This is a purely informative rendering of an RFC that includes verified errata. This rendering may not be used as a reference.
The following 'Verified' errata have been incorporated in this document:
EID 114
Network Working Group J. Sjoberg
Request for Comments: 4352 M. Westerlund
Category: Standards Track Ericsson
A. Lakaniemi
S. Wenger
Nokia
January 2006
RTP Payload Format for the
Extended Adaptive Multi-Rate Wideband (AMR-WB+) Audio Codec
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 (2006).
Abstract
This document specifies a Real-time Transport Protocol (RTP) payload
format for Extended Adaptive Multi-Rate Wideband (AMR-WB+) encoded
audio signals. The AMR-WB+ codec is an audio extension of the AMR-WB
speech codec. It encompasses the AMR-WB frame types and a number of
new frame types designed to support high-quality music and speech. A
media type registration for AMR-WB+ is included in this
specification.
Table of Contents
1. Introduction ....................................................3
2. Definitions .....................................................4
2.1. Glossary ...................................................4
2.2. Terminology ................................................4
3. Background of AMR-WB+ and Design Principles .....................4
3.1. The AMR-WB+ Audio Codec ....................................4
3.2. Multi-rate Encoding and Rate Adaptation ....................8
3.3. Voice Activity Detection and Discontinuous Transmission ....8
3.4. Support for Multi-Channel Session ..........................8
3.5. Unequal Bit-Error Detection and Protection .................9
3.6. Robustness against Packet Loss .............................9
3.6.1. Use of Forward Error Correction (FEC) ...............9
3.6.2. Use of Frame Interleaving ..........................10
3.7. AMR-WB+ Audio over IP Scenarios ...........................11
3.8. Out-of-Band Signaling .....................................11
4. RTP Payload Format for AMR-WB+ .................................12
4.1. RTP Header Usage ..........................................13
4.2. Payload Structure .........................................14
4.3. Payload Definitions .......................................14
4.3.1. Payload Header .....................................14
4.3.2. The Payload Table of Contents ......................15
4.3.3. Audio Data .........................................20
4.3.4. Methods for Forming the Payload ....................21
4.3.5. Payload Examples ...................................21
4.4. Interleaving Considerations ...............................24
4.5. Implementation Considerations .............................25
4.5.1. ISF Recovery in Case of Packet Loss ................26
4.5.2. Decoding Validation ................................28
5. Congestion Control .............................................28
6. Security Considerations ........................................28
6.1. Confidentiality ...........................................29
6.2. Authentication and Integrity ..............................29
7. Payload Format Parameters ......................................29
7.1. Media Type Registration ...................................30
7.2. Mapping Media Type Parameters into SDP ....................32
7.2.1. Offer-Answer Model Considerations ..................32
7.2.2. Examples ...........................................34
8. IANA Considerations ............................................34
9. Contributors ...................................................34
10. Acknowledgements ..............................................34
11. References ....................................................35
11.1. Normative References .....................................35
11.2. Informative References ...................................35
1. Introduction
This document specifies the payload format for packetization of
Extended Adaptive Multi-Rate Wideband (AMR-WB+) [1] encoded audio
signals into the Real-time Transport Protocol (RTP) [3]. The payload
format supports the transmission of mono or stereo audio, aggregating
multiple frames per payload, and mechanisms enhancing the robustness
of the packet stream against packet loss.
The AMR-WB+ codec is an extension of the Adaptive Multi-Rate Wideband
(AMR-WB) speech codec. New features include extended audio bandwidth
to enable high quality for non-speech signals (e.g., music), native
support for stereophonic audio, and the option to operate on, and
switch between, several internal sampling frequencies (ISFs). The
primary usage scenario for AMR-WB+ is the transport over IP.
Therefore, interworking with other transport networks, as discussed
for AMR-WB in [7], is not a major concern and hence not addressed in
this memo.
The expected key application for AMR-WB+ is streaming. To make the
packetization process on a streaming server as efficient as possible,
an octet-aligned payload format is desirable. Therefore, a
bandwidth-efficient mode (as defined for AMR-WB in [7]) is not
specified herein; the bandwidth savings of the bandwidth-efficient
mode would be very small anyway, since all extension frame types are
octet aligned.
The stereo encoding capability of AMR-WB+ renders the support for
multi-channel transport at RTP payload format level, as specified for
AMR-WB [7], obsolete. Therefore, this feature is not included in
this memo.
This specification does not include a definition of a file format for
AMR-WB+. Instead, it refers to the ISO-based 3GP file format [14],
which supports AMR-WB+ and provides all functionality required. The
3GP format also supports storage of AMR, AMR-WB, and many other
multi-media formats, thereby allowing synchronized playback.
The rest of the document is organized as follows: Background
information on the AMR-WB+ codec, and design principles, can be found
in Section 3. The payload format itself is specified in Section 4.
Sections 5 and 6 discuss congestion control and security
considerations, respectively. In Section 7, a media type
registration is provided.
2. Definitions
2.1. Glossary
3GPP - Third Generation Partnership Project
AMR - Adaptive Multi-Rate (Codec)
AMR-WB - Adaptive Multi-Rate Wideband (Codec)
AMR-WB+ - Extended Adaptive Multi-Rate Wideband (Codec)
CN - Comfort Noise
DTX - Discontinuous Transmission
FEC - Forward Error Correction
FT - Frame Type
ISF - Internal Sampling Frequency
SCR - Source-Controlled Rate Operation
SID - Silence Indicator (the frames containing only CN
parameters)
TFI - Transport Frame Index
TS - Timestamp
VAD - Voice Activity Detection
UED - Unequal Error Detection
UEP - Unequal Error Protection
2.2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [2].
3. Background of AMR-WB+ and Design Principles
The Extended Adaptive Multi-Rate Wideband (AMR-WB+) [1] audio codec
is designed to compress speech and audio signals at low bit-rate and
good quality. The codec is specified by the Third Generation
Partnership Project (3GPP). The primary target applications are 1)
the packet-switched streaming service (PSS) [13], 2) multimedia
messaging service (MMS) [18], and 3) multimedia broadcast and
multicast service (MBMS) [19]. However, due to its flexibility and
robustness, AMR-WB+ is also well suited for streaming services in
other highly varying transport environments, for example, the
Internet.
3.1. The AMR-WB+ Audio Codec
3GPP originally developed the AMR-WB+ audio codec for streaming and
messaging services in Global System for Mobile communications (GSM)
and third generation (3G) cellular systems. The codec is designed as
an audio extension of the AMR-WB speech codec. The extension adds
new functionality to the codec in order to provide high audio quality
for a wide range of signals including music. Stereophonic operation
has also been added. A new, high-efficiency hybrid stereo coding
algorithm enables stereo operation at bit-rates as low as 6.2 kbit/s.
The AMR-WB+ codec includes the nine frame types specified for AMR-WB,
extended by new bit-rates ranging from 5.2 to 48 kbit/s. The AMR-WB
frame types can employ only a 16000 Hz sampling frequency and operate
only on monophonic signals. The newly introduced extension frame
types, however, can operate at a number of internal sampling
frequencies (ISFs), both in mono and stereo. Please see Table 24 in
[1] for details. The output sampling frequency of the decoder is
limited to 8, 16, 24, 32, or 48 kHz.
An overview of the AMR-WB+ encoding operations is provided as
follows. The encoder receives the audio sampled at, for example, 48
kHz. The encoding process starts with pre-processing and resampling
to the user-selected ISF. The encoding is performed on equally sized
super-frames. Each super-frame corresponds to 2048 samples per
channel, at the ISF. The codec carries out a number of encoding
decisions for each super-frame, thereby choosing between different
encoding algorithms and block lengths, so as to achieve a fidelity-
optimized encoding adapted to the signal characteristics of the
source. The stereo encoding (if used) executes separately from the
monophonic core encoding, thus enabling the selection of different
combinations of core and stereo encoding rates. The resulting
encoded audio is produced in four transport frames of equal length.
Each transport frame corresponds to 512 samples at the ISF and is
individually usable by the decoder, provided that its position in the
super-frame structure is known.
