path: root/doc/rfc/rfc3190.txt

Network Working Group                                       K. Kobayashi
Request for Comments: 3190             Communication Research Laboratory
Category: Standards Track                                       A. Ogawa
                                                         Keio University
                                                               S. Casner
                                                           Packet Design
                                                              C. Bormann
                                                 Universitaet Bremen TZI
                                                            January 2002


                         RTP Payload Format for
        12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio

Status of this Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2002).  All Rights Reserved.

Abstract

   This document specifies a packetization scheme for encapsulating
   12-bit nonlinear, 20-bit linear, and 24-bit linear audio data streams
   using the Real-time Transport Protocol (RTP).  This document also
   specifies the format of a Session Description Protocol (SDP)
   parameter to indicate when audio data is preemphasized before
   sampling.  The parameter may be used with other audio payload
   formats, in particular L16 (16-bit linear).

1. Introduction

   This document describes the sampling of audio data in 12-bit
   nonlinear, 20-bit linear, and 24-bit linear encodings, and specifies
   the encapsulation of the audio data into the Real-time Transport
   Protocol (RTP), version 2 [1,2].  DAT (digital audio tape) and DV
   (digital video) devices [3,4] use these audio encodings in addition
   to 16-bit linear encoding.  The packetization scheme for 16-bit
   linear audio (L16) is already specified [2,5].  This document
   specifies the packetization scheme for the other encodings following
   that for L16; in particular, when used with the RTP profile [2],
   these payload formats follow the encoding-independent rules for



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   sample ordering and channel interleaving specified in [2] plus
   extensions specified here.  This document also specifies out-of-band
   negotiation methods for the extended channel interleaving rules and
   for use when an analog preemphasis technique is applied to the audio
   data.

1.1 Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [6]

2. The need for RTP encapsulation of 12-, 20- and 24-bit audio

   Many high-quality digital audio and visual systems, such as DAT and
   DV, adopt sample-based audio encodings.  Different audio formats are
   used in various situations.  To transport the audio data using RTP,
   an encapsulation needs to be defined for each specific format.  Only
   16-bit linear audio encapsulation (L16) has thus far been defined.
   Other encoding formats have already appeared, such as the 12-bit
   nonlinear, 20-bit linear and 24-bit linear encodings used in the DAT
   and DV video world.  This specification defines the RTP payload
   encapsulation format in order to use the new encodings in the RTP
   environment.

3. 12-bit nonlinear audio encapsulation

   IEC 61119 [3] specifies the 12-bit nonlinear audio format in DAT and
   DV, called LP (Long Play) audio.  It would be easy to convert 12-bit
   nonlinear audio into 16-bit linear form at the RTP sender and
   transmit it using the L16 audio format already defined.  However,
   this would consume 33% more network bandwidth than necessary.  This
   payload format is specified as a more efficient alternative.

   The 12-bit nonlinear encoding is the same as for 16-bit linear audio
   except for the packing of each sampled data element.  Each sample of
   12-bit nonlinear audio is derived from a single sample of 16-bit
   linear audio by a nonlinear compression.  Table 1 shows the details
   of the conversion from 16 to 12 bits.  The result is a 12-bit signed
   value ranging from -2048 to 2047 and it is represented in two's
   complement notation.  The 12-bit samples are packed contiguously into
   payload octets starting with the most significant bit.  When the
   payload contains an odd number of samples, the four LSBs of the last
   octet are unused.  Parameters other than quantization, e.g., sampling
   frequency and audio channel assignment, are the same as in the L16
   payload format.  In particular, samples are packed into the packet in
   time sequence beginning with the oldest sample.




