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rfc:rfc3119

Network Working Group R. Finlayson Request for Comments: 3119 LIVE.COM Category: Standards Track June 2001

       A More Loss-Tolerant RTP Payload Format for MP3 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 (2001).  All Rights Reserved.

Abstract

 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.

1. 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.

Finlayson Standards Track [Page 1] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

2. 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:
  1. Whether the audio is MPEG 1 or MPEG 2.
  2. Whether the audio is layer I, II, or III.

(The remainder of this document assumes layer III, i.e., "MP3"

    frames)
 -  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:
  1. (optionally) A 2-byte CRC field
  2. A "side info" structure. This has the following length:
    1. 32 bytes for MPEG 1 stereo
    2. 17 bytes for MPEG 1 mono, or for MPEG 2 stereo
    3. 9 bytes for MPEG 2 mono
  3. 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:
  1. "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".)

Finlayson Standards Track [Page 2] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

3. 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".

3.1 ADU frames

 An "ADU frame" is defined as:
  1. 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 6 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)

3.2 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
  1. "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 frames sizes of 64 bytes or
         more.  For smaller ADU frame sizes, senders MAY alternatively

Finlayson Standards Track [Page 3] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

         use a 1-byte ADU descriptor (with a 6-bit "ADU size" field).
         Receivers MUST be able to accept an ADU descriptor of either
         size.
 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)    |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

3.3 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".

3.4 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.

Finlayson Standards Track [Page 4] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

3.5 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.

4. 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 consists *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.

5. Frame Packetizing and Depacketizing

 The transmission of a sequence of MP3 frames takes the following
 steps:
       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 2 and
 3.1 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 6 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

Finlayson Standards Track [Page 5] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

 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 3.3 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 6 below.  (Note also the pseudo-code in appendix
 B.2.)
 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.

6. 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.)

Finlayson Standards Track [Page 6] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

 The structure of the ISN is (a,b), where:
  1. a == bits 0-7: 8-bit Interleave Index (within Cycle)
  2. 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):
          1,3,5,7,0,2,4,6
 (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 [7].)  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.

Finlayson Standards Track [Page 7] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

 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.)

7. MIME registration

    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 3119".
    Security considerations:
       See the "Security Considerations" section of
       "RFC 3119".
    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 3119".

Finlayson Standards Track [Page 8] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

    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@live.com
    Intended usage: COMMON
    Author/Change controller:
       Author: Ross Finlayson
       Change controller: IETF AVT Working Group

8. SDP usage

 When conveying information by SDP [8], the encoding name SHALL be
 "mp3" (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

9. 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].

10. 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.

Finlayson Standards Track [Page 9] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

11. 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] Marshall Eubanks, personal communication, December 2000.
 [8] Handley, M. and V. Jacobson, "SDP: Session Description Protocol",
     RFC 2327, April 1998.

11. Author's Address

 Ross Finlayson,
 Live Networks, Inc. (LIVE.COM)
 EMail: finlayson@live.com
 WWW: http://www.live.com/

Finlayson Standards Track [Page 10] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

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:
  1. "Segment": A record that represents either a "MP3 Frame" or an

"ADU Frame". It consists of the following fields:

  1. "header": the 4-byte MPEG header
  2. "headerSize": a constant (== 4)
  3. "sideInfo": the 'side info' structure, *including* the optional

2-byte CRC field, if present

  1. "sideInfoSize": the size (in bytes) of the above structure
  2. "frameData": the remaining data in this frame
  3. "frameDataSize": the size (in bytes) of the above data
  4. "backpointer": the size (in bytes) of the backpointer for this

frame

  1. "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".
  1. "SegmentQueue": A FIFO queue of "Segment"s, with operations
    1. void enqueue(Segment)
    2. Segment dequeue()
    3. Boolean isEmpty()
    4. Segment head()
    5. Segment tail()
    6. Segment previous(Segment): returns the segment prior to a

given one

  1. Segment next(Segment): returns the segment after a given one
  2. unsigned totalDataSize(): returns the sum of the

"frameDataSize" fields of each entry in the queue

Finlayson Standards Track [Page 11] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

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';
              pendingMP3Frames.enqueue(newFrame);
              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:
      output(tailFrame.header);
      output(tailFrame.sideInfo);
      // 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;
                      break;
              }
      }
      // Dequeue any frames that we no longer need:
      while (pendingMP3Frames.head() != curFrame) {
              pendingMP3Frames.dequeue();
      }

Finlayson Standards Track [Page 12] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

      // 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'; pendingADUFrames.enqueue(newADU); insertDummyADUsIfNecessary(); } generateFrameFromHeadADU(); } 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 +
                                curADU.aduDataSize;
                      if (endOfData >= endOfHeadFrame) {
                              // We have enough data to generate a
                              // frame.

Finlayson Standards Track [Page 13] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

                              needToEnqueue = False;
                              break;
                      }
                      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:
      output(curADU.header);
      output(curADU.sideInfo);
      // 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) {

Finlayson Standards Track [Page 14] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

                              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);
      }
      pendingADUFrames.dequeue();

}

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);
                      prevADUend
                              = prevADU.mp3FrameSize +
                                prevADU.backpointer
                                - prevADU.headerSize
                                - curADU.sideInfoSize;
                      if (prevADU.aduDataSize > prevADUend) {
                              // this shouldn't happen if the previous
                              // ADU was well-formed
                              prevADUend = 0;
                      } else {
                              prevADUend -= prevADU.aduDataSize;

Finlayson Standards Track [Page 15] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

                      }
              } 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 an "aduDataSize" of zero.  The simplest
                      // way to do this is to copy the "sideInfo" from
                      // the tail ADU, and zero out the
                      // "main_data_begin" and all of the
                      // "part2_3_length" fields.
              } 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:
  1. "interleaveCycleSize": an integer in the range [1,256] -

"interleaveCycle": an array, of size "interleaveCycleSize",

    containing some permutation of the integers from the set [0 ..
    interleaveCycleSize-1]
    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

Finlayson Standards Track [Page 16] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

          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:
  1. "ii": the Interleave Index from the current incoming ADU frame
  2. "icc": the Interleave Cycle Count from the current incoming ADU

frame

  1. "iiLastSeen": the most recently seen Interleave Index (initially,

some integer *not* in the range [0,255])

  1. "iccLastSeen": the most recently seen Interleave Cycle Count

(initially, some integer *not* in the range [0,7])

  1. "aduFrameBuffer": an array, of size 32, 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 < 32; ++i) {
                      if (aduFrameBuffer[i] != NULL) {
                              release aduFrameBuffer[i];
                              aduFrameBuffer[i] = NULL;
                      }
              }
      }
      iiLastSeen = ii;

Finlayson Standards Track [Page 17] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

      iccLastSeen = icc;
      aduFrameBuffer[ii] = aduFrame;

}

Finlayson Standards Track [Page 18] RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001

Full Copyright Statement

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 This document and translations of it may be copied and furnished to
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 the copyright notice or references to the Internet Society or other
 Internet organizations, except as needed for the purpose of
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 followed, or as required to translate it into languages other than
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 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
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Acknowledgement

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

Finlayson Standards Track [Page 19]

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