The codec supports 13 different ISFs, ranging from 12.8 to 38.4 kHz,
as described by Table 24 of [1]. The high number of ISFs allows a
trade-off between the audio bandwidth and the target bit-rate. As
encoding is performed on 2048 samples at the ISF, the duration of a
super-frame and the effective bit-rate of the frame type in use
varies.
The ISF of 25600 Hz has a super-frame duration of 80 ms. This is the
'nominal' value used to describe the encoding bit-rates henceforth.
Assuming this normalization, the ISF selection results in bit-rate
variations from 1/2 up to 3/2 of the nominal bit-rate.
The encoding for the extension modes is performed as one monophonic
core encoding and one stereo encoding. The core encoding is executed
by splitting the monophonic signal into a lower and a higher
frequency band. The lower band is encoded employing either algebraic
code excited linear prediction (ACELP) or transform coded excitation
(TCX). This selection can be made once per transport frame, but must
obey certain limitations of legal combinations within the super-
frame. The higher band is encoded using a low-rate parametric
bandwidth extension approach.
The stereo signal is encoded employing a similar frequency band
decomposition; however, here the signal is divided into three bands
that are individually parameterized.
The total bit-rate produced by the extension is the result of the
combination of the encoder's core rate, stereo rate, and ISF. The
extension supports 8 different core encoding rates, producing bit-
rates between 10.4 and 24.0 kbit/s; see Table 22 in [1]. There are
16 stereo encoding rates generating bit-rates between 2.0 and 8.0
kbit/s; see Table 23 in [1]. The frame type uniquely identifies the
AMR-WB modes, 4 fixed extension rates (see below), 24 combinations of
core and stereo rates for stereo signals, and the 8 core rates for
mono signals, as listed in Table 25 in [1]. This implies that the
AMR-WB+ supports encoding rates between 10.4 and 32 kbit/s, assuming
an ISF of 25600 Hz.
Different ISFs allow for additional freedom in the produced bit-rates
and audio quality. The selection of an ISF changes the available
audio bandwidth of the reconstructed signal, and also the total bit-
rate. The bit-rate for a given combination of frame type and ISF is
determined by multiplying the frame type's bit-rate with the used
ISF's bit-rate factor; see Table 24 in [1].
The extension also has four frame types which have fixed ISFs.
Please see frame types 10-13 in Table 21 in [1]. These four pre-
defined frame types have a fixed input sampling frequency at the
encoder, which can be set at either 16 or 24 kHz. Like the AMR-WB
frame types, transport frames encoded utilizing these frame types
represent exactly 20 ms of the audio signal. However, they are also
part of 80 ms super-frames. Frame types 0-13 (AMR-WB and fixed
extension rates), as listed in Table 21 in [1], do not require an
explicit ISF indication. The other frame types, 14-47, require the
ISF employed to be indicated.
The 32 different frame types of the extension, in combination with 13
ISFs, allows for a great flexibility in bit-rate and selection of
desired audio quality. A number of combinations exist that produce
the same codec bit-rate. For example, a 32 kbit/s audio stream can
be produced by utilizing frame type 41 (i.e., 25.6 kbit/s) and the
ISF of 32kHz (5/4 * (19.2+6.4) = 32 kbit/s), or frame type 47 and the
ISF of 25.6 kHz (1 * (24 + 8) = 32 kbit/s). Which combination is
more beneficial for the perceived audio quality depends on the
content. In the above example, the first case provides a higher
audio bandwidth, while the second one spends the same number of bits
on somewhat narrower audio bandwidth but provides higher fidelity.
Encoders are free to select the combination they deem most
beneficial.
Since a transport frame always corresponds to 512 samples at the used
ISF, its duration is limited to the range 13.33 to 40 ms; see Table
1. An RTP Timestamp clock rate of 72000 Hz, as mandated by this
specification, results in AMR-WB+ transport frame lengths of 960 to
2880 timestamp ticks, depending solely on the selected ISF.
Index ISF Duration(ms) Duration(TS Ticks @ 72 kHz)
------------------------------------------------------
0 N/A 20 1440
1 12800 40 2880
2 14400 35.55 2560
3 16000 32 2304
4 17067 30 2160
5 19200 26.67 1920
6 21333 24 1728
7 24000 21.33 1536
8 25600 20 1440
9 28800 17.78 1280
10 32000 16 1152
11 34133 15 1080
12 36000 14.22 1024
13 38400 13.33 960
Table 1: Normative number of RTP Timestamp Ticks for each
Transport Frame depending on ISF (ISF and Duration in
ms are rounded)
The encoder is free to change both the ISF and the encoding frame
type (both mono and stereo) during a session. For the extension
frame types with index 10-13 and 16-47, the ISF and frame type
changes are constrained to occur at super-frame boundaries. This
implies that, for the frame types mentioned, the ISF is constant
throughout a super-frame. This limitation does not apply for frame
types with index 0-9, 14, and 15; i.e., the original AMR-WB frame
types.
A number of features of the AMR-WB+ codec require special
consideration from a transport point of view, and solutions that
could perhaps be viewed as unorthodox. First, there are constraints
on the RTP timestamping, due to the relationship of the frame
duration and the ISFs. Second, each frame of encoded audio must
maintain information about its frame type, ISF, and position in the
super-frame.
3.2. Multi-rate Encoding and Rate Adaptation
The multi-rate encoding capability of AMR-WB+ is designed to preserve
high audio quality under a wide range of bandwidth requirements and
transmission conditions.
AMR-WB+ enables seamless switching between frame types that use the
same number of audio channels and the same ISF. Every AMR-WB+ codec
implementation is required to support all frame types defined by the
codec and must be able to handle switching between any two frame
types. Switching between frame types employing a different number of
audio channels or a different ISF must also be supported, but it may
not be completely seamless. Therefore, it is recommended to perform
such switching infrequently and, if possible, during periods of
silence.
3.3. Voice Activity Detection and Discontinuous Transmission
AMR-WB+ supports the same algorithms as AMR-WB for voice activity
detection (VAD) and generation of comfort noise (CN) parameters
during silence periods. However, these functionalities can only be
used in conjunction with the AMR-WB frame types (FT=0-8). This
option allows reducing the number of transmitted bits and packets
during silence periods to a minimum. The operation of sending CN
parameters at regular intervals during silence periods is usually
called discontinuous transmission (DTX) or source controlled rate
(SCR) operation. The AMR-WB+ frames containing CN parameters are
called Silence Indicator (SID) frames. More details about the VAD
and DTX functionality are provided in [4] and [5].
3.4. Support for Multi-Channel Session
Some of the AMR-WB+ frame types support the encoding of stereophonic
audio. Because of this native support for a two-channel stereophonic
signal, it does not seem necessary to support multi-channel transport
with separate codec instances, as specified in the AMR-WB RTP payload
[7]. The codec has the capability of stereo to mono downmixing as
part of the decoding process. Thus, a receiver that is only capable
of playout of monophonic audio must still be able to decode and play
signals originally encoded and transmitted as stereo. However, to
avoid spending bits on a stereo encoding that is not going to be
utilized, a mechanism is defined in this specification to signal
mono-only audio.
3.5. Unequal Bit-Error Detection and Protection
The audio bits encoded in each AMR-WB frame are sorted according to
their different perceptual sensitivity to bit errors. In cellular
systems, for example, this property can be exploited to achieve
better voice quality, by using unequal error protection and detection
(UEP and UED) mechanisms. However, the bits of the extension frame
types of the AMR-WB+ codec do not have a consistent perceptual
significance property and are not sorted in this order. Thus, UEP or
UED is meaningless with the extension frame types. If there is a
need to use UEP or UED for AMR-WB frame types, it is recommended that
RFC 3267 [7] be used.
3.6. Robustness against Packet Loss
The payload format supports two mechanisms to improve robustness
against packet loss: simple forward error correction (FEC) and frame
interleaving.
3.6.1. Use of Forward Error Correction (FEC)
Generic forward error correction within RTP is defined, for example,
in RFC 2733 [11]. Audio redundancy coding is defined in RFC 2198
[12]. Either scheme can be used to add redundant information to the
RTP packet stream and make it more resilient to packet losses, at the
expense of a higher bit rate. Please see either RFC for a discussion
of the implications of the higher bit rate to network congestion.