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    ------------------------------------------------------------
     32,767 (7FFFh) Y = INT(X/64) + (600h)        2,047 (7FFh)
     16,384 (4000h)                               1,792 (700h)
    ------------------------------------------------------------
     16,383 (3FFFh) Y = INT(X/32) + (500h)        1,791 (6FFh)
      8,192 (2000h)                               1,536 (600h)
    ------------------------------------------------------------
      8,191 (1FFFh) Y = INT(X/16) + (400h)        1,535 (5FFh)
      4,096 (1000h)                               1,280 (500h)
    ------------------------------------------------------------
      4,095 (0FFFh) Y = INT(X/8) + (300h)         1,279 (4FFh)
      2,048 (0800h)                               1,024 (400h)
    ------------------------------------------------------------
      2,047 (07FFh) Y = INT(X/4) + (200h)         1,023 (3FFh)
      1,024 (0400h)                                 768 (300h)
    ------------------------------------------------------------
      1,023 (03FFh) Y = INT(X/2) + (100h)           767 (2FFh)
        512 (0200h)                                 512 (200h)
    ------------------------------------------------------------
        511 (01FFh) Y = X                           511 (1FFh)
          0 (0000h)                                   0 (000h)
    ------------------------------------------------------------
         -1 (FFFFh) Y = X                            -1 (FFFh)
       -512 (FE00h)                                -512 (E00h)
    ------------------------------------------------------------
       -513 (FFFFh) Y = INT((X + 1)/2) - (101h)    -513 (DFFh)
     -1,024 (FE00h)                                -768 (D00h)
    ------------------------------------------------------------
     -1,025 (FBFFh) Y = INT((X + 1)/4) - (201h)    -769 (CFFh)
     -2,048 (F800h)                              -1,024 (C00h)
    ------------------------------------------------------------
     -2,049 (F7FFh) Y = INT((X + 1)/8) - (301h)  -1,025 (BFFh)
     -4,096 (F000h)                              -1,280 (B00h)
    ------------------------------------------------------------
     -4,097 (EFFFh) Y = INT((X + 1)/16) - (401h) -1,281 (AFFh)
     -8,192 (E000h)                              -1,536 (A00h)
    ------------------------------------------------------------
     -8,193 (DFFFh) Y = INT((X + 1)/32) - (501h) -1,537 (9FFh)
    -16,384 (C000h)                              -1,792 (900h)
    ------------------------------------------------------------
    -16,385 (BFFFh) Y = INT((X + 1)/64) - (601h) -1,793 (8FFh)
    -32,768 (8000h)                              -2,048 (800h)
    ------------------------------------------------------------

    Table 1. Conversion from 16-bit linear values (X) to 12-bit
             nonlinear values (Y) [3]





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   When conveying encoding information in an SDP [7] session
   description, the 12-bit nonlinear audio payload format specified here
   is given the encoding name "DAT12".  Thus, the media format
   representation might be:

      m=audio 49230 RTP/AVP 97 98
      a=rtpmap:97 DAT12/32000/2
      a=rtpmap:98 L16/48000/2

4. 20- and 24-bit linear audio encapsulation

   The 20- and 24-bit linear audio encodings are simply an extension of
   the L16 linear audio encoding [2].  The 20- or 24-bit uncompressed
   audio data samples are represented as signed values in two's
   complement notation.  The samples are packed contiguously into
   payload octets starting with the most significant bit.  For the
   20-bit encoding, when the payload contains an odd number of samples,
   the four LSBs of the last octet are unused.  Samples are packed into
   the packet in time sequence beginning with the oldest sample.

   When conveying encoding information in an SDP session description,
   the 20- and 24-bit linear audio payload formats specified here are
   given the encoding names "L20" and "L24", respectively.  An example
   SDP audio media description would be:

      m=audio 49230 RTP/AVP 99 100
      a=rtpmap:99 L20/48000/2
      a=rtpmap:100 L24/48000

