This Telecommunication Standard (TS) describes the detailed mapping from input blocks of 320 speech samples in 16 bit uniform PCM format to encoded blocks of 132, 177, 253, 285, 317, 365, 397, 461 and 477 bits and from encoded blocks of 132, 177, 253, 285, 317, 365, 397, 461 and 477 bits to output blocks of 320 reconstructed speech samples. The sampling rate is 16 000 samples/s leading to a bit rate for the encoded bit stream of 6.60, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 or 23.85 kbit/s. The coding scheme for the multi-rate coding modes is the so called Algebraic Code Excited Linear Prediction Coder, hereafter referred to as ACELP. The multi-rate wideband ACELP coder is referred to as MRWB-ACELP.

This TS incorporates by dated and undated reference, provisions from other publications. These normative references are cited in the appropriate places in the text and the publications are listed hereafter. For dated references, subsequent amendments to or revisions of any of these publications apply to this TS only when incorporated in it by amendment or revision. For undated references, the latest edition of the publication referred to applies.

[1]

GSM 03.50: " Digital cellular telecommunications system (Phase 2); Transmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) system"

[2]
TS 26.201: "AMR wideband speech codec; Frame structure".

[3]
TS 26.194: "AMR wideband speech codec; Voice Activity Detection (VAD)".

[4]
TS 26.173: "AMR wideband speech codec; ANSI C code".

[5]
TS 26.174: "AMR wideband speech codec; Test sequences".

[6]
ITU T Recommendation G.711 (1988): "Coding of analogue signals by pulse code modulation Pulse code modulation (PCM) of voice frequencies".

For the purposes of this TS, the following definitions apply:
adaptive codebook:

The adaptive codebook contains excitation vectors that are adapted for every subframe. The adaptive codebook is derived from the long-term filter state. The lag value can be viewed as an index into the adaptive codebook.

algebraic codebook:
A fixed codebook where algebraic code is used to populate the excitation vectors (innovation vectors). The excitation contains a small number of nonzero pulses with predefined interlaced sets of potential positions. The amplitudes and positions of the pulses of the kth excitation codevector can be derived from its index k through a rule requiring no or minimal physical storage, in contrast with stochastic codebooks whereby the path from the index to the associated codevector involves look-up tables.

anti-sparseness processing:
An adaptive post-processing procedure applied to the fixed codebook vector in order to reduce perceptual artifacts from a sparse fixed codebook vector.

closed loop pitch analysis:
This is the adaptive codebook search, i.e., a process of estimating the pitch (lag) value from the weighted input speech and the long term filter state. In the closed loop search, the lag is searched using error minimization loop (analysis by synthesis). In the adaptive multi-rate wideband codec, closed loop pitch search is performed for every subframe.

direct form coefficients:
One of the formats for storing the short term filter parameters. In the adaptive multi-rate wideband codec, all filters which are used to modify speech samples use direct form coefficients.

fixed codebook:
The fixed codebook contains excitation vectors for speech synthesis filters. The contents of the codebook are non adaptive (i.e., fixed). In the adaptive multi-rate wideband codec, the fixed codebook is implemented using an algebraic codebook.

fractional lags:
A set of lag values having sub sample resolution. In the adaptive multi-rate wideband codec a sub sample resolution of 1/4th or 1/2nd of a sample is used.

frame:
A time interval equal to 20 ms (320 samples at an 16 kHz sampling rate).

Immittance Spectral Frequencies:
(see Immittance Spectral Pair)

Immittance Spectral Pair:
Transformation of LPC parameters. Immittance Spectral Pairs are obtained by decomposing the inverse filter transfer function A(z) to a set of two transfer functions, one having even symmetry and the other having odd symmetry. The Immittance Spectral Pairs (also called as Immittance Spectral Frequencies) are the roots of these polynomials on the z-unit circle.

integer lags:
A set of lag values having whole sample resolution.

interpolating filter:
An FIR filter used to produce an estimate of sub-sample resolution samples, given an input sampled with integer sample resolution. In this implementation, the interpolating filter has low pass filter characteristics. Thus the adaptive codebook consists of the low-pass filtered interpolated past excitation.

inverse filter:
This filter removes the short term correlation from the speech signal. The filter models an inverse frequency response of the vocal tract.

lag:
The long term filter delay. This is typically the true pitch period, or its multiple or sub multiple.

LP analysis window:
For each frame, the short term filter coefficients are computed using the high pass filtered speech samples within the analysis window. In the adaptive multi-rate wideband codec, the length of the analysis window is always 384 samples. For all the modes, a single asymmetric window is used to generate a single set of LP coefficients. The 5 ms look-ahead is used in the analysis.

LP coefficients:
Linear Prediction (LP) coefficients (also referred as Linear Predictive Coding (LPC) coefficients) is a generic descriptive term for the short term filter coefficients.

mode:
When used alone, refers to the source codec mode, i.e., to one of the source codecs employed in the AMR-WB codec.

open loop pitch search:
A process of estimating the near optimal lag directly from the weighted speech input. This is done to simplify the pitch analysis and confine the closed loop pitch search to a small number of lags around the open loop estimated lags. In the adaptive multi-rate wideband codec, an open loop pitch search is performed in every other subframe.

residual:
The output signal resulting from an inverse filtering operation.

short term synthesis filter:
This filter introduces, into the excitation signal, short term correlation which models the impulse response of the vocal tract.

perceptual weighting filter:
This filter is employed in the analysis by synthesis search of the codebooks. The filter exploits the noise masking properties of the formants (vocal tract resonances) by weighting the error less in regions near the formant frequencies and more in regions away from them.

subframe:
A time interval equal to 5 ms (80 samples at 16 kHz sampling rate).

vector quantization:
A method of grouping several parameters into a vector and quantizing them simultaneously.

zero input response:
The output of a filter due to past inputs, i.e. due to the present state of the filter, given that an input of zeros is applied.

zero state response:
The output of a filter due to the present input, given that no past inputs have been applied, i.e., given that the state information in the filter is all zeroes.