In addition to these media-unaware mechanisms, this memo specifies an
AMR-WB+ specific form of audio redundancy coding, which may be
beneficial in terms of packetization overhead.
Conceptually, previously transmitted transport frames are aggregated
together with new ones. A sliding window is used to group the frames
to be sent in each payload. Figure 1 below shows an example.
--+--------+--------+--------+--------+--------+--------+--------+--
| f(n-2) | f(n-1) | f(n) | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
--+--------+--------+--------+--------+--------+--------+--------+--
<---- p(n-1) ---->
<----- p(n) ----->
<---- p(n+1) ---->
<---- p(n+2) ---->
<---- p(n+3) ---->
<---- p(n+4) ---->
Figure 1: An example of redundant transmission
Here, each frame is retransmitted once in the following RTP payload
packet. F(n-2)...f(n+4) denote a sequence of audio frames, and
p(n-1)...p(n+4) a sequence of payload packets.
The mechanism described does not require signaling at the session
setup. In other words, the audio sender can choose to use this
scheme without consulting the receiver. For a certain timestamp, the
receiver may receive multiple copies of a frame containing encoded
audio data or frames indicated as NO_DATA. The cost of this scheme
is bandwidth and the receiver delay necessary to allow the redundant
copy to arrive.
This redundancy scheme provides a functionality similar to the one
described in RFC 2198, but it works only if both original frames and
redundant representations are AMR-WB+ frames. When the use of other
media coding schemes is desirable, one has to resort to RFC 2198.
The sender is responsible for selecting an appropriate amount of
redundancy based on feedback about the channel conditions, e.g., in
the RTP Control Protocol (RTCP) [3] receiver reports. The sender is
also responsible for avoiding congestion, which may be exacerbated by
redundancy (see Section 5 for more details).
3.6.2. Use of Frame Interleaving
To decrease protocol overhead, the payload design allows several
audio transport frames to be encapsulated into a single RTP packet.
One of the drawbacks of such an approach is that in case of packet
loss several consecutive frames are lost. Consecutive frame loss
normally renders error concealment less efficient and usually causes
clearly audible and annoying distortions in the reconstructed audio.
Interleaving of transport frames can improve the audio quality in
such cases by distributing the consecutive losses into a number of
isolated frame losses, which are easier to conceal. However,
interleaving and bundling several frames per payload also increases
end-to-end delay and sets higher buffering requirements. Therefore,
interleaving is not appropriate for all use cases or devices.
Streaming applications should most likely be able to exploit
interleaving to improve audio quality in lossy transmission
conditions.
Note that this payload design supports the use of frame interleaving
as an option. The usage of this feature needs to be negotiated in
the session setup.
The interleaving supported by this format is rather flexible. For
example, a continuous pattern can be defined, as depicted in Figure
2.
--+--------+--------+--------+--------+--------+--------+--------+--
| f(n-2) | f(n-1) | f(n) | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
--+--------+--------+--------+--------+--------+--------+--------+--
[ P(n) ]
[ P(n+1) ] [ P(n+1) ]
[ P(n+2) ] [ P(n+2) ]
[ P(n+3) ] [P(
[ P(n+4) ]
Figure 2: An example of interleaving pattern that has constant delay
In Figure 2 the consecutive frames, denoted f(n-2) to f(n+4), are
aggregated into packets P(n) to P(n+4), each packet carrying two
frames. This approach provides an interleaving pattern that allows
for constant delay in both the interleaving and deinterleaving
processes. The deinterleaving buffer needs to have room for at least
three frames, including the one that is ready to be consumed. The
storage space for three frames is needed, for example, when f(n) is
the next frame to be decoded: since frame f(n) was received in packet
P(n+2), which also carried frame f(n+3), both these frames are stored
in the buffer. Furthermore, frame f(n+1) received in the previous
packet, P(n+1), is also in the deinterleaving buffer. Note also that
in this example the buffer occupancy varies: when frame f(n+1) is the
next one to be decoded, there are only two frames, f(n+1) and f(n+3),
in the buffer.
3.7. AMR-WB+ Audio over IP Scenarios
Since the primary target application for the AMR-WB+ codec is
streaming over packet networks, the most relevant usage scenario for
this payload format is IP end-to-end between a server and a terminal,
as shown in Figure 3.
+----------+ +----------+
| | IP/UDP/RTP/AMR-WB+ | |
| SERVER |<------------------------>| TERMINAL |
| | | |
+----------+ +----------+
Figure 3: Server to terminal IP scenario
3.8. Out-of-Band Signaling
Some of the options of this payload format remain constant throughout
a session. Therefore, they can be controlled/negotiated at the
session setup. Throughout this specification, these options and
variables are denoted as "parameters to be established through out-
of-band means". In Section 7, all the parameters are formally
specified in the form of media type registration for the AMR-WB+
encoding. The method used to signal these parameters at session
setup or to arrange prior agreement of the participants is beyond the
scope of this document; however, Section 7.2 provides a mapping of
the parameters into the Session Description Protocol (SDP) [6] for
those applications that use SDP.
4. RTP Payload Format for AMR-WB+
The main emphasis in the payload design for AMR-WB+ has been to
minimize the overhead in typical use cases, while providing full
flexibility with a slightly higher overhead. In order to keep the
specification reasonably simple, we refrained from defining frame-
specific parameters for each frame type. Instead, a few common
parameters were specified that cover all types of frames.
The payload format has two modes: basic mode and interleaved mode.
The main structural difference between the two modes is the extension
of the table of content entries with frame displacement fields when
operating in the interleaved mode. The basic mode supports
aggregation of multiple consecutive frames in a payload. The
interleaved mode supports aggregation of multiple frames that are
non-consecutive in time. In both modes it is possible to have frames
encoded with different frame types in the same payload. The ISF must
remain constant throughout the payload of a single packet.
The payload format is designed around the property of AMR-WB+ frames
that the frames are consecutive in time and share the same frame
duration (in the absence of an ISF change). This enables the
receiver to derive the timestamp for an individual frame within a
payload. In basic mode, the deriving process is based on the order
of frames. In interleaved mode, it is based on the compact
displacement fields. The frame timestamps are used to regenerate the
correct order of frames after reception, identify duplicates, and
detect lost frames that require concealment.
The interleaving scheme of this payload format is significantly more
flexible than the one specified in RFC 3267. The AMR and AMR-WB
payload format is only capable of using periodic patterns with frames
taken from an interleaving group at fixed intervals. The
interleaving scheme of this specification, in contrast, allows for
any interleaving pattern, as long as the distance in decoding order
between any two adjacent frames is not more than 256 frames. Note
that even at the highest ISF this allows an interleaving depth of up
to 3.41 seconds.
To allow for error resiliency through redundant transmission, the
periods covered by multiple packets MAY overlap in time. A receiver
MUST be prepared to receive any audio frame multiple times. All
redundantly sent frames MUST use the same frame type and ISF, and
MUST have the same RTP timestamp, or MUST be a NO_DATA frame (FT=15).
The payload consists of octet-aligned elements (header, ToC, and
audio frames). Only the audio frames for AMR-WB frame types (0-9)
require padding for octet alignment. If additional padding is
desired, then the P bit in the RTP header MAY be set, and padding MAY
be appended as specified in [3].
4.1. RTP Header Usage
The format of the RTP header is specified in [3]. This payload
format uses the fields of the header in a manner consistent with that
specification.
The RTP timestamp corresponds to the sampling instant of the first
sample encoded for the first frame in the packet. The timestamp
clock frequency SHALL be 72000 Hz. This frequency allows the frame
duration to be integer RTP timestamp ticks for the ISFs specified in
Table 1. It also provides reasonable conversion factors to the
input/output audio sampling frequencies supported by the codec. See
Section 4.3.2.3 for guidance on how to derive the RTP timestamp for
any audio frame beyond the first one.
