Network Working Group                                            B. Link
Request for Comments: 4184                                      T. Hager
Category: Standards Track                             Dolby Laboratories
                                                               J. Flaks
                                                  Microsoft Corporation
                                                           October 2005


                  RTP Payload Format for AC-3 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 (2005).

Abstract

  This document describes an RTP payload format for transporting audio
  data using the AC-3 audio compression standard.  AC-3 is a high
  quality, multichannel audio coding system that is used for United
  States HDTV, DVD, cable television, satellite television and other
  media.  The RTP payload format presented in this document includes
  support for data fragmentation.

1.  Introduction

  AC-3 [ATSC] is a high-quality audio codec (audio coding format)
  designed to encode multiple channels of audio into a low bit-rate
  format.  AC-3 achieves its large compression ratios via encoding a
  multiplicity of channels as a single entity.  Dolby Digital, which is
  a branded version of AC-3, encodes up to 5.1 channels of audio.

  AC-3 has been adopted as an audio compression scheme for many
  consumer and professional applications.  It is a mandatory audio
  codec for DVD-video, Advanced Television Standards Committee (ATSC)
  digital terrestrial television and Digital Living Network Alliance
  (DLNA) home networking, as well as an optional multichannel audio
  format for DVD-audio.

  There is a need to stream AC-3 data over IP networks.  The Internet
  Real Time Protocol (RTP) provides a mechanism for stream



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  synchronization and hence serves as the best transport solution for
  AC-3, which is primarily used in audio-for-video applications.
  Applications for streaming AC-3 include streaming movies from a home
  media server to a display, video on demand, and multichannel Internet
  radio.

  Section 2 gives a brief overview of the AC-3 algorithm.  Section 3
  specifies values for fields in the RTP header, while Section 4
  specifies the AC-3 payload format.  Section 5 discusses media types
  and SDP usage.  Security considerations are covered in Section 6,
  congestion control in Section 7, and IANA considerations in Section
  8.  References are given in Sections 9 and 10.

  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 [RFC2119].

2.  Overview of AC-3

  AC-3 can deliver up to 5.1 channels of audio at data rates
  approximately equal to half of one PCM channel [ATSC], [1994AC3],
  [1996AC3].  The ".1" refers to a band-limited, optional, low-
  frequency effects (LFE) channel.  AC-3 was designed for signals
  sampled at rates of 32, 44.1, or 48 kHz.  Data rates can vary between
  32 kbps and 640 kbps, depending on the number of channels and the
  desired quality.

  AC-3 exploits psycho-acoustic phenomena that cause a significant
  fraction of the information contained in a typical audio signal to be
  inaudible.  Substantial data reduction occurs via the removal of
  inaudible information contained in an audio stream.  Source coding
  techniques are further used to reduce the data rate.

  Like most perceptual coders, AC-3 operates in the frequency domain.
  A 512-point TDAC transform is taken with 50% overlap, providing 256
  new frequency samples.  Frequency samples are then converted to
  exponents and mantissas.  Exponents are differentially encoded.
  Mantissas are allocated a varying number of bits depending on the
  audibility of the associated spectral components.  Audibility is
  determined via a masking curve.  Bits for mantissas are allocated
  from a global bit pool.

2.1.  AC-3 Bit Stream

  AC-3 bit streams are organized into synchronization frames.  Each
  AC-3 frame contains a Synchronization Information (SI) field, a Bit
  Stream Information (BSI) field, and 6 audio blocks (ABs) that each
  represent 256 PCM samples for all channels.  The frame ends with an



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  optional auxiliary data field (AUX) and an error correction field
  (CRC).  The entire frame represents the time duration of 1536 PCM
  samples across all coded channels [ATSC].  AC-3 encodes audio sampled
  at 32 kHz, 44.1 kHz, and 48 kHz.  From Annex A, Part 2, of [ATSC],
  the time duration of an AC-3 frame varies with the sampling rate as
  follows:

     Sampling rate          Frame duration
     _____________________________________
        48   kHz                32    ms
        44.1 kHz        approx. 34.83 ms
        32   kHz                48    ms


  Figure 1 shows the AC-3 frame format.

  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |SI |BSI|  AB0  |  AB1  |  AB2  |  AB3  |  AB4  |  AB5  |AUX|CRC|
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                       Figure 1. AC-3 Frame Format

  The Synchronization Information field contains information needed to
  acquire and maintain codec synchronization.  The Bit Stream
  Information field contains parameters that describe the coded audio
  service [ATSC].  Each audio block contains fields that indicate the
  use of various coding tools: block switching, dither, coupling, and
  exponent strategy.  They also contain metadata, optionally used to
  enhance the playback, such as dynamic range control.  Finally, the
  exponents and bit allocation data needed to decode the mantissas into
  audio data, and the mantissas themselves, are included.  The
  placement of these fields in an AC-3 audio block is shown in Figure
  2.  The fields shown in this figure are described in detail in
  [ATSC].  Note that field sizes vary depending on the coded data.

