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Network Working Group                                         S. Wenger
Internet Draft                                               C. Perkins
Document: draft-ietf-avt-variable-rate-audio-00.txt
Expires: April 2005
                                                           October 2004





         RTP Timestamp Frequency for Variable Rate Audio Codecs



Status of this Memo

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   This document is a submission of the IETF AVT WG.  Comments should
   be directed to the AVT WG mailing list, avt@ietf.org.


Abstract

   This memo discusses the problems of audio codecs with variable
   external sampling rates.  Historically, for audio codecs, the RTP
   timestamp frequency was chosen to match the sampling rate of the
   audio codec.  However, this choice is nowadays more difficult to
   justify, because of the advent of audio codecs (and, even more
   important, practical use cases) that support multiple sample rates
   and the switch between the sample rates during the lifetime of an
   RTP session.  This Internet draft addresses the problem by
   suggesting that RTP Payload RFCs for such codecs to utilize a
   single, high, unified RTP timestamp frequency.

1.   Introduction

Internet Draft                                            October 2004
   One key property of audio codecs is the external input sample rate.
   For many of codecs, this sample rate is fixed.  ITU-T G.711 [2],
   also known as a-law and mu-law, uses, for example, a sample rate of
   8 kHz.  Other audio codecs give the user a choice between different
   sample rates.  However, until recently, applications never changed
   the sample rate during the lifetime of an RTP session, even if this
   is technically feasible and probably advantageous from both the user
   perception, and the network point-of-view.  One example for such a
   codec is MPEG-1 audio, layers 1, 2, or 3 [3].  At the time RTP [1]
   and the AV-profile [4] was developed, it was a reasonable design
   choice to use an RTP timestamp frequency that is identical to the
   codec's input sample rate, as this facilitates sample exact
   synchronization and processing of media data in endpoints, mixers
   and translators, among other advantages.  Although neither RTP [1]
   nor the audio-visual profile [4] require the codec sample rate being
   the same as the RTP timestamp frequency, this paradigm was observed
   in practice.

   Recently, codecs have been developed which do not only support
   variable sample rates, but use unannounced (in-band only signaled)
   changes of the sample rate as one of their key mechanisms.
   Similarly, applications have emerged, that not only support variable
   sample rates, but, to some extend, rely on this feature.  For most
   (if not all) of these codecs, it is true that the required bit rate
   and the user experience scales with the sample rate selected.  This
   allows, in the future, a network-dictated scaling of the
   transmission bit rate of an audio codec -- a feature that was not
   available before -- which could turn out to be very useful in
   Internet environments, for example to support congestion control.

   With the modern codecs mentioned, the current paradigm of RTP time
   stamp frequency equal to codec sample rate does not make much sense
   any more.  The purpose of this draft is to provide guidance for the
   developers of RTP payload specs for codecs with variable sample rate
   to use a single, relatively high, RTP timestamp frequency, which is
   specified in this draft.


2.   Audio codecs with variable sample rates: Examples

   Examples for audio codecs with variable sample rates, that (at least
   in theory) could switch the sample rate on the fly without
   out-of-band signaling support, include:

   *  AMR-WB+ [5] with a choice of 56 different sample rates
   *  VMR-WB [6] with the choice of 8 kHz and 16 kHz sample rates
   *  MPEG-4 AAC+ [7] with the choice of (need details here)
   *  Any others?

   All these codecs use in-band signaling of the sample rate.


3.   Rounding

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Internet Draft                                            October 2004
   It is possible (even likely) that no unified RTP timestamp frequency
   can be found that, on one hand, fulfills one key requirement spelled
   out later (namely: is low enough to make timestamp wrap-around
   during erasure periods unlikely for all practical application
   scenarios) and, one the other hand, is an integer multitude of all
   sampling frequencies the codecs support.  It is well possible that,
   in the future, codecs be developed that can make sample rate choices
   in a granularity of 1 Hz or even finer.  Considering this, it is
   required to specify a rounding algorithm for such cases where no
   sample-exact position of an audio frame can be found in the RTP
   timestamp numbering space.  Specifying this rounding algorithm
   ensures that all equipment conforming to this draft use the same
   rounding algorithm.  If that selected rounding algorithm guaranties
   that inaccuracies do not add up (as spelled out in the requirements
   later), then even frequent transcoding steps will not lead to an
   increase to inaccuracy of the timing beyond the unavoidable minimum.