5. Preemphasized audio data

   In order to improve the higher frequency characteristics of audio
   signals, analog preemphasis is often applied to the signal before
   quantization.  If analog preemphasis was applied before the payload
   data was sampled, the type of the preemphasis SHOULD be conveyed with
   out-of-band signaling.  An "emphasis" parameter is defined for this
   purpose and may be conveyed either as a MIME optional parameter or
   using the SDP format-specific attribute (a=fmtp line) as below:

      a=fmtp:<payload type> emphasis=<emphasis type>

   Only one <emphasis type> value is defined for the parameter at this
   point:

      50-15           <50/15 microsecond CD-type emphasis>






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   The emphasis attribute MUST NOT be included in the SDP record if
   preemphasis was not applied.  This rule allows the emphasis attribute
   to be used with other audio formats, in particular L16 [2], while
   retaining backward compatibility with existing implementations so
   long as preemphasis is not applied.  If an existing application that
   does not implement preemphasis accepts a session description with an
   emphasis attribute but ignores that attribute, the only penalty is
   that the sound will be too "bright" when receiving or "dull" when
   sending.

   A sample SDP record showing preemphasis applied only to payload type
   99 might be as follows:

      m=audio 49230 RTP/AVP 99 100
      a=rtpmap:99 L20/48000/2
      a=fmtp:99 emphasis=50-15
      a=rtpmap:100 L24/48000

6. Translation of DV audio error code

   The DV video specification IEC 61834-4 [4] defines the negative full-
   scale audio sample value to be an audio error code indicating that no
   valid audio sample is available for that sample period.  Such an
   error might occur due to a failure while reading audio data from
   magnetic tape.  The audio error code values for each of the DV audio
   encodings are (in hexadecimal):

      12-bit nonlinear:  800h
      16-bit linear:     8000h
      20-bit linear:     80000h

   For the payload formats defined in this document, as well as for the
   L16 payload format defined in [2], no such error code is defined.
   That is, all possible sample values are valid.  When an RTP sender
   accepts audio samples from a DV video system and encapsulates those
   samples according to one of these payload formats, the RTP sender
   SHOULD perform some error concealment algorithm which may depend upon
   whether a single sample error or multiple sample errors have
   occurred.  The error concealment algorithm is not specified here and
   is left to the implementation.  The RTP sender MAY treat the error
   code as if it were a valid audio sample, but this is likely to cause
   undesirable audio output.

   Conversely, an RTP receiver that accepts audio packets in one of
   these payload formats and delivers the audio samples to a DV video
   system SHOULD translate the audio samples that would be interpreted
   as error codes into the next smaller negative audio value.  Such
   audio samples may be present because the audio packets may have come



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   from a source other than a DV video system.  The DV video
   specification [4] gives the following translations for the defined
   audio encodings:

      12-bit nonlinear:  800h              ->  801h
      16-bit linear:     8000h             ->  8001h
      20-bit linear:     80000h - 8000Fh   ->  80010h

   For the 20-bit linear encoding, note that multiple audio sample
   values are translated in order to allow a 16-bit system to play 20-
   bit audio data by ignoring the least significant four bits.  Note
   also that no translation is specified for 24-bit linear audio because
   that encoding is not included in the DV video specification.

7. Channel interleaving and non-AIFF-C audio channel convention

   When multiple channels of audio, such as in a stereo program, are
   multiplexed into a single RTP stream, the audio samples from each
   channel are interleaved according to the rules specified in [2] to be
   consistent with the L16 payload format.  That is, samples from
   different channels taken at the same sampling instant are packed into
   consecutive octets.  For example, for a two-channel encoding, the
   sample sequence is (left channel, first sample), (right channel,
   first sample), (left channel, second sample), (right channel, second
   sample).  Samples for all channels belonging to a single sampling
   instant MUST be contained in the same packet.

   This sample order differs from the packing of samples into blocks in
   a native DV audio stream.  Therefore, applications transmitting DV
   audio using the payload formats defined in this document MUST
   reshuffle the samples into the order specified here.  This
   requirement is intended to enable interworking between DV systems and
   other digital audio systems.  Applications choosing to send bundled
   DV audio/video streams using the native DV block structure may use
   the payload format defined in [8] instead.

