ietf.org 
Sent: Tuesday, January 18, 2011 2:55:09 PM 
Subject: Re: [codec] Next Steps for WG 

Thank you, this is very helpful. 

What I was considering was something similar: Typically you have one speaker in the conference with short periods of double talk. I was thinking that decoding, mixing and encoding using independent frames the double talk period with the few initial frames of the new speaker, and then switching to a new speaker stream would sound the most natural and will only require re-encoding for short periods of time. 

Thank you for your great work, 
_____________________________ 
Roman Shpount - www.telurix.com 



On Tue, Jan 18, 2011 at 5:35 PM, Koen Vos < koen.vos@skype.net > wrote: 




Roman: 

Jean-Marc and I discussed this a bit, and this is how we see it: 

Resetting the decoder state would lead to a discontinuity in the output signal and may create an objectionable "click" sound. Moreover, the incoming stream that's enabled in the switching node at the moment of the decoder reset may depend on the last frame before the switch. So the first few frames would not be decoded correctly. 

Coding and transmitting the decoder state doesn't overcome the discontinuity problem. And it would add a lot of complications and code to Opus. 

A better method seems to be: 
1. Determine if the first frame after switching has significant dependencies on the previous frame (i.e., long-term prediction in the case of SILK, or energy prediction in the case of CELT) 
2a: If yes: decode the first frame and re-encode it independently. After that, switch to the new stream. 
2b. If no: simply switch to the new stream immediately. 
This ensures a smooth transition and fast convergence to the correct output signal for the new stream. 

best, 
koen. 



From: "Roman Shpount" < roman@telurix.com > 
To: "Kevin P. Fleming" < kpfleming@digium.com > 
Cc: codec@ietf.org 
Sent: Saturday, January 15, 2011 2:07:28 PM 

Subject: Re: [codec] Next Steps for WG 




First of all, CODEC definition should be independent from the transport protocol, and we might need this functionality when RTP SSRC are not available. 

Furthermore, in case of RTP, there are two problems with your suggestion: 

1. Quite a few clients do not allow remote party to change SSRC without the re-Invite. SIP clients note SSRC of the first received RTP packet and ignore RTP packets with different SSRC 

2. Even when the client allows switching of SSRC on the fly, a few initial RTP packets are typically discarded due to RTP probation. After this, clients typically need to pre-fill the jitter buffer, and only after this start audio playback. This produces from 40ms to 100ms gap in audio. This is very audible and highly undesirable. 

Finally, what I was looking for was a bit more then just decoder reset. Ideally I wanted to set decoder state to a known value to start decoding audio packets that I will send to it. 

P.S. As a side note, Intel Performance Primitives CODECs implement a packet type in its wave file format that resets the decoder. This packet is used to simplify implementation of performance and regression tests. I think they are using standard sized packet with all bits set to zero for this purpose. We can at least do something similar. 

_____________ 
Roman Shpount 



On Sat, Jan 15, 2011 at 8:12 AM, Kevin P. Fleming < kpfleming@digium.com > wrote: 



On 01/14/2011 11:27 PM, Roman Shpount wrote: 


My question was not as much about the VAD, as about multiple stream 
combining. What I want is to implement is switching between multiple 
talker based on VAD (computation of which is done by some external 
methods). In this case we need to indicate to the remote party that this 
is a new stream and ideally get the remote decoder in such a state that 
it can immediately start correctly decoding the new audio without a 
glitch. So, the minimum requirement would be to have a flag or a packet 
that indicates the decoder that it needs to reset its state. Ideally, we 
need an algorithm to get the proper decoder state based on some audio 
history stored on the conferencing server, and to send this decoder 
state to the remote party so that it can be synchronized. I don't think 
this should affect the codec performance. This is just something that 
needs to be accommodated in the bitstream by adding a packet type which 
either resets the decoder state or sets it to some specified value. 

This sort of mechanism already exists when using RTP as the transport mechanism, by setting the marker bit and changing the SSRC to indicate that the payload in the packet is from a different source than the previous packet. In my opinion there's no need for the codec bitstream to have any provisions for such an indication. 

-- 
Kevin P. Fleming 
Digium, Inc. | Director of Software Technologies 
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 
skype: kpfleming | jabber: kfleming@digium.com 
Check us out at www.digium.com & www.asterisk.org 



_______________________________________________ 
codec mailing list 
codec@ietf.org 
https://www.ietf.org/mailman/listinfo/codec 


_______________________________________________ 
codec mailing list 
codec@ietf.org 
https://www.ietf.org/mailman/listinfo/codec 


------=_Part_1113906_217516684.1295392430780
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><head><style type=3D'text/css'>p { margin: 0; }</style></head><body><=
div style=3D'font-family: Times New Roman; font-size: 12pt; color: #000000'=
>That should work.&nbsp; <br>Make sure to get the time alignment right by a=
djusting for look-ahead etc.&nbsp; <br>And only the first frame of the doub=
le talk period has to be coded independently; for the others you could let =
the encoder decide.&nbsp; In general, whenever you're adding or removing a =
stream from the mix you want an independent frame.<br><br>best,<br>koen.<br=
><br><br><hr id=3D"zwchr"><b>From: </b>"Roman Shpount" &lt;roman@telurix.co=
m&gt;<br><b>To: </b>"Koen Vos" &lt;koen.vos@skype.net&gt;<br><b>Cc: </b>cod=
ec@ietf.org<br><b>Sent: </b>Tuesday, January 18, 2011 2:55:09 PM<br><b>Subj=
ect: </b>Re: [codec] Next Steps for WG<br><br>Thank you, this is very helpf=
ul. <br><br>What I was considering was something similar: Typically you hav=
e one speaker in the conference with short periods of double talk. I was th=
inking that decoding, mixing and encoding using independent frames the doub=
le talk period with the few initial frames of the new speaker, and then swi=
tching to a new speaker stream would sound the most natural and will only r=
equire re-encoding for short periods of time.<br>
<br>Thank you for your great work,<br clear=3D"all">_______________________=
______<br>Roman Shpount - <a href=3D"http://www.telurix.com" target=3D"_bla=
nk">www.telurix.com</a><br>
<br><br><div class=3D"gmail_quote">On Tue, Jan 18, 2011 at 5:35 PM, Koen Vo=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:koen.vos@skype.net" target=3D"_bl=
ank">koen.vos@skype.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail=
_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204,=
 204, 204); padding-left: 1ex;">
<div><div style=3D"font-family: Times New Roman; font-size: 12pt; color: rg=
b(0, 0, 0);">Roman:<br><br>Jean-Marc and I discussed this a bit, and this i=
s how we see it:<br><br>Resetting the decoder state would lead to a discont=
inuity in the output signal and may create an objectionable "click" sound.&=
nbsp; Moreover, the incoming stream that's enabled in the switching node at=
 the moment of the decoder reset may depend on the last frame before the sw=
itch. So the first few frames would not be decoded correctly.<br>
<br>Coding and transmitting the decoder state doesn't overcome the disconti=
nuity problem.&nbsp; And it would add a lot of complications and code to Op=
us.&nbsp; <br><br>A better method seems to be:<br>1. Determine if the first=
 frame after switching has significant dependencies on the previous frame (=
i.e., long-term prediction in the case of SILK, or energy prediction in the=
 case of CELT)<br>
2a: If yes: decode the first frame and re-encode it independently.&nbsp; Af=
ter that, switch to the new stream.<br>
2b. If no: simply switch to the new stream immediately.<br>This ensures a s=
mooth transition and fast convergence to the correct output signal for the =
new stream.<br><br>best,<br>koen.<br><br><br><hr><b>From: </b>"Roman Shpoun=
t" &lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix=
.com</a>&gt;<br>
<b>To: </b>"Kevin P. Fleming" &lt;<a href=3D"mailto:kpfleming@digium.com" t=
arget=3D"_blank">kpfleming@digium.com</a>&gt;<br><b>Cc: </b><a href=3D"mail=
to:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br><b>Sent: </b>Sat=
urday, January 15, 2011 2:07:28 PM<div class=3D"im">
<br><b>Subject: </b>Re: [codec] Next Steps for WG<br><br></div><div><div></=
div><div class=3D"h5">First of all, CODEC definition should be independent =
from the transport protocol, and we might need this functionality when RTP =
SSRC are not available.<br>
<br>Furthermore, in case of RTP, there are two problems with your suggestio=
n:<br>
<br>1. Quite a few clients do not allow remote party to change SSRC without=
 the re-Invite. SIP clients note SSRC of the first received RTP packet and =
ignore RTP packets with different SSRC<br><br>2. Even when the client allow=
s switching of SSRC on the fly, a few initial RTP packets are typically dis=
carded due to RTP probation. After this, clients typically need to pre-fill=
 the jitter buffer, and only after this start audio playback. This produces=
 from 40ms to 100ms gap in audio. This is very audible and highly undesirab=
le.<br>

<br>Finally, what I was looking for was a bit more then just decoder reset.=
 Ideally I wanted to set decoder state to a known value to start decoding a=
udio packets that I will send to it.<br><br>P.S. As a side note, Intel Perf=
ormance Primitives CODECs implement a packet type in its wave file format t=
hat resets the decoder. This packet is used to simplify implementation of p=
erformance and regression tests. I think they are using standard sized pack=
et with all bits set to zero for this purpose. We can at least do something=
 similar.<br>

<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Sat, Jan 15, 2011 at 8:12 AM, Kevin P=
. Fleming <span dir=3D"ltr">&lt;<a href=3D"mailto:kpfleming@digium.com" tar=
get=3D"_blank">kpfleming@digium.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px so=
lid rgb(204, 204, 204); padding-left: 1ex;">

<div>On 01/14/2011 11:27 PM, Roman Shpount wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
My question was not as much about the VAD, as about multiple stream<br>
combining. What I want is to implement is switching between multiple<br>
talker based on VAD (computation of which is done by some external<br>
methods). In this case we need to indicate to the remote party that this<br=
>
is a new stream and ideally get the remote decoder in such a state that<br>
it can immediately start correctly decoding the new audio without a<br>
glitch. So, the minimum requirement would be to have a flag or a packet<br>
that indicates the decoder that it needs to reset its state. Ideally, we<br=
>
need an algorithm to get the proper decoder state based on some audio<br>
history stored on the conferencing server, and to send this decoder<br>
state to the remote party so that it can be synchronized. I don't think<br>
this should affect the codec performance. This is just something that<br>
needs to be accommodated in the bitstream by adding a packet type which<br>
either resets the decoder state or sets it to some specified value.<br>
</blockquote>
<br></div>
This sort of mechanism already exists when using RTP as the transport mecha=
nism, by setting the marker bit and changing the SSRC to indicate that the =
payload in the packet is from a different source than the previous packet. =
In my opinion there's no need for the codec bitstream to have any provision=
s for such an indication.<br>

<font color=3D"#888888">
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href=3D"mailto:kfleming@digium.com" target=3D=
"_blank">kfleming@digium.com</a><br>
Check us out at <a href=3D"http://www.digium.com" target=3D"_blank">www.dig=
ium.com</a> &amp; <a href=3D"http://www.asterisk.org" target=3D"_blank">www=
.asterisk.org</a></font><div><div></div><div><br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>
<br>_______________________________________________<br>codec mailing list<b=
r><a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br=
><a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></div></div></blockquote></div><br>
</div></body></html>
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Filename        : draft-ietf-codec-guidelines-00.txt
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From jdrosen@jdrosen.net  Sat Jan 22 05:06:14 2011
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Date: Sat, 22 Jan 2011 08:08:58 -0500
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Subject: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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The authors believe that the requirements document is complete. The 
chairs would like to now issue a 3 week working group last call for 
draft-ietf-codec-requirements-02. At the end of 3 weeks, if no comments 
have been received, we will pass the document to the IESG for approval.

Thanks,
Jonathan R.
-- 
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From hoene@uni-tuebingen.de  Sun Jan 23 00:17:05 2011
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
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Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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> The authors believe that the requirements document is complete. 

[Christian Hoene] Which authors? So far this document has only two selected
editors.

>The chairs
> would like to now issue a 3 week working group last call for
draft-ietf-codec-
> requirements-02. At the end of 3 weeks, if no comments have been
> received, we will pass the document to the IESG for approval.

[Christian Hoene] Despite hundreds of emails, what were exchanged on this
mailing list on issues regarding requirements, the editors have been
reluctant to update the requirements document according to the consensus
process on this mailing list. 

If I compare draft-ietf-codec-requirement-02 with
draft-valin-codec-requirements-02, I only see editorial changes.
https://tools.ietf.org/rfcdiff?url1=draft-ietf-codec-requirements-02&difftyp
e=--html&submit=Go!&url2=draft-valin-codec-requirements-02

Sorry guys, if I see this I become very cynical on the standardization
process at the IETF.   The editors have not done their job and the chairs do
not care. At the end, it all looks like rubberstamping with a little bit of
theater.

Christian



From stephen.botzko@gmail.com  Mon Jan 24 10:47:05 2011
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Date: Mon, 24 Jan 2011 13:49:58 -0500
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From: Stephen Botzko <stephen.botzko@gmail.com>
To: Christian Hoene <hoene@uni-tuebingen.de>
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Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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>>>
[Christian Hoene] Despite hundreds of emails, what were exchanged on this
mailing list on issues regarding requirements, the editors have been
reluctant to update the requirements document according to the consensus
process on this mailing list.
>>>
Christian - perhaps you could post a list of the issues you see that haven't
been addressed?

Stephen Botzko

On Sun, Jan 23, 2011 at 3:19 AM, Christian Hoene <hoene@uni-tuebingen.de>wrote:

> > The authors believe that the requirements document is complete.
>
> [Christian Hoene] Which authors? So far this document has only two selected
> editors.
>
> >The chairs
> > would like to now issue a 3 week working group last call for
> draft-ietf-codec-
> > requirements-02. At the end of 3 weeks, if no comments have been
> > received, we will pass the document to the IESG for approval.
>
> [Christian Hoene] Despite hundreds of emails, what were exchanged on this
> mailing list on issues regarding requirements, the editors have been
> reluctant to update the requirements document according to the consensus
> process on this mailing list.
>
> If I compare draft-ietf-codec-requirement-02 with
> draft-valin-codec-requirements-02, I only see editorial changes.
>
> https://tools.ietf.org/rfcdiff?url1=draft-ietf-codec-requirements-02&difftyp
> e=--html&submit=Go!&url2=draft-valin-codec-requirements-02
>
> Sorry guys, if I see this I become very cynical on the standardization
> process at the IETF.   The editors have not done their job and the chairs
> do
> not care. At the end, it all looks like rubberstamping with a little bit of
> theater.
>
> Christian
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

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&gt;&gt;&gt;<br>[Christian Hoene] Despite hundreds of emails, what were exc=
hanged on this<br>
mailing list on issues regarding requirements, the editors have been<br>
reluctant to update the requirements document according to the consensus<br=
>
process on this mailing list.<br>&gt;&gt;&gt;<br>Christian - perhaps you co=
uld post a list of the issues you see that haven&#39;t been addressed?<br><=
br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Sun, Jan 23, 2011 at=
 3:19 AM, Christian Hoene <span dir=3D"ltr">&lt;<a href=3D"mailto:hoene@uni=
-tuebingen.de">hoene@uni-tuebingen.de</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div class=3D"im"=
>&gt; The authors believe that the requirements document is complete.<br>
<br>
</div>[Christian Hoene] Which authors? So far this document has only two se=
lected<br>
editors.<br>
<div class=3D"im"><br>
&gt;The chairs<br>
&gt; would like to now issue a 3 week working group last call for<br>
draft-ietf-codec-<br>
&gt; requirements-02. At the end of 3 weeks, if no comments have been<br>
&gt; received, we will pass the document to the IESG for approval.<br>
<br>
</div>[Christian Hoene] Despite hundreds of emails, what were exchanged on =
this<br>
mailing list on issues regarding requirements, the editors have been<br>
reluctant to update the requirements document according to the consensus<br=
>
process on this mailing list.<br>
<br>
If I compare draft-ietf-codec-requirement-02 with<br>
draft-valin-codec-requirements-02, I only see editorial changes.<br>
<a href=3D"https://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-codec-requireme=
nts-02&amp;difftyp" target=3D"_blank">https://tools.ietf.org/rfcdiff?url1=
=3Ddraft-ietf-codec-requirements-02&amp;difftyp</a><br>
e=3D--html&amp;submit=3DGo!&amp;url2=3Ddraft-valin-codec-requirements-02<br=
>
<br>
Sorry guys, if I see this I become very cynical on the standardization<br>
process at the IETF. =A0 The editors have not done their job and the chairs=
 do<br>
not care. At the end, it all looks like rubberstamping with a little bit of=
<br>
theater.<br>
<font color=3D"#888888"><br>
Christian<br>
</font><div><div></div><div class=3D"h5"><br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br>

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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: "'Stephen Botzko'" <stephen.botzko@gmail.com>
References: <4D3AD6EA.5020607@jdrosen.net>	<000001cbbad6$4f44aea0$edce0be0$@uni-tuebingen.de> <AANLkTi=xTwet-toobezTZAsitgdTnTrMCHDD3OqChxF7@mail.gmail.com>
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Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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Christian - perhaps you could post a list of the issues you see that haven't
been addressed?



[Christian Hoene] No Stephen, these issues have been written down in
previous emails, drafts and issues in the Trac. They can be read by anybody
anytime. Thus, I do not see any benefit of repeating them again if the
editors continue to ignore any input. Indeed, they did not improve the draft
despite sound technical reasons. 

Even if somebody is not fully involved in the technical details: It is very
odd that despite many hundreds emails and many discussions since starting
this WG the editors have not updated the draft beside minor changes such as
the boilerplate and typos. 

Even if the lack of any update was not intentionally, the editors missed to
do their job because they were too lazy or rather too busy doing other
thinks.

I would be sad if all the fruitful discussions here and all the good
contributions of many industry experts should have been in vain. Even if not
all requirements can be met by Opus, a proper requirements document may be
relevant for future solutions or other SDOs.

CH


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4.0pt'><p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
lang=3DEN-US>Christian - perhaps you could post a list of the issues you =
see that haven't been addressed?<br><br><span =
style=3D'color:#1F497D'><o:p></o:p></span></span></p><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><i><span =
lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>[Christian Hoene] No Stephen, these issues have been written down in =
previous emails, drafts and issues in the Trac. They can be read by =
anybody anytime. Thus, I do not see any benefit of repeating them again =
if the editors continue to ignore any input. Indeed, they did not =
improve the draft despite sound technical reasons. =
<o:p></o:p></span></i></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><i><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Even if somebody is not fully involved in the technical details: It =
is very odd that despite many hundreds emails and many discussions since =
starting this WG the editors have not updated the draft beside minor =
changes such as the boilerplate and typos. =
<o:p></o:p></span></i></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><i><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Even if the lack of any update was not intentionally, the editors =
missed to do their job because they were too lazy or rather too busy =
doing other thinks.<o:p></o:p></span></i></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><i><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I would be sad if all the fruitful discussions here and all the good =
contributions of many industry experts should have been in vain. Even if =
not all requirements can be met by Opus, a proper requirements document =
may be relevant for future solutions or other =
SDOs.<o:p></o:p></span></i></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><i><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>CH<o:p></o:p></span></i></b></p></div></div></body></html>
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From stewe@stewe.org  Mon Jan 24 12:33:22 2011
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From: Stephan Wenger <stewe@stewe.org>
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Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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Hi Christian:
I can understand your frustration, and to some extent I share it.  However,
the purpose of a WGLC is to bring up issues that the editors have missed or
not acted upon.  So let's bring them up.

All:
The tracker is supposed to be one organized form of keeping issues alive and
make sure that they have been addressed adequately.  The tracker can be
found at http://trac.tools.ietf.org/wg/codec/trac/report.  I believe that
most requirements-related issues have been entered into the tracker in some
form, although, in order to see whether an issue has been addressed
according to the WG consensus or to get a full picture of all issues, one
would have to review the mailing list as well.  But the tracker is a good
starting point.

Thanks,
Stephan


From:  Christian Hoene <hoene@uni-tuebingen.de>
Organization:  Universitat Tubingen
Date:  Mon, 24 Jan 2011 21:10:35 +0100
To:  'Stephen Botzko' <stephen.botzko@gmail.com>
Cc:  <codec@ietf.org>
Subject:  Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02

Christian - perhaps you could post a list of the issues you see that haven't
been addressed?


[Christian Hoene] No Stephen, these issues have been written down in
previous emails, drafts and issues in the Trac. They can be read by anybody
anytime. Thus, I do not see any benefit of repeating them again if the
editors continue to ignore any input. Indeed, they did not improve the draft
despite sound technical reasons.
Even if somebody is not fully involved in the technical details: It is very
odd that despite many hundreds emails and many discussions since starting
this WG the editors have not updated the draft beside minor changes such as
the boilerplate and typos.
Even if the lack of any update was not intentionally, the editors missed to
do their job because they were too lazy or rather too busy doing other
thinks.
I would be sad if all the fruitful discussions here and all the good
contributions of many industry experts should have been in vain. Even if not
all requirements can be met by Opus, a proper requirements document may be
relevant for future solutions or other SDOs.
CH
_______________________________________________ codec mailing list
codec@ietf.org https://www.ietf.org/mailman/listinfo/codec


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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>Hi Christian:</div><div>I ca=
n understand your frustration, and to some extent I share it. &nbsp;However,=
 the purpose of a WGLC is to bring up issues that the editors have missed or=
 not acted upon. &nbsp;So let's bring them up. &nbsp;</div><div><br></div><d=
iv>All:</div><div>The tracker is supposed to be one organized form of keepin=
g issues alive and make sure that they have been addressed adequately. &nbsp=
;The tracker can be found at&nbsp;<a href=3D"http://trac.tools.ietf.org/wg/cod=
ec/trac/report">http://trac.tools.ietf.org/wg/codec/trac/report</a>. &nbsp;I=
 believe that most requirements-related issues have been entered into the tr=
acker in some form, although, in order to see whether an issue has been addr=
essed according to the WG consensus or to get a full picture of all issues, =
one would have to review the mailing list as well. &nbsp;But the tracker is =
a good starting point.</div><div><br></div><div>Thanks,</div><div>Stephan</d=
iv><div><br></div><div><br></div><span id=3D"OLK_SRC_BODY_SECTION"><div style=3D=
"font-family:Calibri; font-size:11pt; text-align:left; color:black; BORDER-B=
OTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-L=
EFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: m=
edium none; PADDING-TOP: 3pt"><span style=3D"font-weight:bold">From: </span> C=
hristian Hoene &lt;<a href=3D"mailto:hoene@uni-tuebingen.de">hoene@uni-tuebing=
en.de</a>&gt;<br><span style=3D"font-weight:bold">Organization: </span> Univer=
sitat Tubingen<br><span style=3D"font-weight:bold">Date: </span> Mon, 24 Jan 2=
011 21:10:35 +0100<br><span style=3D"font-weight:bold">To: </span> 'Stephen Bo=
tzko' &lt;<a href=3D"mailto:stephen.botzko@gmail.com">stephen.botzko@gmail.com=
</a>&gt;<br><span style=3D"font-weight:bold">Cc: </span> &lt;<a href=3D"mailto:c=
odec@ietf.org">codec@ietf.org</a>&gt;<br><span style=3D"font-weight:bold">Subj=
ect: </span> Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02<br>=
</div><div><br></div><div xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"u=
rn:schemas-microsoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:o=
ffice:word" xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=
=3D"http://www.w3.org/TR/REC-html40"><style><!--
/* Font Definitions */
@font-face
	{font-family:Calibri;
	panose-1:2 15 5 2 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
	{margin:0cm;
	margin-bottom:.0001pt;
	font-size:12.0pt;
	font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
	{mso-style-priority:99;
	color:blue;
	text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
	{mso-style-priority:99;
	color:purple;
	text-decoration:underline;}
span.E-MailFormatvorlage17
	{mso-style-type:personal-reply;
	font-family:"Calibri","sans-serif";
	color:#1F497D;}
.MsoChpDefault
	{mso-style-type:export-only;
	font-family:"Calibri","sans-serif";
	mso-fareast-language:EN-US;}
@page WordSection1
	{size:612.0pt 792.0pt;
	margin:70.85pt 70.85pt 2.0cm 70.85pt;}
div.WordSection1
	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--><div lang=3D"DE" link=3D"blue" vlink=3D"purple"=
><div class=3D"WordSection1"><div style=3D"border:none;border-left:solid blue 1.=
5pt;padding:0cm 0cm 0cm 4.0pt"><p class=3D"MsoNormal" style=3D"margin-bottom:12.=
0pt"><span lang=3D"EN-US">Christian - perhaps you could post a list of the iss=
ues you see that haven't been addressed?<br><br><span style=3D"color:#1F497D">=
<o:p></o:p></span></span></p><p class=3D"MsoNormal" style=3D"margin-bottom:12.0p=
t"><b><i><span lang=3D"EN-US" style=3D"font-size: 11pt; color: rgb(31, 73, 125);=
 font-family: Calibri, sans-serif; ">[Christian Hoene] No Stephen, these iss=
ues have been written down in previous emails, drafts and issues in the Trac=
. They can be read by anybody anytime. Thus, I do not see any benefit of rep=
eating them again if the editors continue to ignore any input. Indeed, they =
did not improve the draft despite sound technical reasons. <o:p></o:p></span=
></i></b></p><p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span l=
ang=3D"EN-US" style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Ca=
libri, sans-serif; ">Even if somebody is not fully involved in the technical=
 details: It is very odd that despite many hundreds emails and many discussi=
ons since starting this WG the editors have not updated the draft beside min=
or changes such as the boilerplate and typos. <o:p></o:p></span></i></b></p>=
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><i><span lang=3D"EN-US" s=
tyle=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calibri, sans-s=
erif; ">Even if the lack of any update was not intentionally, the editors mi=
ssed to do their job because they were too lazy or rather too busy doing oth=
er thinks.<o:p></o:p></span></i></b></p><p class=3D"MsoNormal" style=3D"margin-b=
ottom:12.0pt"><b><i><span lang=3D"EN-US" style=3D"font-size: 11pt; color: rgb(31=
, 73, 125); font-family: Calibri, sans-serif; ">I would be sad if all the fr=
uitful discussions here and all the good contributions of many industry expe=
rts should have been in vain. Even if not all requirements can be met by Opu=
s, a proper requirements document may be relevant for future solutions or ot=
her SDOs.<o:p></o:p></span></i></b></p><p class=3D"MsoNormal" style=3D"margin-bo=
ttom:12.0pt"><b><i><span lang=3D"EN-US" style=3D"font-size: 11pt; color: rgb(31,=
 73, 125); font-family: Calibri, sans-serif; ">CH<o:p></o:p></span></i></b><=
/p></div></div></div></div>_______________________________________________
codec mailing list
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec">https://www.ietf.org/=
mailman/listinfo/codec</a>
</span></body></html>

--B_3378717373_14165870--



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#13: reference use-cases and topologies!


Comment(by gmaxwell@â€¦):

 As far as I can tell, this request is redundant to RFC 5117, as pointed
 out above.

 Can we go ahead and close out this ticket?

-- 
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 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  defect                  |      Status:  new
 Priority:  critical                |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  Active WG Document      |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/13#comment:4>
codec <http://tools.ietf.org/codec/>


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#20: Computational complexity?


Comment(by gmaxwell@â€¦):

 Section 4.4 of draft-ietf-codec-requirements-02 appears to address this
 point. Is there any opposition to closing this ticket?

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------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/20#comment:4>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #21 (new): Supporting Wireless Links?
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#21: Supporting Wireless Links?


Comment(by gmaxwell@â€¦):

 Wireless IP is a packet-loss channel like all other IP networks.  The
 requirements draft suggests that bit-error robustness is useful for non-
 packet-loss channels, but I don't see what more is required on this
 subject.  Is any further activity required on this point?

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------------------------------------+---------------------------------------

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codec <http://tools.ietf.org/codec/>


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#22: Memory Usage?

Changes (by gmaxwell@â€¦):

  * status:  new => closed
  * resolution:  => worksforme


Comment:

 This is addressed in the final paragraph of section 4.4 of draft-ietf-
 codec-requirements-02.txt.

-- 
------------------------------------+---------------------------------------
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     Type:  defect                  |       Status:  closed    
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------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/22#comment:2>
codec <http://tools.ietf.org/codec/>


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#25: Supported audio contents


Comment(by gmaxwell@â€¦):

 Section 2. Lists a number of applications, and for each of those it breaks
 out some of the content requirements.  I believe the actual testing
 procedure is out of scope for the requirements RFC.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  defect                  |      Status:  new
 Priority:  major                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  Active WG Document      |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/25#comment:1>
codec <http://tools.ietf.org/codec/>


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#26: Comparing the CODEC to other codecs...


Comment(by gmaxwell@â€¦):

 There are thousands of codecs in existence. The requirements specify a
 number of baseline comparison points of codecs nearest the current
 application space.  Considering the significant workload required to add
 another codec in a comparison, I don't believe any omission here is
 actually an issue unless there is a clear consensus that we need to add
 additional codecs to the baseline set. I believe we can close this issue.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  defect                  |      Status:  new
 Priority:  major                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/26#comment:1>
codec <http://tools.ietf.org/codec/>


From jean-marc.valin@octasic.com  Mon Jan 24 15:06:02 2011
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Cc: codec@ietf.org, 'Stephen Botzko' <stephen.botzko@gmail.com>
Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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Christian,

I actually responded to the last comments you made a while ago (oct 2010). 
One issue I pointed out was your use of RFC2119 keywords, which (AFAIK) 
aren't appropriate for a requirements draft (the requirements aren't a 
standard). So statements like "Any codec specified by the IETF MUST be well 
specified", besides stating the obvious, are inappropriate.

There were also comments that just did not belong to this draft, such as 
the section on collaboration with other WGs. Collaboration is not a 
characteristic of a codec. So essentially, I merged the uncontroversial 
suggestion, but that's all I could do. As far as I'm aware, there's nothing 
in the current draft that goes against the consensus of the WG. If there 
are, please point to specific issues and to statements made by others (not 
just you) asking for the change.

Cheers,

	Jean-Marc

On 11-01-24 03:10 PM, Christian Hoene wrote:
> Christian - perhaps you could post a list of the issues you see that
> haven't been addressed?
>
> */[Christian Hoene] No Stephen, these issues have been written down in
> previous emails, drafts and issues in the Trac. They can be read by anybody
> anytime. Thus, I do not see any benefit of repeating them again if the
> editors continue to ignore any input. Indeed, they did not improve the
> draft despite sound technical reasons. /*
>
> */Even if somebody is not fully involved in the technical details: It is
> very odd that despite many hundreds emails and many discussions since
> starting this WG the editors have not updated the draft beside minor
> changes such as the boilerplate and typos. /*
>
> */Even if the lack of any update was not intentionally, the editors missed
> to do their job because they were too lazy or rather too busy doing other
> thinks./*
>
> */I would be sad if all the fruitful discussions here and all the good
> contributions of many industry experts should have been in vain. Even if
> not all requirements can be met by Opus, a proper requirements document may
> be relevant for future solutions or other SDOs./*
>
> */CH/*
>
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


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Subject: Re: [codec] requirements #30 (new): Self-awareness
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#30: Self-awareness


Comment(by gmaxwell@â€¦):

 These all sound like implementation properties to me, and could be
 implemented (and would depend on) transport level things like the RTP
 timestamps.  An implementation could do (or fail to do) these things
 completely independent of the the design of the codec working group
 product. I believe that we can close this issue in terms of the
 requirements.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  defect                  |      Status:  new
 Priority:  major                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/30#comment:1>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #3 (new): 2.2. Conferencing: Support of binaural audio?
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#3: 2.2.  Conferencing: Support of binaural audio?


Comment(by gmaxwell@â€¦):

 Stereo support is address in 5.5 of draft-ietf-codec-requirements-02.txt
 and the mixing language in 5.1 is stereo/mono neutral. I am not aware of
 any technical reason raised why stereo _mixing_ would somehow be
 technically distinct from any other form of mixing.  More information is
 required if this issue is to remain open.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  enhancement             |      Status:  new
 Priority:  major                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/3#comment:2>
codec <http://tools.ietf.org/codec/>


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Date: Mon, 24 Jan 2011 17:24:30 -0600
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Cc: codec@ietf.org, Stephen Botzko <stephen.botzko@gmail.com>
Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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The document is currently specified as Standards Track, thus the use of RFC
2119 language is entirely appropriate.  However, ISTM that the document
should really just be Informational, in particular given the language in the
introduction that this document defines a "suggested process", as opposed to
a process that is required for codec development.

Regards,
Mary.

On Mon, Jan 24, 2011 at 5:08 PM, Jean-Marc Valin <
jean-marc.valin@octasic.com> wrote:

> Christian,
>
> I actually responded to the last comments you made a while ago (oct 2010).
> One issue I pointed out was your use of RFC2119 keywords, which (AFAIK)
> aren't appropriate for a requirements draft (the requirements aren't a
> standard). So statements like "Any codec specified by the IETF MUST be well
> specified", besides stating the obvious, are inappropriate.
>
> There were also comments that just did not belong to this draft, such as
> the section on collaboration with other WGs. Collaboration is not a
> characteristic of a codec. So essentially, I merged the uncontroversial
> suggestion, but that's all I could do. As far as I'm aware, there's nothing
> in the current draft that goes against the consensus of the WG. If there
> are, please point to specific issues and to statements made by others (not
> just you) asking for the change.
>
> Cheers,
>
>        Jean-Marc
>
>
> On 11-01-24 03:10 PM, Christian Hoene wrote:
>
>> Christian - perhaps you could post a list of the issues you see that
>> haven't been addressed?
>>
>> */[Christian Hoene] No Stephen, these issues have been written down in
>> previous emails, drafts and issues in the Trac. They can be read by
>> anybody
>> anytime. Thus, I do not see any benefit of repeating them again if the
>> editors continue to ignore any input. Indeed, they did not improve the
>> draft despite sound technical reasons. /*
>>
>> */Even if somebody is not fully involved in the technical details: It is
>> very odd that despite many hundreds emails and many discussions since
>> starting this WG the editors have not updated the draft beside minor
>> changes such as the boilerplate and typos. /*
>>
>> */Even if the lack of any update was not intentionally, the editors missed
>> to do their job because they were too lazy or rather too busy doing other
>> thinks./*
>>
>> */I would be sad if all the fruitful discussions here and all the good
>> contributions of many industry experts should have been in vain. Even if
>> not all requirements can be met by Opus, a proper requirements document
>> may
>> be relevant for future solutions or other SDOs./*
>>
>> */CH/*
>>
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--00163630f5fbdf81b4049a9fe604
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The document is currently specified as Standards Track, thus the use of RFC=
 2119 language is entirely appropriate. =A0However, ISTM that the document =
should really just be Informational, in particular given the language in th=
e introduction that this document defines a <font class=3D"Apple-style-span=
" face=3D"verdana, sans-serif">&quot;</font><span class=3D"Apple-style-span=
" style=3D"white-space: pre-wrap; "><font class=3D"Apple-style-span" face=
=3D"verdana, sans-serif">suggested process&quot;</font><font class=3D"Apple=
-style-span" face=3D"arial, helvetica, sans-serif">, as opposed to a proces=
s that is required for codec development. </font></span><span class=3D"Appl=
e-style-span" style=3D"font-family: monospace; font-size: medium; white-spa=
ce: pre-wrap; ">=A0</span><div>
<font class=3D"Apple-style-span" face=3D"monospace"><span class=3D"Apple-st=
yle-span" style=3D"white-space: pre-wrap; font-size: medium;"><br></span></=
font></div><div><span class=3D"Apple-style-span" style=3D"white-space: pre-=
wrap; "><font class=3D"Apple-style-span" face=3D"arial, helvetica, sans-ser=
if">Regards,</font></span></div>
<div><span class=3D"Apple-style-span" style=3D"white-space: pre-wrap; "><fo=
nt class=3D"Apple-style-span" face=3D"arial, helvetica, sans-serif">Mary.<b=
r></font></span><br><div class=3D"gmail_quote">On Mon, Jan 24, 2011 at 5:08=
 PM, Jean-Marc Valin <span dir=3D"ltr">&lt;<a href=3D"mailto:jean-marc.vali=
n@octasic.com">jean-marc.valin@octasic.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">Christian,<br>
<br>
I actually responded to the last comments you made a while ago (oct 2010). =
One issue I pointed out was your use of RFC2119 keywords, which (AFAIK) are=
n&#39;t appropriate for a requirements draft (the requirements aren&#39;t a=
 standard). So statements like &quot;Any codec specified by the IETF MUST b=
e well specified&quot;, besides stating the obvious, are inappropriate.<br>

<br>
There were also comments that just did not belong to this draft, such as th=
e section on collaboration with other WGs. Collaboration is not a character=
istic of a codec. So essentially, I merged the uncontroversial suggestion, =
but that&#39;s all I could do. As far as I&#39;m aware, there&#39;s nothing=
 in the current draft that goes against the consensus of the WG. If there a=
re, please point to specific issues and to statements made by others (not j=
ust you) asking for the change.<br>

<br>
Cheers,<br>
<br>
 =A0 =A0 =A0 =A0Jean-Marc<div><div></div><div class=3D"h5"><br>
<br>
On 11-01-24 03:10 PM, Christian Hoene wrote:<br>
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div></div><div class=3D"h5=
">
Christian - perhaps you could post a list of the issues you see that<br>
haven&#39;t been addressed?<br>
<br>
*/[Christian Hoene] No Stephen, these issues have been written down in<br>
previous emails, drafts and issues in the Trac. They can be read by anybody=
<br>
anytime. Thus, I do not see any benefit of repeating them again if the<br>
editors continue to ignore any input. Indeed, they did not improve the<br>
draft despite sound technical reasons. /*<br>
<br>
*/Even if somebody is not fully involved in the technical details: It is<br=
>
very odd that despite many hundreds emails and many discussions since<br>
starting this WG the editors have not updated the draft beside minor<br>
changes such as the boilerplate and typos. /*<br>
<br>
*/Even if the lack of any update was not intentionally, the editors missed<=
br>
to do their job because they were too lazy or rather too busy doing other<b=
r>
thinks./*<br>
<br>
*/I would be sad if all the fruitful discussions here and all the good<br>
contributions of many industry experts should have been in vain. Even if<br=
>
not all requirements can be met by Opus, a proper requirements document may=
<br>
be relevant for future solutions or other SDOs./*<br>
<br></div></div>
*/CH/*<div class=3D"im"><br>
<br>
<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></blockquote><div><div></div><div class=3D"h5">
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br></div>

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Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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I agree with Mary here.

