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Versions: (draft-ietf-codec-harmony) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 RFC 6716

Network Working Group                                          JM. Valin
Internet-Draft                                              Octasic Inc.
Intended status: Standards Track                                  K. Vos
Expires: April 18, 2011                          Skype Technologies S.A.
                                                        October 15, 2010


                   Definition of the Opus Audio Codec
                        draft-ietf-codec-opus-00

Abstract

   This document describes the Opus codec, designed for interactive
   speech and audio transmission over the Internet.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 18, 2011.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.





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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Opus Codec . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Source Code  . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Codec Modes  . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  Examples . . . . . . . . . . . . . . . . . . . . . . . . .  6
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . .  8
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  9
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 10
   7.  Informative References . . . . . . . . . . . . . . . . . . . . 11
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12







































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1.  Introduction

   We propose the Opus codec based on a linear prediction layer (LP) and
   an MDCT-based enhancement layer.  The main idea behind the proposal
   is that the speech low frequencies are usually more efficiently coded
   using linear prediction codecs (such as CELP variants), while the
   higher frequencies are more efficiently coded in the transform domain
   (e.g.  MDCT).  For low sampling rates, the MDCT layer is not useful
   and only the LP-based layer is used.  On the other hand, non-speech
   signals are not always adequately coded using linear prediction, so
   for music only the MDCT-based layer is used.

   In this proposed prototype, the LP layer is based on the SILK [1]
   codec [SILK] and the MDCT layer is based on the CELT [2] codec
   [CELT].

   This is a work in progress.


































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2.  Opus Codec

   In hybrid mode, each frame is coded first by the LP layer and then by
   the MDCT layer.  In the current prototype, the cutoff frequency is 8
   kHz.  In the MDCT layer, all bands below 8 kHz are discarded, such
   that there is no coding redundancy between the two layers.  Also both
   layers use the same instance of the range coder to encode the signal,
   which ensures that no "padding bits" are wasted.  The hybrid approach
   makes it easy to support both constant bit-rate (CBR) and varaible
   bit-rate (VBR) coding.  Although the SILK layer used is VBR, it is
   easy to make the bit allocation of the CELT layer produce a final
   stream that is CBR by using all the bits left unused by the SILK
   layer.

   The implementation of SILK-based LP layer is similar to the
   description in the SILK Internet-Draft [SILK] with the main exception
   that SILK was modified to use the same range coder as CELT.  The
   implementation of the CELT-based MDCT layer is available from the
   CELT website and is a more recent version (0.8.1) of the CELT
   Internet-Draft [CELT].  The main changes include better support for
   20 ms frames as well as the ability to encode only the higher bands
   using a range coder partially filled by the SILK layer.

   In addition to their frame size, the SILK and CELT codecs require a
   look-ahead of 5.2 ms and 2.5 ms, respectively.  SILK's look-ahead is
   due to noise shaping estimation (5 ms) and the internal resampling
   (0.2 ms), while CELT's look-ahead is due to the overlapping MDCT
   windows.  To compensate for the difference, the CELT encoder input is
   delayed by 2.7 ms.  This ensures that low frequencies and high
   frequencies arrive at the same time.

2.1.  Source Code

   The source code is currently available in a Git repository [3] which
   references two other repositories (for SILK and CELT).  Some
   snapshots are provided for convenience at
   <http://people.xiph.org/~jm/ietfcodec/> along with sample files.
   Although the build system is very primitive, some instructions are
   provided in the toplevel README file.  This is very early development
   so both the quality and feature set should greatly improve over time.
   In the current version, only 48 kHz audio is supported, but support
   for all configurations listed in Section 3 is planned.









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3.  Codec Modes

   There are three possible operating modes for the proposed prototype:

   1.  A linear prediction (LP) mode for use in low bit-rate connections
       with up to 8 kHz audio bandwidth (16 kHz sampling rate)

   2.  A hybrid (LP+MDCT) mode for full-bandwidth speech at medium
       bitrates

   3.  An MDCT-only mode for very low delay speech transmission as well
       as music transmission.

   Each of these modes supports a number of difference frame sizes and
   sampling rates.  In order to distinguish between the various modes
   and configurations, we need to define a simple header that can used
   in the transport layer (e.g RTP) to signal this information.  The
   following describes the proposed header.

   The LP mode supports the following configurations (numbered from
   00000...01011 in binary):

   o  8 kHz: 10, 20, 40, 60 ms (00000...00011)

   o  12 kHz: 10, 20, 40, 60 ms (00100...00111)

   o  16 kHz: 10, 20, 40, 60 ms (01000...01011)

   for a total of 12 configurations.