   Most of the existing RTP audio payload formats follow the AIFF-C
   convention for channel ordering as specified in [2] when sending more
   than two audio channels within a single RTP stream.  However, this
   convention does not cover some applications.  For example, some DV
   audio formats define a "woofer" channel, but AIFF-C does not include
   this frequency-dependent channel.  Thus, it is necessary to specify
   the audio channel allocation information explicitly when the contents
   of the audio stream are beyond the scope of AIFF-C.

   For DV audio streams of 4 or more channels, the channel order MUST be
   specified out-of-band.  This applies both to the payload formats
   defined in this document and to the L16 payload format.  A "channel-



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   order" parameter is defined here for this purpose and may be conveyed
   either as a MIME optional parameter or with the SDP a=fmtp attribute
   using the following syntax:

      a=fmtp:<payload type> channel-order=<convention>.<order>

   The first component of the value, <convention>, specifies the type of
   channel assignment convention used.  This allows for conventions
   other than AIFF-C and DV to be defined in the future.  This document
   defines only one convention for use in the channel-order parameter:

      DV

   The second component of the value, <order>, indicates the arrangement
   of channels within the stream.  The DV video specification [4]
   defines the types of audio channels that may be present and in what
   order.  The symbols used to denote the channel types are reproduced
   in the Appendix at the end of this document.  For the DV convention,
   the following values, which were formed from the concatenation of the
   individual channel symbols in the allowed channel orders, are defined
   for the <order> component:

      4 channels: LRLsRs, LRCS, LRCWo
      5 channels: LRLsRsC
      6 channels: LRLsRsCS, LmixRmixTWoQ1Q2
      8 channels: LRCWoLsRsLmixRmix, LRCWoLs1Rs1Ls2Rs2, LRCWoLsRsLcRc

   The <convention> and <order> symbols are case-insensitive, but are
   shown here in mixed case to make the individual channel symbols more
   apparent.  These concatenated symbols were deliberately constructed
   without separators to make clear the fact that the channels MUST NOT
   be assembled in other, arbitrary orders.

   For interworking with DV video systems, some of the audio encodings
   are defined only for a subset of the channel combinations listed
   above.  The DV video specification lists the channel combinations
   that are allowed for each encoding.

   The channel-order parameter MUST be consistent with the number of
   channels specified in the MIME optional parameter "channels" or in
   the a=rtpmap attribute of SDP.  For RTP audio streams of 1, 2 or 3
   channels, the AIFF-C channel order is used and is implicit in the
   number of audio channels specified.  To allow backward compatibility,
   the channel-order parameter MUST NOT be included in this case.

   Note that for the DV convention with 5 channels only one channel
   order is allowed, but for consistency the channel-order parameter
   MUST be included nonetheless.



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   An example of an SDP session description using the channel-order
   parameter is:

      v=0
      o=ikob 2890844526 2890842807 IN IP4 126.16.64.4
      s=POI (Audio only)
      i=A Seminar on making Presentations on the Internet
      u=http://www.koganei.wide.ad.jp/~ikob/POI/index.html
      e=ikob@koganei.wide.ad.jp (Katsushi Kobayashi)
      c=IN IP4 224.2.17.12/127
      t=2873397496 2873404696
      m=audio 49170 RTP/AVP 112 113
      a=rtpmap:112 L16/48000/2
      a=rtpmap:113 DAT12/32000/4
      a=fmtp:113 emphasis=50-15; channel-order=DV.LRCWO

   This session description shows the audio medium being transmitted in
   two formats, L16 and DAT12, using payload type numbers 112 and 113,
   respectively.  For the L16 format, the audio data contains 2-channel
   stereo following the implicit, default AIFF-C convention for left
   channel first and right channel second.  For the DAT12 format, the
   audio data contains 4 channels following the DV audio convention for
   the channels left, right, center, and woofer.

