The transcoding procedure specified in the present document is applicable for the full rate Traffic Channel (TCH) in the digital cellular telecommunications system. The use of this transcoding scheme for other applications has not been considered.
In GSM 06.01, a reference configuration for the speech transmission chain of the digital cellular telecommunications system is shown. According to this reference configuration, the speech encoder takes its input as a 13 bit uniform PCM signal either from the audio part of the mobile station or on the network side, from the PSTN via an 8 bit/A or μ-law (PCS 1900) to 13 bit uniform PCM conversion. The encoded speech at the output of the speech encoder is delivered to a channel encoder unit which is specified in GSM 05.03. In the receive direction, the inverse operations take place.
The present document describes the detailed mapping between input blocks of 160 speech samples in 13 bit uniform PCM format to encoded blocks of 260 bits and from encoded blocks of 260 bits to output blocks of 160 reconstructed speech samples. The sampling rate is 8000 sample/s leading to an average bit rate for the encoded bit stream of 13 kbit/s. The coding scheme is the so called Regular Pulse Excitation Long Term prediction Linear Predictive Coder, here after referred to as RPE LTP.
The present document also specifies the conversion between A and μ-law (PCS 1900) PCM and 13 bit uniform PCM. Performance requirements for the audio input and output parts are included only to the extent that they affect the transcoder performance. The present document also describes the codec down to the bit level, thus enabling the verification of compliance to the present document to a high degree of confidence by use of a set of digital test sequences. These test sequences are described and are contained in archive en_300961v080101p0.ZIP which accompanies the present document.
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.
The present document is structured as follows:
Subclause 1.3 contains a functional description of the audio parts including the A/D and D/A functions. Subclause 1.4 describes the conversion between 13 bit uniform and 8 bit A law samples. Subclauses 1.5 and 1.6 present a simplified description of the principles of the RPE LTP encoding and decoding process respectively. In clause 1.7, the sequence and subjective importance of encoded parameters are given.
Clause 2 deals with the transmission characteristics of the audio parts that are relevant for the performance of the RPE LTP codec.
Some transmission characteristics of the RPE LTP codec are also specified in clause 2. Clause 3 presents the functional description of the RPE LTP coding and decoding procedures, whereas clause 4 describes the computational details of the algorithm. Procedures for the verification of the correct functioning of the RPE LTP are described in clause 5.
Performance and network aspects of the RPE LTP codec are contained in Annex A.
The analogue to digital and digital to analogue conversion will in principle comprise the following elements:
Analogue to uniform digital:
microphone;
input level adjustment device;
input anti aliasing filter;
sample hold device sampling at 8 kHz;
analogue to uniform digital conversion to 13 bits representation.
The uniform format shall be represented in two's complement.
Uniform digital to analogue:
conversion from 13 bit /8 kHz uniform PCM to analogue;
a hold device;
reconstruction filter including x/sin x correction;
output level adjustment device;
earphone or loudspeaker.
In the terminal equipment, the A/D function may be achieved either:
by direct conversion to 13 bit uniform PCM format;
or by conversion to 8 bit/A or μ-law (PCS 1900) companded format, based on a standard A or μ-law (PCS 1900) codec/filter according to ITU-T Recommendation G.711/714, followed by the 8 bit to 13 bit conversion according to the procedure specified in clause 1.4.
For the D/A operation, the inverse operations take place.
In the latter case it should be noted that the specifications in ITU-T recommendation G.714 (superseded by G.712) are concerned with PCM equipment located in the central parts of the network. When used in the terminal equipment, the present document does not on its own ensure sufficient out of band attenuation.
The specification of out of band signals is defined in clause 2 between the acoustic signal and the digital interface to take into account that the filtering in the terminal can be achieved both by electronic and acoustical design.
The conversion between 8 bit A or μ-law (PCS 1900) companded format and the 13 bit uniform format shall be as defined in ITU-T Recommendation G.721 (superseded by G.726), clause 4.2.1, sub block EXPAND and clause 4.2.7, sub block COMPRESS. The parameter LAW = 1 should be used for A-law and LAW=0 should be used for μ-law (PCS 1900).