On 1/24/11 4:24 PM, Mary Barnes wrote:
> The document is currently specified as Standards Track, thus the use of=

> RFC 2119 language is entirely appropriate.  However, ISTM that the
> document should really just be Informational, in particular given the
> language in the introduction that this document defines a "suggested
> process", as opposed to a process that is required for codec developmen=
t. =20
>=20
> Regards,
> Mary.
>=20
> On Mon, Jan 24, 2011 at 5:08 PM, Jean-Marc Valin
> <jean-marc.valin@octasic.com <mailto:jean-marc.valin@octasic.com>> wrot=
e:
>=20
>     Christian,
>=20
>     I actually responded to the last comments you made a while ago (oct=

>     2010). One issue I pointed out was your use of RFC2119 keywords,
>     which (AFAIK) aren't appropriate for a requirements draft (the
>     requirements aren't a standard). So statements like "Any codec
>     specified by the IETF MUST be well specified", besides stating the
>     obvious, are inappropriate.
>=20
>     There were also comments that just did not belong to this draft,
>     such as the section on collaboration with other WGs. Collaboration
>     is not a characteristic of a codec. So essentially, I merged the
>     uncontroversial suggestion, but that's all I could do. As far as I'=
m
>     aware, there's nothing in the current draft that goes against the
>     consensus of the WG. If there are, please point to specific issues
>     and to statements made by others (not just you) asking for the chan=
ge.
>=20
>     Cheers,
>=20
>            Jean-Marc
>=20
>=20
>     On 11-01-24 03:10 PM, Christian Hoene wrote:
>=20
>         Christian - perhaps you could post a list of the issues you see=
 that
>         haven't been addressed?
>=20
>         */[Christian Hoene] No Stephen, these issues have been written
>         down in
>         previous emails, drafts and issues in the Trac. They can be rea=
d
>         by anybody
>         anytime. Thus, I do not see any benefit of repeating them again=

>         if the
>         editors continue to ignore any input. Indeed, they did not
>         improve the
>         draft despite sound technical reasons. /*
>=20
>         */Even if somebody is not fully involved in the technical
>         details: It is
>         very odd that despite many hundreds emails and many discussions=

>         since
>         starting this WG the editors have not updated the draft beside =
minor
>         changes such as the boilerplate and typos. /*
>=20
>         */Even if the lack of any update was not intentionally, the
>         editors missed
>         to do their job because they were too lazy or rather too busy
>         doing other
>         thinks./*
>=20
>         */I would be sad if all the fruitful discussions here and all
>         the good
>         contributions of many industry experts should have been in vain=
=2E
>         Even if
>         not all requirements can be met by Opus, a proper requirements
>         document may
>         be relevant for future solutions or other SDOs./*
>=20
>         */CH/*
>=20
>=20
>=20
>=20
>         _______________________________________________
>         codec mailing list
>         codec@ietf.org <mailto:codec@ietf.org>
>         https://www.ietf.org/mailman/listinfo/codec
>=20
>=20
>     _______________________________________________
>     codec mailing list
>     codec@ietf.org <mailto:codec@ietf.org>
>     https://www.ietf.org/mailman/listinfo/codec
>=20
>=20
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


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From jean-marc.valin@octasic.com  Mon Jan 24 15:24:26 2011
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From: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Cc: codec@ietf.org, Stephen Botzko <stephen.botzko@gmail.com>
Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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Sorry, this was indeed intended to be informational (probably bad 
copy/paste). Let me fix that.

	Jean-Marc

On 11-01-24 06:26 PM, Peter Saint-Andre wrote:
> I agree with Mary here.
>
> On 1/24/11 4:24 PM, Mary Barnes wrote:
>> The document is currently specified as Standards Track, thus the use of
>> RFC 2119 language is entirely appropriate.  However, ISTM that the
>> document should really just be Informational, in particular given the
>> language in the introduction that this document defines a "suggested
>> process", as opposed to a process that is required for codec development.
>>
>> Regards,
>> Mary.
>>
>> On Mon, Jan 24, 2011 at 5:08 PM, Jean-Marc Valin
>> <jean-marc.valin@octasic.com<mailto:jean-marc.valin@octasic.com>>  wrote:
>>
>>      Christian,
>>
>>      I actually responded to the last comments you made a while ago (oct
>>      2010). One issue I pointed out was your use of RFC2119 keywords,
>>      which (AFAIK) aren't appropriate for a requirements draft (the
>>      requirements aren't a standard). So statements like "Any codec
>>      specified by the IETF MUST be well specified", besides stating the
>>      obvious, are inappropriate.
>>
>>      There were also comments that just did not belong to this draft,
>>      such as the section on collaboration with other WGs. Collaboration
>>      is not a characteristic of a codec. So essentially, I merged the
>>      uncontroversial suggestion, but that's all I could do. As far as I'm
>>      aware, there's nothing in the current draft that goes against the
>>      consensus of the WG. If there are, please point to specific issues
>>      and to statements made by others (not just you) asking for the change.
>>
>>      Cheers,
>>
>>             Jean-Marc
>>
>>
>>      On 11-01-24 03:10 PM, Christian Hoene wrote:
>>
>>          Christian - perhaps you could post a list of the issues you see that
>>          haven't been addressed?
>>
>>          */[Christian Hoene] No Stephen, these issues have been written
>>          down in
>>          previous emails, drafts and issues in the Trac. They can be read
>>          by anybody
>>          anytime. Thus, I do not see any benefit of repeating them again
>>          if the
>>          editors continue to ignore any input. Indeed, they did not
>>          improve the
>>          draft despite sound technical reasons. /*
>>
>>          */Even if somebody is not fully involved in the technical
>>          details: It is
>>          very odd that despite many hundreds emails and many discussions
>>          since
>>          starting this WG the editors have not updated the draft beside minor
>>          changes such as the boilerplate and typos. /*
>>
>>          */Even if the lack of any update was not intentionally, the
>>          editors missed
>>          to do their job because they were too lazy or rather too busy
>>          doing other
>>          thinks./*
>>
>>          */I would be sad if all the fruitful discussions here and all
>>          the good
>>          contributions of many industry experts should have been in vain.
>>          Even if
>>          not all requirements can be met by Opus, a proper requirements
>>          document may
>>          be relevant for future solutions or other SDOs./*
>>
>>          */CH/*
>>
>>
>>
>>
>>          _______________________________________________
>>          codec mailing list
>>          codec@ietf.org<mailto:codec@ietf.org>
>>          https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>      _______________________________________________
>>      codec mailing list
>>      codec@ietf.org<mailto:codec@ietf.org>
>>      https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>


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Subject: Re: [codec] requirements #18 (closed): Frame Sizes?
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#18: Frame Sizes?

Changes (by gmaxwell@â€¦):

  * status:  new => closed
  * resolution:  => worksforme


Comment:

 The frame sizes are driven by the latent requirements. I don't believe
 that there has been any preference expressed for frame sizes independent
 of the latency preferences.

 The requirements documents discusses latency in depth:

 http://tools.ietf.org/id/draft-ietf-codec-requirements-02.txt

 Section 2.1 "The codec delay must be less than 40 ms, but no more than
 25ms is desirable."
 Section 2.2 "For this reason, the codec delay must be less than 30 ms for
 this application.  An optional low-delay mode with less than 10 ms delay
 is desirable, but not required."
 Section 2.5 " While for most applications a codec delay up to 30 ms is
 acceptable, a low-delay (<10 ms) option is highly desirable, especially
 for games with rapid interactions."
 etc.

 I believe the language of the current requirements draft reflects the best
 consensus that we were able to achieveâ€” a fairly open _requirement_ but a
 strong preference for lower delay support for many applications.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |        Owner:            
     Type:  enhancement             |       Status:  closed    
 Priority:  major                   |    Milestone:            
Component:  requirements            |      Version:            
 Severity:  -                       |   Resolution:  worksforme
 Keywords:                          |  
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/18#comment:3>
codec <http://tools.ietf.org/codec/>


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#29: Packet loss concealment?

Changes (by gmaxwell@â€¦):

  * status:  new => closed
  * resolution:  => fixed


Comment:

 This is addressed in the guidelines:

 http://tools.ietf.org/html/draft-valin-codec-guidelines-08
 4.5: "To ensure freedom of implementation, decoder-side only error
 concealment does not need to be specified, although the reference
 implementation should include the same PLC algorithm as used in the
 testing phase.  Is it up to the working group to decide whether minimum
 requirements on PLC quality will be required for compliance with the
 specification.  Obviously, any information signaled in the bitstream
 intended to aid PLC needs to be specified."

 I don't believe the working group has expressed interest in PLC
 performance, and I know that there is at least some opposition to
 requiring specific that implementers provide a particular PLC or PLC
 performance (e.g. I'd be opposed to requiring a particular kind of PLC).
 As such, Jean-Marc is adding the following parallel language to section 3
 of the requirements document:

 "To ensure freedom of implementation, decoder-side only error concealment
 does not need to be specified, although a functional PLC algorithm is
 desirable as part of the codec reference implementation.  Obviously,
 any information signaled in the bitstream intended to aid PLC needs to be
 specified."
 (Available at http://jmvalin.ca/misc_stuff/draft-ietf-codec-
 requirements.txt)

-- 
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Component:  requirements            |      Version:        
 Severity:  -                       |   Resolution:  fixed 
 Keywords:                          |  
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/29#comment:2>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #23 (new): Impact of packet headers and header compression
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#23: Impact of packet headers and header compression


Comment(by gmaxwell@â€¦):

 Vaguely related: http://tools.ietf.org/id/draft-ietf-codec-
 requirements-02.txt
 Section 1. "Codec bit-rates in bits per second (b/s) will be considered
 without counting any overhead (IP/UDP/RTP headers, padding, ...)." and
 Section 4.2 " When comparing the bit-rate of codecs, the overhead of
 IP/UDP/RTP headers should not be considered, but any additional bits
 required in the RTP payload format after the header (e.g. required
 signalling) should be considered. "

 Because header overhead is generally dependent on the transport, and not
 our activities I don't believe that we really have much to require here
 from the codec itself.

 Unless someone has some specific proposed requirement language to suggest,
 I propose that we close this ticket and leave any requirements related to
 RTP packetization to the specific RFCs describing the packetization.

-- 
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     Type:  enhancement             |      Status:  new
 Priority:  major                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/23#comment:2>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #32 (new): Playout/Dejittering Buffer
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#32: Playout/Dejittering Buffer


Comment(by gmaxwell@â€¦):

 I believe the complete system requirements and detailed testing
 methodology is out of scope for the CODEC requirements document.

 I believe the requirements stipulation on real world testing is satisfied
 by whatever basically reasonable methodology is applied so long as it is
 applied uniformly and does not offend the consensus of the working group.
 In other words, I see no reason for our permissive testing requirement to
 force us to import detailed systems requirements into the codec
 requirement document, no more than does the mention of "quality" in the
 document demand that we specific a particular headphone manufacturer for
 listening tests in this document.

 Am I missing any reason why this ticket can't be closed?

-- 
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 Severity:  Active WG Document      |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/32#comment:2>
codec <http://tools.ietf.org/codec/>


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#27: Testing Impact of IP on Codec Performance

Changes (by gmaxwell@â€¦):

  * status:  new => closed
  * resolution:  => duplicate


Comment:

 I believe that we can fold this one into #32.

-- 
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 Severity:  Active WG Document      |   Resolution:  duplicate
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------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/27#comment:2>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #10 (new): Multi-channel frame ordering?
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#10: Multi-channel frame ordering?


Comment(by gmaxwell@â€¦):

 I believe this is a primarily codec design question. E.g. How the bits are
 written onto the wire.

 The requirements draft covers our multichannel requirements in some of the
 applications sections (e.g. 2.3) and in Section 5.5.

 I believe that we can either close this ticket or convert it to a ticket
 regarding the Opus draft.

-- 
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 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/10#comment:1>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #31 (new): Requirements of high-density VoIP gateways (and low cost VoIP phone)?
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#31: Requirements of high-density VoIP gateways (and low cost VoIP phone)?


Comment(by gmaxwell@â€¦):

 Section 4.4 in the requirement speaks to the consensus demands for
 complexity, and sections 5.1 on mixing addresses conferencing.

 Unless someone has specific requirements language to propose covering
 omitted requirements, I believe that document as standing pays sufficient
 attention to complexity and latency that no further action is required on
 this point.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  enhancement             |      Status:  new
 Priority:  major                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  Active WG Document      |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/31#comment:2>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #34 (new): Speech Quality Aspects in emergency calls
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#34: Speech Quality Aspects in emergency calls


Comment(by gmaxwell@â€¦):

 It isn't clear to me that this subject area has any particular interaction
 with the codec bitstream. I don't doubt that its an important areas, but I
 believe that its primarily a product of the implementation and the system
 and not the codec bitstream.

 Certainly no codec specification is going to require the codec to fail to
 generate packets while the system is requiring it to generate packets.

 As we lack any proposed requirements language is there any opposition to
 closing this ticket?

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  defect                  |      Status:  new
 Priority:  minor                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/34#comment:1>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #8 (new): Sample rates?
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#8: Sample rates?


Comment(by gmaxwell@â€¦):

 I believe the requirements aspect of this has been addressed.  That there
 is consensus that full-band operation is a requirement, and that other
 rates/behaviors are possibly for consideration but not required.

 Have I misunderstood the situation?

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:  jean-marc.valin@â€¦             
     Type:  enhancement             |      Status:  new                           
 Priority:  minor                   |   Milestone:                                
Component:  requirements            |     Version:                                
 Severity:  Active WG Document      |    Keywords:                                
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:3>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #9 (closed): Joint stereo/channel coding?
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#9: Joint stereo/channel coding?

Changes (by gmaxwell@â€¦):

  * status:  new => closed
  * resolution:  => worksforme


Comment:

 The requirements document addresses this. Efficient stereo encoding is
 desirable but not required.

 http://tools.ietf.org/id/draft-ietf-codec-requirements-02.txt

 5.5.  Stereo support

    It is highly desirable for the codec to have stereo support.  At a
    minimum, the codec should be able to encode two channels
    independently without causing significant stereo image artefacts.  It
    is also desirable for the codec to take advantage of the inter-
    channel redundancy in stereo audio to reduce the bitrate (for an
    equivalent quality) of stereo audio compared to coding channels
    independently.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |        Owner:            
     Type:  enhancement             |       Status:  closed    
 Priority:  minor                   |    Milestone:            
Component:  requirements            |      Version:            
 Severity:  Active WG Document      |   Resolution:  worksforme
 Keywords:                          |  
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/9#comment:1>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #8 (new): Sample rates?
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I would like to see 8 and 16 KHz as required rates for the codec to insure
interoperability with existing narrowband and wideband codecs. In other
words we should be able to negotiate 8 or 16 Khz sample rate if audio will
be transcoded for PSTN or wideband codec such as G.722
_____________
Roman Shpount


On Mon, Jan 24, 2011 at 7:10 PM, codec issue tracker <trac@tools.ietf.org>w=
rote:

> #8: Sample rates?
>
>
> Comment(by gmaxwell@=85):
>
>  I believe the requirements aspect of this has been addressed.  That ther=
e
>  is consensus that full-band operation is a requirement, and that other
>  rates/behaviors are possibly for consideration but not required.
>
>  Have I misunderstood the situation?
>
> --
>
> ------------------------------------+------------------------------------=
---
>  Reporter:  hoene@=85                 |       Owner:  jean-marc.valin@=85
>     Type:  enhancement             |      Status:  new
>  Priority:  minor                   |   Milestone:
> Component:  requirements            |     Version:
>  Severity:  Active WG Document      |    Keywords:
>
> ------------------------------------+------------------------------------=
---
>
> Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:3>
> codec <http://tools.ietf.org/codec/>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
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>

--0021cc022382721d5d049aa098c9
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I would like to see 8 and 16 KHz as required rates for the codec to insure =
interoperability with existing narrowband and wideband codecs. In other wor=
ds we should be able to negotiate 8 or 16 Khz sample rate if audio will be =
transcoded for PSTN or wideband codec such as G.722<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Mon, Jan 24, 2011 at 7:10 PM, codec i=
ssue tracker <span dir=3D"ltr">&lt;<a href=3D"mailto:trac@tools.ietf.org">t=
rac@tools.ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204,=
 204); padding-left: 1ex;">
#8: Sample rates?<br>
<br>
<br>
Comment(by gmaxwell@=85):<br>
<br>
=A0I believe the requirements aspect of this has been addressed. =A0That th=
ere<br>
=A0is consensus that full-band operation is a requirement, and that other<b=
r>
=A0rates/behaviors are possibly for consideration but not required.<br>
<br>
=A0Have I misunderstood the situation?<br>
<br>
--<br>
------------------------------------+--------------------------------------=
-<br>
=A0Reporter: =A0hoene@=85 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =A0 Own=
er: =A0jean-marc.valin@=85<br>
 =A0 =A0 Type: =A0enhancement =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =A0Status: =
=A0new<br>
=A0Priority: =A0minor =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 Milestone:<=
br>
Component: =A0requirements =A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0 Version:<br>
=A0Severity: =A0Active WG Document =A0 =A0 =A0| =A0 =A0Keywords:<br>
------------------------------------+--------------------------------------=
-<br>
<br>
Ticket URL: &lt;<a href=3D"http://trac.tools.ietf.org/wg/codec/trac/ticket/=
8#comment:3" target=3D"_blank">http://trac.tools.ietf.org/wg/codec/trac/tic=
ket/8#comment:3</a>&gt;<br>
codec &lt;<a href=3D"http://tools.ietf.org/codec/" target=3D"_blank">http:/=
/tools.ietf.org/codec/</a>&gt;<br>
<br>
_______________________________________________<br>
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</blockquote></div><br>

--0021cc022382721d5d049aa098c9--

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#28: Layered bit-stream

Changes (by gmaxwell@â€¦):

  * status:  new => closed
  * resolution:  => worksforme


Comment:

 I haven't seen any recent discussion or interest in changing the current
 "desirable but not mandatory" language in the requirements draft.

 http://tools.ietf.org/id/draft-ietf-codec-requirements-02.txt
 5.3.  Layered bit-stream

    A layered codec makes it possible to transmit only a certain subset
    of the bits and still obtain a valid bit-stream with a quality that
    is equivalent to the quality that would be obtained from encoding at
    the corresponding rate.  While this is not a necessary feature for
    most applications, it can be desirable for cases where a "mixing
    server" needs to handle a large number of streams with limited
    computational resources.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |        Owner:            
     Type:  defect                  |       Status:  closed    
 Priority:  minor                   |    Milestone:            
Component:  requirements            |      Version:            
 Severity:  Active WG Document      |   Resolution:  worksforme
 Keywords:                          |  
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/28#comment:3>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #8 (new): Sample rates?
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I believe such requirement for narrowband/wideband is already present, but 
I don't mind making it even more explicit is necessary.

     Jean-Marc

On 11-01-24 07:14 PM, Roman Shpount wrote:
> I would like to see 8 and 16 KHz as required rates for the codec to insure
> interoperability with existing narrowband and wideband codecs. In other
> words we should be able to negotiate 8 or 16 Khz sample rate if audio will
> be transcoded for PSTN or wideband codec such as G.722
> _____________
> Roman Shpount
>
>
> On Mon, Jan 24, 2011 at 7:10 PM, codec issue tracker <trac@tools.ietf.org
> <mailto:trac@tools.ietf.org>> wrote:
>
>     #8: Sample rates?
>
>
>     Comment(by gmaxwell@…):
>
>     I believe the requirements aspect of this has been addressed. That there
>     is consensus that full-band operation is a requirement, and that other
>     rates/behaviors are possibly for consideration but not required.
>
>     Have I misunderstood the situation?
>
>     --
>     ------------------------------------+---------------------------------------
>     Reporter: hoene@… | Owner: jean-marc.valin@…
>     Type: enhancement | Status: new
>     Priority: minor | Milestone:
>     Component: requirements | Version:
>     Severity: Active WG Document | Keywords:
>     ------------------------------------+---------------------------------------
>
>     Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:3>
>     codec <http://tools.ietf.org/codec/>
>
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#19: How large is the frame size depended delay / the serialization delay /
frame size depended processing delay?


Comment(by gmaxwell@â€¦):

 There  was a large amount of interesting discussion on this subject, but I
 don't believe that any of it was moving in the direction of creating
 requirements language beyond narrowing our existing latency requirements.

 I believe there was ultimately no consensus to further narrow the latency
 requirements beyond what was established in the current requirements
 document and, accordingly, no further action is required on this
 particular ticket.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  enhancement             |      Status:  new
 Priority:  minor                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/19#comment:6>
codec <http://tools.ietf.org/codec/>


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#8: Sample rates?


Comment(by gmaxwell@â€¦):

 On 11-01-24 07:14 PM, Roman Shpount wrote:
 > I would like to see 8 and 16 KHz as required rates for the codec to
 insure
 > interoperability with existing narrowband and wideband codecs. In other
 > words we should be able to negotiate 8 or 16 Khz sample rate if audio
 will
 > be transcoded for PSTN or wideband codec such as G.722

 Jean-Marc wrote:
 > I believe such requirement for narrowband/wideband is already present,
 but
 > I don't mind making it even more explicit is necessary.



 I believe that we since are close enough to finalizing the draft we should
 try to include proposed language with our issues.  Otherwise there may be
 no clear path forward.

 Some of the requirements specify that compatibility with
 wideband/narrowband is important, but for the avoidance of doubt, I'll
 suggest:

 At the end of 4.1.  Operating space:

 "Because interoperation with existing wideband and narrowband facilities
 is essential at least one method of interoperation must be provided
 regardless of the codec's operating mode, sample rate, or bitrate."

 Of course, I would be perfectly happy to leave this out entirely (as I
 believe the application section 2.1 already implies the requirement) or
 use some other language.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:  jean-marc.valin@â€¦             
     Type:  enhancement             |      Status:  new                           
 Priority:  minor                   |   Milestone:                                
Component:  requirements            |     Version:                                
 Severity:  Active WG Document      |    Keywords:                                
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:4>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #24 (new): Negotiation of codec parameters?
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#24: Negotiation of codec parameters?


Comment(by gmaxwell@â€¦):

 Section 3. of draft-ietf-codec-requirements-02 provides:

 "If any codec parameters need to be negotiated between end-points, the
 negotiation should be as easy as possible to carry over SIP/SDP or
 alternatively over XMPP/Jingle."

 Is anything further required?

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |       Owner:     
     Type:  enhancement             |      Status:  new
 Priority:  minor                   |   Milestone:     
Component:  requirements            |     Version:     
 Severity:  -                       |    Keywords:     
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/24#comment:2>
codec <http://tools.ietf.org/codec/>


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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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#12: bit-exact vs. bit-compatible?

Changes (by gmaxwell@â€¦):

  * status:  new => closed
  * resolution:  => worksforme


Comment:

 http://www.ietf.org/proceedings/78/minutes/codec.txt

 "On the topic of bit exact. Consensus was bit exactness is not required."

 I believe this issue is already closed.

-- 
------------------------------------+---------------------------------------
 Reporter:  hoene@â€¦                 |        Owner:            
     Type:  enhancement             |       Status:  closed    
 Priority:  minor                   |    Milestone:            
Component:  requirements            |      Version:            
 Severity:  Active WG Document      |   Resolution:  worksforme
 Keywords:                          |  
------------------------------------+---------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
codec <http://tools.ietf.org/codec/>


From roman@telurix.com  Mon Jan 24 17:05:01 2011
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Date: Mon, 24 Jan 2011 20:07:54 -0500
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [codec] requirements #8 (new): Sample rates?
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--002215046b87a0770d049aa1589f
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I actually do believe that the requirement is already addressed by section
4.1 of the document. I was responding to a comment that only full band is
required, which is clearly not what we agreed upon and not what's in the
document.
_____________
Roman Shpount


On Mon, Jan 24, 2011 at 7:40 PM, codec issue tracker <trac@tools.ietf.org>w=
rote:

> #8: Sample rates?
>
>
> Comment(by gmaxwell@=85):
>
>  On 11-01-24 07:14 PM, Roman Shpount wrote:
>  > I would like to see 8 and 16 KHz as required rates for the codec to
>  insure
>  > interoperability with existing narrowband and wideband codecs. In othe=
r
>  > words we should be able to negotiate 8 or 16 Khz sample rate if audio
>  will
>  > be transcoded for PSTN or wideband codec such as G.722
>
>  Jean-Marc wrote:
>  > I believe such requirement for narrowband/wideband is already present,
>  but
>  > I don't mind making it even more explicit is necessary.
>
>
>
>  I believe that we since are close enough to finalizing the draft we shou=
ld
>  try to include proposed language with our issues.  Otherwise there may b=
e
>  no clear path forward.
>
>  Some of the requirements specify that compatibility with
>  wideband/narrowband is important, but for the avoidance of doubt, I'll
>  suggest:
>
>  At the end of 4.1.  Operating space:
>
>  "Because interoperation with existing wideband and narrowband facilities
>  is essential at least one method of interoperation must be provided
>  regardless of the codec's operating mode, sample rate, or bitrate."
>
>  Of course, I would be perfectly happy to leave this out entirely (as I
>  believe the application section 2.1 already implies the requirement) or
>  use some other language.
>
> --
>
> ------------------------------------+------------------------------------=
---
>  Reporter:  hoene@=85                 |       Owner:  jean-marc.valin@=85
>     Type:  enhancement             |      Status:  new
>  Priority:  minor                   |   Milestone:
> Component:  requirements            |     Version:
>  Severity:  Active WG Document      |    Keywords:
>
> ------------------------------------+------------------------------------=
---
>
> Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:4>
> codec <http://tools.ietf.org/codec/>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

--002215046b87a0770d049aa1589f
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I actually do believe that the requirement is already addressed by section =
4.1 of the document. I was responding to a comment that only full band is r=
equired, which is clearly not what we agreed upon and not what&#39;s in the=
 document.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Mon, Jan 24, 2011 at 7:40 PM, codec i=
ssue tracker <span dir=3D"ltr">&lt;<a href=3D"mailto:trac@tools.ietf.org">t=
rac@tools.ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204,=
 204); padding-left: 1ex;">
<div class=3D"im">#8: Sample rates?<br>
<br>
<br>
Comment(by gmaxwell@=85):<br>
<br>
</div><div class=3D"im">=A0On 11-01-24 07:14 PM, Roman Shpount wrote:<br>
=A0&gt; I would like to see 8 and 16 KHz as required rates for the codec to=
<br>
=A0insure<br>
=A0&gt; interoperability with existing narrowband and wideband codecs. In o=
ther<br>
=A0&gt; words we should be able to negotiate 8 or 16 Khz sample rate if aud=
io<br>
=A0will<br>
=A0&gt; be transcoded for PSTN or wideband codec such as G.722<br>
<br>
</div><div class=3D"im">=A0Jean-Marc wrote:<br>
=A0&gt; I believe such requirement for narrowband/wideband is already prese=
nt,<br>
=A0but<br>
=A0&gt; I don&#39;t mind making it even more explicit is necessary.<br>
<br>
<br>
<br>
</div>=A0I believe that we since are close enough to finalizing the draft w=
e should<br>
=A0try to include proposed language with our issues. =A0Otherwise there may=
 be<br>
=A0no clear path forward.<br>
<br>
=A0Some of the requirements specify that compatibility with<br>
=A0wideband/narrowband is important, but for the avoidance of doubt, I&#39;=
ll<br>
=A0suggest:<br>
<br>
=A0At the end of 4.1. =A0Operating space:<br>
<br>
=A0&quot;Because interoperation with existing wideband and narrowband facil=
ities<br>
=A0is essential at least one method of interoperation must be provided<br>
=A0regardless of the codec&#39;s operating mode, sample rate, or bitrate.&q=
uot;<br>
<br>
=A0Of course, I would be perfectly happy to leave this out entirely (as I<b=
r>
=A0believe the application section 2.1 already implies the requirement) or<=
br>
=A0use some other language.<br>
<br>
--<br>
<div class=3D"im">------------------------------------+--------------------=
-------------------<br>
=A0Reporter: =A0hoene@=85 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =A0 Own=
er: =A0jean-marc.valin@=85<br>
 =A0 =A0 Type: =A0enhancement =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =A0Status: =
=A0new<br>
=A0Priority: =A0minor =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 Milestone:<=
br>
Component: =A0requirements =A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0 Version:<br>
=A0Severity: =A0Active WG Document =A0 =A0 =A0| =A0 =A0Keywords:<br>
------------------------------------+--------------------------------------=
-<br>
<br>
</div>Ticket URL: &lt;<a href=3D"http://trac.tools.ietf.org/wg/codec/trac/t=
icket/8#comment:4" target=3D"_blank">http://trac.tools.ietf.org/wg/codec/tr=
ac/ticket/8#comment:4</a>&gt;<br>
<div class=3D"im">codec &lt;<a href=3D"http://tools.ietf.org/codec/" target=
=3D"_blank">http://tools.ietf.org/codec/</a>&gt;<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
</div><div><div></div><div class=3D"h5"><a href=3D"https://www.ietf.org/mai=
lman/listinfo/codec" target=3D"_blank">https://www.ietf.org/mailman/listinf=
o/codec</a><br>
</div></div></blockquote></div><br>

--002215046b87a0770d049aa1589f--

From stewe@stewe.org  Mon Jan 24 19:59:27 2011
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Date: Mon, 24 Jan 2011 20:02:12 -0800
From: Stephan Wenger <stewe@stewe.org>
To: <codec@ietf.org>, <gmaxwell@juniper.net>
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Thread-Topic: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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Hi all:

Let me speak once more against this decision (if such a decision were
really made; see the p.s.).

There are currently three IPR disclosures against the codec draft and/or
its predecessors.  The Xiph disclosure is at this point a placeholder
(Xiph folks: it's time to fix that!).  However, the two other disclosures
on file provide a patent grant only for necessary patent claims and only
when the standard is practiced in full compliance.  These terms (in
various formulations) are quite common.

In order to ensure one has a license (or can rely on a non-assert
covenant), one has to ensure one meets the conditions set by the
rightholder.  On stuff such as reciprocity clauses this is simple.  On
compliance, it's not always easy.

The traditional compliance test for a media codec is a stimulus-response
test: you feed test vectors into the codec, and you get results.  If the
results match, you are in compliance, if not, you are not.  Simple.

Without bit exactness, the compliance criteria have to be defined
differently.  We can do so, and, indeed, I recall that this has been
mentioned as one plan forward.  However, I have seen zero activity in this
direction, and I have also not seen any language that mentions this in the
requirements draft.  I think that the subject of compliance tests, at
least in its most basic outline, needs to be documented in the
requirements draft.  The details can be taken care of elsewhere and later,
but not too much later.  It should be clear that a codec candidate (if
there were more than one) needs to have compliance criteria defined before
that codec candidate can become an RFC.  Without that, the key goal of the
WG, a reasonably freely practicable codec, is just not achievable in the
current legal environment (which includes, in this case, the IPR
disclosures on file).

Of course, it would be sooooo much simpler if we would mandate a bit exact
decoder...  Is it really that restricting to require that?

Stephan

P.s.: for the IETF procedures newcomers: humms taken at meetings need to
be confirmed on a mailing list, and consensus needs to be declared by the
chairs.  On this subject, I do recall mailing list discussions after
Maastricht, but I do not recall that consensus was reached, yet alone
declared.  (Unfortunately, I currently don't have the time to go through
the mailing list archives to verify my recollection; Sorry.)
=20


On 1.24.2011 16:45 , "codec issue tracker" <trac@tools.ietf.org> wrote:

>#12: bit-exact vs. bit-compatible?
>
>Changes (by gmaxwell@=8A):
>
>  * status:  new =3D> closed
>  * resolution:  =3D> worksforme
>
>
>Comment:
>
> http://www.ietf.org/proceedings/78/minutes/codec.txt
>
> "On the topic of bit exact. Consensus was bit exactness is not required."
>
> I believe this issue is already closed.
>
>--=20
>------------------------------------+-------------------------------------
>--
> Reporter:  hoene@=8A                 |        Owner:
>     Type:  enhancement             |       Status:  closed
> Priority:  minor                   |    Milestone:
>Component:  requirements            |      Version:
> Severity:  Active WG Document      |   Resolution:  worksforme
> Keywords:                          |
>------------------------------------+-------------------------------------
>--
>
>Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>codec <http://tools.ietf.org/codec/>
>
>_______________________________________________
>codec mailing list
>codec@ietf.org
>https://www.ietf.org/mailman/listinfo/codec



From jmvalin@jmvalin.ca  Mon Jan 24 21:01:07 2011
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Date: Tue, 25 Jan 2011 00:03:06 -0500
From: Jean-Marc Valin <jmvalin@jmvalin.ca>
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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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Hi Stephan,

I understand your concern and I'd be interested if you have alternative 
ways of handling the licensing to avoid any issue. It's not like this is 
a unique situation. As far as I know, most (all?) MPEG codecs have 
similar non-bit exact definitions. I have also heard that they also 
require some IPR licensing...

In general the issue of bit-exactness has been discussed and so far I 
don't recall many arguing in favor of a bit-exact definition. Most of 
the concerns that have been expressed are solved by considering that 
non-bitexact does not mean you cannot be bit-exact with the reference 
encoder. It merely means that you don't *have* to. So regardless of how 
conformance is defined exactly, one always has the option of being 
bit-exact with the reference implementation, which obviously guarantees 
compliance.

As for language mentioning compliance, I believe it belongs more to the 
guidelines (it's not a requirement of the codec itself), which includes 
the following text:

    4.  To reduce the risk of bias towards certain CPU/DSP architectures,
        ideally the decoder specification should not require "bit-exact"
        conformance with the reference implementation.  The output of a
        decoder implementation should only be "close enough" to the
        output of the reference decoder.  A comparison tool should be
        provided along with the codec to verify objectively that the
        output of a decoder is likely to be perceptually
        indistinguishable from that of the reference decoder.  However,
        an implementation may still wish to produce an output that is
        bit-exact with the reference implementation to simplify the
        testing procedure.



Cheers,

	Jean-Marc

On 11-01-24 11:02 PM, Stephan Wenger wrote:
> Hi all:
>
> Let me speak once more against this decision (if such a decision were
> really made; see the p.s.).
>
> There are currently three IPR disclosures against the codec draft and/or
> its predecessors.  The Xiph disclosure is at this point a placeholder
> (Xiph folks: it's time to fix that!).  However, the two other disclosures
> on file provide a patent grant only for necessary patent claims and only
> when the standard is practiced in full compliance.  These terms (in
> various formulations) are quite common.
>
> In order to ensure one has a license (or can rely on a non-assert
> covenant), one has to ensure one meets the conditions set by the
> rightholder.  On stuff such as reciprocity clauses this is simple.  On
> compliance, it's not always easy.
>
> The traditional compliance test for a media codec is a stimulus-response
> test: you feed test vectors into the codec, and you get results.  If the
> results match, you are in compliance, if not, you are not.  Simple.
>
> Without bit exactness, the compliance criteria have to be defined
> differently.  We can do so, and, indeed, I recall that this has been
> mentioned as one plan forward.  However, I have seen zero activity in this
> direction, and I have also not seen any language that mentions this in the
> requirements draft.  I think that the subject of compliance tests, at
> least in its most basic outline, needs to be documented in the
> requirements draft.  The details can be taken care of elsewhere and later,
> but not too much later.  It should be clear that a codec candidate (if
> there were more than one) needs to have compliance criteria defined before
> that codec candidate can become an RFC.  Without that, the key goal of the
> WG, a reasonably freely practicable codec, is just not achievable in the
> current legal environment (which includes, in this case, the IPR
> disclosures on file).
>
> Of course, it would be sooooo much simpler if we would mandate a bit exact
> decoder...  Is it really that restricting to require that?
>
> Stephan
>
> P.s.: for the IETF procedures newcomers: humms taken at meetings need to
> be confirmed on a mailing list, and consensus needs to be declared by the
> chairs.  On this subject, I do recall mailing list discussions after
> Maastricht, but I do not recall that consensus was reached, yet alone
> declared.  (Unfortunately, I currently don't have the time to go through
> the mailing list archives to verify my recollection; Sorry.)
>
>
>
> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org>  wrote:
>
>> #12: bit-exact vs. bit-compatible?
>>
>> Changes (by gmaxwell@Š):
>>
>>   * status:  new =>  closed
>>   * resolution:  =>  worksforme
>>
>>
>> Comment:
>>
>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>
>> "On the topic of bit exact. Consensus was bit exactness is not required."
>>
>> I believe this issue is already closed.
>>
>> --
>> ------------------------------------+-------------------------------------
>> --
>> Reporter:  hoene@Š                 |        Owner:
>>      Type:  enhancement             |       Status:  closed
>> Priority:  minor                   |    Milestone:
>> Component:  requirements            |      Version:
>> Severity:  Active WG Document      |   Resolution:  worksforme
>> Keywords:                          |
>> ------------------------------------+-------------------------------------
>> --
>>
>> Ticket URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>> codec<http://tools.ietf.org/codec/>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>
>

From john.elwell@siemens-enterprise.com  Tue Jan 25 01:59:15 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>
Date: Tue, 25 Jan 2011 11:02:04 +0100
Thread-Topic: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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Cc: "codec@ietf.org" <codec@ietf.org>, Stephen Botzko <stephen.botzko@gmail.com>
Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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Mary,

I am confused which document we are talking about. The subject indicates th=
e requirements document, but your quoted text "suggested process" seems to =
come from the guidelines document.