   This example also shows how multiple MIME optional parameters
   ("emphasis" and "channel-order") for these payload formats are mapped
   to a single a=fmtp attribute as a semicolon separated list of
   parameter=value pairs.

   The channel-order parameter described here provides a generic out-of-
   band mechanism to define alternatives to the AIFF-C channel order.
   However, if multi-channel audio data could be sent following the
   AIFF-C channel convention after simple processing, such as a data
   shuffling on the sender side, the alternative channel order SHOULD
   NOT be defined and instead the AIFF-C order SHOULD be employed to
   maximize interoperability.

   Moreover, multiple channels of audio data should only be multiplexed
   within a single RTP stream when all of the audio channels are
   intended to be reproduced together, such as the left and right
   channels in a stereo program.  Independent audio channels, such as
   multiple language translations, SHOULD be sent in separate RTP
   sessions.  This reduces bandwidth requirements by allowing receivers
   to tune in to only those channels which are desired.







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8. MIME registration

   This document defines some new RTP payload format names which are
   also registered as MIME subtypes: DAT12, L20 and L24.  The
   registration forms for these MIME subtypes are provided in the next
   sections.

8.1 MIME registration form for audio/DAT12

   MIME media type name: audio

   MIME subtype name: DAT12

   Required parameters:
      rate: number of samples per second -- RECOMMENDED values for rate
         are 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000
         samples per second.  Other values are permissible.

   Optional parameters:
      channels: how many audio streams are interleaved -- defaults to 1;
         stereo would be 2, etc.  Interleaving takes place between
         individual 12-bit samples.

      emphasis: analog preemphasis applied to the data before
         quantization.  The only emphasis value defined here is
         emphasis=50-15 to indicate 50/15 microsecond preemphasis.  This
         parameter MUST be omitted if no analog preemphasis was applied.

      channel-order: specifies the sample interleaving order for
         multiple-channel audio streams (see RFC 3190 Section 7).
         Permissible values are DV.LRLsRs, DV.LRCS, DV.LRCWo,
         DV.LRLsRsC, DV.LRLsRsCS, DV.LmixRmixTWoQ1Q2,
         DV.LRCWoLsRsLmixRmix, DV.LRCWoLs1Rs1Ls2Rs2, DV.LRCWoLsRsLcRc.
         For interoperation with DV video systems, only a subset of
         these channel combinations is specified for use with 12-bit
         nonlinear encoding in the DV video specification [4]; that
         subset is all of the above except DV.LmixRmixTWoQ1Q2.  This
         parameter MUST be omitted when the AIFF-C channel order
         convention is in use.

   Encoding considerations:
      DAT12 audio can be transmitted with RTP as specified in RFC 3190.

   Security considerations: See Section 9.

   Interoperability considerations: NONE





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   Published specification:
      IEC 61119 Standard [4] and RFC 3190.

   Applications which use this media type:
      Audio communication.

   Additional information:
      Magic number(s): None
      File extension(s): None
      Macintosh File Type Code(s): None

   Person & email address to contact for further information:
      Katsushi Kobayashi
      e-mail: ikob@koganei.wide.ad.jp

   Intended usage: COMMON

   Author/Change controller:
      Katsushi Kobayashi
      e-mail: ikob@koganei.wide.ad.jp

8.2 MIME registration form for audio/L20

   MIME media type name: audio

   MIME subtype name: L20

   Required parameters:
      rate: number of samples per second -- RECOMMENDED values for rate
         are 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000
         samples per second.  Other values are permissible.

