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Versions: (draft-finlayson-rtp-mp3) 00 01 02 03 04 05 RFC 5219

Network Working Group                                   Ross Finlayson
INTERNET-DRAFT                                          LIVE.COM
Category: Standards Track                               October 5, 2004
Expires: February 5, 2005

             A More Loss-Tolerant RTP Payload Format for MP3 Audio

Status of this Memo

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    This document describes a RTP (Real-Time Protocol) payload format for
    transporting MPEG (Moving Picture Experts Group) 1 or 2, layer III
    audio (commonly known as "MP3").  This format is an alternative to
    that described in RFC 2250, and performs better if there is packet
    loss.  (This document updates RFC 3119, correcting typographical
    errors in the "SDP usage" section and pseudo-code appendices.)

1. Terminology
    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
    document are to be interpreted as described in RFC 2119 [1].

2. Introduction

    While the RTP payload format defined in RFC 2250 [2] is generally
    applicable to all forms of MPEG audio or video, it is sub-optimal for
    MPEG-1 or 2, layer III audio (commonly known as "MP3").  The reason
    for this is that an MP3 frame is not a true "Application Data Unit" -
    it contains a back-pointer to data in earlier frames, and so cannot
    be decoded independently of these earlier frames.  Because RFC 2250
    defines that packet boundaries coincide with frame boundaries, it
    handles packet loss inefficiently when carrying MP3 data.  The loss
    of an MP3 frame will render some data in previous (or future) frames
    useless, even if they are received without loss.

    In this document we define an alternative RTP payload format for MP3
    audio.  This format uses a data-preserving rearrangement of the
    original MPEG frames, so that packet boundaries now coincide with
    true MP3 "Application Data Units", which can also (optionally) be
    rearranged in an interleaving pattern.  This new format is therefore
    more data-efficient than RFC 2250 in the face of packet loss.

3. The Structure of MP3 Frames

    In this section we give a brief overview of the structure of a MP3
    frame.  (For more detailed description, see the MPEG-1 audio [3] and
    MPEG-2 audio [4] specifications.)

    Each MPEG audio frame begins with a 4-byte header.  Information
    defined by this header includes:

    -  Whether the audio is MPEG-1 or MPEG-2.
    -  Whether the audio is layer I, II, or III.
       (The remainder of this document assumes layer III, i.e., "MP3"
    -  Whether the audio is mono or stereo.
    -  Whether or not there is a 2-byte CRC field following the header.
    -  (indirectly) The size of the frame.

    The following structures appear after the header:

    -  (optionally) A 2-byte CRC field
    -  A "side info" structure.  This has the following length:
       -  32 bytes for MPEG-1 stereo
       -  17 bytes for MPEG-1 mono, or for MPEG-2 stereo
       -  9 bytes for MPEG-2 mono
    -  Encoded audio data, plus optional ancillary data (filling out the
       rest of the frame)

    For the purpose of this document, the "side info" structure is the
    most important, because it defines the location and size of the
    "Application Data Unit" (ADU) that an MP3 decoder will process.  In
    particular, the "side info" structure defines:

    -  "main_data_begin": This is a back-pointer (in bytes) to the start
       of the ADU.  The back-pointer is counted from the beginning of the
       frame, and counts only encoded audio data and any ancillary data
       (i.e., ignoring any header, CRC, or "side info" fields).

    An MP3 decoder processes each ADU independently.  The ADUs will
    generally vary in length, but their average length will, of course,
    be that of the of the MP3 frames (minus the length of the header,
    CRC, and "side info" fields).  (In MPEG literature, this ADU is
    sometimes referred to as a "bit reservoir".)

4. A New Payload Format

    As noted in [5], a payload format should be designed so that packet
    boundaries coincide with "codec frame boundaries" - i.e., with ADUs.
    In the RFC 2250 payload format for MPEG audio [2], each RTP packet
    payload contains MP3 frames.  In this new payload format for MP3
    audio, however, each RTP packet payload contains "ADU frames", each
    preceded by an "ADU descriptor".

