TETRA语音编码标准EN 300 395-2 V1.3.1：全速率信道应用

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ETSI EN 300 395-2 V1.3.1 (2005-01)，全称为“欧洲电信标准化协会（ETSI）关于陆地集群无线通信（Terrestrial Trunked Radio, TETRA）的第2部分：TETRA语音编码器标准”，是针对TETRA系统中全速率语音通道的编码技术规范。TETRA是一种专为公共安全、交通管理和应急服务设计的数字集群通信系统，它提供高效、可靠的通信能力，支持多用户同时通话和数据传输。该标准的主要关注点在于定义了一个用于TETRA网络的全速率（Full Rate）语音编码方案，全速率通常指语音通信中的12.2kbps比特率，这是当时业界常见的高质量语音编码标准。TETRA编码器的设计旨在确保在无线通信环境中，语音信号能够保持清晰、准确且易于理解，即使在嘈杂或信号较差的条件下也能维持良好的通话质量。 TETRA协议的这个部分详细阐述了编码算法、参数设置、以及编码器的实现要求。它可能包括诸如语音编码的帧结构、编码算法的选择（例如采用的MPEG-4或者G.723.1等标准）、噪声抑制和回声消除技术、以及与TETRA网络其他部分（如基带单元、多址接入和空中接口）的兼容性要求。此外，标准还引用了参考文献REN/TETRA-05059，表明它是基于或与TETRA技术发展相关的其他研究或技术文档。关键词“TETRA”和“radio codec”强调了标准的核心内容，即TETRA系统的无线通信和语音编码模块。值得注意的是，ETSI EN 300 395-2 V1.3.1（2005-01）作为一项官方标准，其版本可能会随着技术进步和需求变化而更新。用户在使用时需留意文档可能的修订或状态变更，应以PDF格式的印刷版或ETSI官方网站发布的最新版本为准，以确保获取最准确的信息。总结来说，ETSI EN 300 395-2 V1.3.1是TETRA系统的关键技术文档，对于开发TETRA终端设备、基站、网络管理系统和相关通信软件的制造商和开发者具有指导意义，确保了TETRA系统的语音通信性能和互操作性。

ETSI

TSI EN 300 395-2 V1.3.1 (2005

and

() ()

()

Fz Az z Az

11 1'

=−

−−

(14)

respectively. It can be proven that all roots of these polynomials are on the unit circle and they alternate each other

(see annex F).

()

′

has a root

z =− =1( )

and

()

′

has a root

z ==

(

)

To eliminate these two roots, new polynomials are defined:

() ()

()

Fz Fz z

′

−

(15)

and

() ()

()

Fz Fz z

′

−

/ (16)

Each polynomial has 5 conjugate roots on the unit circle

()

, therefore, the polynomials can be written as:

()

Fz qz z

13 9

12=−+

∏

−−

= , ,...,

(17)

and

()

Fz qz z

24 10

12=−+

∏

−−

= ,,...,

(18)

where

()

= cos

, with

being the Line Spectral Frequencies (LSFs). They satisfy the ordering

property

<<<< <

.. .

are referred as the LSPs in the cosine domain.

The first five coefficients of each of the symmetric polynomials

()

and

()

are found by the recursive relations

(for

i = 0 to 4):

() ()

fi a a fi

ipi

11 1

21 2

+= + −

+= − +

+−

(19)

The LSPs are found by evaluating the polynomials

()

and

()

at 60 points equally spaced between 0 and

and checking for sign changes. A sign change signifies the existence of a root and the sign change interval is then

divided 4 times to better track the root. The Chebyshev polynomials have to be used to evaluate

()

and

()

(see annex F). This method is very computationally efficient since it bypasses the cosine computations as the roots are

found directly in the cosine domain

{}

. In the TETRA codec, implementation, quantization and interpolation of the

LSPs are performed in the cosine domain, thus no trigonometric computations are needed to convert to the frequency

domain. The polynomials

()

are given by:

() () () () () () () () () () ()

()

Fz e Tx f Tx f Tx f Tx f Tx f

=+++++

−

2 123452

54 32 1

/ (20)

where

() ( )

Tx m

= cos

is the

order Chebyshev polynomial, and

()

fii, ,...=15, are the coefficients

of either

()

, computed using the equations in (19). The details of the Chebyshev polynomial

evaluation method are found in (see annex F). If this numerical process is not able to find enough roots, the

previous computed set of LSPs is used.