   Optional parameters:
      channels: how many audio streams are interleaved -- defaults to 1;
         stereo would be 2, etc.  Interleaving takes place between
         individual 20-bit samples.

      emphasis: analog preemphasis applied to the data before
         quantization.  The only emphasis value defined here is
         emphasis=50-15 to indicate 50/15 microsecond preemphasis.  This
         parameter MUST be omitted if no analog preemphasis was applied.

      channel-order: specifies the sample interleaving order for
         multiple-channel audio streams (see RFC 3190 Section 7).
         Permissible values are DV.LRLsRs, DV.LRCS, DV.LRCWo,
         DV.LRLsRsC, DV.LRLsRsCS, DV.LmixRmixTWoQ1Q2,
         DV.LRCWoLsRsLmixRmix, DV.LRCWoLs1Rs1Ls2Rs2, DV.LRCWoLsRsLcRc.
         For interoperation with DV video systems, none of these channel



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         combinations is specified for use with 20-bit linear encoding
         in the DV video specification [4]; only mono and stereo are
         allowed.  This parameter MUST be omitted when the AIFF-C
         channel order convention is in use.

   Encoding considerations:
      L20 audio can be transmitted with RTP as specified in RFC 3190.

   Security considerations: See Section 9.

   Interoperability considerations: NONE

   Published specification:
      RFC 3190.

   Applications which use this media type:
      Audio communication.

   Additional information:
      Magic number(s): None
      File extension(s): None
      Macintosh File Type Code(s): None

   Person & email address to contact for further information:
      Katsushi Kobayashi
      e-mail: ikob@koganei.wide.ad.jp

   Intended usage: COMMON

   Author/Change controller:
      Katsushi Kobayashi
      e-mail: ikob@koganei.wide.ad.jp

8.3 MIME registration form for audio/L24

   MIME media type name: audio

   MIME subtype name: L24

   Required parameters:
      rate: number of samples per second -- RECOMMENDED values for rate
         are 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000
         samples per second.  Other values are permissible.

   Optional parameters:
      channels: how many audio streams are interleaved -- defaults to 1;
         stereo would be 2, etc.  Interleaving takes place between
         individual 24-bit samples.



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      emphasis: analog preemphasis applied to the data before
         quantization.  The only emphasis value defined here is
         emphasis=50-15 to indicate 50/15 microsecond preemphasis.  This
         parameter MUST be omitted if no analog preemphasis was applied.

      channel-order: specifies the sample interleaving order for
         multiple-channel audio streams (see Section 7).  Permissible
         values are DV.LRLsRs, DV.LRCS, DV.LRCWo, DV.LRLsRsC,
         DV.LRLsRsCS, DV.LmixRmixTWoQ1Q2, DV.LRCWoLsRsLmixRmix,
         DV.LRCWoLs1Rs1Ls2Rs2, DV.LRCWoLsRsLcRc.  This parameter MUST be
         omitted when the AIFF-C channel order convention is in use.

   Encoding considerations:
      L24 audio can be transmitted with RTP as specified in RFC 3190.

   Security considerations: See Section 9.

   Interoperability considerations: NONE

   Published specification:
      RFC 3190.

   Applications which use this media type:
      Audio communication.

   Additional information:
      Magic number(s): None
      File extension(s): None
      Macintosh File Type Code(s): None

   Person & email address to contact for further information:
      Katsushi Kobayashi
      e-mail: ikob@koganei.wide.ad.jp

   Intended usage: COMMON

   Author/Change controller:
      Katsushi Kobayashi
      e-mail: ikob@koganei.wide.ad.jp












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9.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [1], and any appropriate RTP profile [2].  This implies
   that confidentiality of the media streams is achieved by encryption.
   Because the data compression used along with this payload format is
   applied end-to-end, encryption may be performed after compression so
   there is no conflict between the two operations.

   A potential denial-of-service threat exists for data encodings using
   compression techniques that have non-uniform receiver-end
   computational load.  The attacker can inject pathological datagrams
   into the stream which are complex to decode and cause the receiver to
   be overloaded.  However, this encoding does not exhibit any
   significant non-uniformity.