4.1 ADU frames

    An "ADU frame" is defined as:

       -  The 4-byte MPEG header
          (the same as the original MP3 frame, except that the first 11
          bits are (optionally) replaced by an "Interleaving Sequence
          Number", as described in section 7 below)
       -  The optional 2-byte CRC field
          (the same as the original MP3 frame)
       -  The "side info" structure
          (the same as the original MP3 frame)
       -  The complete sequence of encoded audio data (and any ancillary
          data) for the ADU (i.e., running from the start of this MP3
          frame's "main_data_begin" back-pointer, up to the start of the
          next MP3 frame's back-pointer)

4.2 ADU descriptors

    Within each RTP packet payload, each "ADU frame" is preceded by a 1
    or 2-byte "ADU descriptor", which gives the size of the ADU, and
    indicates whether or not this packet's data is a continuation of the
    previous packet's data.  (This occurs only when a single "ADU
    descriptor"+"ADU frame" is too large to fit within a RTP packet.)

    An ADU descriptor consists of the following fields

    -  "C": Continuation flag (1 bit):  1 if the data following the ADU
            descriptor is a continuation of an ADU frame that was too
            large to fit within a single RTP packet; 0 otherwise.
    -  "T": Descriptor Type flag (1 bit):
            0 if this is a 1-byte ADU descriptor;
            1 if this is a 2-byte ADU descriptor.
    -  "ADU size" (6 or 14 bits):
            The size (in bytes) of the ADU frame that will follow this
            ADU descriptor (i.e., NOT including the size of the
            descriptor itself).  A 2-byte ADU descriptor (with a 14-bit
            "ADU size" field) is used for ADU frame sizes of 64 bytes or
            more.  For smaller ADU frame sizes, senders MAY alternatively
            use a 1-byte ADU descriptor (with a 6-bit "ADU size" field).
            Receivers MUST be able to accept an ADU descriptor of either

    Thus, a 1-byte ADU descriptor is formatted as follows:

           0 1 2 3 4 5 6 7
          |C|0|  ADU size |

    and a 2-byte ADU descriptor is formatted as follows:

           0                   1
           0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
          |C|1|     ADU size (14 bits)    |

4.3 Packing rules

    Each RTP packet payload begins with a "ADU descriptor", followed by
    "ADU frame" data.  Normally, this "ADU descriptor"+"ADU frame" will
    fit completely within the RTP packet.  In this case, more than one
    successive "ADU descriptor"+"ADU frame" MAY be packed into a single
    RTP packet, provided that they all fit completely.

    If, however, a single "ADU descriptor"+"ADU frame" is too large to
    fit within an RTP packet, then the "ADU frame" is split across two or
    more successive RTP packets.  Each such packet begins with an ADU
    descriptor.  The first packet's descriptor has a "C" (continuation)
    flag of 0; the following packets' descriptors each have a "C" flag of
    1.  Each descriptor, in this case, has the same "ADU size" value: the
    size of the entire "ADU frame" (not just the portion that will fit
    within a single RTP packet).  Each such packet (even the last one)
    contains only one "ADU descriptor".

4.4 RTP header fields

       Payload Type: The (static) payload type 14 that was defined for
          MPEG audio [6] MUST NOT be used.  Instead, a different, dynamic
          payload type MUST be used - i.e., one in the range [96,127].

       M bit: This payload format defines no use for this bit.  Senders
          SHOULD set this bit to zero in each outgoing packet.

       Timestamp: This is a 32-bit 90 kHz timestamp, representing the
          presentation time of the first ADU packed within the packet.

4.5 Handling received data

    Note that no information is lost by converting a sequence of MP3
    frames to a corresponding sequence of "ADU frames", so a receiving
    RTP implementation can either feed the ADU frames directly to an
    appropriately modified MP3 decoder, or convert them back into a
    sequence of MP3 frames, as described in appendix A.2 below.

5. Handling Multiple MPEG Audio Layers

    The RTP payload format described here is intended only for MPEG-1 or
    2, layer III audio ("MP3").  In contrast, layer I and layer II frames
    are self-contained, without a back-pointer to earlier frames.
    However, it is possible (although unusual) for a sequence of audio
    frames to consist of a mixture of layer III frames and layer I or II
    frames.  When such a sequence is transmitted, only layer III frames
    are converted to ADUs; layer I or II frames are sent 'as is' (except
    for the prepending of an "ADU descriptor").  Similarly, the receiver
    of a sequence of frames - using this payload format - leaves layer I
    and II frames untouched (after removing the prepended "ADU
    descriptor), but converts layer III frames from "ADU frames" to
    regular MP3 frames.  (Recall that each frame's layer is identified
    from its 4-byte MPEG header.)