  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |  Block  |Dither |Dynamic    |Coupling |Coupling     |Exponent |
  |  Switch |Flags  |Range Ctrl |Strategy |Coordinates  |Strategy |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |     Exponents       | Bit Allocation  |        Mantissas      |
  |                     | Parameters      |                       |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                    Figure 2. AC-3 Audio Block Format







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3.  RTP AC-3 Header Fields

  o Payload Type (PT): The assignment of an RTP payload type for this
    packet format is outside the scope of this document.  It is
    specified by the RTP profile under which this payload format is
    used, or signaled dynamically out-of-band (e.g., using SDP).

  o Marker (M) bit: The M bit is set to one to indicate that the RTP
    packet payload contains at least one complete AC-3 frame or
    contains the final fragment of an AC-3 frame.

  o Extension (X) bit: Defined by the RTP profile used.

  o Timestamp: A 32-bit word that corresponds to the sampling instant
    for the first AC-3 frame in the RTP packet.  Packets containing
    fragments of the same frame MUST have the same time stamp.  The
    timestamp of the first RTP packet sent SHOULD be selected at
    random.  Thereafter, the timestamp increases linearly with the
    number of samples included in each frame (i.e., by 1536 for each
    frame).

4.  RTP AC-3 Payload Format

  This payload format is defined for AC-3, as defined in the main part
  and Annex D of [ATSC].  It is not defined for E-AC-3, as defined in
  Annex E of [ATSC], and it MUST NOT be used to carry E-AC-3.

  According to [RFC2736], RTP payload formats should contain an
  integral number of application data units (ADUs).  The AC-3 frame
  corresponds to an ADU, in the context of this payload format.  Each
  RTP payload MUST start with the two-byte payload header followed by
  an integral number of complete AC-3 frames or by a single fragment of
  an AC-3 frame.

  If an AC-3 frame exceeds the MTU for a network, it SHOULD be
  fragmented for transmission within an RTP packet.  Section 4.2
  provides guidelines for creating frame fragments.














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4.1.  Payload-Specific Header

  There is a two-octet Payload Header at the beginning of each payload.

4.1.1.  Payload Header

  Each AC-3 RTP payload MUST begin with the payload header shown in
  Figure 3.

                 0                   1
                 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
                +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                |    MBZ    | FT|       NF      |
                +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                Figure 3.  AC-3 RTP Payload Header

  o MBZ (Must Be Zero): Bits marked MBZ SHALL be set to the value zero
    and SHALL be ignored by receivers.  The bits are reserved for
    future extensions.

  o FT (Frame Type): This two-bit field indicates the type of frame(s)
    present in the payload.  It takes the following values:

     0 - One or more complete frames.
     1 - Initial fragment of frame which includes the first 5/8ths of
         the frame.  (See Section 4.2.)
     2 - Initial fragment of frame, which does not include the first
         5/8ths of the frame.
     3 - Fragment of frame other than initial fragment.  (Note that M
         bit in RTP header is set for final fragment).

  o NF (Number of frames/fragments): An 8-bit field whose meaning
    depends on the Frame Type (FT) in this payload.  For complete
    frames (FT of 0), it is used to indicate the number of AC-3 frames
    in the RTP payload.  For frame fragments (FT of 1, 2, or 3), it is
    used to indicate the number fragments (and therefore packets) that
    make up the current frame.  NF MUST be identical for packets
    containing fragments of the same frame.












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RFC 4184                  RTP Payload for AC-3              October 2005


  Figure 4 shows the full AC-3 RTP payload format.

        +-+-+-+-+-+-+-+-+-+-+-+-+-+- .. +-+-+-+-+
        | Payload | Frame | Frame |     | Frame |
        | Header  |  (1)  |  (2)  |     |  (n)  |
        +-+-+-+-+-+-+-+-+-+-+-+-+-+- .. +-+-+-+-+

                Figure 4. Full AC-3 RTP payload

  When receiving AC-3 payloads with FT = 0 and more than a single frame
  (NF > 1), a receiver needs to use the "frmsizecod" field in the
  Synchronization Information (syncinfo) block in each AC-3 frame to
  determine the frame's length.  That way a receiver can determine the
  boundary of the next frame.  Note that the frame length may vary from
  frame to frame.

