ETSI TS 126 448 V18.0.0 (2024-05)

Universal Mobile Telecommunications System (UMTS); LTE; 5G; Codec for Enhanced Voice Services (EVS); Jitter Buffer Management (3GPP TS 26.448 version 18.0.0 Release 18)

RTS/TSGS-0426448vi00

General Information

• Status: Not Published
• Technical Committee: 3GPP SA 4 - Codec
• Current Stage: 12 - Citation in the OJ (auto-insert)
• Completion Date: 16-May-2024

Standards Content (Sample)

TECHNICAL SPECIFICATION
Universal Mobile Telecommunications System (UMTS);
LTE;
5G;
Codec for Enhanced Voice Services (EVS);
Jitter Buffer Management
(3GPP TS 26.448 version 18.0.0 Release 18)

3GPP TS 26.448 version 18.0.0 Release 18 1 ETSI TS 126 448 V18.0.0 (2024-05)

Reference
RTS/TSGS-0426448vi00
Keywords
5G,LTE,UMTS
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ETSI
3GPP TS 26.448 version 18.0.0 Release 18 2 ETSI TS 126 448 V18.0.0 (2024-05)
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ETSI
3GPP TS 26.448 version 18.0.0 Release 18 3 ETSI TS 126 448 V18.0.0 (2024-05)
Contents
Intellectual Property Rights . 2
Legal Notice . 2
Modal verbs terminology . 2
Foreword . 4
1 Scope . 5
2 References . 5
3 Definitions, symbols and abbreviations . 5
3.1 Definitions . 5
3.2 Symbols . 5
3.3 Abbreviations . 6
3.4 Mathematical Expressions . 6
4 General . 6
4.1 Introduction . 6
4.2 Packet-based communications . 7
4.3 EVS Receiver architecture overview . 7
5 Jitter Buffer Management . 8
5.1 Overview . 8
5.2 Depacketization of RTP packets (informative) . 9
5.3 Network Jitter Analysis and Delay Estimatio n . 9
5.3.1 General . 9
5.3.2 Long-term Jitter . 10
5.3.3 Short-term jitter . 10
5.3.4 Target Playout Delay . 11
5.3.5 Playout Delay Estimation . 11
5.4 Adaptation Control Logic . 12
5.4.1 Control Logic . 12
5.4.2 Frame-based adaptation . 12
5.4.2.1 General . 12
5.4.2.2 Insertion of Concealed Frames . 12
5.4.2.3 Frame Dropping . 13
5.4.2.4 Comfort Noise Insertion in DTX . 13
5.4.2.5 Comfort Noise Deletion in DTX . 13
5.4.3 Signal-based adaptation . 13
5.4.3.1 General . 13
5.4.3.2 Time-shrinking . 14
5.4.3.3 Time-stretching . 15
5.4.3.4 Energy Estimation . 16
5.4.3.5 Similarity Measurement . 16
5.4.3.6 Quality Control . 17
5.4.3.7 Overlap-add . 17
5.5 Receiver Output Buffer . 18
5.6 De-Jitter Buffer . 18
6 Decoder interaction . 19
6.1 General . 19
6.2 Decoder Requirements . 19
6.3 Partial Redundancy. 19
6.3.1 Computation of the Partial Redundancy Offset . 20
6.3.2 Computation of a frame erasure rate indicator to control the frequency of the Partial Redundancy
transmission . 21
Annex A (informative): Change history . 22
History . 23
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3GPP TS 26.448 version 18.0.0 Release 18 4 ETSI TS 126 448 V18.0.0 (2024-05)
Foreword
rd
This Technical Specification has been produced by the 3 Generation Partnership Project (3GPP).
The contents of the present document are subject to continuing work within the TSG and may change following formal
TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an
identifying change of release date and an increase in version number as follows:
Version x.y.z
where:
x the first digit:
1 presented to TSG for information;
2 presented to TSG for approval;
3 or greater indicates TSG approved document under change control.
y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections,
updates, etc.
z the third digit is incremented when editorial only changes have been incorporated in the document.
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3GPP TS 26.448 version 18.0.0 Release 18 5 ETSI TS 126 448 V18.0.0 (2024-05)
1 Scope
The present document defines the Jitter Buffer Management solution for the Codec for Enhanced Voice Services (EVS).
2 References
The following documents contain provisions which, through reference in this text, constitute provisions of the present
document.
- References are either specific (identified by date of publication, edition number, version number, etc.) or
non-specific.
- For a specific reference, subsequent revisions do not apply.
- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including
a GSM document), a non-specific reference implicitly refers to the latest version of that document in the same
Release as the present document.
[1] 3GPP TR 21.905: "Vocabulary for 3GPP Specifications".
[2] 3GPP TS 26.445: "Codec for Enhanced Voice Services (EVS); Detailed Algorithmic Description".
[3] 3GPP TS 26.114: "IP Multimedia Subsystem (IMS); Multimedia telephony; Media handling and
interaction".
[4] 3GPP TS 26.071: "Mandatory speech CODEC speech processing functions; AMR speech Codec;
General description".
[5] 3GPP TS 26.171: "Speech codec speech processing functions; Adaptive Multi-Rate - Wideband
(AMR-WB) speech codec; General description".
[6] 3GPP TS 26.442: "Codec for Enhanced Voice Services (EVS); ANSI C code (fixed-point)".
[7] 3GPP TS 26.443: "Codec for Enhanced Voice Services (EVS); ANSI C code (floating-point)".
[8] 3GPP TS 26.131: "Terminal acoustic characteristics for telephony; Requirements".
[9] IETF RFC 4867 (2007): "RTP Payload Format and File Storage Format for the Adaptive Multi-
Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs", J. Sjoberg, M.
Westerlund, A. Lakaniemi and Q. Xie.
[10] 3GPP TS 26.452: "Codec for Enhanced Voice Services (EVS); ANSI C code; Alternative fixed-
point using updated basic operators".
3 Definitions, symbols and abbreviations
3.1 Definitions
For the purposes of the present document, the terms and definitions given in TR 21.905 [1] and the following apply. A
term defined in the present document takes precedence over the definition of the same term, if any, in TR 21.905 [1].
3.2 Symbols
For the purposes of the present document, the following symbols apply:
s ()n Time signal and time index n in context x, e.g. x can be inp, out, HP, pre, etc.
x
L Frame length / size of module x
x
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E Energy values in context of x
x
C Correlation function in context x
x
3.3 Abbreviations
For the purposes of the present document, the abbreviations given in TR 21.905 [1] and the following apply. An
abbreviation defined in the present document takes precedence over the definition of the same abbreviation, if any, in
TR 21.905 [1].
AMR Adaptive Multi Rate (codec)
AMR-WB Adaptive Multi Rate Wideband (codec)
CNG Comfort Noise Generator
DTX Discontinuous Transmission
EVS Enhanced Voice Services
FB Fullband
FIFO First In, First Out
IP Internet Protocol
JBM Jitter Buffer Management
MTSI Multimedia Telephony Service for IMS
NB Narrowband
PCM Pulse Code Modulation
PLC Packet Loss Concealment
RTP Real Time Transport Protocol
SID Silence Insertion Descriptor
SOLA Synchronized overlap-add
SWB Super Wideband
TSM Time Scale Modification
VAD Voice Activity Detection
WB Wideband
3.4 Mathematical Expressions
For the purposes of the present document, the following conventions apply to mathematical expressions:
x indicates the smallest integer greater than or equal to x: , 1.1 = 2 2.0 = 2 and −1.1 = −1
     