    If you are transmitting a stream consisting *only* of layer I or
    layer II frames (i.e., without any MP3 data), then there is no
    benefit to using this payload format, *unless* you are using the
    interleaving mechanism described in section 7 below.

6. Frame Packetizing and Depacketizing

    The transmission of a sequence of MP3 frames takes the following

          MP3 frames
                  -1-> ADU frames
                      -2-> interleaved ADU frames
                            -3-> RTP packets

    Step 1, the conversion of a sequence of MP3 frames to a corresponding
    sequence of ADU frames, takes place as described in sections 3 and
    4.1 above.  (Note also the pseudo-code in appendix A.1.)

    Step 2 is the reordering of the sequence of ADU frames in an
    (optional) interleaving pattern, prior to packetization, as described
    in section 7 below.  (Note also the pseudo-code in appendix B.1.)
    Interleaving helps reduce the effect of packet loss, by distributing
    consecutive ADU frames over non-consecutive packets.  (Note that
    because of the back-pointer in MP3 frames, interleaving can be
    applied - in general - only to ADU frames.  Thus, interleaving was
    not possible for RFC 2250.)

    Step 3 is the packetizing of a sequence of (interleaved) ADU frames
    into RTP packets - as described in section 4.3 above.  Each packet's
    RTP timestamp is the presentation time of the first ADU that is
    packed within it.  Note that, if interleaving was done in step 2, the
    RTP timestamps on outgoing packets will not necessarily be
    monotonically nondecreasing.

    Similarly, a sequence of received RTP packets is handled as follows:

          RTP packets
                -4-> RTP packets ordered by RTP sequence number
                      -5-> interleaved ADU frames
                            -6-> ADU frames
                                  -7-> MP3 frames

    Step 4 is the usual sorting of incoming RTP packets using the RTP
    sequence number.

    Step 5 is the depacketizing of ADU frames from RTP packets - i.e.,
    the reverse of step 3.  As part of this process, a receiver uses the
    "C" (continuation) flag in the ADU descriptor to notice when an ADU
    frame is split over more than one packet (and to discard the ADU
    frame entirely if one of these packets is lost).

    Step 6 is the rearranging of the sequence of ADU frames back to its
    original order (except for ADU frames missing due to packet loss), as
    described in section 7 below.  (Note also the pseudo-code in appendix

    Step 7 is the conversion of the sequence of ADU frames into a
    corresponding sequence of MP3 frames - i.e., the reverse of step 1.
    (Note also the pseudo-code in appendix A.2.)  With an appropriately
    modified MP3 decoder, an implementation may omit this step; instead,
    it could feed ADU frames directly to the (modified) MP3 decoder.

7. ADU Frame Interleaving

    In MPEG audio frames (MPEG-1 or 2; all layers) the high-order 11 bits
    of the 4-byte MPEG header ('syncword') are always all-one (i.e.,
    0xFFE).  When reordering a sequence of ADU frames for transmission,
    we reuse these 11 bits as an "Interleaving Sequence Number" (ISN).
    (Upon reception, they are replaced with 0xFFE once again.)

    The structure of the ISN is (a,b), where:

          - a == bits 0-7:      8-bit Interleave Index (within Cycle)
          - b == bits 8-10:     3-bit Interleave Cycle Count

    I.e., the 4-byte MPEG header is reused as follows:

        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
       |Interleave Idx |CycCt|   The rest of the original MPEG header  |

    Example: Consider the following interleave cycle (of size 8):
    (This particular pattern has the property that any loss of up to four
    consecutive ADUs in the interleaved stream will lead to a
    deinterleaved stream with no gaps greater than one.)
    This produces the following sequence of ISNs:

    (1,0) (3,0) (5,0) (7,0) (0,0) (2,0) (4,0) (6,0) (1,1) (3,1)
    (5,1) etc.

    So, in this example, a sequence of ADU frames

    f0 f1 f2 f3 f4 f5 f6 f7 f8 f9 (etc.)

    would get reordered, in step 2, into:

    (1,0)f1 (3,0)f3 (5,0)f5 (7,0)f7 (0,0)f0 (2,0)f2 (4,0)f4 (6,0)f6
    (1,1)f9 (3,1)f11 (5,1)f13 (etc.)

    and the reverse reordering (along with replacement of the 0xFFE)
    would occur upon reception.