4.2.  Fragmentation of AC-3 Frames

  The size of an AC-3 frame depends on the sample rate of the audio and
  the data rate used by the encoder (which are indicated in the
  "Synchronization Information" header in the AC-3 frame).  The size of
  a frame, for a given sample rate and data rate, is specified in Table
  5.18 ("Frame Size Code Table") of [ATSC].  This table shows that AC-3
  frames range in size from a minimum of 128 bytes to a maximum of 3840
  bytes.  If the size of an AC-3 frame exceeds the MTU size, the frame
  SHOULD be fragmented at the RTP level.  The fragmentation MAY be
  performed at any byte boundary in the frame.  RTP packets containing
  fragments of the same AC-3 frame SHALL be sent in consecutive order,
  from first to last fragment.  This enables a receiver to assemble the
  fragments in correct order.

  When an AC-3 frame is fragmented, it MAY be fragmented such that at
  least the first 5/8ths of the frame data is in the first fragment.
  This provides greater resilience to packet loss.  This initial
  portion of any frame is guaranteed to contain the data necessary to
  decode the first two blocks of the frame.  Any frame fragments other
  than those containing the first 5/8ths of frame data are only
  decodable once the complete frame is received.  The 5/8ths point of
  the frame is defined in Table 7.34 ("5/8_frame Size Table") of
  [ATSC].











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5.  Types and Names

5.1.  Media Type Registration

  This registration uses the template defined in [DRAFT-FREED] and
  follows RFC 3555 [RFC3555].

  o Type name:                         audio

  o Subtype name:                      ac3

  o Required parameters:

     rate: The RTP timestamp clock rate that is equal to the audio
        sampling rate.  Permitted rates are 32000, 44100, and 48000.

  o Optional parameters:

     channels: From a sender, the maximum number of channels present in
        the AC3 stream.  From a receiver, the maximum number of output
        channels the receiver will deliver.  This MUST be a number
        between 1 and 6.  The LFE (".1") channel MUST be counted as one
        channel.  Note that the channel order used in AC-3 differs from
        the channel order scheme in [RFC3551].  The AC-3 channel order
        scheme can be found in Table 5.8 of [ATSC].

     ptime: See RFC 2327 [RFC2327].

     maxptime: See RFC 3267 [RFC3267].

  o Encoding considerations:

        This media type is framed (see section 4.8 in [DRAFT-FREED])
        and contains binary data.

  o Security considerations:

        See Section 6 of this document.

  o Interoperability considerations:

        None

  o Published specification:

        This payload format specification and see [ATSC].





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  o Applications that use this media type:

        Multichannel audio compression of audio and audio for video.

  o Additional Information:

        Magic number(s):
                The first two octets of an AC-3 frame are always the
                synchronization word, which has the hex value 0x0B77.

  o Person & email address to contact for further information:

        Brian Link <[email protected]>
        IETF AVT working group.

  o Intended Usage:

        COMMON

  o Restrictions on usage:

        This media type depends on RTP framing, and hence is only
        defined for transfer via RTP [RFC3550].  Transport within other
        framing protocols is not defined at this time.

  Author/Change controller:

        IETF Audio/Video Transport Working Group delegated from the
        IESG.

5.2.  SDP Usage

  The information carried in the MIME media type specification has a
  specific mapping to fields in the Session Description Protocol (SDP)
  [RFC2327], which is commonly used to describe RTP sessions.  When SDP
  is used to specify sessions employing AC-3, the mapping is as
  follows:

  o  The Media type ("audio") goes in SDP "m=" as the media name.

  o  The Media subtype ("ac3") goes in SDP "a=rtpmap" as the encoding
     name.

  o  The required parameter "rate" also goes in "a=rtpmap" as the clock
     rate, optionally followed by the parameter "channel".

  o  The optional parameters "ptime" and "maxptime" go in the SDP
     "a=ptime" and "a=maxptime" attributes, respectively.



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  An example of the SDP data for AC-3:

           m=audio 49111 RTP/AVP 100
           a=rtpmap:100 ac3/48000/6

  Certain considerations are needed when SDP is used to perform
  offer/answer exchanges [RFC3264].

     o  The "rate" is a symmetric parameter, and the answer MUST use
        the same value or remove the payload type.

     o  The "channels" parameter is declarative and indicates, for
        recvonly or sendrecv, the desired channel configuration to
        receive, and for sendonly, the intended channel configuration
        to transmit.  All receivers are capable of receiving any of the
        defined channel configurations, and the parameter exchange
        might be used to help optimize the transmission to the number
        of channels the receiver requests.  If the "channels" parameter
        is omitted, a default maximum value of 6 is implied.

     o  The "ptime" and "maxptime" parameters are negotiated as defined
        for "ptime" in RFC 3264 [RFC3264].