The situation with the requirements document (draft-ietf-codec-requirements=
-02) is that it is intended as Standards Track but contains no RFC 2119 lan=
guage (as far as I can see, e.g., no "MUST" statements, and no reference to=
 RFC 2119).

I think it should indeed be Informational, but this does not prevent us hav=
ing normative statements in which the codec specification that this WG prod=
uces is the subject, e.g., "The codec MUST ..." This would make it much eas=
ier to pick out the actual requirements and to measure the resulting codec =
specification against those requirements. Some WGs adopt the practice of nu=
mbering requirements, for easy reference.

John

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org]=20
> On Behalf Of Mary Barnes
> Sent: 24 January 2011 23:25
> To: Jean-Marc Valin
> Cc: codec@ietf.org; Stephen Botzko
> Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
>=20
> The document is currently specified as Standards Track, thus=20
> the use of RFC 2119 language is entirely appropriate. =20
> However, ISTM that the document should really just be=20
> Informational, in particular given the language in the=20
> introduction that this document defines a "suggested=20
> process", as opposed to a process that is required for codec=20
> development.  =20
>=20
>=20
> Regards,
> Mary.
>=20
>=20
> On Mon, Jan 24, 2011 at 5:08 PM, Jean-Marc Valin=20
> <jean-marc.valin@octasic.com> wrote:
>=20
>=20
> 	Christian,
> =09
> 	I actually responded to the last comments you made a=20
> while ago (oct 2010). One issue I pointed out was your use of=20
> RFC2119 keywords, which (AFAIK) aren't appropriate for a=20
> requirements draft (the requirements aren't a standard). So=20
> statements like "Any codec specified by the IETF MUST be well=20
> specified", besides stating the obvious, are inappropriate.
> =09
> 	There were also comments that just did not belong to=20
> this draft, such as the section on collaboration with other=20
> WGs. Collaboration is not a characteristic of a codec. So=20
> essentially, I merged the uncontroversial suggestion, but=20
> that's all I could do. As far as I'm aware, there's nothing=20
> in the current draft that goes against the consensus of the=20
> WG. If there are, please point to specific issues and to=20
> statements made by others (not just you) asking for the change.
> =09
> 	Cheers,
> =09
> 	       Jean-Marc=20
>=20
>=20
> 	On 11-01-24 03:10 PM, Christian Hoene wrote:
> =09
>=20
> 		Christian - perhaps you could post a list of=20
> the issues you see that
> 		haven't been addressed?
> 	=09
> 		*/[Christian Hoene] No Stephen, these issues=20
> have been written down in
> 		previous emails, drafts and issues in the Trac.=20
> They can be read by anybody
> 		anytime. Thus, I do not see any benefit of=20
> repeating them again if the
> 		editors continue to ignore any input. Indeed,=20
> they did not improve the
> 		draft despite sound technical reasons. /*
> 	=09
> 		*/Even if somebody is not fully involved in the=20
> technical details: It is
> 		very odd that despite many hundreds emails and=20
> many discussions since
> 		starting this WG the editors have not updated=20
> the draft beside minor
> 		changes such as the boilerplate and typos. /*
> 	=09
> 		*/Even if the lack of any update was not=20
> intentionally, the editors missed
> 		to do their job because they were too lazy or=20
> rather too busy doing other
> 		thinks./*
> 	=09
> 		*/I would be sad if all the fruitful=20
> discussions here and all the good
> 		contributions of many industry experts should=20
> have been in vain. Even if
> 		not all requirements can be met by Opus, a=20
> proper requirements document may
> 		be relevant for future solutions or other SDOs./*
> 	=09
> 	=09
> 		*/CH/*=20
>=20
>=20
>=20
>=20
> 		_______________________________________________
> 		codec mailing list
> 		codec@ietf.org
> 		https://www.ietf.org/mailman/listinfo/codec
> 	=09
>=20
>=20
> 	_______________________________________________
> 	codec mailing list
> 	codec@ietf.org
> 	https://www.ietf.org/mailman/listinfo/codec
> =09
>=20
>=20
> =

From hoene@uni-tuebingen.de  Tue Jan 25 05:09:59 2011
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: "'Jean-Marc Valin'" <jean-marc.valin@octasic.com>
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Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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Hi Jean-Marc Valin,

you are referring to the changes in the guideline doc? 

Thank you for considering my comments from October last year. I have to
admit that I have not read the latest version of the guidelines document so
far. I will give you updated comments (with any) well before the next
deadline.

With best regards,

 Christian 


> -----Original Message-----
> From: Jean-Marc Valin [mailto:jean-marc.valin@octasic.com]
> Sent: Tuesday, January 25, 2011 12:09 AM
> To: Christian Hoene
> Cc: 'Stephen Botzko'; codec@ietf.org
> Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
> 
> Christian,
> 
> I actually responded to the last comments you made a while ago (oct 2010).
> One issue I pointed out was your use of RFC2119 keywords, which (AFAIK)
> aren't appropriate for a requirements draft (the requirements aren't a
> standard). So statements like "Any codec specified by the IETF MUST be
well
> specified", besides stating the obvious, are inappropriate.
> 
> There were also comments that just did not belong to this draft, such as
the
> section on collaboration with other WGs. Collaboration is not a
characteristic
> of a codec. So essentially, I merged the uncontroversial suggestion, but
that's
> all I could do. As far as I'm aware, there's nothing in the current draft
that
> goes against the consensus of the WG. If there are, please point to
specific
> issues and to statements made by others (not just you) asking for the
> change.
> 
> Cheers,
> 
> 	Jean-Marc
> 
> On 11-01-24 03:10 PM, Christian Hoene wrote:
> > Christian - perhaps you could post a list of the issues you see that
> > haven't been addressed?
> >
> > */[Christian Hoene] No Stephen, these issues have been written down in
> > previous emails, drafts and issues in the Trac. They can be read by
> > anybody anytime. Thus, I do not see any benefit of repeating them
> > again if the editors continue to ignore any input. Indeed, they did
> > not improve the draft despite sound technical reasons. /*
> >
> > */Even if somebody is not fully involved in the technical details: It
> > is very odd that despite many hundreds emails and many discussions
> > since starting this WG the editors have not updated the draft beside
> > minor changes such as the boilerplate and typos. /*
> >
> > */Even if the lack of any update was not intentionally, the editors
> > missed to do their job because they were too lazy or rather too busy
> > doing other
> > thinks./*
> >
> > */I would be sad if all the fruitful discussions here and all the good
> > contributions of many industry experts should have been in vain. Even
> > if not all requirements can be met by Opus, a proper requirements
> > document may be relevant for future solutions or other SDOs./*
> >
> > */CH/*
> >
> >
> >
> > _______________________________________________
> > codec mailing list
> > codec@ietf.org
> > https://www.ietf.org/mailman/listinfo/codec


From roman@telurix.com  Tue Jan 25 06:31:37 2011
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Date: Tue, 25 Jan 2011 09:34:30 -0500
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From: Roman Shpount <roman@telurix.com>
To: Jean-Marc Valin <jmvalin@jmvalin.ca>
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Cc: codec@ietf.org
Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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One more concern that I have related to bit exactness is codec regression
testing. One can produce a non-bit-exact encoder that produces reasonable
results for a reference decoder, but if paired with a modified,
non-bit-exact decoder will produce significant audio artifacts. Since
neither decoder or encoder are bit exact we will need to have a test
procedure that will validate that both encoder or decoder will not break an=
y
standard compliant and non-bit-exact encoders or decoders. Not really sure
how this can be done.

I might be wrong, but I think MPEG decoders are bit exact and encoders are
not.
_____________
Roman Shpount


On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin <jmvalin@jmvalin.ca>wrote=
:

> Hi Stephan,
>
> I understand your concern and I'd be interested if you have alternative
> ways of handling the licensing to avoid any issue. It's not like this is =
a
> unique situation. As far as I know, most (all?) MPEG codecs have similar
> non-bit exact definitions. I have also heard that they also require some =
IPR
> licensing...
>
> In general the issue of bit-exactness has been discussed and so far I don=
't
> recall many arguing in favor of a bit-exact definition. Most of the conce=
rns
> that have been expressed are solved by considering that non-bitexact does
> not mean you cannot be bit-exact with the reference encoder. It merely me=
ans
> that you don't *have* to. So regardless of how conformance is defined
> exactly, one always has the option of being bit-exact with the reference
> implementation, which obviously guarantees compliance.
>
> As for language mentioning compliance, I believe it belongs more to the
> guidelines (it's not a requirement of the codec itself), which includes t=
he
> following text:
>
>   4.  To reduce the risk of bias towards certain CPU/DSP architectures,
>       ideally the decoder specification should not require "bit-exact"
>       conformance with the reference implementation.  The output of a
>       decoder implementation should only be "close enough" to the
>       output of the reference decoder.  A comparison tool should be
>       provided along with the codec to verify objectively that the
>       output of a decoder is likely to be perceptually
>       indistinguishable from that of the reference decoder.  However,
>       an implementation may still wish to produce an output that is
>       bit-exact with the reference implementation to simplify the
>       testing procedure.
>
>
>
> Cheers,
>
>        Jean-Marc
>
>
> On 11-01-24 11:02 PM, Stephan Wenger wrote:
>
>> Hi all:
>>
>> Let me speak once more against this decision (if such a decision were
>> really made; see the p.s.).
>>
>> There are currently three IPR disclosures against the codec draft and/or
>> its predecessors.  The Xiph disclosure is at this point a placeholder
>> (Xiph folks: it's time to fix that!).  However, the two other disclosure=
s
>> on file provide a patent grant only for necessary patent claims and only
>> when the standard is practiced in full compliance.  These terms (in
>> various formulations) are quite common.
>>
>> In order to ensure one has a license (or can rely on a non-assert
>> covenant), one has to ensure one meets the conditions set by the
>> rightholder.  On stuff such as reciprocity clauses this is simple.  On
>> compliance, it's not always easy.
>>
>> The traditional compliance test for a media codec is a stimulus-response
>> test: you feed test vectors into the codec, and you get results.  If the
>> results match, you are in compliance, if not, you are not.  Simple.
>>
>> Without bit exactness, the compliance criteria have to be defined
>> differently.  We can do so, and, indeed, I recall that this has been
>> mentioned as one plan forward.  However, I have seen zero activity in th=
is
>> direction, and I have also not seen any language that mentions this in t=
he
>> requirements draft.  I think that the subject of compliance tests, at
>> least in its most basic outline, needs to be documented in the
>> requirements draft.  The details can be taken care of elsewhere and late=
r,
>> but not too much later.  It should be clear that a codec candidate (if
>> there were more than one) needs to have compliance criteria defined befo=
re
>> that codec candidate can become an RFC.  Without that, the key goal of t=
he
>> WG, a reasonably freely practicable codec, is just not achievable in the
>> current legal environment (which includes, in this case, the IPR
>> disclosures on file).
>>
>> Of course, it would be sooooo much simpler if we would mandate a bit exa=
ct
>> decoder...  Is it really that restricting to require that?
>>
>> Stephan
>>
>> P.s.: for the IETF procedures newcomers: humms taken at meetings need to
>> be confirmed on a mailing list, and consensus needs to be declared by th=
e
>> chairs.  On this subject, I do recall mailing list discussions after
>> Maastricht, but I do not recall that consensus was reached, yet alone
>> declared.  (Unfortunately, I currently don't have the time to go through
>> the mailing list archives to verify my recollection; Sorry.)
>>
>>
>>
>> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org>  wrote:
>>
>>  #12: bit-exact vs. bit-compatible?
>>>
>>> Changes (by gmaxwell@=C5=A0):
>>>
>>>  * status:  new =3D>  closed
>>>  * resolution:  =3D>  worksforme
>>>
>>>
>>> Comment:
>>>
>>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>>
>>> "On the topic of bit exact. Consensus was bit exactness is not required=
."
>>>
>>> I believe this issue is already closed.
>>>
>>> --
>>>
>>> ------------------------------------+----------------------------------=
---
>>> --
>>> Reporter:  hoene@=C5=A0                 |        Owner:
>>>     Type:  enhancement             |       Status:  closed
>>> Priority:  minor                   |    Milestone:
>>> Component:  requirements            |      Version:
>>> Severity:  Active WG Document      |   Resolution:  worksforme
>>> Keywords:                          |
>>>
>>> ------------------------------------+----------------------------------=
---
>>> --
>>>
>>> Ticket URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:=
1
>>> >
>>> codec<http://tools.ietf.org/codec/>
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org
>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>  _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>

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Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

One more concern that I have related to bit exactness is codec regression t=
esting. One can produce a non-bit-exact encoder that produces reasonable re=
sults for a reference decoder, but if paired with a modified, non-bit-exact=
 decoder will produce significant audio artifacts. Since neither decoder or=
 encoder are bit exact we will need to have a test procedure that will vali=
date that both encoder or decoder will not break any standard compliant and=
 non-bit-exact encoders or decoders. Not really sure how this can be done.<=
div>
<br></div><div>I might be wrong, but I think MPEG decoders are bit exact an=
d encoders are not.<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Tue, Jan 25, 2011 at 12:03 AM, Jean-M=
arc Valin <span dir=3D"ltr">&lt;<a href=3D"mailto:jmvalin@jmvalin.ca">jmval=
in@jmvalin.ca</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi Stephan,<br>
<br>
I understand your concern and I&#39;d be interested if you have alternative=
 ways of handling the licensing to avoid any issue. It&#39;s not like this =
is a unique situation. As far as I know, most (all?) MPEG codecs have simil=
ar non-bit exact definitions. I have also heard that they also require some=
 IPR licensing...<br>

<br>
In general the issue of bit-exactness has been discussed and so far I don&#=
39;t recall many arguing in favor of a bit-exact definition. Most of the co=
ncerns that have been expressed are solved by considering that non-bitexact=
 does not mean you cannot be bit-exact with the reference encoder. It merel=
y means that you don&#39;t *have* to. So regardless of how conformance is d=
efined exactly, one always has the option of being bit-exact with the refer=
ence implementation, which obviously guarantees compliance.<br>

<br>
As for language mentioning compliance, I believe it belongs more to the gui=
delines (it&#39;s not a requirement of the codec itself), which includes th=
e following text:<br>
<br>
 =C2=A0 4. =C2=A0To reduce the risk of bias towards certain CPU/DSP archite=
ctures,<br>
 =C2=A0 =C2=A0 =C2=A0 ideally the decoder specification should not require =
&quot;bit-exact&quot;<br>
 =C2=A0 =C2=A0 =C2=A0 conformance with the reference implementation. =C2=A0=
The output of a<br>
 =C2=A0 =C2=A0 =C2=A0 decoder implementation should only be &quot;close eno=
ugh&quot; to the<br>
 =C2=A0 =C2=A0 =C2=A0 output of the reference decoder. =C2=A0A comparison t=
ool should be<br>
 =C2=A0 =C2=A0 =C2=A0 provided along with the codec to verify objectively t=
hat the<br>
 =C2=A0 =C2=A0 =C2=A0 output of a decoder is likely to be perceptually<br>
 =C2=A0 =C2=A0 =C2=A0 indistinguishable from that of the reference decoder.=
 =C2=A0However,<br>
 =C2=A0 =C2=A0 =C2=A0 an implementation may still wish to produce an output=
 that is<br>
 =C2=A0 =C2=A0 =C2=A0 bit-exact with the reference implementation to simpli=
fy the<br>
 =C2=A0 =C2=A0 =C2=A0 testing procedure.<br>
<br>
<br>
<br>
Cheers,<br><font color=3D"#888888">
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Jean-Marc</font><div><div></div><div class=3D"h=
5"><br>
<br>
On 11-01-24 11:02 PM, Stephan Wenger wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi all:<br>
<br>
Let me speak once more against this decision (if such a decision were<br>
really made; see the p.s.).<br>
<br>
There are currently three IPR disclosures against the codec draft and/or<br=
>
its predecessors. =C2=A0The Xiph disclosure is at this point a placeholder<=
br>
(Xiph folks: it&#39;s time to fix that!). =C2=A0However, the two other disc=
losures<br>
on file provide a patent grant only for necessary patent claims and only<br=
>
when the standard is practiced in full compliance. =C2=A0These terms (in<br=
>
various formulations) are quite common.<br>
<br>
In order to ensure one has a license (or can rely on a non-assert<br>
covenant), one has to ensure one meets the conditions set by the<br>
rightholder. =C2=A0On stuff such as reciprocity clauses this is simple. =C2=
=A0On<br>
compliance, it&#39;s not always easy.<br>
<br>
The traditional compliance test for a media codec is a stimulus-response<br=
>
test: you feed test vectors into the codec, and you get results. =C2=A0If t=
he<br>
results match, you are in compliance, if not, you are not. =C2=A0Simple.<br=
>
<br>
Without bit exactness, the compliance criteria have to be defined<br>
differently. =C2=A0We can do so, and, indeed, I recall that this has been<b=
r>
mentioned as one plan forward. =C2=A0However, I have seen zero activity in =
this<br>
direction, and I have also not seen any language that mentions this in the<=
br>
requirements draft. =C2=A0I think that the subject of compliance tests, at<=
br>
least in its most basic outline, needs to be documented in the<br>
requirements draft. =C2=A0The details can be taken care of elsewhere and la=
ter,<br>
but not too much later. =C2=A0It should be clear that a codec candidate (if=
<br>
there were more than one) needs to have compliance criteria defined before<=
br>
that codec candidate can become an RFC. =C2=A0Without that, the key goal of=
 the<br>
WG, a reasonably freely practicable codec, is just not achievable in the<br=
>
current legal environment (which includes, in this case, the IPR<br>
disclosures on file).<br>
<br>
Of course, it would be sooooo much simpler if we would mandate a bit exact<=
br>
decoder... =C2=A0Is it really that restricting to require that?<br>
<br>
Stephan<br>
<br>
P.s.: for the IETF procedures newcomers: humms taken at meetings need to<br=
>
be confirmed on a mailing list, and consensus needs to be declared by the<b=
r>
chairs. =C2=A0On this subject, I do recall mailing list discussions after<b=
r>
Maastricht, but I do not recall that consensus was reached, yet alone<br>
declared. =C2=A0(Unfortunately, I currently don&#39;t have the time to go t=
hrough<br>
the mailing list archives to verify my recollection; Sorry.)<br>
<br>
<br>
<br>
On 1.24.2011 16:45 , &quot;codec issue tracker&quot;&lt;<a href=3D"mailto:t=
rac@tools.ietf.org" target=3D"_blank">trac@tools.ietf.org</a>&gt; =C2=A0wro=
te:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
#12: bit-exact vs. bit-compatible?<br>
<br>
Changes (by gmaxwell@=C5=A0):<br>
<br>
 =C2=A0* status: =C2=A0new =3D&gt; =C2=A0closed<br>
 =C2=A0* resolution: =C2=A0=3D&gt; =C2=A0worksforme<br>
<br>
<br>
Comment:<br>
<br>
<a href=3D"http://www.ietf.org/proceedings/78/minutes/codec.txt" target=3D"=
_blank">http://www.ietf.org/proceedings/78/minutes/codec.txt</a><br>
<br>
&quot;On the topic of bit exact. Consensus was bit exactness is not require=
d.&quot;<br>
<br>
I believe this issue is already closed.<br>
<br>
--<br>
------------------------------------+-------------------------------------<=
br>
--<br>
Reporter: =C2=A0hoene@=C5=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0Owner:<br>
 =C2=A0 =C2=A0 Type: =C2=A0enhancement =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 | =C2=A0 =C2=A0 =C2=A0 Status: =C2=A0closed<br>
Priority: =C2=A0minor =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 | =C2=A0 =C2=A0Milestone:<br>
Component: =C2=A0requirements =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =
=C2=A0 =C2=A0 =C2=A0Version:<br>
Severity: =C2=A0Active WG Document =C2=A0 =C2=A0 =C2=A0| =C2=A0 Resolution:=
 =C2=A0worksforme<br>
Keywords: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
------------------------------------+-------------------------------------<=
br>
--<br>
<br>
Ticket URL:&lt;<a href=3D"http://trac.tools.ietf.org/wg/codec/trac/ticket/1=
2#comment:1" target=3D"_blank">http://trac.tools.ietf.org/wg/codec/trac/tic=
ket/12#comment:1</a>&gt;<br>
codec&lt;<a href=3D"http://tools.ietf.org/codec/" target=3D"_blank">http://=
tools.ietf.org/codec/</a>&gt;<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</blockquote>
<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
<br>
<br>
</blockquote>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br></div>

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From stephen.botzko@gmail.com  Tue Jan 25 07:39:05 2011
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References: <C963872D.26A05%stewe@stewe.org> <4D3E598A.3010200@jmvalin.ca> <AANLkTinSyfj-r67BhghSbOURTXLvToqNLBtATo+aJYo6@mail.gmail.com>
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From: Stephen Botzko <stephen.botzko@gmail.com>
To: Roman Shpount <roman@telurix.com>
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Cc: codec@ietf.org
Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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It seems to me that the reference encoder and decoder will be bit exact for
a given floating point format, correct?  So one could specify IEEE 754-2008
when bit exactness is needed in the reference (for instance regression
testing), but not require it for compliance.

Folks who are modifying the reference encoder or decoder algorithms are "on
their own" as far as audio quality is concerned.  Though I agree with
Stephan that we need to address compliance (when exactly does a ported or
otherwise modified encoder/decoder become non-compliant?).

Stephen Botzko

On Tue, Jan 25, 2011 at 9:34 AM, Roman Shpount <roman@telurix.com> wrote:

> One more concern that I have related to bit exactness is codec regression
> testing. One can produce a non-bit-exact encoder that produces reasonable
> results for a reference decoder, but if paired with a modified,
> non-bit-exact decoder will produce significant audio artifacts. Since
> neither decoder or encoder are bit exact we will need to have a test
> procedure that will validate that both encoder or decoder will not break =
any
> standard compliant and non-bit-exact encoders or decoders. Not really sur=
e
> how this can be done.
>
> I might be wrong, but I think MPEG decoders are bit exact and encoders ar=
e
> not.
> _____________
> Roman Shpount
>
>
>
> On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin <jmvalin@jmvalin.ca>wro=
te:
>
>> Hi Stephan,
>>
>> I understand your concern and I'd be interested if you have alternative
>> ways of handling the licensing to avoid any issue. It's not like this is=
 a
>> unique situation. As far as I know, most (all?) MPEG codecs have similar
>> non-bit exact definitions. I have also heard that they also require some=
 IPR
>> licensing...
>>
>> In general the issue of bit-exactness has been discussed and so far I
>> don't recall many arguing in favor of a bit-exact definition. Most of th=
e
>> concerns that have been expressed are solved by considering that
>> non-bitexact does not mean you cannot be bit-exact with the reference
>> encoder. It merely means that you don't *have* to. So regardless of how
>> conformance is defined exactly, one always has the option of being bit-e=
xact
>> with the reference implementation, which obviously guarantees compliance=
.
>>
>> As for language mentioning compliance, I believe it belongs more to the
>> guidelines (it's not a requirement of the codec itself), which includes =
the
>> following text:
>>
>>   4.  To reduce the risk of bias towards certain CPU/DSP architectures,
>>       ideally the decoder specification should not require "bit-exact"
>>       conformance with the reference implementation.  The output of a
>>       decoder implementation should only be "close enough" to the
>>       output of the reference decoder.  A comparison tool should be
>>       provided along with the codec to verify objectively that the
>>       output of a decoder is likely to be perceptually
>>       indistinguishable from that of the reference decoder.  However,
>>       an implementation may still wish to produce an output that is
>>       bit-exact with the reference implementation to simplify the
>>       testing procedure.
>>
>>
>>
>> Cheers,
>>
>>        Jean-Marc
>>
>>
>> On 11-01-24 11:02 PM, Stephan Wenger wrote:
>>
>>> Hi all:
>>>
>>> Let me speak once more against this decision (if such a decision were
>>> really made; see the p.s.).
>>>
>>> There are currently three IPR disclosures against the codec draft and/o=
r
>>> its predecessors.  The Xiph disclosure is at this point a placeholder
>>> (Xiph folks: it's time to fix that!).  However, the two other disclosur=
es
>>> on file provide a patent grant only for necessary patent claims and onl=
y
>>> when the standard is practiced in full compliance.  These terms (in
>>> various formulations) are quite common.
>>>
>>> In order to ensure one has a license (or can rely on a non-assert
>>> covenant), one has to ensure one meets the conditions set by the
>>> rightholder.  On stuff such as reciprocity clauses this is simple.  On
>>> compliance, it's not always easy.
>>>
>>> The traditional compliance test for a media codec is a stimulus-respons=
e
>>> test: you feed test vectors into the codec, and you get results.  If th=
e
>>> results match, you are in compliance, if not, you are not.  Simple.
>>>
>>> Without bit exactness, the compliance criteria have to be defined
>>> differently.  We can do so, and, indeed, I recall that this has been
>>> mentioned as one plan forward.  However, I have seen zero activity in
>>> this
>>> direction, and I have also not seen any language that mentions this in
>>> the
>>> requirements draft.  I think that the subject of compliance tests, at
>>> least in its most basic outline, needs to be documented in the
>>> requirements draft.  The details can be taken care of elsewhere and
>>> later,
>>> but not too much later.  It should be clear that a codec candidate (if
>>> there were more than one) needs to have compliance criteria defined
>>> before
>>> that codec candidate can become an RFC.  Without that, the key goal of
>>> the
>>> WG, a reasonably freely practicable codec, is just not achievable in th=
e
>>> current legal environment (which includes, in this case, the IPR
>>> disclosures on file).
>>>
>>> Of course, it would be sooooo much simpler if we would mandate a bit
>>> exact
>>> decoder...  Is it really that restricting to require that?
>>>
>>> Stephan
>>>
>>> P.s.: for the IETF procedures newcomers: humms taken at meetings need t=
o
>>> be confirmed on a mailing list, and consensus needs to be declared by t=
he
>>> chairs.  On this subject, I do recall mailing list discussions after
>>> Maastricht, but I do not recall that consensus was reached, yet alone
>>> declared.  (Unfortunately, I currently don't have the time to go throug=
h
>>> the mailing list archives to verify my recollection; Sorry.)
>>>
>>>
>>>
>>> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org>  wrote:
>>>
>>>  #12: bit-exact vs. bit-compatible?
>>>>
>>>> Changes (by gmaxwell@=C5=A0):
>>>>
>>>>  * status:  new =3D>  closed
>>>>  * resolution:  =3D>  worksforme
>>>>
>>>>
>>>> Comment:
>>>>
>>>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>>>
>>>> "On the topic of bit exact. Consensus was bit exactness is not
>>>> required."
>>>>
>>>> I believe this issue is already closed.
>>>>
>>>> --
>>>>
>>>> ------------------------------------+---------------------------------=
----
>>>> --
>>>> Reporter:  hoene@=C5=A0                 |        Owner:
>>>>     Type:  enhancement             |       Status:  closed
>>>> Priority:  minor                   |    Milestone:
>>>> Component:  requirements            |      Version:
>>>> Severity:  Active WG Document      |   Resolution:  worksforme
>>>> Keywords:                          |
>>>>
>>>> ------------------------------------+---------------------------------=
----
>>>> --
>>>>
>>>> Ticket URL:<
>>>> http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>>>> codec<http://tools.ietf.org/codec/>
>>>>
>>>> _______________________________________________
>>>> codec mailing list
>>>> codec@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>
>>>
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org
>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>>
>>>  _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>
>

--0015175cda2ca68edd049aad8ec4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

It seems to me that the reference encoder and decoder will be bit exact for=
 a given floating point format, correct?=C2=A0 So one could specify IEEE 75=
4-2008 when bit exactness is needed in the reference (for instance regressi=
on testing), but not require it for compliance.<br>
<br>Folks who are modifying the reference encoder or decoder algorithms are=
 &quot;on their own&quot; as far as audio quality is concerned.=C2=A0 Thoug=
h I agree with Stephan that we need to address compliance (when exactly doe=
s a ported or otherwise modified encoder/decoder become non-compliant?).<br=
>
<br>Stephen Botzko<br><br><div class=3D"gmail_quote">On Tue, Jan 25, 2011 a=
t 9:34 AM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telu=
rix.com">roman@telurix.com</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(2=
04, 204, 204); padding-left: 1ex;">
One more concern that I have related to bit exactness is codec regression t=
esting. One can produce a non-bit-exact encoder that produces reasonable re=
sults for a reference decoder, but if paired with a modified, non-bit-exact=
 decoder will produce significant audio artifacts. Since neither decoder or=
 encoder are bit exact we will need to have a test procedure that will vali=
date that both encoder or decoder will not break any standard compliant and=
 non-bit-exact encoders or decoders. Not really sure how this can be done.<=
div>

<br></div><div>I might be wrong, but I think MPEG decoders are bit exact an=
d encoders are not.<br clear=3D"all">_____________<br><font color=3D"#88888=
8">Roman Shpount</font><div><div></div><div class=3D"h5"><br>
<br><br><div class=3D"gmail_quote">On Tue, Jan 25, 2011 at 12:03 AM, Jean-M=
arc Valin <span dir=3D"ltr">&lt;<a href=3D"mailto:jmvalin@jmvalin.ca" targe=
t=3D"_blank">jmvalin@jmvalin.ca</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid=
 rgb(204, 204, 204); padding-left: 1ex;">

Hi Stephan,<br>
<br>
I understand your concern and I&#39;d be interested if you have alternative=
 ways of handling the licensing to avoid any issue. It&#39;s not like this =
is a unique situation. As far as I know, most (all?) MPEG codecs have simil=
ar non-bit exact definitions. I have also heard that they also require some=
 IPR licensing...<br>


<br>
In general the issue of bit-exactness has been discussed and so far I don&#=
39;t recall many arguing in favor of a bit-exact definition. Most of the co=
ncerns that have been expressed are solved by considering that non-bitexact=
 does not mean you cannot be bit-exact with the reference encoder. It merel=
y means that you don&#39;t *have* to. So regardless of how conformance is d=
efined exactly, one always has the option of being bit-exact with the refer=
ence implementation, which obviously guarantees compliance.<br>


<br>
As for language mentioning compliance, I believe it belongs more to the gui=
delines (it&#39;s not a requirement of the codec itself), which includes th=
e following text:<br>
<br>
 =C2=A0 4. =C2=A0To reduce the risk of bias towards certain CPU/DSP archite=
ctures,<br>
 =C2=A0 =C2=A0 =C2=A0 ideally the decoder specification should not require =
&quot;bit-exact&quot;<br>
 =C2=A0 =C2=A0 =C2=A0 conformance with the reference implementation. =C2=A0=
The output of a<br>
 =C2=A0 =C2=A0 =C2=A0 decoder implementation should only be &quot;close eno=
ugh&quot; to the<br>
 =C2=A0 =C2=A0 =C2=A0 output of the reference decoder. =C2=A0A comparison t=
ool should be<br>
 =C2=A0 =C2=A0 =C2=A0 provided along with the codec to verify objectively t=
hat the<br>
 =C2=A0 =C2=A0 =C2=A0 output of a decoder is likely to be perceptually<br>
 =C2=A0 =C2=A0 =C2=A0 indistinguishable from that of the reference decoder.=
 =C2=A0However,<br>
 =C2=A0 =C2=A0 =C2=A0 an implementation may still wish to produce an output=
 that is<br>
 =C2=A0 =C2=A0 =C2=A0 bit-exact with the reference implementation to simpli=
fy the<br>
 =C2=A0 =C2=A0 =C2=A0 testing procedure.<br>
<br>
<br>
<br>
Cheers,<br><font color=3D"#888888">
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Jean-Marc</font><div><div></div><div><br>
<br>
On 11-01-24 11:02 PM, Stephan Wenger wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi all:<br>
<br>
Let me speak once more against this decision (if such a decision were<br>
really made; see the p.s.).<br>
<br>
There are currently three IPR disclosures against the codec draft and/or<br=
>
its predecessors. =C2=A0The Xiph disclosure is at this point a placeholder<=
br>
(Xiph folks: it&#39;s time to fix that!). =C2=A0However, the two other disc=
losures<br>
on file provide a patent grant only for necessary patent claims and only<br=
>
when the standard is practiced in full compliance. =C2=A0These terms (in<br=
>
various formulations) are quite common.<br>
<br>
In order to ensure one has a license (or can rely on a non-assert<br>
covenant), one has to ensure one meets the conditions set by the<br>
rightholder. =C2=A0On stuff such as reciprocity clauses this is simple. =C2=
=A0On<br>
compliance, it&#39;s not always easy.<br>
<br>
The traditional compliance test for a media codec is a stimulus-response<br=
>
test: you feed test vectors into the codec, and you get results. =C2=A0If t=
he<br>
results match, you are in compliance, if not, you are not. =C2=A0Simple.<br=
>
<br>
Without bit exactness, the compliance criteria have to be defined<br>
differently. =C2=A0We can do so, and, indeed, I recall that this has been<b=
r>
mentioned as one plan forward. =C2=A0However, I have seen zero activity in =
this<br>
direction, and I have also not seen any language that mentions this in the<=
br>
requirements draft. =C2=A0I think that the subject of compliance tests, at<=
br>
least in its most basic outline, needs to be documented in the<br>
requirements draft. =C2=A0The details can be taken care of elsewhere and la=
ter,<br>
but not too much later. =C2=A0It should be clear that a codec candidate (if=
<br>
there were more than one) needs to have compliance criteria defined before<=
br>
that codec candidate can become an RFC. =C2=A0Without that, the key goal of=
 the<br>
WG, a reasonably freely practicable codec, is just not achievable in the<br=
>
current legal environment (which includes, in this case, the IPR<br>
disclosures on file).<br>
<br>
Of course, it would be sooooo much simpler if we would mandate a bit exact<=
br>
decoder... =C2=A0Is it really that restricting to require that?<br>
<br>
Stephan<br>
<br>
P.s.: for the IETF procedures newcomers: humms taken at meetings need to<br=
>
be confirmed on a mailing list, and consensus needs to be declared by the<b=
r>
chairs. =C2=A0On this subject, I do recall mailing list discussions after<b=
r>
Maastricht, but I do not recall that consensus was reached, yet alone<br>
declared. =C2=A0(Unfortunately, I currently don&#39;t have the time to go t=
hrough<br>
the mailing list archives to verify my recollection; Sorry.)<br>
<br>
<br>
<br>
On 1.24.2011 16:45 , &quot;codec issue tracker&quot;&lt;<a href=3D"mailto:t=
rac@tools.ietf.org" target=3D"_blank">trac@tools.ietf.org</a>&gt; =C2=A0wro=
te:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
#12: bit-exact vs. bit-compatible?<br>
<br>
Changes (by gmaxwell@=C5=A0):<br>
<br>
 =C2=A0* status: =C2=A0new =3D&gt; =C2=A0closed<br>
 =C2=A0* resolution: =C2=A0=3D&gt; =C2=A0worksforme<br>
<br>
<br>
Comment:<br>
<br>
<a href=3D"http://www.ietf.org/proceedings/78/minutes/codec.txt" target=3D"=
_blank">http://www.ietf.org/proceedings/78/minutes/codec.txt</a><br>
<br>
&quot;On the topic of bit exact. Consensus was bit exactness is not require=
d.&quot;<br>
<br>
I believe this issue is already closed.<br>
<br>
--<br>
------------------------------------+-------------------------------------<=
br>
--<br>
Reporter: =C2=A0hoene@=C5=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0Owner:<br>
 =C2=A0 =C2=A0 Type: =C2=A0enhancement =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 | =C2=A0 =C2=A0 =C2=A0 Status: =C2=A0closed<br>
Priority: =C2=A0minor =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 | =C2=A0 =C2=A0Milestone:<br>
Component: =C2=A0requirements =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =
=C2=A0 =C2=A0 =C2=A0Version:<br>
Severity: =C2=A0Active WG Document =C2=A0 =C2=A0 =C2=A0| =C2=A0 Resolution:=
 =C2=A0worksforme<br>
Keywords: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
------------------------------------+-------------------------------------<=
br>
--<br>
<br>
Ticket URL:&lt;<a href=3D"http://trac.tools.ietf.org/wg/codec/trac/ticket/1=
2#comment:1" target=3D"_blank">http://trac.tools.ietf.org/wg/codec/trac/tic=
ket/12#comment:1</a>&gt;<br>
codec&lt;<a href=3D"http://tools.ietf.org/codec/" target=3D"_blank">http://=
tools.ietf.org/codec/</a>&gt;<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</blockquote>
<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
<br>
<br>
</blockquote>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
</div></div></blockquote></div><br></div></div></div>
<br>_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/codec" target=3D"_blank">h=
ttps://www.ietf.org/mailman/listinfo/codec</a><br>
<br></blockquote></div><br>

--0015175cda2ca68edd049aad8ec4--

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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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As far as I know, MPEG decoders are specified, but not bit exact. With a 
bit-exact specification, there simply could not be any floating-point 
implementation of an MP3 or AAC decoder.