The RTP header marker bit (M) SHALL be set to 1 whenever the first
frame carried in the packet is the first frame in a talkspurt (see
the definition of talkspurt in Section 4.1 of [9]). For all other
packets, the marker bit SHALL be set to zero (M=0).
The assignment of an RTP payload type for the format defined in this
memo is outside the scope of this document. The RTP profile in use
either assigns a static payload type or mandates binding the payload
type dynamically.
The media type parameter "channels" is used to indicate the maximum
number of channels allowed for a given payload type. A payload type
where channels=1 (mono) SHALL only carry mono content. A payload
type for which channels=2 has been declared MAY carry both mono and
stereo content. Note that this definition is different from the one
in RFC 3551 [9]. As mentioned before, the AMR-WB+ codec handles the
support of stereo content and the (eventual) downmixing of stereo to
mono internally. This makes it unnecessary to negotiate for the
number of channels for reasons other than bit-rate efficiency.
4.2. Payload Structure
The payload consists of a payload header, a table of contents, and
the audio data representing one or more audio frames. The following
diagram shows the general payload format layout:
+----------------+-------------------+----------------
| payload header | table of contents | audio data ...
+----------------+-------------------+----------------
Payloads containing more than one audio frame are called compound
payloads.
The following sections describe the variations taken by the payload
format depending on the mode in use: basic mode or interleaved mode.
4.3. Payload Definitions
4.3.1. Payload Header
The payload header carries data that is common for all frames in the
payload. The structure of the payload header is described below.
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
| ISF |TFI|L|
+-+-+-+-+-+-+-+-+
ISF (5 bits): Indicates the Internal Sampling Frequency employed for
all frames in this payload. The index value corresponds to
internal sampling frequency as specified in Table 24 in [1]. This
field SHALL be set to 0 for payloads containing frames with Frame
Type values 0-13.
TFI (2 bits): Transport Frame Index, from 0 (first) to 3 (last),
indicating the position of the first transport frame of this
payload in the AMR-WB+ super-frame structure. For payloads with
frames of only Frame Type values 0-9, this field SHALL be set to 0
by the sender. The TFI value for a frame of type 0-9 SHALL be
ignored by the receiver. Note that the frame type is coded in the
table of contents (as discussed later); hence, the mentioned
dependencies of the frame type can be applied easily by
interpreting only values carried in the payload header. It is not
necessary to interpret the audio bit stream itself.
L (1 bit): Long displacement field flag for payloads in interleaved
mode. If set to 0, four-bit displacement fields are used to
indicate interleaving offset; if set to 1, displacement fields of
eight bits are used (see Section 4.3.2.2). For payloads in the
basic mode, this bit SHALL be set to 0 and SHALL be ignored by the
receiver.
Note that frames employing different ISF values require encapsulation
in separate packets. Thus, special considerations apply when
generating interleaved packets and an ISF change is executed. In
particular, frames that, according to the previously used
interleaving pattern, would be aggregated into a single packet have
to be separated into different packets, so that the aforementioned
condition (all frames in a packet share the ISF) remains true. A
naive implementation that splits the frames with different ISF into
different packets can result in up to twice the number of RTP
packets, when compared to an optimal interleaved solution.
Alteration of the interleaving before and after the ISF change may
reduce the need for extra RTP packets.
4.3.2. The Payload Table of Contents
The table of contents (ToC) consists of a list of entries, each entry
corresponds to a group of audio frames carried in the payload, as
depicted below.
+----------------+----------------+- ... -+----------------+
| ToC entry #1 | ToC entry #2 | ToC entry #N |
+----------------+----------------+- ... -+----------------+
When multiple groups of frames are present in a payload, the ToC
entries SHALL be placed in the packet in order of increasing RTP
timestamp value (modulo 2^32) of the first transport frame the TOC
entry represents.
4.3.2.1. ToC Entry in the Basic Mode
A ToC entry of a payload in the basic mode has the following format:
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| Frame Type | #frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
F (1 bit): If set to 1, indicates that this ToC entry is followed by
another ToC entry; if set to 0, indicates that this ToC entry is
the last one in the ToC.
Frame Type (FT) (7 bits): Indicates the audio codec frame type used
for the group of frames referenced by this ToC entry. FT
designates the combination of AMR-WB+ core and stereo rate, one of
the special AMR-WB+ frame types, the AMR-WB rate, or comfort
noise, as specified by Table 25 in [1].
#frames (8 bits): Indicates the number of frames in the group
referenced by this ToC entry. ToC entries with this field equal
to 0 (which would indicate zero frames) SHALL NOT be used, and
received packets with such a TOC entry SHALL be discarded.
4.3.2.2. ToC Entry in the Interleaved Mode
Two different ToC entry formats are defined in interleaved mode.
They differ in the length of the displacement field, 4 bits or 8
bits. The L-bit in the payload header differentiates between the two
modes.
If L=0, a ToC entry has the following format:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| Frame Type | #frames | DIS1 | ... | DISi | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | ... | DISn | Padd |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
F (1 bit): See definition in 4.3.2.1.
Frame Type (FT) (7 bits): See definition in 4.3.2.1.
#frames (8 bits): See definition in 4.3.2.1.
DIS1...DISn (4 bits): A list of n (n=#frames) displacement fields
indicating the displacement of the i:th (i=1..n) audio frame
relative to the preceding audio frame in the payload, in units of
frames. The four-bit unsigned integer displacement values may be
between 0 and 15, indicating the number of audio frames in
decoding order between the (i-1):th and the i:th frame in the
payload. Note that for the first ToC entry of the payload, the
value of DIS1 is meaningless. It SHALL be set to zero by a sender
and SHALL be ignored by a receiver. This frame's location in the
decoding order is uniquely defined by the RTP timestamp and TFI in
the payload header. Note also that for subsequent ToC entries,
DIS1 indicates the number of frames between the last frame of the
previous group and the first frame of this group.
Padd (4 bits): To ensure octet alignment, four padding bits SHALL be
included at the end of the ToC entry in case there is odd number
of frames in the group referenced by this entry. These bits SHALL
be set to zero and SHALL be ignored by the receiver. If a group
containing an even number of frames is referenced by this ToC
entry, these padding bits SHALL NOT be included in the payload.
If L=1, a ToC entry has the following format:
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| Frame Type | #frames | DIS1 | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | DISn |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
F (1 bit): See definition in 4.3.2.1.
Frame Type (FT) (7 bits): See definition in 4.3.2.1.
#frames (8 bits): See definition in 4.3.2.1.
DIS1...DISn (8 bits): A list of n (n=#frames) displacement fields
indicating the displacement of the i:th (i=1..n) audio frame
relative to the preceding audio frame in the payload, in units of
frames. The eight-bit unsigned integer displacement values may be
between 0 and 255, indicating the number of audio frames in
decoding order between the (i-1):th and the i:th frame in the
payload. Note that for the first ToC entry of the payload, the
value of DIS1 is meaningless. It SHALL be set to zero by a sender
and SHALL be ignored by a receiver. This frame's location in the
decoding order is uniquely defined by the RTP timestamp and TFI in
the payload header. Note also that for subsequent ToC entries,
DIS1 indicates the displacement between the last frame of the
previous group and the first frame of this group.
4.3.2.3. RTP Timestamp Derivation
The RTP Timestamp value for a frame SHALL be the timestamp value of
the first audio sample encoded in the frame. The timestamp value for
a frame is derived differently depending on the payload mode, basic
or interleaved. In both cases, the first frame in a compound packet
has an RTP timestamp equal to the one received in the RTP header. In
the basic mode, the RTP time for any subsequent frame is derived in
two steps. First, the sum of the frame durations (see Table 1) of
all the preceding frames in the payload is calculated. Then, this
sum is added to the RTP header timestamp value. For example, let's
assume that the RTP Header timestamp value is 12345, the payload
carries four frames, and the frame duration is 16 ms (ISF = 32 kHz)
corresponding to 1152 timestamp ticks. Then the RTP timestamp of the
fourth frame in the payload is 12345 + 3 * 1152 = 15801.