4.   Requirements discussion

4.1. Requirements for this draft (general)

   1) This draft MUST specify a unified RTP timestamp frequency that
      fulfills the requirements of section 4.2.
   2) This draft MUST specify a rounding algorithm that can be used for
      non-sample exact alignment of samples stemming from more than one
      audio codec, at least one of which having a variable sample
      rate).  The rounding algorithm MUST fulfill the requirements of
      section 4.3.
   3) This draft SHOULD state that its provisions MUST be used for the
      design of future RTP payload formats for audio codecs with
      variable sample rates
   4) This draft SHOULD state that its provisions SHOULD be considered
      in the design of future RTP payload formats for non-audio codecs
      that have similar problems as variable sample rate audio codecs.
   5) This draft SHOULD provide an application example for a
      well-understood variable sample rate codec.

4.2. Requirements for the unified RTP timestamp rate

   6) The unified RTP timestamp rate (uRTR) MUST be sufficiently high
      to fulfill the requirements for timestamps in RFC3550[1]
   7) The uRTR MUST be low enough to make wrap-arounds of the RTP
      timestamp during erasure periods (packet loss bursts) unlikely in
      all reasonable application scenarios.
         Informative note: Such scenarios include, for example, cell
         handovers in wireless cellular networks, where erasure periods
         of a few seconds can occur.
   8) The uRTR SHOULD share the prime factors of the sample rates of
      the most commonly used fixed sample rate audio codecs, so to
      allow for sample exact mixing of streams coded by those fixed
      sample rate audio codecs.
   9) The uRTR SHOULD be chosen to include a sufficiently high number
      of prime factors so to support as many future variable sample
      rate codec code points as possible for sample-exact mixing

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Internet Draft                                            October 2004
4.3. Requirements for a rounding algorithm

   10)    The rounding algorithm MUST be applicable for all sample
          rates lower than the 0.5 * uRTR specified in this draft.
   11)    The rounding algorithm MAY specify a minimum and maximum
          sample rate, in units of x * uRTR.  Only within this band it
          is a reasonable expectation that the application of the
          rounding algorithm does not lead to audible distortions for
          the common user.
   12)    The rounding algorithm MUST be simple enough to be
          implemented, without a serious cycle burden, in networking
          equipment.
   13)    The rounding algorithm SHOULD be imlementable in fixed-point
          arithmetic
   14)    The rounding algorithm MAY, advantageously, be specified such
          that it does not require division operations
   15)    The rounding algorithm SHOULD be designed such that that
          multiple applications of the algorithm does not lead to the
          introduction of errors larger than one tick of the uRTR.
             Informative Note: this is a much more difficult
             requirement as it seems at the first glance.  Think of a
             transcoding scenario where variable goes to 44.1 kHz goes
             to variable, and the unified timestamp frequency does not
             share all prime factors of 44.1 kHz.  One way out of this
             would be to rewrite all fixed rate payload specs that use
             timestamp frequencies that do not fit into the prime
             factors of the uRTR to be rewritten so to use the uRTR.
             Is it possible to do this for 44.1 -- or is this nailed
             down in RFC3551?

5.   Open issues

   *  Very general: is this a good idea?
   *  What would be a good choice for the uRTR? 192 kHz?
   *  Is it a good idea to require ALL future I-Ds on audio (not only
      the variable clock frequency ones) to use the uRTR?
   *  Or only those that do not fit the uRTR (fit == subset of prime
      factors)?
   *  Revisit CD 44.1.  No variable sample rate needed? Are there
      proposals for an 88.2 CD audio codec?

6.   Security Considerations
   None

7.   Congestion Control
   None

8.   IANA Consideration
   None

9.   Acknowledgements
   None

10.  Full Copyright Statement

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Internet Draft                                            October 2004
   Copyright (C) The Internet Society (2004).  This document is subject
   to the rights, licenses and restrictions contained in BCP 78, and
   except as set forth therein, the authors retain all their rights.

   This document and the information contained herein are provided on
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   IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
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   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.


11.  Intellectual Property Notice

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12.   References

12.1.     Normative References
[1]  RTP, RFC 3550, STD 64

12.2.     Informative References

[2]  G.711
[3]  ISO/IEC 11172 part 3
[4]  RTP AV profile, RFC 3551, STD 65
[5]  AMR-WB+
[6]  VMR-WB
[7]  ISO/IEC 14496 part xxx, AAC+

13.      Author's Addresses

    Stephan Wenger                    Phone: +358-50-486-0637
    Nokia Research Center              Email: stewe@stewe.org
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Internet Draft                                            October 2004
    P.O. Box 100
    FIN-33721 Tampere
    Finland

    Colin Perkins <csp@csperkins.org>
    University of Glasgow
    Department of Computing Science
    17 Lilybank Gardens
    Glasgow G12 8QQ
    United Kingdom


14.  RFC Editor Considerations

none

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