As long as the decoder is correctly specified with a tolerance on the 
deviation compared to the reference decoder, I do not see how an encoder 
could work with one decoder and not with another. As an example of (very 
simple) tolerance, we could say that a decoder implementation has to be 
within 80 dB SNR of the reference implementation on some test vectors. In 
practice we can use slightly more perceptually relevant metrics.

	Jean-Marc

On 11-01-25 09:34 AM, Roman Shpount wrote:
> One more concern that I have related to bit exactness is codec regression
> testing. One can produce a non-bit-exact encoder that produces reasonable
> results for a reference decoder, but if paired with a modified,
> non-bit-exact decoder will produce significant audio artifacts. Since
> neither decoder or encoder are bit exact we will need to have a test
> procedure that will validate that both encoder or decoder will not break
> any standard compliant and non-bit-exact encoders or decoders. Not really
> sure how this can be done.
>
> I might be wrong, but I think MPEG decoders are bit exact and encoders are not.
> _____________
> Roman Shpount
>
>
> On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin <jmvalin@jmvalin.ca
> <mailto:jmvalin@jmvalin.ca>> wrote:
>
>     Hi Stephan,
>
>     I understand your concern and I'd be interested if you have alternative
>     ways of handling the licensing to avoid any issue. It's not like this
>     is a unique situation. As far as I know, most (all?) MPEG codecs have
>     similar non-bit exact definitions. I have also heard that they also
>     require some IPR licensing...
>
>     In general the issue of bit-exactness has been discussed and so far I
>     don't recall many arguing in favor of a bit-exact definition. Most of
>     the concerns that have been expressed are solved by considering that
>     non-bitexact does not mean you cannot be bit-exact with the reference
>     encoder. It merely means that you don't *have* to. So regardless of how
>     conformance is defined exactly, one always has the option of being
>     bit-exact with the reference implementation, which obviously guarantees
>     compliance.
>
>     As for language mentioning compliance, I believe it belongs more to the
>     guidelines (it's not a requirement of the codec itself), which includes
>     the following text:
>
>     4. To reduce the risk of bias towards certain CPU/DSP architectures,
>     ideally the decoder specification should not require "bit-exact"
>     conformance with the reference implementation. The output of a
>     decoder implementation should only be "close enough" to the
>     output of the reference decoder. A comparison tool should be
>     provided along with the codec to verify objectively that the
>     output of a decoder is likely to be perceptually
>     indistinguishable from that of the reference decoder. However,
>     an implementation may still wish to produce an output that is
>     bit-exact with the reference implementation to simplify the
>     testing procedure.
>
>
>
>     Cheers,
>
>     Jean-Marc
>
>
>     On 11-01-24 11:02 PM, Stephan Wenger wrote:
>
>         Hi all:
>
>         Let me speak once more against this decision (if such a decision were
>         really made; see the p.s.).
>
>         There are currently three IPR disclosures against the codec draft
>         and/or
>         its predecessors. The Xiph disclosure is at this point a placeholder
>         (Xiph folks: it's time to fix that!). However, the two other
>         disclosures
>         on file provide a patent grant only for necessary patent claims and
>         only
>         when the standard is practiced in full compliance. These terms (in
>         various formulations) are quite common.
>
>         In order to ensure one has a license (or can rely on a non-assert
>         covenant), one has to ensure one meets the conditions set by the
>         rightholder. On stuff such as reciprocity clauses this is simple. On
>         compliance, it's not always easy.
>
>         The traditional compliance test for a media codec is a
>         stimulus-response
>         test: you feed test vectors into the codec, and you get results. If the
>         results match, you are in compliance, if not, you are not. Simple.
>
>         Without bit exactness, the compliance criteria have to be defined
>         differently. We can do so, and, indeed, I recall that this has been
>         mentioned as one plan forward. However, I have seen zero activity
>         in this
>         direction, and I have also not seen any language that mentions this
>         in the
>         requirements draft. I think that the subject of compliance tests, at
>         least in its most basic outline, needs to be documented in the
>         requirements draft. The details can be taken care of elsewhere and
>         later,
>         but not too much later. It should be clear that a codec candidate (if
>         there were more than one) needs to have compliance criteria defined
>         before
>         that codec candidate can become an RFC. Without that, the key goal
>         of the
>         WG, a reasonably freely practicable codec, is just not achievable
>         in the
>         current legal environment (which includes, in this case, the IPR
>         disclosures on file).
>
>         Of course, it would be sooooo much simpler if we would mandate a
>         bit exact
>         decoder... Is it really that restricting to require that?
>
>         Stephan
>
>         P.s.: for the IETF procedures newcomers: humms taken at meetings
>         need to
>         be confirmed on a mailing list, and consensus needs to be declared
>         by the
>         chairs. On this subject, I do recall mailing list discussions after
>         Maastricht, but I do not recall that consensus was reached, yet alone
>         declared. (Unfortunately, I currently don't have the time to go through
>         the mailing list archives to verify my recollection; Sorry.)
>
>
>
>         On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org
>         <mailto:trac@tools.ietf.org>> wrote:
>
>             #12: bit-exact vs. bit-compatible?
>
>             Changes (by gmaxwell@Å ):
>
>             * status: new => closed
>             * resolution: => worksforme
>
>
>             Comment:
>
>             http://www.ietf.org/proceedings/78/minutes/codec.txt
>
>             "On the topic of bit exact. Consensus was bit exactness is not
>             required."
>
>             I believe this issue is already closed.
>
>             --
>             ------------------------------------+-------------------------------------
>             --
>             Reporter: hoene@Å | Owner:
>             Type: enhancement | Status: closed
>             Priority: minor | Milestone:
>             Component: requirements | Version:
>             Severity: Active WG Document | Resolution: worksforme
>             Keywords: |
>             ------------------------------------+-------------------------------------
>             --
>
>             Ticket
>             URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>             codec<http://tools.ietf.org/codec/>
>
>             _______________________________________________
>             codec mailing list
>             codec@ietf.org <mailto:codec@ietf.org>
>             https://www.ietf.org/mailman/listinfo/codec
>
>
>
>         _______________________________________________
>         codec mailing list
>         codec@ietf.org <mailto:codec@ietf.org>
>         https://www.ietf.org/mailman/listinfo/codec
>
>
>     _______________________________________________
>     codec mailing list
>     codec@ietf.org <mailto:codec@ietf.org>
>     https://www.ietf.org/mailman/listinfo/codec
>
>
>
>
> _______________________________________________
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> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From jean-marc.valin@octasic.com  Tue Jan 25 07:54:38 2011
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Date: Tue, 25 Jan 2011 10:57:25 -0500
From: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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Hi Stephen,

The reference implementation includes both floating-point and fixed-point, 
so I'm not sure it's a good idea to say that one is "better" than the 
other. As for IEEE 754-2008, as far as I know it is not a "bit-exact" 
standard either when it comes to some operations (e.g. transcendental 
function) and even for other operations, most compilers I know of are not 
strict IEEE-754 (at least by default) because of issues such as the x86 
extended precision operations.

Regarding MPEG bit-streams, I'm not sure where to get confirmation, but 
this is what Wikipedia has to say about MPEG-1 Layer 3:


"Decoding, on the other hand, is carefully defined in the standard. Most 
decoders are "bitstream compliant", which means that the decompressed 
output – that they produce from a given MP3 file – will be the same, within 
a specified degree of rounding tolerance, as the output specified 
mathematically in the ISO/IEC standard document (ISO/IEC 11172-3). 
Therefore, comparison of decoders is usually based on how computationally 
efficient they are (i.e., how much memory or CPU time they use in the 
decoding process)."


That implies a standard based on "infinite precision", with a degree of 
tolerance for different implementations.

Cheers,

	Jean-Marc

On 11-01-25 10:42 AM, Stephen Botzko wrote:
> It seems to me that the reference encoder and decoder will be bit exact for
> a given floating point format, correct? So one could specify IEEE 754-2008
> when bit exactness is needed in the reference (for instance regression
> testing), but not require it for compliance.
>
> Folks who are modifying the reference encoder or decoder algorithms are "on
> their own" as far as audio quality is concerned. Though I agree with
> Stephan that we need to address compliance (when exactly does a ported or
> otherwise modified encoder/decoder become non-compliant?).
>
> Stephen Botzko
>
> On Tue, Jan 25, 2011 at 9:34 AM, Roman Shpount <roman@telurix.com
> <mailto:roman@telurix.com>> wrote:
>
>     One more concern that I have related to bit exactness is codec
>     regression testing. One can produce a non-bit-exact encoder that
>     produces reasonable results for a reference decoder, but if paired with
>     a modified, non-bit-exact decoder will produce significant audio
>     artifacts. Since neither decoder or encoder are bit exact we will need
>     to have a test procedure that will validate that both encoder or
>     decoder will not break any standard compliant and non-bit-exact
>     encoders or decoders. Not really sure how this can be done.
>
>     I might be wrong, but I think MPEG decoders are bit exact and encoders
>     are not.
>     _____________
>     Roman Shpount
>
>
>
>     On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin <jmvalin@jmvalin.ca
>     <mailto:jmvalin@jmvalin.ca>> wrote:
>
>         Hi Stephan,
>
>         I understand your concern and I'd be interested if you have
>         alternative ways of handling the licensing to avoid any issue. It's
>         not like this is a unique situation. As far as I know, most (all?)
>         MPEG codecs have similar non-bit exact definitions. I have also
>         heard that they also require some IPR licensing...
>
>         In general the issue of bit-exactness has been discussed and so far
>         I don't recall many arguing in favor of a bit-exact definition.
>         Most of the concerns that have been expressed are solved by
>         considering that non-bitexact does not mean you cannot be bit-exact
>         with the reference encoder. It merely means that you don't *have*
>         to. So regardless of how conformance is defined exactly, one always
>         has the option of being bit-exact with the reference
>         implementation, which obviously guarantees compliance.
>
>         As for language mentioning compliance, I believe it belongs more to
>         the guidelines (it's not a requirement of the codec itself), which
>         includes the following text:
>
>         4. To reduce the risk of bias towards certain CPU/DSP architectures,
>         ideally the decoder specification should not require "bit-exact"
>         conformance with the reference implementation. The output of a
>         decoder implementation should only be "close enough" to the
>         output of the reference decoder. A comparison tool should be
>         provided along with the codec to verify objectively that the
>         output of a decoder is likely to be perceptually
>         indistinguishable from that of the reference decoder. However,
>         an implementation may still wish to produce an output that is
>         bit-exact with the reference implementation to simplify the
>         testing procedure.
>
>
>
>         Cheers,
>
>         Jean-Marc
>
>
>         On 11-01-24 11:02 PM, Stephan Wenger wrote:
>
>             Hi all:
>
>             Let me speak once more against this decision (if such a
>             decision were
>             really made; see the p.s.).
>
>             There are currently three IPR disclosures against the codec
>             draft and/or
>             its predecessors. The Xiph disclosure is at this point a
>             placeholder
>             (Xiph folks: it's time to fix that!). However, the two other
>             disclosures
>             on file provide a patent grant only for necessary patent claims
>             and only
>             when the standard is practiced in full compliance. These terms (in
>             various formulations) are quite common.
>
>             In order to ensure one has a license (or can rely on a non-assert
>             covenant), one has to ensure one meets the conditions set by the
>             rightholder. On stuff such as reciprocity clauses this is
>             simple. On
>             compliance, it's not always easy.
>
>             The traditional compliance test for a media codec is a
>             stimulus-response
>             test: you feed test vectors into the codec, and you get
>             results. If the
>             results match, you are in compliance, if not, you are not. Simple.
>
>             Without bit exactness, the compliance criteria have to be defined
>             differently. We can do so, and, indeed, I recall that this has been
>             mentioned as one plan forward. However, I have seen zero
>             activity in this
>             direction, and I have also not seen any language that mentions
>             this in the
>             requirements draft. I think that the subject of compliance
>             tests, at
>             least in its most basic outline, needs to be documented in the
>             requirements draft. The details can be taken care of elsewhere
>             and later,
>             but not too much later. It should be clear that a codec
>             candidate (if
>             there were more than one) needs to have compliance criteria
>             defined before
>             that codec candidate can become an RFC. Without that, the key
>             goal of the
>             WG, a reasonably freely practicable codec, is just not
>             achievable in the
>             current legal environment (which includes, in this case, the IPR
>             disclosures on file).
>
>             Of course, it would be sooooo much simpler if we would mandate
>             a bit exact
>             decoder... Is it really that restricting to require that?
>
>             Stephan
>
>             P.s.: for the IETF procedures newcomers: humms taken at
>             meetings need to
>             be confirmed on a mailing list, and consensus needs to be
>             declared by the
>             chairs. On this subject, I do recall mailing list discussions after
>             Maastricht, but I do not recall that consensus was reached, yet
>             alone
>             declared. (Unfortunately, I currently don't have the time to go
>             through
>             the mailing list archives to verify my recollection; Sorry.)
>
>
>
>             On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org
>             <mailto:trac@tools.ietf.org>> wrote:
>
>                 #12: bit-exact vs. bit-compatible?
>
>                 Changes (by gmaxwell@Å ):
>
>                 * status: new => closed
>                 * resolution: => worksforme
>
>
>                 Comment:
>
>                 http://www.ietf.org/proceedings/78/minutes/codec.txt
>
>                 "On the topic of bit exact. Consensus was bit exactness is
>                 not required."
>
>                 I believe this issue is already closed.
>
>                 --
>                 ------------------------------------+-------------------------------------
>                 --
>                 Reporter: hoene@Å | Owner:
>                 Type: enhancement | Status: closed
>                 Priority: minor | Milestone:
>                 Component: requirements | Version:
>                 Severity: Active WG Document | Resolution: worksforme
>                 Keywords: |
>                 ------------------------------------+-------------------------------------
>                 --
>
>                 Ticket
>                 URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>                 codec<http://tools.ietf.org/codec/>
>
>                 _______________________________________________
>                 codec mailing list
>                 codec@ietf.org <mailto:codec@ietf.org>
>                 https://www.ietf.org/mailman/listinfo/codec
>
>
>
>             _______________________________________________
>             codec mailing list
>             codec@ietf.org <mailto:codec@ietf.org>
>             https://www.ietf.org/mailman/listinfo/codec
>
>
>         _______________________________________________
>         codec mailing list
>         codec@ietf.org <mailto:codec@ietf.org>
>         https://www.ietf.org/mailman/listinfo/codec
>
>
>
>     _______________________________________________
>     codec mailing list
>     codec@ietf.org <mailto:codec@ietf.org>
>     https://www.ietf.org/mailman/listinfo/codec
>
>
>
>
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From jean-marc.valin@octasic.com  Tue Jan 25 08:26:36 2011
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Just to give an idea of what I'm talking about in terms of compliance test, 
here's a tool I just wrote that could be used for testing. There may need 
to be a bit of refining and playing with the thresholds, but the main idea 
is there:

http://jmvalin.ca/misc_stuff/compare.m

It works with Octave, and is likely very easy to get to work with Matlab. 
Any comments?

	Jean-Marc

On 11-01-25 10:57 AM, Jean-Marc Valin wrote:
> Hi Stephen,
>
> The reference implementation includes both floating-point and fixed-point,
> so I'm not sure it's a good idea to say that one is "better" than the
> other. As for IEEE 754-2008, as far as I know it is not a "bit-exact"
> standard either when it comes to some operations (e.g. transcendental
> function) and even for other operations, most compilers I know of are not
> strict IEEE-754 (at least by default) because of issues such as the x86
> extended precision operations.
>
> Regarding MPEG bit-streams, I'm not sure where to get confirmation, but
> this is what Wikipedia has to say about MPEG-1 Layer 3:
>
>
> "Decoding, on the other hand, is carefully defined in the standard. Most
> decoders are "bitstream compliant", which means that the decompressed
> output – that they produce from a given MP3 file – will be the same, within
> a specified degree of rounding tolerance, as the output specified
> mathematically in the ISO/IEC standard document (ISO/IEC 11172-3).
> Therefore, comparison of decoders is usually based on how computationally
> efficient they are (i.e., how much memory or CPU time they use in the
> decoding process)."
>
>
> That implies a standard based on "infinite precision", with a degree of
> tolerance for different implementations.
>
> Cheers,
>
> Jean-Marc
>
> On 11-01-25 10:42 AM, Stephen Botzko wrote:
>> It seems to me that the reference encoder and decoder will be bit exact for
>> a given floating point format, correct? So one could specify IEEE 754-2008
>> when bit exactness is needed in the reference (for instance regression
>> testing), but not require it for compliance.
>>
>> Folks who are modifying the reference encoder or decoder algorithms are "on
>> their own" as far as audio quality is concerned. Though I agree with
>> Stephan that we need to address compliance (when exactly does a ported or
>> otherwise modified encoder/decoder become non-compliant?).
>>
>> Stephen Botzko
>>
>> On Tue, Jan 25, 2011 at 9:34 AM, Roman Shpount <roman@telurix.com
>> <mailto:roman@telurix.com>> wrote:
>>
>> One more concern that I have related to bit exactness is codec
>> regression testing. One can produce a non-bit-exact encoder that
>> produces reasonable results for a reference decoder, but if paired with
>> a modified, non-bit-exact decoder will produce significant audio
>> artifacts. Since neither decoder or encoder are bit exact we will need
>> to have a test procedure that will validate that both encoder or
>> decoder will not break any standard compliant and non-bit-exact
>> encoders or decoders. Not really sure how this can be done.
>>
>> I might be wrong, but I think MPEG decoders are bit exact and encoders
>> are not.
>> _____________
>> Roman Shpount
>>
>>
>>
>> On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin <jmvalin@jmvalin.ca
>> <mailto:jmvalin@jmvalin.ca>> wrote:
>>
>> Hi Stephan,
>>
>> I understand your concern and I'd be interested if you have
>> alternative ways of handling the licensing to avoid any issue. It's
>> not like this is a unique situation. As far as I know, most (all?)
>> MPEG codecs have similar non-bit exact definitions. I have also
>> heard that they also require some IPR licensing...
>>
>> In general the issue of bit-exactness has been discussed and so far
>> I don't recall many arguing in favor of a bit-exact definition.
>> Most of the concerns that have been expressed are solved by
>> considering that non-bitexact does not mean you cannot be bit-exact
>> with the reference encoder. It merely means that you don't *have*
>> to. So regardless of how conformance is defined exactly, one always
>> has the option of being bit-exact with the reference
>> implementation, which obviously guarantees compliance.
>>
>> As for language mentioning compliance, I believe it belongs more to
>> the guidelines (it's not a requirement of the codec itself), which
>> includes the following text:
>>
>> 4. To reduce the risk of bias towards certain CPU/DSP architectures,
>> ideally the decoder specification should not require "bit-exact"
>> conformance with the reference implementation. The output of a
>> decoder implementation should only be "close enough" to the
>> output of the reference decoder. A comparison tool should be
>> provided along with the codec to verify objectively that the
>> output of a decoder is likely to be perceptually
>> indistinguishable from that of the reference decoder. However,
>> an implementation may still wish to produce an output that is
>> bit-exact with the reference implementation to simplify the
>> testing procedure.
>>
>>
>>
>> Cheers,
>>
>> Jean-Marc
>>
>>
>> On 11-01-24 11:02 PM, Stephan Wenger wrote:
>>
>> Hi all:
>>
>> Let me speak once more against this decision (if such a
>> decision were
>> really made; see the p.s.).
>>
>> There are currently three IPR disclosures against the codec
>> draft and/or
>> its predecessors. The Xiph disclosure is at this point a
>> placeholder
>> (Xiph folks: it's time to fix that!). However, the two other
>> disclosures
>> on file provide a patent grant only for necessary patent claims
>> and only
>> when the standard is practiced in full compliance. These terms (in
>> various formulations) are quite common.
>>
>> In order to ensure one has a license (or can rely on a non-assert
>> covenant), one has to ensure one meets the conditions set by the
>> rightholder. On stuff such as reciprocity clauses this is
>> simple. On
>> compliance, it's not always easy.
>>
>> The traditional compliance test for a media codec is a
>> stimulus-response
>> test: you feed test vectors into the codec, and you get
>> results. If the
>> results match, you are in compliance, if not, you are not. Simple.
>>
>> Without bit exactness, the compliance criteria have to be defined
>> differently. We can do so, and, indeed, I recall that this has been
>> mentioned as one plan forward. However, I have seen zero
>> activity in this
>> direction, and I have also not seen any language that mentions
>> this in the
>> requirements draft. I think that the subject of compliance
>> tests, at
>> least in its most basic outline, needs to be documented in the
>> requirements draft. The details can be taken care of elsewhere
>> and later,
>> but not too much later. It should be clear that a codec
>> candidate (if
>> there were more than one) needs to have compliance criteria
>> defined before
>> that codec candidate can become an RFC. Without that, the key
>> goal of the
>> WG, a reasonably freely practicable codec, is just not
>> achievable in the
>> current legal environment (which includes, in this case, the IPR
>> disclosures on file).
>>
>> Of course, it would be sooooo much simpler if we would mandate
>> a bit exact
>> decoder... Is it really that restricting to require that?
>>
>> Stephan
>>
>> P.s.: for the IETF procedures newcomers: humms taken at
>> meetings need to
>> be confirmed on a mailing list, and consensus needs to be
>> declared by the
>> chairs. On this subject, I do recall mailing list discussions after
>> Maastricht, but I do not recall that consensus was reached, yet
>> alone
>> declared. (Unfortunately, I currently don't have the time to go
>> through
>> the mailing list archives to verify my recollection; Sorry.)
>>
>>
>>
>> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org
>> <mailto:trac@tools.ietf.org>> wrote:
>>
>> #12: bit-exact vs. bit-compatible?
>>
>> Changes (by gmaxwell@Å ):
>>
>> * status: new => closed
>> * resolution: => worksforme
>>
>>
>> Comment:
>>
>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>
>> "On the topic of bit exact. Consensus was bit exactness is
>> not required."
>>
>> I believe this issue is already closed.
>>
>> --
>> ------------------------------------+-------------------------------------
>> --
>> Reporter: hoene@Å | Owner:
>> Type: enhancement | Status: closed
>> Priority: minor | Milestone:
>> Component: requirements | Version:
>> Severity: Active WG Document | Resolution: worksforme
>> Keywords: |
>> ------------------------------------+-------------------------------------
>> --
>>
>> Ticket
>> URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>> codec<http://tools.ietf.org/codec/>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>
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> https://www.ietf.org/mailman/listinfo/codec


From koen.vos@skype.net  Tue Jan 25 12:34:32 2011
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From: Koen Vos <koen.vos@skype.net>
To: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Cc: codec@ietf.org, Stephen Botzko <stephen.botzko@gmail.com>
Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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A similar tool is in the code in silk\test\signalCompare.c.  It's based on =
LPC, and measures the SNR in a partially-whitened domain.  For music the cr=
itical-band approach of Jean-Marc's tool may work better, not sure.

One question is about what implementation should be the reference for compl=
iance testing: floating-point or fixed-point?

A fixed-point reference allows for bit-exact implementations, which is nice=
.  And fixed-point reference implementations are the norm in speech coding.
However, CELT's floating-point implementation has a larger dynamic range th=
an the fixed-point version, and is thus arguable better.  A fixed-point ref=
erence could not test this extra dynamic range.

Any suggestions?
koen.


----- Original Message -----
From: "Jean-Marc Valin" <jean-marc.valin@octasic.com>
To: "Stephen Botzko" <stephen.botzko@gmail.com>
Cc: codec@ietf.org
Sent: Tuesday, January 25, 2011 8:29:25 AM
Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatibl=
e?

Just to give an idea of what I'm talking about in terms of compliance test,=
=20
here's a tool I just wrote that could be used for testing. There may need=
=20
to be a bit of refining and playing with the thresholds, but the main idea=
=20
is there:

http://jmvalin.ca/misc_stuff/compare.m

It works with Octave, and is likely very easy to get to work with Matlab.=
=20
Any comments?

=09Jean-Marc

On 11-01-25 10:57 AM, Jean-Marc Valin wrote:
> Hi Stephen,
>
> The reference implementation includes both floating-point and fixed-point=
,
> so I'm not sure it's a good idea to say that one is "better" than the
> other. As for IEEE 754-2008, as far as I know it is not a "bit-exact"
> standard either when it comes to some operations (e.g. transcendental
> function) and even for other operations, most compilers I know of are not
> strict IEEE-754 (at least by default) because of issues such as the x86
> extended precision operations.
>
> Regarding MPEG bit-streams, I'm not sure where to get confirmation, but
> this is what Wikipedia has to say about MPEG-1 Layer 3:
>
>
> "Decoding, on the other hand, is carefully defined in the standard. Most
> decoders are "bitstream compliant", which means that the decompressed
> output =E2=80=93 that they produce from a given MP3 file =E2=80=93 will b=
e the same, within
> a specified degree of rounding tolerance, as the output specified
> mathematically in the ISO/IEC standard document (ISO/IEC 11172-3).
> Therefore, comparison of decoders is usually based on how computationally
> efficient they are (i.e., how much memory or CPU time they use in the
> decoding process)."
>
>
> That implies a standard based on "infinite precision", with a degree of
> tolerance for different implementations.
>
> Cheers,
>
> Jean-Marc
>
> On 11-01-25 10:42 AM, Stephen Botzko wrote:
>> It seems to me that the reference encoder and decoder will be bit exact =
for
>> a given floating point format, correct? So one could specify IEEE 754-20=
08
>> when bit exactness is needed in the reference (for instance regression
>> testing), but not require it for compliance.
>>
>> Folks who are modifying the reference encoder or decoder algorithms are =
"on
>> their own" as far as audio quality is concerned. Though I agree with
>> Stephan that we need to address compliance (when exactly does a ported o=
r
>> otherwise modified encoder/decoder become non-compliant?).
>>
>> Stephen Botzko
>>
>> On Tue, Jan 25, 2011 at 9:34 AM, Roman Shpount <roman@telurix.com
>> <mailto:roman@telurix.com>> wrote:
>>
>> One more concern that I have related to bit exactness is codec
>> regression testing. One can produce a non-bit-exact encoder that
>> produces reasonable results for a reference decoder, but if paired with
>> a modified, non-bit-exact decoder will produce significant audio
>> artifacts. Since neither decoder or encoder are bit exact we will need
>> to have a test procedure that will validate that both encoder or
>> decoder will not break any standard compliant and non-bit-exact
>> encoders or decoders. Not really sure how this can be done.
>>
>> I might be wrong, but I think MPEG decoders are bit exact and encoders
>> are not.
>> _____________
>> Roman Shpount
>>
>>
>>
>> On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin <jmvalin@jmvalin.ca
>> <mailto:jmvalin@jmvalin.ca>> wrote:
>>
>> Hi Stephan,
>>
>> I understand your concern and I'd be interested if you have
>> alternative ways of handling the licensing to avoid any issue. It's
>> not like this is a unique situation. As far as I know, most (all?)
>> MPEG codecs have similar non-bit exact definitions. I have also
>> heard that they also require some IPR licensing...
>>
>> In general the issue of bit-exactness has been discussed and so far
>> I don't recall many arguing in favor of a bit-exact definition.
>> Most of the concerns that have been expressed are solved by
>> considering that non-bitexact does not mean you cannot be bit-exact
>> with the reference encoder. It merely means that you don't *have*
>> to. So regardless of how conformance is defined exactly, one always
>> has the option of being bit-exact with the reference
>> implementation, which obviously guarantees compliance.
>>
>> As for language mentioning compliance, I believe it belongs more to
>> the guidelines (it's not a requirement of the codec itself), which
>> includes the following text:
>>
>> 4. To reduce the risk of bias towards certain CPU/DSP architectures,
>> ideally the decoder specification should not require "bit-exact"
>> conformance with the reference implementation. The output of a
>> decoder implementation should only be "close enough" to the
>> output of the reference decoder. A comparison tool should be
>> provided along with the codec to verify objectively that the
>> output of a decoder is likely to be perceptually
>> indistinguishable from that of the reference decoder. However,
>> an implementation may still wish to produce an output that is
>> bit-exact with the reference implementation to simplify the
>> testing procedure.
>>
>>
>>
>> Cheers,
>>
>> Jean-Marc
>>
>>
>> On 11-01-24 11:02 PM, Stephan Wenger wrote:
>>
>> Hi all:
>>
>> Let me speak once more against this decision (if such a
>> decision were
>> really made; see the p.s.).
>>
>> There are currently three IPR disclosures against the codec
>> draft and/or
>> its predecessors. The Xiph disclosure is at this point a
>> placeholder
>> (Xiph folks: it's time to fix that!). However, the two other
>> disclosures
>> on file provide a patent grant only for necessary patent claims
>> and only
>> when the standard is practiced in full compliance. These terms (in
>> various formulations) are quite common.
>>
>> In order to ensure one has a license (or can rely on a non-assert
>> covenant), one has to ensure one meets the conditions set by the
>> rightholder. On stuff such as reciprocity clauses this is
>> simple. On
>> compliance, it's not always easy.
>>
>> The traditional compliance test for a media codec is a
>> stimulus-response
>> test: you feed test vectors into the codec, and you get
>> results. If the
>> results match, you are in compliance, if not, you are not. Simple.
>>
>> Without bit exactness, the compliance criteria have to be defined
>> differently. We can do so, and, indeed, I recall that this has been
>> mentioned as one plan forward. However, I have seen zero
>> activity in this
>> direction, and I have also not seen any language that mentions
>> this in the
>> requirements draft. I think that the subject of compliance
>> tests, at
>> least in its most basic outline, needs to be documented in the
>> requirements draft. The details can be taken care of elsewhere
>> and later,
>> but not too much later. It should be clear that a codec
>> candidate (if
>> there were more than one) needs to have compliance criteria
>> defined before
>> that codec candidate can become an RFC. Without that, the key
>> goal of the
>> WG, a reasonably freely practicable codec, is just not
>> achievable in the
>> current legal environment (which includes, in this case, the IPR
>> disclosures on file).
>>
>> Of course, it would be sooooo much simpler if we would mandate
>> a bit exact
>> decoder... Is it really that restricting to require that?
>>
>> Stephan
>>
>> P.s.: for the IETF procedures newcomers: humms taken at
>> meetings need to
>> be confirmed on a mailing list, and consensus needs to be
>> declared by the
>> chairs. On this subject, I do recall mailing list discussions after
>> Maastricht, but I do not recall that consensus was reached, yet
>> alone
>> declared. (Unfortunately, I currently don't have the time to go
>> through
>> the mailing list archives to verify my recollection; Sorry.)
>>
>>
>>
>> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org
>> <mailto:trac@tools.ietf.org>> wrote:
>>
>> #12: bit-exact vs. bit-compatible?
>>
>> Changes (by gmaxwell@=C3=85 ):
>>
>> * status: new =3D> closed
>> * resolution: =3D> worksforme
>>
>>
>> Comment:
>>
>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>
>> "On the topic of bit exact. Consensus was bit exactness is
>> not required."
>>
>> I believe this issue is already closed.
>>
>> --
>> ------------------------------------+-----------------------------------=
--
>> --
>> Reporter: hoene@=C3=85 | Owner:
>> Type: enhancement | Status: closed
>> Priority: minor | Milestone:
>> Component: requirements | Version:
>> Severity: Active WG Document | Resolution: worksforme
>> Keywords: |
>> ------------------------------------+-----------------------------------=
--
>> --
>>
>> Ticket
>> URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>> codec<http://tools.ietf.org/codec/>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org <mailto:codec@ietf.org>
>> https://www.ietf.org/mailman/listinfo/codec
>>
>>
>>
>>
>> _______________________________________________
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>> https://www.ietf.org/mailman/listinfo/codec
>
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From jean-marc.valin@octasic.com  Tue Jan 25 12:55:19 2011
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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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My preference actually goes with using both fixed-point and floating point. 
That way, implementers of floating-point decoder can compare with the float 
version and implementers of the fixed-point decoder can compare with the 
fixed-point version.

This is not unheard of either, if you look at the reference implementation 
of the 3GPP(2?) EVRC codec, the distribution includes two sets of output 
vectors. One is for a fixed-point implementation that uses 31-bit precision 
in the multiplications and the other one is for 32-bit precision. You just 
choose whatever is more efficient for your DSP and use the appropriate 
testvectors for testing.