In interleaved mode, the RTP timestamp for each frame in the payload
is derived from the RTP header timestamp and the sum of the time
offsets of all preceding frames in this payload. The frame
timestamps are computed based on displacement fields and the frame
duration derived from the ISF value. Note that the displacement in
time between frame i-1 and frame i is (DISi + 1) * frame duration
because the duration of the (i-1):th must also be taken into account.
The timestamp of the first frame of the first group of frames (TS(1))
(i.e., the first frame of the payload) is the RTP header timestamp.
For subsequent frames in the group, the timestamp is computed by
TS(i) = TS(i-1) + (DISi + 1) * frame duration, 2 < i < n
For subsequent groups of frames, the timestamp of the first frame is
computed by
TS(1) = TSprev + (DIS1 + 1) * frame duration,
where TSprev denotes the timestamp of the last frame in the previous
group. The timestamps of the subsequent frames in the group are
computed in the same way as for the first group.
The following example derives the RTP timestamps for the frames in an
interleaved mode payload having the following header and ToC
information:
RTP header timestamp: 12345
ISF = 32 kHz
Frame 1 displacement field: DIS1 = 0
Frame 2 displacement field: DIS2 = 6
Frame 3 displacement field: DIS3 = 4
Frame 4 displacement field: DIS4 = 7
Assuming an ISF of 32 kHz, which implies a frame duration of 16 ms,
one frame lasts 1152 ticks. The timestamp of the first frame in the
payload is the RTP timestamp, i.e., TS(1) = RTP TS. Note that the
displacement field value for this frame must be ignored. For the
second frame in the payload, the timestamp can be calculated as TS(2)
= TS(1) + (DIS2 + 1) * 1152 = 20409. For the third frame, the
timestamp is TS(3) = TS(2) + (DIS3 + 1) * 1152 = 26169. Finally, for
the fourth frame of the payload, we have TS(4) = TS(3) + (DIS4 + 1) *
1152 = 35385.
4.3.2.4. Frame Type Considerations
The value of Frame Type (FT) is defined in Table 25 in [1]. FT=14
(AUDIO_LOST) is used to denote frames that are lost. A NO_DATA
(FT=15) frame could result from two situations: First, that no data
has been produced by the audio encoder; and second, that no data is
transmitted in the current payload. An example for the latter would
be that the frame in question has been or will be sent in an earlier
or later packet. The duration for these non-included frames is
dependent on the internal sampling frequency indicated by the ISF
field.
For frame types with index 0-13, the ISF field SHALL be set 0. The
frame duration for these frame types is fixed to 20 ms in time, i.e.,
1440 ticks in 72 kHz. For payloads containing only frames of type
0-9, the TFI field SHALL be set to 0 and SHALL be ignored by the
receiver. In a payload combining frames of type 0-9 and 10-13, the
TFI values need to be set to match the transport frames of type
10-13. Thus, frames of type 0-9 will also have a derived TFI, which
is ignored.
4.3.2.5. Other TOC Considerations
If a ToC entry with an undefined FT value is received, the whole
packet SHALL be discarded. This is to avoid the loss of data
synchronization in the depacketization process, which can result in a
severe degradation in audio quality.
Packets containing only NO_DATA frames SHOULD NOT be transmitted.
Also, NO_DATA frames at the end of a frame sequence to be carried in
a payload SHOULD NOT be included in the transmitted packet. The
AMR-WB+ SCR/DTX is identical with AMR-WB SCR/DTX described in [5] and
can only be used in combination with the AMR-WB frame types (0-8).
When multiple groups of frames are present, their ToC entries SHALL
be placed in the ToC in order of increasing RTP timestamp value
(modulo 2^32) of the first transport frame the TOC entry represents,
independent of the payload mode. In basic mode, the frames SHALL be
consecutive in time, while in interleaved mode the frames MAY not
only be non-consecutive in time but MAY even have varying inter-frame
distances.
4.3.2.6. ToC Examples
The following example illustrates a ToC for three audio frames in
basic mode. Note that in this case all audio frames are encoded
using the same frame type, i.e., there is only one ToC entry.
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Frame Type1 | #frames = 3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The next example depicts a ToC of three entries in basic mode. Note
that in this case the payload also carries three frames, but three
ToC entries are needed because the frames of the payload are encoded
using different frame types.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Frame Type1 | #frames = 1 |1| Frame Type2 | #frames = 1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Frame Type3 | #frames = 1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The following example illustrates a ToC with two entries in
interleaved mode using four-bit displacement fields. The payload
includes two groups of frames, the first one including a single
frame, and the other one consisting of two frames.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Frame Type1 | #frames = 1 | DIS1 | padd |0| Frame Type2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| #frames = 2 | DIS1 | DIS2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4.3.3. Audio Data
Audio data of a payload consists of zero or more audio frames, as
described in the ToC of the payload.
ToC entries with FT=14 or 15 represent frame types with a length of
0. Hence, no data SHALL be placed in the audio data section to
represent frames of this type.
As already discussed, each audio frame of an extension frame type
represents an AMR-WB+ transport frame corresponding to the encoding
of 512 samples of audio, sampled with the internal sampling frequency
specified by the ISF indicator. As an exception, frame types with
index 10-13 are only capable of using a single internal sampling
frequency (25600 Hz). The encoding rates (combination of core bit-
rate and stereo bit-rate) are indicated in the frame type field of
the corresponding ToC entry. The octet length of the audio frame is
implicitly defined by the frame type field and is given in Tables 21
and 25 of [1]. The order and numbering notation of the bits are as
specified in [1]. For the AMR-WB+ extension frame types and comfort
noise frames, the bits are in the order produced by the encoder. The
last octet of each audio frame MUST be padded with zeroes at the end
if not all bits in the octet are used. In other words, each audio
frame MUST be octet-aligned.
4.3.4. Methods for Forming the Payload
The payload begins with the payload header, followed by the table of
contents, which consists of a list of ToC entries.
The audio data follows the table of contents. All the octets
comprising an audio frame SHALL be appended to the payload as a unit.
The audio frames are packetized in timestamp order within each group
of frames (per ToC entry). The groups of frames are packetized in
the same order as their corresponding ToC entries. Note that there
are no data octets in a group having a ToC entry with FT=14 or FT=15.
4.3.5. Payload Examples
4.3.5.1. Example 1: Basic Mode Payload Carrying Multiple Frames Encoded
Using the Same Frame Type
Figure 4 depicts a payload that carries three AMR-WB+ frames encoded
using 14 kbit/s frame type (FT=26) with a frame length of 280 bits
(35 bytes). The internal sampling frequency in this example is 25.6
kHz (ISF = 8). The TFI for the first frame is 2, indicating that the
first transport frame in this payload is the third in a super-frame.
Since this payload is in the basic mode, the subsequent frames of the
payload are consecutive frames in decoding order, i.e., the fourth
transport frame of the current super-frame and the first transport
frame of the next super-frame. Note that because the frames are all
encoded using the same frame type, only one ToC entry is required.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ISF = 8 | 2 |0|0| FT = 26 | #frames = 3 | f1(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f1(272...279) | f2(0...7) | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f2(272...279) | f3(0...7) | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f3(272...279) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: An example of a basic mode payload carrying three frames
of the same frame type
4.3.5.2. Example 2: Basic Mode Payload Carrying Multiple Frames Encoded
Using Different Frame Types
Figure 5 depicts a payload that carries three AMR-WB+ frames; the
first frame is encoded using 18.4 kbit/s frame type (FT=33) with a
frame length of 368 bits (46 bytes), and the two subsequent frames
are encoded using 20 kbit/s frame type (FT=35) having frame length of
400 bits (50 bytes). The internal sampling frequency in this example
is 32 kHz (ISF = 10), implying the overall bit-rates of 23 kbit/s for
the first frame of the payload, and 25 kbit/s for the subsequent
frames. The TFI for the first frame is 3, indicating that the first
transport frame in this payload is the fourth in a super-frame.