    The reason for breaking the ISN into "Interleave Cycle Count" and
    "Interleave Index" (rather than just treating it as a single 11-bit
    counter) is to give receivers a way of knowing when an ADU frame
    should be 'released' to the ADU->MP3 conversion process (step 7
    above), rather than waiting for more interleaved ADU frames to
    arrive.  E.g., in the example above, when the receiver sees a frame
    with ISN (<something>,1), it knows that it can release all
    previously-seen frames with ISN (<something>,0), even if some other
    (<something>,0) frames remain missing due to packet loss.  A 8-bit
    Interleave Index allows interleave cycles of size up to 256.

    The choice of an interleaving order can be made independently of RTP
    packetization.  Thus, a simple implementation could choose an
    interleaving order first, reorder the ADU frames accordingly (step
    2), then simply pack them sequentially into RTP packets (step 3).
    However, the size of ADU frames - and thus the number of ADU frames
    that will fit in each RTP packet - will typically vary in size, so a
    more optimal implementation would combine steps 2 and 3, by choosing
    an interleaving order that better reflected the number of ADU frames
    packed within each RTP packet.

    Each receiving implementation of this payload format MUST recognize
    the ISN and be able to perform deinterleaving of incoming ADU frames
    (step 6).  However, a sending implementation of this payload format
    MAY choose not to perform interleaving - i.e., by omitting step 2.
    In this case, the high-order 11 bits in each 4-byte MPEG header would
    remain at 0xFFE.  Receiving implementations would thus see a sequence
    of identical ISNs (all 0xFFE).  They would handle this in the same
    way as if the Interleave Cycle Count changed with each ADU frame, by
    simply releasing the sequence of incoming ADU frames sequentially to
    the ADU->MP3 conversion process (step 7), without reordering.  (Note
    also the pseudo-code in appendix B.2.)

8. MIME registration

[Note to RFC Editor: Please replace "XXXX" with this document's
RFC number, when it is assigned.]

       MIME media type name: audio

       MIME subtype: mpa-robust

       Required parameters: none

       Optional parameters: none

       Encoding considerations:
          This type is defined only for transfer via RTP, as specified in
          "RFC XXXX".

       Security considerations:
          See the "Security Considerations" section of
          "RFC XXXX".

       Interoperability considerations:
          This encoding is incompatible with both the "audio/mpa"
          and "audio/mpeg" mime types.

       Published specification:
          The ISO/IEC MPEG-1 [3] and MPEG-2 [4] audio specifications,
          and "RFC XXXX".

       Applications which use this media type:
          Audio streaming tools (transmitting and receiving)

       Additional information: none

       Person & email address to contact for further information:
          Ross Finlayson
          finlayson (at) live.com

       Intended usage: COMMON

       Author/Change controller:
          Author: Ross Finlayson
          Change controller: IETF AVT Working Group

9. SDP usage

    When conveying information by SDP [7], the encoding name SHALL be
    "mpa-robust" (the same as the MIME subtype).  An example of the media
    representation in SDP is:

          m=audio 49000 RTP/AVP 121
          a=rtpmap:121 mpa-robust/90000

10. Security Considerations

    If a session using this payload format is being encrypted, and
    interleaving is being used, then the sender SHOULD ensure that any
    change of encryption key coincides with a start of a new interleave
    cycle.  Apart from this, the security considerations for this payload
    format are identical to those noted for RFC 2250 [2].

11. Acknowledgements

    The suggestion of adding an interleaving option (using the first bits
    of the MPEG 'syncword' - which would otherwise be all-ones - as an
    interleaving index) is due to Dave Singer and Stefan Gewinner.  In
    addition, Dave Singer provided valuable feedback that helped clarify
    and improve the description of this payload format.  Feedback from
    Chris Sloan led to the addition of an "ADU descriptor" preceding each
    ADU frame in the RTP packet.

12. Normative References

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

    [2] Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, "RTP
        Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998.

    [3] ISO/IEC International Standard 11172-3; "Coding of moving
        pictures and associated audio for digital storage media up to
        about 1,5 Mbits/s - Part 3: Audio", 1993.

    [4] ISO/IEC International Standard 13818-3; "Generic coding of moving
        pictures and associated audio information - Part 3: Audio", 1998.