	Jean-Marc

On 11-01-25 03:37 PM, Koen Vos wrote:
> A similar tool is in the code in silk\test\signalCompare.c.  It's based on LPC, and measures the SNR in a partially-whitened domain.  For music the critical-band approach of Jean-Marc's tool may work better, not sure.
>
> One question is about what implementation should be the reference for compliance testing: floating-point or fixed-point?
>
> A fixed-point reference allows for bit-exact implementations, which is nice.  And fixed-point reference implementations are the norm in speech coding.
> However, CELT's floating-point implementation has a larger dynamic range than the fixed-point version, and is thus arguable better.  A fixed-point reference could not test this extra dynamic range.
>
> Any suggestions?
> koen.
>
>
> ----- Original Message -----
> From: "Jean-Marc Valin"<jean-marc.valin@octasic.com>
> To: "Stephen Botzko"<stephen.botzko@gmail.com>
> Cc: codec@ietf.org
> Sent: Tuesday, January 25, 2011 8:29:25 AM
> Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
>
> Just to give an idea of what I'm talking about in terms of compliance test,
> here's a tool I just wrote that could be used for testing. There may need
> to be a bit of refining and playing with the thresholds, but the main idea
> is there:
>
> http://jmvalin.ca/misc_stuff/compare.m
>
> It works with Octave, and is likely very easy to get to work with Matlab.
> Any comments?
>
> 	Jean-Marc
>
> On 11-01-25 10:57 AM, Jean-Marc Valin wrote:
>> Hi Stephen,
>>
>> The reference implementation includes both floating-point and fixed-point,
>> so I'm not sure it's a good idea to say that one is "better" than the
>> other. As for IEEE 754-2008, as far as I know it is not a "bit-exact"
>> standard either when it comes to some operations (e.g. transcendental
>> function) and even for other operations, most compilers I know of are not
>> strict IEEE-754 (at least by default) because of issues such as the x86
>> extended precision operations.
>>
>> Regarding MPEG bit-streams, I'm not sure where to get confirmation, but
>> this is what Wikipedia has to say about MPEG-1 Layer 3:
>>
>>
>> "Decoding, on the other hand, is carefully defined in the standard. Most
>> decoders are "bitstream compliant", which means that the decompressed
>> output â€“ that they produce from a given MP3 file â€“ will be the same, within
>> a specified degree of rounding tolerance, as the output specified
>> mathematically in the ISO/IEC standard document (ISO/IEC 11172-3).
>> Therefore, comparison of decoders is usually based on how computationally
>> efficient they are (i.e., how much memory or CPU time they use in the
>> decoding process)."
>>
>>
>> That implies a standard based on "infinite precision", with a degree of
>> tolerance for different implementations.
>>
>> Cheers,
>>
>> Jean-Marc
>>
>> On 11-01-25 10:42 AM, Stephen Botzko wrote:
>>> It seems to me that the reference encoder and decoder will be bit exact for
>>> a given floating point format, correct? So one could specify IEEE 754-2008
>>> when bit exactness is needed in the reference (for instance regression
>>> testing), but not require it for compliance.
>>>
>>> Folks who are modifying the reference encoder or decoder algorithms are "on
>>> their own" as far as audio quality is concerned. Though I agree with
>>> Stephan that we need to address compliance (when exactly does a ported or
>>> otherwise modified encoder/decoder become non-compliant?).
>>>
>>> Stephen Botzko
>>>
>>> On Tue, Jan 25, 2011 at 9:34 AM, Roman Shpount<roman@telurix.com
>>> <mailto:roman@telurix.com>>  wrote:
>>>
>>> One more concern that I have related to bit exactness is codec
>>> regression testing. One can produce a non-bit-exact encoder that
>>> produces reasonable results for a reference decoder, but if paired with
>>> a modified, non-bit-exact decoder will produce significant audio
>>> artifacts. Since neither decoder or encoder are bit exact we will need
>>> to have a test procedure that will validate that both encoder or
>>> decoder will not break any standard compliant and non-bit-exact
>>> encoders or decoders. Not really sure how this can be done.
>>>
>>> I might be wrong, but I think MPEG decoders are bit exact and encoders
>>> are not.
>>> _____________
>>> Roman Shpount
>>>
>>>
>>>
>>> On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin<jmvalin@jmvalin.ca
>>> <mailto:jmvalin@jmvalin.ca>>  wrote:
>>>
>>> Hi Stephan,
>>>
>>> I understand your concern and I'd be interested if you have
>>> alternative ways of handling the licensing to avoid any issue. It's
>>> not like this is a unique situation. As far as I know, most (all?)
>>> MPEG codecs have similar non-bit exact definitions. I have also
>>> heard that they also require some IPR licensing...
>>>
>>> In general the issue of bit-exactness has been discussed and so far
>>> I don't recall many arguing in favor of a bit-exact definition.
>>> Most of the concerns that have been expressed are solved by
>>> considering that non-bitexact does not mean you cannot be bit-exact
>>> with the reference encoder. It merely means that you don't *have*
>>> to. So regardless of how conformance is defined exactly, one always
>>> has the option of being bit-exact with the reference
>>> implementation, which obviously guarantees compliance.
>>>
>>> As for language mentioning compliance, I believe it belongs more to
>>> the guidelines (it's not a requirement of the codec itself), which
>>> includes the following text:
>>>
>>> 4. To reduce the risk of bias towards certain CPU/DSP architectures,
>>> ideally the decoder specification should not require "bit-exact"
>>> conformance with the reference implementation. The output of a
>>> decoder implementation should only be "close enough" to the
>>> output of the reference decoder. A comparison tool should be
>>> provided along with the codec to verify objectively that the
>>> output of a decoder is likely to be perceptually
>>> indistinguishable from that of the reference decoder. However,
>>> an implementation may still wish to produce an output that is
>>> bit-exact with the reference implementation to simplify the
>>> testing procedure.
>>>
>>>
>>>
>>> Cheers,
>>>
>>> Jean-Marc
>>>
>>>
>>> On 11-01-24 11:02 PM, Stephan Wenger wrote:
>>>
>>> Hi all:
>>>
>>> Let me speak once more against this decision (if such a
>>> decision were
>>> really made; see the p.s.).
>>>
>>> There are currently three IPR disclosures against the codec
>>> draft and/or
>>> its predecessors. The Xiph disclosure is at this point a
>>> placeholder
>>> (Xiph folks: it's time to fix that!). However, the two other
>>> disclosures
>>> on file provide a patent grant only for necessary patent claims
>>> and only
>>> when the standard is practiced in full compliance. These terms (in
>>> various formulations) are quite common.
>>>
>>> In order to ensure one has a license (or can rely on a non-assert
>>> covenant), one has to ensure one meets the conditions set by the
>>> rightholder. On stuff such as reciprocity clauses this is
>>> simple. On
>>> compliance, it's not always easy.
>>>
>>> The traditional compliance test for a media codec is a
>>> stimulus-response
>>> test: you feed test vectors into the codec, and you get
>>> results. If the
>>> results match, you are in compliance, if not, you are not. Simple.
>>>
>>> Without bit exactness, the compliance criteria have to be defined
>>> differently. We can do so, and, indeed, I recall that this has been
>>> mentioned as one plan forward. However, I have seen zero
>>> activity in this
>>> direction, and I have also not seen any language that mentions
>>> this in the
>>> requirements draft. I think that the subject of compliance
>>> tests, at
>>> least in its most basic outline, needs to be documented in the
>>> requirements draft. The details can be taken care of elsewhere
>>> and later,
>>> but not too much later. It should be clear that a codec
>>> candidate (if
>>> there were more than one) needs to have compliance criteria
>>> defined before
>>> that codec candidate can become an RFC. Without that, the key
>>> goal of the
>>> WG, a reasonably freely practicable codec, is just not
>>> achievable in the
>>> current legal environment (which includes, in this case, the IPR
>>> disclosures on file).
>>>
>>> Of course, it would be sooooo much simpler if we would mandate
>>> a bit exact
>>> decoder... Is it really that restricting to require that?
>>>
>>> Stephan
>>>
>>> P.s.: for the IETF procedures newcomers: humms taken at
>>> meetings need to
>>> be confirmed on a mailing list, and consensus needs to be
>>> declared by the
>>> chairs. On this subject, I do recall mailing list discussions after
>>> Maastricht, but I do not recall that consensus was reached, yet
>>> alone
>>> declared. (Unfortunately, I currently don't have the time to go
>>> through
>>> the mailing list archives to verify my recollection; Sorry.)
>>>
>>>
>>>
>>> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org
>>> <mailto:trac@tools.ietf.org>>  wrote:
>>>
>>> #12: bit-exact vs. bit-compatible?
>>>
>>> Changes (by gmaxwell@Ã… ):
>>>
>>> * status: new =>  closed
>>> * resolution: =>  worksforme
>>>
>>>
>>> Comment:
>>>
>>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>>
>>> "On the topic of bit exact. Consensus was bit exactness is
>>> not required."
>>>
>>> I believe this issue is already closed.
>>>
>>> --
>>> ------------------------------------+-------------------------------------
>>> --
>>> Reporter: hoene@Ã… | Owner:
>>> Type: enhancement | Status: closed
>>> Priority: minor | Milestone:
>>> Component: requirements | Version:
>>> Severity: Active WG Document | Resolution: worksforme
>>> Keywords: |
>>> ------------------------------------+-------------------------------------
>>> --
>>>
>>> Ticket
>>> URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>>> codec<http://tools.ietf.org/codec/>
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org<mailto:codec@ietf.org>
>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>>
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org<mailto:codec@ietf.org>
>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org<mailto:codec@ietf.org>
>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>>
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org<mailto:codec@ietf.org>
>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org
>>> https://www.ietf.org/mailman/listinfo/codec
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>
> _______________________________________________
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> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From fluffy@cisco.com  Tue Jan 25 13:01:58 2011
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Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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I followed up with Christian on a separate thread and I hope we will =
sort out whatever is needed to fix this document.=20

I just want to say the chairs certainly want this document to reflect =
rough consensus of the WG. Please, help us get there - review this =
document and provide suggested text for any technical changes that you =
think need to be made to the draft.=20

Thanks, Cullen <CODEC co-chair>


On Jan 23, 2011, at 1:19 AM, Christian Hoene wrote:

>> The authors believe that the requirements document is complete.=20
>=20
> [Christian Hoene] Which authors? So far this document has only two =
selected
> editors.
>=20
>> The chairs
>> would like to now issue a 3 week working group last call for
> draft-ietf-codec-
>> requirements-02. At the end of 3 weeks, if no comments have been
>> received, we will pass the document to the IESG for approval.
>=20
> [Christian Hoene] Despite hundreds of emails, what were exchanged on =
this
> mailing list on issues regarding requirements, the editors have been
> reluctant to update the requirements document according to the =
consensus
> process on this mailing list.=20
>=20
> If I compare draft-ietf-codec-requirement-02 with
> draft-valin-codec-requirements-02, I only see editorial changes.
> =
https://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-codec-requirements-02&dif=
ftyp
> e=3D--html&submit=3DGo!&url2=3Ddraft-valin-codec-requirements-02
>=20
> Sorry guys, if I see this I become very cynical on the standardization
> process at the IETF.   The editors have not done their job and the =
chairs do
> not care. At the end, it all looks like rubberstamping with a little =
bit of
> theater.
>=20
> Christian
>=20
>=20
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From gmaxwell@juniper.net  Tue Jan 25 15:49:49 2011
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Date: Tue, 25 Jan 2011 15:45:34 -0800
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I wanted to send a quick update about the recent technical progress
from the CELT development team since the 0.10 software release and
our last public update about a month ago.
(http://people.xiph.org/~xiphmont/demo/celt/demo.html)

Each point is referenced to commits in our public GIT repository.
(http://git.xiph.org/?p=3Dcelt.git;a=3Dshortlog)


=3D=3DAnti-collapse=3D=3D
(87efe1df, 21af73eb, 5da938b2d, 01fa3389, etc.)

In CELT, transient frames use multiple smaller MDCT transforms rather
than a single large transform in order to provide better temporal
control of quantization noise. These sub-frames are grouped and
coded together.

A previously outstanding issue for transient frames with the larger
(480, 960) frame sizes was that the codec was able to starve some quiet
sub-frames, particularly at low bitrates. This starvation resulted in
short collapse=97 falling to zero energy=97 for a whole subframe
per-band, and often for that subframe in all bands. This was not
prevented by our previously existing sparseness prevention mechanisms
which were primarily targeted at whole band starvation.

The effect is most strikingly demonstrated with spectograms on
castanet samples:
  https://people.xiph.org/~tterribe/notes/celt/spec-anti-collapse-off.png
   -vs-
  https://people.xiph.org/~tterribe/notes/celt/spec-anti-collapse-on.png
(It's not as audible as the images suggest, due to forwards/backwards
temporal masking).

The allocation decision logic is correct in so far as the bits were
best allocated to the louder sub-frames, but the short collapse
produces artifacts of its own.

In order to avoid this issue we implemented decoder-side logic which
tracks where collapse is possible and refills it using noise scaled
using the band energy from prior frames. This resolves the issue but
there is still ongoing fine tuning of the specifics.

=3D=3DPrevent 'busting' at low rates=3D=3D
(76469c64)

One of the core designs of the CELT bitstream is the lack of a fixed
sized/fixed position structure. The encoder and decoder dynamically
and continuously adapt to the remaining space in a frame without
signaling according to behavior patterns hardcoded into the
bitstream specification.

Code/Bitstream simplifications last year reduced the ability of the
codec to deactivate some signaling elements as the available space
approached zero.  This is reflected in the 'non-simple' block diagram
by signaling elements with a minimum rate greater than zero bits:
https://people.xiph.org/~greg/celt/full_block_diagram.png
(now outdated by these changes)

When running out of bits the encoder would simply stop coding and the
decoder would be left to pick up the pieces. It turns out that sanely
handling these cases while maintaining synchronization between the
encoder and decoder was at least of comparable complexity to not
permitting this situation in the bitstream at all.

With these changes all the signaling elements can reach zero bits
per frame.

As a side effect some nonsensical bit-streams are no longer possible.
For example, it was previous possible for a stupid encoder to spend all
of the remaining bits in a frame signaling that a band should get more
bits, but then not have any bits left to actually give to it.

It's no longer possible to code sequences like this, which should make
implementation QA somewhat simpler and the elimination of redundancy
(unreasonable signaling combinations) has resulted in some small quality
improvements.

The encoder can now happily encode down to one byte per frame. The
ability to completely eliminate signaling parameters also provided
some quality improvement at ridiculously low rates, and we can give
(barely) intelligible speech with a couple bytes per-frame.

Intelligible speech at zero bytes per frame remains elusive.

=3D=3DPre/Postfilter tap-sets=3D=3D
(dfa847a2, 8d367029, 2ce5c63d, d121260f)

One of the recommendations from Raymond Chen and team was that
the bitstream should allow the selection of multiple filter-tap
configurations for the pitch filters in order to better adapt to
signals with different harmonic shapes.

Spot testing early in the implementation of the pre/post filter didn't
show much improvement from additional tapsets, but now that the PF
implementation is basically complete we reevaluated this.

Using an automated noise to masking ratio test we scanned the
complete range of reasonable 3-tap filters, and many 5-tap filters,
for hundreds of sound clips. Some of the results are visible here:
https://people.xiph.org/~greg/celt/tests/20110116/sqam-taps-256/results.htm=
l

While the differences were generally small, there were classes of
samples which worked better with configurations on one end of the
spectrum or another.  In many cases the difference between a single
'general' value and a specialized one were easily large enough to
justify some increased signaling overhead.

We ultimately adopted three configurations, a 5-tap filter at
{0.12954,0.21701,0.30690,0.21701,0.12954} with a ~13kHz rolloff which
is now our default, our prior 3-tap filter {0.26795,0.46410,0.26795},
and a much flatter 3-tap filter {0.1,0.8,0.1}. The default takes one
bit to signal, and the other two options require two bits (per frame).

We implemented some very basic automatic selection logic in the
encoder, but it isn't yet adequate. Fortunately the selection logic
doesn't have bitstream impact and can be refined for years to come.

These testing results show the three options and the automatic selection
compared to no filtering for a variety of samples and bitrates:
https://people.xiph.org/~greg/celt/tests/20110118/tap-diff/results.html

=3D=3DNumerous small improvements=3D=3D

* The encoder inter/intraframe decision now checks to make sure that
it doesn't prefer a choice which encodes the energy less precisely due
to running out of bits.  (2293e461)

* The encoder could get stuck in an infinite loop while coding an
extremely loud or quiet difference from the predicted energy. =20
(68b8d72e)

* Range coder simplifications were made which reduced the number
of calls to the entropy coder, which is good for performance, and
increases the amount of code shared with Silk.
(845dfa19, 1aaa50d1, 30df6cf3)

* Corrected an encoder bug which caused a mismatch with the decoder
for quiet signals, some other encoder signal processing improvements,
and changes to the energy clamping behavior mean that our floating
point implementation can provide consistent performance for signals
attenuated by more than 138dB without harming compatibility with
fixed point implementations.  (8f02c482, 495114b7)

* The reference implementation now provides a simple comfort noise
generator for discontinuous operation.  (eafd8a7f)

* Miscellaneous stereo quality improvements.
(235c64b9, a387ebfc, 051e044d, fa7215fdb, 79d76a2e3, 620e716b)

* Corrections to the design of the Time-frequency tradeoff filters
for various corner cases.  (dc6d69e64, 2b13401f, ecefde3d)

...Plus many other small improvements, as documented in our version
control system.

We've also been maintaining a short term working todo list on the
Xiph.Org wiki: https://wiki.xiph.org/CELT_TODO

As usual, our realtime technical development discussion takes place in
the IRC channel, #celt on irc.freenode.net. Our development collaboration
is almost exclusively public and we welcome more participation.



From mary.ietf.barnes@gmail.com  Tue Jan 25 16:08:07 2011
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Date: Tue, 25 Jan 2011 18:11:03 -0600
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: "codec@ietf.org" <codec@ietf.org>, Stephen Botzko <stephen.botzko@gmail.com>
Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
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John,

Sorry for the confusion - yes the one document I noted was the guidelines.
(There was no link to the doc in the Last call announcement, so I went to WG
page and was looking at the other documents as well).

However, the same comment does apply to the requirements - i.e., it should
be informational. There is a perenniel debate as to whether RFC 2119
language applies to informational documents - the RFC 2119 abstract
references only standards track documents and notes that the language (i.e.,
the words in CAPS) should be used sparingly as required for interoperability
or to limit behavior that has the potential to cause problems.  I interpret
that to mean that it does not apply to requirements.

I agree with you that clearly enumerating requirements is a good idea and
should be standard practice.

Thanks,
Mary.

On Tue, Jan 25, 2011 at 4:02 AM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> Mary,
>
> I am confused which document we are talking about. The subject indicates
> the requirements document, but your quoted text "suggested process" seems to
> come from the guidelines document.
>
> The situation with the requirements document
> (draft-ietf-codec-requirements-02) is that it is intended as Standards Track
> but contains no RFC 2119 language (as far as I can see, e.g., no "MUST"
> statements, and no reference to RFC 2119).
>
> I think it should indeed be Informational, but this does not prevent us
> having normative statements in which the codec specification that this WG
> produces is the subject, e.g., "The codec MUST ..." This would make it much
> easier to pick out the actual requirements and to measure the resulting
> codec specification against those requirements. Some WGs adopt the practice
> of numbering requirements, for easy reference.
>
> John
>
> > -----Original Message-----
> > From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org]
> > On Behalf Of Mary Barnes
> > Sent: 24 January 2011 23:25
> > To: Jean-Marc Valin
> > Cc: codec@ietf.org; Stephen Botzko
> > Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02
> >
> > The document is currently specified as Standards Track, thus
> > the use of RFC 2119 language is entirely appropriate.
> > However, ISTM that the document should really just be
> > Informational, in particular given the language in the
> > introduction that this document defines a "suggested
> > process", as opposed to a process that is required for codec
> > development.
> >
> >
> > Regards,
> > Mary.
> >
> >
> > On Mon, Jan 24, 2011 at 5:08 PM, Jean-Marc Valin
> > <jean-marc.valin@octasic.com> wrote:
> >
> >
> >       Christian,
> >
> >       I actually responded to the last comments you made a
> > while ago (oct 2010). One issue I pointed out was your use of
> > RFC2119 keywords, which (AFAIK) aren't appropriate for a
> > requirements draft (the requirements aren't a standard). So
> > statements like "Any codec specified by the IETF MUST be well
> > specified", besides stating the obvious, are inappropriate.
> >
> >       There were also comments that just did not belong to
> > this draft, such as the section on collaboration with other
> > WGs. Collaboration is not a characteristic of a codec. So
> > essentially, I merged the uncontroversial suggestion, but
> > that's all I could do. As far as I'm aware, there's nothing
> > in the current draft that goes against the consensus of the
> > WG. If there are, please point to specific issues and to
> > statements made by others (not just you) asking for the change.
> >
> >       Cheers,
> >
> >              Jean-Marc
> >
> >
> >       On 11-01-24 03:10 PM, Christian Hoene wrote:
> >
> >
> >               Christian - perhaps you could post a list of
> > the issues you see that
> >               haven't been addressed?
> >
> >               */[Christian Hoene] No Stephen, these issues
> > have been written down in
> >               previous emails, drafts and issues in the Trac.
> > They can be read by anybody
> >               anytime. Thus, I do not see any benefit of
> > repeating them again if the
> >               editors continue to ignore any input. Indeed,
> > they did not improve the
> >               draft despite sound technical reasons. /*
> >
> >               */Even if somebody is not fully involved in the
> > technical details: It is
> >               very odd that despite many hundreds emails and
> > many discussions since
> >               starting this WG the editors have not updated
> > the draft beside minor
> >               changes such as the boilerplate and typos. /*
> >
> >               */Even if the lack of any update was not
> > intentionally, the editors missed
> >               to do their job because they were too lazy or
> > rather too busy doing other
> >               thinks./*
> >
> >               */I would be sad if all the fruitful
> > discussions here and all the good
> >               contributions of many industry experts should
> > have been in vain. Even if
> >               not all requirements can be met by Opus, a
> > proper requirements document may
> >               be relevant for future solutions or other SDOs./*
> >
> >
> >               */CH/*
> >
> >
> >
> >
> >               _______________________________________________
> >               codec mailing list
> >               codec@ietf.org
> >               https://www.ietf.org/mailman/listinfo/codec
> >
> >
> >
> >       _______________________________________________
> >       codec mailing list
> >       codec@ietf.org
> >       https://www.ietf.org/mailman/listinfo/codec
> >
> >
> >
> >
>

--0023547c895b35a3aa049ab4ab34
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

John,<div><br></div><div>Sorry for the confusion - yes the one document I n=
oted was the guidelines. (There was no link to the doc in the Last call ann=
ouncement, so I went to WG page and was looking at the other documents as w=
ell).</div>
<div><br></div><div>However, the same comment does apply to the requirement=
s - i.e., it should be informational. There is a perenniel debate as to whe=
ther RFC 2119 language applies to informational documents - the RFC 2119 ab=
stract references only standards track documents and notes that the languag=
e (i.e., the words in CAPS) should be used sparingly as required for intero=
perability or to limit behavior that has the potential to cause problems. =
=A0I interpret that to mean that it does not apply to requirements.</div>
<div><br></div><div>I agree with you that clearly enumerating requirements =
is a good idea and should be standard practice.=A0</div><div><br></div><div=
>Thanks,</div><div>Mary.=A0</div><div><br><div class=3D"gmail_quote">On Tue=
, Jan 25, 2011 at 4:02 AM, Elwell, John <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:john.elwell@siemens-enterprise.com">john.elwell@siemens-enterprise.com=
</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">Mary,<br>
<br>
I am confused which document we are talking about. The subject indicates th=
e requirements document, but your quoted text &quot;suggested process&quot;=
 seems to come from the guidelines document.<br>
<br>
The situation with the requirements document (draft-ietf-codec-requirements=
-02) is that it is intended as Standards Track but contains no RFC 2119 lan=
guage (as far as I can see, e.g., no &quot;MUST&quot; statements, and no re=
ference to RFC 2119).<br>

<br>
I think it should indeed be Informational, but this does not prevent us hav=
ing normative statements in which the codec specification that this WG prod=
uces is the subject, e.g., &quot;The codec MUST ...&quot; This would make i=
t much easier to pick out the actual requirements and to measure the result=
ing codec specification against those requirements. Some WGs adopt the prac=
tice of numbering requirements, for easy reference.<br>

<font color=3D"#888888"><br>
John<br>
</font><div><div></div><div class=3D"h5"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:codec-bounces@ietf.org">codec-bounces@ietf.org=
</a> [mailto:<a href=3D"mailto:codec-bounces@ietf.org">codec-bounces@ietf.o=
rg</a>]<br>
&gt; On Behalf Of Mary Barnes<br>
&gt; Sent: 24 January 2011 23:25<br>
&gt; To: Jean-Marc Valin<br>
&gt; Cc: <a href=3D"mailto:codec@ietf.org">codec@ietf.org</a>; Stephen Botz=
ko<br>
&gt; Subject: Re: [codec] 3 week WGLC on draft-ietf-codec-requirements-02<b=
r>
&gt;<br>
&gt; The document is currently specified as Standards Track, thus<br>
&gt; the use of RFC 2119 language is entirely appropriate.<br>
&gt; However, ISTM that the document should really just be<br>
&gt; Informational, in particular given the language in the<br>
&gt; introduction that this document defines a &quot;suggested<br>
&gt; process&quot;, as opposed to a process that is required for codec<br>
&gt; development.<br>
&gt;<br>
&gt;<br>
&gt; Regards,<br>
&gt; Mary.<br>
&gt;<br>
&gt;<br>
&gt; On Mon, Jan 24, 2011 at 5:08 PM, Jean-Marc Valin<br>
&gt; &lt;<a href=3D"mailto:jean-marc.valin@octasic.com">jean-marc.valin@oct=
asic.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 Christian,<br>
&gt;<br>
&gt; =A0 =A0 =A0 I actually responded to the last comments you made a<br>
&gt; while ago (oct 2010). One issue I pointed out was your use of<br>
&gt; RFC2119 keywords, which (AFAIK) aren&#39;t appropriate for a<br>
&gt; requirements draft (the requirements aren&#39;t a standard). So<br>
&gt; statements like &quot;Any codec specified by the IETF MUST be well<br>
&gt; specified&quot;, besides stating the obvious, are inappropriate.<br>
&gt;<br>
&gt; =A0 =A0 =A0 There were also comments that just did not belong to<br>
&gt; this draft, such as the section on collaboration with other<br>
&gt; WGs. Collaboration is not a characteristic of a codec. So<br>
&gt; essentially, I merged the uncontroversial suggestion, but<br>
&gt; that&#39;s all I could do. As far as I&#39;m aware, there&#39;s nothin=
g<br>
&gt; in the current draft that goes against the consensus of the<br>
&gt; WG. If there are, please point to specific issues and to<br>
&gt; statements made by others (not just you) asking for the change.<br>
&gt;<br>
&gt; =A0 =A0 =A0 Cheers,<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0Jean-Marc<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 On 11-01-24 03:10 PM, Christian Hoene wrote:<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 Christian - perhaps you could post a list =
of<br>
&gt; the issues you see that<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 haven&#39;t been addressed?<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 */[Christian Hoene] No Stephen, these issu=
es<br>
&gt; have been written down in<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 previous emails, drafts and issues in the =
Trac.<br>
&gt; They can be read by anybody<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 anytime. Thus, I do not see any benefit of=
<br>
&gt; repeating them again if the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 editors continue to ignore any input. Inde=
ed,<br>
&gt; they did not improve the<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 draft despite sound technical reasons. /*<=
br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 */Even if somebody is not fully involved i=
n the<br>
&gt; technical details: It is<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 very odd that despite many hundreds emails=
 and<br>
&gt; many discussions since<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 starting this WG the editors have not upda=
ted<br>
&gt; the draft beside minor<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 changes such as the boilerplate and typos.=
 /*<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 */Even if the lack of any update was not<b=
r>
&gt; intentionally, the editors missed<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 to do their job because they were too lazy=
 or<br>
&gt; rather too busy doing other<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 thinks./*<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 */I would be sad if all the fruitful<br>
&gt; discussions here and all the good<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 contributions of many industry experts sho=
uld<br>
&gt; have been in vain. Even if<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 not all requirements can be met by Opus, a=
<br>
&gt; proper requirements document may<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 be relevant for future solutions or other =
SDOs./*<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 */CH/*<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 __________________________________________=
_____<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 codec mailing list<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"mailto:codec@ietf.org">codec@ie=
tf.org</a><br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"https://www.ietf.org/mailman/li=
stinfo/codec" target=3D"_blank">https://www.ietf.org/mailman/listinfo/codec=
</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 _______________________________________________<br>
&gt; =A0 =A0 =A0 codec mailing list<br>
&gt; =A0 =A0 =A0 <a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><br>
&gt; =A0 =A0 =A0 <a href=3D"https://www.ietf.org/mailman/listinfo/codec" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/codec</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; </div></div></blockquote></div><br></div>

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From rchen@broadcom.com  Tue Jan 25 16:51:04 2011
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Date: Tue, 25 Jan 2011 16:53:38 -0800
Thread-Topic: [codec] requirements #8 (new): Sample rates?
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Subject: Re: [codec] requirements #8 (new): Sample rates?
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I agree that supporting sampling rates of 48, 16, and 8 kHz all makes sense=
, but I also think it would be highly desirable to support
another common sampling rate used by music CDs: 44.1 kHz.

For music streaming applications, a large percentage of the music sources w=
ill have this 44.1 kHz sampling rate.  I know it is
possible to up-sample them to 48 kHz first and then pass them through the 4=
8 kHz version of Opus.  However, doing such up-
sampling requires additional processing power and additional latency, and i=
f the sampling rate conversion is not done with a
high-quality method (which usually requires higher complexity and higher de=
lay), audio quality degradation may be introduced
in the process.  For these reasons, many people prefer not to do the 44.1 k=
Hz to 48 kHz sampling rate conversion and prefer to
encode the 44.1 kHz music directly instead.  Therefore, it is highly desira=
ble for Opus to support this 44.1 kHz sampling rate.

I understand that it may be inconvenient for the SILK mode or the SILK+CELT=
 hybrid mode to support 44.1 kHz.  However, for the
CELT-only mode, my understanding is that the current CELT C code already su=
pports the 44.1 kHz sampling rate.  Since the CELT-
only mode is also the most suitable mode to encode music, it would then mak=
e perfect sense for the CELT-only mode to also
support this 44.1 kHz in addition to 48 kHz.

Furthermore, currently the CELT mode at 48 kHz supports frame sizes of 2.5,=
 5., 10, and 20 ms to be consistent with the frame
sizes of the SILK mode and SILK+CELT hybrid mode, and this results in frame=
 sizes and FFT window sizes that are not powers of 2,
which reduces the implementation efficiency.  If the CELT mode supports the=
 44.1 kHz sampling rate, then since it is no longer
possible to support exactly 2.5, 5, 10, and 20 ms at this sampling rate any=
way, I think we might as well choose the FFT window
sizes to be powers of 2 to allow more efficient implementation.  Again, suc=
h power of 2 FFT sizes at 44.1 kHz is already supported
in the current C code for CELT, so no new development is needed.

In summary, I propose that we allow the CELT-only mode of the Opus codec to=
 officially support the 44.1 kHz sampling rate using
power of 2 FFT window sizes to avoid sampling rate conversion and to allow =
the most efficient implementation.

Raymond

From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of R=
oman Shpount
Sent: Monday, January 24, 2011 5:08 PM
To: codec@ietf.org
Cc: jean-marc.valin@usherbrooke.ca
Subject: Re: [codec] requirements #8 (new): Sample rates?

I actually do believe that the requirement is already addressed by section =
4.1 of the document. I was responding to a comment that only full band is r=
equired, which is clearly not what we agreed upon and not what's in the doc=
ument.
_____________
Roman Shpount

On Mon, Jan 24, 2011 at 7:40 PM, codec issue tracker <trac@tools.ietf.org<m=
ailto:trac@tools.ietf.org>> wrote:
#8: Sample rates?


Comment(by gmaxwell@...):
 On 11-01-24 07:14 PM, Roman Shpount wrote:
 > I would like to see 8 and 16 KHz as required rates for the codec to
 insure
 > interoperability with existing narrowband and wideband codecs. In other
 > words we should be able to negotiate 8 or 16 Khz sample rate if audio
 will
 > be transcoded for PSTN or wideband codec such as G.722
 Jean-Marc wrote:
 > I believe such requirement for narrowband/wideband is already present,
 but
 > I don't mind making it even more explicit is necessary.


 I believe that we since are close enough to finalizing the draft we should
 try to include proposed language with our issues.  Otherwise there may be
 no clear path forward.

 Some of the requirements specify that compatibility with
 wideband/narrowband is important, but for the avoidance of doubt, I'll
 suggest:

 At the end of 4.1.  Operating space:

 "Because interoperation with existing wideband and narrowband facilities
 is essential at least one method of interoperation must be provided
 regardless of the codec's operating mode, sample rate, or bitrate."

 Of course, I would be perfectly happy to leave this out entirely (as I
 believe the application section 2.1 already implies the requirement) or
 use some other language.

--
------------------------------------+--------------------------------------=
-
 Reporter:  hoene@...                 |       Owner:  jean-marc.valin@...
    Type:  enhancement             |      Status:  new
 Priority:  minor                   |   Milestone:
Component:  requirements            |     Version:
 Severity:  Active WG Document      |    Keywords:
------------------------------------+--------------------------------------=
-
Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:4>
codec <http://tools.ietf.org/codec/>

_______________________________________________
codec mailing list
codec@ietf.org<mailto:codec@ietf.org>
https://www.ietf.org/mailman/listinfo/codec


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<div class=3DWordSection1>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>I agree that supporting sampling rates of 48, 16, and 8 kHz =
all
makes sense, but I also think it would be highly desirable to support <o:p>=
</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>another common sampling rate used by music CDs: 44.1 kHz.&nb=
sp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>For music streaming applications, a large percentage of the =
music
sources will have this 44.1 kHz sampling rate.&nbsp; I know it is <o:p></o:=
p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>possible to up-sample them to 48 kHz first and then pass the=
m through
the 48 kHz version of Opus.&nbsp; However, doing such up-<o:p></o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>sampling requires additional processing power and additional
latency, and if the sampling rate conversion is not done with a <o:p></o:p>=
</span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>high-quality method (which usually requires higher complexit=
y
and higher delay), audio quality degradation may be introduced <o:p></o:p><=
/span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>in the process.&nbsp; For these reasons, many people prefer =
not
to do the 44.1 kHz to 48 kHz sampling rate conversion and prefer to <o:p></=
o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>encode the 44.1 kHz music directly instead.&nbsp; Therefore,=
 it
is highly desirable for Opus to support this 44.1 kHz sampling rate.<o:p></=
o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>I understand that it may be inconvenient for the SILK mode o=
r the
SILK+CELT hybrid mode to support 44.1 kHz.&nbsp; However, for the <o:p></o:=
p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>CELT-only mode, my understanding is that the current CELT C =
code
already supports the 44.1 kHz sampling rate.&nbsp; Since the CELT-<o:p></o:=
p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>only mode is also the most suitable mode to encode music, it
would then make perfect sense for the CELT-only mode to also <o:p></o:p></s=
pan></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>support this 44.1 kHz in addition to 48 kHz.&nbsp; <o:p></o:=
p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Furthermore, currently the CELT mode at 48 kHz supports fram=
e
sizes of 2.5, 5., 10, and 20 ms to be consistent with the frame <o:p></o:p>=
</span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>sizes of the SILK mode and SILK+CELT hybrid mode, and this
results in frame sizes and FFT window sizes that are not powers of 2, <o:p>=
</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>which reduces the implementation efficiency.&nbsp; If the CE=
LT
mode supports the 44.1 kHz sampling rate, then since it is no longer <o:p><=
/o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>possible to support exactly 2.5, 5, 10, and 20 ms at this
sampling rate anyway, I think we might as well choose the FFT window <o:p><=
/o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>sizes to be powers of 2 to allow more efficient
implementation.&nbsp; Again, such power of 2 FFT sizes at 44.1 kHz is alrea=
dy supported
<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>in the current C code for CELT, so no new development is nee=
ded.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>In summary, I propose that we allow the CELT-only mode of th=
e Opus
codec to officially support the 44.1 kHz sampling rate using <o:p></o:p></s=
pan></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>power of 2 FFT window sizes to avoid sampling rate conversio=
n
and to allow the most efficient implementation.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Raymond<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] <b>On Behalf Of </b>=
Roman
Shpount<br>
<b>Sent:</b> Monday, January 24, 2011 5:08 PM<br>
<b>To:</b> codec@ietf.org<br>
<b>Cc:</b> jean-marc.valin@usherbrooke.ca<br>
<b>Subject:</b> Re: [codec] requirements #8 (new): Sample rates?<o:p></o:p>=
</span></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>I actually do believe t=
hat the
requirement is already addressed by section 4.1 of the document. I was
responding to a comment that only full band is required, which is clearly n=
ot
what we agreed upon and not what's in the document.<br clear=3Dall>
_____________<br>
Roman Shpount<br>
<br>
<o:p></o:p></p>

<div>

<p class=3DMsoNormal>On Mon, Jan 24, 2011 at 7:40 PM, codec issue tracker &=
lt;<a
href=3D"mailto:trac@tools.ietf.org">trac@tools.ietf.org</a>&gt; wrote:<o:p>=
</o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>#8: Sample rates?<br>
<br>
<br>
Comment(by gmaxwell@&#8230;):<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>&nbsp;On 11-01-24 07:14=
 PM,
Roman Shpount wrote:<br>
&nbsp;&gt; I would like to see 8 and 16 KHz as required rates for the codec=
 to<br>
&nbsp;insure<br>
&nbsp;&gt; interoperability with existing narrowband and wideband codecs. I=
n
other<br>
&nbsp;&gt; words we should be able to negotiate 8 or 16 Khz sample rate if
audio<br>
&nbsp;will<br>
&nbsp;&gt; be transcoded for PSTN or wideband codec such as G.722<o:p></o:p=
></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>&nbsp;Jean-Marc wrote:<=
br>
&nbsp;&gt; I believe such requirement for narrowband/wideband is already
present,<br>
&nbsp;but<br>
&nbsp;&gt; I don't mind making it even more explicit is necessary.<br>
<br>
<br>
<o:p></o:p></p>

</div>

<p class=3DMsoNormal>&nbsp;I believe that we since are close enough to fina=
lizing
the draft we should<br>
&nbsp;try to include proposed language with our issues. &nbsp;Otherwise the=
re
may be<br>
&nbsp;no clear path forward.<br>
<br>
&nbsp;Some of the requirements specify that compatibility with<br>
&nbsp;wideband/narrowband is important, but for the avoidance of doubt, I'l=
l<br>
&nbsp;suggest:<br>
<br>
&nbsp;At the end of 4.1. &nbsp;Operating space:<br>
<br>
&nbsp;&quot;Because interoperation with existing wideband and narrowband
facilities<br>
&nbsp;is essential at least one method of interoperation must be provided<b=
r>
&nbsp;regardless of the codec's operating mode, sample rate, or bitrate.&qu=
ot;<br>
<br>
&nbsp;Of course, I would be perfectly happy to leave this out entirely (as =
I<br>
&nbsp;believe the application section 2.1 already implies the requirement) =
or<br>
&nbsp;use some other language.<br>
<br>
--<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>-----------------------=
-------------+---------------------------------------<br>
&nbsp;Reporter: &nbsp;hoene@&#8230; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p;
&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; Owner: &nbsp;jean-marc.valin@&#8230;<b=
r>
&nbsp; &nbsp; Type: &nbsp;enhancement &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp;
| &nbsp; &nbsp; &nbsp;Status: &nbsp;new<br>
&nbsp;Priority: &nbsp;minor &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; | &nbsp; Milestone:<br>
Component: &nbsp;requirements &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| &n=
bsp;
&nbsp; Version:<br>
&nbsp;Severity: &nbsp;Active WG Document &nbsp; &nbsp; &nbsp;| &nbsp;
&nbsp;Keywords:<br>
------------------------------------+--------------------------------------=
-<o:p></o:p></p>

</div>

<p class=3DMsoNormal>Ticket URL: &lt;<a
href=3D"http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:4"
target=3D"_blank">http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment=
:4</a>&gt;<o:p></o:p></p>

<div>

<p class=3DMsoNormal>codec &lt;<a href=3D"http://tools.ietf.org/codec/"
target=3D"_blank">http://tools.ietf.org/codec/</a>&gt;<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org">codec@ietf.org</a><o:p></o:p></p>

</div>

<div>

<div>

<p class=3DMsoNormal><a href=3D"https://www.ietf.org/mailman/listinfo/codec=
"
target=3D"_blank">https://www.ietf.org/mailman/listinfo/codec</a><o:p></o:p=
></p>

</div>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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From stewe@stewe.org  Tue Jan 25 19:57:16 2011
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Date: Tue, 25 Jan 2011 19:59:58 -0800
From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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Subject: Re: [codec] requirements #12 (closed): bit-exact vs. bit-compatible?
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So I see I'm outgunned on the bit-exact issue.  Fine.

But please add language in the requirement doc to the extent that there
MUST be a conformance spec for each candidate codec that allows to
automatically determine the conformance of a codec implementation.  Absent
this spec, the IPR statements received are meaningless.