   The hybrid mode supports the following configurations (numbered from
   01100...01111):

   o  32 kHz: 10, 20 ms (01100...01101)

   o  48 kHz: 10, 20 ms (01110...01111)

   for a total of 4 configurations.

   The MDCT-only mode supports the following configurations (numbered
   from 10000...11101):

   o  8 kHz: 2.5, 5, 10, 20 ms (10000...10011)

   o  16 kHz: 2.5, 5, 10, 20 ms (10100...10111)

   o  32 kHz: 2.5, 5, 10, 20 ms (11000...11011)




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   o  48 kHz: 2.5, 5, 10, 20 ms (11100...11111)

   for a total of 16 configurations.

   There is thus a total of 32 configurations, so 5 bits are necessary
   to indicate the mode, frame size and sampling rate (MFS).  This
   leaves 3 bits for the number of frames per packets (codes 0 to 7):

   o  0-2: 1-3 frames in the packet, each with equal compressed size

   o  3: arbitrary number of frames in the packet, each with equal
      compressed size (one size needs to be encoded)

   o  4-5: 2-3 frames in the packet, with different compressed sizes,
      which need to be encoded (except the last one)

   o  6: arbitrary number of frames in the packet, with different
      compressed sizes, each of which needs to be encoded

   o  7: The first frame has this MFS, but others have different MFS.
      Each compressed size needs to be encoded.

   When code 7 is used and the last frames of a packet have the same
   MFS, it is allowed to switch to another code for them.

   The compressed size of the frames (if needed) is indicated -- usually
   -- with one byte, with the following meaning:

   o  0: No frame (DTX or lost packet)

   o  1-251: Size of the frame in bytes

   o  252-255: A second byte is needed.  The total size is (size[1]*4)+
      (size[0]%4)+252

   The maximum size representable is 255*4+3+252=1275 bytes.  For 20 ms
   frames, that represents a bit-rate of 510 kb/s, which is really the
   highest rate anyone would want to use in stereo mode (beyond that
   point, lossless codecs would be more appropriate).

3.1.  Examples

   Simplest case: one packet








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    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   MFS   |0|0|0|               compressed data...              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Four frames of the same compressed size:


    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   MFS   |0|1|1|               compressed data...              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Two frames of different compressed size:


    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   MFS   |1|0|1|   frame size  |        compressed data...     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Three frames of different _durations_:


    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | 1st MFS |1|1|1|   frame size  | 2nd MFS |1|1|1|   frame size  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | 3rd MFS |1|1|1|   frame size  |      compressed data...       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

















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4.  Security Considerations

   The codec needs to take appropriate security considerations into
   account, as outlined in [DOS] and [SECGUIDE].  It is extremely
   important for the decoder to be robust against malicious payloads.
   Malicious payloads must not cause the decoder to overrun its
   allocated memory or to take much more resources to decode.  Although
   problems in encoders are typically rarer, the same applies to the
   encoder.  Malicious audio stream must not cause the encoder to
   misbehave because this would allow an attacker to attack transcoding
   gateways.

   In its current version, the Opus codec likely does NOT meet these
   security considerations, so it should be used with caution.





































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5.  IANA Considerations

   This document has no actions for IANA.
















































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6.  Acknowledgments

   Thanks to all other developers, including Soeren Skak Jensen, Gregory
   Maxwell, Christopher Montgomery, Karsten Vandborg Soerensen, and
   Timothy Terriberry.














































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7.  Informative References

   [SILK]     Vos, K., Jensen, S., and K. Soerensen, "SILK Speech
              Codec", draft-vos-silk-01 (work in progress), March 2010.

   [CELT]     Valin, J-M., Terriberry, T., Maxwell, G., and C.
              Montgomery, "Constrained-Energy Lapped Transform (CELT)
              Codec", draft-valin-celt-codec-02 (work in progress),
              July 2010.

   [DOS]      Handley, M., Rescorla, E., and IAB, "Internet Denial-of-
              Service Considerations", RFC 4732, December 2006.

   [SECGUIDE]
              Rescorla, E. and B. Korver, "Guidelines for Writing RFC
              Text on Security Considerations", BCP 72, RFC 3552,
              July 2003.

   [1]  <http://developer.skype.com/silk>

   [2]  <http://www.celt-codec.org/>

   [3]  <git://git.xiph.org/users/jm/ietfcodec.git>




























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Authors' Addresses

   Jean-Marc Valin
   Octasic Inc.
   4101, Molson Street
   Montreal, Quebec
   Canada

   Phone: +1 514 282-8858
   Email: jean-marc.valin@octasic.com


   Koen Vos
   Skype Technologies S.A.
   Stadsgaarden 6
   Stockholm,   11645
   SE

   Phone: +46 855 921 989
   Email: koen.vos@skype.net































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