Since this is a payload in the basic mode, the subsequent frames of
the payload are consecutive frames in decoding order, i.e., the first
and second transport frames of the current super-frame. Note that
since the payload carries two different frame types, there are two
ToC entries.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ISF=10 | 3 |0|1| FT = 33 | #frames = 1 |0| FT = 35 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| #frames = 2 | f1(0...7) | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f1(360...367) | f2(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| f2(392...399) | f3(0...7) | ... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f3(392...399) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: An example of a basic mode payload carrying three frames
employing two different frame types
4.3.5.3. Example 3: Payload in Interleaved Mode
The example in Figure 6 depicts a payload in interleaved mode,
carrying four frames encoded using 32 kbit/s frame type (FT=47) with
frame length of 640 bits (80 bytes). The internal sampling frequency
is 38.4 kHz (ISF = 13), implying a bit-rate of 48 kbit/s for all
frames in the payload. The TFI for the first frame is 0; hence, it
is the first transport frame of a super-frame. The displacement
fields for the subsequent frames are DIS2=18, DIS3=15, and DIS4=10,
which indicates that the subsequent frames have the TFIs of 3, 3, and
2, respectively. The long displacement field flag L in the payload
header is set to 1, which results in the use of eight bits for the
displacement fields in the ToC entry. Note that since all frames of
this payload are encoded using the same frame type, there is need
only for a single ToC entry. Furthermore, the displacement field for
the first frame (corresponding to the first ToC entry with DIS1=0)
must be ignored, since its timestamp and TFI are defined by the RTP
timestamp and the TFI found in the payload header.
The RTP timestamp values of the frames in this example are:
Frame1: TS1 = RTP Timestamp
Frame2: TS2 = TS1 + 19 * 960
Frame3: TS3 = TS2 + 16 * 960
Frame4: TS4 = TS3 + 11 * 960
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ISF=13 | 0 |1|0| FT = 47 | #frames = 4 | DIS1 = 0 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| DIS2 = 18 | DIS3 = 15 | DIS4 = 10 | f1(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f1(632...639) | f2(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f2(632...639) | f3(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f3(632...639) | f4(0...7) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... | f4(632...639) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: An example of an interleaved mode payload carrying four
frames at the same frame type
4.4. Interleaving Considerations
The use of interleaving requires further considerations. As
presented in the example in Section 3.6.2, a given interleaving
pattern requires a certain amount of the deinterleaving buffer. This
buffer space, expressed in a number of transport frame slots, is
indicated by the "interleaving" media type parameter. The number of
frame slots needed can be converted into actual memory requirements
by considering the 80 bytes per frame used by the largest combination
of AMR-WB+'s core and stereo rates.
The information about the frame buffer size is not always sufficient
to determine when it is appropriate to start consuming frames from
the interleaving buffer. There are two cases in which additional
information is needed: first, when switching of the ISF occurs, and
second, when the interleaving pattern changes. The "int-delay" media
type parameter is defined to convey this information. It allows a
sender to indicate the minimal media time that needs to be present in
the buffer before the decoder can start consuming frames from the
buffer. Because the sender has full control over ISF changes and the
interleaving pattern, it can calculate this value.
In certain cases (for example, if joining a multicast session with
interleaving mid-session), a receiver may initially receive only part
of the packets in the interleaving pattern. This initial partial
reception (in frame sequence order) of frames can yield too few
frames for acceptable quality from the audio decoding. This problem
also arises when using encryption for access control, and the
receiver does not have the previous key.
Although the AMR-WB+ is robust and thus tolerant to a high random
frame erasure rate, it would have difficulties handling consecutive
frame losses at startup. Thus, some special implementation
considerations are described. In order to handle this type of
startup efficiently, it must be noted that decoding is only possible
to start at the beginning of a super-frame, and that holds true even
if the first transport frame is indicated as lost. Secondly,
decoding is only RECOMMENDED to start if at least 2 transport frames
are available out of the 4 belonging to that super-frame.
After receiving a number of packets, in the worst case as many
packets as the interleaving pattern covers, the previously described
effects disappear and normal decoding is resumed.
Similar issues arise when a receiver leaves a session or has lost
access to the stream. If the receiver leaves the session, this would
be a minor issue since playout is normally stopped. It is also a
minor issue for the case of lost access, since the AMR-WB+ error
concealment will fade out the audio if massive consecutive losses are
encountered.
The sender can avoid this type of problem in many sessions by
starting and ending interleaving patterns correctly when risks of
losses occur. One such example is a key-change done for access
control to encrypted streams. If only some keys are provided to
clients and there is a risk of their receiving content for which they
do not have the key, it is recommended that interleaving patterns not
overlap key changes.
4.5. Implementation Considerations
An application implementing this payload format MUST understand all
the payload parameters. Any mapping of the parameters to a signaling
protocol MUST support all parameters. So an implementation of this
payload format in an application using SDP is required to understand
all the payload parameters in their SDP-mapped form. This
requirement ensures that an implementation always can decide whether
it is capable of communicating.
Both basic and interleaved mode SHALL be implemented. The
implementation burden of both is rather small, and requiring both
ensures interoperability. As the AMR-WB+ codec contains the full
functionality of the AMR-WB codec, it is RECOMMENDED to also
implement the payload format in RFC 3267 [7] for the AMR-WB frame
types when implementing this specification. Doing so makes
interoperability with devices that only support AMR-WB more likely.
The switching of ISF, when combined with packet loss, could result in
concealment using the wrong audio frame length. This can occur if
packet losses result in lost frames directly after the point of ISF
change. The packet loss would prevent the receiver from noticing the
changed ISF and thereby conceal the lost transport frame with the
previous ISF, instead of the new one. Although always later
detectable, such an error results in frame boundary misalignment,
which can cause audio distortions and problems with synchronization,
as too many or too few audio samples were created. This problem can
be mitigated in most cases by performing ISF recovery prior to
concealment as outlined in Section 4.5.1.
4.5.1. ISF Recovery in Case of Packet Loss
In case of packet loss, it is important that the AMR-WB+ decoder
initiates a proper error concealment to replace the frames carried in
the lost packet. A loss concealment algorithm requires a codec
framing that matches the timestamps of the correctly received frames.
Hence, it is necessary to recover the timestamps of the lost frames.
Doing so is non-trivial because the codec frame length that is
associated with the ISF may have changed during the frame loss.
In the following, the recovery of the timestamp information of lost
frames is illustrated by the means of an example. Two frames with
timestamps t0 and t1 have been received properly, the first one being
the last packet before the loss, and the latter one being the first
packet after the loss period. The ISF values for these packets are
isf0 and isf1, respectively. The TFIs of these frames are tfi0 and
tfi1, respectively. The associated frame lengths (in timestamp
ticks) are given as L0 and L1, respectively. In this example three
frames with timestamps x1 - x3 have been lost. The example further
assumes that ISF changes once from isf0 to isf1 during the frame loss
period, as shown in the figure below.
Since not all information required for the full recovery of the
timestamps is generally known in the receiver, an algorithm is needed
to estimate the ISF associated with the lost frames. Also, the
number of lost frames needs to be recovered.
|<---L0--->|<---L0--->|<-L1->|<-L1->|<-L1->|
| Rxd | lost | lost | lost | Rxd |
--+----------+----------+------+------+------+--
t0 x1 x2 x3 t1
Example Algorithm:
Start: # check for frame loss
If (t0 + L0) == t1 Then goto End # no frame loss
Step 1: # check case with no ISF change
If (isf0 != isf1) Then goto Step 2 # At least one ISF change
If (isFractional(t1 - t0)/L0) Then goto Step 3
# More than 1 ISF change
Return recovered timestamps as
x(n) = t0 + n*L1 and associated ISF equal to isf0,
for 0 < n < (t1 - t0)/L0
goto End
Step 2:
Loop initialization: n := 4 - tfi0 mod 4
While n <= (t1-t0)/L0
Evaluate m := (t1 - t0 - n*L0)/L1
If (isInteger(m) AND ((tfi0+n+m) mod 4 == tfi1)) Then goto found;
n := n+4
endloop
goto step 3 # More than 1 ISF change
found:
Return recovered timestamps and ISFs as
x(i) = t0 + i*L0 and associated ISF equal to isf0, for 0 < i <= n
x(i) = t0 + n*L0 + (i-n)*L1 and associated ISF equal to isf1,
for n < i <= n+m
goto End
Step 3:
More than 1 ISF change has occurred. Since ISF changes can be
assumed to be infrequent, such a situation occurs only if long
sequences of frames are lost. In that case it is probably not useful
to try to recover the timestamps of the lost frames. Rather, the
AMR-WB+ decoder should be reset, and decoding should be resumed
starting with the frame with timestamp t1.