My preference would be to have that conformance spec in the same document
as the codec, so to make citations in IPR statements easy and to avoid a
charter-update. =20

Stephan


On 1.25.2011 18:24 , "Raymond (Juin-Hwey) Chen" <rchen@broadcom.com> wrote:

>My opinion is that we should not require bit-exactness.  There are many
>examples of previous codec
>standards that did not require bit-exactness, including both
>royalty-bearing codecs (such as MPEG)
>and royalty-free codecs (such as our BroadVoice/BV16/BV32), so I don't
>see any real reason why we
>can't follow that same route.
>
>I can see at least two important advantages for not requiring
>bit-exactness:
>(1) it allows future innovations to improve the performance of the codec
>without breaking the bit-
>stream compatibility, and
>(2) it allows the codec implementers the greatest flexibility in
>implementing the codecs in ways
>that are most efficient for the particular hardware or processor
>architecture they have.
>Having a bit-exact standard will kill both.
>
>The main drawback of a non-bit-exact standard is that the verification of
>the conformance of codec
>implementations is not as straightforward as with a bit-exact standard.
>However, I don't think
>that is a show stopper or even a difficult problem.  There are many
>possible ways to do software-
>based objective measurements to verify that a candidate codec
>implementation is likely to sound
>essentially indistinguishable from the output of the reference codec.  We
>had previously measured
>the SNRs and PESQ scores of a large number of test files using the
>reference codec and candidate
>codec and examined the statistical distribution of the differences in the
>scores.  Jean-Marc just
>posted a nice little Matlab code that built upon psychoacoustic modeling,
>NMR, averages, outliers
>checking, etc.  Koen also mentioned another tool.  Even if these tools
>and methodologies are not
>perfect yet, with some additional tuning of their algorithms and
>parameters, it is really not that
>difficult to come up with objective, software-based verification methods
>that will give you a high
>level of confidence whether a candidate codec will sound at least as good
>as the reference codec
>(or perhaps better if improvements are made).
>
>I like Jean-Marc's idea of having both a floating-point reference codec
>and a fixed-point
>reference codec. For people who don't want to run the risk of degrading
>audio quality through
>their own fixed-point algorithm conversion, they can just implement the
>fixed-point reference
>codec in a bit-exact manner.  For people who want to implement the codec
>in floating-point (e.g.
>on computers), the easiest way is just to use the floating-point
>reference code as is.  In both
>cases, the conformance verification will be quite easy.
>
>Raymond
>
>-----Original Message-----
>From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of
>Jean-Marc Valin
>Sent: Tuesday, January 25, 2011 12:58 PM
>To: Koen Vos
>Cc: codec@ietf.org; Stephen Botzko
>Subject: Re: [codec] requirements #12 (closed): bit-exact vs.
>bit-compatible?
>
>My preference actually goes with using both fixed-point and floating
>point.
>That way, implementers of floating-point decoder can compare with the
>float
>version and implementers of the fixed-point decoder can compare with the
>fixed-point version.
>
>This is not unheard of either, if you look at the reference implementation
>of the 3GPP(2?) EVRC codec, the distribution includes two sets of output
>vectors. One is for a fixed-point implementation that uses 31-bit
>precision
>in the multiplications and the other one is for 32-bit precision. You just
>choose whatever is more efficient for your DSP and use the appropriate
>testvectors for testing.
>
>        Jean-Marc
>
>On 11-01-25 03:37 PM, Koen Vos wrote:
>> A similar tool is in the code in silk\test\signalCompare.c.  It's based
>>on LPC, and measures the SNR in a partially-whitened domain.  For music
>>the critical-band approach of Jean-Marc's tool may work better, not sure.
>>
>> One question is about what implementation should be the reference for
>>compliance testing: floating-point or fixed-point?
>>
>> A fixed-point reference allows for bit-exact implementations, which is
>>nice.  And fixed-point reference implementations are the norm in speech
>>coding.
>> However, CELT's floating-point implementation has a larger dynamic
>>range than the fixed-point version, and is thus arguable better.  A
>>fixed-point reference could not test this extra dynamic range.
>>
>> Any suggestions?
>> koen.
>>
>>
>> ----- Original Message -----
>> From: "Jean-Marc Valin"<jean-marc.valin@octasic.com>
>> To: "Stephen Botzko"<stephen.botzko@gmail.com>
>> Cc: codec@ietf.org
>> Sent: Tuesday, January 25, 2011 8:29:25 AM
>> Subject: Re: [codec] requirements #12 (closed): bit-exact vs.
>>bit-compatible?
>>
>> Just to give an idea of what I'm talking about in terms of compliance
>>test,
>> here's a tool I just wrote that could be used for testing. There may
>>need
>> to be a bit of refining and playing with the thresholds, but the main
>>idea
>> is there:
>>
>> http://jmvalin.ca/misc_stuff/compare.m
>>
>> It works with Octave, and is likely very easy to get to work with
>>Matlab.
>> Any comments?
>>
>>       Jean-Marc
>>
>> On 11-01-25 10:57 AM, Jean-Marc Valin wrote:
>>> Hi Stephen,
>>>
>>> The reference implementation includes both floating-point and
>>>fixed-point,
>>> so I'm not sure it's a good idea to say that one is "better" than the
>>> other. As for IEEE 754-2008, as far as I know it is not a "bit-exact"
>>> standard either when it comes to some operations (e.g. transcendental
>>> function) and even for other operations, most compilers I know of are
>>>not
>>> strict IEEE-754 (at least by default) because of issues such as the x86
>>> extended precision operations.
>>>
>>> Regarding MPEG bit-streams, I'm not sure where to get confirmation, but
>>> this is what Wikipedia has to say about MPEG-1 Layer 3:
>>>
>>>
>>> "Decoding, on the other hand, is carefully defined in the standard.
>>>Most
>>> decoders are "bitstream compliant", which means that the decompressed
>>> output =C3=A2=E2=82=AC=E2=80=9C that they produce from a given MP3 file =C3=A2=E2=82=AC=E2=80=9C will b=
e the
>>>same, within
>>> a specified degree of rounding tolerance, as the output specified
>>> mathematically in the ISO/IEC standard document (ISO/IEC 11172-3).
>>> Therefore, comparison of decoders is usually based on how
>>>computationally
>>> efficient they are (i.e., how much memory or CPU time they use in the
>>> decoding process)."
>>>
>>>
>>> That implies a standard based on "infinite precision", with a degree of
>>> tolerance for different implementations.
>>>
>>> Cheers,
>>>
>>> Jean-Marc
>>>
>>> On 11-01-25 10:42 AM, Stephen Botzko wrote:
>>>> It seems to me that the reference encoder and decoder will be bit
>>>>exact for
>>>> a given floating point format, correct? So one could specify IEEE
>>>>754-2008
>>>> when bit exactness is needed in the reference (for instance regression
>>>> testing), but not require it for compliance.
>>>>
>>>> Folks who are modifying the reference encoder or decoder algorithms
>>>>are "on
>>>> their own" as far as audio quality is concerned. Though I agree with
>>>> Stephan that we need to address compliance (when exactly does a
>>>>ported or
>>>> otherwise modified encoder/decoder become non-compliant?).
>>>>
>>>> Stephen Botzko
>>>>
>>>> On Tue, Jan 25, 2011 at 9:34 AM, Roman Shpount<roman@telurix.com
>>>> <mailto:roman@telurix.com>>  wrote:
>>>>
>>>> One more concern that I have related to bit exactness is codec
>>>> regression testing. One can produce a non-bit-exact encoder that
>>>> produces reasonable results for a reference decoder, but if paired
>>>>with
>>>> a modified, non-bit-exact decoder will produce significant audio
>>>> artifacts. Since neither decoder or encoder are bit exact we will need
>>>> to have a test procedure that will validate that both encoder or
>>>> decoder will not break any standard compliant and non-bit-exact
>>>> encoders or decoders. Not really sure how this can be done.
>>>>
>>>> I might be wrong, but I think MPEG decoders are bit exact and encoders
>>>> are not.
>>>> _____________
>>>> Roman Shpount
>>>>
>>>>
>>>>
>>>> On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin<jmvalin@jmvalin.ca
>>>> <mailto:jmvalin@jmvalin.ca>>  wrote:
>>>>
>>>> Hi Stephan,
>>>>
>>>> I understand your concern and I'd be interested if you have
>>>> alternative ways of handling the licensing to avoid any issue. It's
>>>> not like this is a unique situation. As far as I know, most (all?)
>>>> MPEG codecs have similar non-bit exact definitions. I have also
>>>> heard that they also require some IPR licensing...
>>>>
>>>> In general the issue of bit-exactness has been discussed and so far
>>>> I don't recall many arguing in favor of a bit-exact definition.
>>>> Most of the concerns that have been expressed are solved by
>>>> considering that non-bitexact does not mean you cannot be bit-exact
>>>> with the reference encoder. It merely means that you don't *have*
>>>> to. So regardless of how conformance is defined exactly, one always
>>>> has the option of being bit-exact with the reference
>>>> implementation, which obviously guarantees compliance.
>>>>
>>>> As for language mentioning compliance, I believe it belongs more to
>>>> the guidelines (it's not a requirement of the codec itself), which
>>>> includes the following text:
>>>>
>>>> 4. To reduce the risk of bias towards certain CPU/DSP architectures,
>>>> ideally the decoder specification should not require "bit-exact"
>>>> conformance with the reference implementation. The output of a
>>>> decoder implementation should only be "close enough" to the
>>>> output of the reference decoder. A comparison tool should be
>>>> provided along with the codec to verify objectively that the
>>>> output of a decoder is likely to be perceptually
>>>> indistinguishable from that of the reference decoder. However,
>>>> an implementation may still wish to produce an output that is
>>>> bit-exact with the reference implementation to simplify the
>>>> testing procedure.
>>>>
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Jean-Marc
>>>>
>>>>
>>>> On 11-01-24 11:02 PM, Stephan Wenger wrote:
>>>>
>>>> Hi all:
>>>>
>>>> Let me speak once more against this decision (if such a
>>>> decision were
>>>> really made; see the p.s.).
>>>>
>>>> There are currently three IPR disclosures against the codec
>>>> draft and/or
>>>> its predecessors. The Xiph disclosure is at this point a
>>>> placeholder
>>>> (Xiph folks: it's time to fix that!). However, the two other
>>>> disclosures
>>>> on file provide a patent grant only for necessary patent claims
>>>> and only
>>>> when the standard is practiced in full compliance. These terms (in
>>>> various formulations) are quite common.
>>>>
>>>> In order to ensure one has a license (or can rely on a non-assert
>>>> covenant), one has to ensure one meets the conditions set by the
>>>> rightholder. On stuff such as reciprocity clauses this is
>>>> simple. On
>>>> compliance, it's not always easy.
>>>>
>>>> The traditional compliance test for a media codec is a
>>>> stimulus-response
>>>> test: you feed test vectors into the codec, and you get
>>>> results. If the
>>>> results match, you are in compliance, if not, you are not. Simple.
>>>>
>>>> Without bit exactness, the compliance criteria have to be defined
>>>> differently. We can do so, and, indeed, I recall that this has been
>>>> mentioned as one plan forward. However, I have seen zero
>>>> activity in this
>>>> direction, and I have also not seen any language that mentions
>>>> this in the
>>>> requirements draft. I think that the subject of compliance
>>>> tests, at
>>>> least in its most basic outline, needs to be documented in the
>>>> requirements draft. The details can be taken care of elsewhere
>>>> and later,
>>>> but not too much later. It should be clear that a codec
>>>> candidate (if
>>>> there were more than one) needs to have compliance criteria
>>>> defined before
>>>> that codec candidate can become an RFC. Without that, the key
>>>> goal of the
>>>> WG, a reasonably freely practicable codec, is just not
>>>> achievable in the
>>>> current legal environment (which includes, in this case, the IPR
>>>> disclosures on file).
>>>>
>>>> Of course, it would be sooooo much simpler if we would mandate
>>>> a bit exact
>>>> decoder... Is it really that restricting to require that?
>>>>
>>>> Stephan
>>>>
>>>> P.s.: for the IETF procedures newcomers: humms taken at
>>>> meetings need to
>>>> be confirmed on a mailing list, and consensus needs to be
>>>> declared by the
>>>> chairs. On this subject, I do recall mailing list discussions after
>>>> Maastricht, but I do not recall that consensus was reached, yet
>>>> alone
>>>> declared. (Unfortunately, I currently don't have the time to go
>>>> through
>>>> the mailing list archives to verify my recollection; Sorry.)
>>>>
>>>>
>>>>
>>>> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org
>>>> <mailto:trac@tools.ietf.org>>  wrote:
>>>>
>>>> #12: bit-exact vs. bit-compatible?
>>>>
>>>> Changes (by gmaxwell@=C3=83=E2=80=A6 ):
>>>>
>>>> * status: new =3D>  closed
>>>> * resolution: =3D>  worksforme
>>>>
>>>>
>>>> Comment:
>>>>
>>>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>>>
>>>> "On the topic of bit exact. Consensus was bit exactness is
>>>> not required."
>>>>
>>>> I believe this issue is already closed.
>>>>
>>>> --
>>>>=20
>>>>------------------------------------+----------------------------------
>>>>---
>>>> --
>>>> Reporter: hoene@=C3=83=E2=80=A6 | Owner:
>>>> Type: enhancement | Status: closed
>>>> Priority: minor | Milestone:
>>>> Component: requirements | Version:
>>>> Severity: Active WG Document | Resolution: worksforme
>>>> Keywords: |
>>>>=20
>>>>------------------------------------+----------------------------------
>>>>---
>>>> --
>>>>
>>>> Ticket
>>>> URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>>>> codec<http://tools.ietf.org/codec/>
>>>>
>>>> _______________________________________________
>>>> codec mailing list
>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> codec mailing list
>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>
>>>>
>>>> _______________________________________________
>>>> codec mailing list
>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> codec mailing list
>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> codec mailing list
>>>> codec@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org
>>> https://www.ietf.org/mailman/listinfo/codec
>>
>> _______________________________________________
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>> https://www.ietf.org/mailman/listinfo/codec
>
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On 11-01-25 10:59 PM, Stephan Wenger wrote:
> But please add language in the requirement doc to the extent that there
> MUST be a conformance spec for each candidate codec that allows to
> automatically determine the conformance of a codec implementation.  Absent
> this spec, the IPR statements received are meaningless.

IIRC, the guidelines already say that we need a conformance test.

> My preference would be to have that conformance spec in the same document
> as the codec, so to make citations in IPR statements easy and to avoid a
> charter-update.

Agreed. We really want the conformance tool in the same document as the 
codec itself.

Cheers,

	Jean-Marc

> Stephan
>
>
> On 1.25.2011 18:24 , "Raymond (Juin-Hwey) Chen"<rchen@broadcom.com>  wrote:
>
>> My opinion is that we should not require bit-exactness.  There are many
>> examples of previous codec
>> standards that did not require bit-exactness, including both
>> royalty-bearing codecs (such as MPEG)
>> and royalty-free codecs (such as our BroadVoice/BV16/BV32), so I don't
>> see any real reason why we
>> can't follow that same route.
>>
>> I can see at least two important advantages for not requiring
>> bit-exactness:
>> (1) it allows future innovations to improve the performance of the codec
>> without breaking the bit-
>> stream compatibility, and
>> (2) it allows the codec implementers the greatest flexibility in
>> implementing the codecs in ways
>> that are most efficient for the particular hardware or processor
>> architecture they have.
>> Having a bit-exact standard will kill both.
>>
>> The main drawback of a non-bit-exact standard is that the verification of
>> the conformance of codec
>> implementations is not as straightforward as with a bit-exact standard.
>> However, I don't think
>> that is a show stopper or even a difficult problem.  There are many
>> possible ways to do software-
>> based objective measurements to verify that a candidate codec
>> implementation is likely to sound
>> essentially indistinguishable from the output of the reference codec.  We
>> had previously measured
>> the SNRs and PESQ scores of a large number of test files using the
>> reference codec and candidate
>> codec and examined the statistical distribution of the differences in the
>> scores.  Jean-Marc just
>> posted a nice little Matlab code that built upon psychoacoustic modeling,
>> NMR, averages, outliers
>> checking, etc.  Koen also mentioned another tool.  Even if these tools
>> and methodologies are not
>> perfect yet, with some additional tuning of their algorithms and
>> parameters, it is really not that
>> difficult to come up with objective, software-based verification methods
>> that will give you a high
>> level of confidence whether a candidate codec will sound at least as good
>> as the reference codec
>> (or perhaps better if improvements are made).
>>
>> I like Jean-Marc's idea of having both a floating-point reference codec
>> and a fixed-point
>> reference codec. For people who don't want to run the risk of degrading
>> audio quality through
>> their own fixed-point algorithm conversion, they can just implement the
>> fixed-point reference
>> codec in a bit-exact manner.  For people who want to implement the codec
>> in floating-point (e.g.
>> on computers), the easiest way is just to use the floating-point
>> reference code as is.  In both
>> cases, the conformance verification will be quite easy.
>>
>> Raymond
>>
>> -----Original Message-----
>> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of
>> Jean-Marc Valin
>> Sent: Tuesday, January 25, 2011 12:58 PM
>> To: Koen Vos
>> Cc: codec@ietf.org; Stephen Botzko
>> Subject: Re: [codec] requirements #12 (closed): bit-exact vs.
>> bit-compatible?
>>
>> My preference actually goes with using both fixed-point and floating
>> point.
>> That way, implementers of floating-point decoder can compare with the
>> float
>> version and implementers of the fixed-point decoder can compare with the
>> fixed-point version.
>>
>> This is not unheard of either, if you look at the reference implementation
>> of the 3GPP(2?) EVRC codec, the distribution includes two sets of output
>> vectors. One is for a fixed-point implementation that uses 31-bit
>> precision
>> in the multiplications and the other one is for 32-bit precision. You just
>> choose whatever is more efficient for your DSP and use the appropriate
>> testvectors for testing.
>>
>>         Jean-Marc
>>
>> On 11-01-25 03:37 PM, Koen Vos wrote:
>>> A similar tool is in the code in silk\test\signalCompare.c.  It's based
>>> on LPC, and measures the SNR in a partially-whitened domain.  For music
>>> the critical-band approach of Jean-Marc's tool may work better, not sure.
>>>
>>> One question is about what implementation should be the reference for
>>> compliance testing: floating-point or fixed-point?
>>>
>>> A fixed-point reference allows for bit-exact implementations, which is
>>> nice.  And fixed-point reference implementations are the norm in speech
>>> coding.
>>> However, CELT's floating-point implementation has a larger dynamic
>>> range than the fixed-point version, and is thus arguable better.  A
>>> fixed-point reference could not test this extra dynamic range.
>>>
>>> Any suggestions?
>>> koen.
>>>
>>>
>>> ----- Original Message -----
>>> From: "Jean-Marc Valin"<jean-marc.valin@octasic.com>
>>> To: "Stephen Botzko"<stephen.botzko@gmail.com>
>>> Cc: codec@ietf.org
>>> Sent: Tuesday, January 25, 2011 8:29:25 AM
>>> Subject: Re: [codec] requirements #12 (closed): bit-exact vs.
>>> bit-compatible?
>>>
>>> Just to give an idea of what I'm talking about in terms of compliance
>>> test,
>>> here's a tool I just wrote that could be used for testing. There may
>>> need
>>> to be a bit of refining and playing with the thresholds, but the main
>>> idea
>>> is there:
>>>
>>> http://jmvalin.ca/misc_stuff/compare.m
>>>
>>> It works with Octave, and is likely very easy to get to work with
>>> Matlab.
>>> Any comments?
>>>
>>>        Jean-Marc
>>>
>>> On 11-01-25 10:57 AM, Jean-Marc Valin wrote:
>>>> Hi Stephen,
>>>>
>>>> The reference implementation includes both floating-point and
>>>> fixed-point,
>>>> so I'm not sure it's a good idea to say that one is "better" than the
>>>> other. As for IEEE 754-2008, as far as I know it is not a "bit-exact"
>>>> standard either when it comes to some operations (e.g. transcendental
>>>> function) and even for other operations, most compilers I know of are
>>>> not
>>>> strict IEEE-754 (at least by default) because of issues such as the x86
>>>> extended precision operations.
>>>>
>>>> Regarding MPEG bit-streams, I'm not sure where to get confirmation, but
>>>> this is what Wikipedia has to say about MPEG-1 Layer 3:
>>>>
>>>>
>>>> "Decoding, on the other hand, is carefully defined in the standard.
>>>> Most
>>>> decoders are "bitstream compliant", which means that the decompressed
>>>> output â€“ that they produce from a given MP3 file â€“ will be the
>>>> same, within
>>>> a specified degree of rounding tolerance, as the output specified
>>>> mathematically in the ISO/IEC standard document (ISO/IEC 11172-3).
>>>> Therefore, comparison of decoders is usually based on how
>>>> computationally
>>>> efficient they are (i.e., how much memory or CPU time they use in the
>>>> decoding process)."
>>>>
>>>>
>>>> That implies a standard based on "infinite precision", with a degree of
>>>> tolerance for different implementations.
>>>>
>>>> Cheers,
>>>>
>>>> Jean-Marc
>>>>
>>>> On 11-01-25 10:42 AM, Stephen Botzko wrote:
>>>>> It seems to me that the reference encoder and decoder will be bit
>>>>> exact for
>>>>> a given floating point format, correct? So one could specify IEEE
>>>>> 754-2008
>>>>> when bit exactness is needed in the reference (for instance regression
>>>>> testing), but not require it for compliance.
>>>>>
>>>>> Folks who are modifying the reference encoder or decoder algorithms
>>>>> are "on
>>>>> their own" as far as audio quality is concerned. Though I agree with
>>>>> Stephan that we need to address compliance (when exactly does a
>>>>> ported or
>>>>> otherwise modified encoder/decoder become non-compliant?).
>>>>>
>>>>> Stephen Botzko
>>>>>
>>>>> On Tue, Jan 25, 2011 at 9:34 AM, Roman Shpount<roman@telurix.com
>>>>> <mailto:roman@telurix.com>>   wrote:
>>>>>
>>>>> One more concern that I have related to bit exactness is codec
>>>>> regression testing. One can produce a non-bit-exact encoder that
>>>>> produces reasonable results for a reference decoder, but if paired
>>>>> with
>>>>> a modified, non-bit-exact decoder will produce significant audio
>>>>> artifacts. Since neither decoder or encoder are bit exact we will need
>>>>> to have a test procedure that will validate that both encoder or
>>>>> decoder will not break any standard compliant and non-bit-exact
>>>>> encoders or decoders. Not really sure how this can be done.
>>>>>
>>>>> I might be wrong, but I think MPEG decoders are bit exact and encoders
>>>>> are not.
>>>>> _____________
>>>>> Roman Shpount
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 25, 2011 at 12:03 AM, Jean-Marc Valin<jmvalin@jmvalin.ca
>>>>> <mailto:jmvalin@jmvalin.ca>>   wrote:
>>>>>
>>>>> Hi Stephan,
>>>>>
>>>>> I understand your concern and I'd be interested if you have
>>>>> alternative ways of handling the licensing to avoid any issue. It's
>>>>> not like this is a unique situation. As far as I know, most (all?)
>>>>> MPEG codecs have similar non-bit exact definitions. I have also
>>>>> heard that they also require some IPR licensing...
>>>>>
>>>>> In general the issue of bit-exactness has been discussed and so far
>>>>> I don't recall many arguing in favor of a bit-exact definition.
>>>>> Most of the concerns that have been expressed are solved by
>>>>> considering that non-bitexact does not mean you cannot be bit-exact
>>>>> with the reference encoder. It merely means that you don't *have*
>>>>> to. So regardless of how conformance is defined exactly, one always
>>>>> has the option of being bit-exact with the reference
>>>>> implementation, which obviously guarantees compliance.
>>>>>
>>>>> As for language mentioning compliance, I believe it belongs more to
>>>>> the guidelines (it's not a requirement of the codec itself), which
>>>>> includes the following text:
>>>>>
>>>>> 4. To reduce the risk of bias towards certain CPU/DSP architectures,
>>>>> ideally the decoder specification should not require "bit-exact"
>>>>> conformance with the reference implementation. The output of a
>>>>> decoder implementation should only be "close enough" to the
>>>>> output of the reference decoder. A comparison tool should be
>>>>> provided along with the codec to verify objectively that the
>>>>> output of a decoder is likely to be perceptually
>>>>> indistinguishable from that of the reference decoder. However,
>>>>> an implementation may still wish to produce an output that is
>>>>> bit-exact with the reference implementation to simplify the
>>>>> testing procedure.
>>>>>
>>>>>
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Jean-Marc
>>>>>
>>>>>
>>>>> On 11-01-24 11:02 PM, Stephan Wenger wrote:
>>>>>
>>>>> Hi all:
>>>>>
>>>>> Let me speak once more against this decision (if such a
>>>>> decision were
>>>>> really made; see the p.s.).
>>>>>
>>>>> There are currently three IPR disclosures against the codec
>>>>> draft and/or
>>>>> its predecessors. The Xiph disclosure is at this point a
>>>>> placeholder
>>>>> (Xiph folks: it's time to fix that!). However, the two other
>>>>> disclosures
>>>>> on file provide a patent grant only for necessary patent claims
>>>>> and only
>>>>> when the standard is practiced in full compliance. These terms (in
>>>>> various formulations) are quite common.
>>>>>
>>>>> In order to ensure one has a license (or can rely on a non-assert
>>>>> covenant), one has to ensure one meets the conditions set by the
>>>>> rightholder. On stuff such as reciprocity clauses this is
>>>>> simple. On
>>>>> compliance, it's not always easy.
>>>>>
>>>>> The traditional compliance test for a media codec is a
>>>>> stimulus-response
>>>>> test: you feed test vectors into the codec, and you get
>>>>> results. If the
>>>>> results match, you are in compliance, if not, you are not. Simple.
>>>>>
>>>>> Without bit exactness, the compliance criteria have to be defined
>>>>> differently. We can do so, and, indeed, I recall that this has been
>>>>> mentioned as one plan forward. However, I have seen zero
>>>>> activity in this
>>>>> direction, and I have also not seen any language that mentions
>>>>> this in the
>>>>> requirements draft. I think that the subject of compliance
>>>>> tests, at
>>>>> least in its most basic outline, needs to be documented in the
>>>>> requirements draft. The details can be taken care of elsewhere
>>>>> and later,
>>>>> but not too much later. It should be clear that a codec
>>>>> candidate (if
>>>>> there were more than one) needs to have compliance criteria
>>>>> defined before
>>>>> that codec candidate can become an RFC. Without that, the key
>>>>> goal of the
>>>>> WG, a reasonably freely practicable codec, is just not
>>>>> achievable in the
>>>>> current legal environment (which includes, in this case, the IPR
>>>>> disclosures on file).
>>>>>
>>>>> Of course, it would be sooooo much simpler if we would mandate
>>>>> a bit exact
>>>>> decoder... Is it really that restricting to require that?
>>>>>
>>>>> Stephan
>>>>>
>>>>> P.s.: for the IETF procedures newcomers: humms taken at
>>>>> meetings need to
>>>>> be confirmed on a mailing list, and consensus needs to be
>>>>> declared by the
>>>>> chairs. On this subject, I do recall mailing list discussions after
>>>>> Maastricht, but I do not recall that consensus was reached, yet
>>>>> alone
>>>>> declared. (Unfortunately, I currently don't have the time to go
>>>>> through
>>>>> the mailing list archives to verify my recollection; Sorry.)
>>>>>
>>>>>
>>>>>
>>>>> On 1.24.2011 16:45 , "codec issue tracker"<trac@tools.ietf.org
>>>>> <mailto:trac@tools.ietf.org>>   wrote:
>>>>>
>>>>> #12: bit-exact vs. bit-compatible?
>>>>>
>>>>> Changes (by gmaxwell@Ã… ):
>>>>>
>>>>> * status: new =>   closed
>>>>> * resolution: =>   worksforme
>>>>>
>>>>>
>>>>> Comment:
>>>>>
>>>>> http://www.ietf.org/proceedings/78/minutes/codec.txt
>>>>>
>>>>> "On the topic of bit exact. Consensus was bit exactness is
>>>>> not required."
>>>>>
>>>>> I believe this issue is already closed.
>>>>>
>>>>> --
>>>>>
>>>>> ------------------------------------+----------------------------------
>>>>> ---
>>>>> --
>>>>> Reporter: hoene@Ã… | Owner:
>>>>> Type: enhancement | Status: closed
>>>>> Priority: minor | Milestone:
>>>>> Component: requirements | Version:
>>>>> Severity: Active WG Document | Resolution: worksforme
>>>>> Keywords: |
>>>>>
>>>>> ------------------------------------+----------------------------------
>>>>> ---
>>>>> --
>>>>>
>>>>> Ticket
>>>>> URL:<http://trac.tools.ietf.org/wg/codec/trac/ticket/12#comment:1>
>>>>> codec<http://tools.ietf.org/codec/>
>>>>>
>>>>> _______________________________________________
>>>>> codec mailing list
>>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> codec mailing list
>>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> codec mailing list
>>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> codec mailing list
>>>>> codec@ietf.org<mailto:codec@ietf.org>
>>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> codec mailing list
>>>>> codec@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/codec
>>>>
>>>> _______________________________________________
>>>> codec mailing list
>>>> codec@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/codec
>>>
>>> _______________________________________________
>>> codec mailing list
>>> codec@ietf.org
>>> https://www.ietf.org/mailman/listinfo/codec
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>
>
> _______________________________________________
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From: Koen Vos <koen.vos@skype.net>
To: "Raymond (Juin-Hwey) Chen" <rchen@broadcom.com>
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Subject: Re: [codec] requirements #8 (new): Sample rates?
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------=_Part_1374992_461071571.1296021563187
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Hi Raymond,=20

Streaming of CD music was not an anticipated use case or requirement. That =
said, 44.1 kHz support at the API is certainly nice to have.=20

My understanding of how CELT works is that a *native* 44.1 kHz mode with a =
power-of-2 frame size would break the flexibility to seamlessly switch API =
sampling rates or coded audio bandwidth on the fly. (Jean-Marc can shine mo=
re light on this.) This feature makes it easy for applications to switch du=
ring a call between sources or streams that have different sampling rates.=
=20

I'm also not as negative about resampling as you are: modern hybrid-FIR/IIR=
 designs are very efficient, and I'm convinced a 44.1<->48 kHz resampler wi=
th transparent quality can be built with 10~20 MACs per sample. And resampl=
ing happens everywhere anyway: most hardware ADCs and DACs resample interna=
lly, and even Opus internally resamples in many of its modes already.=20

So if desired we could support 44.1 kHz at the API level, without losing th=
e nice flexibility properties we've so painstakingly built into the design.=
 (Agree we have to think about how to handle 2.5 and 5 ms frames at 44.1 kH=
z, but I'm sure we'll find a solution.)=20

best,=20
koen.=20


----- Original Message -----
From: "Raymond (Juin-Hwey) Chen" <rchen@broadcom.com>=20
To: codec@ietf.org=20
Cc: "jean-marc valin" <jean-marc.valin@usherbrooke.ca>=20
Sent: Tuesday, January 25, 2011 4:53:38 PM=20
Subject: Re: [codec] requirements #8 (new): Sample rates?=20




I agree that supporting sampling rates of 48, 16, and 8 kHz all makes sense=
, but I also think it would be highly desirable to support=20

another common sampling rate used by music CDs: 44.1 kHz.=20



For music streaming applications, a large percentage of the music sources w=
ill have this 44.1 kHz sampling rate. I know it is=20

possible to up-sample them to 48 kHz first and then pass them through the 4=
8 kHz version of Opus. However, doing such up-=20

sampling requires additional processing power and additional latency, and i=
f the sampling rate conversion is not done with a=20

high-quality method (which usually requires higher complexity and higher de=
lay), audio quality degradation may be introduced=20

in the process. For these reasons, many people prefer not to do the 44.1 kH=
z to 48 kHz sampling rate conversion and prefer to=20

encode the 44.1 kHz music directly instead. Therefore, it is highly desirab=
le for Opus to support this 44.1 kHz sampling rate.=20



I understand that it may be inconvenient for the SILK mode or the SILK+CELT=
 hybrid mode to support 44.1 kHz. However, for the=20

CELT-only mode, my understanding is that the current CELT C code already su=
pports the 44.1 kHz sampling rate. Since the CELT-=20

only mode is also the most suitable mode to encode music, it would then mak=
e perfect sense for the CELT-only mode to also=20

support this 44.1 kHz in addition to 48 kHz.=20



Furthermore, currently the CELT mode at 48 kHz supports frame sizes of 2.5,=
 5., 10, and 20 ms to be consistent with the frame=20

sizes of the SILK mode and SILK+CELT hybrid mode, and this results in frame=
 sizes and FFT window sizes that are not powers of 2,=20

which reduces the implementation efficiency. If the CELT mode supports the =
44.1 kHz sampling rate, then since it is no longer=20

possible to support exactly 2.5, 5, 10, and 20 ms at this sampling rate any=
way, I think we might as well choose the FFT window=20

sizes to be powers of 2 to allow more efficient implementation. Again, such=
 power of 2 FFT sizes at 44.1 kHz is already supported=20

in the current C code for CELT, so no new development is needed.=20



In summary, I propose that we allow the CELT-only mode of the Opus codec to=
 officially support the 44.1 kHz sampling rate using=20

power of 2 FFT window sizes to avoid sampling rate conversion and to allow =
the most efficient implementation.=20



Raymond=20




From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf Of R=
oman Shpount=20
Sent: Monday, January 24, 2011 5:08 PM=20
To: codec@ietf.org=20
Cc: jean-marc.valin@usherbrooke.ca=20
Subject: Re: [codec] requirements #8 (new): Sample rates?=20



I actually do believe that the requirement is already addressed by section =
4.1 of the document. I was responding to a comment that only full band is r=
equired, which is clearly not what we agreed upon and not what's in the doc=
ument.=20
_____________=20
Roman Shpount=20




On Mon, Jan 24, 2011 at 7:40 PM, codec issue tracker < trac@tools.ietf.org =
> wrote:=20


#8: Sample rates?=20


Comment(by gmaxwell@=E2=80=A6):=20


On 11-01-24 07:14 PM, Roman Shpount wrote:=20
> I would like to see 8 and 16 KHz as required rates for the codec to=20
insure=20
> interoperability with existing narrowband and wideband codecs. In other=
=20
> words we should be able to negotiate 8 or 16 Khz sample rate if audio=20
will=20
> be transcoded for PSTN or wideband codec such as G.722=20


Jean-Marc wrote:=20
> I believe such requirement for narrowband/wideband is already present,=20
but=20
> I don't mind making it even more explicit is necessary.=20




I believe that we since are close enough to finalizing the draft we should=
=20
try to include proposed language with our issues. Otherwise there may be=20
no clear path forward.=20

Some of the requirements specify that compatibility with=20
wideband/narrowband is important, but for the avoidance of doubt, I'll=20
suggest:=20

At the end of 4.1. Operating space:=20

"Because interoperation with existing wideband and narrowband facilities=20
is essential at least one method of interoperation must be provided=20
regardless of the codec's operating mode, sample rate, or bitrate."=20

Of course, I would be perfectly happy to leave this out entirely (as I=20
believe the application section 2.1 already implies the requirement) or=20
use some other language.=20

--=20


------------------------------------+--------------------------------------=
-=20
Reporter: hoene@=E2=80=A6 | Owner: jean-marc.valin@=E2=80=A6=20
Type: enhancement | Status: new=20
Priority: minor | Milestone:=20
Component: requirements | Version:=20
Severity: Active WG Document | Keywords:=20
------------------------------------+--------------------------------------=
-=20

Ticket URL: < http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:4 >=
=20


codec < http://tools.ietf.org/codec/ >=20

_______________________________________________=20
codec mailing list=20
codec@ietf.org=20



https://www.ietf.org/mailman/listinfo/codec=20


_______________________________________________=20
codec mailing list=20
codec@ietf.org=20
https://www.ietf.org/mailman/listinfo/codec=20

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<html><head><style type=3D'text/css'>p { margin: 0; }</style></head><body><=
div style=3D'font-family: Times New Roman; font-size: 12pt; color: #000000'=
>Hi Raymond,<br><br>Streaming of CD music was not an anticipated use case o=
r requirement.&nbsp; That said, 44.1 kHz support at the API is certainly ni=
ce to have.<br><br>My understanding of how CELT works is that a *native* 44=
.1 kHz mode with a power-of-2 frame size would break the flexibility to sea=
mlessly switch API sampling rates or coded audio bandwidth on the fly.&nbsp=
; (Jean-Marc can shine more light on this.) This feature makes it easy for =
applications to switch during a call between sources or streams that have d=
ifferent sampling rates.<br><br>I'm also not as negative about resampling a=
s you are: modern hybrid-FIR/IIR designs are very efficient, and I'm convin=
ced a 44.1&lt;-&gt;48 kHz resampler with transparent quality can be built w=
ith 10~20 MACs per sample.&nbsp; And resampling happens everywhere anyway: =
most hardware ADCs and DACs resample internally, and even Opus internally r=
esamples in many of its modes already.&nbsp; <br><br>So if desired we could=
 support 44.1 kHz at the API level, without losing the
 nice flexibility properties we've so painstakingly built into the=20
design.&nbsp; (Agree we have to think about how to handle 2.5 and 5 ms fram=
es at 44.1 kHz, but I'm sure we'll find a solution.)<br><br>best,<br>koen.<=
br><br><br><hr id=3D"zwchr"><b>From: </b>"Raymond (Juin-Hwey) Chen" &lt;rch=
en@broadcom.com&gt;<br><b>To: </b>codec@ietf.org<br><b>Cc: </b>"jean-marc v=
alin" &lt;jean-marc.valin@usherbrooke.ca&gt;<br><b>Sent: </b>Tuesday, Janua=
ry 25, 2011 4:53:38 PM<br><b>Subject: </b>Re: [codec] requirements #8 (new)=
: Sample rates?<br><br>