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

    [6] Schulzrinne, H., "RTP Profile for Audio and Video Conferences
        with Minimal Control", RFC 1890, January 1996.

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

13. Author's Address

    Ross Finlayson,
    Live Networks, Inc. (LIVE.COM)
    650 Castro St., suite 120-196
    Mountain View, CA 94041

    EMail: finlayson (at) live.com
    WWW: http://www.live.com/

14. IPR Notice

    The IETF takes no position regarding the validity or scope of any
    Intellectual Property Rights or other rights that might be claimed
    to pertain to the implementation or use of the technology
    described in this document or the extent to which any license
    under such rights might or might not be available; nor does it
    represent that it has made any independent effort to identify any
    such rights.  Information on the procedures with respect to rights
    in RFC documents can be found in BCP 78 and BCP 79.

    Copies of IPR disclosures made to the IETF Secretariat and any
    assurances of licenses to be made available, or the result of an
    attempt made to obtain a general license or permission for the use
    of such proprietary rights by implementers or users of this
    specification can be obtained from the IETF on-line IPR repository
    at http://www.ietf.org/ipr.

    The IETF invites any interested party to bring to its attention
    any copyrights, patents or patent applications, or other
    proprietary rights that may cover technology that may be required
    to implement this standard.  Please address the information to the
    IETF at ietf-ipr@ietf.org.

15. Copyright Notice

    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 an

Appendix A. Translating Between "MP3 Frames" and "ADU Frames"

    The following 'pseudo code' describes how a sender using this payload
    format can translate a sequence of regular "MP3 Frames" to "ADU
    Frames", and how a receiver can perform the reverse translation: from
    "ADU Frames" to "MP3 Frames".

    We first define the following abstract data structures:

    -  "Segment": A record that represents either a "MP3 Frame" or an
       "ADU Frame".  It consists of the following fields:
       -  "header": the 4-byte MPEG header
       -  "headerSize": a constant (== 4)
       -  "sideInfo": the 'side info' structure, *including* the optional
          2-byte CRC field, if present
       -  "sideInfoSize": the size (in bytes) of the above structure
       -  "frameData": the remaining data in this frame
       -  "frameDataSize": the size (in bytes) of the above data
       -  "backpointer": the value (expressed in bytes) of the
           backpointer for this frame
       -  "aduDataSize": the size (in bytes) of the ADU associated with
          this frame.  (If the frame is already an "ADU Frame", then
          aduDataSize == frameDataSize)
       -  "mp3FrameSize": the total size (in bytes) that this frame would
          have if it were a regular "MP3 Frame".  (If it is already a
          "MP3 Frame", then mp3FrameSize == headerSize + sideInfoSize +
          frameDataSize) Note that this size can be derived completely
          from "header".

    -  "SegmentQueue": A FIFO queue of "Segment"s, with operations
       -  void enqueue(Segment)
       -  Segment dequeue()
       -  Boolean isEmpty()
       -  Segment head()
       -  Segment tail()
       -  Segment previous(Segment):  returns the segment prior to a
          given one
       -  Segment next(Segment): returns the segment after a given one
       -  unsigned totalDataSize(): returns the sum of the
          "frameDataSize" fields of each entry in the queue

A.1 Converting a sequence of "MP3 Frames" to a sequence of "ADU Frames":

SegmentQueue pendingMP3Frames; // initially empty
while (1) {
         // Enqueue new MP3 Frames, until we have enough data to generate
         // the ADU for a frame:
         do {
                 int totalDataSizeBefore
                         = pendingMP3Frames.totalDataSize();

                 Segment newFrame = 'the next MP3 Frame';

                 int totalDataSizeAfter
                         = pendingMP3Frames.totalDataSize();
         } while (totalDataSizeBefore < newFrame.backpointer ||
                   totalDataSizeAfter < newFrame.aduDataSize);

         // We now have enough data to generate the ADU for the most
         // recently enqueued frame (i.e., the tail of the queue).
         // (The earlier frames in the queue - if any - must be
         // discarded, as we don't have enough data to generate
         // their ADUs.)
         Segment tailFrame = pendingMP3Frames.tail();

         // Output the header and side info:

         // Go back to the frame that contains the start of our ADU data:
         int offset = 0;
         Segment curFrame = tailFrame;
         int prevBytes = tailFrame.backpointer;
         while (prevBytes > 0) {
                 curFrame = pendingMP3Frames.previous(curFrame);
                 int dataHere = curFrame.frameDataSize;
                 if (dataHere < prevBytes) {
                         prevBytes -= dataHere;
                 } else {
                         offset = dataHere - prevBytes;

         // Dequeue any frames that we no longer need:
         while (pendingMP3Frames.head() != curFrame) {

         // Output, from the remaining frames, the ADU data that we want:
         int bytesToUse = tailFrame.aduDataSize;
         while (bytesToUse > 0) {
                 int dataHere = curFrame.frameDataSize - offset;
                 int bytesUsedHere
                         = dataHere < bytesToUse ? dataHere : bytesToUse;

                 output("bytesUsedHere" bytes from curFrame.frameData,
                         starting from "offset");

                 bytesToUse -= bytesUsedHere;
                 offset = 0;
                 curFrame = pendingMP3Frames.next(curFrame);

A.2 Converting a sequence of "ADU Frames" to a sequence of "MP3 Frames":

SegmentQueue pendingADUFrames; // initially empty
while (1) {
         while (needToGetAnADU()) {
                 Segment newADU = 'the next ADU Frame';



Boolean needToGetAnADU() {
         // Checks whether we need to enqueue one or more new ADUs before
         // we have enough data to generate a frame for the head ADU.
         Boolean needToEnqueue = True;

         if (!pendingADUFrames.isEmpty()) {
                 Segment curADU = pendingADUFrames.head();
                 int endOfHeadFrame = curADU.mp3FrameSize
                         - curADU.headerSize - curADU.sideInfoSize;
                 int frameOffset = 0;

                 while (1) {
                         int endOfData = frameOffset
                                 - curADU.backpointer +
                         if (endOfData >= endOfHeadFrame) {
                                 // We have enough data to generate a
                                 // frame.
                                 needToEnqueue = False;

                         frameOffset += curADU.mp3FrameSize
                                 - curADU.headerSize
                                 - curADU.sideInfoSize;
                         if (curADU == pendingADUFrames.tail()) break;
                         curADU = pendingADUFrames.next(curADU);

     return needToEnqueue;

void generateFrameFromHeadADU() {
         Segment curADU = pendingADUFrames.head();

         // Output the header and side info:

         // Begin by zeroing out the rest of the frame, in case the ADU
         // data doesn't fill it in completely:
         int endOfHeadFrame = curADU.mp3FrameSize
                 - curADU.headerSize - curADU.sideInfoSize;
         output("endOfHeadFrame" zero bytes);

         // Fill in the frame with appropriate ADU data from this and
         // subsequent ADUs:
         int frameOffset = 0;
         int toOffset = 0;

         while (toOffset < endOfHeadFrame) {
                 int startOfData = frameOffset - curADU.backpointer;
                 if (startOfData > endOfHeadFrame) {
                         break; // no more ADUs are needed
                 int endOfData = startOfData + curADU.aduDataSize;
                 if (endOfData > endOfHeadFrame) {
                         endOfData = endOfHeadFrame;

                 int fromOffset;
                 if (startOfData <= toOffset) {
                         fromOffset = toOffset - startOfData;
                         startOfData = toOffset;
                         if (endOfData < startOfData) {
                                 endOfData = startOfData;
                 } else {
                         fromOffset = 0;

                         // leave some zero bytes beforehand:
                         toOffset = startOfData;

                 int bytesUsedHere = endOfData - startOfData;
                 output(starting at offset "toOffset, "bytesUsedHere"
                         bytes from "&curADU.frameData[fromOffset]");
                 toOffset += bytesUsedHere;

                 frameOffset += curADU.mp3FrameSize
                         - curADU.headerSize - curADU.sideInfoSize;
                 curADU = pendingADUFrames.next(curADU);


void insertDummyADUsIfNecessary() {
         // The tail segment (ADU) is assumed to have been recently
         // enqueued.  If its backpointer would overlap the data
         // of the previous ADU, then we need to insert one or more
         // empty, 'dummy' ADUs ahead of it.  (This situation should
         // occur only if an intermediate ADU was missing - e.g., due
         // to packet loss.)
         while (1) {
                 Segment tailADU = pendingADUFrames.tail();
                 int prevADUend; // relative to the start of the tail ADU