6.  Security Considerations

  The payload format described in this document is subject to the
  security considerations defined in the RTP specification [RFC3550]
  and in any applicable RTP profile (e.g., [RFC3551]).  To protect the
  user's privacy and any copyrighted material, confidentiality
  protection would have to be applied.  To also protect against
  modification by intermediate entities and ensure the authenticity of
  the stream, integrity protection and authentication would be
  required.  Confidentiality, integrity protection, and authentication
  have to be provided by a mechanism external to this payload format,
  e.g., SRTP [RFC3711].

  The AC-3 format is designed so that the validity of data frames can
  determined by decoders.  A decoder that encounters a malformed frame
  discards the malformed data and conceals the errors in the audio
  output until a valid frame is detected and decoded.  This is expected
  to prevent crashes and other abnormal decoder behavior in response to
  errors or attacks.

7.  Congestion Control

  The general congestion control considerations for transporting RTP
  data apply to AC-3 audio over RTP as well.  See the RTP specification
  [RFC3550] and any applicable RTP profile (e.g., [RFC3551]).



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RFC 4184                  RTP Payload for AC-3              October 2005


  AC-3 encoders may use a range of bit rates to encode audio data, so
  it is possible to adapt network bandwidth by adjusting the encoder
  bit rate in real time or by having multiple copies of content encoded
  at different bit rates.  Additionally, packing more frames in each
  RTP payload can reduce the number of packets sent and hence the
  overhead from IP/UDP/RTP headers, at the expense of increased delay
  and reduced robustness against packet losses.

8.  IANA Considerations

  A new media subtype has been assigned for AC-3; see Section 5.1.

9.  Normative References

  [RFC2119]     Bradner, S., "Key Words for use in RFCs to Indicate
                Requirement Levels", RFC 2119, March 1997.

  [ATSC]        U.S. Advanced Television Systems Committee (ATSC),
                "ATSC Standard: Digital Audio Compression (AC-3),
                Revision B," Doc A/52B, June 2005.

  [RFC2327]     Handley, M. and V. Jacobson, "SDP: Session Description
                Protocol", RFC 2327, April 1998.

  [RFC3550]     Schulzrinne, H., Casner, S., Frederick, R., and V.
                Jacobson, "RTP: A Transport Protocol for Real-Time
                Applications", STD 64, RFC 3550, July 2003.

  [RFC3264]     Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
                Model with Session Description Protocol (SDP)", RFC
                3264, June 2002.

  [RFC3267]     Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
                "Real-Time Transport Protocol (RTP) Payload Format and
                File Storage Format for the Adaptive Multi-Rate (AMR)
                and Adaptive Multi-Rate Wideband (AMR-WB) Audio
                Codecs", RFC 3267, June 2002.

  [RFC3555]     Casner, S. and P. Hoschka, "MIME Type Registration of
                RTP Payload Formats", RFC 3555, July 2003.











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10.  Informative References

  [RFC2736]     Handley, M. and C. Perkins, "Guidelines for Writers of
                RTP Payload Format Specifications", BCP 36, RFC 2736,
                December 1999.

  [RFC3551]     Schulzrinne, H. and S. Casner, "RTP Profile for Audio
                and Video Conferences with Minimal Control", STD 65,
                RFC 3551, July 2003.

  [1994AC3]     Todd, C., et al., "AC-3: Flexible Perceptual Coding for
                Audio Transmission and Storage," Preprint 3796,
                Presented at the 96th Convention of the Audio
                Engineering Society, May 1994.

  [1996AC3]     Fielder, L., et al., "AC-2 and AC-3: Low-Complexity
                Transform-Based Audio Coding," Collected Papers on
                Digital Audio Bit-Rate Reduction, pp. 54-72, Audio
                Engineering Society, September 1996.

  [RFC3711]     Baugher, M., et al., "The Secure Real-time Transport
                Protocol (SRTP)", RFC 3711, March 2004.

  [DRAFT-FREED] Freed, N. and Klensin, J., "Media Type Specifications
                and Registration Procedures", Work in Progress, April
                2005.

























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Authors' Addresses

  Brian Link
  Dolby Laboratories
  100 Potrero Ave
  San Francisco, CA 94103

  Phone: +1 415 558 0200
  EMail: [email protected]


  Todd Hager
  Dolby Laboratories
  100 Potrero Ave
  San Francisco, CA 94103

  Phone: +1 415 558 0136
  EMail: [email protected]


  Jason Flaks
  Microsoft Corporation
  1 Microsoft Way
  Redmond, WA 98052

  Phone: +1 425 722 2543
  EMail: [email protected]
























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Full Copyright Statement

  Copyright (C) The Internet Society (2005).

  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 an
  "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
  OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
  ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
  INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
  INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
  WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

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  [email protected].

Acknowledgement

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







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