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<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">I agree that=
 supporting sampling rates of 48, 16, and 8 kHz all
makes sense, but I also think it would be highly desirable to support </spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">another comm=
on sampling rate used by music CDs: 44.1 kHz.&nbsp; </span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">&nbsp;</span=
></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">For music st=
reaming applications, a large percentage of the music
sources will have this 44.1 kHz sampling rate.&nbsp; I know it is </span></=
p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">possible to =
up-sample them to 48 kHz first and then pass them through
the 48 kHz version of Opus.&nbsp; However, doing such up-</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">sampling req=
uires additional processing power and additional
latency, and if the sampling rate conversion is not done with a </span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">high-quality=
 method (which usually requires higher complexity
and higher delay), audio quality degradation may be introduced </span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">in the proce=
ss.&nbsp; For these reasons, many people prefer not
to do the 44.1 kHz to 48 kHz sampling rate conversion and prefer to </span>=
</p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">encode the 4=
4.1 kHz music directly instead.&nbsp; Therefore, it
is highly desirable for Opus to support this 44.1 kHz sampling rate.</span>=
</p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">&nbsp;</span=
></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">I understand=
 that it may be inconvenient for the SILK mode or the
SILK+CELT hybrid mode to support 44.1 kHz.&nbsp; However, for the </span></=
p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">CELT-only mo=
de, my understanding is that the current CELT C code
already supports the 44.1 kHz sampling rate.&nbsp; Since the CELT-</span></=
p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">only mode is=
 also the most suitable mode to encode music, it
would then make perfect sense for the CELT-only mode to also </span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">support this=
 44.1 kHz in addition to 48 kHz.&nbsp; </span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">&nbsp;</span=
></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Furthermore,=
 currently the CELT mode at 48 kHz supports frame
sizes of 2.5, 5., 10, and 20 ms to be consistent with the frame </span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">sizes of the=
 SILK mode and SILK+CELT hybrid mode, and this
results in frame sizes and FFT window sizes that are not powers of 2, </spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">which reduce=
s the implementation efficiency.&nbsp; If the CELT
mode supports the 44.1 kHz sampling rate, then since it is no longer </span=
></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">possible to =
support exactly 2.5, 5, 10, and 20 ms at this
sampling rate anyway, I think we might as well choose the FFT window </span=
></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">sizes to be =
powers of 2 to allow more efficient
implementation.&nbsp; Again, such power of 2 FFT sizes at 44.1 kHz is alrea=
dy supported
</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">in the curre=
nt C code for CELT, so no new development is needed.</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">&nbsp;</span=
></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">In summary, =
I propose that we allow the CELT-only mode of the Opus
codec to officially support the 44.1 kHz sampling rate using </span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">power of 2 F=
FT window sizes to avoid sampling rate conversion
and to allow the most efficient implementation.</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">&nbsp;</span=
></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Raymond</spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; font-family: &quot;C=
alibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">&nbsp;</span=
></p>

<div style=3D"border-right: medium none; border-width: 1pt medium medium; b=
order-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-tex=
t-color -moz-use-text-color; padding: 3pt 0in 0in;">

<p class=3D"MsoNormal"><b><span style=3D"font-size: 10pt; font-family: &quo=
t;Tahoma&quot;,&quot;sans-serif&quot;;">From:</span></b><span style=3D"font=
-size: 10pt; font-family: &quot;Tahoma&quot;,&quot;sans-serif&quot;;">
codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] <b>On Behalf Of </b>=
Roman
Shpount<br>
<b>Sent:</b> Monday, January 24, 2011 5:08 PM<br>
<b>To:</b> codec@ietf.org<br>
<b>Cc:</b> jean-marc.valin@usherbrooke.ca<br>
<b>Subject:</b> Re: [codec] requirements #8 (new): Sample rates?</span></p>

</div>

<p class=3D"MsoNormal">&nbsp;</p>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">I actually do believe=
 that the
requirement is already addressed by section 4.1 of the document. I was
responding to a comment that only full band is required, which is clearly n=
ot
what we agreed upon and not what's in the document.<br clear=3D"all">
_____________<br>
Roman Shpount<br>
<br>
</p>

<div>

<p class=3D"MsoNormal">On Mon, Jan 24, 2011 at 7:40 PM, codec issue tracker=
 &lt;<a href=3D"mailto:trac@tools.ietf.org" target=3D"_blank">trac@tools.ie=
tf.org</a>&gt; wrote:</p>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">#8: Sample rates?<br>
<br>
<br>
Comment(by gmaxwell@=E2=80=A6):</p>

</div>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">&nbsp;On 11-01-24 07:=
14 PM,
Roman Shpount wrote:<br>
&nbsp;&gt; I would like to see 8 and 16 KHz as required rates for the codec=
 to<br>
&nbsp;insure<br>
&nbsp;&gt; interoperability with existing narrowband and wideband codecs. I=
n
other<br>
&nbsp;&gt; words we should be able to negotiate 8 or 16 Khz sample rate if
audio<br>
&nbsp;will<br>
&nbsp;&gt; be transcoded for PSTN or wideband codec such as G.722</p>

</div>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">&nbsp;Jean-Marc wrote=
:<br>
&nbsp;&gt; I believe such requirement for narrowband/wideband is already
present,<br>
&nbsp;but<br>
&nbsp;&gt; I don't mind making it even more explicit is necessary.<br>
<br>
<br>
</p>

</div>

<p class=3D"MsoNormal">&nbsp;I believe that we since are close enough to fi=
nalizing
the draft we should<br>
&nbsp;try to include proposed language with our issues. &nbsp;Otherwise the=
re
may be<br>
&nbsp;no clear path forward.<br>
<br>
&nbsp;Some of the requirements specify that compatibility with<br>
&nbsp;wideband/narrowband is important, but for the avoidance of doubt, I'l=
l<br>
&nbsp;suggest:<br>
<br>
&nbsp;At the end of 4.1. &nbsp;Operating space:<br>
<br>
&nbsp;"Because interoperation with existing wideband and narrowband
facilities<br>
&nbsp;is essential at least one method of interoperation must be provided<b=
r>
&nbsp;regardless of the codec's operating mode, sample rate, or bitrate."<b=
r>
<br>
&nbsp;Of course, I would be perfectly happy to leave this out entirely (as =
I<br>
&nbsp;believe the application section 2.1 already implies the requirement) =
or<br>
&nbsp;use some other language.<br>
<br>
--</p>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">---------------------=
---------------+---------------------------------------<br>
&nbsp;Reporter: &nbsp;hoene@=E2=80=A6 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp;
&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; Owner: &nbsp;jean-marc.valin@=E2=80=A6=
<br>
&nbsp; &nbsp; Type: &nbsp;enhancement &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp;
| &nbsp; &nbsp; &nbsp;Status: &nbsp;new<br>
&nbsp;Priority: &nbsp;minor &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; | &nbsp; Milestone:<br>
Component: &nbsp;requirements &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| &n=
bsp;
&nbsp; Version:<br>
&nbsp;Severity: &nbsp;Active WG Document &nbsp; &nbsp; &nbsp;| &nbsp;
&nbsp;Keywords:<br>
------------------------------------+--------------------------------------=
-</p>

</div>

<p class=3D"MsoNormal">Ticket URL: &lt;<a href=3D"http://trac.tools.ietf.or=
g/wg/codec/trac/ticket/8#comment:4" target=3D"_blank">http://trac.tools.iet=
f.org/wg/codec/trac/ticket/8#comment:4</a>&gt;</p>

<div>

<p class=3D"MsoNormal">codec &lt;<a href=3D"http://tools.ietf.org/codec/" t=
arget=3D"_blank">http://tools.ietf.org/codec/</a>&gt;<br>
<br>
_______________________________________________<br>
codec mailing list<br>
<a href=3D"mailto:codec@ietf.org" target=3D"_blank">codec@ietf.org</a></p>

</div>

<div>

<div>

<p class=3D"MsoNormal"><a href=3D"https://www.ietf.org/mailman/listinfo/cod=
ec" target=3D"_blank">https://www.ietf.org/mailman/listinfo/codec</a></p>

</div>

</div>

</div>

<p class=3D"MsoNormal">&nbsp;</p>

</div>

<br>_______________________________________________<br>codec mailing list<b=
r>codec@ietf.org<br>https://www.ietf.org/mailman/listinfo/codec<br></div></=
body></html>
------=_Part_1374992_461071571.1296021563187--

From jean-marc.valin@octasic.com  Wed Jan 26 05:48:30 2011
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Date: Wed, 26 Jan 2011 08:51:16 -0500
From: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Subject: Re: [codec] requirements #8 (new): Sample rates?
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Hi,

I think it goes beyond just having native support for 44.1 kHz. There are 
several existing applications that are currently using CELT in modes that 
are do not have a 2.5/5/10/20 ms frame size. There are different reasons 
for this depending on the application. Some of these are:
1) Some implementers choose frames that are powers of two so that they can 
use their platform's highly-optimized FFT
2) For ultra low delay applications, such as "jamming over the network" 
remote music performances, one can reduce latency by choosing a frame size 
that matches the soundcard's buffer size (which is almost always a power of 
two).
3) There are even users that consider 2.5 ms to be too high of a delay for 
their applications and use frames of 1.33 ms (64 samples at 48 kHz).
4) In some cases, transport or protocol issues impose some constraints on 
the frame size. For example, the DAB digital radio standard requires frames 
of 24 ms, which CELT can also handle.

I agree that the use cases I listed above may not be part of our "core use 
cases". However, CELT already handles all these cases with no change 
required in the code. Considering that people will keep using CELT for 
these "special" applications, the only question is how we want it to 
happen. I can see essentially two possible approaches:

1) Defining "custom modes" that handle the use cases above and that operate 
differently from the "main modes" that we have defined (e.g. no bandwidth 
switching on the fly).
2) We can exclude these modes from Opus and have these people using "CELT" 
as a non-standard codec.

I don't have a strong opinion on the subject, but I tend to lean towards 1) 
mainly because we would avoid having Opus and CELT as two "competing" 
standards. Also, it would mean that the IETF would have change control over 
all the operating modes instead of "splitting" change control in two. OTOH, 
I do agree that we would need a way to avoid causing compatibility issues 
(perhaps by using a different labelling for these "custom modes").

Cheers,

	Jean-Marc

On 11-01-26 12:59 AM, Koen Vos wrote:
> Hi Raymond,
>
> Streaming of CD music was not an anticipated use case or requirement. That
> said, 44.1 kHz support at the API is certainly nice to have.
>
> My understanding of how CELT works is that a *native* 44.1 kHz mode with a
> power-of-2 frame size would break the flexibility to seamlessly switch API
> sampling rates or coded audio bandwidth on the fly. (Jean-Marc can shine
> more light on this.) This feature makes it easy for applications to switch
> during a call between sources or streams that have different sampling rates.
>
> I'm also not as negative about resampling as you are: modern hybrid-FIR/IIR
> designs are very efficient, and I'm convinced a 44.1<->48 kHz resampler
> with transparent quality can be built with 10~20 MACs per sample. And
> resampling happens everywhere anyway: most hardware ADCs and DACs resample
> internally, and even Opus internally resamples in many of its modes already.
>
> So if desired we could support 44.1 kHz at the API level, without losing
> the nice flexibility properties we've so painstakingly built into the
> design. (Agree we have to think about how to handle 2.5 and 5 ms frames at
> 44.1 kHz, but I'm sure we'll find a solution.)
>
> best,
> koen.
>
>
> ---------------------------------------------------------------------------
> *From: *"Raymond (Juin-Hwey) Chen" <rchen@broadcom.com>
> *To: *codec@ietf.org
> *Cc: *"jean-marc valin" <jean-marc.valin@usherbrooke.ca>
> *Sent: *Tuesday, January 25, 2011 4:53:38 PM
> *Subject: *Re: [codec] requirements #8 (new): Sample rates?
>
> I agree that supporting sampling rates of 48, 16, and 8 kHz all makes
> sense, but I also think it would be highly desirable to support
>
> another common sampling rate used by music CDs: 44.1 kHz.
>
> For music streaming applications, a large percentage of the music sources
> will have this 44.1 kHz sampling rate. I know it is
>
> possible to up-sample them to 48 kHz first and then pass them through the
> 48 kHz version of Opus. However, doing such up-
>
> sampling requires additional processing power and additional latency, and
> if the sampling rate conversion is not done with a
>
> high-quality method (which usually requires higher complexity and higher
> delay), audio quality degradation may be introduced
>
> in the process. For these reasons, many people prefer not to do the 44.1
> kHz to 48 kHz sampling rate conversion and prefer to
>
> encode the 44.1 kHz music directly instead. Therefore, it is highly
> desirable for Opus to support this 44.1 kHz sampling rate.
>
> I understand that it may be inconvenient for the SILK mode or the SILK+CELT
> hybrid mode to support 44.1 kHz. However, for the
>
> CELT-only mode, my understanding is that the current CELT C code already
> supports the 44.1 kHz sampling rate. Since the CELT-
>
> only mode is also the most suitable mode to encode music, it would then
> make perfect sense for the CELT-only mode to also
>
> support this 44.1 kHz in addition to 48 kHz.
>
> Furthermore, currently the CELT mode at 48 kHz supports frame sizes of 2.5,
> 5., 10, and 20 ms to be consistent with the frame
>
> sizes of the SILK mode and SILK+CELT hybrid mode, and this results in frame
> sizes and FFT window sizes that are not powers of 2,
>
> which reduces the implementation efficiency. If the CELT mode supports the
> 44.1 kHz sampling rate, then since it is no longer
>
> possible to support exactly 2.5, 5, 10, and 20 ms at this sampling rate
> anyway, I think we might as well choose the FFT window
>
> sizes to be powers of 2 to allow more efficient implementation. Again, such
> power of 2 FFT sizes at 44.1 kHz is already supported
>
> in the current C code for CELT, so no new development is needed.
>
> In summary, I propose that we allow the CELT-only mode of the Opus codec to
> officially support the 44.1 kHz sampling rate using
>
> power of 2 FFT window sizes to avoid sampling rate conversion and to allow
> the most efficient implementation.
>
> Raymond
>
> *From:*codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] *On Behalf Of
> *Roman Shpount
> *Sent:* Monday, January 24, 2011 5:08 PM
> *To:* codec@ietf.org
> *Cc:* jean-marc.valin@usherbrooke.ca
> *Subject:* Re: [codec] requirements #8 (new): Sample rates?
>
> I actually do believe that the requirement is already addressed by section
> 4.1 of the document. I was responding to a comment that only full band is
> required, which is clearly not what we agreed upon and not what's in the
> document.
> _____________
> Roman Shpount
>
> On Mon, Jan 24, 2011 at 7:40 PM, codec issue tracker <trac@tools.ietf.org
> <mailto:trac@tools.ietf.org>> wrote:
>
> #8: Sample rates?
>
>
> Comment(by gmaxwell@â€¦):
>
> On 11-01-24 07:14 PM, Roman Shpount wrote:
>  > I would like to see 8 and 16 KHz as required rates for the codec to
> insure
>  > interoperability with existing narrowband and wideband codecs. In other
>  > words we should be able to negotiate 8 or 16 Khz sample rate if audio
> will
>  > be transcoded for PSTN or wideband codec such as G.722
>
> Jean-Marc wrote:
>  > I believe such requirement for narrowband/wideband is already present,
> but
>  > I don't mind making it even more explicit is necessary.
>
>
> I believe that we since are close enough to finalizing the draft we should
> try to include proposed language with our issues. Otherwise there may be
> no clear path forward.
>
> Some of the requirements specify that compatibility with
> wideband/narrowband is important, but for the avoidance of doubt, I'll
> suggest:
>
> At the end of 4.1. Operating space:
>
> "Because interoperation with existing wideband and narrowband facilities
> is essential at least one method of interoperation must be provided
> regardless of the codec's operating mode, sample rate, or bitrate."
>
> Of course, I would be perfectly happy to leave this out entirely (as I
> believe the application section 2.1 already implies the requirement) or
> use some other language.
>
> --
>
> ------------------------------------+---------------------------------------
> Reporter: hoene@â€¦ | Owner: jean-marc.valin@â€¦
> Type: enhancement | Status: new
> Priority: minor | Milestone:
> Component: requirements | Version:
> Severity: Active WG Document | Keywords:
> ------------------------------------+---------------------------------------
>
> Ticket URL: <http://trac.tools.ietf.org/wg/codec/trac/ticket/8#comment:4>
>
> codec <http://tools.ietf.org/codec/>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org <mailto:codec@ietf.org>
>
> https://www.ietf.org/mailman/listinfo/codec
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
>
>
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


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Subject: Re: [codec] requirements #8 (new): Sample rates?
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Hello,

I just wanted to give in my 2 cents about 44.1Khz native support.

99% of all the audio we use celt and eventually opus for are encoded at
44.1Khz. They are provided to us that way. Which means that without 44.1khz
support we'll have to up-sample 99% of our audio most likely in a
preprocess build making us maintain 2 sets of audibly identical files. We
had to do it before with speex where we converted from 44.1 down to 32khz
to use its native ultrawideband but it really wasn't the easiest. We had
thousands of files duplicated and every now and then a few of these getting
updated from the source, forcing us to redo massive conversions each time
to make sure we didn't miss one somewhere.

Also about upsampling not costing much, I beg to differ. We had to work
with hardware that could decode speex ultrawideband 32Khz just fine but the
decoding alone was eating up all our CPU leaving not much else to do the
real work that we needed to do. We had to use 16khz instead to make it all
work. 48Khz, 44.1khz might not look like a big difference when working on a
desktop but when you're counting bytes to see how you can reduce you memory
consumption it could mean the world.

So strickly from a user of the codec's view, native 44.1Khz would certainly
make working with celt/opus a lot easier. I'm guessing that I'm not the
only one in that case based on this thread. Easier would also lead to
faster adoption.

Sorry I didn't intend to write that much. In short 44.1Khz: great if you
can do it, if not we'll just have to work around it.

-Pascal


From gmaxwell@juniper.net  Wed Jan 26 11:38:23 2011
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To: "Pochol@WebfootGames.com" <Pochol@WebfootGames.com>, Jean-Marc Valin <jean-marc.valin@octasic.com>
Date: Wed, 26 Jan 2011 11:32:01 -0800
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I'm very concerned that some people may believe that 44.1k is just another =
checkbox that can be added
without cost or much consideration. This isn't the case. I think it's essen=
tial that we delineate realtime VoIP
style usage from other applications.

I support Jean Marc's option (1), which I think allows us to have our cake =
and eat it too. It's the closest thing to a=20
"cost free" option that I think we're going to get. This option basically s=
eparates the codec into two profiles,
one which imposes sampling rate restrictions in exchange for many important=
 advantages, and one which
does not, but misses the advantages. =20

JM's option (2) would also work. But I don't like the idea of the market co=
nfusion created by a totally
separate codec which has significant overlap with Opus, but isn't Opus.  Ba=
sically, I think it's silly
for part of the working group's output to compete with itself.  I'd prefer =
to just have "Opus" and
"Opus-custom" or whatever.


There are several reasons that supporting 44.1kHz in the primary Opus profi=
le would be bad:

One reason for this is that quite a bit of hardware (even on desktops) can =
only do 48kHz (or closely related
rates) and even when the hardware can do multiple rates it can only ever do=
 one rate at a time so if=20
its even possible that multiple applications may play sound at once then th=
e only way to avoid resampling
is if they are all running at a common rate.

For the 48kHz related rates we can do very computationally cheap handling o=
f different rates purely
inside the codec. If their hardware supports any mode out of the 48kHz fami=
ly, then they'll need no costly
resampling at all. And if they do  need run at 44.1k they can resample in a=
nd out of the codec without imposing on (or
negotiating with) the far end.

Another one is that Opus (as described in the draft, without 44.1kHz) can s=
witch between any of
its supported modes, all on the fly, without creating any surprising imposi=
tions on the clients.
This is possible only because of the closely related nature of the supporte=
d rates, and 44.1kHz
can't be accommodated in this scheme.

The on the fly switching and lack of requirement to negotiate, also means t=
hat two opus devices can
 communicate without transcoding, even if they were spliced long after nego=
tiation. (e.g. as part
of a conference gateway).


On the other end of the spectrum=97 the current CELT library and bitstream =
is extremely flexible.
It supports a great many frame sizes and sample rates and I've personally a=
rgued against
every limitation we've imposed on it, because I like the idea that CELT can=
 fit into every
niche requirement (like the DAB frame sizes).

But this flexibility has a serious price for interoperable implementations:=
 They must carry substantially
more code (the limited rates/sample sizes means that a simple table can rep=
lace several hundred
lines of tricky bit-exact initialization code), cope with increased peak CP=
U usage (e.g. if some device
only speaks 64 sample frames, 96KHz you might need 10x the CPU power to spe=
ak to it compared=20
to your preferred mode), and undertake more complicated negotiation and tes=
ting (CELT can support
far more unique mixtures of sample rate, frame-size, and channel count then=
 there are RTP payload
types).=20

So basically, I support fully supporting oddball configurations as a well s=
pecified standardized mode,
but I oppose subjecting the general VoIP/RTP users to the increase complexi=
ty and limitations of
the more rate/framesize agile configurations.

I expect that most users which care about the 'custom' modes are doing othe=
r specialized things
and won't be expected to ever interop with a random Opus phone except (mayb=
e) via a gateway,
and that most of them won't even speak RTP=97 so the separation shouldn't e=
ven make=20
much difference to them at all.


Thoughts?




________________________________________
From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of Pascal P=
ochol [Pochol@WebfootGames.com]
Sent: Wednesday, January 26, 2011 6:06 PM
To: Jean-Marc Valin
Cc: codec@ietf.org
Subject: Re: [codec] requirements #8 (new): Sample rates?

Hello,

I just wanted to give in my 2 cents about 44.1Khz native support.

99% of all the audio we use celt and eventually opus for are encoded at
44.1Khz. They are provided to us that way. Which means that without 44.1khz
support we'll have to up-sample 99% of our audio most likely in a
preprocess build making us maintain 2 sets of audibly identical files. We
had to do it before with speex where we converted from 44.1 down to 32khz
to use its native ultrawideband but it really wasn't the easiest. We had
thousands of files duplicated and every now and then a few of these getting
updated from the source, forcing us to redo massive conversions each time
to make sure we didn't miss one somewhere.

Also about upsampling not costing much, I beg to differ. We had to work
with hardware that could decode speex ultrawideband 32Khz just fine but the
decoding alone was eating up all our CPU leaving not much else to do the
real work that we needed to do. We had to use 16khz instead to make it all
work. 48Khz, 44.1khz might not look like a big difference when working on a
desktop but when you're counting bytes to see how you can reduce you memory
consumption it could mean the world.

So strickly from a user of the codec's view, native 44.1Khz would certainly
make working with celt/opus a lot easier. I'm guessing that I'm not the
only one in that case based on this thread. Easier would also lead to
faster adoption.

Sorry I didn't intend to write that much. In short 44.1Khz: great if you
can do it, if not we'll just have to work around it.

-Pascal

_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec

From Pochol@WebfootGames.com  Wed Jan 26 12:50:53 2011
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Gregory,

> I'm very concerned that some people may believe that 44.1k is just
> another checkbox that can be added without cost or much consideration.

I don't think that's the case. We all know that if it was easy to handle
every sample rate imaginable that it would have been done already.

Again talking about our personal case which has nothing to do with VOIP we
have a lot of audio data handed to us in 44.1Khz and 22.5Khz. Other popular
rates for us are 16Khz and 8Khz which are already handled if I'm not
mistaken by opus? Easily derived from 48khz. Before using speex last year I
had never encountered anything at 32khz. It'll probably a few more years
before we're allowed to use cpus powerful enough and enough ram to handle
even 32khz. For now we'll most likely still down sample everything to
16khz. The artists don't like programmers massaging their data but they
understand technical limitations so for now we're in luck.

> I support Jean Marc's option (1), which I think allows us to have our
> cake and eat it too. It's the closest thing to a "cost free" option
> that I think we're going to get. This option basically separates the
> codec into two profiles, one which imposes sampling rate restrictions
> in exchange for many important advantages, and one which does not, but
> misses the advantages.

Out of curiosity, what would be the advantages that we miss in such case?

> JM's option (2) would also work. But I don't like the idea of the
> market confusion created by a totally separate codec which has
> significant overlap with Opus, but isn't Opus.  Basically, I think it's
> silly for part of the working group's output to compete with itself.
> I'd prefer to just have "Opus" and "Opus-custom" or whatever.

I totally agree.

> I expect that most users which care about the 'custom' modes are doing
> other specialized things and won't be expected to ever interop with a
> random Opus phone except (maybe) via a gateway, and that most of them
> won't even speak RTPâ€” so the separation shouldn't even make much
> difference to them at all.

Absolutely right In our case we'd be using for things that wouldn't need to
interact with any other devices actually. We use it as a very efficient way
to decode stored pre-encoded music/speech streams.

-Pascal


From koen.vos@skype.net  Wed Jan 26 13:05:54 2011
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Cc: Pochol@WebfootGames.com, codec@ietf.org
Subject: Re: [codec] requirements #8 (new): Sample rates?
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So what needs to be decided is:

1. How to call these custom modes (for instance in the SDP descriptor).  As=
 long as we agree that it is not "Opus", then we prevent compatibility issu=
es and we keep our on-the-fly switching flexibility within the standard Opu=
s format.  I'm fine with "Opus-custom".

2. How to enable the custom modes in the code.  I think there must be a sma=
ll barrier so that users don't unintentionally start using them.  Maybe a w=
ell-commented #define somewhere, plus an API control flag to put the codec =
in custom mode?

3. Should there be a set of "official" custom modes, to aid interop even wi=
th Opus-custom?  Users could still pick and choose any modes they want, but=
 having the list could make users gravitate towards a few common choices.

koen.



----- Original Message -----
From: "Gregory Maxwell" <gmaxwell@juniper.net>
To: Pochol@WebfootGames.com, "Jean-Marc Valin" <jean-marc.valin@octasic.com=
>
Cc: codec@ietf.org
Sent: Wednesday, January 26, 2011 11:32:01 AM
Subject: Re: [codec] requirements #8 (new): Sample rates?



I'm very concerned that some people may believe that 44.1k is just another =
checkbox that can be added
without cost or much consideration. This isn't the case. I think it's essen=
tial that we delineate realtime VoIP
style usage from other applications.

I support Jean Marc's option (1), which I think allows us to have our cake =
and eat it too. It's the closest thing to a=20
"cost free" option that I think we're going to get. This option basically s=
eparates the codec into two profiles,
one which imposes sampling rate restrictions in exchange for many important=
 advantages, and one which
does not, but misses the advantages. =20

JM's option (2) would also work. But I don't like the idea of the market co=
nfusion created by a totally
separate codec which has significant overlap with Opus, but isn't Opus.  Ba=
sically, I think it's silly
for part of the working group's output to compete with itself.  I'd prefer =
to just have "Opus" and
"Opus-custom" or whatever.


There are several reasons that supporting 44.1kHz in the primary Opus profi=
le would be bad:

One reason for this is that quite a bit of hardware (even on desktops) can =
only do 48kHz (or closely related
rates) and even when the hardware can do multiple rates it can only ever do=
 one rate at a time so if=20
its even possible that multiple applications may play sound at once then th=
e only way to avoid resampling
is if they are all running at a common rate.

For the 48kHz related rates we can do very computationally cheap handling o=
f different rates purely
inside the codec. If their hardware supports any mode out of the 48kHz fami=
ly, then they'll need no costly
resampling at all. And if they do  need run at 44.1k they can resample in a=
nd out of the codec without imposing on (or
negotiating with) the far end.

Another one is that Opus (as described in the draft, without 44.1kHz) can s=
witch between any of
its supported modes, all on the fly, without creating any surprising imposi=
tions on the clients.
This is possible only because of the closely related nature of the supporte=
d rates, and 44.1kHz
can't be accommodated in this scheme.

The on the fly switching and lack of requirement to negotiate, also means t=
hat two opus devices can
 communicate without transcoding, even if they were spliced long after nego=
tiation. (e.g. as part
of a conference gateway).


On the other end of the spectrum=E2=80=94 the current CELT library and bits=
tream is extremely flexible.
It supports a great many frame sizes and sample rates and I've personally a=
rgued against
every limitation we've imposed on it, because I like the idea that CELT can=
 fit into every
niche requirement (like the DAB frame sizes).

But this flexibility has a serious price for interoperable implementations:=
 They must carry substantially
more code (the limited rates/sample sizes means that a simple table can rep=
lace several hundred
lines of tricky bit-exact initialization code), cope with increased peak CP=
U usage (e.g. if some device
only speaks 64 sample frames, 96KHz you might need 10x the CPU power to spe=
ak to it compared=20
to your preferred mode), and undertake more complicated negotiation and tes=
ting (CELT can support
far more unique mixtures of sample rate, frame-size, and channel count then=
 there are RTP payload
types).=20

So basically, I support fully supporting oddball configurations as a well s=
pecified standardized mode,
but I oppose subjecting the general VoIP/RTP users to the increase complexi=
ty and limitations of
the more rate/framesize agile configurations.

I expect that most users which care about the 'custom' modes are doing othe=
r specialized things
and won't be expected to ever interop with a random Opus phone except (mayb=
e) via a gateway,
and that most of them won't even speak RTP=E2=80=94 so the separation shoul=
dn't even make=20
much difference to them at all.


Thoughts?




________________________________________
From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of Pascal P=
ochol [Pochol@WebfootGames.com]
Sent: Wednesday, January 26, 2011 6:06 PM
To: Jean-Marc Valin
Cc: codec@ietf.org
Subject: Re: [codec] requirements #8 (new): Sample rates?

Hello,

I just wanted to give in my 2 cents about 44.1Khz native support.

99% of all the audio we use celt and eventually opus for are encoded at
44.1Khz. They are provided to us that way. Which means that without 44.1khz
support we'll have to up-sample 99% of our audio most likely in a
preprocess build making us maintain 2 sets of audibly identical files. We
had to do it before with speex where we converted from 44.1 down to 32khz
to use its native ultrawideband but it really wasn't the easiest. We had
thousands of files duplicated and every now and then a few of these getting
updated from the source, forcing us to redo massive conversions each time
to make sure we didn't miss one somewhere.

Also about upsampling not costing much, I beg to differ. We had to work
with hardware that could decode speex ultrawideband 32Khz just fine but the
decoding alone was eating up all our CPU leaving not much else to do the
real work that we needed to do. We had to use 16khz instead to make it all
work. 48Khz, 44.1khz might not look like a big difference when working on a
desktop but when you're counting bytes to see how you can reduce you memory
consumption it could mean the world.

So strickly from a user of the codec's view, native 44.1Khz would certainly
make working with celt/opus a lot easier. I'm guessing that I'm not the
only one in that case based on this thread. Easier would also lead to
faster adoption.

Sorry I didn't intend to write that much. In short 44.1Khz: great if you
can do it, if not we'll just have to work around it.

-Pascal

_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec
_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec

From koen.vos@skype.net  Wed Jan 26 13:15:58 2011
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Date: Wed, 26 Jan 2011 22:18:56 +0100 (CET)
From: Koen Vos <koen.vos@skype.net>
To: Gregory Maxwell <gmaxwell@juniper.net>
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Cc: codec@ietf.org, Pochol@WebfootGames.com
Subject: Re: [codec] requirements #8 (new): Sample rates?
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And of course: How to signal the custom modes within the packet?


----- Original Message -----
From: "Koen Vos" <koen.vos@skype.net>
To: "Gregory Maxwell" <gmaxwell@juniper.net>
Cc: Pochol@WebfootGames.com, codec@ietf.org
Sent: Wednesday, January 26, 2011 1:08:51 PM
Subject: Re: [codec] requirements #8 (new): Sample rates?

So what needs to be decided is:

1. How to call these custom modes (for instance in the SDP descriptor).  As=
 long as we agree that it is not "Opus", then we prevent compatibility issu=
es and we keep our on-the-fly switching flexibility within the standard Opu=
s format.  I'm fine with "Opus-custom".

2. How to enable the custom modes in the code.  I think there must be a sma=
ll barrier so that users don't unintentionally start using them.  Maybe a w=
ell-commented #define somewhere, plus an API control flag to put the codec =
in custom mode?

3. Should there be a set of "official" custom modes, to aid interop even wi=
th Opus-custom?  Users could still pick and choose any modes they want, but=
 having the list could make users gravitate towards a few common choices.

koen.



----- Original Message -----
From: "Gregory Maxwell" <gmaxwell@juniper.net>
To: Pochol@WebfootGames.com, "Jean-Marc Valin" <jean-marc.valin@octasic.com=
>
Cc: codec@ietf.org
Sent: Wednesday, January 26, 2011 11:32:01 AM
Subject: Re: [codec] requirements #8 (new): Sample rates?



I'm very concerned that some people may believe that 44.1k is just another =
checkbox that can be added
without cost or much consideration. This isn't the case. I think it's essen=
tial that we delineate realtime VoIP
style usage from other applications.

I support Jean Marc's option (1), which I think allows us to have our cake =
and eat it too. It's the closest thing to a=20
"cost free" option that I think we're going to get. This option basically s=
eparates the codec into two profiles,
one which imposes sampling rate restrictions in exchange for many important=
 advantages, and one which
does not, but misses the advantages. =20

JM's option (2) would also work. But I don't like the idea of the market co=
nfusion created by a totally
separate codec which has significant overlap with Opus, but isn't Opus.  Ba=
sically, I think it's silly
for part of the working group's output to compete with itself.  I'd prefer =
to just have "Opus" and
"Opus-custom" or whatever.


There are several reasons that supporting 44.1kHz in the primary Opus profi=
le would be bad:

One reason for this is that quite a bit of hardware (even on desktops) can =
only do 48kHz (or closely related
rates) and even when the hardware can do multiple rates it can only ever do=
 one rate at a time so if=20
its even possible that multiple applications may play sound at once then th=
e only way to avoid resampling
is if they are all running at a common rate.

For the 48kHz related rates we can do very computationally cheap handling o=
f different rates purely
inside the codec. If their hardware supports any mode out of the 48kHz fami=
ly, then they'll need no costly
resampling at all. And if they do  need run at 44.1k they can resample in a=
nd out of the codec without imposing on (or
negotiating with) the far end.

Another one is that Opus (as described in the draft, without 44.1kHz) can s=
witch between any of
its supported modes, all on the fly, without creating any surprising imposi=
tions on the clients.
This is possible only because of the closely related nature of the supporte=
d rates, and 44.1kHz
can't be accommodated in this scheme.

The on the fly switching and lack of requirement to negotiate, also means t=
hat two opus devices can
 communicate without transcoding, even if they were spliced long after nego=
tiation. (e.g. as part
of a conference gateway).


On the other end of the spectrum=E2=80=94 the current CELT library and bits=
tream is extremely flexible.
It supports a great many frame sizes and sample rates and I've personally a=
rgued against
every limitation we've imposed on it, because I like the idea that CELT can=
 fit into every
niche requirement (like the DAB frame sizes).

But this flexibility has a serious price for interoperable implementations:=
 They must carry substantially
more code (the limited rates/sample sizes means that a simple table can rep=
lace several hundred
lines of tricky bit-exact initialization code), cope with increased peak CP=
U usage (e.g. if some device
only speaks 64 sample frames, 96KHz you might need 10x the CPU power to spe=
ak to it compared=20
to your preferred mode), and undertake more complicated negotiation and tes=
ting (CELT can support
far more unique mixtures of sample rate, frame-size, and channel count then=
 there are RTP payload
types).=20

So basically, I support fully supporting oddball configurations as a well s=
pecified standardized mode,
but I oppose subjecting the general VoIP/RTP users to the increase complexi=
ty and limitations of
the more rate/framesize agile configurations.

I expect that most users which care about the 'custom' modes are doing othe=
r specialized things
and won't be expected to ever interop with a random Opus phone except (mayb=
e) via a gateway,
and that most of them won't even speak RTP=E2=80=94 so the separation shoul=
dn't even make=20
much difference to them at all.


Thoughts?




________________________________________
From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of Pascal P=
ochol [Pochol@WebfootGames.com]
Sent: Wednesday, January 26, 2011 6:06 PM
To: Jean-Marc Valin
Cc: codec@ietf.org
Subject: Re: [codec] requirements #8 (new): Sample rates?

Hello,

I just wanted to give in my 2 cents about 44.1Khz native support.

99% of all the audio we use celt and eventually opus for are encoded at
44.1Khz. They are provided to us that way. Which means that without 44.1khz
support we'll have to up-sample 99% of our audio most likely in a
preprocess build making us maintain 2 sets of audibly identical files. We
had to do it before with speex where we converted from 44.1 down to 32khz
to use its native ultrawideband but it really wasn't the easiest. We had
thousands of files duplicated and every now and then a few of these getting
updated from the source, forcing us to redo massive conversions each time
to make sure we didn't miss one somewhere.

Also about upsampling not costing much, I beg to differ. We had to work
with hardware that could decode speex ultrawideband 32Khz just fine but the
decoding alone was eating up all our CPU leaving not much else to do the
real work that we needed to do. We had to use 16khz instead to make it all
work. 48Khz, 44.1khz might not look like a big difference when working on a
desktop but when you're counting bytes to see how you can reduce you memory
consumption it could mean the world.

So strickly from a user of the codec's view, native 44.1Khz would certainly
make working with celt/opus a lot easier. I'm guessing that I'm not the
only one in that case based on this thread. Easier would also lead to
faster adoption.

Sorry I didn't intend to write that much. In short 44.1Khz: great if you
can do it, if not we'll just have to work around it.

-Pascal

_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec
_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec
_______________________________________________
codec mailing list
codec@ietf.org
https://www.ietf.org/mailman/listinfo/codec

From jean-marc.valin@octasic.com  Wed Jan 26 13:22:00 2011
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Cc: codec@ietf.org, Pochol@WebfootGames.com
Subject: Re: [codec] requirements #8 (new): Sample rates?
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On 11-01-26 04:08 PM, Koen Vos wrote:
> 1. How to call these custom modes (for instance in the SDP descriptor).
> As long as we agree that it is not "Opus", then we prevent compatibility
> issues and we keep our on-the-fly switching flexibility within the
> standard Opus format.  I'm fine with "Opus-custom".