x indicates the largest integer less than or equal to x: , 1.1 = 1 1.0 = 1 and −1.1 = −2
      
min(x ,…x ) indicates the minimum of x ,…, x , N being the number of components
0 N–1 0 N–1
max(x ,…x ) indicates the maximum of x , …, x
0 N–1 0 N–1
indicates summation

4 General
4.1 Introduction
The present document defines the Jitter Buffer Management solution for the Codec for Enhanced Voice Services
(EVS) [2]. Jitter Buffers are required in packet-based communications, such as 3GPP MTSI [2], to smooth the inter-
arrival jitter of incoming media packets for uninterrupted playout.
The solution is used in conjunction with the EVS decoder and can also be used for AMR [4] and AMR-WB [5]. It is
optimized for the Multimedia Telephony Service for IMS (MTSI) and fulfils the requirements for delay and jitter-
induced concealment operations set in [2].
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The procedure of the present document is recommended for implementation in all network entities and UEs supporting
the EVS codec.
The present document does not describe the ANSI C code of this procedure. For a description of the two fixed-point
ANSI C code implementations, using different sets of basic operators, see [6] and [10] respectively; for a description of
the floating-point ANSI C code implementation see [7].
In the case of discrepancy between the EVS Jitter Buffer Management described in the present document and its ANSI-
C code specification contained in [6], the procedure defined by [6] prevails. In the case of discrepancy between the
procedure described in the present document and its ANSI-C code specification contained in [7], the procedure defined
by [7] prevails. In the case of discrepancy between the procedure described in the present document and its ANSI-C
code specifications contained in [10] the procedure defined by [10] prevails.
4.2 Packet-based communications
In packet-based communications, packets arrive at the terminal with random jitters in their arrival time. Packets may
also arrive out of order. Since the decoder expects to be fed a speech packet every 20 milliseconds to output speech
samples in periodic blocks, a de-jitter buffer is required to absorb the jitter in the packet arrival time. The larger the size
of the de-jitter buffer, the better its ability to absorb the jitter in the arrival time and consequently fewer late arriving
packets are discarded. Voice communications is also a delay critical system and therefore it becomes essential to keep
the end to end delay as low as possible so that a two way conversation can be sustained.
The defined adaptive Jitter Buffer Management (JBM) solution reflects the above mentioned trade-offs. While
attempting to minimize packet losses, the JBM algorithm in the receiver also keeps track of the delay in packet delivery
as a result of the buffering. The JBM solution suitably adjusts the depth of the de-jitter buffer in order to achieve the
trade-off between delay and late losses.
4.3 EVS Receiver architecture overview
An EVS receiver for MTSI-based communication is built on top of the EVS Jitter Buffer Management solution. In the
EVS Jitter Buffer Management solution the received EVS frames, contained in RTP packets, are depacketized and fed
to the Jitter Buffer Management (JBM). The JBM smoothes the inter-arrival jitter of incoming packets for uninterrupted
playout of the decoded EVS frames at the Acoustic Frontend of the terminal.