ETSI

TSI EN 300 395-2 V1.3.1 (2005

The quantized LP parameters are used for the fourth sub-frame, whereas the first three sub-frames use a linear

interpolation of the parameters of the present and previous frames. The interpolation is performed on the LSPs in the

domain. Let

be the quantized LSP vector at the present frame and

−

the quantized LSP vector at the past

frame. The interpolated LSP vectors at each of the 4 sub-frames are given by:

qq q

050 050

−

,75

,25

,75

(22)

The initial values of the past quantized LSP vector are given in Q15 by

−

= {30 000, 26 000, 21 000, 15 000, 8 000,

0, -8 000, -15 000, -21 000, -26 000}. (Divide by

to obtain the values in the range [-1,1]). The interpolated LSP

vectors shall be used to compute a different LP filter at each sub-frame.

4.2.2.4 Long-term prediction analysis

The aim of the long term prediction analysis or adaptive codebook search is to find the best pitch parameters, which are

the delay and gain values for the pitch filter. The pitch filter shall be implemented using the so-called adaptive

codebook approach whereby the excitation is repeated for delays less than the sub-frame length (60). In this

implementation the excitation is extended by the LP residual in the search stage to simplify the closed-loop search. In

the first sub-frame, a fractional pitch delay is used with resolutions: 1/3 in the range

19 84

− and integers

only in the range [85 - 143]. For the other sub-frames, a pitch resolution of 1/3 is always used in the range

54−−+, where

is the nearest integer to the fractional pitch lag of the first sub-frame.

To simplify the pitch analysis procedure, a two stage approach shall be used, comprising first an open loop pitch search

followed by a closed loop search.

The open loop pitch has to be computed once every speech frame (30 ms) using a weighted speech signal

()

. A

pole-zero type weighting procedure shall be used to get

()

. This procedure shall be performed with the help of a

shaping filter

()()

Az Az/, / /,095 060 for which the un-quantized LP parameters shall be used.

The open loop pitch search shall then be performed as follows. In a first step, 3 maxima of the correlation:

()( )

Csjsjk

kww

=−

∑

120

(23)

are found in the three ranges, [20 - 39], [40 - 79] and [80 - 142], respectively. The retained maxima

, ,..., ,=13 are normalized by dividing by

()

snki

13−

∑

=, ,..., , respectively. The normalized

maxima and corresponding delays are denoted by

()

Rk i

,, ,...=13. The winner among the three

normalized correlation is selected by favouring the delays in the lower ranges. That is,

is selected if

085

, . This procedure of dividing the delay range into 3 sections and favouring the lower sections

is used to avoid choosing pitch multiples.

NOTE 1: The past weighted speech samples are initialized to zero.

ETSI

TSI EN 300 395-2 V1.3.1 (2005

Having found the open-loop pitch

, a closed-loop pitch analysis has to be performed around the open-loop pitch

delay on a sub-frame basis. In the first sub-frame the range

bounded by [20 - 143] is searched. For the other

sub-frames, closed-loop pitch analysis is performed around the pitch selected in the first sub-frame. As mentioned

earlier, a pitch resolution of 1/3 is always used for the other sub-frames in the range

54−−+, where

is the integer part of the first sub-frame pitch lag. The pitch delay shall be encoded with 8 bits in the first sub-frame

while the relative delays of the other sub-frames shall be encoded with 5 bits per sub-frame.

The closed loop pitch search shall be performed by minimizing the mean-square weighted error between the original

and synthesized speech. This is achieved by maximizing the term:

() ()

xny

∑

(24)

where

()

xn is the target for the adaptive codebook search given by the weighted input speech after

subtracting the zero-input response of the weighted synthesis filter

() ()

HzWz and

()

is the past filtered

excitation at delay

k (the past excitation is initialized to zero).

NOTE 2: The search range is limited around the open-loop pitch as explained earlier.