OK, unless there's objections I think we can go with "Opus-custom".

> 2. How to enable the custom modes in the code.  I think there must be a
> small barrier so that users don't unintentionally start using them.
> Maybe a well-commented #define somewhere, plus an API control flag to
> put the codec in custom mode?

Agreed. The API will *have* to be different because in the normal mode 
there won't even be a need to specify things like frame size. As for a 
#define it also makes sense in terms of saving code size by default.

> 3. Should there be a set of "official" custom modes, to aid interop even
> with Opus-custom?  Users could still pick and choose any modes they
> want, but having the list could make users gravitate towards a few
> common choices.

I think the following modes are main custom modes we want to recommend:
1) 48 kHz, frame sizes 128, 256, 512, 1024 (switchable)
2) 44.1 kHz, frame sizes 128, 256, 512, 1024 (switchable)
3) 48 kHz, frame size 64
4) 44.1 kHz frame size 64

though the other modes will still be available.

Cheers,

	Jean-Marc

> koen.
>
>
>
> ----- Original Message ----- From: "Gregory
> Maxwell"<gmaxwell@juniper.net> To: Pochol@WebfootGames.com, "Jean-Marc
> Valin"<jean-marc.valin@octasic.com> Cc: codec@ietf.org Sent: Wednesday,
> January 26, 2011 11:32:01 AM Subject: Re: [codec] requirements #8 (new):
> Sample rates?
>
>
>
> I'm very concerned that some people may believe that 44.1k is just
> another checkbox that can be added without cost or much consideration.
> This isn't the case. I think it's essential that we delineate realtime
> VoIP style usage from other applications.
>
> I support Jean Marc's option (1), which I think allows us to have our
> cake and eat it too. It's the closest thing to a "cost free" option that
> I think we're going to get. This option basically separates the codec
> into two profiles, one which imposes sampling rate restrictions in
> exchange for many important advantages, and one which does not, but
> misses the advantages.
>
> JM's option (2) would also work. But I don't like the idea of the market
> confusion created by a totally separate codec which has significant
> overlap with Opus, but isn't Opus.  Basically, I think it's silly for
> part of the working group's output to compete with itself.  I'd prefer
> to just have "Opus" and "Opus-custom" or whatever.
>
>
> There are several reasons that supporting 44.1kHz in the primary Opus
> profile would be bad:
>
> One reason for this is that quite a bit of hardware (even on desktops)
> can only do 48kHz (or closely related rates) and even when the hardware
> can do multiple rates it can only ever do one rate at a time so if its
> even possible that multiple applications may play sound at once then the
> only way to avoid resampling is if they are all running at a common
> rate.
>
> For the 48kHz related rates we can do very computationally cheap
> handling of different rates purely inside the codec. If their hardware
> supports any mode out of the 48kHz family, then they'll need no costly
> resampling at all. And if they do  need run at 44.1k they can resample
> in and out of the codec without imposing on (or negotiating with) the
> far end.
>
> Another one is that Opus (as described in the draft, without 44.1kHz)
> can switch between any of its supported modes, all on the fly, without
> creating any surprising impositions on the clients. This is possible
> only because of the closely related nature of the supported rates, and
> 44.1kHz can't be accommodated in this scheme.
>
> The on the fly switching and lack of requirement to negotiate, also
> means that two opus devices can communicate without transcoding, even if
> they were spliced long after negotiation. (e.g. as part of a conference
> gateway).
>
>
> On the other end of the spectrumâ€” the current CELT library and
> bitstream is extremely flexible. It supports a great many frame sizes
> and sample rates and I've personally argued against every limitation
> we've imposed on it, because I like the idea that CELT can fit into
> every niche requirement (like the DAB frame sizes).
>
> But this flexibility has a serious price for interoperable
> implementations: They must carry substantially more code (the limited
> rates/sample sizes means that a simple table can replace several
> hundred lines of tricky bit-exact initialization code), cope with
> increased peak CPU usage (e.g. if some device only speaks 64 sample
> frames, 96KHz you might need 10x the CPU power to speak to it compared
> to your preferred mode), and undertake more complicated negotiation and
> testing (CELT can support far more unique mixtures of sample rate,
> frame-size, and channel count then there are RTP payload types).
>
> So basically, I support fully supporting oddball configurations as a
> well specified standardized mode, but I oppose subjecting the general
> VoIP/RTP users to the increase complexity and limitations of the more
> rate/framesize agile configurations.
>
> I expect that most users which care about the 'custom' modes are doing
> other specialized things and won't be expected to ever interop with a
> random Opus phone except (maybe) via a gateway, and that most of them
> won't even speak RTPâ€” so the separation shouldn't even make much
> difference to them at all.
>
>
> Thoughts?
>
>
>
>
> ________________________________________ From: codec-bounces@ietf.org
> [codec-bounces@ietf.org] On Behalf Of Pascal Pochol
> [Pochol@WebfootGames.com] Sent: Wednesday, January 26, 2011 6:06 PM To:
> Jean-Marc Valin Cc: codec@ietf.org Subject: Re: [codec] requirements #8
> (new): Sample rates?
>
> Hello,
>
> I just wanted to give in my 2 cents about 44.1Khz native support.
>
> 99% of all the audio we use celt and eventually opus for are encoded at
> 44.1Khz. They are provided to us that way. Which means that without
> 44.1khz support we'll have to up-sample 99% of our audio most likely in
> a preprocess build making us maintain 2 sets of audibly identical files.
> We had to do it before with speex where we converted from 44.1 down to
> 32khz to use its native ultrawideband but it really wasn't the easiest.
> We had thousands of files duplicated and every now and then a few of
> these getting updated from the source, forcing us to redo massive
> conversions each time to make sure we didn't miss one somewhere.
>
> Also about upsampling not costing much, I beg to differ. We had to work
> with hardware that could decode speex ultrawideband 32Khz just fine but
> the decoding alone was eating up all our CPU leaving not much else to do
> the real work that we needed to do. We had to use 16khz instead to make
> it all work. 48Khz, 44.1khz might not look like a big difference when
> working on a desktop but when you're counting bytes to see how you can
> reduce you memory consumption it could mean the world.
>
> So strickly from a user of the codec's view, native 44.1Khz would
> certainly make working with celt/opus a lot easier. I'm guessing that
> I'm not the only one in that case based on this thread. Easier would
> also lead to faster adoption.
>
> Sorry I didn't intend to write that much. In short 44.1Khz: great if
> you can do it, if not we'll just have to work around it.
>
> -Pascal
>
> _______________________________________________ codec mailing list
> codec@ietf.org https://www.ietf.org/mailman/listinfo/codec
> _______________________________________________ codec mailing list
> codec@ietf.org https://www.ietf.org/mailman/listinfo/codec


From jean-marc.valin@octasic.com  Wed Jan 26 13:27:56 2011
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Cc: codec@ietf.org, Pochol@WebfootGames.com
Subject: Re: [codec] requirements #8 (new): Sample rates?
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On 11-01-26 04:18 PM, Koen Vos wrote:
> And of course: How to signal the custom modes within the packet?

The custom modes will have to be negotiated at the SDP level. Unlike "Opus" 
packets, you will just not be able to decode an "Opus-custom" packet 
without knowing what was negotiated.

     Jean-Marc

>
> ----- Original Message -----
> From: "Koen Vos"<koen.vos@skype.net>
> To: "Gregory Maxwell"<gmaxwell@juniper.net>
> Cc: Pochol@WebfootGames.com, codec@ietf.org
> Sent: Wednesday, January 26, 2011 1:08:51 PM
> Subject: Re: [codec] requirements #8 (new): Sample rates?
>
> So what needs to be decided is:
>
> 1. How to call these custom modes (for instance in the SDP descriptor).  As long as we agree that it is not "Opus", then we prevent compatibility issues and we keep our on-the-fly switching flexibility within the standard Opus format.  I'm fine with "Opus-custom".
>
> 2. How to enable the custom modes in the code.  I think there must be a small barrier so that users don't unintentionally start using them.  Maybe a well-commented #define somewhere, plus an API control flag to put the codec in custom mode?
>
> 3. Should there be a set of "official" custom modes, to aid interop even with Opus-custom?  Users could still pick and choose any modes they want, but having the list could make users gravitate towards a few common choices.
>
> koen.
>
>
>
> ----- Original Message -----
> From: "Gregory Maxwell"<gmaxwell@juniper.net>
> To: Pochol@WebfootGames.com, "Jean-Marc Valin"<jean-marc.valin@octasic.com>
> Cc: codec@ietf.org
> Sent: Wednesday, January 26, 2011 11:32:01 AM
> Subject: Re: [codec] requirements #8 (new): Sample rates?
>
>
>
> I'm very concerned that some people may believe that 44.1k is just another checkbox that can be added
> without cost or much consideration. This isn't the case. I think it's essential that we delineate realtime VoIP
> style usage from other applications.
>
> I support Jean Marc's option (1), which I think allows us to have our cake and eat it too. It's the closest thing to a
> "cost free" option that I think we're going to get. This option basically separates the codec into two profiles,
> one which imposes sampling rate restrictions in exchange for many important advantages, and one which
> does not, but misses the advantages.
>
> JM's option (2) would also work. But I don't like the idea of the market confusion created by a totally
> separate codec which has significant overlap with Opus, but isn't Opus.  Basically, I think it's silly
> for part of the working group's output to compete with itself.  I'd prefer to just have "Opus" and
> "Opus-custom" or whatever.
>
>
> There are several reasons that supporting 44.1kHz in the primary Opus profile would be bad:
>
> One reason for this is that quite a bit of hardware (even on desktops) can only do 48kHz (or closely related
> rates) and even when the hardware can do multiple rates it can only ever do one rate at a time so if
> its even possible that multiple applications may play sound at once then the only way to avoid resampling
> is if they are all running at a common rate.
>
> For the 48kHz related rates we can do very computationally cheap handling of different rates purely
> inside the codec. If their hardware supports any mode out of the 48kHz family, then they'll need no costly
> resampling at all. And if they do  need run at 44.1k they can resample in and out of the codec without imposing on (or
> negotiating with) the far end.
>
> Another one is that Opus (as described in the draft, without 44.1kHz) can switch between any of
> its supported modes, all on the fly, without creating any surprising impositions on the clients.
> This is possible only because of the closely related nature of the supported rates, and 44.1kHz
> can't be accommodated in this scheme.
>
> The on the fly switching and lack of requirement to negotiate, also means that two opus devices can
>   communicate without transcoding, even if they were spliced long after negotiation. (e.g. as part
> of a conference gateway).
>
>
> On the other end of the spectrumâ€” the current CELT library and bitstream is extremely flexible.
> It supports a great many frame sizes and sample rates and I've personally argued against
> every limitation we've imposed on it, because I like the idea that CELT can fit into every
> niche requirement (like the DAB frame sizes).
>
> But this flexibility has a serious price for interoperable implementations: They must carry substantially
> more code (the limited rates/sample sizes means that a simple table can replace several hundred
> lines of tricky bit-exact initialization code), cope with increased peak CPU usage (e.g. if some device
> only speaks 64 sample frames, 96KHz you might need 10x the CPU power to speak to it compared
> to your preferred mode), and undertake more complicated negotiation and testing (CELT can support
> far more unique mixtures of sample rate, frame-size, and channel count then there are RTP payload
> types).
>
> So basically, I support fully supporting oddball configurations as a well specified standardized mode,
> but I oppose subjecting the general VoIP/RTP users to the increase complexity and limitations of
> the more rate/framesize agile configurations.
>
> I expect that most users which care about the 'custom' modes are doing other specialized things
> and won't be expected to ever interop with a random Opus phone except (maybe) via a gateway,
> and that most of them won't even speak RTPâ€” so the separation shouldn't even make
> much difference to them at all.
>
>
> Thoughts?
>
>
>
>
> ________________________________________
> From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of Pascal Pochol [Pochol@WebfootGames.com]
> Sent: Wednesday, January 26, 2011 6:06 PM
> To: Jean-Marc Valin
> Cc: codec@ietf.org
> Subject: Re: [codec] requirements #8 (new): Sample rates?
>
> Hello,
>
> I just wanted to give in my 2 cents about 44.1Khz native support.
>
> 99% of all the audio we use celt and eventually opus for are encoded at
> 44.1Khz. They are provided to us that way. Which means that without 44.1khz
> support we'll have to up-sample 99% of our audio most likely in a
> preprocess build making us maintain 2 sets of audibly identical files. We
> had to do it before with speex where we converted from 44.1 down to 32khz
> to use its native ultrawideband but it really wasn't the easiest. We had
> thousands of files duplicated and every now and then a few of these getting
> updated from the source, forcing us to redo massive conversions each time
> to make sure we didn't miss one somewhere.
>
> Also about upsampling not costing much, I beg to differ. We had to work
> with hardware that could decode speex ultrawideband 32Khz just fine but the
> decoding alone was eating up all our CPU leaving not much else to do the
> real work that we needed to do. We had to use 16khz instead to make it all
> work. 48Khz, 44.1khz might not look like a big difference when working on a
> desktop but when you're counting bytes to see how you can reduce you memory
> consumption it could mean the world.
>
> So strickly from a user of the codec's view, native 44.1Khz would certainly
> make working with celt/opus a lot easier. I'm guessing that I'm not the
> only one in that case based on this thread. Easier would also lead to
> faster adoption.
>
> Sorry I didn't intend to write that much. In short 44.1Khz: great if you
> can do it, if not we'll just have to work around it.
>
> -Pascal
>
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec
> _______________________________________________
> codec mailing list
> codec@ietf.org
> https://www.ietf.org/mailman/listinfo/codec


From koen.vos@skype.net  Wed Jan 26 16:04:16 2011
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From: Koen Vos <koen.vos@skype.net>
To: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Cc: codec@ietf.org, Pochol@WebfootGames.com
Subject: Re: [codec] requirements #8 (new): Sample rates?
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In your previous email you mentioned two custom modes with switchable frame=
 sizes.  For those modes at least the frame size will have to be signaled i=
n each packet, no?

I could see a benefit in making all recommended custom modes "instantly" de=
codable (meaning without prior knowledge).  Although it does add another la=
yer of "standard custom" vs "custom custom" which may confuse people.

I'm fine either way.
koen.


----- Original Message -----
From: "Jean-Marc Valin" <jean-marc.valin@octasic.com>
To: "Koen Vos" <koen.vos@skype.net>
Cc: "Gregory Maxwell" <gmaxwell@juniper.net>, codec@ietf.org, Pochol@Webfoo=
tGames.com
Sent: Wednesday, January 26, 2011 1:27:01 PM
Subject: Re: [codec] requirements #8 (new): Sample rates?

On 11-01-26 04:18 PM, Koen Vos wrote:
> And of course: How to signal the custom modes within the packet?

The custom modes will have to be negotiated at the SDP level. Unlike "Opus"=
=20
packets, you will just not be able to decode an "Opus-custom" packet=20
without knowing what was negotiated.

     Jean-Marc

>
> ----- Original Message -----
> From: "Koen Vos"<koen.vos@skype.net>
> To: "Gregory Maxwell"<gmaxwell@juniper.net>
> Cc: Pochol@WebfootGames.com, codec@ietf.org
> Sent: Wednesday, January 26, 2011 1:08:51 PM
> Subject: Re: [codec] requirements #8 (new): Sample rates?
>
> So what needs to be decided is:
>
> 1. How to call these custom modes (for instance in the SDP descriptor).  =
As long as we agree that it is not "Opus", then we prevent compatibility is=
sues and we keep our on-the-fly switching flexibility within the standard O=
pus format.  I'm fine with "Opus-custom".
>
> 2. How to enable the custom modes in the code.  I think there must be a s=
mall barrier so that users don't unintentionally start using them.  Maybe a=
 well-commented #define somewhere, plus an API control flag to put the code=
c in custom mode?
>
> 3. Should there be a set of "official" custom modes, to aid interop even =
with Opus-custom?  Users could still pick and choose any modes they want, b=
ut having the list could make users gravitate towards a few common choices.
>
> koen.
>
>
>
> ----- Original Message -----
> From: "Gregory Maxwell"<gmaxwell@juniper.net>
> To: Pochol@WebfootGames.com, "Jean-Marc Valin"<jean-marc.valin@octasic.co=
m>
> Cc: codec@ietf.org
> Sent: Wednesday, January 26, 2011 11:32:01 AM
> Subject: Re: [codec] requirements #8 (new): Sample rates?
>
>
>
> I'm very concerned that some people may believe that 44.1k is just anothe=
r checkbox that can be added
> without cost or much consideration. This isn't the case. I think it's ess=
ential that we delineate realtime VoIP
> style usage from other applications.
>
> I support Jean Marc's option (1), which I think allows us to have our cak=
e and eat it too. It's the closest thing to a
> "cost free" option that I think we're going to get. This option basically=
 separates the codec into two profiles,
> one which imposes sampling rate restrictions in exchange for many importa=
nt advantages, and one which
> does not, but misses the advantages.
>
> JM's option (2) would also work. But I don't like the idea of the market =
confusion created by a totally
> separate codec which has significant overlap with Opus, but isn't Opus.  =
Basically, I think it's silly
> for part of the working group's output to compete with itself.  I'd prefe=
r to just have "Opus" and
> "Opus-custom" or whatever.
>
>
> There are several reasons that supporting 44.1kHz in the primary Opus pro=
file would be bad:
>
> One reason for this is that quite a bit of hardware (even on desktops) ca=
n only do 48kHz (or closely related
> rates) and even when the hardware can do multiple rates it can only ever =
do one rate at a time so if
> its even possible that multiple applications may play sound at once then =
the only way to avoid resampling
> is if they are all running at a common rate.
>
> For the 48kHz related rates we can do very computationally cheap handling=
 of different rates purely
> inside the codec. If their hardware supports any mode out of the 48kHz fa=
mily, then they'll need no costly
> resampling at all. And if they do  need run at 44.1k they can resample in=
 and out of the codec without imposing on (or
> negotiating with) the far end.
>
> Another one is that Opus (as described in the draft, without 44.1kHz) can=
 switch between any of
> its supported modes, all on the fly, without creating any surprising impo=
sitions on the clients.
> This is possible only because of the closely related nature of the suppor=
ted rates, and 44.1kHz
> can't be accommodated in this scheme.
>
> The on the fly switching and lack of requirement to negotiate, also means=
 that two opus devices can
>   communicate without transcoding, even if they were spliced long after n=
egotiation. (e.g. as part
> of a conference gateway).
>
>
> On the other end of the spectrum=C3=A2=E2=82=AC=E2=80=9D the current CELT=
 library and bitstream is extremely flexible.
> It supports a great many frame sizes and sample rates and I've personally=
 argued against
> every limitation we've imposed on it, because I like the idea that CELT c=
an fit into every
> niche requirement (like the DAB frame sizes).
>
> But this flexibility has a serious price for interoperable implementation=
s: They must carry substantially
> more code (the limited rates/sample sizes means that a simple table can r=
eplace several hundred
> lines of tricky bit-exact initialization code), cope with increased peak =
CPU usage (e.g. if some device
> only speaks 64 sample frames, 96KHz you might need 10x the CPU power to s=
peak to it compared
> to your preferred mode), and undertake more complicated negotiation and t=
esting (CELT can support
> far more unique mixtures of sample rate, frame-size, and channel count th=
en there are RTP payload
> types).
>
> So basically, I support fully supporting oddball configurations as a well=
 specified standardized mode,
> but I oppose subjecting the general VoIP/RTP users to the increase comple=
xity and limitations of
> the more rate/framesize agile configurations.
>
> I expect that most users which care about the 'custom' modes are doing ot=
her specialized things
> and won't be expected to ever interop with a random Opus phone except (ma=
ybe) via a gateway,
> and that most of them won't even speak RTP=C3=A2=E2=82=AC=E2=80=9D so the=
 separation shouldn't even make
> much difference to them at all.
>
>
> Thoughts?
>
>
>
>
> ________________________________________
> From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of Pascal=
 Pochol [Pochol@WebfootGames.com]
> Sent: Wednesday, January 26, 2011 6:06 PM
> To: Jean-Marc Valin
> Cc: codec@ietf.org
> Subject: Re: [codec] requirements #8 (new): Sample rates?
>
> Hello,
>
> I just wanted to give in my 2 cents about 44.1Khz native support.
>
> 99% of all the audio we use celt and eventually opus for are encoded at
> 44.1Khz. They are provided to us that way. Which means that without 44.1k=
hz
> support we'll have to up-sample 99% of our audio most likely in a
> preprocess build making us maintain 2 sets of audibly identical files. We
> had to do it before with speex where we converted from 44.1 down to 32khz
> to use its native ultrawideband but it really wasn't the easiest. We had
> thousands of files duplicated and every now and then a few of these getti=
ng
> updated from the source, forcing us to redo massive conversions each time
> to make sure we didn't miss one somewhere.
>
> Also about upsampling not costing much, I beg to differ. We had to work
> with hardware that could decode speex ultrawideband 32Khz just fine but t=
he
> decoding alone was eating up all our CPU leaving not much else to do the
> real work that we needed to do. We had to use 16khz instead to make it al=
l
> work. 48Khz, 44.1khz might not look like a big difference when working on=
 a
> desktop but when you're counting bytes to see how you can reduce you memo=
ry
> consumption it could mean the world.
>
> So strickly from a user of the codec's view, native 44.1Khz would certain=
ly
> make working with celt/opus a lot easier. I'm guessing that I'm not the
> only one in that case based on this thread. Easier would also lead to
> faster adoption.
>
> Sorry I didn't intend to write that much. In short 44.1Khz: great if you
> can do it, if not we'll just have to work around it.
>
> -Pascal
>
> _______________________________________________
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From jean-marc.valin@octasic.com  Wed Jan 26 17:12:06 2011
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Date: Wed, 26 Jan 2011 20:14:55 -0500
From: Jean-Marc Valin <jean-marc.valin@octasic.com>
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Subject: Re: [codec] requirements #8 (new): Sample rates?
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What I meant in my previous email was that these modes are potentially 
switchable. How exactly (or whether) we want to handle that is still an 
open question. It even something we could handle in the RTP payload 
format. Overall, I don't have a strong opinion on switching of custom modes.

	Jean-Marc

On 11-01-26 07:07 PM, Koen Vos wrote:
> In your previous email you mentioned two custom modes with switchable frame sizes.  For those modes at least the frame size will have to be signaled in each packet, no?
>
> I could see a benefit in making all recommended custom modes "instantly" decodable (meaning without prior knowledge).  Although it does add another layer of "standard custom" vs "custom custom" which may confuse people.
>
> I'm fine either way.
> koen.
>
>
> ----- Original Message -----
> From: "Jean-Marc Valin"<jean-marc.valin@octasic.com>
> To: "Koen Vos"<koen.vos@skype.net>
> Cc: "Gregory Maxwell"<gmaxwell@juniper.net>, codec@ietf.org, Pochol@WebfootGames.com
> Sent: Wednesday, January 26, 2011 1:27:01 PM
> Subject: Re: [codec] requirements #8 (new): Sample rates?
>
> On 11-01-26 04:18 PM, Koen Vos wrote:
>> And of course: How to signal the custom modes within the packet?
>
> The custom modes will have to be negotiated at the SDP level. Unlike "Opus"
> packets, you will just not be able to decode an "Opus-custom" packet
> without knowing what was negotiated.
>
>       Jean-Marc
>
>>
>> ----- Original Message -----
>> From: "Koen Vos"<koen.vos@skype.net>
>> To: "Gregory Maxwell"<gmaxwell@juniper.net>
>> Cc: Pochol@WebfootGames.com, codec@ietf.org
>> Sent: Wednesday, January 26, 2011 1:08:51 PM
>> Subject: Re: [codec] requirements #8 (new): Sample rates?
>>
>> So what needs to be decided is:
>>
>> 1. How to call these custom modes (for instance in the SDP descriptor).  As long as we agree that it is not "Opus", then we prevent compatibility issues and we keep our on-the-fly switching flexibility within the standard Opus format.  I'm fine with "Opus-custom".
>>
>> 2. How to enable the custom modes in the code.  I think there must be a small barrier so that users don't unintentionally start using them.  Maybe a well-commented #define somewhere, plus an API control flag to put the codec in custom mode?
>>
>> 3. Should there be a set of "official" custom modes, to aid interop even with Opus-custom?  Users could still pick and choose any modes they want, but having the list could make users gravitate towards a few common choices.
>>
>> koen.
>>
>>
>>
>> ----- Original Message -----
>> From: "Gregory Maxwell"<gmaxwell@juniper.net>
>> To: Pochol@WebfootGames.com, "Jean-Marc Valin"<jean-marc.valin@octasic.com>
>> Cc: codec@ietf.org
>> Sent: Wednesday, January 26, 2011 11:32:01 AM
>> Subject: Re: [codec] requirements #8 (new): Sample rates?
>>
>>
>>
>> I'm very concerned that some people may believe that 44.1k is just another checkbox that can be added
>> without cost or much consideration. This isn't the case. I think it's essential that we delineate realtime VoIP
>> style usage from other applications.
>>
>> I support Jean Marc's option (1), which I think allows us to have our cake and eat it too. It's the closest thing to a
>> "cost free" option that I think we're going to get. This option basically separates the codec into two profiles,
>> one which imposes sampling rate restrictions in exchange for many important advantages, and one which
>> does not, but misses the advantages.
>>
>> JM's option (2) would also work. But I don't like the idea of the market confusion created by a totally
>> separate codec which has significant overlap with Opus, but isn't Opus.  Basically, I think it's silly
>> for part of the working group's output to compete with itself.  I'd prefer to just have "Opus" and
>> "Opus-custom" or whatever.
>>
>>
>> There are several reasons that supporting 44.1kHz in the primary Opus profile would be bad:
>>
>> One reason for this is that quite a bit of hardware (even on desktops) can only do 48kHz (or closely related
>> rates) and even when the hardware can do multiple rates it can only ever do one rate at a time so if
>> its even possible that multiple applications may play sound at once then the only way to avoid resampling
>> is if they are all running at a common rate.
>>
>> For the 48kHz related rates we can do very computationally cheap handling of different rates purely
>> inside the codec. If their hardware supports any mode out of the 48kHz family, then they'll need no costly
>> resampling at all. And if they do  need run at 44.1k they can resample in and out of the codec without imposing on (or
>> negotiating with) the far end.
>>
>> Another one is that Opus (as described in the draft, without 44.1kHz) can switch between any of
>> its supported modes, all on the fly, without creating any surprising impositions on the clients.
>> This is possible only because of the closely related nature of the supported rates, and 44.1kHz
>> can't be accommodated in this scheme.
>>
>> The on the fly switching and lack of requirement to negotiate, also means that two opus devices can
>>    communicate without transcoding, even if they were spliced long after negotiation. (e.g. as part
>> of a conference gateway).
>>
>>
>> On the other end of the spectrumâ€” the current CELT library and bitstream is extremely flexible.
>> It supports a great many frame sizes and sample rates and I've personally argued against
>> every limitation we've imposed on it, because I like the idea that CELT can fit into every
>> niche requirement (like the DAB frame sizes).
>>
>> But this flexibility has a serious price for interoperable implementations: They must carry substantially
>> more code (the limited rates/sample sizes means that a simple table can replace several hundred
>> lines of tricky bit-exact initialization code), cope with increased peak CPU usage (e.g. if some device
>> only speaks 64 sample frames, 96KHz you might need 10x the CPU power to speak to it compared
>> to your preferred mode), and undertake more complicated negotiation and testing (CELT can support
>> far more unique mixtures of sample rate, frame-size, and channel count then there are RTP payload
>> types).
>>
>> So basically, I support fully supporting oddball configurations as a well specified standardized mode,
>> but I oppose subjecting the general VoIP/RTP users to the increase complexity and limitations of
>> the more rate/framesize agile configurations.
>>
>> I expect that most users which care about the 'custom' modes are doing other specialized things
>> and won't be expected to ever interop with a random Opus phone except (maybe) via a gateway,
>> and that most of them won't even speak RTPâ€” so the separation shouldn't even make
>> much difference to them at all.
>>
>>
>> Thoughts?
>>
>>
>>
>>
>> ________________________________________
>> From: codec-bounces@ietf.org [codec-bounces@ietf.org] On Behalf Of Pascal Pochol [Pochol@WebfootGames.com]
>> Sent: Wednesday, January 26, 2011 6:06 PM
>> To: Jean-Marc Valin
>> Cc: codec@ietf.org
>> Subject: Re: [codec] requirements #8 (new): Sample rates?
>>
>> Hello,
>>
>> I just wanted to give in my 2 cents about 44.1Khz native support.
>>
>> 99% of all the audio we use celt and eventually opus for are encoded at
>> 44.1Khz. They are provided to us that way. Which means that without 44.1khz
>> support we'll have to up-sample 99% of our audio most likely in a
>> preprocess build making us maintain 2 sets of audibly identical files. We
>> had to do it before with speex where we converted from 44.1 down to 32khz
>> to use its native ultrawideband but it really wasn't the easiest. We had
>> thousands of files duplicated and every now and then a few of these getting
>> updated from the source, forcing us to redo massive conversions each time
>> to make sure we didn't miss one somewhere.
>>
>> Also about upsampling not costing much, I beg to differ. We had to work
>> with hardware that could decode speex ultrawideband 32Khz just fine but the
>> decoding alone was eating up all our CPU leaving not much else to do the
>> real work that we needed to do. We had to use 16khz instead to make it all
>> work. 48Khz, 44.1khz might not look like a big difference when working on a
>> desktop but when you're counting bytes to see how you can reduce you memory
>> consumption it could mean the world.
>>
>> So strickly from a user of the codec's view, native 44.1Khz would certainly
>> make working with celt/opus a lot easier. I'm guessing that I'm not the
>> only one in that case based on this thread. Easier would also lead to
>> faster adoption.
>>
>> Sorry I didn't intend to write that much. In short 44.1Khz: great if you
>> can do it, if not we'll just have to work around it.
>>
>> -Pascal
>>
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
>> _______________________________________________
>> codec mailing list
>> codec@ietf.org
>> https://www.ietf.org/mailman/listinfo/codec
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>> https://www.ietf.org/mailman/listinfo/codec
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>> https://www.ietf.org/mailman/listinfo/codec
>
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From Pochol@WebfootGames.com  Wed Jan 26 19:08:25 2011
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Jean-Marc,

> > And of course: How to signal the custom modes within the packet?
> 
> The custom modes will have to be negotiated at the SDP level. Unlike
> "Opus" packets, you will just not be able to decode an "Opus-custom"
> packet without knowing what was negotiated.

in our case I don't even think that we need a per packet signal to tell the
decoder. Unless the Opus-custom encoder does something that varies per
packet we usually set the parameters up front and never change them when
encoding a stream. We could have a little header packet to set things up or
since we're in total control simply set the parameter in the initialization
of the decoder with a few api calls. Not sure if that would work for others
but in our case I think it would without adding extra information to the
packets.

-Pascal


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Subject: Re: [codec] requirements #8 (new): Sample rates?
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On Thu, Jan 27, 2011 at 4:12 AM, Pascal Pochol <Pochol@webfootgames.com> wrote:
> Jean-Marc,
>
>> > And of course: How to signal the custom modes within the packet?
>>
>> The custom modes will have to be negotiated at the SDP level. Unlike
>> "Opus" packets, you will just not be able to decode an "Opus-custom"
>> packet without knowing what was negotiated.
>
> in our case I don't even think that we need a per packet signal to tell the
> decoder. Unless the Opus-custom encoder does something that varies per
> packet we usually set the parameters up front and never change them when
> encoding a stream. We could have a little header packet to set things up or
> since we're in total control simply set the parameter in the initialization
> of the decoder with a few api calls. Not sure if that would work for others
> but in our case I think it would without adding extra information to the
> packets.
>

Well, but you probably need at least one bit to signal header/data packets, no?

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Cc: codec@ietf.org
Subject: Re: [codec] requirements #8 (new): Sample rates?
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Brutus,

> Well, but you probably need at least one bit to signal header/data
> packets, no?

for what we do, no. We open a stream we can easily assume the header is at
the beginning and everything after that is data packets. We don't have
anything that can join in midstream or change the format midstream or
anything like that. Again this is very specific to what we do it might not
fit others needs. I wouldn't even have a problem if we simply had to call a
few api functions before encoding to set everything up and call a few api
functions before decoding with the same type of parameters to let the
decoder know what it's about to receive and not even use a header. Since we
write both encoder and decoder programs to fit our needs and nobody else
reads our encoded data.

when we used speex we started with the standard speexenc/speexdec format
encapsulated into an ogg. But that was too complex so we dropped the ogg
container and used samplexenc.c and sampledec.c instead. We made our
version of sampleenc.c spit out a tiny header at the beginning that would
give us some basic information that we needed in order to decode
everything. I don't remember off hand what it contained one was the
original number of samples I think. And we loaded that in our decoder and
that was it. the rest of the files were all the smallest speex packets we
could make. We noticed that the first byte was a constant size of the frame
so we dropped that out as well and moved it into the header. That saved a
few hundreds to a few thousands bytes for each file (a few thousands of
files total) for a huge saving. For us a byte is still a full 8 bits which
can be used to store a lot of information, so I perfectly understand and
appreciate this group's concern about adding a few bits to each packets. My
personal vote for opus-custom would be to use api calls and not store any
more data than absolutely necessary in the stream. Since using the custom
mode mean using custom code on both ends it's not a problem for us.

-Pascal


From gmaxwell@juniper.net  Thu Jan 27 10:27:33 2011
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From: Gregory Maxwell <gmaxwell@juniper.net>
To: Butrus Damaskus <butrus.butrus@gmail.com>, "Pochol@webfootgames.com" <Pochol@webfootgames.com>
Date: Thu, 27 Jan 2011 10:21:06 -0800
Thread-Topic: [codec] requirements #8 (new): Sample rates?
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Cc: "codec@ietf.org" <codec@ietf.org>
Subject: Re: [codec] requirements #8 (new): Sample rates?
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Butrus Damaskus [butrus.butrus@gmail.com] wrote:
> On Thu, Jan 27, 2011 at 4:12 AM, Pascal Pochol <Pochol@webfootgames.com> =
wrote:
>> in our case I don't even think that we need a per packet signal to tell =
the
>> decoder. Unless the Opus-custom encoder does something that varies per
>> packet we usually set the parameters up front and never change them when
>> encoding a stream. We could have a little header packet to set things up=
 or
>> since we're in total control simply set the parameter in the initializat=
ion
>> of the decoder with a few api calls. Not sure if that would work for oth=
ers
>> but in our case I think it would without adding extra information to the
>> packets.
>
>Well, but you probably need at least one bit to signal header/data packets=
, no?

No=97 the codec itself doesn't carry quite a bit of information which is im=
portant
to applications already. E.g. it doesn't carry timing.  These things are le=
ft
to upper level protocols to handle.

The same can, and should, be done for configuration negotiation. The reason=
 for
this is that the transport can be well adapted to the environment.   If the=
re were
a 'header packet' that the codec generated only once and sent as part of
the bitstream, if that packet were lost the all the rest of the audio would=
 be
undecodable.   So the only real options are: send it every packet or leave =
it
for another mechanism to handle.

Even if you wanted to reinvent a whole two way negotiation machine inside t=
he
codec=97 you couldn't really, because you can't depend on the usage being
duplex.

For modes which can be reasonably switched on the fly sending it in every
packet provides some value.  For ones which can't be=97 sending it in every
packet is overhead, and worse=97 a possible opening for a malicious
transmitter to trigger poorly tested corner-cases.

A complete selection of modes that CELT can operate in (when restrictions t=
o
simplify implementation are not imposed) takes about 4 bytes, more if you=20
want to add some length coding in order to support packing multiple codec
frames in one packet.   Thats a pretty high overhead for a 64 sample/frame=
=20
mode which is sending 700pps, especially in the case of a lossless
transport for which the configuration would only ever have to be sent once.

So, I think that in packet signaling should generally be limited only
to signalling switches that the codec can actually do on the fly, and
I think for the opus-custom modes it's probably better to not signal in
the packets at all.  If an application needs switching for opus-custom,
it could define a new encapsulation for its transport which provides for it=
.






From hoene@uni-tuebingen.de  Fri Jan 28 02:32:18 2011
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From: "Christian Hoene" <hoene@uni-tuebingen.de>
To: <codec@ietf.org>
Date: Fri, 28 Jan 2011 11:35:19 +0100
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Cc: catherine.quinquis@orange-ftgroup.com, Paolo.Usai@etsi.org
Subject: [codec] Handbook of subjective testing practical procedures
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Hi,

for those how are interested in codec testing, I would like to bring the
attention to newly finished standard draft TD 477 of ITU-T study group =
12:
http://www.itu.int/md/T09-SG12-110118-TD-GEN-0477/en . It will be a =
standard
in about three months' time and then available to the public. Currently,
only people with TIES account can access it.
Thanks to Paolo Usai and Catherine Quinquis, who have written this =
excellent
handbook.

With best regards,
 Christian Hoene

---------------------------------------------------------------
Dr.-Ing. Christian Hoene
Interactive Communication Systems (ICS), Universit=E4t T=FCbingen=20
Sand 13, 72076 T=FCbingen, Germany, Phone +49 7071 2970532=20
http://www.net.uni-tuebingen.de/