Figure 1: Receiver architecture for the EVS Jitter Buffer Management Solution
Figure 1 illustrates the architecture and data flow of the receiver side of an EVS terminal. Note that the architecture
serves only as an example to outline the integration of the JBM in a terminal. This specification defines the JBM
module and its interfaces to the RTP Depacker, the EVS Decoder [2], and the Acoustic Frontend [8]. The modules for
Modem and Acoustic Frontend are outside the scope of the present document. The actual implementation of the RTP
Depacker is outlined in a basic form; more complex depacketization scenarios depend on the usage of RTP.
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Real-time implementations of this architecture typically use independent processing threads for reacting on arriving
RTP packets from the modem and for requesting PCM data for the Acoustic Frontend. Arriving packets are typically
handled by listening for packets received on the network socket related to the RTP session. Incoming packets are
pushed into the RTP Depacker module which extracts the frames contained in an RTP packet. These frame are then
pushed into the JBM where the statistics are updated and the frames are stored for later decoding and playout. The
Acoustic Frontend contains the audio interface which, concurrently to the push operation of EVS frames, pulls PCM
buffers from the JBM. The JBM is therefore required to provide PCM buffers, which are normally generated by
decoding EVS frames by the EVS decoder or by other means to allow uninterrupted playout. Although the JBM is
described for a multi-threaded architecture it does not specify thread-safe data structures due to the dependency on a
particular implementation.
Note that the JBM does not directly forward frames from the RTP Depacker to the EVS decoder but instead uses frame-
based adaptation to smooth the network jitter. In addition signal-based adaptation is executed on the decoded PCM
buffers before they are pulled by the Acoustic Frontend. The corresponding algorithms are described in the following
clauses.
5 Jitter Buffer Management
5.1 Overview
Jitter Buffer Management (JBM) includes the jitter estimation, control and jitter buffer adaptation algorithm to manage
the inter-arrival jitter of the incoming packet stream. The entire solution for EVS consists of the following components:
- RTP Depacker (clause 5.2) to analyse the incoming RTP packet stream and to extract the EVS speech frames
along with meta data to estimate the network jitter
- De-jitter Buffer (clause 5.6) to store the extracted EVS speech frames before decoding and to perform frame-
based adaptation
- EVS decoder [1] for decoding the received EVS speech frames to PCM data
- Time-Scale Modification (clause 5.4.3) to perform signal-based adaptation for changing the playout delay
- Receiver Output Buffer (clause 5.5) to provide PCM data with a fixed frame size to the Acoustic Frontend
- Playout Delay Estimation Module (clause 5.3.5) to provide information on the current playout delay due to JBM
- Network Jitter Analysis (clause 5.3) for estimating the packet inter-arrival jitter and target playout delay
- Adaptation Control Logic (clause 5.4) to decide on actions for changing the playout delay based on the target
playout delay
Figure 2: Modules of the EVS Jitter Buffer Management Solution
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5.2 Depacketization of RTP packets (informative)
The RTP Depacker module of the JBM performs the depacketization of the incoming RTP packet stream. During this
operation the EVS frames, embedded in RTP packets according to the respective RTP payload format [2], [9], are
extracted and pushed to the de-jitter buffer. The RTP timestamp in an RTP packet for EVS always refers to the first
EVS frame in the RTP payload. Any further EVS frames in the RTP payload are indexed in the RTP Payload Format
Header by a Table of Contents (ToC) [2], [9]. The RTP Depacker performs the unpacking and calculates and assigns a
media timestamp to every speech frame present in each received RTP packet.
The Jitter Buffer Management (JBM) for the EVS codec depends on information that is part of the received RTP packet
stream. Each RTP packet consists of an RTP header and the RTP payload. The following data fields of the RTP header
are of relevance for the JBM:
- RTP timestamp
- RTP sequence number
The marker bit in the RTP header is not evaluated by this JBM solution. Other fields in the RTP header are needed to
correctly assign the incoming RTP packets to an RTP session, which is outside the scope of this specification.
All extracted frames (without NO_DATA frames) are fed to the JBM. The data structure for one frame consists of:
- Frame payload data, including the size of the payload
- Arrival timestamp of the RTP packet containing the frame