                 if (pendingADUFrames.head() != tailADU) {
                         // there is a previous ADU
                         Segment prevADU
                                 = pendingADUFrames.previous(tailADU);
                                 = prevADU.mp3FrameSize +
                                   - prevADU.headerSize
                                   - prevADU.sideInfoSize;
                         if (prevADU.aduDataSize > prevADUend) {
                                 // this shouldn't happen if the previous
                                 // ADU was well-formed
                                 prevADUend = 0;
                         } else {
                                 prevADUend -= prevADU.aduDataSize;
                 } else {
                         prevADUend = 0;

                 if (tailADU.backpointer > prevADUend) {
                    // Insert a 'dummy' ADU in front of the tail.
                    // This ADU can have the same "header" (and thus
                    // "mp3FrameSize") as the tail ADU, but should
                    // have a "backpointer" of "prevADUend", and
                    // an "aduDataSize" of zero.  The simplest
                    // way to do this is to copy the "sideInfo" from
                    // the tail ADU, replace the value of
                    // "main_data_begin" with "prevADUend", and set
                    // all of the "part2_3_length" fields to zero.
                 } else {
                         break; // no more dummy ADUs need to be inserted

Appendix B: Interleaving and Deinterleaving

    The following 'pseudo code' describes how a sender can reorder a
    sequence of "ADU Frames" according to an interleaving pattern (step
    2), and how a receiver can perform the reverse reordering (step 6).

B.1 Interleaving a sequence of "ADU Frames":

    We first define the following abstract data structures:

    -  "interleaveCycleSize": an integer in the range [1,256] -
       "interleaveCycle": an array, of size "interleaveCycleSize",
       containing some permutation of the integers from the set [0 ..
       e.g., if "interleaveCycleSize" == 8, "interleaveCycle" might
       contain: 1,3,5,7,0,2,4,6
    -  "inverseInterleaveCycle": an array containing the inverse of the
       permutation in "interleaveCycle" - i.e., such that
       interleaveCycle[inverseInterleaveCycle[i]] == i
    -  "ii": the current Interleave Index (initially 0)
    -  "icc": the current Interleave Cycle Count (initially 0)
    -  "aduFrameBuffer": an array, of size "interleaveCycleSize", of ADU
       Frames that are awaiting packetization

while (1) {
         int positionOfNextFrame = inverseInterleaveCycle[ii];
         aduFrameBuffer[positionOfNextFrame] = the next ADU frame;
         replace the high-order 11 bits of this frame's MPEG header
             with (ii,icc);
                 // Note: Be sure to leave the remaining 21 bits as is
         if (++ii == interleaveCycleSize) {
                 // We've finished this cycle, so pass all
                 // pending frames to the packetizing step
                 for (int i = 0; i < interleaveCycleSize; ++i) {
                         pass aduFrameBuffer[i] to the packetizing step;

                 ii = 0;
                 icc = (icc+1)%8;

B.2 Deinterleaving a sequence of (interleaved) "ADU Frames":

    We first define the following abstract data structures:

    -  "ii": the Interleave Index from the current incoming ADU frame
    -  "icc": the Interleave Cycle Count from the current incoming ADU
    -  "iiLastSeen": the most recently seen Interleave Index (initially,
       some integer *not* in the range [0,255])
    -  "iccLastSeen": the most recently seen Interleave Cycle Count
       (initially, some integer *not* in the range [0,7])
    -  "aduFrameBuffer": an array, of size 256, of (pointers to) ADU
       Frames that have just been depacketized (initially, all entries
       are NULL)

while (1) {
         aduFrame = the next ADU frame from the depacketizing step;
         (ii,icc) = "the high-order 11 bits of aduFrame's MPEG header";
         "the high-order 11 bits of aduFrame's MPEG header" = 0xFFE;
                 // Note: Be sure to leave the remaining 21 bits as is

         if (icc != iccLastSeen || ii == iiLastSeen) {
                 // We've started a new interleave cycle
                 // (or interleaving was not used).  Release all
                 // pending ADU frames to the ADU->MP3 conversion step:
                 for (int i = 0; i < 256; ++i) {
                         if (aduFrameBuffer[i] != NULL) {
                                 release aduFrameBuffer[i];
                                 aduFrameBuffer[i] = NULL;

         iiLastSeen = ii;
         iccLastSeen = icc;
         aduFrameBuffer[ii] = aduFrame;

Expires: February 5, 2005

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