   As with any IP-based protocol, in some circumstances a receiver may
   be overloaded simply by the receipt of too many packets, either
   desired or undesired.  Network-layer authentication may be used to
   discard packets from undesired sources, but the processing cost of
   the authentication itself may be too high.  In a multicast
   environment, pruning of specific sources may be implemented in future
   versions of IGMP [9] and in multicast routing protocols to allow a
   receiver to select which sources are allowed to reach it.

10.  IANA Considerations

   This document defines two new MIME subtype-specific optional
   parameters "emphasis" and "channel-order", which are specified with
   the sets of permissible values for those parameters in Sections 5 and
   7, respectively.  Section 8 includes registrations for three new MIME
   subtypes that use those optional parameters.  These registrations
   define the subset of the optional parameter values allowed for each
   subtype.  It is expected that the number of additional values that
   will need to be defined for these optional parameters is small.
   Therefore, anyone wishing to define new values is required to produce
   a revision of this document to be vetted through the normal Internet
   Standards process.












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11.  References

   [1] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:
       A Transport Protocol for real-time applications," RFC 1889,
       January 1996.

   [2] H. Schulzrinne, "RTP profile for audio and video conferences with
       minimal control", RFC 1890, January 1996.

   [3] IEC 61119, Digital audio tape cassette system (DAT), November
       1992.

   [4] IEC 61834, Helical-scan digital video cassette recording system
       using 6,35 mm magnetic tape for consumer use (525-60, 625-50,
       1125-60 and 1250-50 systems), August 1998.

   [5] Salsman, J., "The Audio/L16 MIME content type", RFC 2586, May
       1999.

   [6] Bradner, S., "Key words for use in RFCs to Indicate Requirement
       Levels", BCP 14, RFC 2119, March 1997.

   [7] Handley, M. and V. Jacobson, "SDP: Session Description Protocol",
       RFC 2327, April 1998.

   [8] Kobayashi, K., Ogawa, A., Casner, S. and C. Bormann, "RTP Payload
       Format for DV (IEC 61834) Video", RFC 3189, January 2002.

   [9] Deering, S., "Host Extensions for IP Multicasting", STD 5, RFC
       1112, August 1989.





















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Appendix

   The DV audio channel symbols are specified in Table 2.  These symbols
   are taken from the notation used in the DV video specification IEC
   61834-4 [4], chapter 8.1.  For the exact meaning of each symbol, the
   original DV video specification should be consulted.

      L: Left channel of stereo
      R: Right channel of stereo
      M: Monaural signal
      C: Center channel of 3,4,6 or 8 channel audio
      S: Surround channel of 4 channel audio
      Ls, Ls1, Ls2: Left surround channel
      Rs, Rs1, Rs2: Right surround channel
      Lc: Left center channel of 8 channel audio
      Rc: Right center channel of 8 channel audio
      Wo: Woofer channel
      Lmix: L + 0.7071C + 0.7071LS
      Rmix: R + 0.7071C + 0.7071RS
      T: 0.7071C
      Q1: 0.7071LS + 0.7071RS
      Q2: 0.7071LS - 0.7071RS

      Table 2. Channel symbols for audio channels in DV video [4]



























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Authors' Addresses

   Katsushi Kobayashi
   Communication Research Laboratory
   4-2-1 Nukii-kita machi, Koganei
   Tokyo 184-8795 JAPAN

   Phone: +81 42 327 6174
   EMail: ikob@koganei.wide.ad.jp


   Akimichi Ogawa
   Keio University
   5322 Endo, Fujisawa
   Kanagawa 252 JAPAN

   EMail:  akimichi@sfc.wide.ad.jp


   Stephen L. Casner
   Packet Design
   2465 Latham Street
   Mountain View, CA 94040
   United States

   Phone: +1 650-943-1843
   EMail: casner@acm.org


   Carsten Bormann
   Universitaet Bremen TZI
   Postfach 330440
   D-28334 Bremen, Germany

   Phone: +49 421 218 7024
   Fax:   +49 421 218 7000
   EMail: cabo@tzi.org














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Full Copyright Statement

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Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.



















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