For delays

k < 60 the excitation signal

()

un is extended by the LP residual signal. Once the optimum integer pitch

delay is determined, the fractions

−

, −

and

around that integer are tested.

NOTE 3: For the first sub-frame, the fractions are tested only if the integer pitch lag is less than 85.

The fractional pitch search is performed by interpolating the normalized correlation in equation (24) and searching for

its maximum. Once the non-integer pitch is determined, the adaptive codebook vector

()

vn is computed by

interpolating the past excitation signal

()

un . The interpolation shall be performed using two FIR filters (Hamming

windowed sinc functions); one for interpolating the term in equation (24) with the sinc truncated at

±12

(8 multiplications per fraction) and the other for interpolating the past excitation with the sinc truncated at

±48

(32 multiplications per sample). The pitch gain is then found by:

()()

xnyn

ynyn

∑

bounded by

12≤≤g

, (25)

where

() () ()

yn vn hn=∗ is the filtered adaptive codebook vector (zero-state response of

() ()

HzWz to

()

vn).

NOTE 4: Only positive pitch gains are allowed since by maximizing the term in equation (24) the negative

correlations are eliminated.

ETSI

TSI EN 300 395-2 V1.3.1 (2005

4.2.2.5 Algebraic codebook: structure and search

A 16-bit algebraic codebook shall be used in the innovative codebook search, the aim of which is to find the best

innovation and gain parameters. The innovation vector contains, at most, four non-zero pulses. The 4 pulses can assume

the amplitudes and positions given in table 2.

Table 2

Codebook parameters Positions of the pulses Codebook

bit allocation

Pulse amplitude: +1,4142 0, 2, 4, 6, 8, 10, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30,

32, 34, 36, 38, 40, 42, 44, 46, 48, 50, 52, 54, 56, 58

Pulse amplitude: -1 2, 10, 18, 26, 34, 42, 50, 58 3

Pulse amplitude: +1 4, 12, 20, 28, 36, 44, 52, (60) 3

Pulse amplitude: -1 6, 14, 22, 30, 38, 46, 54, (62) 3

Global sign flag 1

Shift flag 1

The pulses shall have fixed amplitudes of +1,4142, -1, +1 and -1, respectively. The first pulse position shall be encoded

with 5 bits while the positions of the other pulses shall be encoded with 3 bits. The positions of all pulses can be

simultaneously shifted by one, to occupy odd positions. One bit shall be used to encode this shift and a global sign bit

shall be used to invert all pulses simultaneously, giving a total of 16 bits.

NOTE 1: From table 2, it is possible to position the last two pulses outside the sub-frame which indicates that these

pulses are not present.

The codebook is searched by minimizing the mean squared error between the weighted input speech and the weighted

synthesis speech. The target signal used in the closed-loop pitch search is updated by subtracting the adaptive codebook

contribution. That is, the target for the innovation is computed using:

() () ()

xn xn gyn n

059=− =, ,... (26)

where

() () ()

yn vn hn=∗ is the filtered adaptive codebook vector, with

()

hn being the impulse response of

the weighted synthesis filter

() () ( )

HzWz Az=1/ /

As described in clause 4.1 the algebraic codebook is dynamically shaped to enhance the important frequency regions.

The used shaping matrix is a lower triangular convolution matrix consisting of the impulse response of the filter

()

in equation (4). Thus the shaping can be performed as a filtering process. To maintain the simplicity of the algebraic

codebook search, the filter

()

Fz is combined with the weighted synthesis filter

() ()

HzWz and the impulse response

()

′

of the combined filter is computed (see annex F). If

is the algebraic codeword at index k , then the algebraic

codebook is searched by maximizing the term:

()

(27)

where

H is a lower triangular Toeplitz convolution matrix with diagonal

()

′

h 0

and lower diagonals

() ( )

′′

hh159,...,

and dH=

is the backward filtered target vector and

ΦΦ

Φ=HH

The algebraic structure of the codebook allows for very fast search procedures since the innovation vector

contains

only 4 non-zero pulses. The search shall be performed in 4 nested loops, corresponding to each pulse positions, where

in each loop the contribution of a new pulse is added. The correlation in equation (27) is given by:

() () () ()

C adm dm dm dm=−+−

0123

(28)

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