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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.
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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:
- 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"
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:
- (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".)
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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:
- 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
- "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
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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.
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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
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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.)
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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):
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.
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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".
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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.
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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/
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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:
- "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 size (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
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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();
}
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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.
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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) {
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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;
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}
} 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:
- "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
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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
frame
- "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 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;
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iccLastSeen = icc;
aduFrameBuffer[ii] = aduFrame;
}
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RFC 3119 Loss-Tolerant RTP Payload Format for MP3 Audio June 2001
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