End:
The above algorithm still does not solve the issue when the receiver
buffer depth is shallower than the loss burst. In this kind of case,
where the concealment must be done without any knowledge about future
frames, the concealment may result in loss of frame boundary
alignment. If that occurs, it may be necessary to reset and restart
the codec to perform resynchronization.
4.5.2. Decoding Validation
If the receiver finds a mismatch between the size of a received
payload and the size indicated by the ToC of the payload, the
receiver SHOULD discard the packet. This is recommended because
decoding a frame parsed from a payload based on erroneous ToC data
could severely degrade the audio quality.
5. Congestion Control
The general congestion control considerations for transporting RTP
data apply; see RTP [3] and any applicable RTP profile like AVP [9].
However, the multi-rate capability of AMR-WB+ audio coding provides a
mechanism that may help to control congestion, since the bandwidth
demand can be adjusted (within the limits of the codec) by selecting
a different coding frame type or lower internal sampling rate.
The number of frames encapsulated in each RTP payload highly
influences the overall bandwidth of the RTP stream due to header
overhead constraints. Packetizing more frames in each RTP payload
can reduce the number of packets sent and hence the header overhead,
at the expense of increased delay and reduced error robustness.
If forward error correction (FEC) is used, the amount of FEC-induced
redundancy needs to be regulated such that the use of FEC itself does
not cause a congestion problem.
6. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in RTP
[3] and any applicable profile such as AVP [9] or SAVP [10]. As this
format transports encoded audio, the main security issues include
confidentiality, integrity protection, and data origin authentication
of the audio itself. The payload format itself does not have any
built-in security mechanisms. Any suitable external mechanisms, such
as SRTP [10], MAY be used.
This payload format and the AMR-WB+ decoder do not exhibit any
significant non-uniformity in the receiver-side computational
complexity for packet processing, and thus are unlikely to pose a
denial-of-service threat due to the receipt of pathological data.
6.1. Confidentiality
In order to ensure confidentiality of the encoded audio, all audio
data bits MUST be encrypted. There is less need to encrypt the
payload header or the table of contents since they only carry
information about the frame type. This information could also be
useful to a third party, for example, for quality monitoring.
The use of interleaving in conjunction with encryption can have a
negative impact on confidentiality, for a short period of time.
Consider the following packets (in brackets) containing frame numbers
as indicated: {10, 14, 18}, {13, 17, 21}, {16, 20, 24} (a popular
continuous diagonal interleaving pattern). The originator wishes to
deny some participants the ability to hear material starting at time
16. Simply changing the key on the packet with the timestamp at or
after 16, and denying that new key to those participants, does not
achieve this; frames 17, 18, and 21 have been supplied in prior
packets under the prior key, and error concealment may make the audio
intelligible at least as far as frame 18 or 19, and possibly further.
6.2. Authentication and Integrity
To authenticate the sender of the speech, an external mechanism MUST
be used. It is RECOMMENDED that such a mechanism protects both the
complete RTP header and the payload (speech and data bits).
Data tampering by a man-in-the-middle attacker could replace audio
content and also result in erroneous depacketization/decoding that
could lower the audio quality.
7. Payload Format Parameters
This section defines the parameters that may be used to select
features of the AMR-WB+ payload format. The parameters are defined
as part of the media type registration for the AMR-WB+ audio codec.
A mapping of the parameters into the Session Description Protocol
(SDP) [6] is also provided for those applications that use SDP.
Equivalent parameters could be defined elsewhere for use with control
protocols that do not use MIME or SDP.
The data format and parameters are only specified for real-time
transport in RTP.
7.1. Media Type Registration
The media type for the Extended Adaptive Multi-Rate Wideband
(AMR-WB+) codec is allocated from the IETF tree, since AMR-WB+ is
expected to be a widely used audio codec in general streaming
applications.
Note: Parameters not listed below MUST be ignored by the receiver.
Media Type name: audio
Media subtype name: AMR-WB+
Required parameters:
None
Optional parameters:
channels: The maximum number of audio channels used by the
audio frames. Permissible values are 1 (mono) or 2
(stereo). If no parameter is present, the maximum
number of channels is 2 (stereo). Note: When set to
1, implicitly the stereo frame types cannot be used.
interleaving: Indicates that interleaved mode SHALL
be used for the payload. The parameter specifies
the number of transport frame slots required in a
deinterleaving buffer (including the frame that is
ready to be consumed). Its value is equal to one
plus the maximum number of frames that precede any
frame in transmission order and follow the frame in
RTP timestamp order. The value MUST be greater than
zero. If this parameter is not present,
interleaved mode SHALL NOT be used.
int-delay: The minimal media time delay in RTP timestamp ticks
that is needed in the deinterleaving buffer, i.e.,
the difference in RTP timestamp ticks between the
earliest and latest audio frame present in the
deinterleaving buffer.
ptime: See Section 6 in RFC 2327 [6].
maxptime: See Section 8 in RFC 3267 [7].
Restriction on Usage:
This type is only defined for transfer via RTP (STD 64).
Encoding considerations:
An RTP payload according to this format is binary data
and thus may need to be appropriately encoded in non-
binary environments. However, as long as used within
RTP, no encoding is necessary.
Security considerations:
See Section 6 of RFC 4352.
Interoperability considerations:
To maintain interoperability with AMR-WB-capable end-
points, in cases where negotiation is possible and the
AMR-WB+ end-point supporting this format also supports
RFC 3267 for AMR-WB transport, an AMR-WB+ end-point
SHOULD declare itself also as AMR-WB capable (i.e.,
supporting also "audio/AMR-WB" as specified in RFC
3267).
As the AMR-WB+ decoder is capable of performing stereo
to mono conversions, all receivers of AMR-WB+ should be
able to receive both stereo and mono, although the
receiver is only capable of playout of mono signals.
Public specification:
RFC 4352
3GPP TS 26.290, see reference [1] of RFC 4352
Additional information:
This MIME type is not applicable for file storage.
Instead, file storage of AMR-WB+ encoded audio is
specified within the 3GPP-defined ISO-based multimedia
file format defined in 3GPP TS 26.244; see reference
[14] of RFC 4352. This file format has the MIME types
"audio/3GPP" or "video/3GPP" as defined by RFC 3839
[15].
Person & email address to contact for further information:
magnus.westerlund@ericsson.com
ari.lakaniemi@nokia.com
Intended usage: COMMON.
It is expected that many IP-based streaming
applications will use this type.
Change controller:
IETF Audio/Video Transport working group delegated from
the IESG.
7.2. Mapping Media Type Parameters into SDP
The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[6], which is commonly used to describe RTP sessions. When SDP is
used to specify an RTP session using this RTP payload format, the
mapping is as follows:
- The media type ("audio") is used in SDP "m=" as the media name.
- The media type (payload format name) is used in SDP "a=rtpmap" as
the encoding name. The RTP clock rate in "a=rtpmap" SHALL be
72000 for AMR-WB+, and the encoding parameter number of channels
MUST either be explicitly set to 1 or 2, or be omitted, implying
the default value of 2.
- The parameters "ptime" and "maxptime" are placed in the SDP
attributes "a=ptime" and "a=maxptime", respectively.
- Any remaining parameters are placed in the SDP "a=fmtp" attribute
by copying them directly from the MIME media type string as a
semicolon-separated list of parameter=value pairs.