A simplified block diagram of the RPE LTP encoder is shown in Figure 1.1. In this diagram the coding and quantization functions are not shown explicitly.
The input speech frame, consisting of 160 signal samples (uniform 13 bit PCM samples), is first pre processed to produce an offset free signal, which is then subjected to a first order pre emphasis filter. The 160 samples obtained are then analysed to determine the coefficients for the short term analysis filter (LPC analysis). These parameters are then used for the filtering of the same 160 samples. The result is 160 samples of the short term residual signal. The filter parameters, termed reflection coefficients, are transformed to log.area ratios, LARs, before transmission.
For the following operations, the speech frame is divided into 4 sub frames with 40 samples of the short term residual signal in each. Each sub frame is processed blockwise by the subsequent functional elements.
Before the processing of each sub block of 40 short term residual samples, the parameters of the long term analysis filter, the LTP lag and the LTP gain, are estimated and updated in the LTP analysis block, on the basis of the current sub block of the present and a stored sequence of the 120 previous reconstructed short term residual samples.
A block of 40 long term residual signal samples is obtained by subtracting 40 estimates of the short term residual signal from the short term residual signal itself. The resulting block of 40 long term residual samples is fed to the Regular Pulse Excitation analysis which performs the basic compression function of the algorithm.
As a result of the RPE analysis, the block of 40 input long term residual samples are represented by one of 4 candidate sub sequences of 13 pulses each. The subsequence selected is identified by the RPE grid position (M). The 13 RPE pulses are encoded using Adaptive Pulse Code Modulation (APCM) with estimation of the sub block amplitude which is transmitted to the decoder as side information.
The RPE parameters are also fed to a local RPE decoding and reconstruction module which produces a block of 40 samples of the quantized version of the long term residual signal.
By adding these 40 quantized samples of the long term residual to the previous block of short term residual signal estimates, a reconstructed version of the current short term residual signal is obtained.
The block of reconstructed short term residual signal samples is then fed to the long term analysis filter which produces the new block of 40 short term residual signal estimates to be used for the next sub block thereby completing the feedback loop.
The simplified block diagram of the RPE LTP decoder is shown in Figure 1.2. The decoder includes the same structure as the feed back loop of the encoder. In error free transmission, the output of this stage will be the reconstructed short term residual samples. These samples are then applied to the short term synthesis filter followed by the de emphasis filter resulting in the reconstructed speech signal samples.
As indicated in Figure 1.1 the three different groups of data are produced by the encoder are:
the short term filter parameters;
the Long Term Prediction (LTP) parameters;
the RPE parameters.
The encoder will produce this information in a unique sequence and format, and the decoder shall receive the same information in the same way. In Table 1.1, the sequence of output bits b1 to b260 and the bit allocation for each parameter is shown.
The different parameters of the encoded speech and their individual bits have unequal importance with respect to subjective quality. Before being submitted to the channel encoding function the bits have to be rearranged in the sequence of importance as given in GSM 05.03. The ranking has been determined by subjective testing and the procedure used is described in Annex A, clause A.2.
This clause specifies the necessary performance characteristics of the audio parts for proper functioning of the speech transcoder. Some transmission performance characteristics of the RPE LTP transcoder are also given to assist the designer of the speech transcoder function. The information given here is redundant and the detailed specifications are contained in recommendation GSM 11.10.
The performance characteristics are referred to the 13 bit uniform PCM interface.
Concerning 1) discrimination against out of band signals (sending) and 2) spurious out of band signals (receiving), the same requirements as defined in ETSI standard TE 04 15 (digital telephone, candidate NET33) apply.
Consider a back to back configuration where the parameters generated by the encoder are delivered to the speech decoder as soon as they are available.
The transcoder delay is defined as the time interval between the instant a speech frame of 160 samples has been received at the encoder input and the instant the corresponding 160 reconstructed speech samples have been out put by the speech decoder at an 8 kHz sample rate.
The theoretical minimum delay which can be achieved is 20 ms. The requirement is that the transcoder delay should be less than 30 ms.