- Media timestamp in RTP timescale units, derived from the RTP timestamp of the packet
- Media duration in RTP timescale units (20 ms for EVS frames)
- RTP timescale as specified in the specification of the RTP payload format
- RTP sequence number
- SID flag
- Partial copy flag
To optimize the JBM behaviour for DTX, the JBM needs to be aware of SID frames. Determining this information
depends on the implementation of the underlying audio codec. To keep the JBM independent of the audio codec, the
SID flag needs to be fed to the JBM. In case of the EVS, AMR and AMR-WB codecs the SID flag can be determined
from the size of the frame payload data.
Audio encoders supporting DTX typically output NO_DATA frames between SID frames to signal that a frame was not
encoded because it does not contain an active signal and should be substituted with comfort noise by the audio decoder.
Instead of NO_DATA frames this JBM solution uses the RTP timestamp for media time calculation. Therefore the RTP
Depacker should not feed NO_DATA frames into the JBM.
The JBM handles packet reordering and duplication on the network and so the RTP Depacker can feed those frames
into the JBM exactly as received, therefore a typical RTP Depacker implementation might be state-less.
5.3 Network Jitter Analysis and Delay Estimation
5.3.1 General
Estimates of the network jitter are required to control the JBM playout delay. The Jitter Buffer Management for EVS
combines a short-term and a long-term jitter estimate to set the target playout delay. The playout is smoothly adapted to
continuously minimize the difference between playout delay and target playout delay.
The transmission delay of a packet on the network can be seen as the sum of a fixed component (consisting of
unavoidable factors such as propagation times through physical materials and minimum processing times) and a varying
component (dominated by network jitter e.g. due to scheduling). As JBM does not expect synchronized clocks between
the sender and receiver, the fixed delay value cannot be estimated using only the information available from the
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received RTP packets. Therefore JBM ignores the fixed delay component. To estimate the varying delay component,
the two basic values delay d and offset o are calculated using the arrival timestamp r and the media timestamp t .
i i i i
This is done for every received frame, where i relates to the current frame (most recently received), and i −1 relates to
the previously received frame. Note that for the first received frame of a session no delay value will be calculated ( d is
set to 0).
d = (r − r ) − (t − t ) + d (1)
i i i−1 i i−1 i−1
Delay calculations are done in millisecond units. A sliding window stores the offset o and delay values d for received
i i
frames.
o = r − t (2)
i i i
The difference of the stored maximum and minimum delay values is used as an estimate for the varying delay
component (network jitter). Both the target playout delay and the playout delay calculations are based on the minimum
stored offset, i.e. the offset calculated for the frame received with the lowest transmission delay relative to all frames
currently contained in the sliding window. The details on how this approach is used are described in the following
sections.
5.3.2 Long-term Jitter
To calculate the long term jitter j , an array of network statistics entries (FIFO queue) is used. For each frame received
i
in an RTP packet on the network an entry will be added to the array. An entry contains three values: delay d , offset o
i i
and RTP timestamp t . The time span stored in the FIFO might be different to the number of stored entries if DTX is
i
used, for that reason the window size of the FIFO is limited in two ways. It may contain at most 500 entries (equal to 10
seconds at 50 packets per second) and at most a time span (RTP timestamp difference between newest and oldest frame)
of 10 seconds. If more entries need to be stored, the oldest entry is removed. Note that in the following operations the
arrays are assumed to contain only valid entries, i.e. non-arriving packets are left out from the operations when e.g.
calculating minimum or maximum values.
The long term jitter j for the i-th received frame is calculated as the difference between the maximum delay value
i
currently stored in the array and the minimum delay value:
j = max(d ,.,d ) − min(d ,.,d )
(3)
i i−500 i i−500 i
5.3.3 Short-term jitter
The short term jitter estimation is done in two steps.
In the first step, the same jitter calculation is used as for long term estimation with the following modifications: The
window size of a first array ("Fifo1") is limited to a maximum of 50 entries and a maximum time span of one second.
The resulting temporary jitter value k is calculated as the differen
...