7.2.1. Offer-Answer Model Considerations
To achieve good interoperability in an Offer-Answer [8] negotiation
usage, the following considerations should be taken into account:
For negotiable offer/answer usage the following interpretation rules
SHALL be applied:
- The "interleaving" parameter is symmetric, thus requiring that the
answerer must also include it for the answer to an offered payload
type that contains the parameter. However, the buffer space value
is declarative in usage in unicast. For multicast usage, the same
value in the response is required in order to accept the payload
type. For streams declared as sendrecv or recvonly: The receiver
will accept reception of streams using the interleaved mode of the
payload format. The value declares the amount of buffer space the
receiver has available for the sender to utilize. For sendonly
streams, the parameter indicates the desired configuration and
amount of buffer space. An answerer is RECOMMENDED to respond
using the offered value, if capable of using it.
- The "int-delay" parameter is declarative. For streams declared as
sendrecv or recvonly, the value indicates the maximum initial
delay the receiver will accept in the deinterleaving buffer. For
sendonly streams, the value is the amount of media time the sender
desires to use. The value SHOULD be copied into any response.
- The "channels" parameter is declarative. For "sendonly" streams,
it indicates the desired channel usage, stereo and mono, or mono
only. For "recvonly" and "sendrecv" streams, the parameter
indicates what the receiver accepts to use. As any receiver will
be capable of receiving stereo frame type and perform local mixing
within the AMR-WB+ decoder, there is normally only one reason to
restrict to mono only: to avoid spending bit-rate on data that are
not utilized if the front-end is only capable of mono.
- The "ptime" parameter works as indicated by the offer/answer model
[8]; "maxptime" SHALL be used in the same way.
- To maintain interoperability with AMR-WB in cases where
negotiation is possible, an AMR-WB+ capable end-point that also
implements the AMR-WB payload format [7] is RECOMMENDED to declare
itself capable of AMR-WB as it is a subset of the AMR-WB+ codec.
In declarative usage, like SDP in RTSP [16] or SAP [17], the
following interpretation of the parameters SHALL be done:
- The "interleaving" parameter, if present, configures the payload
format in that mode, and the value indicates the number of frames
that the deinterleaving buffer is required to support to be able
to handle this session correctly.
- The "int-delay" parameter indicates the initial buffering delay
required to receive this stream correctly.
- The "channels" parameter indicates if the content being
transmitted can contain either both stereo and mono rates, or only
mono.
- All other parameters indicate values that are being used by the
sending entity.
7.2.2. Examples
One example of an SDP session description utilizing AMR-WB+ mono and
stereo encoding follows.
m=audio 49120 RTP/AVP 99
a=rtpmap:99 AMR-WB+/72000/2
a=fmtp:99 interleaving=30; int-delay=86400
a=maxptime:100
Note that the payload format (encoding) names are commonly shown in
uppercase. Media subtypes are commonly shown in lowercase. These
names are case-insensitive in both places. Similarly, parameter
names are case-insensitive both in MIME types and in the default
mapping to the SDP a=fmtp attribute.
8. IANA Considerations
The IANA has registered one new MIME subtype (audio/amr-wb+); see
Section 7.
9. Contributors
Daniel Enstrom has contributed in writing the codec introduction
section. Stefan Bruhn has contributed by writing the ISF recovery
algorithm.
10. Acknowledgements
The authors would like to thank Redwan Salami and Stefan Bruhn for
their significant contributions made throughout the writing and
reviewing of this document. Dave Singer contributed by reviewing and
suggesting improved language. Anisse Taleb and Ingemar Johansson
contributed by implementing the payload format and thus helped locate
some flaws. We would also like to acknowledge Qiaobing Xie, coauthor
of RFC 3267, on which this document is based.
11. References
11.1. Normative References
[1] 3GPP TS 26.290 "Audio codec processing functions; Extended
Adaptive Multi-Rate Wideband (AMR-WB+) codec; Transcoding
functions", version 6.3.0 (2005-06), 3rd Generation Partnership
Project (3GPP).
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[4] 3GPP TS 26.192 "AMR Wideband speech codec; Comfort Noise
aspects", version 6.0.0 (2004-12), 3rd Generation Partnership
Project (3GPP).
[5] 3GPP TS 26.193 "AMR Wideband speech codec; Source Controlled
Rate operation", version 6.0.0 (2004-12), 3rd Generation
Partnership Project (3GPP).
[6] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[7] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "Real-
Time Transport Protocol (RTP) Payload Format and File Storage
Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
Wideband (AMR-WB) Audio Codecs", RFC 3267, June 2002.
[8] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[9] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
11.2. Informative References
[10] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
3711, March 2004.
[11] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
Generic Forward Error Correction", RFC 2733, December 1999.
[12] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload
for Redundant Audio Data", RFC 2198, September 1997.
[13] 3GPP TS 26.233 "Packet Switched Streaming service", version
5.7.0 (2005-03), 3rd Generation Partnership Project (3GPP).
[14] 3GPP TS 26.244 "Transparent end-to-end packet switched streaming
service (PSS); 3GPP file format (3GP)", version 6.4.0 (2005-09),
3rd Generation Partnership Project (3GPP).
[15] Castagno, R. and D. Singer, "MIME Type Registrations for 3rd
Generation Partnership Project (3GPP) Multimedia files", RFC
3839, July 2004.
[16] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
[17] Handley, M., Perkins, C., and E. Whelan, "Session Announcement
Protocol", RFC 2974, October 2000.
[18] 3GPP TS 26.140 "Multimedia Messaging Service (MMS); Media
formats and codes", version 6.2.0 (2005-03), 3rd Generation
Partnership Project (3GPP).
[19] 3GPP TS 26.140 "Multimedia Broadcast/Multicast Service (MBMS);
Protocols and codecs", version 6.3.0 (2005-12), 3rd Generation
Partnership Project (3GPP).
Any 3GPP document can be downloaded from the 3GPP webserver,
"http://www.3gpp.org/", see specifications.
Authors' Addresses
Johan Sjoberg
Ericsson Research
Ericsson AB
SE-164 80 Stockholm
SWEDEN
Phone: +46 8 7190000
EMail: Johan.Sjoberg@ericsson.com
Magnus Westerlund
Ericsson Research
Ericsson AB
SE-164 80 Stockholm
SWEDEN
Phone: +46 8 7190000
EMail: Magnus.Westerlund@ericsson.com
Ari Lakaniemi
Nokia Research Center
P.O. Box 407
FIN-00045 Nokia Group
FINLAND
Phone: +358-71-8008000
EMail: ari.lakaniemi@nokia.com
Stephan Wenger
Nokia Corporation
P.O. Box 100
FIN-33721 Tampere
FINLAND
Phone: +358-50-486-0637
EMail: Stephan.Wenger@nokia.com
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EID 114 (Verified) is as follows:
Section: 99
Original Text:
(1)
In Section 4.3.2.3 of RFC 4352, the first formula line on page 18,
TS(i) = TS(i-1) + (DISi + 1) * frame duration, 2 < i < n
should say:
TS(i) = TS(i-1) + (DISi + 1) * frame duration, 2 <= i <= n
(2)
The 'Example Algorithm' in Section 4.5.1, on page 27, in the second
half of Step 1, says:
Return recovered timestamps as
x(n) = t0 + n*L1 and associated ISF equal to isf0,
for 0 < n < (t1 - t0)/L0
goto End
This pseudocode fragment should say:
for 0 < n < (t1 - t0)/L0
Return recovered timestamps as
x(n) = t0 + n*L0 and associated ISF equal to isf0
goto End
(3)
In Section 7.2, on page 32, the second item of the unnumbered list
says:
- The media type (payload format name) is used in SDP "a=rtpmap" as
the encoding name. [...]
It should say:
- The media subtype (payload format name) is used in SDP "a=rtpmap"
as the encoding name. [...]
(4)
Within the second paragraph on page 33, Section 7.2.1 of RFC 4352
says:
[...] As any receiver will
be capable of receiving stereo frame type and perform local mixing
within the AMR-WB+ decoder, there is normally only one reason to
restrict to mono only: to avoid spending bit-rate on data that are
not utilized if the front-end is only capable of mono.
It should say:
[...] As any receiver will
be capable of receiving stereo frame types and perform local
mixing within the AMR-WB+ decoder, there is normally only one
reason to restrict to mono only: to avoid spending bit-rate on
data that are not utilized if the front-end is only capable of
mono.
Corrected Text:
Notes:
from pending